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OpenCloudOS-Kernel/sound/soc/codecs/twl4030.c

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ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
/*
* ALSA SoC TWL4030 codec driver
*
* Author: Steve Sakoman, <steve@sakoman.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* version 2 as published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
* 02110-1301 USA
*
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
mfd: Rename twl4030* driver files to enable re-use The upcoming TWL6030 is companion chip for OMAP4 like the current TWL4030 for OMAP3. The common modules like RTC, Regulator creates opportunity to re-use the most of the code from twl4030. This patch renames few common drivers twl4030* files to twl* to enable the code re-use. Signed-off-by: Rajendra Nayak <rnayak@ti.com> Signed-off-by: Balaji T K <balajitk@ti.com> Signed-off-by: Santosh Shilimkar <santosh.shilimkar@ti.com> Acked-by: Kevin Hilman <khilman@deeprootsystems.com> Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
2009-12-14 03:05:51 +08:00
#include <linux/i2c/twl.h>
include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-24 16:04:11 +08:00
#include <linux/slab.h>
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>
ASoC: TWL4030: Change the Master volume control to TLV TWL4030 FGAIN volume control has a range: -62 to 0 dB in 1 dB steps, 0 in the FGAIN means mute. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-24 19:49:35 +08:00
#include <sound/tlv.h>
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
/* Register descriptions are here */
#include <linux/mfd/twl4030-codec.h>
/* Shadow register used by the audio driver */
#define TWL4030_REG_SW_SHADOW 0x4A
#define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1)
/* TWL4030_REG_SW_SHADOW (0x4A) Fields */
#define TWL4030_HFL_EN 0x01
#define TWL4030_HFR_EN 0x02
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
/*
* twl4030 register cache & default register settings
*/
static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
0x00, /* this register not used */
ASoC: TWL4030: Revisit codec defaults Reset most of the codec registers to their chip reset value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:14 +08:00
0x00, /* REG_CODEC_MODE (0x1) */
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
0x00, /* REG_OPTION (0x2) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_UNKNOWN (0x3) */
0x00, /* REG_MICBIAS_CTL (0x4) */
ASoC: TWL4030: Make offset cancellation path configurable Add means for machine drivers to select the path for offset cancellation. Reset the reg cache value to the chip reset value at the same time. Machine drivers can specify which path need to be used for offset cancellation via the twl4030_setup.offset_cncl_path. For paths use the defines from include/linux/mfd/twl4030-codec.h: TWL4030_OFFSET_CNCL_SEL_* Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:16 +08:00
0x00, /* REG_ANAMICL (0x5) */
ASoC: TWL4030: Add input selection and gain controls The TWL4030 codec device has two ADCs. Both of them can have several inputs routed to them, but TRM says that only one source can be selected for every ADC, even though every source has a dedicated bit in the registers. This patch adds input source controls. It modifies default register values to have no inputs selected and ADCs disabled. When some input is selected, control handlers enable apropriate input amplifier and ADC. If a microphone is selected, bias power is automatically enabled. When some input is deselected, unused chip parts are disabled. Microphone and line input recording tested on OMAP3 pandora board. Signed-off-by: Grazvydas Ignotas <notasas@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-03 02:48:58 +08:00
0x00, /* REG_ANAMICR (0x6) */
0x00, /* REG_AVADC_CTL (0x7) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_ADCMICSEL (0x8) */
0x00, /* REG_DIGMIXING (0x9) */
ASoC: TWL4030: Revisit codec defaults Reset most of the codec registers to their chip reset value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:14 +08:00
0x0f, /* REG_ATXL1PGA (0xA) */
0x0f, /* REG_ATXR1PGA (0xB) */
0x0f, /* REG_AVTXL2PGA (0xC) */
0x0f, /* REG_AVTXR2PGA (0xD) */
ASoC: TWL4030: Add supply for audio serial interface control The serial interface (TDM/I2S) for the audio block have been constantly enabled. Introduce a new DAPM_SUPPLY for handling the AIF_EN bit, so the interface is only enabled, when there is a need for it. For example when only the analog loopback is enabled, there is no need to keep the serial interface active. I have added the persons who contributed to the Voice path of twl4030 codec driver, so they might have the ability to test this patch, and send an update for the Voice path, if it is necessary Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-09 21:24:04 +08:00
0x00, /* REG_AUDIO_IF (0xE) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_VOICE_IF (0xF) */
ASoC: TWL4030: Revisit codec defaults Reset most of the codec registers to their chip reset value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:14 +08:00
0x3f, /* REG_ARXR1PGA (0x10) */
0x3f, /* REG_ARXL1PGA (0x11) */
0x3f, /* REG_ARXR2PGA (0x12) */
0x3f, /* REG_ARXL2PGA (0x13) */
0x25, /* REG_VRXPGA (0x14) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_VSTPGA (0x15) */
0x00, /* REG_VRX2ARXPGA (0x16) */
ASoC: TWL4030: Modify codec default settings Change the legacy default register configuration, which left some internal components on. Now we have either DAPM, or other ways to control these bits, so there is no need to enable them by default. The affected parts: Disable ADCL and ADCR Disable ARXL2 and ARXR2 analog PGA (playback) Disable APLL by default Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-28 21:57:04 +08:00
0x00, /* REG_AVDAC_CTL (0x17) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_ARX2VTXPGA (0x18) */
ASoC: TWL4030: Revisit codec defaults Reset most of the codec registers to their chip reset value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:14 +08:00
0x32, /* REG_ARXL1_APGA_CTL (0x19) */
0x32, /* REG_ARXR1_APGA_CTL (0x1A) */
0x32, /* REG_ARXL2_APGA_CTL (0x1B) */
0x32, /* REG_ARXR2_APGA_CTL (0x1C) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_ATX2ARXPGA (0x1D) */
0x00, /* REG_BT_IF (0x1E) */
ASoC: TWL4030: Revisit codec defaults Reset most of the codec registers to their chip reset value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:14 +08:00
0x55, /* REG_BTPGA (0x1F) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_BTSTPGA (0x20) */
0x00, /* REG_EAR_CTL (0x21) */
ASoC: TWL4030: Use codec defaults for Headset initial configuration Disable the amplifiers for the headset outputs, and do not select routings by default to the headset outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-17 15:49:54 +08:00
0x00, /* REG_HS_SEL (0x22) */
0x00, /* REG_HS_GAIN_SET (0x23) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_HS_POPN_SET (0x24) */
0x00, /* REG_PREDL_CTL (0x25) */
0x00, /* REG_PREDR_CTL (0x26) */
0x00, /* REG_PRECKL_CTL (0x27) */
0x00, /* REG_PRECKR_CTL (0x28) */
0x00, /* REG_HFL_CTL (0x29) */
0x00, /* REG_HFR_CTL (0x2A) */
ASoC: TWL4030: Revisit codec defaults Reset most of the codec registers to their chip reset value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:14 +08:00
0x05, /* REG_ALC_CTL (0x2B) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_ALC_SET1 (0x2C) */
0x00, /* REG_ALC_SET2 (0x2D) */
0x00, /* REG_BOOST_CTL (0x2E) */
ASoC: TWL4030: Disable soft-volume Keep Soft-volume disabled for now, since if it is enabled the FGAIN volume controls are not working in the current configuration: CODEC_MODE:OPT_MODE = 1 OPTION:ARXR2_EN = 1 OPTION:ARXL2_EN = 1 OPTION:ARXR1_EN = 0 OPTION:ARXL1_VRX_EN = 0 RX_PATH_SEL:RXL1_SEL = 0x0 (or 0x1) RX_PATH_SEL:RXR1_SEL = 0x0 (or 0x1) After the patch, FGAIN volume control works. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-24 14:25:45 +08:00
0x00, /* REG_SOFTVOL_CTL (0x2F) */
ASoC: TWL4030: Revisit codec defaults Reset most of the codec registers to their chip reset value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:14 +08:00
0x13, /* REG_DTMF_FREQSEL (0x30) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_DTMF_TONEXT1H (0x31) */
0x00, /* REG_DTMF_TONEXT1L (0x32) */
0x00, /* REG_DTMF_TONEXT2H (0x33) */
0x00, /* REG_DTMF_TONEXT2L (0x34) */
ASoC: TWL4030: Revisit codec defaults Reset most of the codec registers to their chip reset value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:14 +08:00
0x79, /* REG_DTMF_TONOFF (0x35) */
0x11, /* REG_DTMF_WANONOFF (0x36) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */
0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */
0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */
ASoC: TWL4030: Modify codec default settings Change the legacy default register configuration, which left some internal components on. Now we have either DAPM, or other ways to control these bits, so there is no need to enable them by default. The affected parts: Disable ADCL and ADCR Disable ARXL2 and ARXR2 analog PGA (playback) Disable APLL by default Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-28 21:57:04 +08:00
0x06, /* REG_APLL_CTL (0x3A) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_DTMF_CTL (0x3B) */
ASoC: TWL4030: Revisit codec defaults Reset most of the codec registers to their chip reset value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:14 +08:00
0x44, /* REG_DTMF_PGA_CTL2 (0x3C) */
0x69, /* REG_DTMF_PGA_CTL1 (0x3D) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_MISC_SET_1 (0x3E) */
0x00, /* REG_PCMBTMUX (0x3F) */
0x00, /* not used (0x40) */
0x00, /* not used (0x41) */
0x00, /* not used (0x42) */
0x00, /* REG_RX_PATH_SEL (0x43) */
ASoC: TWL4030: Revisit codec defaults Reset most of the codec registers to their chip reset value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:14 +08:00
0x32, /* REG_VDL_APGA_CTL (0x44) */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
0x00, /* REG_VIBRA_CTL (0x45) */
0x00, /* REG_VIBRA_SET (0x46) */
0x00, /* REG_VIBRA_PWM_SET (0x47) */
0x00, /* REG_ANAMIC_GAIN (0x48) */
0x00, /* REG_MISC_SET_2 (0x49) */
ASoC: TWL4030: Add shadow register Shadow, non HW register for dealing with the HandsfreeL/R muting. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:12 +08:00
0x00, /* REG_SW_SHADOW (0x4A) - Shadow, non HW register */
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
};
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
/* codec private data */
struct twl4030_priv {
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
struct snd_soc_codec codec;
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
unsigned int codec_powered;
ASoC: TWL4030: AIF/APLL fix in DAPM domain This patch orders the APLL and AIF power sequence in case of HiFi (audio in TWL4030 terms) playback/capture. We also need to make sure that the AIF is running during playback/capture, when there is no valid DAPM route available. For this purpose I introduce these virtual widgets: /* To have complete playback route all the time */ DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */ /* To have complete capture route all the time */ DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */ /* To have complete playback route for the voice module */ DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */ The DAPM_SUPPLY widgets for APLL and AIF are placed in a way, that during any audio activity the needed configuration of AIF and APLL will be enabled (playback, capture, analog loopback, digital loopback, and voice activity). The apll reference counting code has been lifted, and modified from Liam Girdwood's earlier patch. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-29 15:58:08 +08:00
/* reference counts of AIF/APLL users */
ASoC: TWL4030: Change codec_muted to apll_enabled codec_muted is missleading, change it to apll_enabled, which is what it is doing: enabing and disabling the APLL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-28 16:57:05 +08:00
unsigned int apll_enabled;
ASoC: TWL4030: Add constrains for second stream In case of duplex mode (capture and playback at the same time), the second stream has to have the same parameters (rate, sample size) as the already running stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-27 16:39:08 +08:00
struct snd_pcm_substream *master_substream;
struct snd_pcm_substream *slave_substream;
ASoC: TWL4030: Fix for the constraint handling The original implementation of the constraints were good against sane applications. If the opening sequence is: stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the constraints are set correctly for stream2. But if the sequence is: stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2 would receive constraint rate = 0, sample_bits = 0, since the stream1 has not yet called hw_params... The command to trigger this event: gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false This patch does some 'black magic' in order to always set the correct constraints and sets it only when it is needed for the other stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-17 20:55:08 +08:00
unsigned int configured;
unsigned int rate;
unsigned int sample_bits;
unsigned int channels;
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
unsigned int sysclk;
ASoC: TWL4030: PM fix for output amplifiers Gain controls on outputs affect the power consumption when the gain is set to non 0 value. Outputs with amps have one register to configure the routing and the gain: PREDL_CTL (0x25): bit 0: Voice enable bit 1: Audio L1 enable bit 2: Audio L2 enable bit 3: Audio R2 enable bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB) bit 0 - 3: is handled in DAPM domain (DAPM_MIXER) bit 4 - 5: has simple volume control If there is no audio activity (BIAS_STANDBY), and user changes the volume, than the output amplifier will be enabled. If the user changes the routing (but the codec remains in BIAS_STANDBY), than the cached gain value also be written to the register, which enables the amplifier. The existing workaround for this is to have virtual PGAs associated with the outputs, and whit DAPM PMD the gain on the output will be forced to 0 (off) by bypassing the regcache. This failed to disable the amplifiers in several scenario (as mentioned above). Also if the codec is in BIAS_ON state, and user modifies a volume control, which path is actually not enabled, than that amplifier will be enabled as well, but it will be not turned off, since there is no DAPM path, which would make mute it. To prevent amps being enabled, when they are not needed, introduce the following workaround: Track the state of each of this type of output. In twl4030_write only allow actual write, when the given output is enabled, otherwise only update the reg_cache. The PGA event handlers on power up will write the cached value to the chip (restoring gain, routing selection). On power down 0 is written to the register (disabling the amp, and also just in case clearing the routing). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-22 23:46:37 +08:00
/* Output (with associated amp) states */
u8 hsl_enabled, hsr_enabled;
u8 earpiece_enabled;
u8 predrivel_enabled, predriver_enabled;
u8 carkitl_enabled, carkitr_enabled;
ASoC: TWL4030: Add configurable delay after digimic enable When digital microphones are connected to twl, delay is needed after enabling the digimic interface of the codec. Add new parameter for the setup data, which can be used to pass the apropriate delay in ms after the digimic interface has been enabled. Without certain delay (in certain HW configuration) the beggining of the recorded sample contains a glitch, which is generated by the digital microphones. Delaying the micbias1, 2 (which is the bias for the digimic0 or 1) does not help, since the glitch is coming after switching the digimic interface. Reversing the micbias and digimic enable order does not work either (in that case the wait need to be added after the micbias enabled). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-20 20:49:09 +08:00
/* Delay needed after enabling the digimic interface */
unsigned int digimic_delay;
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
};
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
/*
* read twl4030 register cache
*/
static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec,
unsigned int reg)
{
ASoC: Fix build error in twl4030.c Fix the (likely cut-n-paste) error by commit 16a30fbb0d3aa4ee829a2dd3d0e314e2b5ae96a9, which causes the error below: sound/soc/codecs/twl4030.c: In function 'twl4030_read_reg_cache': sound/soc/codecs/twl4030.c:152: error: 'cache' undeclared (first use in this function) Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-04 15:58:18 +08:00
u8 *cache = codec->reg_cache;
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
ASoC: fixes to caching implementations This patch takes fixes a number of bugs in the caching code used by several ASoC codec drivers. Mostly off-by-one fixes. Signed-off-by: Ian Molton <ian@mnementh.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-18 01:44:23 +08:00
if (reg >= TWL4030_CACHEREGNUM)
return -EIO;
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
return cache[reg];
}
/*
* write twl4030 register cache
*/
static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec,
u8 reg, u8 value)
{
u8 *cache = codec->reg_cache;
if (reg >= TWL4030_CACHEREGNUM)
return;
cache[reg] = value;
}
/*
* write to the twl4030 register space
*/
static int twl4030_write(struct snd_soc_codec *codec,
unsigned int reg, unsigned int value)
{
ASoC: Add indirection for CODEC private data One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-14 14:35:19 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: TWL4030: PM fix for output amplifiers Gain controls on outputs affect the power consumption when the gain is set to non 0 value. Outputs with amps have one register to configure the routing and the gain: PREDL_CTL (0x25): bit 0: Voice enable bit 1: Audio L1 enable bit 2: Audio L2 enable bit 3: Audio R2 enable bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB) bit 0 - 3: is handled in DAPM domain (DAPM_MIXER) bit 4 - 5: has simple volume control If there is no audio activity (BIAS_STANDBY), and user changes the volume, than the output amplifier will be enabled. If the user changes the routing (but the codec remains in BIAS_STANDBY), than the cached gain value also be written to the register, which enables the amplifier. The existing workaround for this is to have virtual PGAs associated with the outputs, and whit DAPM PMD the gain on the output will be forced to 0 (off) by bypassing the regcache. This failed to disable the amplifiers in several scenario (as mentioned above). Also if the codec is in BIAS_ON state, and user modifies a volume control, which path is actually not enabled, than that amplifier will be enabled as well, but it will be not turned off, since there is no DAPM path, which would make mute it. To prevent amps being enabled, when they are not needed, introduce the following workaround: Track the state of each of this type of output. In twl4030_write only allow actual write, when the given output is enabled, otherwise only update the reg_cache. The PGA event handlers on power up will write the cached value to the chip (restoring gain, routing selection). On power down 0 is written to the register (disabling the amp, and also just in case clearing the routing). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-22 23:46:37 +08:00
int write_to_reg = 0;
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
twl4030_write_reg_cache(codec, reg, value);
ASoC: TWL4030: PM fix for output amplifiers Gain controls on outputs affect the power consumption when the gain is set to non 0 value. Outputs with amps have one register to configure the routing and the gain: PREDL_CTL (0x25): bit 0: Voice enable bit 1: Audio L1 enable bit 2: Audio L2 enable bit 3: Audio R2 enable bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB) bit 0 - 3: is handled in DAPM domain (DAPM_MIXER) bit 4 - 5: has simple volume control If there is no audio activity (BIAS_STANDBY), and user changes the volume, than the output amplifier will be enabled. If the user changes the routing (but the codec remains in BIAS_STANDBY), than the cached gain value also be written to the register, which enables the amplifier. The existing workaround for this is to have virtual PGAs associated with the outputs, and whit DAPM PMD the gain on the output will be forced to 0 (off) by bypassing the regcache. This failed to disable the amplifiers in several scenario (as mentioned above). Also if the codec is in BIAS_ON state, and user modifies a volume control, which path is actually not enabled, than that amplifier will be enabled as well, but it will be not turned off, since there is no DAPM path, which would make mute it. To prevent amps being enabled, when they are not needed, introduce the following workaround: Track the state of each of this type of output. In twl4030_write only allow actual write, when the given output is enabled, otherwise only update the reg_cache. The PGA event handlers on power up will write the cached value to the chip (restoring gain, routing selection). On power down 0 is written to the register (disabling the amp, and also just in case clearing the routing). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-22 23:46:37 +08:00
if (likely(reg < TWL4030_REG_SW_SHADOW)) {
/* Decide if the given register can be written */
switch (reg) {
case TWL4030_REG_EAR_CTL:
if (twl4030->earpiece_enabled)
write_to_reg = 1;
break;
case TWL4030_REG_PREDL_CTL:
if (twl4030->predrivel_enabled)
write_to_reg = 1;
break;
case TWL4030_REG_PREDR_CTL:
if (twl4030->predriver_enabled)
write_to_reg = 1;
break;
case TWL4030_REG_PRECKL_CTL:
if (twl4030->carkitl_enabled)
write_to_reg = 1;
break;
case TWL4030_REG_PRECKR_CTL:
if (twl4030->carkitr_enabled)
write_to_reg = 1;
break;
case TWL4030_REG_HS_GAIN_SET:
if (twl4030->hsl_enabled || twl4030->hsr_enabled)
write_to_reg = 1;
break;
default:
/* All other register can be written */
write_to_reg = 1;
break;
}
if (write_to_reg)
return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
value, reg);
}
return 0;
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
}
ASoC: TWL4030: Code clean up for codec power up and down Merge the codec up and down functions to a simple one. Codec is powered down by default (reg_cache change). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:40 +08:00
static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
{
ASoC: Add indirection for CODEC private data One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-14 14:35:19 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
int mode;
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
if (enable == twl4030->codec_powered)
return;
ASoC: TWL4030: Code clean up for codec power up and down Merge the codec up and down functions to a simple one. Codec is powered down by default (reg_cache change). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:40 +08:00
if (enable)
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
mode = twl4030_codec_enable_resource(TWL4030_CODEC_RES_POWER);
ASoC: TWL4030: Code clean up for codec power up and down Merge the codec up and down functions to a simple one. Codec is powered down by default (reg_cache change). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:40 +08:00
else
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
mode = twl4030_codec_disable_resource(TWL4030_CODEC_RES_POWER);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
if (mode >= 0) {
twl4030_write_reg_cache(codec, TWL4030_REG_CODEC_MODE, mode);
twl4030->codec_powered = enable;
}
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
/* REVISIT: this delay is present in TI sample drivers */
/* but there seems to be no TRM requirement for it */
udelay(10);
}
ASoC: TWL4030: Helper to check chip default registers Since the twl4030 codec driver supports different version of the PM chip, a helper function can come handy, which can check the driver's default versus the chip values. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:18 +08:00
static inline void twl4030_check_defaults(struct snd_soc_codec *codec)
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
{
ASoC: TWL4030: Helper to check chip default registers Since the twl4030 codec driver supports different version of the PM chip, a helper function can come handy, which can check the driver's default versus the chip values. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:18 +08:00
int i, difference = 0;
u8 val;
dev_dbg(codec->dev, "Checking TWL audio default configuration\n");
for (i = 1; i <= TWL4030_REG_MISC_SET_2; i++) {
twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &val, i);
if (val != twl4030_reg[i]) {
difference++;
dev_dbg(codec->dev,
"Reg 0x%02x: chip: 0x%02x driver: 0x%02x\n",
i, val, twl4030_reg[i]);
}
}
dev_dbg(codec->dev, "Found %d non maching registers. %s\n",
difference, difference ? "Not OK" : "OK");
}
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
ASoC: TWL4030: Add functionalty to reset the registers Machine driver can instruct the codec driver to reset the chip registers to their default values at probe time. If machine driver does not provide setup data, then the registers are going to be reseted to their defaults, to be safe. If the developer on the platform confirms that the register reset is not needed, than it can be skipped, saving ~20ms time in probe. As safety measure do the register reset at remove time also. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:21 +08:00
static inline void twl4030_reset_registers(struct snd_soc_codec *codec)
{
int i;
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
/* set all audio section registers to reasonable defaults */
for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
ASoC: TWL4030: Do not modify the APLL_CTL register APLL_CTL register is configured by the twl4030-codec MFD driver. Remove code, which makes changes in the APLL_CTL register, and replace those with checks against the configured audio_mclk configuration done in the MFD driver. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-11-04 15:58:20 +08:00
if (i != TWL4030_REG_APLL_CTL)
ASoC: TWL4030: Add functionalty to reset the registers Machine driver can instruct the codec driver to reset the chip registers to their default values at probe time. If machine driver does not provide setup data, then the registers are going to be reseted to their defaults, to be safe. If the developer on the platform confirms that the register reset is not needed, than it can be skipped, saving ~20ms time in probe. As safety measure do the register reset at remove time also. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:21 +08:00
twl4030_write(codec, i, twl4030_reg[i]);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static void twl4030_init_chip(struct snd_soc_codec *codec)
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
struct twl4030_codec_audio_data *pdata = dev_get_platdata(codec->dev);
ASoC: Add indirection for CODEC private data One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-14 14:35:19 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
u8 reg, byte;
int i = 0;
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
ASoC: TWL4030: Helper to check chip default registers Since the twl4030 codec driver supports different version of the PM chip, a helper function can come handy, which can check the driver's default versus the chip values. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:18 +08:00
/* Check defaults, if instructed before anything else */
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
if (pdata && pdata->check_defaults)
ASoC: TWL4030: Helper to check chip default registers Since the twl4030 codec driver supports different version of the PM chip, a helper function can come handy, which can check the driver's default versus the chip values. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:18 +08:00
twl4030_check_defaults(codec);
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
ASoC: TWL4030: Add functionalty to reset the registers Machine driver can instruct the codec driver to reset the chip registers to their default values at probe time. If machine driver does not provide setup data, then the registers are going to be reseted to their defaults, to be safe. If the developer on the platform confirms that the register reset is not needed, than it can be skipped, saving ~20ms time in probe. As safety measure do the register reset at remove time also. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:21 +08:00
/* Reset registers, if no setup data or if instructed to do so */
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
if (!pdata || (pdata && pdata->reset_registers))
ASoC: TWL4030: Add functionalty to reset the registers Machine driver can instruct the codec driver to reset the chip registers to their default values at probe time. If machine driver does not provide setup data, then the registers are going to be reseted to their defaults, to be safe. If the developer on the platform confirms that the register reset is not needed, than it can be skipped, saving ~20ms time in probe. As safety measure do the register reset at remove time also. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:21 +08:00
twl4030_reset_registers(codec);
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
/* Refresh APLL_CTL register from HW */
ASoC: TWL4030: Helper to check chip default registers Since the twl4030 codec driver supports different version of the PM chip, a helper function can come handy, which can check the driver's default versus the chip values. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:18 +08:00
twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
TWL4030_REG_APLL_CTL);
twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, byte);
ASoC: TWL4030: Move the twl4030_power_up and _power_down function Move the twl4030_power_up and twl4030_power_down function earlier to facilitate the analog bypass implementation. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:43 +08:00
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
/* anti-pop when changing analog gain */
reg = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
twl4030_write(codec, TWL4030_REG_MISC_SET_1,
reg | TWL4030_SMOOTH_ANAVOL_EN);
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
twl4030_write(codec, TWL4030_REG_OPTION,
TWL4030_ATXL1_EN | TWL4030_ATXR1_EN |
TWL4030_ARXL2_EN | TWL4030_ARXR2_EN);
ASoC: TWL4030: Move the twl4030_power_up and _power_down function Move the twl4030_power_up and twl4030_power_down function earlier to facilitate the analog bypass implementation. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:43 +08:00
ASoC: TWL4030: Correct the ARXR2_APGA_CTL chip default It seams at least on twl5031 that the ARXR2_APGA_CTL register does not have the same default value as it is written in the TRM. Since the codec part of the PM chip has not been actually changed according to TI, assuming, that all version has the same problem, so writing there the TRM value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:19 +08:00
/* REG_ARXR2_APGA_CTL reset according to the TRM: 0dB, DA_EN */
twl4030_write(codec, TWL4030_REG_ARXR2_APGA_CTL, 0x32);
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
/* Machine dependent setup */
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
if (!pdata)
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
return;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
twl4030->digimic_delay = pdata->digimic_delay;
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
reg = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
reg &= ~TWL4030_RAMP_DELAY;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
reg |= (pdata->ramp_delay_value << 2);
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, reg);
ASoC: TWL4030: Move the twl4030_power_up and _power_down function Move the twl4030_power_up and twl4030_power_down function earlier to facilitate the analog bypass implementation. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:43 +08:00
/* initiate offset cancellation */
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
twl4030_codec_enable(codec, 1);
reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
reg &= ~TWL4030_OFFSET_CNCL_SEL;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
reg |= pdata->offset_cncl_path;
ASoC: TWL4030: Move the twl4030_power_up and _power_down function Move the twl4030_power_up and twl4030_power_down function earlier to facilitate the analog bypass implementation. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:43 +08:00
twl4030_write(codec, TWL4030_REG_ANAMICL,
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
reg | TWL4030_CNCL_OFFSET_START);
ASoC: TWL4030: Move the twl4030_power_up and _power_down function Move the twl4030_power_up and twl4030_power_down function earlier to facilitate the analog bypass implementation. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:43 +08:00
/* wait for offset cancellation to complete */
do {
/* this takes a little while, so don't slam i2c */
udelay(2000);
mfd: Rename all twl4030_i2c* This patch renames function names like twl4030_i2c_write_u8, twl4030_i2c_read_u8 to twl_i2c_write_u8, twl_i2c_read_u8 and also common variable in twl-core.c Signed-off-by: Rajendra Nayak <rnayak@ti.com> Signed-off-by: Balaji T K <balajitk@ti.com> Signed-off-by: Santosh Shilimkar <santosh.shilimkar@ti.com> Acked-by: Kevin Hilman <khilman@deeprootsystems.com> Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
2009-12-14 04:23:33 +08:00
twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
ASoC: TWL4030: Move the twl4030_power_up and _power_down function Move the twl4030_power_up and twl4030_power_down function earlier to facilitate the analog bypass implementation. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:43 +08:00
TWL4030_REG_ANAMICL);
} while ((i++ < 100) &&
((byte & TWL4030_CNCL_OFFSET_START) ==
TWL4030_CNCL_OFFSET_START));
/* Make sure that the reg_cache has the same value as the HW */
twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte);
twl4030_codec_enable(codec, 0);
}
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
ASoC: TWL4030: Move the twl4030_power_up and _power_down function Move the twl4030_power_up and twl4030_power_down function earlier to facilitate the analog bypass implementation. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:43 +08:00
{
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
int status = -1;
if (enable) {
twl4030->apll_enabled++;
if (twl4030->apll_enabled == 1)
status = twl4030_codec_enable_resource(
TWL4030_CODEC_RES_APLL);
} else {
twl4030->apll_enabled--;
if (!twl4030->apll_enabled)
status = twl4030_codec_disable_resource(
TWL4030_CODEC_RES_APLL);
}
if (status >= 0)
twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status);
ASoC: TWL4030: Move the twl4030_power_up and _power_down function Move the twl4030_power_up and twl4030_power_down function earlier to facilitate the analog bypass implementation. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:43 +08:00
}
ASoC: TWL4030: DAPM mapping of the Earpiece output Adds DAPM muxing, routing for the Earpiece output. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:47 +08:00
/* Earpiece */
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = {
SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0),
SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_EAR_CTL, 1, 1, 0),
SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_EAR_CTL, 2, 1, 0),
SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_EAR_CTL, 3, 1, 0),
};
ASoC: TWL4030: DAPM mapping of the Earpiece output Adds DAPM muxing, routing for the Earpiece output. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:47 +08:00
ASoC: TWL4030: DAPM mapping of the PreDriv outputs Adds DAPM muxing, routing for the PreDrive outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:48 +08:00
/* PreDrive Left */
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
static const struct snd_kcontrol_new twl4030_dapm_predrivel_controls[] = {
SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDL_CTL, 0, 1, 0),
SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PREDL_CTL, 1, 1, 0),
SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDL_CTL, 2, 1, 0),
SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDL_CTL, 3, 1, 0),
};
ASoC: TWL4030: DAPM mapping of the PreDriv outputs Adds DAPM muxing, routing for the PreDrive outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:48 +08:00
/* PreDrive Right */
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
static const struct snd_kcontrol_new twl4030_dapm_predriver_controls[] = {
SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDR_CTL, 0, 1, 0),
SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PREDR_CTL, 1, 1, 0),
SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDR_CTL, 2, 1, 0),
SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDR_CTL, 3, 1, 0),
};
ASoC: TWL4030: DAPM mapping of the PreDriv outputs Adds DAPM muxing, routing for the PreDrive outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:48 +08:00
ASoC: TWL4030: DAPM mapping of the Headset outputs Adds DAPM muxing, routing for the Headset outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:49 +08:00
/* Headset Left */
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
static const struct snd_kcontrol_new twl4030_dapm_hsol_controls[] = {
SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 0, 1, 0),
SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_HS_SEL, 1, 1, 0),
SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_HS_SEL, 2, 1, 0),
};
ASoC: TWL4030: DAPM mapping of the Headset outputs Adds DAPM muxing, routing for the Headset outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:49 +08:00
/* Headset Right */
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
static const struct snd_kcontrol_new twl4030_dapm_hsor_controls[] = {
SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 3, 1, 0),
SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_HS_SEL, 4, 1, 0),
SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_HS_SEL, 5, 1, 0),
};
ASoC: TWL4030: DAPM mapping of the Headset outputs Adds DAPM muxing, routing for the Headset outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:49 +08:00
ASoC: TWL4030: DAPM mapping of the Carkit outputs Adds DAPM muxing, routing for the Carkit outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:50 +08:00
/* Carkit Left */
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
static const struct snd_kcontrol_new twl4030_dapm_carkitl_controls[] = {
SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKL_CTL, 0, 1, 0),
SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PRECKL_CTL, 1, 1, 0),
SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PRECKL_CTL, 2, 1, 0),
};
ASoC: TWL4030: DAPM mapping of the Carkit outputs Adds DAPM muxing, routing for the Carkit outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:50 +08:00
/* Carkit Right */
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = {
SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKR_CTL, 0, 1, 0),
SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PRECKR_CTL, 1, 1, 0),
SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PRECKR_CTL, 2, 1, 0),
};
ASoC: TWL4030: DAPM mapping of the Carkit outputs Adds DAPM muxing, routing for the Carkit outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:50 +08:00
ASoC: TWL4030: DAPM mapping of the Handsfree outputs Adds DAPM muxing, routing for the Handsfree outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:51 +08:00
/* Handsfree Left */
static const char *twl4030_handsfreel_texts[] =
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
{"Voice", "AudioL1", "AudioL2", "AudioR2"};
ASoC: TWL4030: DAPM mapping of the Handsfree outputs Adds DAPM muxing, routing for the Handsfree outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:51 +08:00
static const struct soc_enum twl4030_handsfreel_enum =
SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0,
ARRAY_SIZE(twl4030_handsfreel_texts),
twl4030_handsfreel_texts);
static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control =
SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum);
ASoC: TWL4030: HandsfreeL/R mute DAPM switch Add DAPM switch for HeadsetL/R mute. Since all bits are are needed for the HFL/R pop removal to work the switch is using the SW_SHADOW no HW register for the HandsfreeL/R mute. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:13 +08:00
/* Handsfree Left virtual mute */
static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control =
SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 0, 1, 0);
ASoC: TWL4030: DAPM mapping of the Handsfree outputs Adds DAPM muxing, routing for the Handsfree outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:51 +08:00
/* Handsfree Right */
static const char *twl4030_handsfreer_texts[] =
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
{"Voice", "AudioR1", "AudioR2", "AudioL2"};
ASoC: TWL4030: DAPM mapping of the Handsfree outputs Adds DAPM muxing, routing for the Handsfree outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:51 +08:00
static const struct soc_enum twl4030_handsfreer_enum =
SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0,
ARRAY_SIZE(twl4030_handsfreer_texts),
twl4030_handsfreer_texts);
static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
ASoC: TWL4030: HandsfreeL/R mute DAPM switch Add DAPM switch for HeadsetL/R mute. Since all bits are are needed for the HFL/R pop removal to work the switch is using the SW_SHADOW no HW register for the HandsfreeL/R mute. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:13 +08:00
/* Handsfree Right virtual mute */
static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control =
SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 1, 1, 0);
ASoC: TWL4030: Add VIBRA output This patch adds support for the VIBRA output on TWL4030 codec. The VIBRA output can be driven with audio data or with local vibrator driver. Add the needed DAPM elements and routes for the VIBRA output and controls for the VIBRA driver configuration. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-05 13:55:47 +08:00
/* Vibra */
/* Vibra audio path selection */
static const char *twl4030_vibra_texts[] =
{"AudioL1", "AudioR1", "AudioL2", "AudioR2"};
static const struct soc_enum twl4030_vibra_enum =
SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 2,
ARRAY_SIZE(twl4030_vibra_texts),
twl4030_vibra_texts);
static const struct snd_kcontrol_new twl4030_dapm_vibra_control =
SOC_DAPM_ENUM("Route", twl4030_vibra_enum);
/* Vibra path selection: local vibrator (PWM) or audio driven */
static const char *twl4030_vibrapath_texts[] =
{"Local vibrator", "Audio"};
static const struct soc_enum twl4030_vibrapath_enum =
SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 4,
ARRAY_SIZE(twl4030_vibrapath_texts),
twl4030_vibrapath_texts);
static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control =
SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum);
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
/* Left analog microphone selection */
ASoC: TWL4030: change DAPM for analog microphone selection The inputs of the twl4030 codec can be mixed, so we will use the mixer DAPM for the analog microphone registers(0x05, 0x06), but if we enable more than one input at the same time, the input impedance of the input amplifier will be reduced. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-11 19:36:08 +08:00
static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = {
ASoC: TWL4030: Fix for capture mixer strings Change the strings related to capture in order to be interpreted correctly by alsamixer and possible other UI based mixer applications. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 13:44:00 +08:00
SOC_DAPM_SINGLE("Main Mic Capture Switch",
TWL4030_REG_ANAMICL, 0, 1, 0),
SOC_DAPM_SINGLE("Headset Mic Capture Switch",
TWL4030_REG_ANAMICL, 1, 1, 0),
SOC_DAPM_SINGLE("AUXL Capture Switch",
TWL4030_REG_ANAMICL, 2, 1, 0),
SOC_DAPM_SINGLE("Carkit Mic Capture Switch",
TWL4030_REG_ANAMICL, 3, 1, 0),
ASoC: TWL4030: change DAPM for analog microphone selection The inputs of the twl4030 codec can be mixed, so we will use the mixer DAPM for the analog microphone registers(0x05, 0x06), but if we enable more than one input at the same time, the input impedance of the input amplifier will be reduced. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-11 19:36:08 +08:00
};
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
/* Right analog microphone selection */
ASoC: TWL4030: change DAPM for analog microphone selection The inputs of the twl4030 codec can be mixed, so we will use the mixer DAPM for the analog microphone registers(0x05, 0x06), but if we enable more than one input at the same time, the input impedance of the input amplifier will be reduced. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-11 19:36:08 +08:00
static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = {
ASoC: TWL4030: Fix for capture mixer strings Change the strings related to capture in order to be interpreted correctly by alsamixer and possible other UI based mixer applications. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 13:44:00 +08:00
SOC_DAPM_SINGLE("Sub Mic Capture Switch", TWL4030_REG_ANAMICR, 0, 1, 0),
SOC_DAPM_SINGLE("AUXR Capture Switch", TWL4030_REG_ANAMICR, 2, 1, 0),
ASoC: TWL4030: change DAPM for analog microphone selection The inputs of the twl4030 codec can be mixed, so we will use the mixer DAPM for the analog microphone registers(0x05, 0x06), but if we enable more than one input at the same time, the input impedance of the input amplifier will be reduced. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-11 19:36:08 +08:00
};
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
/* TX1 L/R Analog/Digital microphone selection */
static const char *twl4030_micpathtx1_texts[] =
{"Analog", "Digimic0"};
static const struct soc_enum twl4030_micpathtx1_enum =
SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 0,
ARRAY_SIZE(twl4030_micpathtx1_texts),
twl4030_micpathtx1_texts);
static const struct snd_kcontrol_new twl4030_dapm_micpathtx1_control =
SOC_DAPM_ENUM("Route", twl4030_micpathtx1_enum);
/* TX2 L/R Analog/Digital microphone selection */
static const char *twl4030_micpathtx2_texts[] =
{"Analog", "Digimic1"};
static const struct soc_enum twl4030_micpathtx2_enum =
SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 2,
ARRAY_SIZE(twl4030_micpathtx2_texts),
twl4030_micpathtx2_texts);
static const struct snd_kcontrol_new twl4030_dapm_micpathtx2_control =
SOC_DAPM_ENUM("Route", twl4030_micpathtx2_enum);
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
/* Analog bypass for AudioR1 */
static const struct snd_kcontrol_new twl4030_dapm_abypassr1_control =
SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXR1_APGA_CTL, 2, 1, 0);
/* Analog bypass for AudioL1 */
static const struct snd_kcontrol_new twl4030_dapm_abypassl1_control =
SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL1_APGA_CTL, 2, 1, 0);
/* Analog bypass for AudioR2 */
static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control =
SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXR2_APGA_CTL, 2, 1, 0);
/* Analog bypass for AudioL2 */
static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control =
SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0);
ASoC: TWL4030: Add VDL analog bypass This patch adds voice downlink analog bypass switch. It follows the same approach as in other analog bypass switches. DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer', that will also allow voice DAC to be powered in digital voice loopback (sidetone). Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-01 10:47:22 +08:00
/* Analog bypass for Voice */
static const struct snd_kcontrol_new twl4030_dapm_abypassv_control =
SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0);
ASoC: TWL4030: Fix for digital loopback gain range When the gain is configured using dB value it was not possible to use -24dB since the loopback got muted instead of -24dB. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-12 16:50:06 +08:00
/* Digital bypass gain, mute instead of -30dB */
ASoC: TWL4030: Add digital loopback support This patch adds the digital loopback/bypass support for twl4030 codec. The digital loopback will let the digimic0 (routed in the TX1 capture path inside of TWL4030) data to be routed back to the RX2 playback path (I2S stereo). It can also route the analog capture date routed through the TX1 back to RX2. Effectively the digital loopback is routing the audio from the TX1 capture path to the RX2 playback path. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 20:39:05 +08:00
static const unsigned int twl4030_dapm_dbypass_tlv[] = {
ASoC: TWL4030: Fix for digital loopback gain range When the gain is configured using dB value it was not possible to use -24dB since the loopback got muted instead of -24dB. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-12 16:50:06 +08:00
TLV_DB_RANGE_HEAD(3),
0, 1, TLV_DB_SCALE_ITEM(-3000, 600, 1),
2, 3, TLV_DB_SCALE_ITEM(-2400, 0, 0),
ASoC: TWL4030: Add digital loopback support This patch adds the digital loopback/bypass support for twl4030 codec. The digital loopback will let the digimic0 (routed in the TX1 capture path inside of TWL4030) data to be routed back to the RX2 playback path (I2S stereo). It can also route the analog capture date routed through the TX1 back to RX2. Effectively the digital loopback is routing the audio from the TX1 capture path to the RX2 playback path. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 20:39:05 +08:00
4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0),
};
/* Digital bypass left (TX1L -> RX2L) */
static const struct snd_kcontrol_new twl4030_dapm_dbypassl_control =
SOC_DAPM_SINGLE_TLV("Volume",
TWL4030_REG_ATX2ARXPGA, 3, 7, 0,
twl4030_dapm_dbypass_tlv);
/* Digital bypass right (TX1R -> RX2R) */
static const struct snd_kcontrol_new twl4030_dapm_dbypassr_control =
SOC_DAPM_SINGLE_TLV("Volume",
TWL4030_REG_ATX2ARXPGA, 0, 7, 0,
twl4030_dapm_dbypass_tlv);
ASoC: TWL4030: Add voice digital loopback: sidetone This patch add voice digital loopback (sidetone) to the twl4030 driver. It mixes voice uplink attenuated (by sidetone gain) with voice downlink when the codec is working in option2 (voice/audio mode). Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-01 10:48:08 +08:00
/*
* Voice Sidetone GAIN volume control:
* from -51 to -10 dB in 1 dB steps (mute instead of -51 dB)
*/
static DECLARE_TLV_DB_SCALE(twl4030_dapm_dbypassv_tlv, -5100, 100, 1);
/* Digital bypass voice: sidetone (VUL -> VDL)*/
static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control =
SOC_DAPM_SINGLE_TLV("Volume",
TWL4030_REG_VSTPGA, 0, 0x29, 0,
twl4030_dapm_dbypassv_tlv);
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
/*
* Output PGA builder:
* Handle the muting and unmuting of the given output (turning off the
* amplifier associated with the output pin)
ASoC: TWL4030: PM fix for output amplifiers Gain controls on outputs affect the power consumption when the gain is set to non 0 value. Outputs with amps have one register to configure the routing and the gain: PREDL_CTL (0x25): bit 0: Voice enable bit 1: Audio L1 enable bit 2: Audio L2 enable bit 3: Audio R2 enable bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB) bit 0 - 3: is handled in DAPM domain (DAPM_MIXER) bit 4 - 5: has simple volume control If there is no audio activity (BIAS_STANDBY), and user changes the volume, than the output amplifier will be enabled. If the user changes the routing (but the codec remains in BIAS_STANDBY), than the cached gain value also be written to the register, which enables the amplifier. The existing workaround for this is to have virtual PGAs associated with the outputs, and whit DAPM PMD the gain on the output will be forced to 0 (off) by bypassing the regcache. This failed to disable the amplifiers in several scenario (as mentioned above). Also if the codec is in BIAS_ON state, and user modifies a volume control, which path is actually not enabled, than that amplifier will be enabled as well, but it will be not turned off, since there is no DAPM path, which would make mute it. To prevent amps being enabled, when they are not needed, introduce the following workaround: Track the state of each of this type of output. In twl4030_write only allow actual write, when the given output is enabled, otherwise only update the reg_cache. The PGA event handlers on power up will write the cached value to the chip (restoring gain, routing selection). On power down 0 is written to the register (disabling the amp, and also just in case clearing the routing). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-22 23:46:37 +08:00
* On mute bypass the reg_cache and write 0 to the register
* On unmute: restore the register content from the reg_cache
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
* Outputs handled in this way: Earpiece, PreDrivL/R, CarkitL/R
*/
#define TWL4030_OUTPUT_PGA(pin_name, reg, mask) \
static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \
struct snd_kcontrol *kcontrol, int event) \
{ \
ASoC: Add indirection for CODEC private data One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-14 14:35:19 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); \
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
\
switch (event) { \
case SND_SOC_DAPM_POST_PMU: \
ASoC: TWL4030: PM fix for output amplifiers Gain controls on outputs affect the power consumption when the gain is set to non 0 value. Outputs with amps have one register to configure the routing and the gain: PREDL_CTL (0x25): bit 0: Voice enable bit 1: Audio L1 enable bit 2: Audio L2 enable bit 3: Audio R2 enable bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB) bit 0 - 3: is handled in DAPM domain (DAPM_MIXER) bit 4 - 5: has simple volume control If there is no audio activity (BIAS_STANDBY), and user changes the volume, than the output amplifier will be enabled. If the user changes the routing (but the codec remains in BIAS_STANDBY), than the cached gain value also be written to the register, which enables the amplifier. The existing workaround for this is to have virtual PGAs associated with the outputs, and whit DAPM PMD the gain on the output will be forced to 0 (off) by bypassing the regcache. This failed to disable the amplifiers in several scenario (as mentioned above). Also if the codec is in BIAS_ON state, and user modifies a volume control, which path is actually not enabled, than that amplifier will be enabled as well, but it will be not turned off, since there is no DAPM path, which would make mute it. To prevent amps being enabled, when they are not needed, introduce the following workaround: Track the state of each of this type of output. In twl4030_write only allow actual write, when the given output is enabled, otherwise only update the reg_cache. The PGA event handlers on power up will write the cached value to the chip (restoring gain, routing selection). On power down 0 is written to the register (disabling the amp, and also just in case clearing the routing). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-22 23:46:37 +08:00
twl4030->pin_name##_enabled = 1; \
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
twl4030_write(w->codec, reg, \
twl4030_read_reg_cache(w->codec, reg)); \
break; \
case SND_SOC_DAPM_POST_PMD: \
ASoC: TWL4030: PM fix for output amplifiers Gain controls on outputs affect the power consumption when the gain is set to non 0 value. Outputs with amps have one register to configure the routing and the gain: PREDL_CTL (0x25): bit 0: Voice enable bit 1: Audio L1 enable bit 2: Audio L2 enable bit 3: Audio R2 enable bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB) bit 0 - 3: is handled in DAPM domain (DAPM_MIXER) bit 4 - 5: has simple volume control If there is no audio activity (BIAS_STANDBY), and user changes the volume, than the output amplifier will be enabled. If the user changes the routing (but the codec remains in BIAS_STANDBY), than the cached gain value also be written to the register, which enables the amplifier. The existing workaround for this is to have virtual PGAs associated with the outputs, and whit DAPM PMD the gain on the output will be forced to 0 (off) by bypassing the regcache. This failed to disable the amplifiers in several scenario (as mentioned above). Also if the codec is in BIAS_ON state, and user modifies a volume control, which path is actually not enabled, than that amplifier will be enabled as well, but it will be not turned off, since there is no DAPM path, which would make mute it. To prevent amps being enabled, when they are not needed, introduce the following workaround: Track the state of each of this type of output. In twl4030_write only allow actual write, when the given output is enabled, otherwise only update the reg_cache. The PGA event handlers on power up will write the cached value to the chip (restoring gain, routing selection). On power down 0 is written to the register (disabling the amp, and also just in case clearing the routing). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-22 23:46:37 +08:00
twl4030->pin_name##_enabled = 0; \
twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \
0, reg); \
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
break; \
} \
return 0; \
}
TWL4030_OUTPUT_PGA(earpiece, TWL4030_REG_EAR_CTL, TWL4030_EAR_GAIN);
TWL4030_OUTPUT_PGA(predrivel, TWL4030_REG_PREDL_CTL, TWL4030_PREDL_GAIN);
TWL4030_OUTPUT_PGA(predriver, TWL4030_REG_PREDR_CTL, TWL4030_PREDR_GAIN);
TWL4030_OUTPUT_PGA(carkitl, TWL4030_REG_PRECKL_CTL, TWL4030_PRECKL_GAIN);
TWL4030_OUTPUT_PGA(carkitr, TWL4030_REG_PRECKR_CTL, TWL4030_PRECKR_GAIN);
ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:11 +08:00
static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp)
ASoC: TWL4030: hands-free start-up sequence. A special start-up sequence is required to reduce the pop-noise of Class D amplifier when enable hands-free on TWL4030. Signed-off-by: Stanley.Miao <stanley.miao@windriver.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-11 23:28:10 +08:00
{
unsigned char hs_ctl;
ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:11 +08:00
hs_ctl = twl4030_read_reg_cache(codec, reg);
ASoC: TWL4030: hands-free start-up sequence. A special start-up sequence is required to reduce the pop-noise of Class D amplifier when enable hands-free on TWL4030. Signed-off-by: Stanley.Miao <stanley.miao@windriver.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-11 23:28:10 +08:00
ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:11 +08:00
if (ramp) {
/* HF ramp-up */
hs_ctl |= TWL4030_HF_CTL_REF_EN;
twl4030_write(codec, reg, hs_ctl);
udelay(10);
ASoC: TWL4030: hands-free start-up sequence. A special start-up sequence is required to reduce the pop-noise of Class D amplifier when enable hands-free on TWL4030. Signed-off-by: Stanley.Miao <stanley.miao@windriver.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-11 23:28:10 +08:00
hs_ctl |= TWL4030_HF_CTL_RAMP_EN;
ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:11 +08:00
twl4030_write(codec, reg, hs_ctl);
udelay(40);
ASoC: TWL4030: hands-free start-up sequence. A special start-up sequence is required to reduce the pop-noise of Class D amplifier when enable hands-free on TWL4030. Signed-off-by: Stanley.Miao <stanley.miao@windriver.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-11 23:28:10 +08:00
hs_ctl |= TWL4030_HF_CTL_LOOP_EN;
hs_ctl |= TWL4030_HF_CTL_HB_EN;
ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:11 +08:00
twl4030_write(codec, reg, hs_ctl);
ASoC: TWL4030: hands-free start-up sequence. A special start-up sequence is required to reduce the pop-noise of Class D amplifier when enable hands-free on TWL4030. Signed-off-by: Stanley.Miao <stanley.miao@windriver.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-11 23:28:10 +08:00
} else {
ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:11 +08:00
/* HF ramp-down */
hs_ctl &= ~TWL4030_HF_CTL_LOOP_EN;
hs_ctl &= ~TWL4030_HF_CTL_HB_EN;
twl4030_write(codec, reg, hs_ctl);
hs_ctl &= ~TWL4030_HF_CTL_RAMP_EN;
twl4030_write(codec, reg, hs_ctl);
udelay(40);
hs_ctl &= ~TWL4030_HF_CTL_REF_EN;
twl4030_write(codec, reg, hs_ctl);
ASoC: TWL4030: hands-free start-up sequence. A special start-up sequence is required to reduce the pop-noise of Class D amplifier when enable hands-free on TWL4030. Signed-off-by: Stanley.Miao <stanley.miao@windriver.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-11 23:28:10 +08:00
}
ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:11 +08:00
}
ASoC: TWL4030: hands-free start-up sequence. A special start-up sequence is required to reduce the pop-noise of Class D amplifier when enable hands-free on TWL4030. Signed-off-by: Stanley.Miao <stanley.miao@windriver.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-11 23:28:10 +08:00
ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:11 +08:00
static int handsfreelpga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
switch (event) {
case SND_SOC_DAPM_POST_PMU:
handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 1);
break;
case SND_SOC_DAPM_POST_PMD:
handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 0);
break;
}
return 0;
}
static int handsfreerpga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
switch (event) {
case SND_SOC_DAPM_POST_PMU:
handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 1);
break;
case SND_SOC_DAPM_POST_PMD:
handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 0);
break;
}
ASoC: TWL4030: hands-free start-up sequence. A special start-up sequence is required to reduce the pop-noise of Class D amplifier when enable hands-free on TWL4030. Signed-off-by: Stanley.Miao <stanley.miao@windriver.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-11 23:28:10 +08:00
return 0;
}
ASoC: TWL4030: Vibra motor stop fix when it is driven with audio This patch fixes vibrator playing incoherently, when driven with audio. There is something wrong in switch 3 at H-bridge and VIBRA_SET still affects PWM generator. Slowest value fixes things. Signed-off-by: Jari Vanhala <ext-jari.vanhala@nokia.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-29 17:58:09 +08:00
static int vibramux_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
twl4030_write(w->codec, TWL4030_REG_VIBRA_SET, 0xff);
return 0;
}
ASoC: TWL4030: Change APLL powering sequence It seams that certain part of the twl4030 codec needs the APLL enabled before they are enabled. Paths which has any digital processing needs need the APLL enabled before they can function. For example the vibra output will have some random data after it is enabled and before the APLL also enabled. If only analog components are in use (analog bypass), than it seams, that the APLL does not need to be enabled. This lowers the power consumption with around ~0.005A. Adding DAPM_SUPPLY to the Digital playback route and also to the capture route to enable and disable the APLL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-29 17:58:10 +08:00
static int apll_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
twl4030_apll_enable(w->codec, 1);
break;
case SND_SOC_DAPM_POST_PMD:
twl4030_apll_enable(w->codec, 0);
break;
}
return 0;
}
ASoC: TWL4030: AIF/APLL fix in DAPM domain This patch orders the APLL and AIF power sequence in case of HiFi (audio in TWL4030 terms) playback/capture. We also need to make sure that the AIF is running during playback/capture, when there is no valid DAPM route available. For this purpose I introduce these virtual widgets: /* To have complete playback route all the time */ DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */ /* To have complete capture route all the time */ DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */ /* To have complete playback route for the voice module */ DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */ The DAPM_SUPPLY widgets for APLL and AIF are placed in a way, that during any audio activity the needed configuration of AIF and APLL will be enabled (playback, capture, analog loopback, digital loopback, and voice activity). The apll reference counting code has been lifted, and modified from Liam Girdwood's earlier patch. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-29 15:58:08 +08:00
static int aif_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
u8 audio_if;
audio_if = twl4030_read_reg_cache(w->codec, TWL4030_REG_AUDIO_IF);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
/* Enable AIF */
/* enable the PLL before we use it to clock the DAI */
twl4030_apll_enable(w->codec, 1);
twl4030_write(w->codec, TWL4030_REG_AUDIO_IF,
audio_if | TWL4030_AIF_EN);
break;
case SND_SOC_DAPM_POST_PMD:
/* disable the DAI before we stop it's source PLL */
twl4030_write(w->codec, TWL4030_REG_AUDIO_IF,
audio_if & ~TWL4030_AIF_EN);
twl4030_apll_enable(w->codec, 0);
break;
}
return 0;
}
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
static void headset_ramp(struct snd_soc_codec *codec, int ramp)
ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use The Headset Left anti-pop and bias ramp does not need to be enabled, if the headset is not in use. Move the code to DAPM event handler for HeadsetL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:41 +08:00
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
struct twl4030_codec_audio_data *pdata = codec->dev->platform_data;
ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use The Headset Left anti-pop and bias ramp does not need to be enabled, if the headset is not in use. Move the code to DAPM event handler for HeadsetL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:41 +08:00
unsigned char hs_gain, hs_pop;
ASoC: Add indirection for CODEC private data One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-14 14:35:19 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
/* Base values for ramp delay calculation: 2^19 - 2^26 */
unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304,
8388608, 16777216, 33554432, 67108864};
ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use The Headset Left anti-pop and bias ramp does not need to be enabled, if the headset is not in use. Move the code to DAPM event handler for HeadsetL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:41 +08:00
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET);
hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use The Headset Left anti-pop and bias ramp does not need to be enabled, if the headset is not in use. Move the code to DAPM event handler for HeadsetL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:41 +08:00
ASoC: TWL4030: Add EXTMUTE to reduce pop-noise effect According to TRM, an external FET controlled by a 1.8V output signal can be used to reduce the pop-noise heard when the audio amplifier is switched on. It is suggested that GPIO6 of TWL4030 be used, but any other gpio can be used instead. This is indicated in machine driver with the following twl4030_setup_data members: -hs_extmute. Set to 1 if board has support for EXTMUTE. -set_hs_extmute. Set to a callback funcion to control an external gpio line. Set to NULL if MUTE[GPIO6] pin is used. Codec driver takes care of enabling and disabling this output during the headset pop attenuation sequence. Also add a delay to let VMID settle in ramp up sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-02 08:17:43 +08:00
/* Enable external mute control, this dramatically reduces
* the pop-noise */
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
if (pdata && pdata->hs_extmute) {
if (pdata->set_hs_extmute) {
pdata->set_hs_extmute(1);
ASoC: TWL4030: Add EXTMUTE to reduce pop-noise effect According to TRM, an external FET controlled by a 1.8V output signal can be used to reduce the pop-noise heard when the audio amplifier is switched on. It is suggested that GPIO6 of TWL4030 be used, but any other gpio can be used instead. This is indicated in machine driver with the following twl4030_setup_data members: -hs_extmute. Set to 1 if board has support for EXTMUTE. -set_hs_extmute. Set to a callback funcion to control an external gpio line. Set to NULL if MUTE[GPIO6] pin is used. Codec driver takes care of enabling and disabling this output during the headset pop attenuation sequence. Also add a delay to let VMID settle in ramp up sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-02 08:17:43 +08:00
} else {
hs_pop |= TWL4030_EXTMUTE;
twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
}
}
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
if (ramp) {
/* Headset ramp-up according to the TRM */
ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use The Headset Left anti-pop and bias ramp does not need to be enabled, if the headset is not in use. Move the code to DAPM event handler for HeadsetL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:41 +08:00
hs_pop |= TWL4030_VMID_EN;
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
ASoC: TWL4030: PM fix for output amplifiers Gain controls on outputs affect the power consumption when the gain is set to non 0 value. Outputs with amps have one register to configure the routing and the gain: PREDL_CTL (0x25): bit 0: Voice enable bit 1: Audio L1 enable bit 2: Audio L2 enable bit 3: Audio R2 enable bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB) bit 0 - 3: is handled in DAPM domain (DAPM_MIXER) bit 4 - 5: has simple volume control If there is no audio activity (BIAS_STANDBY), and user changes the volume, than the output amplifier will be enabled. If the user changes the routing (but the codec remains in BIAS_STANDBY), than the cached gain value also be written to the register, which enables the amplifier. The existing workaround for this is to have virtual PGAs associated with the outputs, and whit DAPM PMD the gain on the output will be forced to 0 (off) by bypassing the regcache. This failed to disable the amplifiers in several scenario (as mentioned above). Also if the codec is in BIAS_ON state, and user modifies a volume control, which path is actually not enabled, than that amplifier will be enabled as well, but it will be not turned off, since there is no DAPM path, which would make mute it. To prevent amps being enabled, when they are not needed, introduce the following workaround: Track the state of each of this type of output. In twl4030_write only allow actual write, when the given output is enabled, otherwise only update the reg_cache. The PGA event handlers on power up will write the cached value to the chip (restoring gain, routing selection). On power down 0 is written to the register (disabling the amp, and also just in case clearing the routing). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-03-22 23:46:37 +08:00
/* Actually write to the register */
twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
hs_gain,
TWL4030_REG_HS_GAIN_SET);
ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use The Headset Left anti-pop and bias ramp does not need to be enabled, if the headset is not in use. Move the code to DAPM event handler for HeadsetL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:41 +08:00
hs_pop |= TWL4030_RAMP_EN;
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
ASoC: TWL4030: Add EXTMUTE to reduce pop-noise effect According to TRM, an external FET controlled by a 1.8V output signal can be used to reduce the pop-noise heard when the audio amplifier is switched on. It is suggested that GPIO6 of TWL4030 be used, but any other gpio can be used instead. This is indicated in machine driver with the following twl4030_setup_data members: -hs_extmute. Set to 1 if board has support for EXTMUTE. -set_hs_extmute. Set to a callback funcion to control an external gpio line. Set to NULL if MUTE[GPIO6] pin is used. Codec driver takes care of enabling and disabling this output during the headset pop attenuation sequence. Also add a delay to let VMID settle in ramp up sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-02 08:17:43 +08:00
/* Wait ramp delay time + 1, so the VMID can settle */
mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] /
twl4030->sysclk) + 1);
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
} else {
/* Headset ramp-down _not_ according to
* the TRM, but in a way that it is working */
ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use The Headset Left anti-pop and bias ramp does not need to be enabled, if the headset is not in use. Move the code to DAPM event handler for HeadsetL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:41 +08:00
hs_pop &= ~TWL4030_RAMP_EN;
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
/* Wait ramp delay time + 1, so the VMID can settle */
mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] /
twl4030->sysclk) + 1);
ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use The Headset Left anti-pop and bias ramp does not need to be enabled, if the headset is not in use. Move the code to DAPM event handler for HeadsetL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:41 +08:00
/* Bypass the reg_cache to mute the headset */
mfd: Rename all twl4030_i2c* This patch renames function names like twl4030_i2c_write_u8, twl4030_i2c_read_u8 to twl_i2c_write_u8, twl_i2c_read_u8 and also common variable in twl-core.c Signed-off-by: Rajendra Nayak <rnayak@ti.com> Signed-off-by: Balaji T K <balajitk@ti.com> Signed-off-by: Santosh Shilimkar <santosh.shilimkar@ti.com> Acked-by: Kevin Hilman <khilman@deeprootsystems.com> Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
2009-12-14 04:23:33 +08:00
twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use The Headset Left anti-pop and bias ramp does not need to be enabled, if the headset is not in use. Move the code to DAPM event handler for HeadsetL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:41 +08:00
hs_gain & (~0x0f),
TWL4030_REG_HS_GAIN_SET);
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use The Headset Left anti-pop and bias ramp does not need to be enabled, if the headset is not in use. Move the code to DAPM event handler for HeadsetL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:41 +08:00
hs_pop &= ~TWL4030_VMID_EN;
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
}
ASoC: TWL4030: Add EXTMUTE to reduce pop-noise effect According to TRM, an external FET controlled by a 1.8V output signal can be used to reduce the pop-noise heard when the audio amplifier is switched on. It is suggested that GPIO6 of TWL4030 be used, but any other gpio can be used instead. This is indicated in machine driver with the following twl4030_setup_data members: -hs_extmute. Set to 1 if board has support for EXTMUTE. -set_hs_extmute. Set to a callback funcion to control an external gpio line. Set to NULL if MUTE[GPIO6] pin is used. Codec driver takes care of enabling and disabling this output during the headset pop attenuation sequence. Also add a delay to let VMID settle in ramp up sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-02 08:17:43 +08:00
/* Disable external mute */
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
if (pdata && pdata->hs_extmute) {
if (pdata->set_hs_extmute) {
pdata->set_hs_extmute(0);
ASoC: TWL4030: Add EXTMUTE to reduce pop-noise effect According to TRM, an external FET controlled by a 1.8V output signal can be used to reduce the pop-noise heard when the audio amplifier is switched on. It is suggested that GPIO6 of TWL4030 be used, but any other gpio can be used instead. This is indicated in machine driver with the following twl4030_setup_data members: -hs_extmute. Set to 1 if board has support for EXTMUTE. -set_hs_extmute. Set to a callback funcion to control an external gpio line. Set to NULL if MUTE[GPIO6] pin is used. Codec driver takes care of enabling and disabling this output during the headset pop attenuation sequence. Also add a delay to let VMID settle in ramp up sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-02 08:17:43 +08:00
} else {
hs_pop &= ~TWL4030_EXTMUTE;
twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
}
}
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
}
static int headsetlpga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
ASoC: Add indirection for CODEC private data One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-14 14:35:19 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/* Do the ramp-up only once */
if (!twl4030->hsr_enabled)
headset_ramp(w->codec, 1);
twl4030->hsl_enabled = 1;
break;
case SND_SOC_DAPM_POST_PMD:
/* Do the ramp-down only if both headsetL/R is disabled */
if (!twl4030->hsr_enabled)
headset_ramp(w->codec, 0);
twl4030->hsl_enabled = 0;
break;
}
return 0;
}
static int headsetrpga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
ASoC: Add indirection for CODEC private data One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-14 14:35:19 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/* Do the ramp-up only once */
if (!twl4030->hsl_enabled)
headset_ramp(w->codec, 1);
twl4030->hsr_enabled = 1;
break;
case SND_SOC_DAPM_POST_PMD:
/* Do the ramp-down only if both headsetL/R is disabled */
if (!twl4030->hsl_enabled)
headset_ramp(w->codec, 0);
twl4030->hsr_enabled = 0;
ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use The Headset Left anti-pop and bias ramp does not need to be enabled, if the headset is not in use. Move the code to DAPM event handler for HeadsetL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:41 +08:00
break;
}
return 0;
}
ASoC: TWL4030: Add configurable delay after digimic enable When digital microphones are connected to twl, delay is needed after enabling the digimic interface of the codec. Add new parameter for the setup data, which can be used to pass the apropriate delay in ms after the digimic interface has been enabled. Without certain delay (in certain HW configuration) the beggining of the recorded sample contains a glitch, which is generated by the digital microphones. Delaying the micbias1, 2 (which is the bias for the digimic0 or 1) does not help, since the glitch is coming after switching the digimic interface. Reversing the micbias and digimic enable order does not work either (in that case the wait need to be added after the micbias enabled). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-20 20:49:09 +08:00
static int digimic_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
if (twl4030->digimic_delay)
mdelay(twl4030->digimic_delay);
return 0;
}
ASoC: TWL4030: Add helper function for output gain controls Some of the gain controls in TWL (mostly those which are associated with the outputs) are implemented in an interesting way: 0x0 : Power down (mute) 0x1 : 6dB 0x2 : 0 dB 0x3 : -6 dB Inverting not going to help with these. Custom volsw and volsw_2r get/put functions to handle these gains. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-24 19:49:38 +08:00
/*
* Some of the gain controls in TWL (mostly those which are associated with
* the outputs) are implemented in an interesting way:
* 0x0 : Power down (mute)
* 0x1 : 6dB
* 0x2 : 0 dB
* 0x3 : -6 dB
* Inverting not going to help with these.
* Custom volsw and volsw_2r get/put functions to handle these gain bits.
*/
#define SOC_DOUBLE_TLV_TWL4030(xname, xreg, shift_left, shift_right, xmax,\
xinvert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw, \
.get = snd_soc_get_volsw_twl4030, \
.put = snd_soc_put_volsw_twl4030, \
.private_value = (unsigned long)&(struct soc_mixer_control) \
{.reg = xreg, .shift = shift_left, .rshift = shift_right,\
.max = xmax, .invert = xinvert} }
#define SOC_DOUBLE_R_TLV_TWL4030(xname, reg_left, reg_right, xshift, xmax,\
xinvert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw_2r, \
.get = snd_soc_get_volsw_r2_twl4030,\
.put = snd_soc_put_volsw_r2_twl4030, \
.private_value = (unsigned long)&(struct soc_mixer_control) \
{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
ASoC: Register non-AC97 codec DAIs Currently this is done at module probe time since ASoC ties in codec device probe to the instantiation of the entire ASoC device. Subsequent patches will refactor the codec drivers to handle probing separately. Note that the core does not yet use this information. AC97 is special since the codec is controlled over the AC97 link but we want to give the machine driver a chance to set up the system before trying to instantiate since it may need to do configuration before the AC97 link will operate Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 03:17:58 +08:00
.rshift = xshift, .max = xmax, .invert = xinvert} }
ASoC: TWL4030: Add helper function for output gain controls Some of the gain controls in TWL (mostly those which are associated with the outputs) are implemented in an interesting way: 0x0 : Power down (mute) 0x1 : 6dB 0x2 : 0 dB 0x3 : -6 dB Inverting not going to help with these. Custom volsw and volsw_2r get/put functions to handle these gains. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-24 19:49:38 +08:00
#define SOC_SINGLE_TLV_TWL4030(xname, xreg, xshift, xmax, xinvert, tlv_array) \
SOC_DOUBLE_TLV_TWL4030(xname, xreg, xshift, xshift, xmax, \
xinvert, tlv_array)
static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int shift = mc->shift;
unsigned int rshift = mc->rshift;
int max = mc->max;
int mask = (1 << fls(max)) - 1;
ucontrol->value.integer.value[0] =
(snd_soc_read(codec, reg) >> shift) & mask;
if (ucontrol->value.integer.value[0])
ucontrol->value.integer.value[0] =
max + 1 - ucontrol->value.integer.value[0];
if (shift != rshift) {
ucontrol->value.integer.value[1] =
(snd_soc_read(codec, reg) >> rshift) & mask;
if (ucontrol->value.integer.value[1])
ucontrol->value.integer.value[1] =
max + 1 - ucontrol->value.integer.value[1];
}
return 0;
}
static int snd_soc_put_volsw_twl4030(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int shift = mc->shift;
unsigned int rshift = mc->rshift;
int max = mc->max;
int mask = (1 << fls(max)) - 1;
unsigned short val, val2, val_mask;
val = (ucontrol->value.integer.value[0] & mask);
val_mask = mask << shift;
if (val)
val = max + 1 - val;
val = val << shift;
if (shift != rshift) {
val2 = (ucontrol->value.integer.value[1] & mask);
val_mask |= mask << rshift;
if (val2)
val2 = max + 1 - val2;
val |= val2 << rshift;
}
return snd_soc_update_bits(codec, reg, val_mask, val);
}
static int snd_soc_get_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int reg2 = mc->rreg;
unsigned int shift = mc->shift;
int max = mc->max;
int mask = (1<<fls(max))-1;
ucontrol->value.integer.value[0] =
(snd_soc_read(codec, reg) >> shift) & mask;
ucontrol->value.integer.value[1] =
(snd_soc_read(codec, reg2) >> shift) & mask;
if (ucontrol->value.integer.value[0])
ucontrol->value.integer.value[0] =
max + 1 - ucontrol->value.integer.value[0];
if (ucontrol->value.integer.value[1])
ucontrol->value.integer.value[1] =
max + 1 - ucontrol->value.integer.value[1];
return 0;
}
static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int reg2 = mc->rreg;
unsigned int shift = mc->shift;
int max = mc->max;
int mask = (1 << fls(max)) - 1;
int err;
unsigned short val, val2, val_mask;
val_mask = mask << shift;
val = (ucontrol->value.integer.value[0] & mask);
val2 = (ucontrol->value.integer.value[1] & mask);
if (val)
val = max + 1 - val;
if (val2)
val2 = max + 1 - val2;
val = val << shift;
val2 = val2 << shift;
err = snd_soc_update_bits(codec, reg, val_mask, val);
if (err < 0)
return err;
err = snd_soc_update_bits(codec, reg2, val_mask, val2);
return err;
}
ASoC: TWL4030: Add control for selecting codec operation mode Add a control for selecting the codec operation mode. TWL4030 codec has two modes: - Option 1. Audio only (4 audio DACs) - Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC) Control is restricted when a stream is ongoing, since codec's operation mode cannot be changed on-the-fly. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-19 00:52:55 +08:00
/* Codec operation modes */
static const char *twl4030_op_modes_texts[] = {
"Option 2 (voice/audio)", "Option 1 (audio)"
};
static const struct soc_enum twl4030_op_modes_enum =
SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0,
ARRAY_SIZE(twl4030_op_modes_texts),
twl4030_op_modes_texts);
ASoC: Staticise put_twl4030_opmode_enum_double() It's an operation for a control and doesn't need to be exported. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-20 20:54:02 +08:00
static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
ASoC: TWL4030: Add control for selecting codec operation mode Add a control for selecting the codec operation mode. TWL4030 codec has two modes: - Option 1. Audio only (4 audio DACs) - Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC) Control is restricted when a stream is ongoing, since codec's operation mode cannot be changed on-the-fly. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-19 00:52:55 +08:00
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
ASoC: Add indirection for CODEC private data One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-14 14:35:19 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: TWL4030: Add control for selecting codec operation mode Add a control for selecting the codec operation mode. TWL4030 codec has two modes: - Option 1. Audio only (4 audio DACs) - Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC) Control is restricted when a stream is ongoing, since codec's operation mode cannot be changed on-the-fly. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-19 00:52:55 +08:00
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
unsigned short val;
unsigned short mask, bitmask;
if (twl4030->configured) {
printk(KERN_ERR "twl4030 operation mode cannot be "
"changed on-the-fly\n");
return -EBUSY;
}
for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
;
if (ucontrol->value.enumerated.item[0] > e->max - 1)
return -EINVAL;
val = ucontrol->value.enumerated.item[0] << e->shift_l;
mask = (bitmask - 1) << e->shift_l;
if (e->shift_l != e->shift_r) {
if (ucontrol->value.enumerated.item[1] > e->max - 1)
return -EINVAL;
val |= ucontrol->value.enumerated.item[1] << e->shift_r;
mask |= (bitmask - 1) << e->shift_r;
}
return snd_soc_update_bits(codec, e->reg, mask, val);
}
ASoC: TWL4030: Change the Master volume control to TLV TWL4030 FGAIN volume control has a range: -62 to 0 dB in 1 dB steps, 0 in the FGAIN means mute. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-24 19:49:35 +08:00
/*
* FGAIN volume control:
* from -62 to 0 dB in 1 dB steps (mute instead of -63 dB)
*/
ASoC: TWL4030: Change the common playback volume controls Add Playback volume controls for all four DACs. All four paths has three levels of volume controls: Digital Fine gain, Digital Coarse gain, Analog gain. The controls are named to reflect their connection to the DACs. Per DAC volume can be performed, if needed: amixer sset 'DAC1 Analog' 5,10 DACL1 analog gain to 5 DACR1 analog gain to 10 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-01 16:03:46 +08:00
static DECLARE_TLV_DB_SCALE(digital_fine_tlv, -6300, 100, 1);
ASoC: TWL4030: Change the Master volume control to TLV TWL4030 FGAIN volume control has a range: -62 to 0 dB in 1 dB steps, 0 in the FGAIN means mute. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-24 19:49:35 +08:00
ASoC: TWL4030: Add CGAIN volume control Add CGAIN (Coarse gain control) to TWL4030 codec. The range of the CGAIN is: 0 dB to 12 dB in 6 dB steps. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-24 19:49:36 +08:00
/*
* CGAIN volume control:
* 0 dB to 12 dB in 6 dB steps
* value 2 and 3 means 12 dB
*/
ASoC: TWL4030: Change the common playback volume controls Add Playback volume controls for all four DACs. All four paths has three levels of volume controls: Digital Fine gain, Digital Coarse gain, Analog gain. The controls are named to reflect their connection to the DACs. Per DAC volume can be performed, if needed: amixer sset 'DAC1 Analog' 5,10 DACL1 analog gain to 5 DACR1 analog gain to 10 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-01 16:03:46 +08:00
static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0);
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
/*
* Voice Downlink GAIN volume control:
* from -37 to 12 dB in 1 dB steps (mute instead of -37 dB)
*/
static DECLARE_TLV_DB_SCALE(digital_voice_downlink_tlv, -3700, 100, 1);
ASoC: TWL4030: Change the common playback volume controls Add Playback volume controls for all four DACs. All four paths has three levels of volume controls: Digital Fine gain, Digital Coarse gain, Analog gain. The controls are named to reflect their connection to the DACs. Per DAC volume can be performed, if needed: amixer sset 'DAC1 Analog' 5,10 DACL1 analog gain to 5 DACR1 analog gain to 10 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-01 16:03:46 +08:00
/*
* Analog playback gain
* -24 dB to 12 dB in 2 dB steps
*/
static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
ASoC: TWL4030: Add CGAIN volume control Add CGAIN (Coarse gain control) to TWL4030 codec. The range of the CGAIN is: 0 dB to 12 dB in 6 dB steps. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-24 19:49:36 +08:00
ASoC: TWL4030: Add volume controls for outputs All outputs have dedicated gain controls except the HandsFree output. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-01 16:03:47 +08:00
/*
* Gain controls tied to outputs
* -6 dB to 6 dB in 6 dB steps (mute instead of -12)
*/
static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
ASoC: TWL4030: Fix gain control for earpiece amplifier The gain control for earpiece amplifier uses 0dB ~ 12dB according to the TRM, but the present code is implemented to -6dB ~ 6dB. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-28 17:18:05 +08:00
/*
* Gain control for earpiece amplifier
* 0 dB to 12 dB in 6 dB steps (mute instead of -6)
*/
static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1);
ASoC: TWL4030: Change the capture volume control to TLV The digital Capture gain control has a range: 0 to 31 dB in 1 dB steps. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-01 16:03:45 +08:00
/*
* Capture gain after the ADCs
* from 0 dB to 31 dB in 1 dB steps
*/
static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0);
ASoC: TWL4030: Add input selection and gain controls The TWL4030 codec device has two ADCs. Both of them can have several inputs routed to them, but TRM says that only one source can be selected for every ADC, even though every source has a dedicated bit in the registers. This patch adds input source controls. It modifies default register values to have no inputs selected and ADCs disabled. When some input is selected, control handlers enable apropriate input amplifier and ADC. If a microphone is selected, bias power is automatically enabled. When some input is deselected, unused chip parts are disabled. Microphone and line input recording tested on OMAP3 pandora board. Signed-off-by: Grazvydas Ignotas <notasas@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-03 02:48:58 +08:00
/*
* Gain control for input amplifiers
* 0 dB to 30 dB in 6 dB steps
*/
static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0);
ASoC: TWL4030: Add AVADC Clock Priority AVDAC clk priority allows to determine the path ADC must be connected when the codec is in option2 and both HiFi and Voice paths are enabled. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-22 23:51:52 +08:00
/* AVADC clock priority */
static const char *twl4030_avadc_clk_priority_texts[] = {
"Voice high priority", "HiFi high priority"
};
static const struct soc_enum twl4030_avadc_clk_priority_enum =
SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2,
ARRAY_SIZE(twl4030_avadc_clk_priority_texts),
twl4030_avadc_clk_priority_texts);
ASoC: TWL4030: Make the HS ramp delay configurable Enum type for selecting the desired ramp delay for the headset output. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-05 18:48:49 +08:00
static const char *twl4030_rampdelay_texts[] = {
"27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms",
"437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms",
"3495/2581/1748 ms"
};
static const struct soc_enum twl4030_rampdelay_enum =
SOC_ENUM_SINGLE(TWL4030_REG_HS_POPN_SET, 2,
ARRAY_SIZE(twl4030_rampdelay_texts),
twl4030_rampdelay_texts);
ASoC: TWL4030: Add VIBRA output This patch adds support for the VIBRA output on TWL4030 codec. The VIBRA output can be driven with audio data or with local vibrator driver. Add the needed DAPM elements and routes for the VIBRA output and controls for the VIBRA driver configuration. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-05 13:55:47 +08:00
/* Vibra H-bridge direction mode */
static const char *twl4030_vibradirmode_texts[] = {
"Vibra H-bridge direction", "Audio data MSB",
};
static const struct soc_enum twl4030_vibradirmode_enum =
SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 5,
ARRAY_SIZE(twl4030_vibradirmode_texts),
twl4030_vibradirmode_texts);
/* Vibra H-bridge direction */
static const char *twl4030_vibradir_texts[] = {
"Positive polarity", "Negative polarity",
};
static const struct soc_enum twl4030_vibradir_enum =
SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 1,
ARRAY_SIZE(twl4030_vibradir_texts),
twl4030_vibradir_texts);
ASoC: TWL4030: Add control for digimic Left Right swap The codec has support for swapping the left and right channels in the digimic interface. New kcontrol to handle this bit. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-12 15:35:36 +08:00
/* Digimic Left and right swapping */
static const char *twl4030_digimicswap_texts[] = {
"Not swapped", "Swapped",
};
static const struct soc_enum twl4030_digimicswap_enum =
SOC_ENUM_SINGLE(TWL4030_REG_MISC_SET_1, 0,
ARRAY_SIZE(twl4030_digimicswap_texts),
twl4030_digimicswap_texts);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
static const struct snd_kcontrol_new twl4030_snd_controls[] = {
ASoC: TWL4030: Add control for selecting codec operation mode Add a control for selecting the codec operation mode. TWL4030 codec has two modes: - Option 1. Audio only (4 audio DACs) - Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC) Control is restricted when a stream is ongoing, since codec's operation mode cannot be changed on-the-fly. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-19 00:52:55 +08:00
/* Codec operation mode control */
SOC_ENUM_EXT("Codec Operation Mode", twl4030_op_modes_enum,
snd_soc_get_enum_double,
snd_soc_put_twl4030_opmode_enum_double),
ASoC: TWL4030: Change the common playback volume controls Add Playback volume controls for all four DACs. All four paths has three levels of volume controls: Digital Fine gain, Digital Coarse gain, Analog gain. The controls are named to reflect their connection to the DACs. Per DAC volume can be performed, if needed: amixer sset 'DAC1 Analog' 5,10 DACL1 analog gain to 5 DACR1 analog gain to 10 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-01 16:03:46 +08:00
/* Common playback gain controls */
SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
0, 0x3f, 0, digital_fine_tlv),
SOC_DOUBLE_R_TLV("DAC2 Digital Fine Playback Volume",
TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA,
0, 0x3f, 0, digital_fine_tlv),
SOC_DOUBLE_R_TLV("DAC1 Digital Coarse Playback Volume",
TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
6, 0x2, 0, digital_coarse_tlv),
SOC_DOUBLE_R_TLV("DAC2 Digital Coarse Playback Volume",
TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA,
6, 0x2, 0, digital_coarse_tlv),
SOC_DOUBLE_R_TLV("DAC1 Analog Playback Volume",
TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL,
3, 0x12, 1, analog_tlv),
SOC_DOUBLE_R_TLV("DAC2 Analog Playback Volume",
TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
3, 0x12, 1, analog_tlv),
ASoC: TWL4030: Add Analog PGA control switch to DAPM Add all four APGA switch to DAPM routing and widgets. Add user control for DA enable for all APGA as normal control. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 14:45:44 +08:00
SOC_DOUBLE_R("DAC1 Analog Playback Switch",
TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL,
1, 1, 0),
SOC_DOUBLE_R("DAC2 Analog Playback Switch",
TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
1, 1, 0),
ASoC: TWL4030: Change the capture volume control to TLV The digital Capture gain control has a range: 0 to 31 dB in 1 dB steps. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-01 16:03:45 +08:00
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
/* Common voice downlink gain controls */
SOC_SINGLE_TLV("DAC Voice Digital Downlink Volume",
TWL4030_REG_VRXPGA, 0, 0x31, 0, digital_voice_downlink_tlv),
SOC_SINGLE_TLV("DAC Voice Analog Downlink Volume",
TWL4030_REG_VDL_APGA_CTL, 3, 0x12, 1, analog_tlv),
SOC_SINGLE("DAC Voice Analog Downlink Switch",
TWL4030_REG_VDL_APGA_CTL, 1, 1, 0),
ASoC: TWL4030: Add volume controls for outputs All outputs have dedicated gain controls except the HandsFree output. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-01 16:03:47 +08:00
/* Separate output gain controls */
SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume",
TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL,
4, 3, 0, output_tvl),
SOC_DOUBLE_TLV_TWL4030("Headset Playback Volume",
TWL4030_REG_HS_GAIN_SET, 0, 2, 3, 0, output_tvl),
SOC_DOUBLE_R_TLV_TWL4030("Carkit Playback Volume",
TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL,
4, 3, 0, output_tvl),
SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
ASoC: TWL4030: Fix gain control for earpiece amplifier The gain control for earpiece amplifier uses 0dB ~ 12dB according to the TRM, but the present code is implemented to -6dB ~ 6dB. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-28 17:18:05 +08:00
TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl),
ASoC: TWL4030: Add volume controls for outputs All outputs have dedicated gain controls except the HandsFree output. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-01 16:03:47 +08:00
ASoC: TWL4030: Change the capture volume control to TLV The digital Capture gain control has a range: 0 to 31 dB in 1 dB steps. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-01 16:03:45 +08:00
/* Common capture gain controls */
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
ASoC: TWL4030: Change the capture volume control to TLV The digital Capture gain control has a range: 0 to 31 dB in 1 dB steps. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-01 16:03:45 +08:00
TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA,
0, 0x1f, 0, digital_capture_tlv),
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
SOC_DOUBLE_R_TLV("TX2 Digital Capture Volume",
TWL4030_REG_AVTXL2PGA, TWL4030_REG_AVTXR2PGA,
0, 0x1f, 0, digital_capture_tlv),
ASoC: TWL4030: Add input selection and gain controls The TWL4030 codec device has two ADCs. Both of them can have several inputs routed to them, but TRM says that only one source can be selected for every ADC, even though every source has a dedicated bit in the registers. This patch adds input source controls. It modifies default register values to have no inputs selected and ADCs disabled. When some input is selected, control handlers enable apropriate input amplifier and ADC. If a microphone is selected, bias power is automatically enabled. When some input is deselected, unused chip parts are disabled. Microphone and line input recording tested on OMAP3 pandora board. Signed-off-by: Grazvydas Ignotas <notasas@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-03 02:48:58 +08:00
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN,
ASoC: TWL4030: Add input selection and gain controls The TWL4030 codec device has two ADCs. Both of them can have several inputs routed to them, but TRM says that only one source can be selected for every ADC, even though every source has a dedicated bit in the registers. This patch adds input source controls. It modifies default register values to have no inputs selected and ADCs disabled. When some input is selected, control handlers enable apropriate input amplifier and ADC. If a microphone is selected, bias power is automatically enabled. When some input is deselected, unused chip parts are disabled. Microphone and line input recording tested on OMAP3 pandora board. Signed-off-by: Grazvydas Ignotas <notasas@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-03 02:48:58 +08:00
0, 3, 5, 0, input_gain_tlv),
ASoC: TWL4030: Make the HS ramp delay configurable Enum type for selecting the desired ramp delay for the headset output. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-05 18:48:49 +08:00
ASoC: TWL4030: Add AVADC Clock Priority AVDAC clk priority allows to determine the path ADC must be connected when the codec is in option2 and both HiFi and Voice paths are enabled. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-22 23:51:52 +08:00
SOC_ENUM("AVADC Clock Priority", twl4030_avadc_clk_priority_enum),
ASoC: TWL4030: Make the HS ramp delay configurable Enum type for selecting the desired ramp delay for the headset output. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-05 18:48:49 +08:00
SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum),
ASoC: TWL4030: Add VIBRA output This patch adds support for the VIBRA output on TWL4030 codec. The VIBRA output can be driven with audio data or with local vibrator driver. Add the needed DAPM elements and routes for the VIBRA output and controls for the VIBRA driver configuration. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-05 13:55:47 +08:00
SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum),
SOC_ENUM("Vibra H-bridge direction", twl4030_vibradir_enum),
ASoC: TWL4030: Add control for digimic Left Right swap The codec has support for swapping the left and right channels in the digimic interface. New kcontrol to handle this bit. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-12 15:35:36 +08:00
SOC_ENUM("Digimic LR Swap", twl4030_digimicswap_enum),
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
};
static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
/* Left channel inputs */
SND_SOC_DAPM_INPUT("MAINMIC"),
SND_SOC_DAPM_INPUT("HSMIC"),
SND_SOC_DAPM_INPUT("AUXL"),
SND_SOC_DAPM_INPUT("CARKITMIC"),
/* Right channel inputs */
SND_SOC_DAPM_INPUT("SUBMIC"),
SND_SOC_DAPM_INPUT("AUXR"),
/* Digital microphones (Stereo) */
SND_SOC_DAPM_INPUT("DIGIMIC0"),
SND_SOC_DAPM_INPUT("DIGIMIC1"),
/* Outputs */
ASoC: TWL4030: DAPM mapping of the Earpiece output Adds DAPM muxing, routing for the Earpiece output. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:47 +08:00
SND_SOC_DAPM_OUTPUT("EARPIECE"),
ASoC: TWL4030: DAPM mapping of the PreDriv outputs Adds DAPM muxing, routing for the PreDrive outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:48 +08:00
SND_SOC_DAPM_OUTPUT("PREDRIVEL"),
SND_SOC_DAPM_OUTPUT("PREDRIVER"),
ASoC: TWL4030: DAPM mapping of the Headset outputs Adds DAPM muxing, routing for the Headset outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:49 +08:00
SND_SOC_DAPM_OUTPUT("HSOL"),
SND_SOC_DAPM_OUTPUT("HSOR"),
ASoC: TWL4030: Add missing Carkit output SND_SOC_DAPM_OUTPUT definition for carkitL/R was missing. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-10 18:51:46 +08:00
SND_SOC_DAPM_OUTPUT("CARKITL"),
SND_SOC_DAPM_OUTPUT("CARKITR"),
ASoC: TWL4030: DAPM mapping of the Handsfree outputs Adds DAPM muxing, routing for the Handsfree outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:51 +08:00
SND_SOC_DAPM_OUTPUT("HFL"),
SND_SOC_DAPM_OUTPUT("HFR"),
ASoC: TWL4030: Add VIBRA output This patch adds support for the VIBRA output on TWL4030 codec. The VIBRA output can be driven with audio data or with local vibrator driver. Add the needed DAPM elements and routes for the VIBRA output and controls for the VIBRA driver configuration. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-05 13:55:47 +08:00
SND_SOC_DAPM_OUTPUT("VIBRA"),
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
ASoC: TWL4030: AIF/APLL fix in DAPM domain This patch orders the APLL and AIF power sequence in case of HiFi (audio in TWL4030 terms) playback/capture. We also need to make sure that the AIF is running during playback/capture, when there is no valid DAPM route available. For this purpose I introduce these virtual widgets: /* To have complete playback route all the time */ DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */ /* To have complete capture route all the time */ DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */ /* To have complete playback route for the voice module */ DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */ The DAPM_SUPPLY widgets for APLL and AIF are placed in a way, that during any audio activity the needed configuration of AIF and APLL will be enabled (playback, capture, analog loopback, digital loopback, and voice activity). The apll reference counting code has been lifted, and modified from Liam Girdwood's earlier patch. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-29 15:58:08 +08:00
/* AIF and APLL clocks for running DAIs (including loopback) */
SND_SOC_DAPM_OUTPUT("Virtual HiFi OUT"),
SND_SOC_DAPM_INPUT("Virtual HiFi IN"),
SND_SOC_DAPM_OUTPUT("Virtual Voice OUT"),
ASoC: TWL4030: Correct DAPM_DAC with power control Add all four DACs to dapm_widgets with power switch. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 14:45:43 +08:00
/* DACs */
ASoC: TWL4030: Differentiate the playback streams Give unique stream names for the two playback streams so DAPM can figure out which codec_dai is in use. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-22 20:12:15 +08:00
SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback",
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
SND_SOC_NOPM, 0, 0),
ASoC: TWL4030: Differentiate the playback streams Give unique stream names for the two playback streams so DAPM can figure out which codec_dai is in use. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-22 20:12:15 +08:00
SND_SOC_DAPM_DAC("DAC Left1", "Left Front HiFi Playback",
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
SND_SOC_NOPM, 0, 0),
ASoC: TWL4030: Differentiate the playback streams Give unique stream names for the two playback streams so DAPM can figure out which codec_dai is in use. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-22 20:12:15 +08:00
SND_SOC_DAPM_DAC("DAC Right2", "Right Rear HiFi Playback",
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
SND_SOC_NOPM, 0, 0),
ASoC: TWL4030: Differentiate the playback streams Give unique stream names for the two playback streams so DAPM can figure out which codec_dai is in use. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-22 20:12:15 +08:00
SND_SOC_DAPM_DAC("DAC Left2", "Left Rear HiFi Playback",
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
SND_SOC_NOPM, 0, 0),
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback",
ASoC: TWL4030: Add VDL analog bypass This patch adds voice downlink analog bypass switch. It follows the same approach as in other analog bypass switches. DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer', that will also allow voice DAC to be powered in digital voice loopback (sidetone). Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-01 10:47:22 +08:00
SND_SOC_NOPM, 0, 0),
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
/* Analog bypasses */
ASoC: TWL4030: Remove bypass tracking Since ASoC core now handling the codec bias differently there is no need to do the tracking of bypass switch states anymore. Handling of the common bit for analog loopbacks is done with DAPM_SUPPLY for the bypass paths. Now this bit is only enabled when there is a complete analog bypass path, compared to the previous implementation, when the global switch was enabled if there were any of the analog bypass switch was on (regardless if there were complete path or not) Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-28 16:57:04 +08:00
SND_SOC_DAPM_SWITCH("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_abypassr1_control),
SND_SOC_DAPM_SWITCH("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_abypassl1_control),
SND_SOC_DAPM_SWITCH("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_abypassr2_control),
SND_SOC_DAPM_SWITCH("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_abypassl2_control),
SND_SOC_DAPM_SWITCH("Voice Analog Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_abypassv_control),
/* Master analog loopback switch */
SND_SOC_DAPM_SUPPLY("FM Loop Enable", TWL4030_REG_MISC_SET_1, 5, 0,
NULL, 0),
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
ASoC: TWL4030: Add digital loopback support This patch adds the digital loopback/bypass support for twl4030 codec. The digital loopback will let the digimic0 (routed in the TX1 capture path inside of TWL4030) data to be routed back to the RX2 playback path (I2S stereo). It can also route the analog capture date routed through the TX1 back to RX2. Effectively the digital loopback is routing the audio from the TX1 capture path to the RX2 playback path. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 20:39:05 +08:00
/* Digital bypasses */
ASoC: TWL4030: Remove bypass tracking Since ASoC core now handling the codec bias differently there is no need to do the tracking of bypass switch states anymore. Handling of the common bit for analog loopbacks is done with DAPM_SUPPLY for the bypass paths. Now this bit is only enabled when there is a complete analog bypass path, compared to the previous implementation, when the global switch was enabled if there were any of the analog bypass switch was on (regardless if there were complete path or not) Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-28 16:57:04 +08:00
SND_SOC_DAPM_SWITCH("Left Digital Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_dbypassl_control),
SND_SOC_DAPM_SWITCH("Right Digital Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_dbypassr_control),
SND_SOC_DAPM_SWITCH("Voice Digital Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_dbypassv_control),
ASoC: TWL4030: Add digital loopback support This patch adds the digital loopback/bypass support for twl4030 codec. The digital loopback will let the digimic0 (routed in the TX1 capture path inside of TWL4030) data to be routed back to the RX2 playback path (I2S stereo). It can also route the analog capture date routed through the TX1 back to RX2. Effectively the digital loopback is routing the audio from the TX1 capture path to the RX2 playback path. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 20:39:05 +08:00
ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:04 +08:00
/* Digital mixers, power control for the physical DACs */
SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer",
TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Digital L1 Playback Mixer",
TWL4030_REG_AVDAC_CTL, 1, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Digital R2 Playback Mixer",
TWL4030_REG_AVDAC_CTL, 2, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Digital L2 Playback Mixer",
TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Digital Voice Playback Mixer",
TWL4030_REG_AVDAC_CTL, 4, 0, NULL, 0),
/* Analog mixers, power control for the physical PGAs */
SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer",
TWL4030_REG_ARXR1_APGA_CTL, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer",
TWL4030_REG_ARXL1_APGA_CTL, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer",
TWL4030_REG_ARXR2_APGA_CTL, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer",
TWL4030_REG_ARXL2_APGA_CTL, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer",
TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0),
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
ASoC: TWL4030: Change APLL powering sequence It seams that certain part of the twl4030 codec needs the APLL enabled before they are enabled. Paths which has any digital processing needs need the APLL enabled before they can function. For example the vibra output will have some random data after it is enabled and before the APLL also enabled. If only analog components are in use (analog bypass), than it seams, that the APLL does not need to be enabled. This lowers the power consumption with around ~0.005A. Adding DAPM_SUPPLY to the Digital playback route and also to the capture route to enable and disable the APLL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-29 17:58:10 +08:00
SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event,
SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
ASoC: TWL4030: AIF/APLL fix in DAPM domain This patch orders the APLL and AIF power sequence in case of HiFi (audio in TWL4030 terms) playback/capture. We also need to make sure that the AIF is running during playback/capture, when there is no valid DAPM route available. For this purpose I introduce these virtual widgets: /* To have complete playback route all the time */ DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */ /* To have complete capture route all the time */ DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */ /* To have complete playback route for the voice module */ DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */ The DAPM_SUPPLY widgets for APLL and AIF are placed in a way, that during any audio activity the needed configuration of AIF and APLL will be enabled (playback, capture, analog loopback, digital loopback, and voice activity). The apll reference counting code has been lifted, and modified from Liam Girdwood's earlier patch. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-29 15:58:08 +08:00
SND_SOC_DAPM_SUPPLY("AIF Enable", SND_SOC_NOPM, 0, 0, aif_event,
SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
ASoC: TWL4030: Add supply for audio serial interface control The serial interface (TDM/I2S) for the audio block have been constantly enabled. Introduce a new DAPM_SUPPLY for handling the AIF_EN bit, so the interface is only enabled, when there is a need for it. For example when only the analog loopback is enabled, there is no need to keep the serial interface active. I have added the persons who contributed to the Voice path of twl4030 codec driver, so they might have the ability to test this patch, and send an update for the Voice path, if it is necessary Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-09 21:24:04 +08:00
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
/* Output MIXER controls */
ASoC: TWL4030: DAPM mapping of the Earpiece output Adds DAPM muxing, routing for the Earpiece output. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:47 +08:00
/* Earpiece */
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_earpiece_controls[0],
ARRAY_SIZE(twl4030_dapm_earpiece_controls)),
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
SND_SOC_DAPM_PGA_E("Earpiece PGA", SND_SOC_NOPM,
0, 0, NULL, 0, earpiecepga_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
ASoC: TWL4030: DAPM mapping of the PreDriv outputs Adds DAPM muxing, routing for the PreDrive outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:48 +08:00
/* PreDrivL/R */
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_predrivel_controls[0],
ARRAY_SIZE(twl4030_dapm_predrivel_controls)),
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
SND_SOC_DAPM_PGA_E("PredriveL PGA", SND_SOC_NOPM,
0, 0, NULL, 0, predrivelpga_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_predriver_controls[0],
ARRAY_SIZE(twl4030_dapm_predriver_controls)),
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
SND_SOC_DAPM_PGA_E("PredriveR PGA", SND_SOC_NOPM,
0, 0, NULL, 0, predriverpga_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
ASoC: TWL4030: DAPM mapping of the Headset outputs Adds DAPM muxing, routing for the Headset outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:49 +08:00
/* HeadsetL/R */
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
SND_SOC_DAPM_MIXER("HeadsetL Mixer", SND_SOC_NOPM, 0, 0,
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
&twl4030_dapm_hsol_controls[0],
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
ARRAY_SIZE(twl4030_dapm_hsol_controls)),
SND_SOC_DAPM_PGA_E("HeadsetL PGA", SND_SOC_NOPM,
0, 0, NULL, 0, headsetlpga_event,
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_hsor_controls[0],
ARRAY_SIZE(twl4030_dapm_hsor_controls)),
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
SND_SOC_DAPM_PGA_E("HeadsetR PGA", SND_SOC_NOPM,
0, 0, NULL, 0, headsetrpga_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
ASoC: TWL4030: DAPM mapping of the Carkit outputs Adds DAPM muxing, routing for the Carkit outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:50 +08:00
/* CarkitL/R */
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_carkitl_controls[0],
ARRAY_SIZE(twl4030_dapm_carkitl_controls)),
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
SND_SOC_DAPM_PGA_E("CarkitL PGA", SND_SOC_NOPM,
0, 0, NULL, 0, carkitlpga_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_carkitr_controls[0],
ARRAY_SIZE(twl4030_dapm_carkitr_controls)),
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
SND_SOC_DAPM_PGA_E("CarkitR PGA", SND_SOC_NOPM,
0, 0, NULL, 0, carkitrpga_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
/* Output MUX controls */
ASoC: TWL4030: DAPM mapping of the Handsfree outputs Adds DAPM muxing, routing for the Handsfree outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:51 +08:00
/* HandsfreeL/R */
ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:11 +08:00
SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_handsfreel_control),
ASoC: Remove word "Switch" from Handsfree switch name SoC dapm adds the suffix "Switch" to SND_SOC_DAPM_SWITCH controls, removing word "Switch" from HandsfreeL/HandsfreeR widget name for avoiding to duplicate it. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-26 01:36:14 +08:00
SND_SOC_DAPM_SWITCH("HandsfreeL", SND_SOC_NOPM, 0, 0,
ASoC: TWL4030: HandsfreeL/R mute DAPM switch Add DAPM switch for HeadsetL/R mute. Since all bits are are needed for the HFL/R pop removal to work the switch is using the SW_SHADOW no HW register for the HandsfreeL/R mute. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:13 +08:00
&twl4030_dapm_handsfreelmute_control),
ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:11 +08:00
SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM,
0, 0, NULL, 0, handsfreelpga_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0,
&twl4030_dapm_handsfreer_control),
ASoC: Remove word "Switch" from Handsfree switch name SoC dapm adds the suffix "Switch" to SND_SOC_DAPM_SWITCH controls, removing word "Switch" from HandsfreeL/HandsfreeR widget name for avoiding to duplicate it. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-26 01:36:14 +08:00
SND_SOC_DAPM_SWITCH("HandsfreeR", SND_SOC_NOPM, 0, 0,
ASoC: TWL4030: HandsfreeL/R mute DAPM switch Add DAPM switch for HeadsetL/R mute. Since all bits are are needed for the HFL/R pop removal to work the switch is using the SW_SHADOW no HW register for the HandsfreeL/R mute. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:13 +08:00
&twl4030_dapm_handsfreermute_control),
ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:11 +08:00
SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM,
0, 0, NULL, 0, handsfreerpga_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
ASoC: TWL4030: Add VIBRA output This patch adds support for the VIBRA output on TWL4030 codec. The VIBRA output can be driven with audio data or with local vibrator driver. Add the needed DAPM elements and routes for the VIBRA output and controls for the VIBRA driver configuration. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-05 13:55:47 +08:00
/* Vibra */
ASoC: TWL4030: Vibra motor stop fix when it is driven with audio This patch fixes vibrator playing incoherently, when driven with audio. There is something wrong in switch 3 at H-bridge and VIBRA_SET still affects PWM generator. Slowest value fixes things. Signed-off-by: Jari Vanhala <ext-jari.vanhala@nokia.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-29 17:58:09 +08:00
SND_SOC_DAPM_MUX_E("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0,
&twl4030_dapm_vibra_control, vibramux_event,
SND_SOC_DAPM_PRE_PMU),
ASoC: TWL4030: Add VIBRA output This patch adds support for the VIBRA output on TWL4030 codec. The VIBRA output can be driven with audio data or with local vibrator driver. Add the needed DAPM elements and routes for the VIBRA output and controls for the VIBRA driver configuration. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-05 13:55:47 +08:00
SND_SOC_DAPM_MUX("Vibra Route", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_vibrapath_control),
ASoC: TWL4030: DAPM mapping of the Earpiece output Adds DAPM muxing, routing for the Earpiece output. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:47 +08:00
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
/* Introducing four virtual ADC, since TWL4030 have four channel for
capture */
SND_SOC_DAPM_ADC("ADC Virtual Left1", "Left Front Capture",
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_ADC("ADC Virtual Right1", "Right Front Capture",
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_ADC("ADC Virtual Left2", "Left Rear Capture",
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_ADC("ADC Virtual Right2", "Right Rear Capture",
SND_SOC_NOPM, 0, 0),
/* Analog/Digital mic path selection.
TX1 Left/Right: either analog Left/Right or Digimic0
TX2 Left/Right: either analog Left/Right or Digimic1 */
ASoC: TWL4030: Capture route runtime DAPM ordering fix Fix the ordering problem in DAPM domain, when the user changes between digital and analog sources during active capture (or loopback) scenario. Before this patch, when the user changed from analog source to digital there were a short time, when the codec enabled analog mic bias (2.2 volts) instead of the correct digital mic bias (1.8 volts) to the digital microphones. This behaviour caused by the former implementation of selecting the correct type of bias. This was done at the POST_REG event of the DAPM_MUX_E("TXx Capture Route") widget. By moving the bias type selection as DAPM_SUPPLY and connecting it to the corresponding digimic widget the problematic situation can be avoided. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-08-03 17:01:01 +08:00
SND_SOC_DAPM_MUX("TX1 Capture Route", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_micpathtx1_control),
SND_SOC_DAPM_MUX("TX2 Capture Route", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_micpathtx2_control),
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
ASoC: TWL4030: change DAPM for analog microphone selection The inputs of the twl4030 codec can be mixed, so we will use the mixer DAPM for the analog microphone registers(0x05, 0x06), but if we enable more than one input at the same time, the input impedance of the input amplifier will be reduced. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-11 19:36:08 +08:00
/* Analog input mixers for the capture amplifiers */
ASoC: TWL4030: Fix for capture mixer strings Change the strings related to capture in order to be interpreted correctly by alsamixer and possible other UI based mixer applications. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 13:44:00 +08:00
SND_SOC_DAPM_MIXER("Analog Left",
ASoC: TWL4030: change DAPM for analog microphone selection The inputs of the twl4030 codec can be mixed, so we will use the mixer DAPM for the analog microphone registers(0x05, 0x06), but if we enable more than one input at the same time, the input impedance of the input amplifier will be reduced. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-11 19:36:08 +08:00
TWL4030_REG_ANAMICL, 4, 0,
&twl4030_dapm_analoglmic_controls[0],
ARRAY_SIZE(twl4030_dapm_analoglmic_controls)),
ASoC: TWL4030: Fix for capture mixer strings Change the strings related to capture in order to be interpreted correctly by alsamixer and possible other UI based mixer applications. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 13:44:00 +08:00
SND_SOC_DAPM_MIXER("Analog Right",
ASoC: TWL4030: change DAPM for analog microphone selection The inputs of the twl4030 codec can be mixed, so we will use the mixer DAPM for the analog microphone registers(0x05, 0x06), but if we enable more than one input at the same time, the input impedance of the input amplifier will be reduced. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-11 19:36:08 +08:00
TWL4030_REG_ANAMICR, 4, 0,
&twl4030_dapm_analogrmic_controls[0],
ARRAY_SIZE(twl4030_dapm_analogrmic_controls)),
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
ASoC: TWL4030: Physical ADC and amplifier power switch change Change the power switches for the physical ADC and for the amplifiers for the analog capture path to map more closely the actual path inside of the codec. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:42 +08:00
SND_SOC_DAPM_PGA("ADC Physical Left",
TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("ADC Physical Right",
TWL4030_REG_AVADC_CTL, 1, 0, NULL, 0),
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
ASoC: TWL4030: Add configurable delay after digimic enable When digital microphones are connected to twl, delay is needed after enabling the digimic interface of the codec. Add new parameter for the setup data, which can be used to pass the apropriate delay in ms after the digimic interface has been enabled. Without certain delay (in certain HW configuration) the beggining of the recorded sample contains a glitch, which is generated by the digital microphones. Delaying the micbias1, 2 (which is the bias for the digimic0 or 1) does not help, since the glitch is coming after switching the digimic interface. Reversing the micbias and digimic enable order does not work either (in that case the wait need to be added after the micbias enabled). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-20 20:49:09 +08:00
SND_SOC_DAPM_PGA_E("Digimic0 Enable",
TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0,
digimic_event, SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("Digimic1 Enable",
TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0,
digimic_event, SND_SOC_DAPM_POST_PMU),
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
ASoC: TWL4030: Capture route runtime DAPM ordering fix Fix the ordering problem in DAPM domain, when the user changes between digital and analog sources during active capture (or loopback) scenario. Before this patch, when the user changed from analog source to digital there were a short time, when the codec enabled analog mic bias (2.2 volts) instead of the correct digital mic bias (1.8 volts) to the digital microphones. This behaviour caused by the former implementation of selecting the correct type of bias. This was done at the POST_REG event of the DAPM_MUX_E("TXx Capture Route") widget. By moving the bias type selection as DAPM_SUPPLY and connecting it to the corresponding digimic widget the problematic situation can be avoided. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-08-03 17:01:01 +08:00
SND_SOC_DAPM_SUPPLY("micbias1 select", TWL4030_REG_MICBIAS_CTL, 5, 0,
NULL, 0),
SND_SOC_DAPM_SUPPLY("micbias2 select", TWL4030_REG_MICBIAS_CTL, 6, 0,
NULL, 0),
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0),
SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0),
SND_SOC_DAPM_MICBIAS("Headset Mic Bias", TWL4030_REG_MICBIAS_CTL, 2, 0),
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
};
static const struct snd_soc_dapm_route intercon[] = {
ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:04 +08:00
{"Digital L1 Playback Mixer", NULL, "DAC Left1"},
{"Digital R1 Playback Mixer", NULL, "DAC Right1"},
{"Digital L2 Playback Mixer", NULL, "DAC Left2"},
{"Digital R2 Playback Mixer", NULL, "DAC Right2"},
{"Digital Voice Playback Mixer", NULL, "DAC Voice"},
ASoC: TWL4030: Change APLL powering sequence It seams that certain part of the twl4030 codec needs the APLL enabled before they are enabled. Paths which has any digital processing needs need the APLL enabled before they can function. For example the vibra output will have some random data after it is enabled and before the APLL also enabled. If only analog components are in use (analog bypass), than it seams, that the APLL does not need to be enabled. This lowers the power consumption with around ~0.005A. Adding DAPM_SUPPLY to the Digital playback route and also to the capture route to enable and disable the APLL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-29 17:58:10 +08:00
/* Supply for the digital part (APLL) */
{"Digital Voice Playback Mixer", NULL, "APLL Enable"},
ASoC: TWL4030: DAC power optimization Restructure the DAPM connections in order to enable only the needed DAC (out of four in twl4030 series). I need to keep the 'AIF Enable' supply connected to the L2/R2 digital path, since the digital loopback needs AIF and APLL running. If no valid route available, than none of the DAC will be powered, but the AIF and APLL is going to be enabled. Furthermore, if only one audio path have valid route, than only the corresponding DAC will be powered. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-13 17:07:44 +08:00
{"DAC Left1", NULL, "AIF Enable"},
{"DAC Right1", NULL, "AIF Enable"},
{"DAC Left2", NULL, "AIF Enable"},
{"DAC Right1", NULL, "AIF Enable"},
ASoC: TWL4030: Add supply for audio serial interface control The serial interface (TDM/I2S) for the audio block have been constantly enabled. Introduce a new DAPM_SUPPLY for handling the AIF_EN bit, so the interface is only enabled, when there is a need for it. For example when only the analog loopback is enabled, there is no need to keep the serial interface active. I have added the persons who contributed to the Voice path of twl4030 codec driver, so they might have the ability to test this patch, and send an update for the Voice path, if it is necessary Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-09 21:24:04 +08:00
{"Digital R2 Playback Mixer", NULL, "AIF Enable"},
{"Digital L2 Playback Mixer", NULL, "AIF Enable"},
ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:04 +08:00
{"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"},
{"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"},
{"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"},
{"Analog R2 Playback Mixer", NULL, "Digital R2 Playback Mixer"},
{"Analog Voice Playback Mixer", NULL, "Digital Voice Playback Mixer"},
ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 12:13:34 +08:00
ASoC: TWL4030: DAPM mapping of the Earpiece output Adds DAPM muxing, routing for the Earpiece output. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:47 +08:00
/* Internal playback routings */
/* Earpiece */
ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:04 +08:00
{"Earpiece Mixer", "Voice", "Analog Voice Playback Mixer"},
{"Earpiece Mixer", "AudioL1", "Analog L1 Playback Mixer"},
{"Earpiece Mixer", "AudioL2", "Analog L2 Playback Mixer"},
{"Earpiece Mixer", "AudioR1", "Analog R1 Playback Mixer"},
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
{"Earpiece PGA", NULL, "Earpiece Mixer"},
ASoC: TWL4030: DAPM mapping of the PreDriv outputs Adds DAPM muxing, routing for the PreDrive outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:48 +08:00
/* PreDrivL */
ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:04 +08:00
{"PredriveL Mixer", "Voice", "Analog Voice Playback Mixer"},
{"PredriveL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
{"PredriveL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
{"PredriveL Mixer", "AudioR2", "Analog R2 Playback Mixer"},
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
{"PredriveL PGA", NULL, "PredriveL Mixer"},
ASoC: TWL4030: DAPM mapping of the PreDriv outputs Adds DAPM muxing, routing for the PreDrive outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:48 +08:00
/* PreDrivR */
ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:04 +08:00
{"PredriveR Mixer", "Voice", "Analog Voice Playback Mixer"},
{"PredriveR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
{"PredriveR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
{"PredriveR Mixer", "AudioL2", "Analog L2 Playback Mixer"},
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
{"PredriveR PGA", NULL, "PredriveR Mixer"},
ASoC: TWL4030: DAPM mapping of the Headset outputs Adds DAPM muxing, routing for the Headset outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:49 +08:00
/* HeadsetL */
ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:04 +08:00
{"HeadsetL Mixer", "Voice", "Analog Voice Playback Mixer"},
{"HeadsetL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
{"HeadsetL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
{"HeadsetL PGA", NULL, "HeadsetL Mixer"},
ASoC: TWL4030: DAPM mapping of the Headset outputs Adds DAPM muxing, routing for the Headset outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:49 +08:00
/* HeadsetR */
ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:04 +08:00
{"HeadsetR Mixer", "Voice", "Analog Voice Playback Mixer"},
{"HeadsetR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
{"HeadsetR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
{"HeadsetR PGA", NULL, "HeadsetR Mixer"},
ASoC: TWL4030: DAPM mapping of the Carkit outputs Adds DAPM muxing, routing for the Carkit outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:50 +08:00
/* CarkitL */
ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:04 +08:00
{"CarkitL Mixer", "Voice", "Analog Voice Playback Mixer"},
{"CarkitL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
{"CarkitL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
{"CarkitL PGA", NULL, "CarkitL Mixer"},
ASoC: TWL4030: DAPM mapping of the Carkit outputs Adds DAPM muxing, routing for the Carkit outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:50 +08:00
/* CarkitR */
ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:04 +08:00
{"CarkitR Mixer", "Voice", "Analog Voice Playback Mixer"},
{"CarkitR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
{"CarkitR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
{"CarkitR PGA", NULL, "CarkitR Mixer"},
ASoC: TWL4030: DAPM mapping of the Handsfree outputs Adds DAPM muxing, routing for the Handsfree outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:51 +08:00
/* HandsfreeL */
ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:04 +08:00
{"HandsfreeL Mux", "Voice", "Analog Voice Playback Mixer"},
{"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"},
{"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"},
{"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"},
ASoC: Remove word "Switch" from Handsfree switch name SoC dapm adds the suffix "Switch" to SND_SOC_DAPM_SWITCH controls, removing word "Switch" from HandsfreeL/HandsfreeR widget name for avoiding to duplicate it. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-26 01:36:14 +08:00
{"HandsfreeL", "Switch", "HandsfreeL Mux"},
{"HandsfreeL PGA", NULL, "HandsfreeL"},
ASoC: TWL4030: DAPM mapping of the Handsfree outputs Adds DAPM muxing, routing for the Handsfree outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:51 +08:00
/* HandsfreeR */
ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:04 +08:00
{"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"},
{"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"},
{"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"},
{"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"},
ASoC: Remove word "Switch" from Handsfree switch name SoC dapm adds the suffix "Switch" to SND_SOC_DAPM_SWITCH controls, removing word "Switch" from HandsfreeL/HandsfreeR widget name for avoiding to duplicate it. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-26 01:36:14 +08:00
{"HandsfreeR", "Switch", "HandsfreeR Mux"},
{"HandsfreeR PGA", NULL, "HandsfreeR"},
ASoC: TWL4030: Add VIBRA output This patch adds support for the VIBRA output on TWL4030 codec. The VIBRA output can be driven with audio data or with local vibrator driver. Add the needed DAPM elements and routes for the VIBRA output and controls for the VIBRA driver configuration. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-05 13:55:47 +08:00
/* Vibra */
{"Vibra Mux", "AudioL1", "DAC Left1"},
{"Vibra Mux", "AudioR1", "DAC Right1"},
{"Vibra Mux", "AudioL2", "DAC Left2"},
{"Vibra Mux", "AudioR2", "DAC Right2"},
ASoC: TWL4030: DAPM mapping of the Earpiece output Adds DAPM muxing, routing for the Earpiece output. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 18:35:47 +08:00
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
/* outputs */
ASoC: TWL4030: AIF/APLL fix in DAPM domain This patch orders the APLL and AIF power sequence in case of HiFi (audio in TWL4030 terms) playback/capture. We also need to make sure that the AIF is running during playback/capture, when there is no valid DAPM route available. For this purpose I introduce these virtual widgets: /* To have complete playback route all the time */ DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */ /* To have complete capture route all the time */ DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */ /* To have complete playback route for the voice module */ DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */ The DAPM_SUPPLY widgets for APLL and AIF are placed in a way, that during any audio activity the needed configuration of AIF and APLL will be enabled (playback, capture, analog loopback, digital loopback, and voice activity). The apll reference counting code has been lifted, and modified from Liam Girdwood's earlier patch. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-29 15:58:08 +08:00
/* Must be always connected (for AIF and APLL) */
ASoC: TWL4030: DAC power optimization Restructure the DAPM connections in order to enable only the needed DAC (out of four in twl4030 series). I need to keep the 'AIF Enable' supply connected to the L2/R2 digital path, since the digital loopback needs AIF and APLL running. If no valid route available, than none of the DAC will be powered, but the AIF and APLL is going to be enabled. Furthermore, if only one audio path have valid route, than only the corresponding DAC will be powered. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-13 17:07:44 +08:00
{"Virtual HiFi OUT", NULL, "DAC Left1"},
{"Virtual HiFi OUT", NULL, "DAC Right1"},
{"Virtual HiFi OUT", NULL, "DAC Left2"},
{"Virtual HiFi OUT", NULL, "DAC Right2"},
ASoC: TWL4030: AIF/APLL fix in DAPM domain This patch orders the APLL and AIF power sequence in case of HiFi (audio in TWL4030 terms) playback/capture. We also need to make sure that the AIF is running during playback/capture, when there is no valid DAPM route available. For this purpose I introduce these virtual widgets: /* To have complete playback route all the time */ DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */ /* To have complete capture route all the time */ DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */ /* To have complete playback route for the voice module */ DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */ The DAPM_SUPPLY widgets for APLL and AIF are placed in a way, that during any audio activity the needed configuration of AIF and APLL will be enabled (playback, capture, analog loopback, digital loopback, and voice activity). The apll reference counting code has been lifted, and modified from Liam Girdwood's earlier patch. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-29 15:58:08 +08:00
/* Must be always connected (for APLL) */
{"Virtual Voice OUT", NULL, "Digital Voice Playback Mixer"},
/* Physical outputs */
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
{"EARPIECE", NULL, "Earpiece PGA"},
{"PREDRIVEL", NULL, "PredriveL PGA"},
{"PREDRIVER", NULL, "PredriveR PGA"},
ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:05 +08:00
{"HSOL", NULL, "HeadsetL PGA"},
{"HSOR", NULL, "HeadsetR PGA"},
ASoC: TWL4030: Introduce PGAs for outputs Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 20:59:34 +08:00
{"CARKITL", NULL, "CarkitL PGA"},
{"CARKITR", NULL, "CarkitR PGA"},
ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-25 16:12:11 +08:00
{"HFL", NULL, "HandsfreeL PGA"},
{"HFR", NULL, "HandsfreeR PGA"},
ASoC: TWL4030: Add VIBRA output This patch adds support for the VIBRA output on TWL4030 codec. The VIBRA output can be driven with audio data or with local vibrator driver. Add the needed DAPM elements and routes for the VIBRA output and controls for the VIBRA driver configuration. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-05 13:55:47 +08:00
{"Vibra Route", "Audio", "Vibra Mux"},
{"VIBRA", NULL, "Vibra Route"},
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
/* Capture path */
ASoC: TWL4030: AIF/APLL fix in DAPM domain This patch orders the APLL and AIF power sequence in case of HiFi (audio in TWL4030 terms) playback/capture. We also need to make sure that the AIF is running during playback/capture, when there is no valid DAPM route available. For this purpose I introduce these virtual widgets: /* To have complete playback route all the time */ DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */ /* To have complete capture route all the time */ DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */ /* To have complete playback route for the voice module */ DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */ The DAPM_SUPPLY widgets for APLL and AIF are placed in a way, that during any audio activity the needed configuration of AIF and APLL will be enabled (playback, capture, analog loopback, digital loopback, and voice activity). The apll reference counting code has been lifted, and modified from Liam Girdwood's earlier patch. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-29 15:58:08 +08:00
/* Must be always connected (for AIF and APLL) */
{"ADC Virtual Left1", NULL, "Virtual HiFi IN"},
{"ADC Virtual Right1", NULL, "Virtual HiFi IN"},
{"ADC Virtual Left2", NULL, "Virtual HiFi IN"},
{"ADC Virtual Right2", NULL, "Virtual HiFi IN"},
/* Physical inputs */
ASoC: TWL4030: Fix for capture mixer strings Change the strings related to capture in order to be interpreted correctly by alsamixer and possible other UI based mixer applications. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 13:44:00 +08:00
{"Analog Left", "Main Mic Capture Switch", "MAINMIC"},
{"Analog Left", "Headset Mic Capture Switch", "HSMIC"},
{"Analog Left", "AUXL Capture Switch", "AUXL"},
{"Analog Left", "Carkit Mic Capture Switch", "CARKITMIC"},
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
ASoC: TWL4030: Fix for capture mixer strings Change the strings related to capture in order to be interpreted correctly by alsamixer and possible other UI based mixer applications. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 13:44:00 +08:00
{"Analog Right", "Sub Mic Capture Switch", "SUBMIC"},
{"Analog Right", "AUXR Capture Switch", "AUXR"},
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
ASoC: TWL4030: Fix for capture mixer strings Change the strings related to capture in order to be interpreted correctly by alsamixer and possible other UI based mixer applications. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 13:44:00 +08:00
{"ADC Physical Left", NULL, "Analog Left"},
{"ADC Physical Right", NULL, "Analog Right"},
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
{"Digimic0 Enable", NULL, "DIGIMIC0"},
{"Digimic1 Enable", NULL, "DIGIMIC1"},
ASoC: TWL4030: Capture route runtime DAPM ordering fix Fix the ordering problem in DAPM domain, when the user changes between digital and analog sources during active capture (or loopback) scenario. Before this patch, when the user changed from analog source to digital there were a short time, when the codec enabled analog mic bias (2.2 volts) instead of the correct digital mic bias (1.8 volts) to the digital microphones. This behaviour caused by the former implementation of selecting the correct type of bias. This was done at the POST_REG event of the DAPM_MUX_E("TXx Capture Route") widget. By moving the bias type selection as DAPM_SUPPLY and connecting it to the corresponding digimic widget the problematic situation can be avoided. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-08-03 17:01:01 +08:00
{"DIGIMIC0", NULL, "micbias1 select"},
{"DIGIMIC1", NULL, "micbias2 select"},
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
/* TX1 Left capture path */
ASoC: TWL4030: Physical ADC and amplifier power switch change Change the power switches for the physical ADC and for the amplifiers for the analog capture path to map more closely the actual path inside of the codec. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:42 +08:00
{"TX1 Capture Route", "Analog", "ADC Physical Left"},
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
{"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
/* TX1 Right capture path */
ASoC: TWL4030: Physical ADC and amplifier power switch change Change the power switches for the physical ADC and for the amplifiers for the analog capture path to map more closely the actual path inside of the codec. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:42 +08:00
{"TX1 Capture Route", "Analog", "ADC Physical Right"},
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
{"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
/* TX2 Left capture path */
ASoC: TWL4030: Physical ADC and amplifier power switch change Change the power switches for the physical ADC and for the amplifiers for the analog capture path to map more closely the actual path inside of the codec. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:42 +08:00
{"TX2 Capture Route", "Analog", "ADC Physical Left"},
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
{"TX2 Capture Route", "Digimic1", "Digimic1 Enable"},
/* TX2 Right capture path */
ASoC: TWL4030: Physical ADC and amplifier power switch change Change the power switches for the physical ADC and for the amplifiers for the analog capture path to map more closely the actual path inside of the codec. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-27 17:29:42 +08:00
{"TX2 Capture Route", "Analog", "ADC Physical Right"},
ASoC: TWL4030: DAPM based capture implementation This patch adds DAPM implementaion for the capture path on twlx030. TWL has two physical ADC and two digital microphone (stereo) connections. The CPU interface has four microphone channels. For simplicity the microphone channel paths are named as: TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid TX2 (Left/Right) Input routing (simplified version): There is two levels of mux settings for TWL in input path: Analog input mux: ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic} ADCR <- {Off, Sub mic, AUXR} Analog/Digital mux: TX1 Analog mode: TX1L <- ADCL TX1R <- ADCR TX1 Digital mode: TX1L <- Digimic0 (Left) TX1R <- Digimic0 (Right) TX2 Analog mode: TX2L <- ADCL TX2R <- ADCR TX2 Digital mode: TX2L <- Digimic1 (Left) TX2R <- Digimic1 (Right) The patch provides the following user controls for the capture path: Mux settings: "TX1 Capture Route": {Analog, Digimic0} "TX2 Capture Route": {Analog, Digimic1} "Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic} "Analog Right Capture Route": {Off, Sub Mic, AUXR} Volume/Gain controls: "TX1 Digital Capture Volume": Stereo gain control for TX1 path "TX2 Digital Capture Volume": Stereo gain control for TX2 path "Analog Capture Volume": Stereo gain control for the analog path only Important things for the board files: Microphone bias: "Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path) "Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path) "Headset Mic Bias": Bias for Headset mic When the routing configured correctly only the needed components will be powered/enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-31 16:08:38 +08:00
{"TX2 Capture Route", "Digimic1", "Digimic1 Enable"},
{"ADC Virtual Left1", NULL, "TX1 Capture Route"},
{"ADC Virtual Right1", NULL, "TX1 Capture Route"},
{"ADC Virtual Left2", NULL, "TX2 Capture Route"},
{"ADC Virtual Right2", NULL, "TX2 Capture Route"},
ASoC: TWL4030: Add supply for audio serial interface control The serial interface (TDM/I2S) for the audio block have been constantly enabled. Introduce a new DAPM_SUPPLY for handling the AIF_EN bit, so the interface is only enabled, when there is a need for it. For example when only the analog loopback is enabled, there is no need to keep the serial interface active. I have added the persons who contributed to the Voice path of twl4030 codec driver, so they might have the ability to test this patch, and send an update for the Voice path, if it is necessary Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-09 21:24:04 +08:00
{"ADC Virtual Left1", NULL, "AIF Enable"},
{"ADC Virtual Right1", NULL, "AIF Enable"},
{"ADC Virtual Left2", NULL, "AIF Enable"},
{"ADC Virtual Right2", NULL, "AIF Enable"},
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
/* Analog bypass routes */
ASoC: TWL4030: Fix for capture mixer strings Change the strings related to capture in order to be interpreted correctly by alsamixer and possible other UI based mixer applications. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 13:44:00 +08:00
{"Right1 Analog Loopback", "Switch", "Analog Right"},
{"Left1 Analog Loopback", "Switch", "Analog Left"},
{"Right2 Analog Loopback", "Switch", "Analog Right"},
{"Left2 Analog Loopback", "Switch", "Analog Left"},
{"Voice Analog Loopback", "Switch", "Analog Left"},
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
ASoC: TWL4030: Remove bypass tracking Since ASoC core now handling the codec bias differently there is no need to do the tracking of bypass switch states anymore. Handling of the common bit for analog loopbacks is done with DAPM_SUPPLY for the bypass paths. Now this bit is only enabled when there is a complete analog bypass path, compared to the previous implementation, when the global switch was enabled if there were any of the analog bypass switch was on (regardless if there were complete path or not) Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-28 16:57:04 +08:00
/* Supply for the Analog loopbacks */
{"Right1 Analog Loopback", NULL, "FM Loop Enable"},
{"Left1 Analog Loopback", NULL, "FM Loop Enable"},
{"Right2 Analog Loopback", NULL, "FM Loop Enable"},
{"Left2 Analog Loopback", NULL, "FM Loop Enable"},
{"Voice Analog Loopback", NULL, "FM Loop Enable"},
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
{"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"},
{"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"},
{"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"},
{"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"},
ASoC: TWL4030: Add VDL analog bypass This patch adds voice downlink analog bypass switch. It follows the same approach as in other analog bypass switches. DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer', that will also allow voice DAC to be powered in digital voice loopback (sidetone). Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-01 10:47:22 +08:00
{"Analog Voice Playback Mixer", NULL, "Voice Analog Loopback"},
ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-29 20:57:50 +08:00
ASoC: TWL4030: Add digital loopback support This patch adds the digital loopback/bypass support for twl4030 codec. The digital loopback will let the digimic0 (routed in the TX1 capture path inside of TWL4030) data to be routed back to the RX2 playback path (I2S stereo). It can also route the analog capture date routed through the TX1 back to RX2. Effectively the digital loopback is routing the audio from the TX1 capture path to the RX2 playback path. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 20:39:05 +08:00
/* Digital bypass routes */
{"Right Digital Loopback", "Volume", "TX1 Capture Route"},
{"Left Digital Loopback", "Volume", "TX1 Capture Route"},
ASoC: TWL4030: Add voice digital loopback: sidetone This patch add voice digital loopback (sidetone) to the twl4030 driver. It mixes voice uplink attenuated (by sidetone gain) with voice downlink when the codec is working in option2 (voice/audio mode). Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-01 10:48:08 +08:00
{"Voice Digital Loopback", "Volume", "TX2 Capture Route"},
ASoC: TWL4030: Add digital loopback support This patch adds the digital loopback/bypass support for twl4030 codec. The digital loopback will let the digimic0 (routed in the TX1 capture path inside of TWL4030) data to be routed back to the RX2 playback path (I2S stereo). It can also route the analog capture date routed through the TX1 back to RX2. Effectively the digital loopback is routing the audio from the TX1 capture path to the RX2 playback path. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 20:39:05 +08:00
ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 21:02:04 +08:00
{"Digital R2 Playback Mixer", NULL, "Right Digital Loopback"},
{"Digital L2 Playback Mixer", NULL, "Left Digital Loopback"},
{"Digital Voice Playback Mixer", NULL, "Voice Digital Loopback"},
ASoC: TWL4030: Add digital loopback support This patch adds the digital loopback/bypass support for twl4030 codec. The digital loopback will let the digimic0 (routed in the TX1 capture path inside of TWL4030) data to be routed back to the RX2 playback path (I2S stereo). It can also route the analog capture date routed through the TX1 back to RX2. Effectively the digital loopback is routing the audio from the TX1 capture path to the RX2 playback path. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-02-18 20:39:05 +08:00
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
};
static int twl4030_add_widgets(struct snd_soc_codec *codec)
{
snd_soc_dapm_new_controls(codec, twl4030_dapm_widgets,
ARRAY_SIZE(twl4030_dapm_widgets));
snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
return 0;
}
static int twl4030_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
ASoC: TWL4030: Remove bypass tracking Since ASoC core now handling the codec bias differently there is no need to do the tracking of bypass switch states anymore. Handling of the common bit for analog loopbacks is done with DAPM_SUPPLY for the bypass paths. Now this bit is only enabled when there is a complete analog bypass path, compared to the previous implementation, when the global switch was enabled if there were any of the analog bypass switch was on (regardless if there were complete path or not) Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-28 16:57:04 +08:00
if (codec->bias_level == SND_SOC_BIAS_OFF)
ASoC: TWL4030: Optimize the power up sequence Since the reg cache now contains the chip default values for all registers (REG_OPTION is reset to it's default within this patch), there is no longer need to rewrite _all_ registers. Initialize only few selected registers. According to the latest information, the offset cancellation need to be done only once, when the codec is powered on, so there is no need for the power up wrapper. Move all chip initialization code under chip_init, and do it when the codec is initialized. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:17 +08:00
twl4030_codec_enable(codec, 1);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
break;
case SND_SOC_BIAS_OFF:
ASoC: TWL4030: Remove wrapper for power down There is no need for the power down wrapper. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:15 +08:00
twl4030_codec_enable(codec, 0);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
break;
}
codec->bias_level = level;
return 0;
}
ASoC: TWL4030: Fix for the constraint handling The original implementation of the constraints were good against sane applications. If the opening sequence is: stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the constraints are set correctly for stream2. But if the sequence is: stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2 would receive constraint rate = 0, sample_bits = 0, since the stream1 has not yet called hw_params... The command to trigger this event: gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false This patch does some 'black magic' in order to always set the correct constraints and sets it only when it is needed for the other stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-17 20:55:08 +08:00
static void twl4030_constraints(struct twl4030_priv *twl4030,
struct snd_pcm_substream *mst_substream)
{
struct snd_pcm_substream *slv_substream;
/* Pick the stream, which need to be constrained */
if (mst_substream == twl4030->master_substream)
slv_substream = twl4030->slave_substream;
else if (mst_substream == twl4030->slave_substream)
slv_substream = twl4030->master_substream;
else /* This should not happen.. */
return;
/* Set the constraints according to the already configured stream */
snd_pcm_hw_constraint_minmax(slv_substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
twl4030->rate,
twl4030->rate);
snd_pcm_hw_constraint_minmax(slv_substream->runtime,
SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
twl4030->sample_bits,
twl4030->sample_bits);
snd_pcm_hw_constraint_minmax(slv_substream->runtime,
SNDRV_PCM_HW_PARAM_CHANNELS,
twl4030->channels,
twl4030->channels);
}
ASoC: TWL4030: Add 4 channel TDM support Support for 4 channel TDM (SND_SOC_DAIFMT_DSP_A) for twl4030 codec. The channel allocations are: Playback: TDM i2s TWL RX Channel 1 Left SDRL2 Channel 3 Right SDRR2 Channel 2 -- SDRL1 Channel 4 -- SDRR1 Capture: TDM i2s TWL TX Channel 1 Left TXL1 Channel 3 Right TXR1 Channel 2 -- TXL2 Channel 4 -- TXR2 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-23 19:36:49 +08:00
/* In case of 4 channel mode, the RX1 L/R for playback and the TX2 L/R for
* capture has to be enabled/disabled. */
static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction,
int enable)
{
u8 reg, mask;
reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION);
if (direction == SNDRV_PCM_STREAM_PLAYBACK)
mask = TWL4030_ARXL1_VRX_EN | TWL4030_ARXR1_EN;
else
mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN;
if (enable)
reg |= mask;
else
reg &= ~mask;
twl4030_write(codec, TWL4030_REG_OPTION, reg);
}
ASoC: TWL4030: Compillation error fix Fix for compillation error introduced by the constrain patch. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-07 14:14:00 +08:00
static int twl4030_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
ASoC: TWL4030: Add constrains for second stream In case of duplex mode (capture and playback at the same time), the second stream has to have the same parameters (rate, sample size) as the already running stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-27 16:39:08 +08:00
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
struct snd_soc_codec *codec = rtd->codec;
ASoC: Add indirection for CODEC private data One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-14 14:35:19 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: TWL4030: Add constrains for second stream In case of duplex mode (capture and playback at the same time), the second stream has to have the same parameters (rate, sample size) as the already running stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-27 16:39:08 +08:00
if (twl4030->master_substream) {
twl4030->slave_substream = substream;
ASoC: TWL4030: Fix for the constraint handling The original implementation of the constraints were good against sane applications. If the opening sequence is: stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the constraints are set correctly for stream2. But if the sequence is: stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2 would receive constraint rate = 0, sample_bits = 0, since the stream1 has not yet called hw_params... The command to trigger this event: gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false This patch does some 'black magic' in order to always set the correct constraints and sets it only when it is needed for the other stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-17 20:55:08 +08:00
/* The DAI has one configuration for playback and capture, so
* if the DAI has been already configured then constrain this
* substream to match it. */
if (twl4030->configured)
twl4030_constraints(twl4030, twl4030->master_substream);
} else {
ASoC: TWL4030: Add 4 channel TDM support Support for 4 channel TDM (SND_SOC_DAIFMT_DSP_A) for twl4030 codec. The channel allocations are: Playback: TDM i2s TWL RX Channel 1 Left SDRL2 Channel 3 Right SDRR2 Channel 2 -- SDRL1 Channel 4 -- SDRR1 Capture: TDM i2s TWL TX Channel 1 Left TXL1 Channel 3 Right TXR1 Channel 2 -- TXL2 Channel 4 -- TXR2 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-23 19:36:49 +08:00
if (!(twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) &
TWL4030_OPTION_1)) {
/* In option2 4 channel is not supported, set the
* constraint for the first stream for channels, the
* second stream will 'inherit' this cosntraint */
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_CHANNELS,
2, 2);
}
ASoC: TWL4030: Add constrains for second stream In case of duplex mode (capture and playback at the same time), the second stream has to have the same parameters (rate, sample size) as the already running stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-27 16:39:08 +08:00
twl4030->master_substream = substream;
ASoC: TWL4030: Fix for the constraint handling The original implementation of the constraints were good against sane applications. If the opening sequence is: stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the constraints are set correctly for stream2. But if the sequence is: stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2 would receive constraint rate = 0, sample_bits = 0, since the stream1 has not yet called hw_params... The command to trigger this event: gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false This patch does some 'black magic' in order to always set the correct constraints and sets it only when it is needed for the other stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-17 20:55:08 +08:00
}
ASoC: TWL4030: Add constrains for second stream In case of duplex mode (capture and playback at the same time), the second stream has to have the same parameters (rate, sample size) as the already running stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-27 16:39:08 +08:00
return 0;
}
ASoC: TWL4030: Compillation error fix Fix for compillation error introduced by the constrain patch. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-07 14:14:00 +08:00
static void twl4030_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
ASoC: TWL4030: Add constrains for second stream In case of duplex mode (capture and playback at the same time), the second stream has to have the same parameters (rate, sample size) as the already running stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-27 16:39:08 +08:00
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
struct snd_soc_codec *codec = rtd->codec;
ASoC: Add indirection for CODEC private data One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-14 14:35:19 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: TWL4030: Add constrains for second stream In case of duplex mode (capture and playback at the same time), the second stream has to have the same parameters (rate, sample size) as the already running stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-27 16:39:08 +08:00
if (twl4030->master_substream == substream)
twl4030->master_substream = twl4030->slave_substream;
twl4030->slave_substream = NULL;
ASoC: TWL4030: Fix for the constraint handling The original implementation of the constraints were good against sane applications. If the opening sequence is: stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the constraints are set correctly for stream2. But if the sequence is: stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2 would receive constraint rate = 0, sample_bits = 0, since the stream1 has not yet called hw_params... The command to trigger this event: gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false This patch does some 'black magic' in order to always set the correct constraints and sets it only when it is needed for the other stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-17 20:55:08 +08:00
/* If all streams are closed, or the remaining stream has not yet
* been configured than set the DAI as not configured. */
if (!twl4030->master_substream)
twl4030->configured = 0;
else if (!twl4030->master_substream->runtime->channels)
twl4030->configured = 0;
ASoC: TWL4030: Add 4 channel TDM support Support for 4 channel TDM (SND_SOC_DAIFMT_DSP_A) for twl4030 codec. The channel allocations are: Playback: TDM i2s TWL RX Channel 1 Left SDRL2 Channel 3 Right SDRR2 Channel 2 -- SDRL1 Channel 4 -- SDRR1 Capture: TDM i2s TWL TX Channel 1 Left TXL1 Channel 3 Right TXR1 Channel 2 -- TXL2 Channel 4 -- TXR2 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-23 19:36:49 +08:00
/* If the closing substream had 4 channel, do the necessary cleanup */
if (substream->runtime->channels == 4)
twl4030_tdm_enable(codec, substream->stream, 0);
ASoC: TWL4030: Add constrains for second stream In case of duplex mode (capture and playback at the same time), the second stream has to have the same parameters (rate, sample size) as the already running stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-27 16:39:08 +08:00
}
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
static int twl4030_hw_params(struct snd_pcm_substream *substream,
ASoC: Merge snd_soc_ops into snd_soc_dai_ops Liam Girdwood's ASoC v2 work avoids having two different ops structures for DAIs by merging the members of struct snd_soc_ops into struct snd_soc_dai_ops, allowing per DAI configuration for everything. Backport this change. This paves the way for future work allowing any combination of DAIs to be connected rather than having fixed purpose CODEC and CPU DAIs and only allowing CODEC<->CPU interconnections. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-19 06:11:38 +08:00
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
struct snd_soc_codec *codec = rtd->codec;
ASoC: Add indirection for CODEC private data One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-14 14:35:19 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
u8 mode, old_mode, format, old_format;
ASoC: TWL4030: Add 4 channel TDM support Support for 4 channel TDM (SND_SOC_DAIFMT_DSP_A) for twl4030 codec. The channel allocations are: Playback: TDM i2s TWL RX Channel 1 Left SDRL2 Channel 3 Right SDRR2 Channel 2 -- SDRL1 Channel 4 -- SDRR1 Capture: TDM i2s TWL TX Channel 1 Left TXL1 Channel 3 Right TXR1 Channel 2 -- TXL2 Channel 4 -- TXR2 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-23 19:36:49 +08:00
/* If the substream has 4 channel, do the necessary setup */
if (params_channels(params) == 4) {
ASoC: TWL4030: Check the interface format for 4 channel mode In addition to the operating mode check, also check the codec's interface format in case of four channel mode. If the codec is not in TDM (DSP_A) mode, return with error. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-01 19:06:40 +08:00
format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
/* Safety check: are we in the correct operating mode and
* the interface is in TDM mode? */
if ((mode & TWL4030_OPTION_1) &&
((format & TWL4030_AIF_FORMAT) == TWL4030_AIF_FORMAT_TDM))
ASoC: TWL4030: Add 4 channel TDM support Support for 4 channel TDM (SND_SOC_DAIFMT_DSP_A) for twl4030 codec. The channel allocations are: Playback: TDM i2s TWL RX Channel 1 Left SDRL2 Channel 3 Right SDRR2 Channel 2 -- SDRL1 Channel 4 -- SDRR1 Capture: TDM i2s TWL TX Channel 1 Left TXL1 Channel 3 Right TXR1 Channel 2 -- TXL2 Channel 4 -- TXR2 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-23 19:36:49 +08:00
twl4030_tdm_enable(codec, substream->stream, 1);
else
return -EINVAL;
}
ASoC: TWL4030: Fix for the constraint handling The original implementation of the constraints were good against sane applications. If the opening sequence is: stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the constraints are set correctly for stream2. But if the sequence is: stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2 would receive constraint rate = 0, sample_bits = 0, since the stream1 has not yet called hw_params... The command to trigger this event: gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false This patch does some 'black magic' in order to always set the correct constraints and sets it only when it is needed for the other stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-17 20:55:08 +08:00
if (twl4030->configured)
/* Ignoring hw_params for already configured DAI */
ASoC: TWL4030: Add constrains for second stream In case of duplex mode (capture and playback at the same time), the second stream has to have the same parameters (rate, sample size) as the already running stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-27 16:39:08 +08:00
return 0;
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
/* bit rate */
old_mode = twl4030_read_reg_cache(codec,
TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
mode = old_mode & ~TWL4030_APLL_RATE;
switch (params_rate(params)) {
case 8000:
mode |= TWL4030_APLL_RATE_8000;
break;
case 11025:
mode |= TWL4030_APLL_RATE_11025;
break;
case 12000:
mode |= TWL4030_APLL_RATE_12000;
break;
case 16000:
mode |= TWL4030_APLL_RATE_16000;
break;
case 22050:
mode |= TWL4030_APLL_RATE_22050;
break;
case 24000:
mode |= TWL4030_APLL_RATE_24000;
break;
case 32000:
mode |= TWL4030_APLL_RATE_32000;
break;
case 44100:
mode |= TWL4030_APLL_RATE_44100;
break;
case 48000:
mode |= TWL4030_APLL_RATE_48000;
break;
ASoC: TWL4030: Add actual support for 96KHz playback support Adds the needed code to be able to use 96KHz playback. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-03 19:39:05 +08:00
case 96000:
mode |= TWL4030_APLL_RATE_96000;
break;
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
default:
printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n",
params_rate(params));
return -EINVAL;
}
/* sample size */
old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
format = old_format;
format &= ~TWL4030_DATA_WIDTH;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
format |= TWL4030_DATA_WIDTH_16S_16W;
break;
case SNDRV_PCM_FORMAT_S24_LE:
format |= TWL4030_DATA_WIDTH_32S_24W;
break;
default:
printk(KERN_ERR "TWL4030 hw params: unknown format %d\n",
params_format(params));
return -EINVAL;
}
ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use Restructure the codec power code in order to be able to hit off when the codec is not in use. Since the audio registers are accessible while the codec is powered down, there is no need for additional safety mechanism. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:20 +08:00
if (format != old_format || mode != old_mode) {
if (twl4030->codec_powered) {
/*
* If the codec is powered, than we need to toggle the
* codec power.
*/
twl4030_codec_enable(codec, 0);
twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
twl4030_codec_enable(codec, 1);
} else {
twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
}
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
}
ASoC: TWL4030: Fix for the constraint handling The original implementation of the constraints were good against sane applications. If the opening sequence is: stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the constraints are set correctly for stream2. But if the sequence is: stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2 would receive constraint rate = 0, sample_bits = 0, since the stream1 has not yet called hw_params... The command to trigger this event: gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false This patch does some 'black magic' in order to always set the correct constraints and sets it only when it is needed for the other stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-17 20:55:08 +08:00
/* Store the important parameters for the DAI configuration and set
* the DAI as configured */
twl4030->configured = 1;
twl4030->rate = params_rate(params);
twl4030->sample_bits = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min;
twl4030->channels = params_channels(params);
/* If both playback and capture streams are open, and one of them
* is setting the hw parameters right now (since we are here), set
* constraints to the other stream to match the current one. */
if (twl4030->slave_substream)
twl4030_constraints(twl4030, substream);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
return 0;
}
static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
ASoC: Add indirection for CODEC private data One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-14 14:35:19 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
switch (freq) {
case 19200000:
case 26000000:
case 38400000:
break;
default:
ASoC: TWL4030: Do not modify the APLL_CTL register APLL_CTL register is configured by the twl4030-codec MFD driver. Remove code, which makes changes in the APLL_CTL register, and replace those with checks against the configured audio_mclk configuration done in the MFD driver. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-11-04 15:58:20 +08:00
dev_err(codec->dev, "Unsupported APLL mclk: %u\n", freq);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
return -EINVAL;
}
ASoC: TWL4030: Do not modify the APLL_CTL register APLL_CTL register is configured by the twl4030-codec MFD driver. Remove code, which makes changes in the APLL_CTL register, and replace those with checks against the configured audio_mclk configuration done in the MFD driver. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-11-04 15:58:20 +08:00
if ((freq / 1000) != twl4030->sysclk) {
dev_err(codec->dev,
"Mismatch in APLL mclk: %u (configured: %u)\n",
freq, twl4030->sysclk * 1000);
return -EINVAL;
}
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
return 0;
}
static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use Restructure the codec power code in order to be able to hit off when the codec is not in use. Since the audio registers are accessible while the codec is powered down, there is no need for additional safety mechanism. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:20 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
u8 old_format, format;
/* get format */
old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
format = old_format;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
format &= ~(TWL4030_AIF_SLAVE_EN);
ALSA: ASoC: TWL4030 codec - fix 256*Fs clock According to TRM, 256*Fs clock output should be enabled when TWL4030 is in slave mode, not master. This allows sound to work on OMAP3 Pandora, which uses 256*Fs clock. Signed-off-by: Grazvydas Ignotas <notasas@gmail.com> Acked-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-06 05:51:05 +08:00
format &= ~(TWL4030_CLK256FS_EN);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
break;
case SND_SOC_DAIFMT_CBS_CFS:
format |= TWL4030_AIF_SLAVE_EN;
ALSA: ASoC: TWL4030 codec - fix 256*Fs clock According to TRM, 256*Fs clock output should be enabled when TWL4030 is in slave mode, not master. This allows sound to work on OMAP3 Pandora, which uses 256*Fs clock. Signed-off-by: Grazvydas Ignotas <notasas@gmail.com> Acked-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-06 05:51:05 +08:00
format |= TWL4030_CLK256FS_EN;
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
break;
default:
return -EINVAL;
}
/* interface format */
format &= ~TWL4030_AIF_FORMAT;
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
format |= TWL4030_AIF_FORMAT_CODEC;
break;
ASoC: TWL4030: Add 4 channel TDM support Support for 4 channel TDM (SND_SOC_DAIFMT_DSP_A) for twl4030 codec. The channel allocations are: Playback: TDM i2s TWL RX Channel 1 Left SDRL2 Channel 3 Right SDRR2 Channel 2 -- SDRL1 Channel 4 -- SDRR1 Capture: TDM i2s TWL TX Channel 1 Left TXL1 Channel 3 Right TXR1 Channel 2 -- TXL2 Channel 4 -- TXR2 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-23 19:36:49 +08:00
case SND_SOC_DAIFMT_DSP_A:
format |= TWL4030_AIF_FORMAT_TDM;
break;
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
default:
return -EINVAL;
}
if (format != old_format) {
ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use Restructure the codec power code in order to be able to hit off when the codec is not in use. Since the audio registers are accessible while the codec is powered down, there is no need for additional safety mechanism. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:20 +08:00
if (twl4030->codec_powered) {
/*
* If the codec is powered, than we need to toggle the
* codec power.
*/
twl4030_codec_enable(codec, 0);
twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
twl4030_codec_enable(codec, 1);
} else {
twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
}
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
}
return 0;
}
ASoC: TWL4030: Add tristate callbacks for HiFi and Voice Add "set_tristate" callbacks for HiFi and Voice DAIs. Machine drivers can enable and disable tristate for each DAI with "snd_soc_dai_set_tristate" function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-03 15:21:39 +08:00
static int twl4030_set_tristate(struct snd_soc_dai *dai, int tristate)
{
struct snd_soc_codec *codec = dai->codec;
u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
if (tristate)
reg |= TWL4030_AIF_TRI_EN;
else
reg &= ~TWL4030_AIF_TRI_EN;
return twl4030_write(codec, TWL4030_REG_AUDIO_IF, reg);
}
ASoC: TWL4030: Enable/disable voice digital filters Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink and downlink paths, respectively. This patch also corrects voice 8/16kHz mode selection bit (SEL_16K) of CODEC_MODE register. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 09:02:31 +08:00
/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R
* (VTXL, VTXR) for uplink has to be enabled/disabled. */
static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction,
int enable)
{
u8 reg, mask;
reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION);
if (direction == SNDRV_PCM_STREAM_PLAYBACK)
mask = TWL4030_ARXL1_VRX_EN;
else
mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN;
if (enable)
reg |= mask;
else
reg &= ~mask;
twl4030_write(codec, TWL4030_REG_OPTION, reg);
}
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
static int twl4030_voice_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
struct snd_soc_codec *codec = rtd->codec;
ASoC: Add indirection for CODEC private data One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-04-14 14:35:19 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
u8 mode;
/* If the system master clock is not 26MHz, the voice PCM interface is
* not avilable.
*/
ASoC: TWL4030: Do not modify the APLL_CTL register APLL_CTL register is configured by the twl4030-codec MFD driver. Remove code, which makes changes in the APLL_CTL register, and replace those with checks against the configured audio_mclk configuration done in the MFD driver. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-11-04 15:58:20 +08:00
if (twl4030->sysclk != 26000) {
dev_err(codec->dev, "The board is configured for %u Hz, while"
"the Voice interface needs 26MHz APLL mclk\n",
twl4030->sysclk * 1000);
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
return -EINVAL;
}
/* If the codec mode is not option2, the voice PCM interface is not
* avilable.
*/
mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
& TWL4030_OPT_MODE;
if (mode != TWL4030_OPTION_2) {
printk(KERN_ERR "TWL4030 voice startup: "
"the codec mode is not option2\n");
return -EINVAL;
}
return 0;
}
ASoC: TWL4030: Enable/disable voice digital filters Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink and downlink paths, respectively. This patch also corrects voice 8/16kHz mode selection bit (SEL_16K) of CODEC_MODE register. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 09:02:31 +08:00
static void twl4030_voice_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
struct snd_soc_codec *codec = rtd->codec;
ASoC: TWL4030: Enable/disable voice digital filters Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink and downlink paths, respectively. This patch also corrects voice 8/16kHz mode selection bit (SEL_16K) of CODEC_MODE register. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 09:02:31 +08:00
/* Enable voice digital filters */
twl4030_voice_enable(codec, substream->stream, 0);
}
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
struct snd_soc_codec *codec = rtd->codec;
ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use Restructure the codec power code in order to be able to hit off when the codec is not in use. Since the audio registers are accessible while the codec is powered down, there is no need for additional safety mechanism. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:20 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
u8 old_mode, mode;
ASoC: TWL4030: Enable/disable voice digital filters Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink and downlink paths, respectively. This patch also corrects voice 8/16kHz mode selection bit (SEL_16K) of CODEC_MODE register. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 09:02:31 +08:00
/* Enable voice digital filters */
twl4030_voice_enable(codec, substream->stream, 1);
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
/* bit rate */
old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
& ~(TWL4030_CODECPDZ);
mode = old_mode;
switch (params_rate(params)) {
case 8000:
mode &= ~(TWL4030_SEL_16K);
break;
case 16000:
mode |= TWL4030_SEL_16K;
break;
default:
printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n",
params_rate(params));
return -EINVAL;
}
if (mode != old_mode) {
ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use Restructure the codec power code in order to be able to hit off when the codec is not in use. Since the audio registers are accessible while the codec is powered down, there is no need for additional safety mechanism. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:20 +08:00
if (twl4030->codec_powered) {
/*
* If the codec is powered, than we need to toggle the
* codec power.
*/
twl4030_codec_enable(codec, 0);
twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
twl4030_codec_enable(codec, 1);
} else {
twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
}
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
}
return 0;
}
static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
ASoC: missing conversions to snd_soc_codec_*_drvdata() Conversions to snd_soc_codec_{get|set}_drvdata() were missing in some files in the previous commit. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-04-20 14:20:31 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
ASoC: TWL4030: Do not modify the APLL_CTL register APLL_CTL register is configured by the twl4030-codec MFD driver. Remove code, which makes changes in the APLL_CTL register, and replace those with checks against the configured audio_mclk configuration done in the MFD driver. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-11-04 15:58:20 +08:00
if (freq != 26000000) {
dev_err(codec->dev, "Unsupported APLL mclk: %u, the Voice"
"interface needs 26MHz APLL mclk\n", freq);
return -EINVAL;
}
if ((freq / 1000) != twl4030->sysclk) {
dev_err(codec->dev,
"Mismatch in APLL mclk: %u (configured: %u)\n",
freq, twl4030->sysclk * 1000);
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
return -EINVAL;
}
return 0;
}
static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use Restructure the codec power code in order to be able to hit off when the codec is not in use. Since the audio registers are accessible while the codec is powered down, there is no need for additional safety mechanism. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:20 +08:00
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
u8 old_format, format;
/* get format */
old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF);
format = old_format;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
ASoC: TWL4030: Fix voice interface clock masters Voice interface of twl4030 codec supports: CBM_CFM and CBS_CFS. It doesn't support CBS_CFM. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-By: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-19 16:23:42 +08:00
case SND_SOC_DAIFMT_CBM_CFM:
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
format &= ~(TWL4030_VIF_SLAVE_EN);
break;
case SND_SOC_DAIFMT_CBS_CFS:
format |= TWL4030_VIF_SLAVE_EN;
break;
default:
return -EINVAL;
}
/* clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_IB_NF:
format &= ~(TWL4030_VIF_FORMAT);
break;
case SND_SOC_DAIFMT_NB_IF:
format |= TWL4030_VIF_FORMAT;
break;
default:
return -EINVAL;
}
if (format != old_format) {
ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use Restructure the codec power code in order to be able to hit off when the codec is not in use. Since the audio registers are accessible while the codec is powered down, there is no need for additional safety mechanism. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-26 16:38:20 +08:00
if (twl4030->codec_powered) {
/*
* If the codec is powered, than we need to toggle the
* codec power.
*/
twl4030_codec_enable(codec, 0);
twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
twl4030_codec_enable(codec, 1);
} else {
twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
}
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
}
return 0;
}
ASoC: TWL4030: Add tristate callbacks for HiFi and Voice Add "set_tristate" callbacks for HiFi and Voice DAIs. Machine drivers can enable and disable tristate for each DAI with "snd_soc_dai_set_tristate" function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-03 15:21:39 +08:00
static int twl4030_voice_set_tristate(struct snd_soc_dai *dai, int tristate)
{
struct snd_soc_codec *codec = dai->codec;
u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF);
if (tristate)
reg |= TWL4030_VIF_TRI_EN;
else
reg &= ~TWL4030_VIF_TRI_EN;
return twl4030_write(codec, TWL4030_REG_VOICE_IF, reg);
}
ASoC: Fix supported sample rates of TWL4030 audio codec TWL4030 currently supports rates between 8 kHz and 48 kHz and sets the codec mode register accordingly in twl4030_hw_params. Expose this info so that ASoC can match other rates than 44.1 kHz or 48 kHz as well. Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com> Acked-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-12 23:05:41 +08:00
#define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000)
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static struct snd_soc_dai_ops twl4030_dai_hifi_ops = {
ASoC: TWL4030: Add constrains for second stream In case of duplex mode (capture and playback at the same time), the second stream has to have the same parameters (rate, sample size) as the already running stream. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-27 16:39:08 +08:00
.startup = twl4030_startup,
.shutdown = twl4030_shutdown,
ASoC: twl4030 - Fix build error CC sound/soc/codecs/twl4030.o sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops') Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-16 20:23:35 +08:00
.hw_params = twl4030_hw_params,
.set_sysclk = twl4030_set_dai_sysclk,
.set_fmt = twl4030_set_dai_fmt,
ASoC: TWL4030: Add tristate callbacks for HiFi and Voice Add "set_tristate" callbacks for HiFi and Voice DAIs. Machine drivers can enable and disable tristate for each DAI with "snd_soc_dai_set_tristate" function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-03 15:21:39 +08:00
.set_tristate = twl4030_set_tristate,
ASoC: twl4030 - Fix build error CC sound/soc/codecs/twl4030.o sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops') Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-16 20:23:35 +08:00
};
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
static struct snd_soc_dai_ops twl4030_dai_voice_ops = {
.startup = twl4030_voice_startup,
ASoC: TWL4030: Enable/disable voice digital filters Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink and downlink paths, respectively. This patch also corrects voice 8/16kHz mode selection bit (SEL_16K) of CODEC_MODE register. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 09:02:31 +08:00
.shutdown = twl4030_voice_shutdown,
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
.hw_params = twl4030_voice_hw_params,
.set_sysclk = twl4030_voice_set_dai_sysclk,
.set_fmt = twl4030_voice_set_dai_fmt,
ASoC: TWL4030: Add tristate callbacks for HiFi and Voice Add "set_tristate" callbacks for HiFi and Voice DAIs. Machine drivers can enable and disable tristate for each DAI with "snd_soc_dai_set_tristate" function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-03 15:21:39 +08:00
.set_tristate = twl4030_voice_set_tristate,
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
};
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static struct snd_soc_dai_driver twl4030_dai[] = {
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
.name = "twl4030-hifi",
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
.playback = {
ASoC: TWL4030: Differentiate the playback streams Give unique stream names for the two playback streams so DAPM can figure out which codec_dai is in use. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-22 20:12:15 +08:00
.stream_name = "HiFi Playback",
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
.channels_min = 2,
ASoC: TWL4030: Add 4 channel TDM support Support for 4 channel TDM (SND_SOC_DAIFMT_DSP_A) for twl4030 codec. The channel allocations are: Playback: TDM i2s TWL RX Channel 1 Left SDRL2 Channel 3 Right SDRR2 Channel 2 -- SDRL1 Channel 4 -- SDRR1 Capture: TDM i2s TWL TX Channel 1 Left TXL1 Channel 3 Right TXR1 Channel 2 -- TXL2 Channel 4 -- TXR2 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-23 19:36:49 +08:00
.channels_max = 4,
ASoC: TWL4030: 96KHz playback support TWL4030 supports 96KHz sample playback, but only playback. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-27 16:39:07 +08:00
.rates = TWL4030_RATES | SNDRV_PCM_RATE_96000,
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
.formats = TWL4030_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
ASoC: TWL4030: Add 4 channel TDM support Support for 4 channel TDM (SND_SOC_DAIFMT_DSP_A) for twl4030 codec. The channel allocations are: Playback: TDM i2s TWL RX Channel 1 Left SDRL2 Channel 3 Right SDRR2 Channel 2 -- SDRL1 Channel 4 -- SDRR1 Capture: TDM i2s TWL TX Channel 1 Left TXL1 Channel 3 Right TXR1 Channel 2 -- TXL2 Channel 4 -- TXR2 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-23 19:36:49 +08:00
.channels_max = 4,
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
.rates = TWL4030_RATES,
.formats = TWL4030_FORMATS,},
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
.ops = &twl4030_dai_hifi_ops,
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
},
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
.name = "twl4030-voice",
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
.playback = {
ASoC: TWL4030: Differentiate the playback streams Give unique stream names for the two playback streams so DAPM can figure out which codec_dai is in use. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-22 20:12:15 +08:00
.stream_name = "Voice Playback",
ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:21:35 +08:00
.channels_min = 1,
.channels_max = 1,
.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.ops = &twl4030_dai_voice_ops,
},
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
};
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static int twl4030_soc_suspend(struct snd_soc_codec *codec, pm_message_t state)
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
{
twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static int twl4030_soc_resume(struct snd_soc_codec *codec)
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
{
twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static int twl4030_soc_probe(struct snd_soc_codec *codec)
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
struct twl4030_priv *twl4030;
ASoC: TWL4030: Add support for platform dependent configuration twl4030_setup_data structure can be passed from platform drivers to the codec via the snd_soc_device->codec_data pointer. Currently the setup data has support for the Headset pop-removal related configuration, which differs from board to board. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-22 15:13:15 +08:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL);
if (twl4030 == NULL) {
printk("Can not allocate memroy\n");
return -ENOMEM;
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
snd_soc_codec_set_drvdata(codec, twl4030);
/* Set the defaults, and power up the codec */
twl4030->sysclk = twl4030_codec_get_mclk() / 1000;
codec->idle_bias_off = 1;
twl4030_init_chip(codec);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
ASoC: cleanup duplicated code. Many codec drivers were implementing cookie-cutter copies of the function that adds kcontrols to the codec. This patch moves this code to a common function snd_soc_add_controls() in soc-core.c and updates all drivers using copies of this function to use the new common version. [Edited to raise priority of error log message and document parameters. -- broonie] Signed-off-by: Ian Molton <ian@mnementh.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-09 08:23:21 +08:00
snd_soc_add_controls(codec, twl4030_snd_controls,
ARRAY_SIZE(twl4030_snd_controls));
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
twl4030_add_widgets(codec);
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
return 0;
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static int twl4030_soc_remove(struct snd_soc_codec *codec)
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
{
ASoC: multi-component: TWL4030: Restore registers on removal Add back the register restore call, when the codec driver is removed. This does not affect normal operation, but it is usefull when debugging audio through the twl4030 class codecs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-08-12 14:29:52 +08:00
/* Reset registers to their chip default before leaving */
twl4030_reset_registers(codec);
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static struct snd_soc_codec_driver soc_codec_dev_twl4030 = {
.probe = twl4030_soc_probe,
.remove = twl4030_soc_remove,
.suspend = twl4030_soc_suspend,
.resume = twl4030_soc_resume,
.read = twl4030_read_reg_cache,
.write = twl4030_write,
.set_bias_level = twl4030_set_bias_level,
.reg_cache_size = sizeof(twl4030_reg),
.reg_word_size = sizeof(u8),
.reg_cache_default = twl4030_reg,
};
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
static int __devinit twl4030_codec_probe(struct platform_device *pdev)
{
struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data;
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
ASoC: TWL4030: Do not modify the APLL_CTL register APLL_CTL register is configured by the twl4030-codec MFD driver. Remove code, which makes changes in the APLL_CTL register, and replace those with checks against the configured audio_mclk configuration done in the MFD driver. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-11-04 15:58:20 +08:00
if (!pdata) {
dev_err(&pdev->dev, "platform_data is missing\n");
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
return -EINVAL;
}
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_twl4030,
twl4030_dai, ARRAY_SIZE(twl4030_dai));
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
}
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
static int __devexit twl4030_codec_remove(struct platform_device *pdev)
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
struct twl4030_priv *twl4030 = dev_get_drvdata(&pdev->dev);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
snd_soc_unregister_codec(&pdev->dev);
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
kfree(twl4030);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
return 0;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
MODULE_ALIAS("platform:twl4030-codec");
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
static struct platform_driver twl4030_codec_driver = {
.probe = twl4030_codec_probe,
.remove = __devexit_p(twl4030_codec_remove),
.driver = {
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
.name = "twl4030-codec",
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
.owner = THIS_MODULE,
},
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
};
ALSA: ASoC - Fix module init entry for twl4030.c Fixed the function name of module init entry for twl4030.c, which conflicted with the existing hardware init function: sound/soc/codecs/twl4030.c:1278: error: conflicting types for 'twl4030_init' sound/soc/codecs/twl4030.c:1187: error: previous definition of 'twl4030_init' was here Also fixed the section type of init function. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-10 14:40:24 +08:00
static int __init twl4030_modinit(void)
ASoC: Register non-AC97 codec DAIs Currently this is done at module probe time since ASoC ties in codec device probe to the instantiation of the entire ASoC device. Subsequent patches will refactor the codec drivers to handle probing separately. Note that the core does not yet use this information. AC97 is special since the codec is controlled over the AC97 link but we want to give the machine driver a chance to set up the system before trying to instantiate since it may need to do configuration before the AC97 link will operate Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 03:17:58 +08:00
{
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
return platform_driver_register(&twl4030_codec_driver);
ASoC: Register non-AC97 codec DAIs Currently this is done at module probe time since ASoC ties in codec device probe to the instantiation of the entire ASoC device. Subsequent patches will refactor the codec drivers to handle probing separately. Note that the core does not yet use this information. AC97 is special since the codec is controlled over the AC97 link but we want to give the machine driver a chance to set up the system before trying to instantiate since it may need to do configuration before the AC97 link will operate Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 03:17:58 +08:00
}
ALSA: ASoC - Fix module init entry for twl4030.c Fixed the function name of module init entry for twl4030.c, which conflicted with the existing hardware init function: sound/soc/codecs/twl4030.c:1278: error: conflicting types for 'twl4030_init' sound/soc/codecs/twl4030.c:1187: error: previous definition of 'twl4030_init' was here Also fixed the section type of init function. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-10 14:40:24 +08:00
module_init(twl4030_modinit);
ASoC: Register non-AC97 codec DAIs Currently this is done at module probe time since ASoC ties in codec device probe to the instantiation of the entire ASoC device. Subsequent patches will refactor the codec drivers to handle probing separately. Note that the core does not yet use this information. AC97 is special since the codec is controlled over the AC97 link but we want to give the machine driver a chance to set up the system before trying to instantiate since it may need to do configuration before the AC97 link will operate Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 03:17:58 +08:00
static void __exit twl4030_exit(void)
{
ASoC: TWL4030: Driver registration via twl4030_codec MFD Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-22 18:26:48 +08:00
platform_driver_unregister(&twl4030_codec_driver);
ASoC: Register non-AC97 codec DAIs Currently this is done at module probe time since ASoC ties in codec device probe to the instantiation of the entire ASoC device. Subsequent patches will refactor the codec drivers to handle probing separately. Note that the core does not yet use this information. AC97 is special since the codec is controlled over the AC97 link but we want to give the machine driver a chance to set up the system before trying to instantiate since it may need to do configuration before the AC97 link will operate Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-09 03:17:58 +08:00
}
module_exit(twl4030_exit);
ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:35:26 +08:00
MODULE_DESCRIPTION("ASoC TWL4030 codec driver");
MODULE_AUTHOR("Steve Sakoman");
MODULE_LICENSE("GPL");