March 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v3 0/9] soundwire: qcom: various improvements
by Srinivas Kandagatla 08 Mar '21

08 Mar '21

Thanks for reviewing v2 of this patchset! During testing SoundWire controller on SM8250 MTP, we found few issues like all the interrupts are not handled, all transport parameters are not read from device tree. Patch to add Auto Enumeration supported by the controller is also part of this series. Other major issue was register read/writes which was interrupt based was an overhead and puts lot of limitation on context it can be used from. With previous approach number of interrupts generated after enumeration are around 130: $ cat /proc/interrupts | grep soundwire 21: 130 0 0 0 0 0 0 0 GICv3 234 Edge soundwire after this patch they are just 3 interrupts $ cat /proc/interrupts | grep soundwire 21: 3 0 0 0 0 0 0 0 GICv3 234 Edge soundwire So this patchset add various improvements to the existing driver to address above issues. Tested it on SM8250 MTP with 2x WSA881x speakers, HeadPhones on WCD938x via lpass-rx-macro and Analog MICs via lpass-tx-macro. Also tested on DragonBoard DB845c with 2xWSA881x speakers. Changes since v1: - added auto enumeration patches to this patchset - added comments to qcom bindings to clarify invalid values - cleanup trasport parameters to use internal port config Srinivas Kandagatla (9): dt-bindings: soundwire: qcom: clarify data port bus parameters soundwire: qcom: add support to missing transport params soundwire: qcom: set continue execution flag for ignored commands soundwire: qcom: start the clock during initialization soundwire: qcom: update register read/write routine soundwire: qcom: add support to new interrupts soundwire: export sdw_compare_devid() and sdw_extract_slave_id() soundwire: qcom: add auto enumeration support soundwire: qcom: wait for enumeration to be complete in probe .../bindings/soundwire/qcom,sdw.txt | 20 + drivers/soundwire/bus.c | 4 +- drivers/soundwire/qcom.c | 529 ++++++++++++++---- include/linux/soundwire/sdw.h | 2 + 4 files changed, 441 insertions(+), 114 deletions(-) -- 2.21.0

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[PATCH] ALSA: hda: Drop the BATCH workaround for AMD controllers
by Takashi Iwai 08 Mar '21

08 Mar '21

The commit c02f77d32d2c ("ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)") introduced a few workarounds for the recent AMD HD-audio controller, and one of them is the forced BATCH PCM mode so that PulseAudio avoids the timer-based scheduling. This was thought to cover for some badly working applications, but this actually worsens for more others. In total, this wasn't a good idea to enforce it. This is a partial revert of the commit above for dropping the PCM BATCH enforcement part to recover from the regression again. Fixes: c02f77d32d2c ("ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 9087981cd1f7..ca2f2ecd1488 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -609,13 +609,6 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) 20, 178000000); - /* by some reason, the playback stream stalls on PulseAudio with - * tsched=1 when a capture stream triggers. Until we figure out the - * real cause, disable tsched mode by telling the PCM info flag. - */ - if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) - runtime->hw.info |= SNDRV_PCM_INFO_BATCH; - if (chip->align_buffer_size) /* constrain buffer sizes to be multiple of 128 bytes. This is more efficient in terms of memory -- 2.26.2

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[PATCH v2] ASoC: wm8962: Relax bit clock divider searching
by Shengjiu Wang 08 Mar '21

08 Mar '21

With S20_3LE format case, the sysclk = rate * 384, the bclk = rate * 20 * 2, there is no proper bclk divider for 384 / 40, because current condition needs exact match. So driver fails to configure the clocking: wm8962 3-001a: Unsupported BCLK ratio 9 Fix this by relaxing bitclk divider searching, so that when no exact value can be derived from sysclk pick the closest value greater than expected bitclk. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- changes in v2: - Add debug message according to Charles's comments sound/soc/codecs/wm8962.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index ce4666a74793..34080f497584 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2403,6 +2403,7 @@ static const int sysclk_rates[] = { static void wm8962_configure_bclk(struct snd_soc_component *component) { struct wm8962_priv *wm8962 = snd_soc_component_get_drvdata(component); + int best, min_diff, diff; int dspclk, i; int clocking2 = 0; int clocking4 = 0; @@ -2473,23 +2474,25 @@ static void wm8962_configure_bclk(struct snd_soc_component *component) dev_dbg(component->dev, "DSPCLK is %dHz, BCLK %d\n", dspclk, wm8962->bclk); - /* We're expecting an exact match */ + /* Search a proper bclk, not exact match. */ + best = 0; + min_diff = INT_MAX; for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { if (bclk_divs[i] < 0) continue; - if (dspclk / bclk_divs[i] == wm8962->bclk) { - dev_dbg(component->dev, "Selected BCLK_DIV %d for %dHz\n", - bclk_divs[i], wm8962->bclk); - clocking2 |= i; + diff = (dspclk / bclk_divs[i]) - wm8962->bclk; + if (diff < 0) /* Table is sorted */ break; + if (diff < min_diff) { + best = i; + min_diff = diff; } } - if (i == ARRAY_SIZE(bclk_divs)) { - dev_err(component->dev, "Unsupported BCLK ratio %d\n", - dspclk / wm8962->bclk); - return; - } + wm8962->bclk = dspclk / bclk_divs[best]; + clocking2 |= best; + dev_dbg(component->dev, "Selected BCLK_DIV %d for %dHz\n", + bclk_divs[best], wm8962->bclk); aif2 |= wm8962->bclk / wm8962->lrclk; dev_dbg(component->dev, "Selected LRCLK divisor %d for %dHz\n", -- 2.27.0

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[PATCH 1/2] ASoC: remove remnants of sirf prima/atlas audio codec
by Peter Robinson 08 Mar '21

08 Mar '21

In 61fbeb5 the sirf prima/atlas drivers were removed. This cleans up a stray header and some Kconfig entries for the codec that were missed in the process. Fixes: 61fbeb5dcb3d (ASoC: remove sirf prima/atlas drivers) Signed-off-by: Peter Robinson <pbrobinson(a)gmail.com> Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 5 -- sound/soc/codecs/sirf-audio-codec.h | 124 ---------------------------- 2 files changed, 129 deletions(-) delete mode 100644 sound/soc/codecs/sirf-audio-codec.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e4cf14e66a51..1c87b42606c9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -186,7 +186,6 @@ config SND_SOC_ALL_CODECS imply SND_SOC_SI476X imply SND_SOC_SIMPLE_AMPLIFIER imply SND_SOC_SIMPLE_MUX - imply SND_SOC_SIRF_AUDIO_CODEC imply SND_SOC_SPDIF imply SND_SOC_SSM2305 imply SND_SOC_SSM2518 @@ -1279,10 +1278,6 @@ config SND_SOC_SIMPLE_MUX tristate "Simple Audio Mux" select GPIOLIB -config SND_SOC_SIRF_AUDIO_CODEC - tristate "SiRF SoC internal audio codec" - select REGMAP_MMIO - config SND_SOC_SPDIF tristate "S/PDIF CODEC" diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h deleted file mode 100644 index a7fe2680f4c7..000000000000 --- a/sound/soc/codecs/sirf-audio-codec.h +++ /dev/null @@ -1,124 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-or-later */ -/* - * SiRF inner codec controllers define - * - * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. - */ - -#ifndef _SIRF_AUDIO_CODEC_H -#define _SIRF_AUDIO_CODEC_H - - -#define AUDIO_IC_CODEC_PWR (0x00E0) -#define AUDIO_IC_CODEC_CTRL0 (0x00E4) -#define AUDIO_IC_CODEC_CTRL1 (0x00E8) -#define AUDIO_IC_CODEC_CTRL2 (0x00EC) -#define AUDIO_IC_CODEC_CTRL3 (0x00F0) - -#define MICBIASEN (1 << 3) - -#define IC_RDACEN (1 << 0) -#define IC_LDACEN (1 << 1) -#define IC_HSREN (1 << 2) -#define IC_HSLEN (1 << 3) -#define IC_SPEN (1 << 4) -#define IC_CPEN (1 << 5) - -#define IC_HPRSELR (1 << 6) -#define IC_HPLSELR (1 << 7) -#define IC_HPRSELL (1 << 8) -#define IC_HPLSELL (1 << 9) -#define IC_SPSELR (1 << 10) -#define IC_SPSELL (1 << 11) - -#define IC_MONOR (1 << 12) -#define IC_MONOL (1 << 13) - -#define IC_RXOSRSEL (1 << 28) -#define IC_CPFREQ (1 << 29) -#define IC_HSINVEN (1 << 30) - -#define IC_MICINREN (1 << 0) -#define IC_MICINLEN (1 << 1) -#define IC_MICIN1SEL (1 << 2) -#define IC_MICIN2SEL (1 << 3) -#define IC_MICDIFSEL (1 << 4) -#define IC_LINEIN1SEL (1 << 5) -#define IC_LINEIN2SEL (1 << 6) -#define IC_RADCEN (1 << 7) -#define IC_LADCEN (1 << 8) -#define IC_ALM (1 << 9) - -#define IC_DIGMICEN (1 << 22) -#define IC_DIGMICFREQ (1 << 23) -#define IC_ADC14B_12 (1 << 24) -#define IC_FIRDAC_HSL_EN (1 << 25) -#define IC_FIRDAC_HSR_EN (1 << 26) -#define IC_FIRDAC_LOUT_EN (1 << 27) -#define IC_POR (1 << 28) -#define IC_CODEC_CLK_EN (1 << 29) -#define IC_HP_3DB_BOOST (1 << 30) - -#define IC_ADC_LEFT_GAIN_SHIFT 16 -#define IC_ADC_RIGHT_GAIN_SHIFT 10 -#define IC_ADC_GAIN_MASK 0x3F -#define IC_MIC_MAX_GAIN 0x39 - -#define IC_RXPGAR_MASK 0x3F -#define IC_RXPGAR_SHIFT 14 -#define IC_RXPGAL_MASK 0x3F -#define IC_RXPGAL_SHIFT 21 -#define IC_RXPGAR 0x7B -#define IC_RXPGAL 0x7B - -#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK 0x3F -#define AUDIO_PORT_TX_FIFO_SC_OFFSET 0 -#define AUDIO_PORT_TX_FIFO_LC_OFFSET 10 -#define AUDIO_PORT_TX_FIFO_HC_OFFSET 20 - -#define TX_FIFO_SC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_SC_OFFSET) -#define TX_FIFO_LC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_LC_OFFSET) -#define TX_FIFO_HC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_HC_OFFSET) - -#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK 0x0F -#define AUDIO_PORT_RX_FIFO_SC_OFFSET 0 -#define AUDIO_PORT_RX_FIFO_LC_OFFSET 10 -#define AUDIO_PORT_RX_FIFO_HC_OFFSET 20 - -#define RX_FIFO_SC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_SC_OFFSET) -#define RX_FIFO_LC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_LC_OFFSET) -#define RX_FIFO_HC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_HC_OFFSET) -#define AUDIO_PORT_IC_CODEC_TX_CTRL (0x00F4) -#define AUDIO_PORT_IC_CODEC_RX_CTRL (0x00F8) - -#define AUDIO_PORT_IC_TXFIFO_OP (0x00FC) -#define AUDIO_PORT_IC_TXFIFO_LEV_CHK (0x0100) -#define AUDIO_PORT_IC_TXFIFO_STS (0x0104) -#define AUDIO_PORT_IC_TXFIFO_INT (0x0108) -#define AUDIO_PORT_IC_TXFIFO_INT_MSK (0x010C) - -#define AUDIO_PORT_IC_RXFIFO_OP (0x0110) -#define AUDIO_PORT_IC_RXFIFO_LEV_CHK (0x0114) -#define AUDIO_PORT_IC_RXFIFO_STS (0x0118) -#define AUDIO_PORT_IC_RXFIFO_INT (0x011C) -#define AUDIO_PORT_IC_RXFIFO_INT_MSK (0x0120) - -#define AUDIO_FIFO_START (1 << 0) -#define AUDIO_FIFO_RESET (1 << 1) - -#define AUDIO_FIFO_FULL (1 << 0) -#define AUDIO_FIFO_EMPTY (1 << 1) -#define AUDIO_FIFO_OFLOW (1 << 2) -#define AUDIO_FIFO_UFLOW (1 << 3) - -#define IC_TX_ENABLE (0x03) -#define IC_RX_ENABLE_MONO (0x01) -#define IC_RX_ENABLE_STEREO (0x03) - -#endif /*__SIRF_AUDIO_CODEC_H*/ -- 2.29.2

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[PATCH] ASoC: codecs/jz4770: Remove superfluous error message
by Tang Bin 08 Mar '21

08 Mar '21

The function devm_platform_ioremap_resource has already contained error message if failed, so remove superfluous dev_err here. Signed-off-by: Zhang Shengju <zhangshengju(a)cmss.chinamobile.com> Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/codecs/jz4770.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c index 298689a07..5a24471a5 100644 --- a/sound/soc/codecs/jz4770.c +++ b/sound/soc/codecs/jz4770.c @@ -900,11 +900,8 @@ static int jz4770_codec_probe(struct platform_device *pdev) codec->dev = dev; codec->base = devm_platform_ioremap_resource(pdev, 0); - if (IS_ERR(codec->base)) { - ret = PTR_ERR(codec->base); - dev_err(dev, "Failed to ioremap mmio memory: %d\n", ret); - return ret; - } + if (IS_ERR(codec->base)) + return PTR_ERR(codec->base); codec->regmap = devm_regmap_init(dev, NULL, codec, &jz4770_codec_regmap_config); -- 2.20.1.windows.1

3 2

[PATCH 1/2] ASoC: rt5645: Move rt5645_platform_data to sound/soc/codecs/rt5645.c
by Hans de Goede 08 Mar '21

08 Mar '21

sound/soc/codecs/rt5645.c is the only user of the rt5645_platform_data, move its definition to sound/soc/codecs/rt5645.c and remove the now empty include/sound/rt5645.h file. Note since the DMI quirk mechanism uses pointers to the rt5645_platform_data struct we can NOT simply add its members to the rt5645_priv struct and completely remove the struct. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- include/sound/rt5645.h | 32 -------------------------------- sound/soc/codecs/rt5645.c | 23 ++++++++++++++++++++++- sound/soc/codecs/rt5645.h | 2 -- 3 files changed, 22 insertions(+), 35 deletions(-) delete mode 100644 include/sound/rt5645.h diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h deleted file mode 100644 index 710c95be5509..000000000000 --- a/include/sound/rt5645.h +++ /dev/null @@ -1,32 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * linux/sound/rt5645.h -- Platform data for RT5645 - * - * Copyright 2013 Realtek Microelectronics - */ - -#ifndef __LINUX_SND_RT5645_H -#define __LINUX_SND_RT5645_H - -struct rt5645_platform_data { - /* IN2 can optionally be differential */ - bool in2_diff; - - unsigned int dmic1_data_pin; - /* 0 = IN2N; 1 = GPIO5; 2 = GPIO11 */ - unsigned int dmic2_data_pin; - /* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */ - - unsigned int jd_mode; - /* Use level triggered irq */ - bool level_trigger_irq; - /* Invert JD1_1 status polarity */ - bool inv_jd1_1; - /* Invert HP detect status polarity */ - bool inv_hp_pol; - - /* Value to asign to snd_soc_card.long_name */ - const char *long_name; -}; - -#endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 63a7e052eaa0..06ea6fce427c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -411,6 +411,27 @@ static const char *const rt5645_supply_names[] = { "cpvdd", }; +struct rt5645_platform_data { + /* IN2 can optionally be differential */ + bool in2_diff; + + unsigned int dmic1_data_pin; + /* 0 = IN2N; 1 = GPIO5; 2 = GPIO11 */ + unsigned int dmic2_data_pin; + /* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */ + + unsigned int jd_mode; + /* Use level triggered irq */ + bool level_trigger_irq; + /* Invert JD1_1 status polarity */ + bool inv_jd1_1; + /* Invert HP detect status polarity */ + bool inv_hp_pol; + + /* Value to assign to snd_soc_card.long_name */ + const char *long_name; +}; + struct rt5645_priv { struct snd_soc_component *component; struct rt5645_platform_data pdata; @@ -3834,7 +3855,7 @@ static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) static int rt5645_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct rt5645_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5645_platform_data *pdata = NULL; const struct dmi_system_id *dmi_data; struct rt5645_priv *rt5645; int ret, i; diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index e2d72ae17484..ac3de6f3bc2f 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -9,8 +9,6 @@ #ifndef __RT5645_H__ #define __RT5645_H__ -#include <sound/rt5645.h> - /* Info */ #define RT5645_RESET 0x00 #define RT5645_VENDOR_ID 0xfd -- 2.30.1

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[PATCH v2 00/15] Report jack and button detection + Capture Support
by Lucas Tanure 08 Mar '21

08 Mar '21

Hi All, Here is a patch series for reporting to user space jack and button events and add the support for Capture. With some cleanups and fixes along the way. Regards, Lucas Tanure Lucas Tanure (12): ASoC: cs42l42: Fix Bitclock polarity inversion ASoC: cs42l42: Fix channel width support ASoC: cs42l42: Fix mixer volume control ASoC: cs42l42: Don't enable/disable regulator at Bias Level ASoC: cs42l42: Always wait at least 3ms after reset ASoC: cs42l42: Remove power if the driver is being removed ASoC: cs42l42: Disable regulators if probe fails ASoC: cs42l42: Provide finer control on playback path ASoC: cs42l42: Set clock source for both ways of stream ASoC: cs42l42: Add Capture Support ASoC: cs42l42: Report jack and button detection ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called Richard Fitzgerald (3): ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT ASoC: cs42l42: Only start PLL if it is needed ASoC: cs42l42: Wait for PLL to lock before switching to it sound/soc/codecs/cs42l42.c | 437 +++++++++++++++++++++---------------- sound/soc/codecs/cs42l42.h | 41 +++- 2 files changed, 282 insertions(+), 196 deletions(-) -- 2.30.1

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[PATCH] ASoC: codecs: lpass-wsa-macro: fix RX MIX input controls
by Jonathan Marek 08 Mar '21

08 Mar '21

Attempting to use the RX MIX path at 48kHz plays at 96kHz, because these controls are incorrectly toggling the first bit of the register, which is part of the FS_RATE field. Fix the problem by using the same method used by the "WSA RX_MIX EC0_MUX" control, which is to use SND_SOC_NOPM as the register and use an enum in the shift field instead. Fixes: 2c4066e5d428 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route") Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> --- sound/soc/codecs/lpass-wsa-macro.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index f399f4dff5511..bd2561f9fb9fa 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1211,14 +1211,16 @@ static int wsa_macro_enable_mix_path(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - u16 gain_reg; + u16 path_reg, gain_reg; int val; - switch (w->reg) { - case CDC_WSA_RX0_RX_PATH_MIX_CTL: + switch (w->shift) { + case WSA_MACRO_RX_MIX0: + path_reg = CDC_WSA_RX0_RX_PATH_MIX_CTL; gain_reg = CDC_WSA_RX0_RX_VOL_MIX_CTL; break; - case CDC_WSA_RX1_RX_PATH_MIX_CTL: + case WSA_MACRO_RX_MIX1: + path_reg = CDC_WSA_RX1_RX_PATH_MIX_CTL; gain_reg = CDC_WSA_RX1_RX_VOL_MIX_CTL; break; default: @@ -1231,7 +1233,7 @@ static int wsa_macro_enable_mix_path(struct snd_soc_dapm_widget *w, snd_soc_component_write(component, gain_reg, val); break; case SND_SOC_DAPM_POST_PMD: - snd_soc_component_update_bits(component, w->reg, + snd_soc_component_update_bits(component, path_reg, CDC_WSA_RX_PATH_MIX_CLK_EN_MASK, CDC_WSA_RX_PATH_MIX_CLK_DISABLE); break; @@ -2068,14 +2070,14 @@ static const struct snd_soc_dapm_widget wsa_macro_dapm_widgets[] = { SND_SOC_DAPM_MUX("WSA_RX0 INP0", SND_SOC_NOPM, 0, 0, &rx0_prim_inp0_mux), SND_SOC_DAPM_MUX("WSA_RX0 INP1", SND_SOC_NOPM, 0, 0, &rx0_prim_inp1_mux), SND_SOC_DAPM_MUX("WSA_RX0 INP2", SND_SOC_NOPM, 0, 0, &rx0_prim_inp2_mux), - SND_SOC_DAPM_MUX_E("WSA_RX0 MIX INP", CDC_WSA_RX0_RX_PATH_MIX_CTL, - 0, 0, &rx0_mix_mux, wsa_macro_enable_mix_path, + SND_SOC_DAPM_MUX_E("WSA_RX0 MIX INP", SND_SOC_NOPM, WSA_MACRO_RX_MIX0, + 0, &rx0_mix_mux, wsa_macro_enable_mix_path, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("WSA_RX1 INP0", SND_SOC_NOPM, 0, 0, &rx1_prim_inp0_mux), SND_SOC_DAPM_MUX("WSA_RX1 INP1", SND_SOC_NOPM, 0, 0, &rx1_prim_inp1_mux), SND_SOC_DAPM_MUX("WSA_RX1 INP2", SND_SOC_NOPM, 0, 0, &rx1_prim_inp2_mux), - SND_SOC_DAPM_MUX_E("WSA_RX1 MIX INP", CDC_WSA_RX1_RX_PATH_MIX_CTL, - 0, 0, &rx1_mix_mux, wsa_macro_enable_mix_path, + SND_SOC_DAPM_MUX_E("WSA_RX1 MIX INP", SND_SOC_NOPM, WSA_MACRO_RX_MIX1, + 0, &rx1_mix_mux, wsa_macro_enable_mix_path, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER_E("WSA_RX INT0 MIX", SND_SOC_NOPM, 0, 0, NULL, 0, -- 2.26.1

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[PATCH] ASoC: codecs: lpass-va-macro: mute/unmute all active decimators
by Jonathan Marek 08 Mar '21

08 Mar '21

An interface can have multiple decimators enabled, so loop over all active decimators. Otherwise only one channel will be unmuted, and other channels will be zero. This fixes recording from dual DMIC as a single two channel stream. Also remove the now unused "active_decimator" field. Fixes: 908e6b1df26e ("ASoC: codecs: lpass-va-macro: Add support to VA Macro") Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> --- sound/soc/codecs/lpass-va-macro.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 91e6890d6efcb..3d6976a3d9e42 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -189,7 +189,6 @@ struct va_macro { struct device *dev; unsigned long active_ch_mask[VA_MACRO_MAX_DAIS]; unsigned long active_ch_cnt[VA_MACRO_MAX_DAIS]; - unsigned long active_decimator[VA_MACRO_MAX_DAIS]; u16 dmic_clk_div; int dec_mode[VA_MACRO_NUM_DECIMATORS]; @@ -549,11 +548,9 @@ static int va_macro_tx_mixer_put(struct snd_kcontrol *kcontrol, if (enable) { set_bit(dec_id, &va->active_ch_mask[dai_id]); va->active_ch_cnt[dai_id]++; - va->active_decimator[dai_id] = dec_id; } else { clear_bit(dec_id, &va->active_ch_mask[dai_id]); va->active_ch_cnt[dai_id]--; - va->active_decimator[dai_id] = -1; } snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, enable, update); @@ -880,18 +877,19 @@ static int va_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) struct va_macro *va = snd_soc_component_get_drvdata(component); u16 tx_vol_ctl_reg, decimator; - decimator = va->active_decimator[dai->id]; - - tx_vol_ctl_reg = CDC_VA_TX0_TX_PATH_CTL + - VA_MACRO_TX_PATH_OFFSET * decimator; - if (mute) - snd_soc_component_update_bits(component, tx_vol_ctl_reg, - CDC_VA_TX_PATH_PGA_MUTE_EN_MASK, - CDC_VA_TX_PATH_PGA_MUTE_EN); - else - snd_soc_component_update_bits(component, tx_vol_ctl_reg, - CDC_VA_TX_PATH_PGA_MUTE_EN_MASK, - CDC_VA_TX_PATH_PGA_MUTE_DISABLE); + for_each_set_bit(decimator, &va->active_ch_mask[dai->id], + VA_MACRO_DEC_MAX) { + tx_vol_ctl_reg = CDC_VA_TX0_TX_PATH_CTL + + VA_MACRO_TX_PATH_OFFSET * decimator; + if (mute) + snd_soc_component_update_bits(component, tx_vol_ctl_reg, + CDC_VA_TX_PATH_PGA_MUTE_EN_MASK, + CDC_VA_TX_PATH_PGA_MUTE_EN); + else + snd_soc_component_update_bits(component, tx_vol_ctl_reg, + CDC_VA_TX_PATH_PGA_MUTE_EN_MASK, + CDC_VA_TX_PATH_PGA_MUTE_DISABLE); + } return 0; } -- 2.26.1

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[PATCH] ASoC: fsl_xcvr: Use devm_platform_ioremap_resource_byname() to simplify code
by Tang Bin 08 Mar '21

08 Mar '21

In this function, devm_platform_ioremap_resource_byname() should be suitable to simplify code. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/fsl/fsl_xcvr.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 6dd0a5fcd455..5e8284db857b 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1131,7 +1131,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct fsl_xcvr *xcvr; - struct resource *ram_res, *regs_res, *rx_res, *tx_res; + struct resource *rx_res, *tx_res; void __iomem *regs; int ret, irq; @@ -1166,13 +1166,11 @@ static int fsl_xcvr_probe(struct platform_device *pdev) return PTR_ERR(xcvr->pll_ipg_clk); } - ram_res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "ram"); - xcvr->ram_addr = devm_ioremap_resource(dev, ram_res); + xcvr->ram_addr = devm_platform_ioremap_resource_byname(pdev, "ram"); if (IS_ERR(xcvr->ram_addr)) return PTR_ERR(xcvr->ram_addr); - regs_res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "regs"); - regs = devm_ioremap_resource(dev, regs_res); + regs = devm_platform_ioremap_resource_byname(pdev, "regs"); if (IS_ERR(regs)) return PTR_ERR(regs); -- 2.18.2

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