OpenCloudOS-Kernel/sound/soc/sunxi/sun8i-codec.c

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treewide: Replace GPLv2 boilerplate/reference with SPDX - rule 157 Based on 3 normalized pattern(s): 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 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 [author] [kishon] [vijay] [abraham] [i] [kishon]@[ti] [com] 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 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 [author] [graeme] [gregory] [gg]@[slimlogic] [co] [uk] [author] [kishon] [vijay] [abraham] [i] [kishon]@[ti] [com] [based] [on] [twl6030]_[usb] [c] [author] [hema] [hk] [hemahk]@[ti] [com] 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 extracted by the scancode license scanner the SPDX license identifier GPL-2.0-or-later has been chosen to replace the boilerplate/reference in 1105 file(s). Signed-off-by: Thomas Gleixner <tglx@linutronix.de> Reviewed-by: Allison Randal <allison@lohutok.net> Reviewed-by: Richard Fontana <rfontana@redhat.com> Reviewed-by: Kate Stewart <kstewart@linuxfoundation.org> Cc: linux-spdx@vger.kernel.org Link: https://lkml.kernel.org/r/20190527070033.202006027@linutronix.de Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-05-27 14:55:06 +08:00
// SPDX-License-Identifier: GPL-2.0-or-later
/*
* This driver supports the digital controls for the internal codec
* found in Allwinner's A33 SoCs.
*
* (C) Copyright 2010-2016
* Reuuimlla Technology Co., Ltd. <www.reuuimllatech.com>
* huangxin <huangxin@Reuuimllatech.com>
* Mylène Josserand <mylene.josserand@free-electrons.com>
*/
#include <linux/module.h>
#include <linux/delay.h>
#include <linux/clk.h>
#include <linux/io.h>
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
#include <linux/of_device.h>
#include <linux/pm_runtime.h>
#include <linux/regmap.h>
#include <linux/log2.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#define SUN8I_SYSCLK_CTL 0x00c
#define SUN8I_SYSCLK_CTL_AIF1CLK_ENA 11
#define SUN8I_SYSCLK_CTL_AIF1CLK_SRC_PLL (0x2 << 8)
#define SUN8I_SYSCLK_CTL_AIF2CLK_ENA 7
#define SUN8I_SYSCLK_CTL_AIF2CLK_SRC_PLL (0x2 << 4)
#define SUN8I_SYSCLK_CTL_SYSCLK_ENA 3
#define SUN8I_SYSCLK_CTL_SYSCLK_SRC 0
#define SUN8I_SYSCLK_CTL_SYSCLK_SRC_AIF1CLK (0x0 << 0)
#define SUN8I_SYSCLK_CTL_SYSCLK_SRC_AIF2CLK (0x1 << 0)
#define SUN8I_MOD_CLK_ENA 0x010
#define SUN8I_MOD_CLK_ENA_AIF1 15
#define SUN8I_MOD_CLK_ENA_AIF2 14
#define SUN8I_MOD_CLK_ENA_ADC 3
#define SUN8I_MOD_CLK_ENA_DAC 2
#define SUN8I_MOD_RST_CTL 0x014
#define SUN8I_MOD_RST_CTL_AIF1 15
#define SUN8I_MOD_RST_CTL_AIF2 14
#define SUN8I_MOD_RST_CTL_ADC 3
#define SUN8I_MOD_RST_CTL_DAC 2
#define SUN8I_SYS_SR_CTRL 0x018
#define SUN8I_SYS_SR_CTRL_AIF1_FS 12
#define SUN8I_SYS_SR_CTRL_AIF2_FS 8
#define SUN8I_AIF_CLK_CTRL(n) (0x040 * (1 + (n)))
#define SUN8I_AIF_CLK_CTRL_MSTR_MOD 15
#define SUN8I_AIF_CLK_CTRL_CLK_INV 13
#define SUN8I_AIF_CLK_CTRL_BCLK_DIV 9
#define SUN8I_AIF_CLK_CTRL_LRCK_DIV 6
#define SUN8I_AIF_CLK_CTRL_WORD_SIZ 4
#define SUN8I_AIF_CLK_CTRL_DATA_FMT 2
#define SUN8I_AIF1_ADCDAT_CTRL 0x044
#define SUN8I_AIF1_ADCDAT_CTRL_AIF1_AD0L_ENA 15
#define SUN8I_AIF1_ADCDAT_CTRL_AIF1_AD0R_ENA 14
#define SUN8I_AIF1_ADCDAT_CTRL_AIF1_AD0L_SRC 10
#define SUN8I_AIF1_ADCDAT_CTRL_AIF1_AD0R_SRC 8
#define SUN8I_AIF1_DACDAT_CTRL 0x048
#define SUN8I_AIF1_DACDAT_CTRL_AIF1_DA0L_ENA 15
#define SUN8I_AIF1_DACDAT_CTRL_AIF1_DA0R_ENA 14
#define SUN8I_AIF1_DACDAT_CTRL_AIF1_DA0L_SRC 10
#define SUN8I_AIF1_DACDAT_CTRL_AIF1_DA0R_SRC 8
#define SUN8I_AIF1_MXR_SRC 0x04c
#define SUN8I_AIF1_MXR_SRC_AD0L_MXR_SRC_AIF1DA0L 15
#define SUN8I_AIF1_MXR_SRC_AD0L_MXR_SRC_AIF2DACL 14
#define SUN8I_AIF1_MXR_SRC_AD0L_MXR_SRC_ADCL 13
#define SUN8I_AIF1_MXR_SRC_AD0L_MXR_SRC_AIF2DACR 12
#define SUN8I_AIF1_MXR_SRC_AD0R_MXR_SRC_AIF1DA0R 11
#define SUN8I_AIF1_MXR_SRC_AD0R_MXR_SRC_AIF2DACR 10
#define SUN8I_AIF1_MXR_SRC_AD0R_MXR_SRC_ADCR 9
#define SUN8I_AIF1_MXR_SRC_AD0R_MXR_SRC_AIF2DACL 8
#define SUN8I_AIF2_ADCDAT_CTRL 0x084
#define SUN8I_AIF2_ADCDAT_CTRL_AIF2_ADCL_ENA 15
#define SUN8I_AIF2_ADCDAT_CTRL_AIF2_ADCR_ENA 14
#define SUN8I_AIF2_ADCDAT_CTRL_AIF2_ADCL_SRC 10
#define SUN8I_AIF2_ADCDAT_CTRL_AIF2_ADCR_SRC 8
#define SUN8I_AIF2_DACDAT_CTRL 0x088
#define SUN8I_AIF2_DACDAT_CTRL_AIF2_DACL_ENA 15
#define SUN8I_AIF2_DACDAT_CTRL_AIF2_DACR_ENA 14
#define SUN8I_AIF2_DACDAT_CTRL_AIF2_DACL_SRC 10
#define SUN8I_AIF2_DACDAT_CTRL_AIF2_DACR_SRC 8
#define SUN8I_AIF2_MXR_SRC 0x08c
#define SUN8I_AIF2_MXR_SRC_ADCL_MXR_SRC_AIF1DA0L 15
#define SUN8I_AIF2_MXR_SRC_ADCL_MXR_SRC_AIF1DA1L 14
#define SUN8I_AIF2_MXR_SRC_ADCL_MXR_SRC_AIF2DACR 13
#define SUN8I_AIF2_MXR_SRC_ADCL_MXR_SRC_ADCL 12
#define SUN8I_AIF2_MXR_SRC_ADCR_MXR_SRC_AIF1DA0R 11
#define SUN8I_AIF2_MXR_SRC_ADCR_MXR_SRC_AIF1DA1R 10
#define SUN8I_AIF2_MXR_SRC_ADCR_MXR_SRC_AIF2DACL 9
#define SUN8I_AIF2_MXR_SRC_ADCR_MXR_SRC_ADCR 8
#define SUN8I_AIF3_PATH_CTRL 0x0cc
#define SUN8I_AIF3_PATH_CTRL_AIF3_ADC_SRC 10
#define SUN8I_AIF3_PATH_CTRL_AIF2_DAC_SRC 8
#define SUN8I_AIF3_PATH_CTRL_AIF3_PINS_TRI 7
#define SUN8I_ADC_DIG_CTRL 0x100
#define SUN8I_ADC_DIG_CTRL_ENAD 15
#define SUN8I_ADC_DIG_CTRL_ADOUT_DTS 2
#define SUN8I_ADC_DIG_CTRL_ADOUT_DLY 1
#define SUN8I_DAC_DIG_CTRL 0x120
#define SUN8I_DAC_DIG_CTRL_ENDA 15
#define SUN8I_DAC_MXR_SRC 0x130
#define SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF1DA0L 15
#define SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF1DA1L 14
#define SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF2DACL 13
#define SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_ADCL 12
#define SUN8I_DAC_MXR_SRC_DACR_MXR_SRC_AIF1DA0R 11
#define SUN8I_DAC_MXR_SRC_DACR_MXR_SRC_AIF1DA1R 10
#define SUN8I_DAC_MXR_SRC_DACR_MXR_SRC_AIF2DACR 9
#define SUN8I_DAC_MXR_SRC_DACR_MXR_SRC_ADCR 8
#define SUN8I_SYSCLK_CTL_AIF1CLK_SRC_MASK GENMASK(9, 8)
#define SUN8I_SYSCLK_CTL_AIF2CLK_SRC_MASK GENMASK(5, 4)
#define SUN8I_SYS_SR_CTRL_AIF1_FS_MASK GENMASK(15, 12)
#define SUN8I_SYS_SR_CTRL_AIF2_FS_MASK GENMASK(11, 8)
#define SUN8I_AIF_CLK_CTRL_CLK_INV_MASK GENMASK(14, 13)
#define SUN8I_AIF_CLK_CTRL_BCLK_DIV_MASK GENMASK(12, 9)
#define SUN8I_AIF_CLK_CTRL_LRCK_DIV_MASK GENMASK(8, 6)
#define SUN8I_AIF_CLK_CTRL_WORD_SIZ_MASK GENMASK(5, 4)
#define SUN8I_AIF_CLK_CTRL_DATA_FMT_MASK GENMASK(3, 2)
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
#define SUN8I_CODEC_PASSTHROUGH_SAMPLE_RATE 48000
#define SUN8I_CODEC_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 |\
SNDRV_PCM_FMTBIT_S16_LE |\
SNDRV_PCM_FMTBIT_S20_LE |\
SNDRV_PCM_FMTBIT_S24_LE |\
SNDRV_PCM_FMTBIT_S20_3LE|\
SNDRV_PCM_FMTBIT_S24_3LE)
#define SUN8I_CODEC_PCM_RATES (SNDRV_PCM_RATE_8000_48000|\
SNDRV_PCM_RATE_88200 |\
SNDRV_PCM_RATE_96000 |\
SNDRV_PCM_RATE_176400 |\
SNDRV_PCM_RATE_192000 |\
SNDRV_PCM_RATE_KNOT)
enum {
SUN8I_CODEC_AIF1,
SUN8I_CODEC_AIF2,
SUN8I_CODEC_NAIFS
};
struct sun8i_codec_aif {
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
unsigned int sample_rate;
unsigned int slots;
unsigned int slot_width;
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
unsigned int active_streams : 2;
unsigned int open_streams : 2;
};
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
struct sun8i_codec_quirks {
bool legacy_widgets : 1;
bool lrck_inversion : 1;
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
};
struct sun8i_codec {
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
struct regmap *regmap;
struct clk *clk_module;
const struct sun8i_codec_quirks *quirks;
struct sun8i_codec_aif aifs[SUN8I_CODEC_NAIFS];
unsigned int sysclk_rate;
int sysclk_refcnt;
};
static int sun8i_codec_runtime_resume(struct device *dev)
{
struct sun8i_codec *scodec = dev_get_drvdata(dev);
int ret;
regcache_cache_only(scodec->regmap, false);
ret = regcache_sync(scodec->regmap);
if (ret) {
dev_err(dev, "Failed to sync regmap cache\n");
return ret;
}
return 0;
}
static int sun8i_codec_runtime_suspend(struct device *dev)
{
struct sun8i_codec *scodec = dev_get_drvdata(dev);
regcache_cache_only(scodec->regmap, true);
regcache_mark_dirty(scodec->regmap);
return 0;
}
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
static int sun8i_codec_get_hw_rate(unsigned int sample_rate)
{
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
switch (sample_rate) {
case 7350:
case 8000:
return 0x0;
case 11025:
return 0x1;
case 12000:
return 0x2;
case 14700:
case 16000:
return 0x3;
case 22050:
return 0x4;
case 24000:
return 0x5;
case 29400:
case 32000:
return 0x6;
case 44100:
return 0x7;
case 48000:
return 0x8;
case 88200:
case 96000:
return 0x9;
case 176400:
case 192000:
return 0xa;
default:
return -EINVAL;
}
}
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
static int sun8i_codec_update_sample_rate(struct sun8i_codec *scodec)
{
unsigned int max_rate = 0;
int hw_rate, i;
for (i = SUN8I_CODEC_AIF1; i < SUN8I_CODEC_NAIFS; ++i) {
struct sun8i_codec_aif *aif = &scodec->aifs[i];
if (aif->active_streams)
max_rate = max(max_rate, aif->sample_rate);
}
/* Set the sample rate for ADC->DAC passthrough when no AIF is active. */
if (!max_rate)
max_rate = SUN8I_CODEC_PASSTHROUGH_SAMPLE_RATE;
hw_rate = sun8i_codec_get_hw_rate(max_rate);
if (hw_rate < 0)
return hw_rate;
regmap_update_bits(scodec->regmap, SUN8I_SYS_SR_CTRL,
SUN8I_SYS_SR_CTRL_AIF1_FS_MASK,
hw_rate << SUN8I_SYS_SR_CTRL_AIF1_FS);
return 0;
}
static int sun8i_codec_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct sun8i_codec *scodec = snd_soc_dai_get_drvdata(dai);
u32 dsp_format, format, invert, value;
/* clock masters */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS: /* Codec slave, DAI master */
value = 0x1;
break;
case SND_SOC_DAIFMT_CBM_CFM: /* Codec Master, DAI slave */
value = 0x0;
break;
default:
return -EINVAL;
}
regmap_update_bits(scodec->regmap, SUN8I_AIF_CLK_CTRL(dai->id),
BIT(SUN8I_AIF_CLK_CTRL_MSTR_MOD),
value << SUN8I_AIF_CLK_CTRL_MSTR_MOD);
/* DAI format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
format = 0x0;
break;
case SND_SOC_DAIFMT_LEFT_J:
format = 0x1;
break;
case SND_SOC_DAIFMT_RIGHT_J:
format = 0x2;
break;
case SND_SOC_DAIFMT_DSP_A:
format = 0x3;
dsp_format = 0x0; /* Set LRCK_INV to 0 */
break;
case SND_SOC_DAIFMT_DSP_B:
format = 0x3;
dsp_format = 0x1; /* Set LRCK_INV to 1 */
break;
default:
return -EINVAL;
}
regmap_update_bits(scodec->regmap, SUN8I_AIF_CLK_CTRL(dai->id),
SUN8I_AIF_CLK_CTRL_DATA_FMT_MASK,
format << SUN8I_AIF_CLK_CTRL_DATA_FMT);
/* clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF: /* Normal */
invert = 0x0;
break;
case SND_SOC_DAIFMT_NB_IF: /* Inverted LRCK */
invert = 0x1;
break;
case SND_SOC_DAIFMT_IB_NF: /* Inverted BCLK */
invert = 0x2;
break;
case SND_SOC_DAIFMT_IB_IF: /* Both inverted */
invert = 0x3;
break;
default:
return -EINVAL;
}
if (format == 0x3) {
/* Inverted LRCK is not available in DSP mode. */
if (invert & BIT(0))
return -EINVAL;
/* Instead, the bit selects between DSP A/B formats. */
invert |= dsp_format;
} else {
/*
* It appears that the DAI and the codec in the A33 SoC don't
* share the same polarity for the LRCK signal when they mean
* 'normal' and 'inverted' in the datasheet.
*
* Since the DAI here is our regular i2s driver that have been
* tested with way more codecs than just this one, it means
* that the codec probably gets it backward, and we have to
* invert the value here.
*/
invert ^= scodec->quirks->lrck_inversion;
}
regmap_update_bits(scodec->regmap, SUN8I_AIF_CLK_CTRL(dai->id),
SUN8I_AIF_CLK_CTRL_CLK_INV_MASK,
invert << SUN8I_AIF_CLK_CTRL_CLK_INV);
return 0;
}
static int sun8i_codec_set_tdm_slot(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask,
int slots, int slot_width)
{
struct sun8i_codec *scodec = snd_soc_dai_get_drvdata(dai);
struct sun8i_codec_aif *aif = &scodec->aifs[dai->id];
if (slot_width && !is_power_of_2(slot_width))
return -EINVAL;
aif->slots = slots;
aif->slot_width = slot_width;
return 0;
}
static const unsigned int sun8i_codec_rates[] = {
7350, 8000, 11025, 12000, 14700, 16000, 22050, 24000,
29400, 32000, 44100, 48000, 88200, 96000, 176400, 192000,
};
static const struct snd_pcm_hw_constraint_list sun8i_codec_all_rates = {
.list = sun8i_codec_rates,
.count = ARRAY_SIZE(sun8i_codec_rates),
};
static const struct snd_pcm_hw_constraint_list sun8i_codec_22M_rates = {
.list = sun8i_codec_rates,
.count = ARRAY_SIZE(sun8i_codec_rates),
.mask = 0x5555,
};
static const struct snd_pcm_hw_constraint_list sun8i_codec_24M_rates = {
.list = sun8i_codec_rates,
.count = ARRAY_SIZE(sun8i_codec_rates),
.mask = 0xaaaa,
};
static int sun8i_codec_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct sun8i_codec *scodec = snd_soc_dai_get_drvdata(dai);
const struct snd_pcm_hw_constraint_list *list;
/* hw_constraints is not relevant for codec2codec DAIs. */
if (dai->id != SUN8I_CODEC_AIF1)
return 0;
if (!scodec->sysclk_refcnt)
list = &sun8i_codec_all_rates;
else if (scodec->sysclk_rate == 22579200)
list = &sun8i_codec_22M_rates;
else if (scodec->sysclk_rate == 24576000)
list = &sun8i_codec_24M_rates;
else
return -EINVAL;
return snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE, list);
}
struct sun8i_codec_clk_div {
u8 div;
u8 val;
};
static const struct sun8i_codec_clk_div sun8i_codec_bclk_div[] = {
{ .div = 1, .val = 0 },
{ .div = 2, .val = 1 },
{ .div = 4, .val = 2 },
{ .div = 6, .val = 3 },
{ .div = 8, .val = 4 },
{ .div = 12, .val = 5 },
{ .div = 16, .val = 6 },
{ .div = 24, .val = 7 },
{ .div = 32, .val = 8 },
{ .div = 48, .val = 9 },
{ .div = 64, .val = 10 },
{ .div = 96, .val = 11 },
{ .div = 128, .val = 12 },
{ .div = 192, .val = 13 },
};
static int sun8i_codec_get_bclk_div(unsigned int sysclk_rate,
unsigned int lrck_div_order,
unsigned int sample_rate)
{
unsigned int div = sysclk_rate / sample_rate >> lrck_div_order;
int i;
for (i = 0; i < ARRAY_SIZE(sun8i_codec_bclk_div); i++) {
const struct sun8i_codec_clk_div *bdiv = &sun8i_codec_bclk_div[i];
if (bdiv->div == div)
return bdiv->val;
}
return -EINVAL;
}
static int sun8i_codec_get_lrck_div_order(unsigned int slots,
unsigned int slot_width)
{
unsigned int div = slots * slot_width;
if (div < 16 || div > 256)
return -EINVAL;
return order_base_2(div);
}
static unsigned int sun8i_codec_get_sysclk_rate(unsigned int sample_rate)
{
return sample_rate % 4000 ? 22579200 : 24576000;
}
static int sun8i_codec_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct sun8i_codec *scodec = snd_soc_dai_get_drvdata(dai);
struct sun8i_codec_aif *aif = &scodec->aifs[dai->id];
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
unsigned int sample_rate = params_rate(params);
unsigned int slots = aif->slots ?: params_channels(params);
unsigned int slot_width = aif->slot_width ?: params_width(params);
unsigned int sysclk_rate = sun8i_codec_get_sysclk_rate(sample_rate);
int bclk_div, lrck_div_order, ret, word_size;
/* word size */
switch (params_width(params)) {
case 8:
word_size = 0x0;
break;
case 16:
word_size = 0x1;
break;
case 20:
word_size = 0x2;
break;
case 24:
word_size = 0x3;
break;
default:
return -EINVAL;
}
regmap_update_bits(scodec->regmap, SUN8I_AIF_CLK_CTRL(dai->id),
SUN8I_AIF_CLK_CTRL_WORD_SIZ_MASK,
word_size << SUN8I_AIF_CLK_CTRL_WORD_SIZ);
/* LRCK divider (BCLK/LRCK ratio) */
lrck_div_order = sun8i_codec_get_lrck_div_order(slots, slot_width);
if (lrck_div_order < 0)
return lrck_div_order;
regmap_update_bits(scodec->regmap, SUN8I_AIF_CLK_CTRL(dai->id),
SUN8I_AIF_CLK_CTRL_LRCK_DIV_MASK,
(lrck_div_order - 4) << SUN8I_AIF_CLK_CTRL_LRCK_DIV);
/* BCLK divider (SYSCLK/BCLK ratio) */
bclk_div = sun8i_codec_get_bclk_div(sysclk_rate, lrck_div_order, sample_rate);
if (bclk_div < 0)
return bclk_div;
regmap_update_bits(scodec->regmap, SUN8I_AIF_CLK_CTRL(dai->id),
SUN8I_AIF_CLK_CTRL_BCLK_DIV_MASK,
bclk_div << SUN8I_AIF_CLK_CTRL_BCLK_DIV);
/*
* SYSCLK rate
*
* Clock rate protection is reference counted; but hw_params may be
* called many times per substream, without matching calls to hw_free.
* Protect the clock rate once per AIF, on the first hw_params call
* for the first substream. clk_set_rate() will allow clock rate
* changes on subsequent calls if only one AIF has open streams.
*/
ret = (aif->open_streams ? clk_set_rate : clk_set_rate_exclusive)(scodec->clk_module,
sysclk_rate);
if (ret == -EBUSY)
dev_err(dai->dev,
"%s sample rate (%u Hz) conflicts with other audio streams\n",
dai->name, sample_rate);
if (ret < 0)
return ret;
if (!aif->open_streams)
scodec->sysclk_refcnt++;
scodec->sysclk_rate = sysclk_rate;
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
aif->sample_rate = sample_rate;
aif->open_streams |= BIT(substream->stream);
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
return sun8i_codec_update_sample_rate(scodec);
}
static int sun8i_codec_hw_free(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct sun8i_codec *scodec = snd_soc_dai_get_drvdata(dai);
struct sun8i_codec_aif *aif = &scodec->aifs[dai->id];
/* Drop references when the last substream for the AIF is freed. */
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
if (aif->open_streams != BIT(substream->stream))
goto done;
clk_rate_exclusive_put(scodec->clk_module);
scodec->sysclk_refcnt--;
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
aif->sample_rate = 0;
done:
aif->open_streams &= ~BIT(substream->stream);
return 0;
}
static const struct snd_soc_dai_ops sun8i_codec_dai_ops = {
.set_fmt = sun8i_codec_set_fmt,
.set_tdm_slot = sun8i_codec_set_tdm_slot,
.startup = sun8i_codec_startup,
.hw_params = sun8i_codec_hw_params,
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
.hw_free = sun8i_codec_hw_free,
};
static struct snd_soc_dai_driver sun8i_codec_dais[] = {
{
.name = "sun8i-codec-aif1",
.id = SUN8I_CODEC_AIF1,
.ops = &sun8i_codec_dai_ops,
/* capture capabilities */
.capture = {
.stream_name = "AIF1 Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SUN8I_CODEC_PCM_RATES,
.formats = SUN8I_CODEC_PCM_FORMATS,
.sig_bits = 24,
},
/* playback capabilities */
.playback = {
.stream_name = "AIF1 Playback",
.channels_min = 1,
.channels_max = 2,
.rates = SUN8I_CODEC_PCM_RATES,
.formats = SUN8I_CODEC_PCM_FORMATS,
},
.symmetric_rates = true,
.symmetric_channels = true,
.symmetric_samplebits = true,
},
{
.name = "sun8i-codec-aif2",
.id = SUN8I_CODEC_AIF2,
.ops = &sun8i_codec_dai_ops,
/* capture capabilities */
.capture = {
.stream_name = "AIF2 Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SUN8I_CODEC_PCM_RATES,
.formats = SUN8I_CODEC_PCM_FORMATS,
.sig_bits = 24,
},
/* playback capabilities */
.playback = {
.stream_name = "AIF2 Playback",
.channels_min = 1,
.channels_max = 2,
.rates = SUN8I_CODEC_PCM_RATES,
.formats = SUN8I_CODEC_PCM_FORMATS,
},
.symmetric_rates = true,
.symmetric_channels = true,
.symmetric_samplebits = true,
},
};
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
static int sun8i_codec_aif_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
struct sun8i_codec *scodec = snd_soc_component_get_drvdata(component);
struct sun8i_codec_aif *aif = &scodec->aifs[w->sname[3] - '1'];
int stream = w->id == snd_soc_dapm_aif_out;
if (SND_SOC_DAPM_EVENT_ON(event))
aif->active_streams |= BIT(stream);
else
aif->active_streams &= ~BIT(stream);
return sun8i_codec_update_sample_rate(scodec);
}
static const char *const sun8i_aif_stereo_mux_enum_values[] = {
"Stereo", "Reverse Stereo", "Sum Mono", "Mix Mono"
};
static SOC_ENUM_DOUBLE_DECL(sun8i_aif1_ad0_stereo_mux_enum,
SUN8I_AIF1_ADCDAT_CTRL,
SUN8I_AIF1_ADCDAT_CTRL_AIF1_AD0L_SRC,
SUN8I_AIF1_ADCDAT_CTRL_AIF1_AD0R_SRC,
sun8i_aif_stereo_mux_enum_values);
static const struct snd_kcontrol_new sun8i_aif1_ad0_stereo_mux_control =
SOC_DAPM_ENUM("AIF1 AD0 Stereo Capture Route",
sun8i_aif1_ad0_stereo_mux_enum);
static SOC_ENUM_DOUBLE_DECL(sun8i_aif2_adc_stereo_mux_enum,
SUN8I_AIF2_ADCDAT_CTRL,
SUN8I_AIF2_ADCDAT_CTRL_AIF2_ADCL_SRC,
SUN8I_AIF2_ADCDAT_CTRL_AIF2_ADCR_SRC,
sun8i_aif_stereo_mux_enum_values);
static const struct snd_kcontrol_new sun8i_aif2_adc_stereo_mux_control =
SOC_DAPM_ENUM("AIF2 ADC Stereo Capture Route",
sun8i_aif2_adc_stereo_mux_enum);
static const struct snd_kcontrol_new sun8i_aif1_ad0_mixer_controls[] = {
SOC_DAPM_DOUBLE("AIF1 Slot 0 Digital ADC Capture Switch",
SUN8I_AIF1_MXR_SRC,
SUN8I_AIF1_MXR_SRC_AD0L_MXR_SRC_AIF1DA0L,
SUN8I_AIF1_MXR_SRC_AD0R_MXR_SRC_AIF1DA0R, 1, 0),
SOC_DAPM_DOUBLE("AIF2 Digital ADC Capture Switch",
SUN8I_AIF1_MXR_SRC,
SUN8I_AIF1_MXR_SRC_AD0L_MXR_SRC_AIF2DACL,
SUN8I_AIF1_MXR_SRC_AD0R_MXR_SRC_AIF2DACR, 1, 0),
SOC_DAPM_DOUBLE("AIF1 Data Digital ADC Capture Switch",
SUN8I_AIF1_MXR_SRC,
SUN8I_AIF1_MXR_SRC_AD0L_MXR_SRC_ADCL,
SUN8I_AIF1_MXR_SRC_AD0R_MXR_SRC_ADCR, 1, 0),
SOC_DAPM_DOUBLE("AIF2 Inv Digital ADC Capture Switch",
SUN8I_AIF1_MXR_SRC,
SUN8I_AIF1_MXR_SRC_AD0L_MXR_SRC_AIF2DACR,
SUN8I_AIF1_MXR_SRC_AD0R_MXR_SRC_AIF2DACL, 1, 0),
};
static const struct snd_kcontrol_new sun8i_aif2_adc_mixer_controls[] = {
SOC_DAPM_DOUBLE("AIF2 ADC Mixer AIF1 DA0 Capture Switch",
SUN8I_AIF2_MXR_SRC,
SUN8I_AIF2_MXR_SRC_ADCL_MXR_SRC_AIF1DA0L,
SUN8I_AIF2_MXR_SRC_ADCR_MXR_SRC_AIF1DA0R, 1, 0),
SOC_DAPM_DOUBLE("AIF2 ADC Mixer AIF1 DA1 Capture Switch",
SUN8I_AIF2_MXR_SRC,
SUN8I_AIF2_MXR_SRC_ADCL_MXR_SRC_AIF1DA1L,
SUN8I_AIF2_MXR_SRC_ADCR_MXR_SRC_AIF1DA1R, 1, 0),
SOC_DAPM_DOUBLE("AIF2 ADC Mixer AIF2 DAC Rev Capture Switch",
SUN8I_AIF2_MXR_SRC,
SUN8I_AIF2_MXR_SRC_ADCL_MXR_SRC_AIF2DACR,
SUN8I_AIF2_MXR_SRC_ADCR_MXR_SRC_AIF2DACL, 1, 0),
SOC_DAPM_DOUBLE("AIF2 ADC Mixer ADC Capture Switch",
SUN8I_AIF2_MXR_SRC,
SUN8I_AIF2_MXR_SRC_ADCL_MXR_SRC_ADCL,
SUN8I_AIF2_MXR_SRC_ADCR_MXR_SRC_ADCR, 1, 0),
};
static const char *const sun8i_aif2_dac_mux_enum_values[] = {
"AIF2", "AIF3+2", "AIF2+3"
};
static SOC_ENUM_SINGLE_DECL(sun8i_aif2_dac_mux_enum,
SUN8I_AIF3_PATH_CTRL,
SUN8I_AIF3_PATH_CTRL_AIF2_DAC_SRC,
sun8i_aif2_dac_mux_enum_values);
static const struct snd_kcontrol_new sun8i_aif2_dac_mux_control =
SOC_DAPM_ENUM("AIF2 DAC Source Playback Route",
sun8i_aif2_dac_mux_enum);
static SOC_ENUM_DOUBLE_DECL(sun8i_aif1_da0_stereo_mux_enum,
SUN8I_AIF1_DACDAT_CTRL,
SUN8I_AIF1_DACDAT_CTRL_AIF1_DA0L_SRC,
SUN8I_AIF1_DACDAT_CTRL_AIF1_DA0R_SRC,
sun8i_aif_stereo_mux_enum_values);
static const struct snd_kcontrol_new sun8i_aif1_da0_stereo_mux_control =
SOC_DAPM_ENUM("AIF1 DA0 Stereo Playback Route",
sun8i_aif1_da0_stereo_mux_enum);
static SOC_ENUM_DOUBLE_DECL(sun8i_aif2_dac_stereo_mux_enum,
SUN8I_AIF2_DACDAT_CTRL,
SUN8I_AIF2_DACDAT_CTRL_AIF2_DACL_SRC,
SUN8I_AIF2_DACDAT_CTRL_AIF2_DACR_SRC,
sun8i_aif_stereo_mux_enum_values);
static const struct snd_kcontrol_new sun8i_aif2_dac_stereo_mux_control =
SOC_DAPM_ENUM("AIF2 DAC Stereo Playback Route",
sun8i_aif2_dac_stereo_mux_enum);
static const struct snd_kcontrol_new sun8i_dac_mixer_controls[] = {
SOC_DAPM_DOUBLE("AIF1 Slot 0 Digital DAC Playback Switch",
SUN8I_DAC_MXR_SRC,
SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF1DA0L,
SUN8I_DAC_MXR_SRC_DACR_MXR_SRC_AIF1DA0R, 1, 0),
SOC_DAPM_DOUBLE("AIF1 Slot 1 Digital DAC Playback Switch",
SUN8I_DAC_MXR_SRC,
SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF1DA1L,
SUN8I_DAC_MXR_SRC_DACR_MXR_SRC_AIF1DA1R, 1, 0),
SOC_DAPM_DOUBLE("AIF2 Digital DAC Playback Switch", SUN8I_DAC_MXR_SRC,
SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF2DACL,
SUN8I_DAC_MXR_SRC_DACR_MXR_SRC_AIF2DACR, 1, 0),
SOC_DAPM_DOUBLE("ADC Digital DAC Playback Switch", SUN8I_DAC_MXR_SRC,
SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_ADCL,
SUN8I_DAC_MXR_SRC_DACR_MXR_SRC_ADCR, 1, 0),
};
static const struct snd_soc_dapm_widget sun8i_codec_dapm_widgets[] = {
/* System Clocks */
SND_SOC_DAPM_CLOCK_SUPPLY("mod"),
SND_SOC_DAPM_SUPPLY("AIF1CLK",
SUN8I_SYSCLK_CTL,
SUN8I_SYSCLK_CTL_AIF1CLK_ENA, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("AIF2CLK",
SUN8I_SYSCLK_CTL,
SUN8I_SYSCLK_CTL_AIF2CLK_ENA, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("SYSCLK",
SUN8I_SYSCLK_CTL,
SUN8I_SYSCLK_CTL_SYSCLK_ENA, 0, NULL, 0),
/* Module Clocks */
SND_SOC_DAPM_SUPPLY("CLK AIF1",
SUN8I_MOD_CLK_ENA,
SUN8I_MOD_CLK_ENA_AIF1, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("CLK AIF2",
SUN8I_MOD_CLK_ENA,
SUN8I_MOD_CLK_ENA_AIF2, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("CLK ADC",
SUN8I_MOD_CLK_ENA,
SUN8I_MOD_CLK_ENA_ADC, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("CLK DAC",
SUN8I_MOD_CLK_ENA,
SUN8I_MOD_CLK_ENA_DAC, 0, NULL, 0),
/* Module Resets */
SND_SOC_DAPM_SUPPLY("RST AIF1",
SUN8I_MOD_RST_CTL,
SUN8I_MOD_RST_CTL_AIF1, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("RST AIF2",
SUN8I_MOD_RST_CTL,
SUN8I_MOD_RST_CTL_AIF2, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("RST ADC",
SUN8I_MOD_RST_CTL,
SUN8I_MOD_RST_CTL_ADC, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("RST DAC",
SUN8I_MOD_RST_CTL,
SUN8I_MOD_RST_CTL_DAC, 0, NULL, 0),
/* Module Supplies */
SND_SOC_DAPM_SUPPLY("ADC",
SUN8I_ADC_DIG_CTRL,
SUN8I_ADC_DIG_CTRL_ENAD, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("DAC",
SUN8I_DAC_DIG_CTRL,
SUN8I_DAC_DIG_CTRL_ENDA, 0, NULL, 0),
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
/* AIF "ADC" Outputs */
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
SND_SOC_DAPM_AIF_OUT_E("AIF1 AD0L", "AIF1 Capture", 0,
SUN8I_AIF1_ADCDAT_CTRL,
SUN8I_AIF1_ADCDAT_CTRL_AIF1_AD0L_ENA, 0,
sun8i_codec_aif_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_OUT("AIF1 AD0R", "AIF1 Capture", 1,
SUN8I_AIF1_ADCDAT_CTRL,
SUN8I_AIF1_ADCDAT_CTRL_AIF1_AD0R_ENA, 0),
SND_SOC_DAPM_AIF_OUT_E("AIF2 ADCL", "AIF2 Capture", 0,
SUN8I_AIF2_ADCDAT_CTRL,
SUN8I_AIF2_ADCDAT_CTRL_AIF2_ADCL_ENA, 0,
sun8i_codec_aif_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_OUT("AIF2 ADCR", "AIF2 Capture", 1,
SUN8I_AIF2_ADCDAT_CTRL,
SUN8I_AIF2_ADCDAT_CTRL_AIF2_ADCR_ENA, 0),
/* AIF "ADC" Mono/Stereo Muxes */
SND_SOC_DAPM_MUX("AIF1 AD0L Stereo Mux", SND_SOC_NOPM, 0, 0,
&sun8i_aif1_ad0_stereo_mux_control),
SND_SOC_DAPM_MUX("AIF1 AD0R Stereo Mux", SND_SOC_NOPM, 0, 0,
&sun8i_aif1_ad0_stereo_mux_control),
SND_SOC_DAPM_MUX("AIF2 ADCL Stereo Mux", SND_SOC_NOPM, 0, 0,
&sun8i_aif2_adc_stereo_mux_control),
SND_SOC_DAPM_MUX("AIF2 ADCR Stereo Mux", SND_SOC_NOPM, 0, 0,
&sun8i_aif2_adc_stereo_mux_control),
/* AIF "ADC" Mixers */
SOC_MIXER_ARRAY("AIF1 AD0L Mixer", SND_SOC_NOPM, 0, 0,
sun8i_aif1_ad0_mixer_controls),
SOC_MIXER_ARRAY("AIF1 AD0R Mixer", SND_SOC_NOPM, 0, 0,
sun8i_aif1_ad0_mixer_controls),
SOC_MIXER_ARRAY("AIF2 ADCL Mixer", SND_SOC_NOPM, 0, 0,
sun8i_aif2_adc_mixer_controls),
SOC_MIXER_ARRAY("AIF2 ADCR Mixer", SND_SOC_NOPM, 0, 0,
sun8i_aif2_adc_mixer_controls),
/* AIF "DAC" Input Muxes */
SND_SOC_DAPM_MUX("AIF2 DACL Source", SND_SOC_NOPM, 0, 0,
&sun8i_aif2_dac_mux_control),
SND_SOC_DAPM_MUX("AIF2 DACR Source", SND_SOC_NOPM, 0, 0,
&sun8i_aif2_dac_mux_control),
/* AIF "DAC" Mono/Stereo Muxes */
SND_SOC_DAPM_MUX("AIF1 DA0L Stereo Mux", SND_SOC_NOPM, 0, 0,
&sun8i_aif1_da0_stereo_mux_control),
SND_SOC_DAPM_MUX("AIF1 DA0R Stereo Mux", SND_SOC_NOPM, 0, 0,
&sun8i_aif1_da0_stereo_mux_control),
SND_SOC_DAPM_MUX("AIF2 DACL Stereo Mux", SND_SOC_NOPM, 0, 0,
&sun8i_aif2_dac_stereo_mux_control),
SND_SOC_DAPM_MUX("AIF2 DACR Stereo Mux", SND_SOC_NOPM, 0, 0,
&sun8i_aif2_dac_stereo_mux_control),
/* AIF "DAC" Inputs */
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
SND_SOC_DAPM_AIF_IN_E("AIF1 DA0L", "AIF1 Playback", 0,
SUN8I_AIF1_DACDAT_CTRL,
SUN8I_AIF1_DACDAT_CTRL_AIF1_DA0L_ENA, 0,
sun8i_codec_aif_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_IN("AIF1 DA0R", "AIF1 Playback", 1,
SUN8I_AIF1_DACDAT_CTRL,
SUN8I_AIF1_DACDAT_CTRL_AIF1_DA0R_ENA, 0),
SND_SOC_DAPM_AIF_IN_E("AIF2 DACL", "AIF2 Playback", 0,
SUN8I_AIF2_DACDAT_CTRL,
SUN8I_AIF2_DACDAT_CTRL_AIF2_DACL_ENA, 0,
sun8i_codec_aif_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_IN("AIF2 DACR", "AIF2 Playback", 1,
SUN8I_AIF2_DACDAT_CTRL,
SUN8I_AIF2_DACDAT_CTRL_AIF2_DACR_ENA, 0),
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
/* ADC Inputs (connected to analog codec DAPM context) */
SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0),
/* DAC Outputs (connected to analog codec DAPM context) */
SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0),
/* DAC Mixers */
SOC_MIXER_ARRAY("DACL Mixer", SND_SOC_NOPM, 0, 0,
sun8i_dac_mixer_controls),
SOC_MIXER_ARRAY("DACR Mixer", SND_SOC_NOPM, 0, 0,
sun8i_dac_mixer_controls),
};
static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = {
/* Clock Routes */
{ "AIF1CLK", NULL, "mod" },
{ "SYSCLK", NULL, "AIF1CLK" },
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
{ "CLK AIF1", NULL, "AIF1CLK" },
{ "CLK AIF1", NULL, "SYSCLK" },
{ "RST AIF1", NULL, "CLK AIF1" },
{ "AIF1 AD0L", NULL, "RST AIF1" },
{ "AIF1 AD0R", NULL, "RST AIF1" },
{ "AIF1 DA0L", NULL, "RST AIF1" },
{ "AIF1 DA0R", NULL, "RST AIF1" },
{ "CLK AIF2", NULL, "AIF2CLK" },
{ "CLK AIF2", NULL, "SYSCLK" },
{ "RST AIF2", NULL, "CLK AIF2" },
{ "AIF2 ADCL", NULL, "RST AIF2" },
{ "AIF2 ADCR", NULL, "RST AIF2" },
{ "AIF2 DACL", NULL, "RST AIF2" },
{ "AIF2 DACR", NULL, "RST AIF2" },
{ "CLK ADC", NULL, "SYSCLK" },
{ "RST ADC", NULL, "CLK ADC" },
{ "ADC", NULL, "RST ADC" },
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
{ "ADCL", NULL, "ADC" },
{ "ADCR", NULL, "ADC" },
{ "CLK DAC", NULL, "SYSCLK" },
{ "RST DAC", NULL, "CLK DAC" },
{ "DAC", NULL, "RST DAC" },
{ "DACL", NULL, "DAC" },
{ "DACR", NULL, "DAC" },
/* AIF "ADC" Output Routes */
{ "AIF1 AD0L", NULL, "AIF1 AD0L Stereo Mux" },
{ "AIF1 AD0R", NULL, "AIF1 AD0R Stereo Mux" },
{ "AIF2 ADCL", NULL, "AIF2 ADCL Stereo Mux" },
{ "AIF2 ADCR", NULL, "AIF2 ADCR Stereo Mux" },
/* AIF "ADC" Mono/Stereo Mux Routes */
{ "AIF1 AD0L Stereo Mux", "Stereo", "AIF1 AD0L Mixer" },
{ "AIF1 AD0L Stereo Mux", "Reverse Stereo", "AIF1 AD0R Mixer" },
{ "AIF1 AD0L Stereo Mux", "Sum Mono", "AIF1 AD0L Mixer" },
{ "AIF1 AD0L Stereo Mux", "Sum Mono", "AIF1 AD0R Mixer" },
{ "AIF1 AD0L Stereo Mux", "Mix Mono", "AIF1 AD0L Mixer" },
{ "AIF1 AD0L Stereo Mux", "Mix Mono", "AIF1 AD0R Mixer" },
{ "AIF1 AD0R Stereo Mux", "Stereo", "AIF1 AD0R Mixer" },
{ "AIF1 AD0R Stereo Mux", "Reverse Stereo", "AIF1 AD0L Mixer" },
{ "AIF1 AD0R Stereo Mux", "Sum Mono", "AIF1 AD0L Mixer" },
{ "AIF1 AD0R Stereo Mux", "Sum Mono", "AIF1 AD0R Mixer" },
{ "AIF1 AD0R Stereo Mux", "Mix Mono", "AIF1 AD0L Mixer" },
{ "AIF1 AD0R Stereo Mux", "Mix Mono", "AIF1 AD0R Mixer" },
{ "AIF2 ADCL Stereo Mux", "Stereo", "AIF2 ADCL Mixer" },
{ "AIF2 ADCL Stereo Mux", "Reverse Stereo", "AIF2 ADCR Mixer" },
{ "AIF2 ADCL Stereo Mux", "Sum Mono", "AIF2 ADCL Mixer" },
{ "AIF2 ADCL Stereo Mux", "Sum Mono", "AIF2 ADCR Mixer" },
{ "AIF2 ADCL Stereo Mux", "Mix Mono", "AIF2 ADCL Mixer" },
{ "AIF2 ADCL Stereo Mux", "Mix Mono", "AIF2 ADCR Mixer" },
{ "AIF2 ADCR Stereo Mux", "Stereo", "AIF2 ADCR Mixer" },
{ "AIF2 ADCR Stereo Mux", "Reverse Stereo", "AIF2 ADCL Mixer" },
{ "AIF2 ADCR Stereo Mux", "Sum Mono", "AIF2 ADCL Mixer" },
{ "AIF2 ADCR Stereo Mux", "Sum Mono", "AIF2 ADCR Mixer" },
{ "AIF2 ADCR Stereo Mux", "Mix Mono", "AIF2 ADCL Mixer" },
{ "AIF2 ADCR Stereo Mux", "Mix Mono", "AIF2 ADCR Mixer" },
/* AIF "ADC" Mixer Routes */
{ "AIF1 AD0L Mixer", "AIF1 Slot 0 Digital ADC Capture Switch", "AIF1 DA0L Stereo Mux" },
{ "AIF1 AD0L Mixer", "AIF2 Digital ADC Capture Switch", "AIF2 DACL Source" },
{ "AIF1 AD0L Mixer", "AIF1 Data Digital ADC Capture Switch", "ADCL" },
{ "AIF1 AD0L Mixer", "AIF2 Inv Digital ADC Capture Switch", "AIF2 DACR Source" },
{ "AIF1 AD0R Mixer", "AIF1 Slot 0 Digital ADC Capture Switch", "AIF1 DA0R Stereo Mux" },
{ "AIF1 AD0R Mixer", "AIF2 Digital ADC Capture Switch", "AIF2 DACR Source" },
{ "AIF1 AD0R Mixer", "AIF1 Data Digital ADC Capture Switch", "ADCR" },
{ "AIF1 AD0R Mixer", "AIF2 Inv Digital ADC Capture Switch", "AIF2 DACL Source" },
{ "AIF2 ADCL Mixer", "AIF2 ADC Mixer AIF1 DA0 Capture Switch", "AIF1 DA0L Stereo Mux" },
{ "AIF2 ADCL Mixer", "AIF2 ADC Mixer AIF2 DAC Rev Capture Switch", "AIF2 DACR Source" },
{ "AIF2 ADCL Mixer", "AIF2 ADC Mixer ADC Capture Switch", "ADCL" },
{ "AIF2 ADCR Mixer", "AIF2 ADC Mixer AIF1 DA0 Capture Switch", "AIF1 DA0R Stereo Mux" },
{ "AIF2 ADCR Mixer", "AIF2 ADC Mixer AIF2 DAC Rev Capture Switch", "AIF2 DACL Source" },
{ "AIF2 ADCR Mixer", "AIF2 ADC Mixer ADC Capture Switch", "ADCR" },
/* AIF "DAC" Input Mux Routes */
{ "AIF2 DACL Source", "AIF2", "AIF2 DACL Stereo Mux" },
{ "AIF2 DACL Source", "AIF2+3", "AIF2 DACL Stereo Mux" },
{ "AIF2 DACR Source", "AIF2", "AIF2 DACR Stereo Mux" },
{ "AIF2 DACR Source", "AIF3+2", "AIF2 DACR Stereo Mux" },
/* AIF "DAC" Mono/Stereo Mux Routes */
{ "AIF1 DA0L Stereo Mux", "Stereo", "AIF1 DA0L" },
{ "AIF1 DA0L Stereo Mux", "Reverse Stereo", "AIF1 DA0R" },
{ "AIF1 DA0L Stereo Mux", "Sum Mono", "AIF1 DA0L" },
{ "AIF1 DA0L Stereo Mux", "Sum Mono", "AIF1 DA0R" },
{ "AIF1 DA0L Stereo Mux", "Mix Mono", "AIF1 DA0L" },
{ "AIF1 DA0L Stereo Mux", "Mix Mono", "AIF1 DA0R" },
{ "AIF1 DA0R Stereo Mux", "Stereo", "AIF1 DA0R" },
{ "AIF1 DA0R Stereo Mux", "Reverse Stereo", "AIF1 DA0L" },
{ "AIF1 DA0R Stereo Mux", "Sum Mono", "AIF1 DA0L" },
{ "AIF1 DA0R Stereo Mux", "Sum Mono", "AIF1 DA0R" },
{ "AIF1 DA0R Stereo Mux", "Mix Mono", "AIF1 DA0L" },
{ "AIF1 DA0R Stereo Mux", "Mix Mono", "AIF1 DA0R" },
{ "AIF2 DACL Stereo Mux", "Stereo", "AIF2 DACL" },
{ "AIF2 DACL Stereo Mux", "Reverse Stereo", "AIF2 DACR" },
{ "AIF2 DACL Stereo Mux", "Sum Mono", "AIF2 DACL" },
{ "AIF2 DACL Stereo Mux", "Sum Mono", "AIF2 DACR" },
{ "AIF2 DACL Stereo Mux", "Mix Mono", "AIF2 DACL" },
{ "AIF2 DACL Stereo Mux", "Mix Mono", "AIF2 DACR" },
{ "AIF2 DACR Stereo Mux", "Stereo", "AIF2 DACR" },
{ "AIF2 DACR Stereo Mux", "Reverse Stereo", "AIF2 DACL" },
{ "AIF2 DACR Stereo Mux", "Sum Mono", "AIF2 DACL" },
{ "AIF2 DACR Stereo Mux", "Sum Mono", "AIF2 DACR" },
{ "AIF2 DACR Stereo Mux", "Mix Mono", "AIF2 DACL" },
{ "AIF2 DACR Stereo Mux", "Mix Mono", "AIF2 DACR" },
/* DAC Output Routes */
{ "DACL", NULL, "DACL Mixer" },
{ "DACR", NULL, "DACR Mixer" },
/* DAC Mixer Routes */
{ "DACL Mixer", "AIF1 Slot 0 Digital DAC Playback Switch", "AIF1 DA0L Stereo Mux" },
{ "DACL Mixer", "AIF2 Digital DAC Playback Switch", "AIF2 DACL Source" },
{ "DACL Mixer", "ADC Digital DAC Playback Switch", "ADCL" },
{ "DACR Mixer", "AIF1 Slot 0 Digital DAC Playback Switch", "AIF1 DA0R Stereo Mux" },
{ "DACR Mixer", "AIF2 Digital DAC Playback Switch", "AIF2 DACR Source" },
{ "DACR Mixer", "ADC Digital DAC Playback Switch", "ADCR" },
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
};
static const struct snd_soc_dapm_widget sun8i_codec_legacy_widgets[] = {
/* Legacy ADC Inputs (connected to analog codec DAPM context) */
SND_SOC_DAPM_ADC("AIF1 Slot 0 Left ADC", NULL, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_ADC("AIF1 Slot 0 Right ADC", NULL, SND_SOC_NOPM, 0, 0),
/* Legacy DAC Outputs (connected to analog codec DAPM context) */
SND_SOC_DAPM_DAC("AIF1 Slot 0 Left", NULL, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("AIF1 Slot 0 Right", NULL, SND_SOC_NOPM, 0, 0),
};
static const struct snd_soc_dapm_route sun8i_codec_legacy_routes[] = {
/* Legacy ADC Routes */
{ "ADCL", NULL, "AIF1 Slot 0 Left ADC" },
{ "ADCR", NULL, "AIF1 Slot 0 Right ADC" },
/* Legacy DAC Routes */
{ "AIF1 Slot 0 Left", NULL, "DACL" },
{ "AIF1 Slot 0 Right", NULL, "DACR" },
};
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
static int sun8i_codec_component_probe(struct snd_soc_component *component)
{
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
struct sun8i_codec *scodec = snd_soc_component_get_drvdata(component);
int ret;
/* Add widgets for backward compatibility with old device trees. */
if (scodec->quirks->legacy_widgets) {
ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_legacy_widgets,
ARRAY_SIZE(sun8i_codec_legacy_widgets));
if (ret)
return ret;
ret = snd_soc_dapm_add_routes(dapm, sun8i_codec_legacy_routes,
ARRAY_SIZE(sun8i_codec_legacy_routes));
if (ret)
return ret;
}
/*
* AIF1CLK and AIF2CLK share a pair of clock parents: PLL_AUDIO ("mod")
* and MCLK (from the CPU DAI connected to AIF1). MCLK's parent is also
* PLL_AUDIO, so using it adds no additional flexibility. Use PLL_AUDIO
* directly to simplify the clock tree.
*/
regmap_update_bits(scodec->regmap, SUN8I_SYSCLK_CTL,
SUN8I_SYSCLK_CTL_AIF1CLK_SRC_MASK |
SUN8I_SYSCLK_CTL_AIF2CLK_SRC_MASK,
SUN8I_SYSCLK_CTL_AIF1CLK_SRC_PLL |
SUN8I_SYSCLK_CTL_AIF2CLK_SRC_PLL);
/* Use AIF1CLK as the SYSCLK parent since AIF1 is used most often. */
regmap_update_bits(scodec->regmap, SUN8I_SYSCLK_CTL,
BIT(SUN8I_SYSCLK_CTL_SYSCLK_SRC),
SUN8I_SYSCLK_CTL_SYSCLK_SRC_AIF1CLK);
ASoC: sun8i-codec: Automatically set the system sample rate The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201014061941.4306-11-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-10-14 14:19:34 +08:00
/* Program the default sample rate. */
sun8i_codec_update_sample_rate(scodec);
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
return 0;
}
static const struct snd_soc_component_driver sun8i_soc_component = {
.dapm_widgets = sun8i_codec_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(sun8i_codec_dapm_widgets),
.dapm_routes = sun8i_codec_dapm_routes,
.num_dapm_routes = ARRAY_SIZE(sun8i_codec_dapm_routes),
.probe = sun8i_codec_component_probe,
.idle_bias_on = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
};
static const struct regmap_config sun8i_codec_regmap_config = {
.reg_bits = 32,
.reg_stride = 4,
.val_bits = 32,
.max_register = SUN8I_DAC_MXR_SRC,
.cache_type = REGCACHE_FLAT,
};
static int sun8i_codec_probe(struct platform_device *pdev)
{
struct sun8i_codec *scodec;
void __iomem *base;
int ret;
scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL);
if (!scodec)
return -ENOMEM;
scodec->clk_module = devm_clk_get(&pdev->dev, "mod");
if (IS_ERR(scodec->clk_module)) {
dev_err(&pdev->dev, "Failed to get the module clock\n");
return PTR_ERR(scodec->clk_module);
}
base = devm_platform_ioremap_resource(pdev, 0);
if (IS_ERR(base)) {
dev_err(&pdev->dev, "Failed to map the registers\n");
return PTR_ERR(base);
}
scodec->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "bus", base,
&sun8i_codec_regmap_config);
if (IS_ERR(scodec->regmap)) {
dev_err(&pdev->dev, "Failed to create our regmap\n");
return PTR_ERR(scodec->regmap);
}
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
scodec->quirks = of_device_get_match_data(&pdev->dev);
platform_set_drvdata(pdev, scodec);
pm_runtime_enable(&pdev->dev);
if (!pm_runtime_enabled(&pdev->dev)) {
ret = sun8i_codec_runtime_resume(&pdev->dev);
if (ret)
goto err_pm_disable;
}
ret = devm_snd_soc_register_component(&pdev->dev, &sun8i_soc_component,
sun8i_codec_dais,
ARRAY_SIZE(sun8i_codec_dais));
if (ret) {
dev_err(&pdev->dev, "Failed to register codec\n");
goto err_suspend;
}
return ret;
err_suspend:
if (!pm_runtime_status_suspended(&pdev->dev))
sun8i_codec_runtime_suspend(&pdev->dev);
err_pm_disable:
pm_runtime_disable(&pdev->dev);
return ret;
}
static int sun8i_codec_remove(struct platform_device *pdev)
{
pm_runtime_disable(&pdev->dev);
if (!pm_runtime_status_suspended(&pdev->dev))
sun8i_codec_runtime_suspend(&pdev->dev);
return 0;
}
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
static const struct sun8i_codec_quirks sun8i_a33_quirks = {
.legacy_widgets = true,
.lrck_inversion = true,
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
};
static const struct sun8i_codec_quirks sun50i_a64_quirks = {
};
static const struct of_device_id sun8i_codec_of_match[] = {
ASoC: sun8i-codec: Fix DAPM to match the hardware topology The A33/A64 digital codec has 4 physical inputs and 4 physical outputs: 3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by a 4-channel mixer connected to each output. The analog and digital sides of the ADC/DAC are in separate ASoC components, so card-level DAPM routes (provided in the device tree) are necessary to connect them together. Currently, these routes are wrong. For AIF1 Playback, the correct topology is: ||<<============ sun8i-codec ===========>>|| || || CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog) || || but the driver and device trees currently describe: || || CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog) || \--> DAC Mixer -> ??? [dead end] || For AIF1 Capture, there is an additional problem, because the Mixer route is backward. The topology should be: || || ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI || || but the driver and device trees currently describe: || || ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI || \--> ADC Mixer -> ??? [dead end] || The ADC/DAC are only powered because AIF1 AD0 (capture) has supply routes from the ADC, and AIF1 DA0 (playback) has supply routes from the DAC. However, neither set of supply routes matches the hardware topology. Audio can be routed among AIF1/2/3 without using the ADC or DAC at all; and audio can be routed from the ADC to the DAC without using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the DAPM routes are wrong, both of these use cases are currently broken. This commit adds the necessary widgets and routes to represent the real hardware topology, with functionality equivalent to the current driver. For the existing "allwinner,sun8i-a33-codec" compatible, widgets with the old names are kept as wrappers around the new widgets, so existing device trees will continue to work. For "allwinner,sun50i-a64-codec", the old widgets can be omitted, because no device trees yet use that compatible. Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-26 09:25:52 +08:00
{ .compatible = "allwinner,sun8i-a33-codec", .data = &sun8i_a33_quirks },
{ .compatible = "allwinner,sun50i-a64-codec", .data = &sun50i_a64_quirks },
{}
};
MODULE_DEVICE_TABLE(of, sun8i_codec_of_match);
static const struct dev_pm_ops sun8i_codec_pm_ops = {
SET_RUNTIME_PM_OPS(sun8i_codec_runtime_suspend,
sun8i_codec_runtime_resume, NULL)
};
static struct platform_driver sun8i_codec_driver = {
.driver = {
.name = "sun8i-codec",
.of_match_table = sun8i_codec_of_match,
.pm = &sun8i_codec_pm_ops,
},
.probe = sun8i_codec_probe,
.remove = sun8i_codec_remove,
};
module_platform_driver(sun8i_codec_driver);
MODULE_DESCRIPTION("Allwinner A33 (sun8i) codec driver");
MODULE_AUTHOR("Mylène Josserand <mylene.josserand@free-electrons.com>");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:sun8i-codec");