OpenCloudOS-Kernel/sound/soc/mxs/mxs-sgtl5000.c

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/*
* Copyright 2011 Freescale Semiconductor, Inc.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* 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 Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <linux/module.h>
#include <linux/device.h>
#include <linux/of.h>
#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include <sound/soc-dapm.h>
#include "../codecs/sgtl5000.h"
#include "mxs-saif.h"
static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
unsigned int rate = params_rate(params);
u32 mclk;
int ret;
/* sgtl5000 does not support 512*rate when in 96000 fs */
switch (rate) {
case 96000:
mclk = 256 * rate;
break;
default:
mclk = 512 * rate;
break;
}
/* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */
ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0);
if (ret) {
dev_err(codec_dai->dev, "Failed to set sysclk to %u.%03uMHz\n",
mclk / 1000000, mclk / 1000 % 1000);
return ret;
}
/* The SAIF MCLK should be the same as SGTL5000_SYSCLK */
ret = snd_soc_dai_set_sysclk(cpu_dai, MXS_SAIF_MCLK, mclk, 0);
if (ret) {
dev_err(cpu_dai->dev, "Failed to set sysclk to %u.%03uMHz\n",
mclk / 1000000, mclk / 1000 % 1000);
return ret;
}
return 0;
}
ASoC: constify snd_soc_ops structures Check for snd_soc_ops structures that are only stored in the ops field of a snd_soc_dai_link structure. This field is declared const, so snd_soc_ops structures that have this property can be declared as const also. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r disable optional_qualifier@ identifier i; position p; @@ static struct snd_soc_ops i@p = { ... }; @ok1@ identifier r.i; struct snd_soc_dai_link e; position p; @@ e.ops = &i@p; @ok2@ identifier r.i, e; position p; @@ struct snd_soc_dai_link e[] = { ..., { .ops = &i@p, }, ..., }; @bad@ position p != {r.p,ok1.p,ok2.p}; identifier r.i; struct snd_soc_ops e; @@ e@i@p @depends on !bad disable optional_qualifier@ identifier r.i; @@ static +const struct snd_soc_ops i = { ... }; // </smpl> The effect on the layout of the .o files is shown by the following output of the size command, first before then after the transformation: text data bss dec hex filename 4500 696 0 5196 144c sound/soc/generic/simple-card.o 4564 632 0 5196 144c sound/soc/generic/simple-card.o text data bss dec hex filename 3018 608 0 3626 e2a sound/soc/generic/simple-scu-card.o 3074 544 0 3618 e22 sound/soc/generic/simple-scu-card.o text data bss dec hex filename 4148 2448 768 7364 1cc4 sound/soc/intel/boards/bdw-rt5677.o 4212 2384 768 7364 1cc4 sound/soc/intel/boards/bdw-rt5677.o text data bss dec hex filename 5403 4628 384 10415 28af sound/soc/intel/boards/bxt_da7219_max98357a.o 5531 4516 384 10431 28bf sound/soc/intel/boards/bxt_da7219_max98357a.o text data bss dec hex filename 5275 4496 384 10155 27ab sound/soc/intel/boards/bxt_rt298.o 5403 4368 384 10155 27ab sound/soc/intel/boards/bxt_rt298.o text data bss dec hex filename 10017 2344 48 12409 3079 sound/soc/intel/boards/bytcr_rt5640.o 10145 2232 48 12425 3089 sound/soc/intel/boards/bytcr_rt5640.o text data bss dec hex filename 3719 2356 0 6075 17bb sound/soc/intel/boards/bytcr_rt5651.o 3847 2244 0 6091 17cb sound/soc/intel/boards/bytcr_rt5651.o text data bss dec hex filename 3598 2392 0 5990 1766 sound/soc/intel/boards/cht_bsw_max98090_ti.o 3726 2280 0 6006 1776 sound/soc/intel/boards/cht_bsw_max98090_ti.o text data bss dec hex filename 5343 3624 16 8983 2317 sound/soc/intel/boards/cht_bsw_rt5645.o 5471 3496 16 8983 2317 sound/soc/intel/boards/cht_bsw_rt5645.o text data bss dec hex filename 4662 2592 384 7638 1dd6 sound/soc/intel/boards/cht_bsw_rt5672.o 4790 2464 384 7638 1dd6 sound/soc/intel/boards/cht_bsw_rt5672.o text data bss dec hex filename 1595 2528 0 4123 101b sound/soc/intel/boards/haswell.o 1659 2472 0 4131 1023 sound/soc/intel/boards/haswell.o text data bss dec hex filename 6272 4760 416 11448 2cb8 sound/soc/intel/boards/skl_nau88l25_max98357a.o 6464 4568 416 11448 2cb8 sound/soc/intel/boards/skl_nau88l25_max98357a.o text data bss dec hex filename 7075 4888 416 12379 305b sound/soc/intel/boards/skl_nau88l25_ssm4567.o 7267 4696 416 12379 305b sound/soc/intel/boards/skl_nau88l25_ssm4567.o text data bss dec hex filename 5659 4496 384 10539 292b sound/soc/intel/boards/skl_rt286.o 5787 4368 384 10539 292b sound/soc/intel/boards/skl_rt286.o text data bss dec hex filename 1721 2048 0 3769 eb9 sound/soc/kirkwood/armada-370-db.o 1769 1976 0 3745 ea1 sound/soc/kirkwood/armada-370-db.o text data bss dec hex filename 1363 1792 0 3155 c53 sound/soc/mxs/mxs-sgtl5000.o 1427 1728 0 3155 c53 sound/soc/mxs/mxs-sgtl5000.o Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-10-15 22:55:49 +08:00
static const struct snd_soc_ops mxs_sgtl5000_hifi_ops = {
.hw_params = mxs_sgtl5000_hw_params,
};
#define MXS_SGTL5000_DAI_FMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \
SND_SOC_DAIFMT_CBS_CFS)
static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
{
.name = "HiFi Tx",
.stream_name = "HiFi Playback",
.codec_dai_name = "sgtl5000",
.dai_fmt = MXS_SGTL5000_DAI_FMT,
.ops = &mxs_sgtl5000_hifi_ops,
ASoC: mxs-sgtl5000: Configure the dai_links as unidirectional On a mx28 board, running "aplay -l" and "arecord -l" results in the following: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 ,which is not correct because we got a capture device listed in aplay and a playback device listed in arecord. On mx28 there are two serial audio interface ports (SAIF0 and SAIF1) and each one of them are unidirectional. Allow to specify a dai link as 'playback_only' or 'capture_only', which suits well for this case. After this change we can correctly report the capabilities as follows: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: HiFi Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 1: HiFi Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Also tested playback and capture on the mx28evk board. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-29 21:32:14 +08:00
.playback_only = true,
}, {
.name = "HiFi Rx",
.stream_name = "HiFi Capture",
.codec_dai_name = "sgtl5000",
.dai_fmt = MXS_SGTL5000_DAI_FMT,
.ops = &mxs_sgtl5000_hifi_ops,
ASoC: mxs-sgtl5000: Configure the dai_links as unidirectional On a mx28 board, running "aplay -l" and "arecord -l" results in the following: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 ,which is not correct because we got a capture device listed in aplay and a playback device listed in arecord. On mx28 there are two serial audio interface ports (SAIF0 and SAIF1) and each one of them are unidirectional. Allow to specify a dai link as 'playback_only' or 'capture_only', which suits well for this case. After this change we can correctly report the capabilities as follows: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: HiFi Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 1: HiFi Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Also tested playback and capture on the mx28evk board. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-29 21:32:14 +08:00
.capture_only = true,
},
};
static const struct snd_soc_dapm_widget mxs_sgtl5000_dapm_widgets[] = {
SND_SOC_DAPM_MIC("Mic Jack", NULL),
SND_SOC_DAPM_LINE("Line In Jack", NULL),
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_SPK("Line Out Jack", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
};
static struct snd_soc_card mxs_sgtl5000 = {
.name = "mxs_sgtl5000",
.owner = THIS_MODULE,
.dai_link = mxs_sgtl5000_dai,
.num_links = ARRAY_SIZE(mxs_sgtl5000_dai),
};
static int mxs_sgtl5000_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &mxs_sgtl5000;
int ret, i;
struct device_node *np = pdev->dev.of_node;
struct device_node *saif_np[2], *codec_np;
saif_np[0] = of_parse_phandle(np, "saif-controllers", 0);
saif_np[1] = of_parse_phandle(np, "saif-controllers", 1);
codec_np = of_parse_phandle(np, "audio-codec", 0);
if (!saif_np[0] || !saif_np[1] || !codec_np) {
dev_err(&pdev->dev, "phandle missing or invalid\n");
return -EINVAL;
}
for (i = 0; i < 2; i++) {
mxs_sgtl5000_dai[i].codec_name = NULL;
mxs_sgtl5000_dai[i].codec_of_node = codec_np;
mxs_sgtl5000_dai[i].cpu_dai_name = NULL;
ASoC: make snd_soc_dai_link more symmetrical Prior to this patch, the CPU side of a DAI link was specified using a single name. Often, this was the result of calling dev_name() on the device providing the DAI, but in the case of a CPU DAI driver that provided multiple DAIs, it needed to mix together both the device name and some device-relative name, in order to form a single globally unique name. However, the CODEC side of the DAI link was specified using separate fields for device (name or OF node) and device-relative DAI name. This patch allows the CPU side of a DAI link to be specified in the same way as the CODEC side, separating concepts of device and device-relative DAI name. I believe this will be important in multi-codec and/or dynamic PCM scenarios, where a single CPU driver provides multiple DAIs, while also booting using device tree, with accompanying desire not to hard-code the CPU side device's name into the original .cpu_dai_name field. Ideally, both the CPU DAI and CODEC DAI loops in soc_bind_dai_link() would now be identical. However, two things prevent that at present: 1) The need to save rtd->codec for the CODEC side, which means we have to search for the CODEC explicitly, and not just the CODEC side DAI. 2) Since we know the CODEC side DAI is part of a codec, and not just a standalone DAI, it's slightly more efficient to convert .codec_name/ .codec_of_node into a codec first, and then compare each DAI's .codec field, since this avoids strcmp() on each DAI's CODEC's name within the loop. However, the two loops are essentially semantically equivalent. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-26 08:22:11 +08:00
mxs_sgtl5000_dai[i].cpu_of_node = saif_np[i];
mxs_sgtl5000_dai[i].platform_name = NULL;
mxs_sgtl5000_dai[i].platform_of_node = saif_np[i];
}
of_node_put(codec_np);
of_node_put(saif_np[0]);
of_node_put(saif_np[1]);
/*
* Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w).
* The Sgtl5000 sysclk is derived from saif0 mclk and it's range
* should be >= 8MHz and <= 27M.
*/
ret = mxs_saif_get_mclk(0, 44100 * 256, 44100);
if (ret) {
dev_err(&pdev->dev, "failed to get mclk\n");
return ret;
}
card->dev = &pdev->dev;
if (of_find_property(np, "audio-routing", NULL)) {
card->dapm_widgets = mxs_sgtl5000_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(mxs_sgtl5000_dapm_widgets);
ret = snd_soc_of_parse_audio_routing(card, "audio-routing");
if (ret) {
dev_err(&pdev->dev, "failed to parse audio-routing (%d)\n",
ret);
return ret;
}
}
ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
if (ret != -EPROBE_DEFER)
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
ret);
return ret;
}
return 0;
}
static int mxs_sgtl5000_remove(struct platform_device *pdev)
{
mxs_saif_put_mclk(0);
return 0;
}
static const struct of_device_id mxs_sgtl5000_dt_ids[] = {
{ .compatible = "fsl,mxs-audio-sgtl5000", },
{ /* sentinel */ }
};
MODULE_DEVICE_TABLE(of, mxs_sgtl5000_dt_ids);
static struct platform_driver mxs_sgtl5000_audio_driver = {
.driver = {
.name = "mxs-sgtl5000",
.of_match_table = mxs_sgtl5000_dt_ids,
},
.probe = mxs_sgtl5000_probe,
.remove = mxs_sgtl5000_remove,
};
module_platform_driver(mxs_sgtl5000_audio_driver);
MODULE_AUTHOR("Freescale Semiconductor, Inc.");
MODULE_DESCRIPTION("MXS ALSA SoC Machine driver");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:mxs-sgtl5000");