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[alsa-devel] soc-core: Questions about cpus->name && cpus->of_node
by Daniel Baluta 20 Jan '20

20 Jan '20

Hello Morimoto-san, I have a question about this snippet from sound/soc/soc-core.c » /* » * CPU device may be specified by either name or OF node, but » * can be left unspecified, and will be matched based on DAI » * name alone.. » */ » if (link->cpus->name && link->cpus->of_node) { » » dev_err(card->dev, » » » "ASoC: Neither/both cpu name/of_node are set for %s\n", » » » link->name); » » return -EINVAL; » } Here either the warning message is wrong OR the if () condition is wrong. If the link->cpus->name = NULL && link->cpus->of_node == NULL the warning message will be not be printed.

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Re: [alsa-devel] [PATCH 01/01] Add VID to support native DSD reproduction on FiiO devices.
by Mihai Luizescu 19 Jan '20

19 Jan '20

Tested-by: Mihai Luizescu <m.luizescu(a)gmail.com> --- I tested the patch today on Archlinux and I can confirm that it is working. I am able to play DSD files up to DSD256, the maximum supported by the sound card. I only compiled the snd_usb_audio module with the patch and replaced the stock one. DSD playback was confirmed by the green LED on the soundcard and by checking /proc/asound (altset 3 is used during DSD playback).

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Re: [alsa-devel] [PATCH 01/01] Add VID to support native DSD reproduction on FiiO devices.
by Mihai Luizescu 19 Jan '20

19 Jan '20

Tested-by: Mihai Luizescu <m.luizescu(a)gmail.com> --- I tested the patch today on Archlinux and I can confirm that it is working. I am able to play DSD files up to DSD256, the maximum supported by the sound card. I only compiled the snd_usb_audio module with the patch and replaced the stock one. DSD playback was confirmed by the green LED on the soundcard and by checking /proc/asound (altset 3 is used during DSD playback).

1 0

[alsa-devel] [PATCH] ASoC: atmel-pcm: Fix build error
by y00444492 19 Jan '20

19 Jan '20

From: Ye Bin <yebin10(a)huawei.com> If SND_ATMEL_SOC_SSC is y and SND_ATMEL_SOC_PDC is m, building fails: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': (.text+0x6f6): undefined reference to `atmel_pcm_pdc_platform_register' Select SND_ATMEL_SOC_PDC to fix this. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 92dfa6198623 ("ASoC: atmel-pcm: split into two file") Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/atmel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index f118c229ed82..78e7d40c3c24 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -19,11 +19,11 @@ config SND_ATMEL_SOC_DMA config SND_ATMEL_SOC_SSC tristate + select SND_ATMEL_SOC_PDC config SND_ATMEL_SOC_SSC_PDC tristate "SoC PCM DAI support for AT91 SSC controller using PDC" depends on ATMEL_SSC - select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC help Say Y or M if you want to add support for Atmel SSC interface -- 2.17.2

1 0

[alsa-devel] [GIT PULL]: soundwire updates for v5.6-rc1
by Vinod Koul 18 Jan '20

18 Jan '20

Hello Greg, Here are the updates for soundwire for v5.6-rc1. I have also shared tag 'sdw_interfaces_2_5.6' to ASoC folks, they might be merging that for cross tree dependency of ASoC drivers (soundwire slaves) The following changes since commit e42617b825f8073569da76dc4510bfa019b1c35a: Linux 5.5-rc1 (2019-12-08 14:57:55 -0800) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-5.6-rc1 for you to fetch changes up to 5098cae1f79cc0580dc2741ce250307a60451eca: dt-bindings: soundwire: fix example (2020-01-16 17:36:40 +0530) ---------------------------------------------------------------- soundwire updates for v5.6-rc1 This round we have bunch of updates to interfaces for ASoC (audio) subsystem by Intel and a new Qualcomm controller driver Details - Updates for sdw_slave interfaces for ASoC - Updates to cadence library and intel driver - New Soundwire controller for Qualcomm masters - Rework of device number assignment ---------------------------------------------------------------- Bard Liao (4): soundwire: intel: update headers for interrupts soundwire: intel: add link_list to handle interrupts with a single thread soundwire: intel: fix factor of two in MCLK handling soundwire: intel: report slave_ids for each link to SOF driver Pierre-Louis Bossart (16): soundwire: sdw_slave: add probe_complete structure and new fields soundwire: sdw_slave: add enumeration_complete structure soundwire: sdw_slave: add initialization_complete definition soundwire: sdw_slave: track unattach_request to handle all init sequences soundwire: intel: update interfaces between ASoC and SoundWire soundwire: intel: add mutex for shared SHIM register access soundwire: intel: add clock stop quirks soundwire: stream: remove redundant pr_err traces soundwire: cadence_master: filter out bad interrupts soundwire: cadence_master: log more useful information during timeouts soundwire: cadence_master: handle multiple status reports per Slave soundwire: bus: check first if Slaves become UNATTACHED soundwire: cadence: update kernel-doc parameter descriptions soundwire: cadence: remove useless variable incrementation soundwire: bus: fix device number leak on errors soundwire: cadence: fix kernel-doc parameter descriptions Rander Wang (4): soundwire: intel: update stream callbacks for hwparams/free stream operations soundwire: intel: add prototype for WAKEEN interrupt processing soundwire: cadence_master: clear interrupt status before enabling interrupt soundwire: cadence_master: remove config update for interrupt setting Srinivas Kandagatla (3): dt-bindings: soundwire: add bindings for Qcom controller soundwire: qcom: add support for SoundWire controller dt-bindings: soundwire: fix example Vinod Koul (1): Merge branch 'topic/sdw_intel' into next .../devicetree/bindings/soundwire/qcom,sdw.txt | 167 ++++ .../bindings/soundwire/soundwire-controller.yaml | 2 + drivers/soundwire/Kconfig | 9 + drivers/soundwire/Makefile | 4 + drivers/soundwire/bus.c | 55 +- drivers/soundwire/cadence_master.c | 66 +- drivers/soundwire/intel.c | 23 +- drivers/soundwire/intel.h | 13 +- drivers/soundwire/intel_init.c | 32 +- drivers/soundwire/qcom.c | 861 +++++++++++++++++++++ drivers/soundwire/stream.c | 8 - include/linux/soundwire/sdw.h | 23 +- include/linux/soundwire/sdw_intel.h | 167 +++- 13 files changed, 1351 insertions(+), 79 deletions(-) create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 drivers/soundwire/qcom.c Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH V6 0/4] update supported sample format
by Shengjiu Wang 18 Jan '20

18 Jan '20

This patch serial is to update the supported format for fsl_asrc and fix some format issue Shengjiu Wang (4): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 changes in v2 - extract snd_dmaengine_pcm_set_runtime_hwparams in one separate path. - 4th patch depends on 3rd patch changes in v3 - Fix build report by kbuild test robot <lkp(a)intel.com> - change snd_dmaengine_pcm_set_runtime_hwparams to snd_dmaengine_pcm_refine_runtime_hwparams changes in v4 - update according to Nicolin's comments. changes in v5 - free asrc pair when error happens in patch 4/4 changes in v6 - ignore return value of snd_soc_set_runtime_hwparams in patch 4/4 include/sound/dmaengine_pcm.h | 5 ++ sound/core/pcm_dmaengine.c | 83 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 64 ++++++++++++++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 61 ++------------------ 6 files changed, 199 insertions(+), 86 deletions(-) -- 2.21.0

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[alsa-devel] [PATCH 1/4] ASoC: wm_adsp: Correct cache handling of new kernel control API
by Charles Keepax 17 Jan '20

17 Jan '20

The recently added API that exposes firmware mixer controls to the kernel is missing cache handling and all writes bypass the cache, this obviously causes the cache to get out of sync with the hardware. Factor out the cache handling into two new helper functions and call those from both the normal ALSA control handlers and the new kernel API. Fixes: eb65ccdb0836 ("ASoC: wm_adsp: Expose mixer control API") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 98 ++++++++++++++++++++++++---------------------- 1 file changed, 51 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 2a9b610f6d435..d3d32b501acae 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1030,8 +1030,8 @@ static int wm_coeff_write_acked_control(struct wm_coeff_ctl *ctl, return -ETIMEDOUT; } -static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, - const void *buf, size_t len) +static int wm_coeff_write_ctrl_raw(struct wm_coeff_ctl *ctl, + const void *buf, size_t len) { struct wm_adsp *dsp = ctl->dsp; void *scratch; @@ -1061,6 +1061,23 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, return 0; } +static int wm_coeff_write_ctrl(struct wm_coeff_ctl *ctl, + const void *buf, size_t len) +{ + int ret = 0; + + if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) + ret = -EPERM; + else if (buf != ctl->cache) + memcpy(ctl->cache, buf, len); + + ctl->set = 1; + if (ctl->enabled && ctl->dsp->running) + ret = wm_coeff_write_ctrl_raw(ctl, buf, len); + + return ret; +} + static int wm_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { @@ -1071,16 +1088,7 @@ static int wm_coeff_put(struct snd_kcontrol *kctl, int ret = 0; mutex_lock(&ctl->dsp->pwr_lock); - - if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) - ret = -EPERM; - else - memcpy(ctl->cache, p, ctl->len); - - ctl->set = 1; - if (ctl->enabled && ctl->dsp->running) - ret = wm_coeff_write_control(ctl, p, ctl->len); - + ret = wm_coeff_write_ctrl(ctl, p, ctl->len); mutex_unlock(&ctl->dsp->pwr_lock); return ret; @@ -1096,15 +1104,10 @@ static int wm_coeff_tlv_put(struct snd_kcontrol *kctl, mutex_lock(&ctl->dsp->pwr_lock); - if (copy_from_user(ctl->cache, bytes, size)) { + if (copy_from_user(ctl->cache, bytes, size)) ret = -EFAULT; - } else { - ctl->set = 1; - if (ctl->enabled && ctl->dsp->running) - ret = wm_coeff_write_control(ctl, ctl->cache, size); - else if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) - ret = -EPERM; - } + else + ret = wm_coeff_write_ctrl(ctl, ctl->cache, size); mutex_unlock(&ctl->dsp->pwr_lock); @@ -1135,8 +1138,8 @@ static int wm_coeff_put_acked(struct snd_kcontrol *kctl, return ret; } -static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, - void *buf, size_t len) +static int wm_coeff_read_ctrl_raw(struct wm_coeff_ctl *ctl, + void *buf, size_t len) { struct wm_adsp *dsp = ctl->dsp; void *scratch; @@ -1166,29 +1169,37 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, return 0; } -static int wm_coeff_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int wm_coeff_read_ctrl(struct wm_coeff_ctl *ctl, void *buf, size_t len) { - struct soc_bytes_ext *bytes_ext = - (struct soc_bytes_ext *)kctl->private_value; - struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); - char *p = ucontrol->value.bytes.data; int ret = 0; - mutex_lock(&ctl->dsp->pwr_lock); - if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { if (ctl->enabled && ctl->dsp->running) - ret = wm_coeff_read_control(ctl, p, ctl->len); + return wm_coeff_read_ctrl_raw(ctl, buf, len); else - ret = -EPERM; + return -EPERM; } else { if (!ctl->flags && ctl->enabled && ctl->dsp->running) - ret = wm_coeff_read_control(ctl, ctl->cache, ctl->len); + ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, ctl->len); - memcpy(p, ctl->cache, ctl->len); + if (buf != ctl->cache) + memcpy(buf, ctl->cache, len); } + return ret; +} + +static int wm_coeff_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + char *p = ucontrol->value.bytes.data; + int ret; + + mutex_lock(&ctl->dsp->pwr_lock); + ret = wm_coeff_read_ctrl(ctl, p, ctl->len); mutex_unlock(&ctl->dsp->pwr_lock); return ret; @@ -1204,15 +1215,7 @@ static int wm_coeff_tlv_get(struct snd_kcontrol *kctl, mutex_lock(&ctl->dsp->pwr_lock); - if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { - if (ctl->enabled && ctl->dsp->running) - ret = wm_coeff_read_control(ctl, ctl->cache, size); - else - ret = -EPERM; - } else { - if (!ctl->flags && ctl->enabled && ctl->dsp->running) - ret = wm_coeff_read_control(ctl, ctl->cache, size); - } + ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, size); if (!ret && copy_to_user(bytes, ctl->cache, size)) ret = -EFAULT; @@ -1340,7 +1343,7 @@ static int wm_coeff_init_control_caches(struct wm_adsp *dsp) * created so we don't need to do anything. */ if (!ctl->flags || (ctl->flags & WMFW_CTL_FLAG_READABLE)) { - ret = wm_coeff_read_control(ctl, ctl->cache, ctl->len); + ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, ctl->len); if (ret < 0) return ret; } @@ -1358,7 +1361,8 @@ static int wm_coeff_sync_controls(struct wm_adsp *dsp) if (!ctl->enabled) continue; if (ctl->set && !(ctl->flags & WMFW_CTL_FLAG_VOLATILE)) { - ret = wm_coeff_write_control(ctl, ctl->cache, ctl->len); + ret = wm_coeff_write_ctrl_raw(ctl, ctl->cache, + ctl->len); if (ret < 0) return ret; } @@ -2048,7 +2052,7 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, if (len > ctl->len) return -EINVAL; - ret = wm_coeff_write_control(ctl, buf, len); + ret = wm_coeff_write_ctrl(ctl, buf, len); kcontrol = snd_soc_card_get_kcontrol(dsp->component->card, ctl->name); snd_ctl_notify(dsp->component->card->snd_card, @@ -2070,7 +2074,7 @@ int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, int type, if (len > ctl->len) return -EINVAL; - return wm_coeff_read_control(ctl, buf, len); + return wm_coeff_read_ctrl(ctl, buf, len); } EXPORT_SYMBOL_GPL(wm_adsp_read_ctl); -- 2.11.0

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[alsa-devel] [PATCH -next] ASoC: rt700: fix return value check in rt700_sdw_probe()
by Wei Yongjun 17 Jan '20

17 Jan '20

In case of error, the function devm_regmap_init() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Fixes: 7d2a5f9ae41e ("ASoC: rt700: add rt700 codec driver") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rt700-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c index 314103601af3..a4b95425886f 100644 --- a/sound/soc/codecs/rt700-sdw.c +++ b/sound/soc/codecs/rt700-sdw.c @@ -460,8 +460,8 @@ static int rt700_sdw_probe(struct sdw_slave *slave, regmap = devm_regmap_init(&slave->dev, NULL, &slave->dev, &rt700_regmap); - if (!regmap) - return -EINVAL; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); rt700_init(&slave->dev, sdw_regmap, regmap, slave);

2 1

[alsa-devel] [PATCH -next] ASoC: rt715: remove unused including <linux/version.h>
by YueHaibing 17 Jan '20

17 Jan '20

Remove including <linux/version.h> that don't need it. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/rt715.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 4c746938a062..2cbc57b16b13 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -10,7 +10,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h>

2 1

[alsa-devel] [PATCH] ASoC: rt1011: set the different setting for QFN/WLCSP package
by shumingf＠realtek.com 17 Jan '20

17 Jan '20

From: Shuming Fan <shumingf(a)realtek.com> The QFN package is a new one. There is a different initial setting to the chip of QFN and WLCSP package. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 17 +++++++++++++++-- sound/soc/codecs/rt1011.h | 1 + 2 files changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 2552073e54ce..dec5638060c3 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -40,7 +40,6 @@ static const struct reg_sequence init_list[] = { { RT1011_ADC_SET_5, 0x0a20 }, { RT1011_DAC_SET_2, 0xa032 }, - { RT1011_ADC_SET_1, 0x2925 }, { RT1011_SPK_PRO_DC_DET_1, 0xb00c }, { RT1011_SPK_PRO_DC_DET_2, 0xcccc }, @@ -2186,7 +2185,6 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) /* ADC/DAC setting */ regmap_write(rt1011->regmap, RT1011_ADC_SET_5, 0x0a20); regmap_write(rt1011->regmap, RT1011_DAC_SET_2, 0xe232); - regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x2925); regmap_write(rt1011->regmap, RT1011_ADC_SET_4, 0xc000); /* DC detection */ @@ -2235,8 +2233,18 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) dc_offset |= (value & 0xffff); dev_info(dev, "Gain1 offset=0x%x\n", dc_offset); + /* check the package info. */ + regmap_read(rt1011->regmap, RT1011_EFUSE_MATCH_DONE, &value); + if (value & 0x4) + rt1011->pack_id = 1; if (cali_flag) { + + if (rt1011->pack_id) + regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x292c); + else + regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x2925); + /* Class D on */ regmap_write(rt1011->regmap, RT1011_CLASS_D_POS, 0x010e); regmap_write(rt1011->regmap, @@ -2361,6 +2369,11 @@ static void rt1011_calibration_work(struct work_struct *work) rt1011_r0_load(rt1011); } + + if (rt1011->pack_id) + snd_soc_component_write(component, RT1011_ADC_SET_1, 0x292c); + else + snd_soc_component_write(component, RT1011_ADC_SET_1, 0x2925); } static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev) diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 68fadc15fa8c..f3a9a96640f1 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -692,6 +692,7 @@ struct rt1011_priv { unsigned int r0_reg, cali_done; unsigned int r0_calib, temperature_calib; int recv_spk_mode; + unsigned int pack_id; /* 0: WLCSP; 1: QFN */ }; #endif /* end of _RT1011_H_ */ -- 2.24.1

2 1

[alsa-devel] [PATCH] ASoC: SOF: Intel: hda: Fix SKL dai count
by Cezary Rojewski 17 Jan '20

17 Jan '20

With fourth pin added for iDisp for skl_dai, update SOF_SKL_DAI_NUM to account for the change. Without this, dais from the bottom of the list are skipped. In current state that's the case for 'Alt Analog CPU DAI'. Fixes: ac42b142cd76 ("ASoC: SOF: Intel: hda: Add iDisp4 DAI") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/sof/intel/hda.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 47408ec0de40..a4d030bfeee1 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -348,7 +348,7 @@ /* Number of DAIs */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) -#define SOF_SKL_NUM_DAIS 14 +#define SOF_SKL_NUM_DAIS 15 #else #define SOF_SKL_NUM_DAIS 8 #endif -- 2.17.1

3 4

[alsa-devel] [PATCH v2 1/2] ASoC: msm8916-wcd-analog: Simplify MIC BIAS Internal
by Stephan Gerhold 17 Jan '20

17 Jan '20

At the moment, MIC BIAS Internal* and MIC BIAS External* both reference the same register, and have a part of their initialization sequence duplicated. For example, the sequence for enabling MIC BIAS Internal1 is: I1. Enable MIC_BIAS1 supply (MICB_EN bit in CDC_A_MICB_1_EN) I2. Enable internal RBIAS (TX1_INT_RBIAS_EN bit in CDC_A_MICB_1_INT_RBIAS) The sequence for enabling MIC BIAS External1 is: E1. Enable MIC_BIAS1 supply (MICB_EN bit in CDC_A_MICB_1_EN) (E2. Ideally, make sure internal RBIAS is disabled. However, this should not happen in practice because DAPM will disable unused supplies...) Right now we have: SND_SOC_DAPM_SUPPLY("MIC BIAS Internal1", CDC_A_MICB_1_EN, 7, 0, ...) // I1 SND_SOC_DAPM_SUPPLY("MIC BIAS External1", CDC_A_MICB_1_EN, 7, 0, ...) // E1 and I2 is done in the PM event handler (pm8916_wcd_analog_enable_micbias_int1). We can simplify this by defining a common DAPM supply for I1/E1 ("MIC_BIAS1"), and one DAPM supply for I2 ("MIC BIAS Internal1"). Additional DAPM routes ensure that we also enable the MIC_BIAS1 supply for the internal and external pull up resistor. Another advantage of this is that we now disable the internal RBIAS when it is not needed. This makes it much easier to add support for MIC BIAS Internal3 as a next step. Tested-by: Nikita Travkin <nikitos.tr(a)gmail.com> # longcheer-l8150 Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- v1: https://lore.kernel.org/alsa-devel/20200111164006.43074-4-stephan@gerhold.n… Changes in v2: - Use an extra DAPM supply for MIC_BIASX that is referenced by MIC BIAS InternalX and MIC BIAS ExternalX, instead of combining it with MIC BIAS ExternalX - Clarify commit message --- sound/soc/codecs/msm8916-wcd-analog.c | 112 ++++++++++---------------- 1 file changed, 42 insertions(+), 70 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 1f7964beb20c..d6d727f5bae5 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -374,9 +374,8 @@ static void pm8916_wcd_analog_micbias_enable(struct snd_soc_component *component } -static int pm8916_wcd_analog_enable_micbias_ext(struct snd_soc_component - *component, int event, - int reg, unsigned int cap_mode) +static int pm8916_wcd_analog_enable_micbias(struct snd_soc_component *component, + int event, unsigned int cap_mode) { switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -389,72 +388,44 @@ static int pm8916_wcd_analog_enable_micbias_ext(struct snd_soc_component return 0; } -static int pm8916_wcd_analog_enable_micbias_int(struct snd_soc_component - *component, int event, - int reg, u32 cap_mode) +static int pm8916_wcd_analog_enable_micbias_int(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) { + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_component_update_bits(component, reg, MICB_1_EN_PULL_DOWN_EN_MASK, 0); snd_soc_component_update_bits(component, CDC_A_MICB_1_EN, MICB_1_EN_OPA_STG2_TAIL_CURR_MASK, MICB_1_EN_OPA_STG2_TAIL_CURR_1_60UA); - - break; - case SND_SOC_DAPM_POST_PMU: - pm8916_wcd_analog_micbias_enable(component); - snd_soc_component_update_bits(component, CDC_A_MICB_1_EN, - MICB_1_EN_BYP_CAP_MASK, cap_mode); break; } return 0; } -static int pm8916_wcd_analog_enable_micbias_ext1(struct - snd_soc_dapm_widget - *w, struct snd_kcontrol - *kcontrol, int event) +static int pm8916_wcd_analog_enable_micbias1(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 pm8916_wcd_analog_priv *wcd = snd_soc_component_get_drvdata(component); - return pm8916_wcd_analog_enable_micbias_ext(component, event, w->reg, - wcd->micbias1_cap_mode); + return pm8916_wcd_analog_enable_micbias(component, event, + wcd->micbias1_cap_mode); } -static int pm8916_wcd_analog_enable_micbias_ext2(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 pm8916_wcd_analog_priv *wcd = snd_soc_component_get_drvdata(component); - - return pm8916_wcd_analog_enable_micbias_ext(component, event, w->reg, - wcd->micbias2_cap_mode); - -} - -static int pm8916_wcd_analog_enable_micbias_int1(struct - snd_soc_dapm_widget - *w, struct snd_kcontrol - *kcontrol, int event) +static int pm8916_wcd_analog_enable_micbias2(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 pm8916_wcd_analog_priv *wcd = snd_soc_component_get_drvdata(component); - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - snd_soc_component_update_bits(component, CDC_A_MICB_1_INT_RBIAS, - MICB_1_INT_TX1_INT_RBIAS_EN_MASK, - MICB_1_INT_TX1_INT_RBIAS_EN_ENABLE); - break; - } + return pm8916_wcd_analog_enable_micbias(component, event, + wcd->micbias2_cap_mode); - return pm8916_wcd_analog_enable_micbias_int(component, event, w->reg, - wcd->micbias1_cap_mode); } static int pm8916_mbhc_configure_bias(struct pm8916_wcd_analog_priv *priv, @@ -564,9 +535,8 @@ static int pm8916_wcd_analog_enable_micbias_int2(struct switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_component_update_bits(component, CDC_A_MICB_1_INT_RBIAS, - MICB_1_INT_TX2_INT_RBIAS_EN_MASK, - MICB_1_INT_TX2_INT_RBIAS_EN_ENABLE); + snd_soc_component_update_bits(component, CDC_A_MICB_2_EN, + CDC_A_MICB_2_PULL_DOWN_EN_MASK, 0); break; case SND_SOC_DAPM_POST_PMU: pm8916_mbhc_configure_bias(wcd, true); @@ -576,8 +546,7 @@ static int pm8916_wcd_analog_enable_micbias_int2(struct break; } - return pm8916_wcd_analog_enable_micbias_int(component, event, w->reg, - wcd->micbias2_cap_mode); + return pm8916_wcd_analog_enable_micbias_int(w, kcontrol, event); } static int pm8916_wcd_analog_enable_adc(struct snd_soc_dapm_widget *w, @@ -878,14 +847,15 @@ static const struct snd_soc_dapm_route pm8916_wcd_analog_audio_map[] = { {"SPK PA", NULL, "SPK DAC"}, {"SPK DAC", "Switch", "PDM_RX3"}, - {"MIC BIAS Internal1", NULL, "INT_LDO_H"}, - {"MIC BIAS Internal2", NULL, "INT_LDO_H"}, - {"MIC BIAS External1", NULL, "INT_LDO_H"}, - {"MIC BIAS External2", NULL, "INT_LDO_H"}, - {"MIC BIAS Internal1", NULL, "vdd-micbias"}, - {"MIC BIAS Internal2", NULL, "vdd-micbias"}, - {"MIC BIAS External1", NULL, "vdd-micbias"}, - {"MIC BIAS External2", NULL, "vdd-micbias"}, + {"MIC_BIAS1", NULL, "INT_LDO_H"}, + {"MIC_BIAS2", NULL, "INT_LDO_H"}, + {"MIC_BIAS1", NULL, "vdd-micbias"}, + {"MIC_BIAS2", NULL, "vdd-micbias"}, + + {"MIC BIAS External1", NULL, "MIC_BIAS1"}, + {"MIC BIAS Internal1", NULL, "MIC_BIAS1"}, + {"MIC BIAS External2", NULL, "MIC_BIAS2"}, + {"MIC BIAS Internal2", NULL, "MIC_BIAS2"}, }; static const struct snd_soc_dapm_widget pm8916_wcd_analog_dapm_widgets[] = { @@ -937,22 +907,24 @@ static const struct snd_soc_dapm_widget pm8916_wcd_analog_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("RX_BIAS", CDC_A_RX_COM_BIAS_DAC, 7, 0, NULL, 0), /* TX */ - SND_SOC_DAPM_SUPPLY("MIC BIAS Internal1", CDC_A_MICB_1_EN, 7, 0, - pm8916_wcd_analog_enable_micbias_int1, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SUPPLY("MIC BIAS Internal2", CDC_A_MICB_2_EN, 7, 0, + SND_SOC_DAPM_SUPPLY("MIC_BIAS1", CDC_A_MICB_1_EN, 7, 0, + pm8916_wcd_analog_enable_micbias1, + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("MIC_BIAS2", CDC_A_MICB_2_EN, 7, 0, + pm8916_wcd_analog_enable_micbias2, + SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_SUPPLY("MIC BIAS External1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC BIAS External2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("MIC BIAS Internal1", CDC_A_MICB_1_INT_RBIAS, 7, 0, + pm8916_wcd_analog_enable_micbias_int, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("MIC BIAS Internal2", CDC_A_MICB_1_INT_RBIAS, 4, 0, pm8916_wcd_analog_enable_micbias_int2, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SUPPLY("MIC BIAS External1", CDC_A_MICB_1_EN, 7, 0, - pm8916_wcd_analog_enable_micbias_ext1, - SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_SUPPLY("MIC BIAS External2", CDC_A_MICB_2_EN, 7, 0, - pm8916_wcd_analog_enable_micbias_ext2, - SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_ADC_E("ADC1", NULL, CDC_A_TX_1_EN, 7, 0, pm8916_wcd_analog_enable_adc, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | -- 2.24.1

3 5

[alsa-devel] [PATCH -next] ASoC: rt711: fix return value check in rt711_sdw_probe()
by Wei Yongjun 17 Jan '20

17 Jan '20

In case of error, the function devm_regmap_init() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Fixes: 320b8b0d13b8 ("ASoC: rt711: add rt711 codec driver") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rt711-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index e28dc84ede5b..85e62e1059cd 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -460,8 +460,8 @@ static int rt711_sdw_probe(struct sdw_slave *slave, regmap = devm_regmap_init(&slave->dev, NULL, &slave->dev, &rt711_regmap); - if (!regmap) - return -EINVAL; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); rt711_init(&slave->dev, sdw_regmap, regmap, slave);

2 1

[alsa-devel] [PATCH -next] ASoC: rt715: fix return value check in rt715_sdw_probe()
by Wei Yongjun 17 Jan '20

17 Jan '20

In case of error, the function devm_regmap_init() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Fixes: d1ede0641b05 ("ASoC: rt715: add RT715 codec driver") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rt715-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index 18868e4ae6e8..6d892c44c522 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -535,8 +535,8 @@ static int rt715_sdw_probe(struct sdw_slave *slave, regmap = devm_regmap_init(&slave->dev, NULL, &slave->dev, &rt715_regmap); - if (!regmap) - return -EINVAL; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); rt715_init(&slave->dev, sdw_regmap, regmap, slave);

2 1

[alsa-devel] [PATCH] ASoC: amd: Additional DAI for I2S SP instance.
by Ravulapati Vishnu vardhan rao 17 Jan '20

17 Jan '20

I2S SP instance has separate BCLK and LRCLK for Tx and Rx. Creating additional DAI for Rx. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x.h | 2 +- sound/soc/amd/raven/pci-acp3x.c | 8 +++++++- 2 files changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index b6a80dc..21e7ac0 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -13,7 +13,7 @@ #define TDM_ENABLE 1 #define TDM_DISABLE 0 -#define ACP3x_DEVS 3 +#define ACP3x_DEVS 4 #define ACP3x_PHY_BASE_ADDRESS 0x1240000 #define ACP3x_I2S_MODE 0 #define ACP3x_REG_START 0x1240000 diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index 2f9f5290..65330bb 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -225,7 +225,13 @@ static int snd_acp3x_probe(struct pci_dev *pci, pdevinfo[2].id = 1; pdevinfo[2].parent = &pci->dev; pdevinfo[2].num_res = 1; - pdevinfo[2].res = &adata->res[2]; + pdevinfo[2].res = &adata->res[1]; + + pdevinfo[3].name = "acp3x_i2s_playcap"; + pdevinfo[3].id = 2; + pdevinfo[3].parent = &pci->dev; + pdevinfo[3].num_res = 1; + pdevinfo[3].res = &adata->res[2]; for (i = 0; i < ACP3x_DEVS; i++) { adata->pdev[i] = platform_device_register_full(&pdevinfo[i]); -- 2.7.4

2 1

[alsa-devel] [PATCH 0/4] ASoC: Intel: 4 small cleanups
by Hans de Goede 17 Jan '20

17 Jan '20

Hi All, Here are 4 ASoC/Intel cleanup patches, these were written to get to know the kernel-devel process by a group of students which I was coaching on a kernel project. Their original submission needed some cleanup which they never got around to. So I've done the cleanup now and I'm submitting a cleaned-up version of the cleanup patches. Regards, Hans

3 9

[alsa-devel] HP ALC671 support headset Mic
by Kailang 17 Jan '20

17 Jan '20

Hi Takashi, HP cPC need to support headset Mic. Attach patch was the support patch. BR, Kailang

2 3

[alsa-devel] simple-card: DAI link creation
by Daniel Baluta 17 Jan '20

17 Jan '20

Hello Morimoto-san, I want to know your opinion on enhancing simple-card in order to be able to create more flexible links. So far, simple-card allows the following cases: 1) normal sound all DAIs are detected as CPU - Codec (+optional platform). 2) DPCM sound 2a) all CPUs are detected CPU - <dummy_codec> [FE] 2b) all codecs are detected <dummy_cpu> - codec [BE] I will need to have a third case where I need to create a back end link like this: link->cpus->of_node = <&my_dai>; link->codecs->of_node = <&my_codec>; link->platforms->of_node = <&my_platform>; link->no_pcm = 1; Somehow, similar with qcom_snd_parse_of from sound/soc/qcom/common.c. Do you think it would be OK to create another file to handle this case in sound/soc/generic? Also, I would really like one day to really understand the role of cpu/platform/codec for a FE/BE. In the case of the FE: - cpu dai - platform - codec, here it makes sense to be dummy codec. In the case of BE: - cpu_dai - platform - codec, here it make sense to be reference to a real codec In the case of normal link: - cpu_dai, here it makes sense is a physical DAI - platform - codec, here it makes sense it is a physical codec. thanks, Daniel.

2 1

[alsa-devel] [PATCH RFC v2 0/4] ASoC: Add Multi CPU DAI support
by Bard liao 17 Jan '20

17 Jan '20

As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html changes in v2: - rebase on asoc-next - fix some typo - compare and merge Kuninori Morimoto's version - add warning if the function is not support multi cpu yet Bard liao (1): ASoC: add warning if the function is not support multi cpu yet. Shreyas NC (3): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add multiple CPU DAI support in DAPM include/sound/soc.h | 15 + sound/soc/soc-compress.c | 7 +- sound/soc/soc-core.c | 222 ++++++----- sound/soc/soc-dapm.c | 131 +++--- sound/soc/soc-generic-dmaengine-pcm.c | 12 + sound/soc/soc-pcm.c | 547 +++++++++++++++++--------- 6 files changed, 599 insertions(+), 335 deletions(-) -- 2.17.1

4 15

[alsa-devel] [PATCH v2] ASoC: dt-bindings: stm32: convert spdfirx to json-schema
by Olivier Moysan 17 Jan '20

17 Jan '20

Convert the STM32 SPDIFRX bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- Changes in v2: - Add "additionalProperties: false" - Also change minItems to 2 for dmas property, as both DMAs are required. --- .../bindings/sound/st,stm32-spdifrx.txt | 56 ------------- .../bindings/sound/st,stm32-spdifrx.yaml | 84 +++++++++++++++++++ 2 files changed, 84 insertions(+), 56 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt deleted file mode 100644 index 33826f2459fa..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt +++ /dev/null @@ -1,56 +0,0 @@ -STMicroelectronics STM32 S/PDIF receiver (SPDIFRX). - -The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with -IEC-60958 and IEC-61937. - -Required properties: - - compatible: should be "st,stm32h7-spdifrx" - - reg: cpu DAI IP base address and size - - clocks: must contain an entry for kclk (used as S/PDIF signal reference) - - clock-names: must contain "kclk" - - interrupts: cpu DAI interrupt line - - dmas: DMA specifiers for audio data DMA and iec control flow DMA - See STM32 DMA bindings, Documentation/devicetree/bindings/dma/stm32-dma.txt - - dma-names: two dmas have to be defined, "rx" and "rx-ctrl" - -Optional properties: - - resets: Reference to a reset controller asserting the SPDIFRX - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -spdifrx: spdifrx@40004000 { - compatible = "st,stm32h7-spdifrx"; - reg = <0x40004000 0x400>; - clocks = <&rcc SPDIFRX_CK>; - clock-names = "kclk"; - interrupts = <97>; - dmas = <&dmamux1 2 93 0x400 0x0>, - <&dmamux1 3 94 0x400 0x0>; - dma-names = "rx", "rx-ctrl"; - pinctrl-0 = <&spdifrx_pins>; - pinctrl-names = "default"; - - spdifrx_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - }; - }; -}; - -spdif_in: spdif-in { - compatible = "linux,spdif-dir"; - - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; - -soundcard { - compatible = "audio-graph-card"; - dais = <&spdifrx_port>; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml new file mode 100644 index 000000000000..e19313e7a552 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-spdifrx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 S/PDIF receiver (SPDIFRX) + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: | + The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with + IEC-60958 and IEC-61937. + +properties: + compatible: + enum: + - st,stm32h7-spdifrx + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: kclk + + interrupts: + maxItems: 1 + + dmas: + description: | + Two DMA channel specifiers are requested: + - audio data capture DMA + - IEC status bits capture DMA + See STM32 DMA bindings /bindings/dma/stm32-dma.txt. + minItems: 2 + maxItems: 2 + + dma-names: + items: + - const: rx + - const: rx-ctrl + + resets: + maxItems: 1 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + spdifrx: spdifrx@40004000 { + compatible = "st,stm32h7-spdifrx"; + #sound-dai-cells = <0>; + reg = <0x40004000 0x400>; + clocks = <&rcc SPDIF_K>; + clock-names = "kclk"; + interrupts = <GIC_SPI 97 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 2 93 0x400 0x0>, + <&dmamux1 3 94 0x400 0x0>; + dma-names = "rx", "rx-ctrl"; + pinctrl-0 = <&spdifrx_pins>; + pinctrl-names = "default"; + }; + +... -- 2.17.1

2 1

[alsa-devel] [PATCH AUTOSEL 4.4 122/174] ASoC: wm8737: Fix copy-paste error in wm8737_snd_controls
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit 554b75bde64bcad9662530726d1483f7ef012069 ] sound/soc/codecs/wm8737.c:112:29: warning: high_3d defined but not used [-Wunused-const-variable=] 'high_3d' should be used for 3D High Cut-off. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 2a9ae13a2641 ("ASoC: Add initial WM8737 driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20190815091920.64480-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8737.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index e7807601e675..ae69cb790ac3 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -170,7 +170,7 @@ SOC_DOUBLE("Polarity Invert Switch", WM8737_ADC_CONTROL, 5, 6, 1, 0), SOC_SINGLE("3D Switch", WM8737_3D_ENHANCE, 0, 1, 0), SOC_SINGLE("3D Depth", WM8737_3D_ENHANCE, 1, 15, 0), SOC_ENUM("3D Low Cut-off", low_3d), -SOC_ENUM("3D High Cut-off", low_3d), +SOC_ENUM("3D High Cut-off", high_3d), SOC_SINGLE_TLV("3D ADC Volume", WM8737_3D_ENHANCE, 7, 1, 1, adc_tlv), SOC_SINGLE("Noise Gate Switch", WM8737_NOISE_GATE, 0, 1, 0), -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.4 121/174] ASoC: cs4349: Use PM ops 'cs4349_runtime_pm'
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit 9b4275c415acca6264a3d7f1182589959c93d530 ] sound/soc/codecs/cs4349.c:358:32: warning: cs4349_runtime_pm defined but not used [-Wunused-const-variable=] cs4349_runtime_pm ops already defined, it seems we should enable it. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: e40da86 ("ASoC: cs4349: Add support for Cirrus Logic CS4349") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20190815090157.70036-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs4349.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 0ac8fc5ed4ae..9ebd500ecf38 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -379,6 +379,7 @@ static struct i2c_driver cs4349_i2c_driver = { .driver = { .name = "cs4349", .of_match_table = cs4349_of_match, + .pm = &cs4349_runtime_pm, }, .id_table = cs4349_i2c_id, .probe = cs4349_i2c_probe, -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.4 120/174] ASoC: es8328: Fix copy-paste error in es8328_right_line_controls
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit 630742c296341a8cfe00dfd941392025ba8dd4e8 ] It seems 'es8328_rline_enum' should be used in es8328_right_line_controls Fixes: 567e4f98922c ("ASoC: add es8328 codec driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20190815092300.68712-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index afa6c5db9dcc..2bf30d0eb82f 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -210,7 +210,7 @@ static const struct soc_enum es8328_rline_enum = ARRAY_SIZE(es8328_line_texts), es8328_line_texts); static const struct snd_kcontrol_new es8328_right_line_controls = - SOC_DAPM_ENUM("Route", es8328_lline_enum); + SOC_DAPM_ENUM("Route", es8328_rline_enum); /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.4 115/174] ALSA: aoa: onyx: always initialize register read value
by Sasha Levin 16 Jan '20

16 Jan '20

From: Johannes Berg <johannes(a)sipsolutions.net> [ Upstream commit f474808acb3c4b30552d9c59b181244e0300d218 ] A lot of places in the driver use onyx_read_register() without checking the return value, and it's been working OK for ~10 years or so, so probably never fails ... Rather than trying to check the return value everywhere, which would be relatively intrusive, at least make sure we don't use an uninitialized value. Fixes: f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/aoa/codecs/onyx.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index a04edff8b729..ae50d59fb810 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -74,8 +74,10 @@ static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value) return 0; } v = i2c_smbus_read_byte_data(onyx->i2c, reg); - if (v < 0) + if (v < 0) { + *value = 0; return -1; + } *value = (u8)v; onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value; return 0; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.4 103/174] ASoC: ti: davinci-mcasp: Fix slot mask settings when using multiple AXRs
by Sasha Levin 16 Jan '20

16 Jan '20

From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> [ Upstream commit fd14f4436fd47d5418023c90e933e66d3645552e ] If multiple serializers are connected in the system and the number of channels will need to use more than one serializer the mask to enable the serializers were left to 0 if tdm_mask is provided Fixes: dd55ff8346a97 ("ASoC: davinci-mcasp: Add set_tdm_slots() support") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/davinci/davinci-mcasp.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 2f7be6cee98e..fc0a73227b02 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -875,14 +875,13 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, active_slots = hweight32(mcasp->tdm_mask[stream]); active_serializers = (channels + active_slots - 1) / active_slots; - if (active_serializers == 1) { + if (active_serializers == 1) active_slots = channels; - for (i = 0; i < total_slots; i++) { - if ((1 << i) & mcasp->tdm_mask[stream]) { - mask |= (1 << i); - if (--active_slots <= 0) - break; - } + for (i = 0; i < total_slots; i++) { + if ((1 << i) & mcasp->tdm_mask[stream]) { + mask |= (1 << i); + if (--active_slots <= 0) + break; } } } else { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.4 083/174] ASoC: fix valid stream condition
by Sasha Levin 16 Jan '20

16 Jan '20

From: Jerome Brunet <jbrunet(a)baylibre.com> [ Upstream commit 6a7c59c6d9f3b280e81d7a04bbe4e55e90152dce ] A stream may specify a rate range using 'rate_min' and 'rate_max', so a stream may be valid and not specify any rates. However, as stream cannot be valid and not have any channel. Let's use this condition instead to determine if a stream is valid or not. Fixes: cde79035c6cf ("ASoC: Handle multiple codecs with split playback / capture") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 78813057167d..dbdea1975f90 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -48,8 +48,8 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) else codec_stream = &dai->driver->capture; - /* If the codec specifies any rate at all, it supports the stream. */ - return codec_stream->rates; + /* If the codec specifies any channels at all, it supports the stream */ + return codec_stream->channels_min; } /** -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.4 081/174] ALSA: usb-audio: Handle the error from snd_usb_mixer_apply_create_quirk()
by Sasha Levin 16 Jan '20

16 Jan '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 328e9f6973be2ee67862cb17bf6c0c5c5918cd72 ] The error from snd_usb_mixer_apply_create_quirk() is ignored in the current usb-audio driver code, which will continue the probing even after the error. Let's take it more serious. Fixes: 7b1eda223deb ("ALSA: usb-mixer: factor out quirks") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 1b81f18010d2..73149b9be29c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2552,7 +2552,9 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, (err = snd_usb_mixer_status_create(mixer)) < 0) goto _error; - snd_usb_mixer_apply_create_quirk(mixer); + err = snd_usb_mixer_apply_create_quirk(mixer); + if (err < 0) + goto _error; err = snd_device_new(chip->card, SNDRV_DEV_CODEC, mixer, &dev_ops); if (err < 0) -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.4 050/174] ASoC: qcom: Fix of-node refcount unbalance in apq8016_sbc_parse_of()
by Sasha Levin 16 Jan '20

16 Jan '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 8d1667200850f8753c0265fa4bd25c9a6e5f94ce ] The apq8016 driver leaves the of-node refcount at aborting from the loop of for_each_child_of_node() in the error path. Not only the iterator node of for_each_child_of_node(), the children nodes referred from it for codec and cpu have to be properly unreferenced. Fixes: bdb052e81f62 ("ASoC: qcom: add apq8016 sound card support") Cc: Patrick Lai <plai(a)codeaurora.org> Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/apq8016_sbc.c | 21 ++++++++++++++++----- 1 file changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 1efdf0088ecd..886f2027e671 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -98,13 +98,15 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) if (!cpu || !codec) { dev_err(dev, "Can't find cpu/codec DT node\n"); - return ERR_PTR(-EINVAL); + ret = -EINVAL; + goto error; } link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); if (!link->cpu_of_node) { dev_err(card->dev, "error getting cpu phandle\n"); - return ERR_PTR(-EINVAL); + ret = -EINVAL; + goto error; } link->codec_of_node = of_parse_phandle(codec, "sound-dai", 0); @@ -116,13 +118,13 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); if (ret) { dev_err(card->dev, "error getting cpu dai name\n"); - return ERR_PTR(ret); + goto error; } ret = snd_soc_of_get_dai_name(codec, &link->codec_dai_name); if (ret) { dev_err(card->dev, "error getting codec dai name\n"); - return ERR_PTR(ret); + goto error; } link->platform_of_node = link->cpu_of_node; @@ -132,15 +134,24 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) ret = of_property_read_string(np, "link-name", &link->name); if (ret) { dev_err(card->dev, "error getting codec dai_link name\n"); - return ERR_PTR(ret); + goto error; } link->stream_name = link->name; link->init = apq8016_sbc_dai_init; link++; + + of_node_put(cpu); + of_node_put(codec); } return data; + + error: + of_node_put(np); + of_node_put(cpu); + of_node_put(codec); + return ERR_PTR(ret); } static int apq8016_sbc_platform_probe(struct platform_device *pdev) -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.4 034/174] ASoC: imx-sgtl5000: put of nodes if finding codec fails
by Sasha Levin 16 Jan '20

16 Jan '20

From: Stefan Agner <stefan(a)agner.ch> [ Upstream commit d9866572486802bc598a3e8576a5231378d190de ] Make sure to properly put the of node in case finding the codec fails. Fixes: 81e8e4926167 ("ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000") Signed-off-by: Stefan Agner <stefan(a)agner.ch> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 8e525f7ac08d..3d99a8579c99 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -119,7 +119,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EPROBE_DEFER; + ret = -EPROBE_DEFER; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.4 004/174] ALSA: usb-audio: update quirk for B&W PX to remove microphone
by Sasha Levin 16 Jan '20

16 Jan '20

From: Nicolas Huaman <nicolas(a)herochao.de> [ Upstream commit c369c8db15d51fa175d2ba85928f79d16af6b562 ] A quirk in snd-usb-audio was added to automate setting sample rate to 4800k and remove the previously exposed nonfunctional microphone for the Bowers & Wilkins PX: commit 240a8af929c7c57dcde28682725b29cf8474e8e5 https://lore.kernel.org/patchwork/patch/919689/ However the headphones where updated shortly after that to remove the unintentional microphone functionality. I guess because of this the headphones now crash when connecting them via USB while the quirk is active. Dmesg: snd-usb-audio: probe of 2-3:1.0 failed with error -22 usb 2-3: 2:1: cannot get min/max values for control 2 (id 2) This patch removes the microfone and allows the headphones to connect and work out of the box. It is based on the current mainline kernel and successfully applied an tested on my machine (4.18.10.arch1-1). Fixes: 240a8af929c7 ("ALSA: usb-audio: Add a quirck for B&W PX headphones") Signed-off-by: Nicolas Huaman <nicolas(a)herochao.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d32727c74a16..c892b4d1e733 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3293,19 +3293,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = 0, .type = QUIRK_AUDIO_STANDARD_MIXER, }, - /* Capture */ - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, - }, /* Playback */ { - .ifnum = 2, + .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, - .iface = 2, + .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_FILL_MAX | -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.4 002/174] ALSA: hda: fix unused variable warning
by Sasha Levin 16 Jan '20

16 Jan '20

From: Anders Roxell <anders.roxell(a)linaro.org> [ Upstream commit 5b03006d5c58ddd31caf542eef4d0269bcf265b3 ] When CONFIG_X86=n function azx_snoop doesn't use the variable chip it only returns true. sound/pci/hda/hda_intel.c: In function ‘dma_alloc_pages’: sound/pci/hda/hda_intel.c:2002:14: warning: unused variable ‘chip’ [-Wunused-variable] struct azx *chip = bus_to_azx(bus); ^~~~ Create a inline function of azx_snoop. Fixes: a41d122449be ("ALSA: hda - Embed bus into controller object") Signed-off-by: Anders Roxell <anders.roxell(a)linaro.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_controller.h | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 55ec4470f6b6..499873d29cc1 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -164,11 +164,10 @@ struct azx { #define azx_bus(chip) (&(chip)->bus.core) #define bus_to_azx(_bus) container_of(_bus, struct azx, bus.core) -#ifdef CONFIG_X86 -#define azx_snoop(chip) ((chip)->snoop) -#else -#define azx_snoop(chip) true -#endif +static inline bool azx_snoop(struct azx *chip) +{ + return !IS_ENABLED(CONFIG_X86) || chip->snoop; +} /* * macros for easy use -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.9 184/251] ASoC: sun4i-i2s: RX and TX counter registers are swapped
by Sasha Levin 16 Jan '20

16 Jan '20

From: Maxime Ripard <maxime.ripard(a)bootlin.com> [ Upstream commit cf2c0e1ce9544df42170fb921f12da82dc0cc8d6 ] The RX and TX counters registers offset have been swapped, fix that. Fixes: fa7c0d13cb26 ("ASoC: sunxi: Add Allwinner A10 Digital Audio driver") Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Link: https://lore.kernel.org/r/8b26477560ad5fd8f69e037b167c5e61de5c26a3.15662424… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sunxi/sun4i-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 15c92400cea4..02c373c65e19 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -78,8 +78,8 @@ #define SUN4I_I2S_CLK_DIV_MCLK_MASK GENMASK(3, 0) #define SUN4I_I2S_CLK_DIV_MCLK(mclk) ((mclk) << 0) -#define SUN4I_I2S_RX_CNT_REG 0x28 -#define SUN4I_I2S_TX_CNT_REG 0x2c +#define SUN4I_I2S_TX_CNT_REG 0x28 +#define SUN4I_I2S_RX_CNT_REG 0x2c #define SUN4I_I2S_TX_CHAN_SEL_REG 0x30 #define SUN4I_I2S_TX_CHAN_SEL(num_chan) (((num_chan) - 1) << 0) -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.9 182/251] ASoC: wm8737: Fix copy-paste error in wm8737_snd_controls
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit 554b75bde64bcad9662530726d1483f7ef012069 ] sound/soc/codecs/wm8737.c:112:29: warning: high_3d defined but not used [-Wunused-const-variable=] 'high_3d' should be used for 3D High Cut-off. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 2a9ae13a2641 ("ASoC: Add initial WM8737 driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20190815091920.64480-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8737.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index f0cb1c4afe3c..c5a8d758f58b 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -170,7 +170,7 @@ SOC_DOUBLE("Polarity Invert Switch", WM8737_ADC_CONTROL, 5, 6, 1, 0), SOC_SINGLE("3D Switch", WM8737_3D_ENHANCE, 0, 1, 0), SOC_SINGLE("3D Depth", WM8737_3D_ENHANCE, 1, 15, 0), SOC_ENUM("3D Low Cut-off", low_3d), -SOC_ENUM("3D High Cut-off", low_3d), +SOC_ENUM("3D High Cut-off", high_3d), SOC_SINGLE_TLV("3D ADC Volume", WM8737_3D_ENHANCE, 7, 1, 1, adc_tlv), SOC_SINGLE("Noise Gate Switch", WM8737_NOISE_GATE, 0, 1, 0), -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.9 181/251] ASoC: cs4349: Use PM ops 'cs4349_runtime_pm'
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit 9b4275c415acca6264a3d7f1182589959c93d530 ] sound/soc/codecs/cs4349.c:358:32: warning: cs4349_runtime_pm defined but not used [-Wunused-const-variable=] cs4349_runtime_pm ops already defined, it seems we should enable it. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: e40da86 ("ASoC: cs4349: Add support for Cirrus Logic CS4349") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20190815090157.70036-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs4349.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 231ca935cdf3..c232c42ccead 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -380,6 +380,7 @@ static struct i2c_driver cs4349_i2c_driver = { .driver = { .name = "cs4349", .of_match_table = cs4349_of_match, + .pm = &cs4349_runtime_pm, }, .id_table = cs4349_i2c_id, .probe = cs4349_i2c_probe, -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.9 180/251] ASoC: es8328: Fix copy-paste error in es8328_right_line_controls
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit 630742c296341a8cfe00dfd941392025ba8dd4e8 ] It seems 'es8328_rline_enum' should be used in es8328_right_line_controls Fixes: 567e4f98922c ("ASoC: add es8328 codec driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20190815092300.68712-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 37722194b107..6b22700842e2 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -234,7 +234,7 @@ static const struct soc_enum es8328_rline_enum = ARRAY_SIZE(es8328_line_texts), es8328_line_texts); static const struct snd_kcontrol_new es8328_right_line_controls = - SOC_DAPM_ENUM("Route", es8328_lline_enum); + SOC_DAPM_ENUM("Route", es8328_rline_enum); /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.9 174/251] ALSA: aoa: onyx: always initialize register read value
by Sasha Levin 16 Jan '20

16 Jan '20

From: Johannes Berg <johannes(a)sipsolutions.net> [ Upstream commit f474808acb3c4b30552d9c59b181244e0300d218 ] A lot of places in the driver use onyx_read_register() without checking the return value, and it's been working OK for ~10 years or so, so probably never fails ... Rather than trying to check the return value everywhere, which would be relatively intrusive, at least make sure we don't use an uninitialized value. Fixes: f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/aoa/codecs/onyx.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index a04edff8b729..ae50d59fb810 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -74,8 +74,10 @@ static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value) return 0; } v = i2c_smbus_read_byte_data(onyx->i2c, reg); - if (v < 0) + if (v < 0) { + *value = 0; return -1; + } *value = (u8)v; onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value; return 0; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.9 156/251] ASoC: ti: davinci-mcasp: Fix slot mask settings when using multiple AXRs
by Sasha Levin 16 Jan '20

16 Jan '20

From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> [ Upstream commit fd14f4436fd47d5418023c90e933e66d3645552e ] If multiple serializers are connected in the system and the number of channels will need to use more than one serializer the mask to enable the serializers were left to 0 if tdm_mask is provided Fixes: dd55ff8346a97 ("ASoC: davinci-mcasp: Add set_tdm_slots() support") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/davinci/davinci-mcasp.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 624c209c9498..d1935c5c3602 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -882,14 +882,13 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, active_slots = hweight32(mcasp->tdm_mask[stream]); active_serializers = (channels + active_slots - 1) / active_slots; - if (active_serializers == 1) { + if (active_serializers == 1) active_slots = channels; - for (i = 0; i < total_slots; i++) { - if ((1 << i) & mcasp->tdm_mask[stream]) { - mask |= (1 << i); - if (--active_slots <= 0) - break; - } + for (i = 0; i < total_slots; i++) { + if ((1 << i) & mcasp->tdm_mask[stream]) { + mask |= (1 << i); + if (--active_slots <= 0) + break; } } } else { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.9 124/251] ASoC: fix valid stream condition
by Sasha Levin 16 Jan '20

16 Jan '20

From: Jerome Brunet <jbrunet(a)baylibre.com> [ Upstream commit 6a7c59c6d9f3b280e81d7a04bbe4e55e90152dce ] A stream may specify a rate range using 'rate_min' and 'rate_max', so a stream may be valid and not specify any rates. However, as stream cannot be valid and not have any channel. Let's use this condition instead to determine if a stream is valid or not. Fixes: cde79035c6cf ("ASoC: Handle multiple codecs with split playback / capture") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d69559e45872..635b22fa1101 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -48,8 +48,8 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) else codec_stream = &dai->driver->capture; - /* If the codec specifies any rate at all, it supports the stream. */ - return codec_stream->rates; + /* If the codec specifies any channels at all, it supports the stream */ + return codec_stream->channels_min; } /** -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.9 122/251] ALSA: usb-audio: Handle the error from snd_usb_mixer_apply_create_quirk()
by Sasha Levin 16 Jan '20

16 Jan '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 328e9f6973be2ee67862cb17bf6c0c5c5918cd72 ] The error from snd_usb_mixer_apply_create_quirk() is ignored in the current usb-audio driver code, which will continue the probing even after the error. Let's take it more serious. Fixes: 7b1eda223deb ("ALSA: usb-mixer: factor out quirks") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 64fa1bbf0acb..54011f8543a7 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2626,7 +2626,9 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, (err = snd_usb_mixer_status_create(mixer)) < 0) goto _error; - snd_usb_mixer_apply_create_quirk(mixer); + err = snd_usb_mixer_apply_create_quirk(mixer); + if (err < 0) + goto _error; err = snd_device_new(chip->card, SNDRV_DEV_CODEC, mixer, &dev_ops); if (err < 0) -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.9 082/251] ASoC: qcom: Fix of-node refcount unbalance in apq8016_sbc_parse_of()
by Sasha Levin 16 Jan '20

16 Jan '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 8d1667200850f8753c0265fa4bd25c9a6e5f94ce ] The apq8016 driver leaves the of-node refcount at aborting from the loop of for_each_child_of_node() in the error path. Not only the iterator node of for_each_child_of_node(), the children nodes referred from it for codec and cpu have to be properly unreferenced. Fixes: bdb052e81f62 ("ASoC: qcom: add apq8016 sound card support") Cc: Patrick Lai <plai(a)codeaurora.org> Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/apq8016_sbc.c | 21 ++++++++++++++++----- 1 file changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 07f91e918b23..754742018515 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -114,13 +114,15 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) if (!cpu || !codec) { dev_err(dev, "Can't find cpu/codec DT node\n"); - return ERR_PTR(-EINVAL); + ret = -EINVAL; + goto error; } link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); if (!link->cpu_of_node) { dev_err(card->dev, "error getting cpu phandle\n"); - return ERR_PTR(-EINVAL); + ret = -EINVAL; + goto error; } link->codec_of_node = of_parse_phandle(codec, "sound-dai", 0); @@ -132,28 +134,37 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); if (ret) { dev_err(card->dev, "error getting cpu dai name\n"); - return ERR_PTR(ret); + goto error; } ret = snd_soc_of_get_dai_name(codec, &link->codec_dai_name); if (ret) { dev_err(card->dev, "error getting codec dai name\n"); - return ERR_PTR(ret); + goto error; } link->platform_of_node = link->cpu_of_node; ret = of_property_read_string(np, "link-name", &link->name); if (ret) { dev_err(card->dev, "error getting codec dai_link name\n"); - return ERR_PTR(ret); + goto error; } link->stream_name = link->name; link->init = apq8016_sbc_dai_init; link++; + + of_node_put(cpu); + of_node_put(codec); } return data; + + error: + of_node_put(np); + of_node_put(cpu); + of_node_put(codec); + return ERR_PTR(ret); } static const struct snd_soc_dapm_widget apq8016_sbc_dapm_widgets[] = { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.9 043/251] ASoC: imx-sgtl5000: put of nodes if finding codec fails
by Sasha Levin 16 Jan '20

16 Jan '20

From: Stefan Agner <stefan(a)agner.ch> [ Upstream commit d9866572486802bc598a3e8576a5231378d190de ] Make sure to properly put the of node in case finding the codec fails. Fixes: 81e8e4926167 ("ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000") Signed-off-by: Stefan Agner <stefan(a)agner.ch> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 8e525f7ac08d..3d99a8579c99 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -119,7 +119,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EPROBE_DEFER; + ret = -EPROBE_DEFER; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.9 006/251] ALSA: usb-audio: update quirk for B&W PX to remove microphone
by Sasha Levin 16 Jan '20

16 Jan '20

From: Nicolas Huaman <nicolas(a)herochao.de> [ Upstream commit c369c8db15d51fa175d2ba85928f79d16af6b562 ] A quirk in snd-usb-audio was added to automate setting sample rate to 4800k and remove the previously exposed nonfunctional microphone for the Bowers & Wilkins PX: commit 240a8af929c7c57dcde28682725b29cf8474e8e5 https://lore.kernel.org/patchwork/patch/919689/ However the headphones where updated shortly after that to remove the unintentional microphone functionality. I guess because of this the headphones now crash when connecting them via USB while the quirk is active. Dmesg: snd-usb-audio: probe of 2-3:1.0 failed with error -22 usb 2-3: 2:1: cannot get min/max values for control 2 (id 2) This patch removes the microfone and allows the headphones to connect and work out of the box. It is based on the current mainline kernel and successfully applied an tested on my machine (4.18.10.arch1-1). Fixes: 240a8af929c7 ("ALSA: usb-audio: Add a quirck for B&W PX headphones") Signed-off-by: Nicolas Huaman <nicolas(a)herochao.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d32727c74a16..c892b4d1e733 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3293,19 +3293,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = 0, .type = QUIRK_AUDIO_STANDARD_MIXER, }, - /* Capture */ - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, - }, /* Playback */ { - .ifnum = 2, + .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, - .iface = 2, + .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_FILL_MAX | -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.9 003/251] ALSA: hda: fix unused variable warning
by Sasha Levin 16 Jan '20

16 Jan '20

From: Anders Roxell <anders.roxell(a)linaro.org> [ Upstream commit 5b03006d5c58ddd31caf542eef4d0269bcf265b3 ] When CONFIG_X86=n function azx_snoop doesn't use the variable chip it only returns true. sound/pci/hda/hda_intel.c: In function ‘dma_alloc_pages’: sound/pci/hda/hda_intel.c:2002:14: warning: unused variable ‘chip’ [-Wunused-variable] struct azx *chip = bus_to_azx(bus); ^~~~ Create a inline function of azx_snoop. Fixes: a41d122449be ("ALSA: hda - Embed bus into controller object") Signed-off-by: Anders Roxell <anders.roxell(a)linaro.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_controller.h | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index b83feecf1e40..8f9386998270 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -171,11 +171,10 @@ struct azx { #define azx_bus(chip) (&(chip)->bus.core) #define bus_to_azx(_bus) container_of(_bus, struct azx, bus.core) -#ifdef CONFIG_X86 -#define azx_snoop(chip) ((chip)->snoop) -#else -#define azx_snoop(chip) true -#endif +static inline bool azx_snoop(struct azx *chip) +{ + return !IS_ENABLED(CONFIG_X86) || chip->snoop; +} /* * macros for easy use -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.14 271/371] ASoC: sun4i-i2s: RX and TX counter registers are swapped
by Sasha Levin 16 Jan '20

16 Jan '20

From: Maxime Ripard <maxime.ripard(a)bootlin.com> [ Upstream commit cf2c0e1ce9544df42170fb921f12da82dc0cc8d6 ] The RX and TX counters registers offset have been swapped, fix that. Fixes: fa7c0d13cb26 ("ASoC: sunxi: Add Allwinner A10 Digital Audio driver") Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Link: https://lore.kernel.org/r/8b26477560ad5fd8f69e037b167c5e61de5c26a3.15662424… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sunxi/sun4i-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index da0a2083e12a..d2802fd8c1dd 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -80,8 +80,8 @@ #define SUN4I_I2S_CLK_DIV_MCLK_MASK GENMASK(3, 0) #define SUN4I_I2S_CLK_DIV_MCLK(mclk) ((mclk) << 0) -#define SUN4I_I2S_RX_CNT_REG 0x28 -#define SUN4I_I2S_TX_CNT_REG 0x2c +#define SUN4I_I2S_TX_CNT_REG 0x28 +#define SUN4I_I2S_RX_CNT_REG 0x2c #define SUN4I_I2S_TX_CHAN_SEL_REG 0x30 #define SUN4I_I2S_CHAN_SEL(num_chan) (((num_chan) - 1) << 0) -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.14 267/371] ASoC: wm8737: Fix copy-paste error in wm8737_snd_controls
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit 554b75bde64bcad9662530726d1483f7ef012069 ] sound/soc/codecs/wm8737.c:112:29: warning: high_3d defined but not used [-Wunused-const-variable=] 'high_3d' should be used for 3D High Cut-off. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 2a9ae13a2641 ("ASoC: Add initial WM8737 driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20190815091920.64480-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8737.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index f0cb1c4afe3c..c5a8d758f58b 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -170,7 +170,7 @@ SOC_DOUBLE("Polarity Invert Switch", WM8737_ADC_CONTROL, 5, 6, 1, 0), SOC_SINGLE("3D Switch", WM8737_3D_ENHANCE, 0, 1, 0), SOC_SINGLE("3D Depth", WM8737_3D_ENHANCE, 1, 15, 0), SOC_ENUM("3D Low Cut-off", low_3d), -SOC_ENUM("3D High Cut-off", low_3d), +SOC_ENUM("3D High Cut-off", high_3d), SOC_SINGLE_TLV("3D ADC Volume", WM8737_3D_ENHANCE, 7, 1, 1, adc_tlv), SOC_SINGLE("Noise Gate Switch", WM8737_NOISE_GATE, 0, 1, 0), -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.14 266/371] ASoC: cs4349: Use PM ops 'cs4349_runtime_pm'
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit 9b4275c415acca6264a3d7f1182589959c93d530 ] sound/soc/codecs/cs4349.c:358:32: warning: cs4349_runtime_pm defined but not used [-Wunused-const-variable=] cs4349_runtime_pm ops already defined, it seems we should enable it. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: e40da86 ("ASoC: cs4349: Add support for Cirrus Logic CS4349") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20190815090157.70036-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs4349.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 0a749c79ef57..1d38e53dc95c 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -380,6 +380,7 @@ static struct i2c_driver cs4349_i2c_driver = { .driver = { .name = "cs4349", .of_match_table = cs4349_of_match, + .pm = &cs4349_runtime_pm, }, .id_table = cs4349_i2c_id, .probe = cs4349_i2c_probe, -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.14 265/371] ASoC: es8328: Fix copy-paste error in es8328_right_line_controls
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit 630742c296341a8cfe00dfd941392025ba8dd4e8 ] It seems 'es8328_rline_enum' should be used in es8328_right_line_controls Fixes: 567e4f98922c ("ASoC: add es8328 codec driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20190815092300.68712-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index bcdb8914ec16..e2f44fa46262 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -231,7 +231,7 @@ static const struct soc_enum es8328_rline_enum = ARRAY_SIZE(es8328_line_texts), es8328_line_texts); static const struct snd_kcontrol_new es8328_right_line_controls = - SOC_DAPM_ENUM("Route", es8328_lline_enum); + SOC_DAPM_ENUM("Route", es8328_rline_enum); /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 257/371] ALSA: aoa: onyx: always initialize register read value
by Sasha Levin 16 Jan '20

16 Jan '20

From: Johannes Berg <johannes(a)sipsolutions.net> [ Upstream commit f474808acb3c4b30552d9c59b181244e0300d218 ] A lot of places in the driver use onyx_read_register() without checking the return value, and it's been working OK for ~10 years or so, so probably never fails ... Rather than trying to check the return value everywhere, which would be relatively intrusive, at least make sure we don't use an uninitialized value. Fixes: f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/aoa/codecs/onyx.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index d2d96ca082b7..6224fd3bbf7c 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -74,8 +74,10 @@ static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value) return 0; } v = i2c_smbus_read_byte_data(onyx->i2c, reg); - if (v < 0) + if (v < 0) { + *value = 0; return -1; + } *value = (u8)v; onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value; return 0; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.14 233/371] ASoC: ti: davinci-mcasp: Fix slot mask settings when using multiple AXRs
by Sasha Levin 16 Jan '20

16 Jan '20

From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> [ Upstream commit fd14f4436fd47d5418023c90e933e66d3645552e ] If multiple serializers are connected in the system and the number of channels will need to use more than one serializer the mask to enable the serializers were left to 0 if tdm_mask is provided Fixes: dd55ff8346a97 ("ASoC: davinci-mcasp: Add set_tdm_slots() support") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/davinci/davinci-mcasp.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 07bac9ea65c4..e10e03800cce 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -882,14 +882,13 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, active_slots = hweight32(mcasp->tdm_mask[stream]); active_serializers = (channels + active_slots - 1) / active_slots; - if (active_serializers == 1) { + if (active_serializers == 1) active_slots = channels; - for (i = 0; i < total_slots; i++) { - if ((1 << i) & mcasp->tdm_mask[stream]) { - mask |= (1 << i); - if (--active_slots <= 0) - break; - } + for (i = 0; i < total_slots; i++) { + if ((1 << i) & mcasp->tdm_mask[stream]) { + mask |= (1 << i); + if (--active_slots <= 0) + break; } } } else { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 500/671] ASoC: sun4i-i2s: RX and TX counter registers are swapped
by Sasha Levin 16 Jan '20

16 Jan '20

From: Maxime Ripard <maxime.ripard(a)bootlin.com> [ Upstream commit cf2c0e1ce9544df42170fb921f12da82dc0cc8d6 ] The RX and TX counters registers offset have been swapped, fix that. Fixes: fa7c0d13cb26 ("ASoC: sunxi: Add Allwinner A10 Digital Audio driver") Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Link: https://lore.kernel.org/r/8b26477560ad5fd8f69e037b167c5e61de5c26a3.15662424… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sunxi/sun4i-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 18cf8404d27c..f248e563986c 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -80,8 +80,8 @@ #define SUN4I_I2S_CLK_DIV_MCLK_MASK GENMASK(3, 0) #define SUN4I_I2S_CLK_DIV_MCLK(mclk) ((mclk) << 0) -#define SUN4I_I2S_RX_CNT_REG 0x28 -#define SUN4I_I2S_TX_CNT_REG 0x2c +#define SUN4I_I2S_TX_CNT_REG 0x28 +#define SUN4I_I2S_RX_CNT_REG 0x2c #define SUN4I_I2S_TX_CHAN_SEL_REG 0x30 #define SUN4I_I2S_CHAN_SEL(num_chan) (((num_chan) - 1) << 0) -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 492/671] ASoC: wm8737: Fix copy-paste error in wm8737_snd_controls
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit 554b75bde64bcad9662530726d1483f7ef012069 ] sound/soc/codecs/wm8737.c:112:29: warning: high_3d defined but not used [-Wunused-const-variable=] 'high_3d' should be used for 3D High Cut-off. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 2a9ae13a2641 ("ASoC: Add initial WM8737 driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20190815091920.64480-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8737.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index e9ae821e7609..568b77692f5f 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -170,7 +170,7 @@ SOC_DOUBLE("Polarity Invert Switch", WM8737_ADC_CONTROL, 5, 6, 1, 0), SOC_SINGLE("3D Switch", WM8737_3D_ENHANCE, 0, 1, 0), SOC_SINGLE("3D Depth", WM8737_3D_ENHANCE, 1, 15, 0), SOC_ENUM("3D Low Cut-off", low_3d), -SOC_ENUM("3D High Cut-off", low_3d), +SOC_ENUM("3D High Cut-off", high_3d), SOC_SINGLE_TLV("3D ADC Volume", WM8737_3D_ENHANCE, 7, 1, 1, adc_tlv), SOC_SINGLE("Noise Gate Switch", WM8737_NOISE_GATE, 0, 1, 0), -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 491/671] ASoC: cs4349: Use PM ops 'cs4349_runtime_pm'
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit 9b4275c415acca6264a3d7f1182589959c93d530 ] sound/soc/codecs/cs4349.c:358:32: warning: cs4349_runtime_pm defined but not used [-Wunused-const-variable=] cs4349_runtime_pm ops already defined, it seems we should enable it. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: e40da86 ("ASoC: cs4349: Add support for Cirrus Logic CS4349") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20190815090157.70036-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs4349.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index bee0e343723f..f08d7a296c0c 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -381,6 +381,7 @@ static struct i2c_driver cs4349_i2c_driver = { .driver = { .name = "cs4349", .of_match_table = cs4349_of_match, + .pm = &cs4349_runtime_pm, }, .id_table = cs4349_i2c_id, .probe = cs4349_i2c_probe, -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 490/671] ASoC: es8328: Fix copy-paste error in es8328_right_line_controls
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit 630742c296341a8cfe00dfd941392025ba8dd4e8 ] It seems 'es8328_rline_enum' should be used in es8328_right_line_controls Fixes: 567e4f98922c ("ASoC: add es8328 codec driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20190815092300.68712-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index e9fc2fd97d2f..3afa163f7652 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -231,7 +231,7 @@ static const struct soc_enum es8328_rline_enum = ARRAY_SIZE(es8328_line_texts), es8328_line_texts); static const struct snd_kcontrol_new es8328_right_line_controls = - SOC_DAPM_ENUM("Route", es8328_lline_enum); + SOC_DAPM_ENUM("Route", es8328_rline_enum); /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 475/671] ALSA: aoa: onyx: always initialize register read value
by Sasha Levin 16 Jan '20

16 Jan '20

From: Johannes Berg <johannes(a)sipsolutions.net> [ Upstream commit f474808acb3c4b30552d9c59b181244e0300d218 ] A lot of places in the driver use onyx_read_register() without checking the return value, and it's been working OK for ~10 years or so, so probably never fails ... Rather than trying to check the return value everywhere, which would be relatively intrusive, at least make sure we don't use an uninitialized value. Fixes: f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/aoa/codecs/onyx.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index d2d96ca082b7..6224fd3bbf7c 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -74,8 +74,10 @@ static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value) return 0; } v = i2c_smbus_read_byte_data(onyx->i2c, reg); - if (v < 0) + if (v < 0) { + *value = 0; return -1; + } *value = (u8)v; onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value; return 0; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 424/671] ASoC: ti: davinci-mcasp: Fix slot mask settings when using multiple AXRs
by Sasha Levin 16 Jan '20

16 Jan '20

From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> [ Upstream commit fd14f4436fd47d5418023c90e933e66d3645552e ] If multiple serializers are connected in the system and the number of channels will need to use more than one serializer the mask to enable the serializers were left to 0 if tdm_mask is provided Fixes: dd55ff8346a97 ("ASoC: davinci-mcasp: Add set_tdm_slots() support") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/davinci/davinci-mcasp.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6a8c279a4b20..14ab16e1369f 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -874,14 +874,13 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, active_slots = hweight32(mcasp->tdm_mask[stream]); active_serializers = (channels + active_slots - 1) / active_slots; - if (active_serializers == 1) { + if (active_serializers == 1) active_slots = channels; - for (i = 0; i < total_slots; i++) { - if ((1 << i) & mcasp->tdm_mask[stream]) { - mask |= (1 << i); - if (--active_slots <= 0) - break; - } + for (i = 0; i < total_slots; i++) { + if ((1 << i) & mcasp->tdm_mask[stream]) { + mask |= (1 << i); + if (--active_slots <= 0) + break; } } } else { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 411/671] ASoC: meson: axg-tdmout: right_j is not supported
by Sasha Levin 16 Jan '20

16 Jan '20

From: Jerome Brunet <jbrunet(a)baylibre.com> [ Upstream commit 7e0d7d0fbd06af0507611f85dba8daf24832abd9 ] Right justified format is actually not supported by the amlogic tdm output encoder. Fixes: c41c2a355b86 ("ASoC: meson: add tdm output driver") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/meson/axg-tdmout.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c index f73368ee1088..d11acb3cc696 100644 --- a/sound/soc/meson/axg-tdmout.c +++ b/sound/soc/meson/axg-tdmout.c @@ -136,7 +136,6 @@ static int axg_tdmout_prepare(struct regmap *map, struct axg_tdm_stream *ts) break; case SND_SOC_DAIFMT_LEFT_J: - case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_DSP_B: val |= TDMOUT_CTRL0_INIT_BITNUM(2); break; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 410/671] ASoC: meson: axg-tdmin: right_j is not supported
by Sasha Levin 16 Jan '20

16 Jan '20

From: Jerome Brunet <jbrunet(a)baylibre.com> [ Upstream commit 47c317b786b6c1efc2cb3cdb894fd323422fe5ea ] Right justified format is actually not supported by the amlogic tdm input decoder. Fixes: 13a22e6a98f8 ("ASoC: meson: add tdm input driver") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/meson/axg-tdmin.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c index bbac44c81688..37207bbebb2a 100644 --- a/sound/soc/meson/axg-tdmin.c +++ b/sound/soc/meson/axg-tdmin.c @@ -119,7 +119,6 @@ static int axg_tdmin_prepare(struct regmap *map, struct axg_tdm_stream *ts) break; case SND_SOC_DAIFMT_LEFT_J: - case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_DSP_B: val = TDMIN_CTRL_IN_BIT_SKEW(2); break; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 344/671] ALSA: aica: Fix a long-time build breakage
by Sasha Levin 16 Jan '20

16 Jan '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 534420c6ff87d3052540f1fd346e0adcff440819 ] The build of aica sound driver has been broken since the timer API conversion and some code rewrite. This patch fixes the breakage by using the common substream field, as well as a bit cleaning up wrt the timer handling in the code. Fixes: d522bb6a105f ("ALSA: sh: aica: Convert timers to use timer_setup()") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/aica.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 2b26311405a4..ad3f71358486 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -303,7 +303,7 @@ static void aica_period_elapsed(struct timer_list *t) { struct snd_card_aica *dreamcastcard = from_timer(dreamcastcard, t, timer); - struct snd_pcm_substream *substream = dreamcastcard->timer_substream; + struct snd_pcm_substream *substream = dreamcastcard->substream; /*timer function - so cannot sleep */ int play_period; struct snd_pcm_runtime *runtime; @@ -335,13 +335,6 @@ static void spu_begin_dma(struct snd_pcm_substream *substream) dreamcastcard = substream->pcm->private_data; /*get the queue to do the work */ schedule_work(&(dreamcastcard->spu_dma_work)); - /* Timer may already be running */ - if (unlikely(dreamcastcard->timer_substream)) { - mod_timer(&dreamcastcard->timer, jiffies + 4); - return; - } - timer_setup(&dreamcastcard->timer, aica_period_elapsed, 0); - dreamcastcard->timer_substream = substream; mod_timer(&dreamcastcard->timer, jiffies + 4); } @@ -379,8 +372,8 @@ static int snd_aicapcm_pcm_close(struct snd_pcm_substream { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; flush_work(&(dreamcastcard->spu_dma_work)); - if (dreamcastcard->timer_substream) - del_timer(&dreamcastcard->timer); + del_timer(&dreamcastcard->timer); + dreamcastcard->substream = NULL; kfree(dreamcastcard->channel); spu_disable(); return 0; @@ -615,6 +608,7 @@ static int snd_aica_probe(struct platform_device *devptr) "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); /* Prepare to use the queue */ INIT_WORK(&(dreamcastcard->spu_dma_work), run_spu_dma); + timer_setup(&dreamcastcard->timer, aica_period_elapsed, 0); /* Load the PCM 'chip' */ err = snd_aicapcmchip(dreamcastcard, 0); if (unlikely(err < 0)) -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 324/671] ASoC: fix valid stream condition
by Sasha Levin 16 Jan '20

16 Jan '20

From: Jerome Brunet <jbrunet(a)baylibre.com> [ Upstream commit 6a7c59c6d9f3b280e81d7a04bbe4e55e90152dce ] A stream may specify a rate range using 'rate_min' and 'rate_max', so a stream may be valid and not specify any rates. However, as stream cannot be valid and not have any channel. Let's use this condition instead to determine if a stream is valid or not. Fixes: cde79035c6cf ("ASoC: Handle multiple codecs with split playback / capture") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 551bfc581fc1..53fefa7c982f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -42,8 +42,8 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) else codec_stream = &dai->driver->capture; - /* If the codec specifies any rate at all, it supports the stream. */ - return codec_stream->rates; + /* If the codec specifies any channels at all, it supports the stream */ + return codec_stream->channels_min; } /** -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 309/671] ALSA: usb-audio: Handle the error from snd_usb_mixer_apply_create_quirk()
by Sasha Levin 16 Jan '20

16 Jan '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 328e9f6973be2ee67862cb17bf6c0c5c5918cd72 ] The error from snd_usb_mixer_apply_create_quirk() is ignored in the current usb-audio driver code, which will continue the probing even after the error. Let's take it more serious. Fixes: 7b1eda223deb ("ALSA: usb-mixer: factor out quirks") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index d7778f2bcbf8..6ac6a0980124 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3480,7 +3480,9 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, if (err < 0) goto _error; - snd_usb_mixer_apply_create_quirk(mixer); + err = snd_usb_mixer_apply_create_quirk(mixer); + if (err < 0) + goto _error; err = snd_device_new(chip->card, SNDRV_DEV_CODEC, mixer, &dev_ops); if (err < 0) -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 206/671] ASoC: qcom: Fix of-node refcount unbalance in apq8016_sbc_parse_of()
by Sasha Levin 16 Jan '20

16 Jan '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 8d1667200850f8753c0265fa4bd25c9a6e5f94ce ] The apq8016 driver leaves the of-node refcount at aborting from the loop of for_each_child_of_node() in the error path. Not only the iterator node of for_each_child_of_node(), the children nodes referred from it for codec and cpu have to be properly unreferenced. Fixes: bdb052e81f62 ("ASoC: qcom: add apq8016 sound card support") Cc: Patrick Lai <plai(a)codeaurora.org> Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/apq8016_sbc.c | 21 ++++++++++++++++----- 1 file changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 1dd23bba1bed..4b559932adc3 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -164,41 +164,52 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) if (!cpu || !codec) { dev_err(dev, "Can't find cpu/codec DT node\n"); - return ERR_PTR(-EINVAL); + ret = -EINVAL; + goto error; } link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); if (!link->cpu_of_node) { dev_err(card->dev, "error getting cpu phandle\n"); - return ERR_PTR(-EINVAL); + ret = -EINVAL; + goto error; } ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); if (ret) { dev_err(card->dev, "error getting cpu dai name\n"); - return ERR_PTR(ret); + goto error; } ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); if (ret < 0) { dev_err(card->dev, "error getting codec dai name\n"); - return ERR_PTR(ret); + goto error; } link->platform_of_node = link->cpu_of_node; ret = of_property_read_string(np, "link-name", &link->name); if (ret) { dev_err(card->dev, "error getting codec dai_link name\n"); - return ERR_PTR(ret); + goto error; } link->stream_name = link->name; link->init = apq8016_sbc_dai_init; link++; + + of_node_put(cpu); + of_node_put(codec); } return data; + + error: + of_node_put(np); + of_node_put(cpu); + of_node_put(codec); + return ERR_PTR(ret); } static const struct snd_soc_dapm_widget apq8016_sbc_dapm_widgets[] = { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 122/671] ASoC: imx-sgtl5000: put of nodes if finding codec fails
by Sasha Levin 16 Jan '20

16 Jan '20

From: Stefan Agner <stefan(a)agner.ch> [ Upstream commit d9866572486802bc598a3e8576a5231378d190de ] Make sure to properly put the of node in case finding the codec fails. Fixes: 81e8e4926167 ("ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000") Signed-off-by: Stefan Agner <stefan(a)agner.ch> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 9b9a7ec52905..4bd8da3a5f5b 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -112,7 +112,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EPROBE_DEFER; + ret = -EPROBE_DEFER; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 049/671] ASoC: sun8i-codec: add missing route for ADC
by Sasha Levin 16 Jan '20

16 Jan '20

From: Vasily Khoruzhick <anarsoul(a)gmail.com> [ Upstream commit 9ee325d029c4abb75716851ce38863845911d605 ] sun8i-codec misses a route from ADC to AIF1 Slot 0 ADC. Add it to the driver to avoid adding it to every dts. Fixes: eda85d1fee05d ("ASoC: sun8i-codec: Add ADC support for a33") Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sunxi/sun8i-codec.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index bf615fa16dc8..a3db6a68dfe6 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -465,7 +465,11 @@ static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = { { "Right Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch", "AIF1 Slot 0 Right"}, - /* ADC routes */ + /* ADC Routes */ + { "AIF1 Slot 0 Right ADC", NULL, "ADC" }, + { "AIF1 Slot 0 Left ADC", NULL, "ADC" }, + + /* ADC Mixer Routes */ { "Left Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch", "AIF1 Slot 0 Left ADC" }, { "Right Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch", -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 046/671] ASoC: wm97xx: fix uninitialized regmap pointer problem
by Sasha Levin 16 Jan '20

16 Jan '20

From: Arnd Bergmann <arnd(a)arndb.de> [ Upstream commit 576ce4075bfa0f03e0e91a89eecc539b3b828b08 ] gcc notices that without either the ac97 bus or the pdata, we never initialize the regmap pointer, which leads to an uninitialized variable access: sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc/codecs/wm9712.c:666:2: error: 'regmap' may be used uninitialized in this function [-Werror=maybe-uninitialized] Since that configuration is invalid, it's better to return an error here. I tried to avoid adding complexity to the conditions, and turned the #ifdef into a regular if(IS_ENABLED()) check for readability. This in turn requires moving some header file declarations out of an #ifdef. The same code is used in three drivers, all of which I'm changing the same way. Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc.h | 2 +- sound/soc/codecs/wm9705.c | 10 ++++------ sound/soc/codecs/wm9712.c | 10 ++++------ sound/soc/codecs/wm9713.c | 10 ++++------ 4 files changed, 13 insertions(+), 19 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 41cec42fb456..88aa48e5485f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -548,12 +548,12 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, } #endif -#ifdef CONFIG_SND_SOC_AC97_BUS struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component); struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component, unsigned int id, unsigned int id_mask); void snd_soc_free_ac97_component(struct snd_ac97 *ac97); +#ifdef CONFIG_SND_SOC_AC97_BUS int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, struct platform_device *pdev); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index ccdf088461b7..54c306707c02 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -325,8 +325,7 @@ static int wm9705_soc_probe(struct snd_soc_component *component) if (wm9705->mfd_pdata) { wm9705->ac97 = wm9705->mfd_pdata->ac97; regmap = wm9705->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9705->ac97 = snd_soc_new_ac97_component(component, WM9705_VENDOR_ID, WM9705_VENDOR_ID_MASK); if (IS_ERR(wm9705->ac97)) { @@ -339,7 +338,8 @@ static int wm9705_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9705->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_set_drvdata(component, wm9705->ac97); @@ -350,14 +350,12 @@ static int wm9705_soc_probe(struct snd_soc_component *component) static void wm9705_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9705_priv *wm9705 = snd_soc_component_get_drvdata(component); - if (!wm9705->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9705->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9705->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9705 = { diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e873baa9e778..01949eaba4fd 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -642,8 +642,7 @@ static int wm9712_soc_probe(struct snd_soc_component *component) if (wm9712->mfd_pdata) { wm9712->ac97 = wm9712->mfd_pdata->ac97; regmap = wm9712->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { int ret; wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, @@ -660,7 +659,8 @@ static int wm9712_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9712->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -673,14 +673,12 @@ static int wm9712_soc_probe(struct snd_soc_component *component) static void wm9712_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); - if (!wm9712->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9712->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9712->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9712 = { diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 643863bb32e0..5a2fdf4f69bf 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1214,8 +1214,7 @@ static int wm9713_soc_probe(struct snd_soc_component *component) if (wm9713->mfd_pdata) { wm9713->ac97 = wm9713->mfd_pdata->ac97; regmap = wm9713->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID, WM9713_VENDOR_ID_MASK); if (IS_ERR(wm9713->ac97)) @@ -1225,7 +1224,8 @@ static int wm9713_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9713->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -1238,14 +1238,12 @@ static int wm9713_soc_probe(struct snd_soc_component *component) static void wm9713_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component); - if (!wm9713->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9713->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9713->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9713 = { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 029/671] ALSA: usb-audio: update quirk for B&W PX to remove microphone
by Sasha Levin 16 Jan '20

16 Jan '20

From: Nicolas Huaman <nicolas(a)herochao.de> [ Upstream commit c369c8db15d51fa175d2ba85928f79d16af6b562 ] A quirk in snd-usb-audio was added to automate setting sample rate to 4800k and remove the previously exposed nonfunctional microphone for the Bowers & Wilkins PX: commit 240a8af929c7c57dcde28682725b29cf8474e8e5 https://lore.kernel.org/patchwork/patch/919689/ However the headphones where updated shortly after that to remove the unintentional microphone functionality. I guess because of this the headphones now crash when connecting them via USB while the quirk is active. Dmesg: snd-usb-audio: probe of 2-3:1.0 failed with error -22 usb 2-3: 2:1: cannot get min/max values for control 2 (id 2) This patch removes the microfone and allows the headphones to connect and work out of the box. It is based on the current mainline kernel and successfully applied an tested on my machine (4.18.10.arch1-1). Fixes: 240a8af929c7 ("ALSA: usb-audio: Add a quirck for B&W PX headphones") Signed-off-by: Nicolas Huaman <nicolas(a)herochao.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 65f9c4ba62ee..90d4f61cc230 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3349,19 +3349,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = 0, .type = QUIRK_AUDIO_STANDARD_MIXER, }, - /* Capture */ - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, - }, /* Playback */ { - .ifnum = 2, + .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, - .iface = 2, + .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_FILL_MAX | -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 017/671] ASoC: wm9712: fix unused variable warning
by Sasha Levin 16 Jan '20

16 Jan '20

From: Arnd Bergmann <arnd(a)arndb.de> [ Upstream commit 18380dcc52cc8965e5144ce33fdfad7e168679a5 ] The 'ret' variable is now only used in an #ifdef, and causes a warning if it is declared outside of that block: sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc/codecs/wm9712.c:641:6: error: unused variable 'ret' [-Werror=unused-variable] Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm9712.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index ade34c26ad2f..e873baa9e778 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -638,13 +638,14 @@ static int wm9712_soc_probe(struct snd_soc_component *component) { struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); struct regmap *regmap; - int ret; if (wm9712->mfd_pdata) { wm9712->ac97 = wm9712->mfd_pdata->ac97; regmap = wm9712->mfd_pdata->regmap; } else { #ifdef CONFIG_SND_SOC_AC97_BUS + int ret; + wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, WM9712_VENDOR_ID_MASK); if (IS_ERR(wm9712->ac97)) { -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 008/671] ALSA: hda: fix unused variable warning
by Sasha Levin 16 Jan '20

16 Jan '20

From: Anders Roxell <anders.roxell(a)linaro.org> [ Upstream commit 5b03006d5c58ddd31caf542eef4d0269bcf265b3 ] When CONFIG_X86=n function azx_snoop doesn't use the variable chip it only returns true. sound/pci/hda/hda_intel.c: In function ‘dma_alloc_pages’: sound/pci/hda/hda_intel.c:2002:14: warning: unused variable ‘chip’ [-Wunused-variable] struct azx *chip = bus_to_azx(bus); ^~~~ Create a inline function of azx_snoop. Fixes: a41d122449be ("ALSA: hda - Embed bus into controller object") Signed-off-by: Anders Roxell <anders.roxell(a)linaro.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_controller.h | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 8a9dd4767b1e..63cc10604afc 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -176,11 +176,10 @@ struct azx { #define azx_bus(chip) (&(chip)->bus.core) #define bus_to_azx(_bus) container_of(_bus, struct azx, bus.core) -#ifdef CONFIG_X86 -#define azx_snoop(chip) ((chip)->snoop) -#else -#define azx_snoop(chip) true -#endif +static inline bool azx_snoop(struct azx *chip) +{ + return !IS_ENABLED(CONFIG_X86) || chip->snoop; +} /* * macros for easy use -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 170/205] ASoC: rsnd: fix DALIGN register for SSIU
by Sasha Levin 16 Jan '20

16 Jan '20

From: Nilkanth Ahirrao <anilkanth(a)jp.adit-jv.com> [ Upstream commit ef8e14794308a428b194f8b06ad9ae06b43466e4 ] The current driver only sets 0x76543210 and 0x67452301 for DALIGN. This doesn’t work well for TDM split and ex-split mode for all SSIU. This patch programs the DALIGN registers based on the SSIU number. Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Jiada Wang <jiada_wang(a)mentor.com> Cc: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> Fixes: a914e44693d41b ("ASoC: rsnd: more clear rsnd_get_dalign() for DALIGN") Signed-off-by: Nilkanth Ahirrao <anilkanth(a)jp.adit-jv.com> Signed-off-by: Eugeniu Rosca <erosca(a)de.adit-jv.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/20191121111023.10976-1-erosca@de.adit-jv.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/core.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e9596c2096cd..a6c1cf987e6e 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -376,6 +376,17 @@ u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io) */ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { + static const u32 dalign_values[8][2] = { + {0x76543210, 0x67452301}, + {0x00000032, 0x00000023}, + {0x00007654, 0x00006745}, + {0x00000076, 0x00000067}, + {0xfedcba98, 0xefcdab89}, + {0x000000ba, 0x000000ab}, + {0x0000fedc, 0x0000efcd}, + {0x000000fe, 0x000000ef}, + }; + int id = 0, inv; struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); struct rsnd_mod *target; struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); @@ -411,13 +422,18 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) target = cmd ? cmd : ssiu; } + if (mod == ssiu) + id = rsnd_mod_id_sub(mod); + /* Non target mod or non 16bit needs normal DALIGN */ if ((snd_pcm_format_width(runtime->format) != 16) || (mod != target)) - return 0x76543210; + inv = 0; /* Target mod needs inverted DALIGN when 16bit */ else - return 0x67452301; + inv = 1; + + return dalign_values[id][inv]; } u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod) -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 117/205] ASoC: core: Fix compile warning with CONFIG_DEBUG_FS=n
by Sasha Levin 16 Jan '20

16 Jan '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit bd0b609e0c3362cb167c51d4bd4330d79fc00987 ] Paper over a compile warning: sound/soc/soc-pcm.c:1185:8: warning: unused variable ‘name’ Fixes: 0632fa042541 ("ASoC: core: Fix pcm code debugfs error") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20191107134833.1502-1-tiwai@suse.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a6e96cf1d8ff..d07026a846b9 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1148,7 +1148,9 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; unsigned long flags; +#ifdef CONFIG_DEBUG_FS char *name; +#endif /* only add new dpcms */ for_each_dpcm_be(fe, stream, dpcm) { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 072/205] ASoC: fsl_esai: Add spin lock to protect reset, stop and start
by Sasha Levin 16 Jan '20

16 Jan '20

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 35dac627471938eda89fa39ee4ead1f7667e0f57 ] xrun may happen at the end of stream, the trigger->fsl_esai_trigger_stop maybe called in the middle of fsl_esai_hw_reset, this may cause esai in wrong state after stop, and there may be endless xrun interrupt. This issue may also happen with trigger->fsl_esai_trigger_start. So Add spin lock to lock those functions. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Link: https://lore.kernel.org/r/52e92c4221a83e39a84a6cd92fc3d5479b44894c.15722523… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index a78e4ab478df..c7a49d03463a 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -33,6 +33,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @spbaclk: SPBA clock (optional, depending on SoC design) * @task: tasklet to handle the reset operation + * @lock: spin lock between hw_reset() and trigger() * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot * @slots: number of slots @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; struct clk *spbaclk; struct tasklet_struct task; + spinlock_t lock; /* Protect hw_reset and trigger */ u32 fifo_depth; u32 slot_width; u32 slots; @@ -676,8 +678,10 @@ static void fsl_esai_hw_reset(unsigned long arg) { struct fsl_esai *esai_priv = (struct fsl_esai *)arg; bool tx = true, rx = false, enabled[2]; + unsigned long lock_flags; u32 tfcr, rfcr; + spin_lock_irqsave(&esai_priv->lock, lock_flags); /* Save the registers */ regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr); regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr); @@ -715,6 +719,8 @@ static void fsl_esai_hw_reset(unsigned long arg) fsl_esai_trigger_start(esai_priv, tx); if (enabled[rx]) fsl_esai_trigger_start(esai_priv, rx); + + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); } static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -722,6 +728,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned long lock_flags; esai_priv->channels[tx] = substream->runtime->channels; @@ -729,12 +736,16 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_start(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_stop(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; default: return -EINVAL; @@ -1002,6 +1013,7 @@ static int fsl_esai_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, esai_priv); + spin_lock_init(&esai_priv->lock); ret = fsl_esai_hw_init(esai_priv); if (ret) return ret; -- 2.20.1

1 0

[alsa-devel] [PATCH] ALSA: pcm: Make snd_pcm_hw_constraints_init() and _complete() static
by Takashi Iwai 16 Jan '20

16 Jan '20

Both snd_pcm_hw_constraints_init() and _complete() functions are called only from pcm_native.c, hence they can be static for further optimization. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_local.h | 3 --- sound/core/pcm_native.c | 4 ++-- 2 files changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index 384efd002984..17a1a5d87098 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -20,9 +20,6 @@ void snd_interval_muldivk(const struct snd_interval *a, void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, const struct snd_interval *b, struct snd_interval *c); -int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream); -int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream); - int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int32_t mask); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c375c41496f8..bb23f5066654 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2308,7 +2308,7 @@ static int snd_pcm_hw_rule_buffer_bytes_max(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) +static int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; @@ -2432,7 +2432,7 @@ int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) return 0; } -int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) +static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hardware *hw = &runtime->hw; -- 2.16.4

1 0

[alsa-devel] [PATCH 0/3] SOF: Fix HDMI probe errors on GLK devices
by Kai Vehmanen 16 Jan '20

16 Jan '20

Hi all, here's a patch series to address an issue that popped up after SOF changed from hdac-hdmi to snd-hda-codec-hdmi: "failed to get afg sub nodes on ChromeOS devices" https://github.com/thesofproject/linux/issues/1642 This is fairly hard to hit and only occurs on some devices (most current reports are Gemini Lake based Chromebooks), but when it does, it is rather severe as the whole SOF probe fails because of this, and user is left without sound. First fix is to optimize out one power down/up cycle from the probe process. On platforms such as GLK, each time audio driver does acomp's get_power(), graphics typically needs to do a modeset due to clocking requirements for the HDA bus. This can cause display to flash when audio is probed, plus in the reported cases on GLK, can lead to probe failures. The above change doesn't cover all reported cases, so additionally retry logic is added to snd_hda_get_sub_nodes() calls on Intel platforms. Multiple other approaches were investigated, but a simple retry -- although less than ideal -- in the end proved to be most reliable solution. HDA dump on one affected Acer Chromebook looks like this: # codec initialization goes as normal with multiple successful cmds udevd-9486 [001] .... 3910.087791: hda_send_cmd: [0000:00:0e.0:2] val=0x207f1c00 udevd-9486 [001] .... 3910.087857: hda_get_response: [0000:00:0e.0:2] val=0x18560010 udevd-9486 [001] .... 3910.087858: hda_send_cmd: [0000:00:0e.0:2] val=0x207f0700 udevd-9486 [001] .... 3910.087931: hda_get_response: [0000:00:0e.0:2] val=0x00000000 udevd-9486 [001] .... 3910.087932: hda_send_cmd: [0000:00:0e.0:2] val=0x20bf8100 udevd-9486 [001] .... 3910.088040: hda_get_response: [0000:00:0e.0:2] val=0x00000001 udevd-9486 [001] .... 3910.088044: hda_send_cmd: [0000:00:0e.0:2] val=0x20b78103 # here get params for AC_PAR_NODE_COUNT fails, 0xffffffff is returned udevd-9486 [001] .... 3910.088048: hda_send_cmd: [0000:00:0e.0:2] val=0x201f0004 udevd-9486 [001] .... 3911.090131: hda_get_response: [0000:00:0e.0:2] val=0xffffffff # retry is successful udevd-9486 [001] .... 3911.090152: hda_send_cmd: [0000:00:0e.0:2] val=0x201f0004 udevd-9486 [001] .... 3911.090288: hda_get_response: [0000:00:0e.0:2] val=0x00020006 Kai Vehmanen (3): ASoC: SOF: Intel: refactor i915_get/put functions ASoC: SOF: Intel: do not disable i915 power during probe ALSA: hda/hdmi - add retry logic to parse_intel_hdmi() sound/pci/hda/patch_hdmi.c | 7 +++++-- sound/soc/sof/intel/hda-codec.c | 21 ++++++--------------- sound/soc/sof/intel/hda.c | 3 ++- sound/soc/sof/intel/hda.h | 7 +++---- 4 files changed, 16 insertions(+), 22 deletions(-) -- 2.17.1

2 4

[alsa-devel] [PATCH] soundwire: cadence: fix kernel-doc parameter descriptions
by Pierre-Louis Bossart 16 Jan '20

16 Jan '20

Fix previous update, bad git merge likely. oops. Fixes: 39737a313085fa ("soundwire: cadence: update kernel-doc parameter descriptions") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/cadence_master.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 12d3ac56e01b..9bec270d0fa4 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -819,7 +819,6 @@ EXPORT_SYMBOL(sdw_cdns_thread); /** * sdw_cdns_exit_reset() - Program reset parameters and start bus operations * @cdns: Cadence instance - * @state: True if we are trying to enable interrupt. */ int sdw_cdns_exit_reset(struct sdw_cdns *cdns) { @@ -846,6 +845,7 @@ EXPORT_SYMBOL(sdw_cdns_exit_reset); /** * sdw_cdns_enable_interrupt() - Enable SDW interrupts * @cdns: Cadence instance + * @state: True if we are trying to enable interrupt. */ int sdw_cdns_enable_interrupt(struct sdw_cdns *cdns, bool state) { -- 2.20.1

2 3

[alsa-devel] [GIT PULL] ASoC fixes for v5.5
by Mark Brown 16 Jan '20

16 Jan '20

The following changes since commit ff3b57417012fcc963ec281f5705bed837e4b1ac: ASoC: Intel: boards: Fix compile-testing RT1011/RT5682 (2020-01-03 00:56:47 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.5-rc6 for you to fetch changes up to 85578bbd642f65065039b1765ebe1a867d5435b0: ASoC: msm8916-wcd-digital: Reset RX interpolation path after use (2020-01-13 15:12:28 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.5 This is mostly driver specific fixes, plus an error handling fix in the core. There is a rather large diffstat for the stm32 SAI driver, this is a very large but mostly mechanical update which wraps every register access in the driver to allow a fix to the locking which avoids circular locks, the active change is much smaller and more reasonably sized. ---------------------------------------------------------------- Arnaud Pouliquen (1): ASoC: sti: fix possible sleep-in-atomic Bard liao (1): ASoC: SOF: Intel: lower print level to dbg if we will reinit DSP Dmitry Osipenko (1): ASoC: rt5640: Fix NULL dereference on module unload Hans de Goede (1): ASoC: Intel: bytcht_es8316: Fix Irbis NB41 netbook quirk Kai Vehmanen (2): ASoC: SOF: Intel: fix HDA codec driver probe with multiple controllers ASoC: hdac_hda: Fix error in driver removal after failed probe Olivier Moysan (2): ASoC: stm32: sai: fix possible circular locking ASoC: stm32: dfsdm: fix 16 bits record Stephan Gerhold (3): ASoC: msm8916-wcd-analog: Fix selected events for MIC BIAS External1 ASoC: msm8916-wcd-analog: Fix MIC BIAS Internal1 ASoC: msm8916-wcd-digital: Reset RX interpolation path after use Takashi Iwai (1): ASoC: Fix NULL dereference at freeing Yu-Hsuan Hsu (1): ASoC: cros_ec_codec: Make the device acpi compatible sound/soc/codecs/cros_ec_codec.c | 8 ++ sound/soc/codecs/hdac_hda.c | 4 +- sound/soc/codecs/msm8916-wcd-analog.c | 20 +++- sound/soc/codecs/msm8916-wcd-digital.c | 6 + sound/soc/codecs/rt5640.c | 7 ++ sound/soc/intel/boards/bytcht_es8316.c | 3 +- sound/soc/soc-component.c | 3 + sound/soc/sof/intel/hda-codec.c | 19 ++-- sound/soc/sof/intel/hda-loader.c | 6 +- sound/soc/sti/uniperif_player.c | 7 +- sound/soc/stm/stm32_adfsdm.c | 12 +- sound/soc/stm/stm32_sai_sub.c | 194 ++++++++++++++++++++++++--------- 12 files changed, 209 insertions(+), 80 deletions(-)

2 1

[alsa-devel] [PATCH] dt-bindings: soundwire: fix example
by Srinivas Kandagatla 16 Jan '20

16 Jan '20

As wsa881x schema mentions #sound-dai-cells as required property, so update soundwire-controller.yaml example so that dt_bindings_check does not fail as below: Documentation/devicetree/bindings/soundwire/soundwire-controller.example.dt.yaml: speaker@0,1: '#sound-dai-cells' is a required property Documentation/devicetree/bindings/soundwire/soundwire-controller.example.dt.yaml: speaker@0,2: '#sound-dai-cells' is a required property Reported-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../devicetree/bindings/soundwire/soundwire-controller.yaml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml b/Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml index 1b43993bccdb..330924b8618e 100644 --- a/Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml +++ b/Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml @@ -69,6 +69,7 @@ examples: reg = <0 1>; powerdown-gpios = <&wcdpinctrl 2 0>; #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; }; speaker@0,2 { @@ -76,6 +77,7 @@ examples: reg = <0 2>; powerdown-gpios = <&wcdpinctrl 2 0>; #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; }; }; -- 2.21.0

2 1

[alsa-devel] [PATCH] soundwire: intel: report slave_ids for each link to SOF driver
by Pierre-Louis Bossart 16 Jan '20

16 Jan '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> The existing link_mask flag is no longer sufficient to detect the hardware and identify which topology file and a machine driver to load. By reporting the slave_ids exposed in ACPI tables, the parent SOF driver will be able to compare against a set of static configurations. This patch only adds the interface change, the functionality is added in future patches. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/linux/soundwire/sdw_intel.h | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 93b83bdf8035..979b41b5dcb4 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -5,6 +5,7 @@ #define __SDW_INTEL_H #include <linux/irqreturn.h> +#include <linux/soundwire/sdw.h> /** * struct sdw_intel_stream_params_data: configuration passed during @@ -93,6 +94,11 @@ struct sdw_intel_link_res; */ #define SDW_INTEL_CLK_STOP_BUS_RESET BIT(3) +struct sdw_intel_slave_id { + int link_id; + struct sdw_slave_id id; +}; + /** * struct sdw_intel_ctx - context allocated by the controller * driver probe @@ -101,9 +107,12 @@ struct sdw_intel_link_res; * hardware capabilities after all power dependencies are settled. * @link_mask: bit-wise mask listing SoundWire links reported by the * Controller + * @num_slaves: total number of devices exposed across all enabled links * @handle: ACPI parent handle * @links: information for each link (controller-specific and kept * opaque here) + * @ids: array of slave_id, representing Slaves exposed across all enabled + * links * @link_list: list to handle interrupts across all links * @shim_lock: mutex to handle concurrent rmw access to shared SHIM registers. */ @@ -111,8 +120,10 @@ struct sdw_intel_ctx { int count; void __iomem *mmio_base; u32 link_mask; + int num_slaves; acpi_handle handle; struct sdw_intel_link_res *links; + struct sdw_intel_slave_id *ids; struct list_head link_list; struct mutex shim_lock; /* lock for access to shared SHIM registers */ }; -- 2.20.1

2 4

[alsa-devel] [PATCH V2 1/2] regulator: core: Add regulator_is_equal() helper
by Marek Vasut 16 Jan '20

16 Jan '20

Add regulator_is_equal() helper to compare whether two regulators are the same. This is useful for checking whether two separate regulators in a driver are actually the same supply. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Igor Opaniuk <igor.opaniuk(a)toradex.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Oleksandr Suvorov <oleksandr.suvorov(a)toradex.com> --- V2: New patch --- drivers/regulator/helpers.c | 14 ++++++++++++++ include/linux/regulator/consumer.h | 7 +++++++ 2 files changed, 21 insertions(+) diff --git a/drivers/regulator/helpers.c b/drivers/regulator/helpers.c index ca3dc3f3bb292..bb16c465426ef 100644 --- a/drivers/regulator/helpers.c +++ b/drivers/regulator/helpers.c @@ -13,6 +13,8 @@ #include <linux/regulator/driver.h> #include <linux/module.h> +#include "internal.h" + /** * regulator_is_enabled_regmap - standard is_enabled() for regmap users * @@ -881,3 +883,15 @@ void regulator_bulk_set_supply_names(struct regulator_bulk_data *consumers, consumers[i].supply = supply_names[i]; } EXPORT_SYMBOL_GPL(regulator_bulk_set_supply_names); + +/** + * regulator_is_equal - test whether two regulators are the same + * + * @reg1: first regulator to operate on + * @reg2: second regulator to operate on + */ +bool regulator_is_equal(struct regulator *reg1, struct regulator *reg2) +{ + return reg1->rdev == reg2->rdev; +} +EXPORT_SYMBOL_GPL(regulator_is_equal); diff --git a/include/linux/regulator/consumer.h b/include/linux/regulator/consumer.h index 337a463915278..2c89d886595cb 100644 --- a/include/linux/regulator/consumer.h +++ b/include/linux/regulator/consumer.h @@ -287,6 +287,8 @@ void regulator_bulk_set_supply_names(struct regulator_bulk_data *consumers, const char *const *supply_names, unsigned int num_supplies); +bool regulator_is_equal(struct regulator *reg1, struct regulator *reg2); + #else /* @@ -593,6 +595,11 @@ regulator_bulk_set_supply_names(struct regulator_bulk_data *consumers, { } +static inline bool +regulator_is_equal(struct regulator *reg1, struct regulator *reg2); +{ + return false; +} #endif static inline int regulator_set_voltage_triplet(struct regulator *regulator, -- 2.24.1

3 5

[alsa-devel] [PATCH] ALSA: usb-audio: add implicit fb quirk for MOTU M Series
by Alexander Tsoy 16 Jan '20

16 Jan '20

This fixes crackling sound during playback. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- MOTU is known for reusing Product IDs for different devices or different generations of the device (e.g. MicroBook I/II/IIc shares a single Product ID). This patch was only tested with M4 audio interface, but the same Product ID is also used by M2. Hope it will work for M2 as well. sound/usb/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a11c8150af58..4f1c00615d94 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -348,6 +348,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x84; ifnum = 0; goto add_sync_ep_from_ifnum; + case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + ep = 0x81; + ifnum = 2; + goto add_sync_ep_from_ifnum; case USB_ID(0x0582, 0x01d8): /* BOSS Katana */ /* BOSS Katana amplifiers do not need quirks */ return 0; -- 2.24.1

2 1

[alsa-devel] Question about runtime PM for HDA codecs
by Sridharan, Ranjani 16 Jan '20

16 Jan '20

Hi Takashi, Can I please bother you with a question about how to handle the situation with runtime PM for HDA codec on one of the Intel CML-based platforms with an ALC285 codec. Basically, the problem we're facing is that when using the SOF driver, jack detection doesn't seem to work as intended because the codec gets suspended right after the headset is plugged in. After a bit of experimenting, what we found was that calling snd_hda_set_power_save() with a delay of 0 or something > 0, both help with fixing the problem. But, I have a basic question about this. What is the expectation for enabling the codec runtime PM? I see that the legacy driver calls snd_hda_set_power_save() based on the CONFIG_SND_HDA_POWER_SAVE_DEFAULT value. In the case of SOF, we do not explicitly set this value so it is 0 by default. Also, when the codec is registered, the runtime PM for the codec is enabled by default without checking if the power_save delay is set to 0 or not. And later when the snd_hda_set_power_save() is called from the legacy HDA driver probe, it sets the use of auto suspend and the delay to be used based on the config. Would it be correct to remove the pm_runtime_enable() call in snd_hda_codec_register() and let the codec runtime PM enabling be set with the call to snd_hda_set_power_save() for both the legacy HDA driver and the SOF driver? Thanks, Ranjani

3 5

[alsa-devel] [PATCH] ALSA: seq: Fix racy access for queue timer in proc read
by Takashi Iwai 15 Jan '20

15 Jan '20

snd_seq_info_timer_read() reads the information of the timer assigned for each queue, but it's done in a racy way which may lead to UAF as spotted by syzkaller. This patch applies the missing q->timer_mutex lock while accessing the timer object as well as a slight code change to adapt the standard coding style. Reported-by: syzbot+2b2ef983f973e5c40943(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_timer.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 63dc7bdb622d..be59b59c9be4 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -471,15 +471,19 @@ void snd_seq_info_timer_read(struct snd_info_entry *entry, q = queueptr(idx); if (q == NULL) continue; - if ((tmr = q->timer) == NULL || - (ti = tmr->timeri) == NULL) { - queuefree(q); - continue; - } + mutex_lock(&q->timer_mutex); + tmr = q->timer; + if (!tmr) + goto unlock; + ti = tmr->timeri; + if (!ti) + goto unlock; snd_iprintf(buffer, "Timer for queue %i : %s\n", q->queue, ti->timer->name); resolution = snd_timer_resolution(ti) * tmr->ticks; snd_iprintf(buffer, " Period time : %lu.%09lu\n", resolution / 1000000000, resolution % 1000000000); snd_iprintf(buffer, " Skew : %u / %u\n", tmr->skew, tmr->skew_base); +unlock: + mutex_unlock(&q->timer_mutex); queuefree(q); } } -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: unlock on error in probe
by Dan Carpenter 15 Jan '20

15 Jan '20

We need to unlock before we returning on this error path. Fixes: 73ac9f5e5b43 ("ALSA: usb-audio: Add boot quirk for MOTU M Series") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 2f582ac7cf78..827fb0bc8b56 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -602,7 +602,7 @@ static int usb_audio_probe(struct usb_interface *intf, if (! chip) { err = snd_usb_apply_boot_quirk_once(dev, intf, quirk, id); if (err < 0) - return err; + goto __error; /* it's a fresh one. * now look for an empty slot and create a new card instance -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: dt-bindings: stm32: convert spdfirx to json-schema
by Olivier Moysan 15 Jan '20

15 Jan '20

Convert the STM32 SPDIFRX bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- .../bindings/sound/st,stm32-spdifrx.txt | 56 ------------- .../bindings/sound/st,stm32-spdifrx.yaml | 80 +++++++++++++++++++ 2 files changed, 80 insertions(+), 56 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt deleted file mode 100644 index 33826f2459fa..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt +++ /dev/null @@ -1,56 +0,0 @@ -STMicroelectronics STM32 S/PDIF receiver (SPDIFRX). - -The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with -IEC-60958 and IEC-61937. - -Required properties: - - compatible: should be "st,stm32h7-spdifrx" - - reg: cpu DAI IP base address and size - - clocks: must contain an entry for kclk (used as S/PDIF signal reference) - - clock-names: must contain "kclk" - - interrupts: cpu DAI interrupt line - - dmas: DMA specifiers for audio data DMA and iec control flow DMA - See STM32 DMA bindings, Documentation/devicetree/bindings/dma/stm32-dma.txt - - dma-names: two dmas have to be defined, "rx" and "rx-ctrl" - -Optional properties: - - resets: Reference to a reset controller asserting the SPDIFRX - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -spdifrx: spdifrx@40004000 { - compatible = "st,stm32h7-spdifrx"; - reg = <0x40004000 0x400>; - clocks = <&rcc SPDIFRX_CK>; - clock-names = "kclk"; - interrupts = <97>; - dmas = <&dmamux1 2 93 0x400 0x0>, - <&dmamux1 3 94 0x400 0x0>; - dma-names = "rx", "rx-ctrl"; - pinctrl-0 = <&spdifrx_pins>; - pinctrl-names = "default"; - - spdifrx_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - }; - }; -}; - -spdif_in: spdif-in { - compatible = "linux,spdif-dir"; - - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; - -soundcard { - compatible = "audio-graph-card"; - dais = <&spdifrx_port>; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml new file mode 100644 index 000000000000..ab8e9d74ac3c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-spdifrx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 S/PDIF receiver (SPDIFRX) + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: | + The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with + IEC-60958 and IEC-61937. + +properties: + compatible: + enum: + - st,stm32h7-spdifrx + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: kclk + + interrupts: + maxItems: 1 + + dmas: + items: + - description: audio data capture DMA + - description: IEC status bits capture DMA + minItems: 1 + maxItems: 2 + + dma-names: + items: + - const: rx + - const: rx-ctrl + + resets: + maxItems: 1 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + spdifrx: spdifrx@40004000 { + compatible = "st,stm32h7-spdifrx"; + #sound-dai-cells = <0>; + reg = <0x40004000 0x400>; + clocks = <&rcc SPDIF_K>; + clock-names = "kclk"; + interrupts = <GIC_SPI 97 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 2 93 0x400 0x0>, + <&dmamux1 3 94 0x400 0x0>; + dma-names = "rx", "rx-ctrl"; + pinctrl-0 = <&spdifrx_pins>; + pinctrl-names = "default"; + }; + +... -- 2.17.1

2 1

[alsa-devel] Simultaneous play/record on implicit feedback device causes 'endpoint in use' error
by Robert Giaraffa 15 Jan '20

15 Jan '20

My company (Silicon Labs) makes a CP2615 USB audio class 1.0 device which uses implicit feedback. I saw these two issues when testing with aplay/arecord/alsabat/alsaloop: * During playback-only operation, the implicit feedback (IN) endpoint was not active. * During play/record operation, the implicit feedback values were not acted on, i.e. the host always sent 192 bytes per OUT packet even if previous IN packets had been 188 or 196 bytes. This behavior was observed on these systems: RaspberryPi2: Raspbian 9 (stretch), kernel v4.14.72 x86-64: Ubuntu 18.04, kernel v4.15.0-36 I was able to get implicit feedback working properly in playback mode by adding a device-specific quirk to 'pcm.c' that is similar to existing quirks for implicit-feedback devices, e.g. M-Audio Fast Track Ultra. (Note: The following patch applies to kernel v4.14.72. In v4.18, the set_sync_ep_implicit_fb_quirk function was changed so that there is a single call to usb_ifnum_to_if(), rather than a call for each device-dependent quirk.) $ git diff pcm.c diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b9c9a19f..6fb5567b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -358,6 +358,21 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, alts = &iface->altsetting[1]; goto add_sync_ep; + case USB_ID(0x10C4, 0xEAC1): // Silicon Labs CP2615 + // Ensure CP2615 is configured in Async Mode + if (attr != USB_ENDPOINT_SYNC_ASYNC) + return 0; + + // In P16R16 async mode, AudioIN EP (IF:4 EP:0x83) used for implicit feedback + ep = 0x83; + iface = usb_ifnum_to_if(dev, 4); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + } if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && After adding the patch, I verified that implicit feedback worked properly for playback, i.e. the OUT samples per packet tracked previous IN values. However, when attempting to perform simultaneous play and record via alsabat, alsaloop, or separate aplay/arecord, the system reports the 'endpoint already in use' error when attempting to set the format for the IN endpoint that is already in use (note that this dmesg log has customized printk messages): [Oct23 10:13] snd_usb_hw_params->set_format() [ +0.000014] set_format(): dev:b3856c00 alts:b67e8824 ifnum: 3 [ +0.001654] set_format()->usb_set_interface 3:1 [ +0.000006] Creating new playback data endpoint 03 [ +0.000004] Creating new capture data endpoint 83 <Begin Playback. SYNC endpoint also starts because feedback is implicit> [ +0.000027] snd_usb_pcm_prepare->set_format() [ +0.000005] set_format(): dev:b3856c00 alts:b67e8824 ifnum: 3 [ +0.000755] Setting params for EP 03 (type 0, 12 urbs), ret=0 [ +0.000017] Setting params for EP 83 (type 0, 12 urbs), ret=0 [ +0.000004] Starting DATA_EP 03 @b9224000 [ +0.000004] snd_usb_endpoint_implicit_feedback_sink (03) is TRUE [ +0.001495] Starting SYNC_EP 83 @b9222000 <After playing for 0.5s, start recording. Failure occurs because EP83 is already in use.> [ +0.493035] snd_usb_hw_params->set_format() [ +0.000016] set_format(): dev:b3856c00 alts:b67e88a4 ifnum: 4 [ +0.001549] set_format()->usb_set_interface 4:1 [ +0.000005] Re-using EP 83 in iface 4:1 [ +0.000028] snd_usb_pcm_prepare->set_format() [ +0.000004] set_format(): dev:b3856c00 alts:b67e88a4 ifnum: 4 *** [ +0.003196] *** snd_usb_endpoint_set_params(): EP #83: already in use [ +0.000003] configure_endpoint->endpoint_set_params: err=FFFFFFF0 I bought a used Fast Track Ultra device for comparison testing, and found that it behaved the same way as the CP2615 does with the implicit feedback quirk: in playback-only mode implicit feedback works properly, but play/record causes the 'EP already in use' error. Are there any suggestions on how I can resolve this issue? Any help would be greatly appreciated! Thanks, Bob Giaraffa

3 4

[alsa-devel] arecord: buffer-size dependency on period-size and other params
by Rajwa, Marcin 15 Jan '20

15 Jan '20

Hello, I have spotted strange behavior of alsa-lib when I try to set buffer-size in arecord. It looks like the size of the buffer is very dependent on other parameters like period-size and rate for example. At first it sounds like normal behaviour, after all buffer size smaller than period size doesn’t make much sense. However what if we go in the reverse direction and set buffer-size much larger than period size? Then I see no reason to forbid such request (at least on the user space level). So to be specific I issue such command: arecod -Dplughw:0,8 -r 16000 -c 2 -f S24_LE --buffer_size 1280000 tmp.wav -vvv. The arecord response for this is buffer size of 31992 frames, 255936 bytes. Now let me also tell the buffer size of 1280000 bytes is the maximum buffer size our driver allows, likewise max period is 268800 bytes. Now if inside our driver I double max period size limit I get buffer size accordingly bigger. Similarly, if I change the format in the command line request from S24_LE to i.e S16_LE then again I will get buffer size closer to my request (bigger). So below are my questions: 1. Why alsa-lib behaves in a way I described above? Why I can not control buffer size independently from other parameters (providing it has physical sense). 2. It looks like alsa-lib first calculates initial buffer-size base on other input params like mentioned period size or sampling rate and then ask driver to refine it according to its capabilities. Is that the case? 3. I also tried to dump hw capabilities by adding –dump-hw-param flag to my command line request and here yet another surprise – in a row dedicated for buffer size it says “BUFFER_SIZE [12 4294967294]”. Where is this “4294967294” comes from, and on what basics is it calculated? I would expect at the bare minimum it should ask driver but in the snd_pcm_hw_param_dump() function I don’t see any communication with the driver. Kind Regards, Marcin Rajwa

3 2

[alsa-devel] Applied "regulator fix for "regulator: core: Add regulator_is_equal() helper"" to the asoc tree
by Mark Brown 15 Jan '20

15 Jan '20

The patch regulator fix for "regulator: core: Add regulator_is_equal() helper" has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0468e667a5bead9c1b7ded92861b5a98d8d78745 Mon Sep 17 00:00:00 2001 From: Stephen Rothwell <sfr(a)canb.auug.org.au> Date: Wed, 15 Jan 2020 12:02:58 +1100 Subject: [PATCH] regulator fix for "regulator: core: Add regulator_is_equal() helper" Signed-off-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Link: https://lore.kernel.org/r/20200115120258.0e535fcb@canb.auug.org.au Acked-by: Marek Vasut <marex(a)denx.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/linux/regulator/consumer.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/linux/regulator/consumer.h b/include/linux/regulator/consumer.h index 2c89d886595c..6a92fd3105a3 100644 --- a/include/linux/regulator/consumer.h +++ b/include/linux/regulator/consumer.h @@ -596,7 +596,7 @@ regulator_bulk_set_supply_names(struct regulator_bulk_data *consumers, } static inline bool -regulator_is_equal(struct regulator *reg1, struct regulator *reg2); +regulator_is_equal(struct regulator *reg1, struct regulator *reg2) { return false; } -- 2.20.1

1 0

[alsa-devel] Bug report for changes in ucm2 with chtrt5645 card on Lenovo Miix 320
by Tim Schumacher 15 Jan '20

15 Jan '20

Hi alsa devs, I'm on Arch Linux on a rather uncommon laptop (Lenovo Miix 320) and my sound stopped working on the upgrade from alsa-lib 1.1.9 to 1.2.1. wabbits on IRC helped me track down the problem to a change in the /usr/share/alsa/ucm (now /usr/share/alsa/ucm2) files. It seems like /usr/share/alsa/ucm2/chtrt5645/HiFi-dmic2.conf and a symlink at /usr/share/alsa/ucm2/chtrt5645/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf were specifically added for this laptop but in the contrary they break the sound for me. After the upgrade to alsa-lib 1.2.1 the sound is not working. The commands play no sound: pasuspender -- speaker-test --nloops=1 --channels=2 --test=wav --device=hw:0,0 pasuspender -- speaker-test --nloops=1 --channels=2 --test=wav --device=hw:0,1 And this is the alsa-info.sh output http://alsa-project.org/db/?f=f883910a5c5101b4b1ea4202d1fe84ccd139f796 After deleting the /usr/share/alsa/ucm2/chtrt5645/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf symlink both commands from above play sound and this is the alsa-info.sh output http://alsa-project.org/db/?f=e759eb9118a191b6c3b8c021fed58abc9cf95076 Thanks. - Tim

2 6

[alsa-devel] [PATCH] ALSA: hda/analog - Minor optimization for SPDIF mux connections
by Takashi Iwai 15 Jan '20

15 Jan '20

AD HD-audio codec driver has a few code lines invoking snd_get_num_conns() and using its return value as the array index without checking. This is basically safe in all those places; at the second and later calls snd_get_num_conns() returns the value cached from the first invocation, hence the value is always consistent. However, it looks a bit confusing as if a lack of the proper check. This patch introduces a new field num_smux_conns in ad198x_spec for simplifying the code. Now we store and refer to the value more locally without invoking the extra function at each time. Reported-by: Colin King <colin.king(a)canonical.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 88c46b051d14..2132b2acec4d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -28,6 +28,7 @@ struct ad198x_spec { hda_nid_t eapd_nid; unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ + int num_smux_conns; }; @@ -453,8 +454,7 @@ static int ad1983_auto_smux_enum_info(struct snd_kcontrol *kcontrol, struct ad198x_spec *spec = codec->spec; static const char * const texts2[] = { "PCM", "ADC" }; static const char * const texts3[] = { "PCM", "ADC1", "ADC2" }; - hda_nid_t dig_out = spec->gen.multiout.dig_out_nid; - int num_conns = snd_hda_get_num_conns(codec, dig_out); + int num_conns = spec->num_smux_conns; if (num_conns == 2) return snd_hda_enum_helper_info(kcontrol, uinfo, 2, texts2); @@ -481,7 +481,7 @@ static int ad1983_auto_smux_enum_put(struct snd_kcontrol *kcontrol, struct ad198x_spec *spec = codec->spec; unsigned int val = ucontrol->value.enumerated.item[0]; hda_nid_t dig_out = spec->gen.multiout.dig_out_nid; - int num_conns = snd_hda_get_num_conns(codec, dig_out); + int num_conns = spec->num_smux_conns; if (val >= num_conns) return -EINVAL; @@ -512,6 +512,7 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec) num_conns = snd_hda_get_num_conns(codec, dig_out); if (num_conns != 2 && num_conns != 3) return 0; + spec->num_smux_conns = num_conns; if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &ad1983_auto_smux_mixer)) return -ENOMEM; return 0; @@ -730,10 +731,12 @@ static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad198x_spec *spec = codec->spec; static const char * const texts[] = { "PCM", "ADC1", "ADC2", "ADC3", }; - int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; + int num_conns = spec->num_smux_conns; + if (num_conns > 4) num_conns = 4; return snd_hda_enum_helper_info(kcontrol, uinfo, num_conns, texts); @@ -756,7 +759,7 @@ static int ad1988_auto_smux_enum_put(struct snd_kcontrol *kcontrol, struct ad198x_spec *spec = codec->spec; unsigned int val = ucontrol->value.enumerated.item[0]; struct nid_path *path; - int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; + int num_conns = spec->num_smux_conns; if (val >= num_conns) return -EINVAL; @@ -847,6 +850,7 @@ static int ad1988_add_spdif_mux_ctl(struct hda_codec *codec) num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; if (num_conns != 3 && num_conns != 4) return 0; + spec->num_smux_conns = num_conns; for (i = 0; i < num_conns; i++) { struct nid_path *path = snd_array_new(&spec->gen.paths); -- 2.16.4

1 0

[alsa-devel] ABI breakage in alsa-lib 1.2
by Jordi Mallach 15 Jan '20

15 Jan '20

[Please Cc me on replies, I'm not currently subscribed to alsa-devel] Hello, I'm one of the ALSA Debian maintainers, and some weeks ago I looked into updating our ALSA packages to 1.2.1. I quickly noticed there's been some fundamental change to libasound2, one that appears to break ABI and should have warranted an soname bump at least. With the split of the snd_tplg API to the new libatopology2, libasound2 has been left missing that part of the API that was part of the asound.h header, etc. This could have been a somewhat lesser issue if libasound2 linked to libatopology, but unfortunately this goes in the other direction, so the ABI and API breakage is not avoided. I cannot update alsa-lib in this state. I could hack around and patch it to reinstate the missing api in libasound, but this sounds like pain and suffering for everyone involved. How should we move forward with this? I am sorry I only come up with this 2 months after the release when I had identified the problem many weeks ago, I've been extremely busy. Thanks, Jordi -- Jordi Mallach <jordi(a)debian.org> Debian Project

2 1

[alsa-devel] [PATCH RFC 0/3] ASoC: Add Multi CPU DAI support
by Bard liao 15 Jan '20

15 Jan '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html Shreyas NC (3): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add multiple CPU DAI support in DAPM include/sound/soc.h | 15 ++ sound/soc/soc-core.c | 183 ++++++++++-------- sound/soc/soc-dapm.c | 131 +++++++------ sound/soc/soc-pcm.c | 432 ++++++++++++++++++++++++++++--------------- 4 files changed, 483 insertions(+), 278 deletions(-) -- 2.17.1

4 7

[alsa-devel] [PATCH] ALSA: hda - fix out of bounds read on spec->smux_paths
by Colin King 14 Jan '20

14 Jan '20

From: Colin Ian King <colin.king(a)canonical.com> It is possible for the call to snd_hda_get_num_conns to fail and return a negative error code that gets assigned to num_conns. In that specific case, the check of very large values of val against num_conns will not fail the -EINVAL check and later on an out of bounds array read on spec->smux_paths will occur. Fix this by sanity checking for an error return from the call to snd_hda_get_num_conns. Addresses-Coverity: ("Out-of-bounds read") Fixes: 272f3ea31776 ("ALSA: hda - Add SPDIF mux control to AD codec auto-parser") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/hda/patch_analog.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 88c46b051d14..399561369495 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -756,9 +756,11 @@ static int ad1988_auto_smux_enum_put(struct snd_kcontrol *kcontrol, struct ad198x_spec *spec = codec->spec; unsigned int val = ucontrol->value.enumerated.item[0]; struct nid_path *path; - int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; + int num_conns = snd_hda_get_num_conns(codec, 0x0b); - if (val >= num_conns) + if (num_conns < 0) + return num_conns; + if (val >= num_conns + 1) return -EINVAL; if (spec->cur_smux == val) return 0; -- 2.24.0

3 2

[alsa-devel] [PATCH 0/5] soundwire: intel: add DAI callbacks
by Pierre-Louis Bossart 14 Jan '20

14 Jan '20

The existing mainline code is missing most of the DAI callbacks needed for a functional implementation, and the existing ones need to be modified to provide the relevant information to SOF drivers. As suggested by Vinod, these patches are shared first - with the risk that they are separated from the actual DAI enablement, so reviewers might wonder why they are needed in the first place. For reference, the complete set of 100+ patches required for SoundWire on Intel platforms is available here: https://github.com/thesofproject/linux/pull/1692 Pierre-Louis Bossart (2): soundwire: intel: rename res field as link_res soundwire: intel: free all resources on hw_free() Rander Wang (3): soundwire: intel: add prepare support in sdw dai driver soundwire: intel: add trigger support in sdw dai driver soundwire: intel: add sdw_stream_setup helper for .startup callback drivers/soundwire/intel.c | 198 ++++++++++++++++++++++++++++++++++---- 1 file changed, 178 insertions(+), 20 deletions(-) base-commit: b637124800a157c4df3699d1137d8533394f7678 -- 2.20.1

2 7

[alsa-devel] how to use DPCM to alter audio path between headphones and speaker
by Tim Harvey 14 Jan '20

14 Jan '20

Greetings, I'm working with a board that has a TI TLV320AIC3105 codec connected to an IMX6 SSI with a headphone jack and a TS3A227E headphone detect chip (HPROUT/HPLOUT), mono speaker amp output (RIGHT_LOP/RIGHT_LOM), and mic in (MIC1L). It seems that Dynamic PCM allows for the automatic routing of audio to change between headphone out and speaker out depending on jack insertion/removal event but I'm not clear how to configure that. Can this be done via simple-audio-card bindings or does it require a modified machine driver? Any pointers or references are greatly appreciated! Best Regards, Tim

1 0

[alsa-devel] [PATCH 0/4] ASoC: msm8916-wcd-analog: MIC BIAS fixes/additions
by Stephan Gerhold 14 Jan '20

14 Jan '20

This patch series fixes some bugs in the MIC BIAS implementation in msm8916-wcd-analog. Finally, it adds support for MIC BIAS Internal3, which is needed for msm8916-longcheer-l8150. Stephan Gerhold (4): ASoC: msm8916-wcd-analog: Fix selected events for MIC BIAS External1 ASoC: msm8916-wcd-analog: Fix MIC BIAS Internal1 ASoC: msm8916-wcd-analog: Simplify MIC BIAS Internal ASoC: msm8916-wcd-analog: Add MIC BIAS Internal3 sound/soc/codecs/msm8916-wcd-analog.c | 59 ++++++++++----------------- 1 file changed, 21 insertions(+), 38 deletions(-) -- 2.24.1

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[alsa-devel] [PATCH] ASoC: sgtl5000: add multi endpoint support
by Marek Vasut 14 Jan '20

14 Jan '20

Support multiple endpoints on SGTL5000 codec port when used in of_graph context. This patch allows to share the codec port between two CPU DAIs. Example: Custom STM32MP157C board uses SGTL5000 audio codec. This codec is connected to two serial audio interfaces, which are configured either as rx or tx. >From AsoC point of view the topolgy is the following: // 2 CPU DAIs (SAI2A/B), 1 Codec (SGTL5000) Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0 Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0 In the DT two endpoints have to be associated to the codec port: sgtl5000_port: port { sgtl5000_tx_endpoint: endpoint@0 { remote-endpoint = <&sai2a_endpoint>; }; sgtl5000_rx_endpoint: endpoint@1 { remote-endpoint = <&sai2b_endpoint>; }; }; However, when the audio graph card parses the codec nodes, it expects to find DAI interface indexes matching the endpoints indexes. The current patch forces the use of DAI id 0 for both endpoints, which allows to share the codec DAI between the two CPU DAIs for playback and capture streams respectively. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> --- NOTE: This is the same as commit ad6bb3067c4d ("ASoC: cs42l51: add multi endpoint support") --- sound/soc/codecs/sgtl5000.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index aa1f9637d8950..93da6b58c97df 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1513,6 +1513,13 @@ static int sgtl5000_probe(struct snd_soc_component *component) return ret; } +static int sgtl5000_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + /* return dai id 0, whatever the endpoint index */ + return 0; +} + static const struct snd_soc_component_driver sgtl5000_driver = { .probe = sgtl5000_probe, .set_bias_level = sgtl5000_set_bias_level, @@ -1522,6 +1529,7 @@ static const struct snd_soc_component_driver sgtl5000_driver = { .num_dapm_widgets = ARRAY_SIZE(sgtl5000_dapm_widgets), .dapm_routes = sgtl5000_dapm_routes, .num_dapm_routes = ARRAY_SIZE(sgtl5000_dapm_routes), + .of_xlate_dai_id = sgtl5000_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, -- 2.24.1

2 1

[alsa-devel] [PATCH] soundwire: bus: fix device number leak on errors
by Pierre-Louis Bossart 14 Jan '20

14 Jan '20

If the programming of the dev_number fails due to an IO error, a new device_number will be assigned, resulting in a leak. Make sure we only assign a device_number once per Slave device. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/bus.c | 37 ++++++++++++++++++++++------------- include/linux/soundwire/sdw.h | 4 +++- 2 files changed, 26 insertions(+), 15 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index be5d437058ed..8f973e70e6f7 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -456,26 +456,35 @@ static int sdw_get_device_num(struct sdw_slave *slave) static int sdw_assign_device_num(struct sdw_slave *slave) { int ret, dev_num; + bool new_device = false; /* check first if device number is assigned, if so reuse that */ if (!slave->dev_num) { - mutex_lock(&slave->bus->bus_lock); - dev_num = sdw_get_device_num(slave); - mutex_unlock(&slave->bus->bus_lock); - if (dev_num < 0) { - dev_err(slave->bus->dev, "Get dev_num failed: %d\n", - dev_num); - return dev_num; + if (!slave->dev_num_sticky) { + mutex_lock(&slave->bus->bus_lock); + dev_num = sdw_get_device_num(slave); + mutex_unlock(&slave->bus->bus_lock); + if (dev_num < 0) { + dev_err(slave->bus->dev, "Get dev_num failed: %d\n", + dev_num); + return dev_num; + } + slave->dev_num = dev_num; + slave->dev_num_sticky = dev_num; + new_device = true; + } else { + slave->dev_num = slave->dev_num_sticky; } - } else { + } + + if (!new_device) dev_info(slave->bus->dev, - "Slave already registered dev_num:%d\n", + "Slave already registered, reusing dev_num:%d\n", slave->dev_num); - /* Clear the slave->dev_num to transfer message on device 0 */ - dev_num = slave->dev_num; - slave->dev_num = 0; - } + /* Clear the slave->dev_num to transfer message on device 0 */ + dev_num = slave->dev_num; + slave->dev_num = 0; ret = sdw_write(slave, SDW_SCP_DEVNUMBER, dev_num); if (ret < 0) { @@ -485,7 +494,7 @@ static int sdw_assign_device_num(struct sdw_slave *slave) } /* After xfer of msg, restore dev_num */ - slave->dev_num = dev_num; + slave->dev_num = slave->dev_num_sticky; return 0; } diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index b7c9eca4332a..b451bb622335 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -546,7 +546,8 @@ struct sdw_slave_ops { * @debugfs: Slave debugfs * @node: node for bus list * @port_ready: Port ready completion flag for each Slave port - * @dev_num: Device Number assigned by Bus + * @dev_num: Current Device Number, values can be 0 or dev_num_sticky + * @dev_num_sticky: one-time static Device Number assigned by Bus * @probed: boolean tracking driver state * @probe_complete: completion utility to control potential races * on startup between driver probe/initialization and SoundWire @@ -575,6 +576,7 @@ struct sdw_slave { struct list_head node; struct completion *port_ready; u16 dev_num; + u16 dev_num_sticky; bool probed; struct completion probe_complete; struct completion enumeration_complete; -- 2.20.1

2 2

[alsa-devel] [PATCH -next] ASoC: rt700: remove unused including <linux/version.h>
by YueHaibing 14 Jan '20

14 Jan '20

Remove including <linux/version.h> that don't need it. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/rt700.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index b1830c1ebf8a..ff68f0e4f629 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -8,7 +8,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h>

2 1

[alsa-devel] [PATCH] ASoC: SOF: Intel: hda-dai: fix compilation warning in pcm_prepare
by Pierre-Louis Bossart 14 Jan '20

14 Jan '20

Fix GCC warning with W=1, previous cleanup did not remove unnecessary variable. sound/soc/sof/intel/hda-dai.c: In function ‘hda_link_pcm_prepare’: sound/soc/sof/intel/hda-dai.c:265:31: warning: variable ‘hda_stream’ set but not used [-Wunused-but-set-variable] 265 | struct sof_intel_hda_stream *hda_stream; | ^~~~~~~~~~ Fixes: a3ebccb52efdf ("ASoC: SOF: Intel: hda: reset link DMA state in prepare") Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-dai.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 313611dcb5e4..9c6e3f990ee3 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -262,14 +262,11 @@ static int hda_link_pcm_prepare(struct snd_pcm_substream *substream, { struct hdac_ext_stream *link_dev = snd_soc_dai_get_dma_data(dai, substream); - struct sof_intel_hda_stream *hda_stream; struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(dai->component); struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); int stream = substream->stream; - hda_stream = hstream_to_sof_hda_stream(link_dev); - if (link_dev->link_prepared) return 0; -- 2.20.1

2 1

[alsa-devel] [PATCH 00/18] ASoC: Intel: remove compilation/static analysis warnings
by Pierre-Louis Bossart 14 Jan '20

14 Jan '20

More trivial fixes for Intel machine drivers in my quixotic quest for zero warnings. None of them are critical but they generate noise that prevents developers from using those tools to detect real issues. Pierre-Louis Bossart (18): ASoC: Intel: bxt_da7219_max98357a: rename shadowed variable ASoC: Intel: bxt_rt298: rename shadowed variable ASoC: Intel: glk_rt5682_max98357a: rename shadowed variable ASoC: Intel: glk_rt5682_max98357a: removed unused variables ASoC: Intel: kbl_da7219_max98357a: rename shadowed variable ASoC: Intel: kbl_da7219_max98927: test return value on route add ASoC: Intel: kbl_da7219_max98927: rename shadowed variable ASoC: Intel: kbl_da7219_max98927: remove unnecessary initialization ASoC: Intel: kbl_rt5660: rename shadowed variable ASoC: Intel: kbl_rt5663_max98927: rename shadowed variable ASoC: Intel: kbl_rt5663_rt5514_max98927: rename shadowed variable ASOC: Intel: kbl_rt5663_rt5514_max98927: remove useless initialization ASoC: Intel: skl_nau88l25_max98357a: rename shadowed variable ASoC: Intel: skl_nau88l25_ssm4567: rename shadowed variable ASoC: Intel: skl_rt286: rename shadowed variable ASoC: Intel: bytcht_es8316: removed unused variable ASoC: Intel: bytcr_rt5640: remove unused variable ASoC: Intel: bytcr_rt5651: remove unused variable sound/soc/intel/boards/bxt_da7219_max98357a.c | 10 +++++----- sound/soc/intel/boards/bxt_rt298.c | 8 ++++---- sound/soc/intel/boards/bytcht_es8316.c | 8 -------- sound/soc/intel/boards/bytcr_rt5640.c | 8 -------- sound/soc/intel/boards/bytcr_rt5651.c | 8 -------- sound/soc/intel/boards/glk_rt5682_max98357a.c | 18 ++++-------------- sound/soc/intel/boards/kbl_da7219_max98357a.c | 10 +++++----- sound/soc/intel/boards/kbl_da7219_max98927.c | 19 +++++++++++-------- sound/soc/intel/boards/kbl_rt5660.c | 4 ++-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 10 +++++----- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 10 +++++----- .../soc/intel/boards/skl_nau88l25_max98357a.c | 10 +++++----- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 10 +++++----- sound/soc/intel/boards/skl_rt286.c | 10 +++++----- 14 files changed, 56 insertions(+), 87 deletions(-) -- 2.20.1

2 36

[alsa-devel] [PATCH] ASoC: rt715: remove warnings
by Pierre-Louis Bossart 14 Jan '20

14 Jan '20

make W=1 reports defined-but-not-used and kernel-doc formatting issues. No functionality change. Fixes: d1ede0641b05e4 ("ASoC: rt715: add RT715 codec driver") Cc: Jack Yu <jack.yu(a)realtek.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/rt715.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 5c6f05b8d8ab..4c746938a062 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -203,7 +203,6 @@ static int rt715_set_amp_gain_get(struct snd_kcontrol *kcontrol, return 0; } -static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); @@ -366,7 +365,7 @@ static const char * const adc_22_23_mux_text[] = { "DMIC4", }; -/** +/* * Due to mux design for nid 24 (MUX_IN3)/25 (MUX_IN4), connection index 0 and * 1 will be connected to the same dmic source, therefore we skip index 1 to * avoid misunderstanding on usage of dapm routing. @@ -404,6 +403,7 @@ static SOC_ENUM_SINGLE_DECL( static SOC_VALUE_ENUM_SINGLE_DECL(rt715_adc24_enum, RT715_MUX_IN3, 0, 0xf, adc_24_mux_text, rt715_adc_24_25_values); + static SOC_VALUE_ENUM_SINGLE_DECL(rt715_adc25_enum, RT715_MUX_IN4, 0, 0xf, adc_25_mux_text, rt715_adc_24_25_values); -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: rt715: use dev_to_sdw_dev() instead of to_sdw_slave_device()
by Kuninori Morimoto 14 Jan '20

14 Jan '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch fixup this error CC sound/soc/codecs/rt715-sdw.o linux/sound/soc/codecs/rt715-sdw.c: In function 'rt715_dev_resume': linux/sound/soc/codecs/rt715-sdw.c:568:28: error: implicit declaration\ of function 'to_sdw_slave_device'; did you mean 'sdw_slave_modalias'?\ [-Werror=implicit-function-declaration] struct sdw_slave *slave = to_sdw_slave_device(dev); ^~~~~~~~~~~~~~~~~~~ sdw_slave_modalias linux/sound/soc/codecs/rt715-sdw.c:568:28: warning: initialization of\ 'struct sdw_slave *' from 'int' makes pointer from integer without a\ cast [-Wint-conversion] cc1: some warnings being treated as errors Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/rt715-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index c35591f..18868e4 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -565,7 +565,7 @@ static int rt715_dev_suspend(struct device *dev) static int rt715_dev_resume(struct device *dev) { - struct sdw_slave *slave = to_sdw_slave_device(dev); + struct sdw_slave *slave = dev_to_sdw_dev(dev); struct rt715_priv *rt715 = dev_get_drvdata(dev); unsigned long time; -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] dt-bindings: sound: Convert jz47*-codec doc to YAML
by Paul Cercueil 14 Jan '20

14 Jan '20

Convert ingenic,jz4740-codec.txt and ingenic,jz4725b-codec.txt to one single ingenic,codec.yaml file, since they share the same binding. Add the ingenic,jz4770-codec compatible string in the process. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> --- .../bindings/sound/ingenic,codec.yaml | 55 +++++++++++++++++++ .../bindings/sound/ingenic,jz4725b-codec.txt | 20 ------- .../bindings/sound/ingenic,jz4740-codec.txt | 20 ------- 3 files changed, 55 insertions(+), 40 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt delete mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt diff --git a/Documentation/devicetree/bindings/sound/ingenic,codec.yaml b/Documentation/devicetree/bindings/sound/ingenic,codec.yaml new file mode 100644 index 000000000000..eb4be86464bb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,codec.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ingenic,codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Ingenic JZ47xx internal codec DT bindings + +maintainers: + - Paul Cercueil <paul(a)crapouillou.net> + +properties: + $nodename: + pattern: '^audio-codec@.*' + + compatible: + oneOf: + - const: ingenic,jz4770-codec + - const: ingenic,jz4725b-codec + - const: ingenic,jz4740-codec + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: aic + + '#sound-dai-cells': + const: 0 + +additionalProperties: false + +required: + - compatible + - reg + - clocks + - clock-names + - '#sound-dai-cells' + +examples: + - | + #include <dt-bindings/clock/jz4740-cgu.h> + codec: audio-codec@10020080 { + compatible = "ingenic,jz4740-codec"; + reg = <0x10020080 0x8>; + #sound-dai-cells = <0>; + clocks = <&cgu JZ4740_CLK_AIC>; + clock-names = "aic"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt deleted file mode 100644 index 05adc0d47b13..000000000000 --- a/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt +++ /dev/null @@ -1,20 +0,0 @@ -Ingenic JZ4725B codec controller - -Required properties: -- compatible : "ingenic,jz4725b-codec" -- reg : codec registers location and length -- clocks : phandle to the AIC clock. -- clock-names: must be set to "aic". -- #sound-dai-cells: Must be set to 0. - -Example: - -codec: audio-codec@100200a4 { - compatible = "ingenic,jz4725b-codec"; - reg = <0x100200a4 0x8>; - - #sound-dai-cells = <0>; - - clocks = <&cgu JZ4725B_CLK_AIC>; - clock-names = "aic"; -}; diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt deleted file mode 100644 index 1ffcade87e7b..000000000000 --- a/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt +++ /dev/null @@ -1,20 +0,0 @@ -Ingenic JZ4740 codec controller - -Required properties: -- compatible : "ingenic,jz4740-codec" -- reg : codec registers location and length -- clocks : phandle to the AIC clock. -- clock-names: must be set to "aic". -- #sound-dai-cells: Must be set to 0. - -Example: - -codec: audio-codec@10020080 { - compatible = "ingenic,jz4740-codec"; - reg = <0x10020080 0x8>; - - #sound-dai-cells = <0>; - - clocks = <&cgu JZ4740_CLK_AIC>; - clock-names = "aic"; -}; -- 2.24.0

3 4

[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt5651: switch to using devm_fwnode_gpiod_get()
by Dmitry Torokhov 14 Jan '20

14 Jan '20

devm_fwnode_get_index_gpiod_from_child() is going away as the name is too unwieldy, let's switch to using the new devm_fwnode_gpiod_get(). Signed-off-by: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> --- sound/soc/intel/boards/bytcr_rt5651.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index c0d322a859f71..4495834e37653 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -998,10 +998,11 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) if (byt_rt5651_gpios) { devm_acpi_dev_add_driver_gpios(codec_dev, byt_rt5651_gpios); - priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child( - &pdev->dev, "ext-amp-enable", 0, - codec_dev->fwnode, - GPIOD_OUT_LOW, "speaker-amp"); + priv->ext_amp_gpio = devm_fwnode_gpiod_get(&pdev->dev, + codec_dev->fwnode, + "ext-amp-enable", + GPIOD_OUT_LOW, + "speaker-amp"); if (IS_ERR(priv->ext_amp_gpio)) { ret_val = PTR_ERR(priv->ext_amp_gpio); switch (ret_val) { @@ -1017,10 +1018,11 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) return ret_val; } } - priv->hp_detect = devm_fwnode_get_index_gpiod_from_child( - &pdev->dev, "hp-detect", 0, - codec_dev->fwnode, - GPIOD_IN, "hp-detect"); + priv->hp_detect = devm_fwnode_gpiod_get(&pdev->dev, + codec_dev->fwnode, + "hp-detect", + GPIOD_IN, + "hp-detect"); if (IS_ERR(priv->hp_detect)) { ret_val = PTR_ERR(priv->hp_detect); switch (ret_val) { -- 2.24.1.735.g03f4e72817-goog -- Dmitry

2 1

[alsa-devel] [PATCH v5 1/2] ASoC: dt-bindings: fsl_asrc: add compatible string for imx8qm & imx8qxp
by Shengjiu Wang 14 Jan '20

14 Jan '20

Add compatible string "fsl,imx8qm-asrc" for imx8qm platform, "fsl,imx8qxp-asrc" for imx8qxp platform. There are two asrc modules in imx8qm & imx8qxp, the clock mapping is different for each other, so add new property "fsl,asrc-clk-map" to distinguish them. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 -none changes in v3 -use only one compatible string "fsl,imx8qm-asrc", -add new property "fsl,asrc-clk-map". changes in v4 -add "fsl,imx8qxp-asrc" changes in v5 -refine the comments for compatible Documentation/devicetree/bindings/sound/fsl,asrc.txt | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt index 1d4d9f938689..cb9a25165503 100644 --- a/Documentation/devicetree/bindings/sound/fsl,asrc.txt +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -8,7 +8,12 @@ three substreams within totally 10 channels. Required properties: - - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + - compatible : Compatible list, should contain one of the following + compatibles: + "fsl,imx35-asrc", + "fsl,imx53-asrc", + "fsl,imx8qm-asrc", + "fsl,imx8qxp-asrc", - reg : Offset and length of the register set for the device. @@ -35,6 +40,11 @@ Required properties: - fsl,asrc-width : Defines a mutual sample width used by DPCM Back Ends. + - fsl,asrc-clk-map : Defines clock map used in driver. which is required + by imx8qm/imx8qxp platform + <0> - select the map for asrc0 in imx8qm/imx8qxp + <1> - select the map for asrc1 in imx8qm/imx8qxp + Optional properties: - big-endian : If this property is absent, the little endian mode -- 2.21.0

3 4

[alsa-devel] [PATCH] ASoC: rt715: use dev_to_sdw_dev() macro
by Pierre-Louis Bossart 14 Jan '20

14 Jan '20

The driver used a macro which was renamed during the joint Intel/SoundWire development, but since those renames were not merged by the SoundWire maintainer we have to revert to the original macros as done with the other Realtek codecs. Fixes: d1ede0641b05e4 ("ASoC: rt715: add RT715 codec driver") Cc: Jack Yu <jack.yu(a)realtek.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/rt715-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index c35591fd281b..18868e4ae6e8 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -565,7 +565,7 @@ static int rt715_dev_suspend(struct device *dev) static int rt715_dev_resume(struct device *dev) { - struct sdw_slave *slave = to_sdw_slave_device(dev); + struct sdw_slave *slave = dev_to_sdw_dev(dev); struct rt715_priv *rt715 = dev_get_drvdata(dev); unsigned long time; -- 2.20.1

2 1

[alsa-devel] [PATCH 00/16] ASoC: remove DAI suspend/resume
by Kuninori Morimoto 14 Jan '20

14 Jan '20

Hi Mark Now, ALSA SoC calls 2 type of suspend/resume suspend/resume for Component suspend/resume for DAI When ALSA SoC calls suspend/resume, its orders are 1) Suspend/Resume all CPU DAI if bus-control was 0 2) Suspend/Resume all Component 3) Suspend/Resume all CPU DAI if bus-control was 1 Historically 2)'s "Component" was "Codec", before. In total, CPU has 3 chance to suspend/resume(= 1/2/3), but Codec has 1 chance to suspend/resume(= 2). And now, no driver which is supporting suspend/resume has bus-control. This means 3) is never used. Almost all drivers which is using DAI suspend/resume can simply switch to Component suspend/resume. Few drivers want to use/keep DAI, but, we can call all DAI from Component by using this if needed for_each_component_dais() These patches switches all DAI's suspend/resume to Component one, and removes it. Kuninori Morimoto (16): ASoC: atmel: atmel_ssc_dai: move .suspend/.resume to component ASoC: bcm: cygnus-ssp: move .suspend/.resume to component ASoC: cirrus: ep93xx-i2s: move .suspend/.resume to component ASoC: jz4740: jz4740-i2s: move .suspend/.resume to component ASoC: mediatek: move .suspend/.resume to component ASoC: samsung: s3c24xx-i2s: move .suspend/.resume to component ASoC: samsung: spdif: move .suspend/.resume to component ASoC: sti: sti_uniperif: move .suspend/.resume to component ASoC: ti: omap-mcpdm: move .suspend/.resume to component ASoC: uniphier: move .suspend/.resume to component ASoC: dwc: dwc-i2s: move .suspend/.resume to component ASoC: samsung: i2s: move .suspend/.resume to component ASoC: ux500: ux500_msp_dai: remove unused DAI .suspend/.resume ASoC: pxa: pxa-ssp: move .suspend/.resume to component ASoC: pxa: pxa2xx-i2s: move .suspend/.resume to component ASoC: soc-core: remove DAI suspend/resume include/sound/soc-dai.h | 2 -- sound/soc/atmel/atmel_ssc_dai.c | 18 ++++++------- sound/soc/bcm/cygnus-ssp.c | 39 +++++++++++++++++++++++----- sound/soc/cirrus/ep93xx-i2s.c | 16 ++++++------ sound/soc/dwc/dwc-i2s.c | 32 +++++++++++++---------- sound/soc/jz4740/jz4740-i2s.c | 18 ++++++------- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 12 ++++----- sound/soc/mediatek/common/mtk-afe-fe-dai.h | 4 +-- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 14 ++-------- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 10 +++----- sound/soc/pxa/pxa-ssp.c | 16 ++++++------ sound/soc/pxa/pxa2xx-i2s.c | 12 ++++----- sound/soc/samsung/i2s.c | 13 +++++----- sound/soc/samsung/s3c24xx-i2s.c | 8 +++--- sound/soc/samsung/spdif.c | 18 ++++++++----- sound/soc/soc-core.c | 41 ------------------------------ sound/soc/soc-dai.c | 12 --------- sound/soc/sti/sti_uniperif.c | 12 ++++----- sound/soc/ti/omap-mcpdm.c | 16 ++++++------ sound/soc/uniphier/aio-cpu.c | 31 +++++++++++++++++++--- sound/soc/uniphier/aio-ld11.c | 18 ------------- sound/soc/uniphier/aio-pxs2.c | 14 ---------- sound/soc/uniphier/aio.h | 2 -- sound/soc/ux500/ux500_msp_dai.c | 2 -- 24 files changed, 167 insertions(+), 213 deletions(-) -- 2.7.4

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Re: [alsa-devel] [PATCH v6] ASoC: Add MediaTek MT6660 Speaker Amp Driver
by Takashi Iwai 14 Jan '20

14 Jan '20

On Tue, 14 Jan 2020 10:48:24 +0100, jeff_chang(張世佳) wrote: > > Dear Takashi: > > Thank for your replying. > > 1.> +static int mt6660_component_get_volsw(struct snd_kcontrol *kcontrol, > > + struct snd_ctl_elem_value *ucontrol) { > > +struct snd_soc_component *component = > > +snd_soc_kcontrol_component(kcontrol); > > +struct mt6660_chip *chip = (struct mt6660_chip *) > > +snd_soc_component_get_drvdata(component); > > +int ret = -EINVAL; > > + > > +if (!strcmp(kcontrol->id.name, "Chip Rev")) { > > +ucontrol->value.integer.value[0] = chip->chip_rev & 0x0f; > > +ret = 0; > > +} > > +return ret; > > So, "T0 SEL" control gets always an error when reading? > Then can't we pass simply NULL for get ops instead? > > Jeff : T0 SEL use snd_soc_get_volsw, it will not use this function. Then what's the reason of this hackish check? > 2. So here both 24 and 32 bits data are handled equally, and... > > .... > > +ret = snd_soc_component_update_bits(dai->component, > > +MT6660_REG_TDM_CFG3, 0x3f0, word_len << 4); > > ... word_len is same for both S32 and S24 formats, so there can be no difference between S24 and S32 format handling in the code. > Meanwhile, the supported formats are: > > > +#define STUB_FORMATS(SNDRV_PCM_FMTBIT_S16_LE | \ > > +SNDRV_PCM_FMTBIT_U16_LE | \ > > +SNDRV_PCM_FMTBIT_S24_LE | \ > > +SNDRV_PCM_FMTBIT_U24_LE | \ > > +SNDRV_PCM_FMTBIT_S32_LE | \ > > +SNDRV_PCM_FMTBIT_U32_LE) > > Are you sure that S24_* formats really work properly? > > Also, the code has no check / setup of the format signedness. > Do unsigned formats (U16, U24, etc) really work as expected, too? > > > Jeff : Yes, it works. So, for the codec, it doesn't matter at all about the signedness and the alingment of 32bit / 24bit of the incoming signals, but magically handled as is? Interesting... thanks, Takashi

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[alsa-devel] [PATCH v6] ASoC: Add MediaTek MT6660 Speaker Amp Driver
by Jeff Chang 14 Jan '20

14 Jan '20

From: Jeff Chang <jeff_chang(a)richtek.com> The MT6660 is a boosted BTL class-D amplifier with V/I sensing. A built-in DC-DC step-up converter is used to provide efficient power for class-D amplifier with multi-level class-G operation. The digital audio interface supports I2S, left-justified, right-justified, TDM and DSP A/B format for audio in with a data out used for chip information like voltage sense and current sense, which are able to be monitored via DATAO through proper Signed-off-by: Jeff Chang <jeff_chang(a)richtek.com> --- sound/soc/codecs/Kconfig | 11 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6660.c | 533 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mt6660.h | 77 +++++++ 4 files changed, 623 insertions(+) create mode 100644 sound/soc/codecs/mt6660.c create mode 100644 sound/soc/codecs/mt6660.h changelogs between v6 & v5 - remove parse DT and apply function about Chip INIT SETTING. - remove DT binding Documentation. changelogs between v5 & v4 - remove redundant initialization. - use packed structures. - remove useless switch case. - take care of memory allocation failures. changelogs between v4 & v3 - remove unnecessary kcontrols. - modify copy right header. - use dev_dbg instead of dev_info. - add necessary debug message. - add DT binding documentation. - add space before } at every table. changelogs between v3 & v2 - modify MT6660 Kconfig, remove unnecessary selection. - remove my own debug io interface. use standard regmap for debugging. - remove regmap volatile ops, we do not use cache. - remove component io read/write function, use snd_soc_component_init_regmap. - remove init setting write code. Using parsing dts to set them. - remove unnecessary pr_info log message. - remove mt6660_component_put_volsw. Using snd_soc_get_volsw. diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 229cc89..f135fbb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_MT6358 if MTK_PMIC_WRAP + select SND_SOC_MT6660 if I2C select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8822 if I2C @@ -1465,6 +1466,16 @@ config SND_SOC_MT6358 Enable support for the platform which uses MT6358 as external codec device. +config SND_SOC_MT6660 + tristate "Mediatek MT6660 Speaker Amplifier" + depends on I2C + help + MediaTek MT6660 is a smart power amplifier which contain + speaker protection, multi-band DRC, equalizer functions. + Select N if you don't have MT6660 on board. + Select M to build this as module. + + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c498373..2b6814c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -119,6 +119,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o +snd-soc-mt6660-objs := mt6660.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -403,6 +404,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o +obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c new file mode 100644 index 0000000..b2523b3 --- /dev/null +++ b/sound/soc/codecs/mt6660.c @@ -0,0 +1,533 @@ +// SPDX-License-Identifier: GPL-2.0 // + +// Copyright (c) 2019 MediaTek Inc. + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/version.h> +#include <linux/err.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/pcm_params.h> +#include <linux/debugfs.h> + +#include "mt6660.h" + +#pragma pack(push, 1) +struct codec_reg_val { + u32 addr; + u32 mask; + u32 data; +}; + +struct reg_size_table { + u32 addr; + u8 size; +}; +#pragma pack(pop) + +static const struct reg_size_table mt6660_reg_size_table[] = { + { MT6660_REG_HPF1_COEF, 4 }, + { MT6660_REG_HPF2_COEF, 4 }, + { MT6660_REG_TDM_CFG3, 2 }, + { MT6660_REG_RESV17, 2 }, + { MT6660_REG_RESV23, 2 }, + { MT6660_REG_SIGMAX, 2 }, + { MT6660_REG_DEVID, 2 }, + { MT6660_REG_HCLIP_CTRL, 2 }, + { MT6660_REG_DA_GAIN, 2 }, +}; + +static int mt6660_get_reg_size(uint32_t addr) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(mt6660_reg_size_table); i++) { + if (mt6660_reg_size_table[i].addr == addr) + return mt6660_reg_size_table[i].size; + } + return 1; +} + +static int mt6660_reg_write(void *context, unsigned int reg, unsigned int val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + u8 reg_data[4]; + int i, ret; + + for (i = 0; i < size; i++) + reg_data[size - i - 1] = (val >> (8 * i)) & 0xff; + + ret = i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); + return ret; +} + +static int mt6660_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + int i, ret; + u8 data[4]; + u32 reg_data = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, reg, size, data); + if (ret < 0) + return ret; + for (i = 0; i < size; i++) { + reg_data <<= 8; + reg_data |= data[i]; + } + *val = reg_data; + return 0; +} + +static struct regmap_config mt6660_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .reg_write = mt6660_reg_write, + .reg_read = mt6660_reg_read, +}; + +static int mt6660_codec_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + + if (event == SND_SOC_DAPM_POST_PMU) + usleep_range(1000, 1100); + return 0; +} + +static int mt6660_codec_classd_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); + int ret; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dev_dbg(component->dev, + "%s: before classd turn on\n", __func__); + /* config to adaptive mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x03); + if (ret < 0) { + dev_err(component->dev, "config mode adaptive fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMU: + /* voltage sensing enable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x04); + if (ret < 0) { + dev_err(component->dev, + "enable voltage sensing fail\n"); + return ret; + } + dev_dbg(component->dev, "Amp on\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + dev_dbg(component->dev, "Amp off\n"); + /* voltage sensing disable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x00); + if (ret < 0) { + dev_err(component->dev, + "disable voltage sensing fail\n"); + return ret; + } + /* pop-noise improvement 1 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x10); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 1 fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, + "%s: after classd turn off\n", __func__); + /* pop-noise improvement 2 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x00); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 2 fail\n"); + return ret; + } + /* config to off mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x00); + if (ret < 0) { + dev_err(component->dev, "config mode off fail\n"); + return ret; + } + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget mt6660_component_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("DAC", NULL, MT6660_REG_PLL_CFG1, + 0, 1, mt6660_codec_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC("VI ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("ClassD", MT6660_REG_SYSTEM_CTRL, 2, 0, + NULL, 0, mt6660_codec_classd_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("OUTP"), + SND_SOC_DAPM_OUTPUT("OUTN"), +}; + +static const struct snd_soc_dapm_route mt6660_component_dapm_routes[] = { + { "DAC", NULL, "aif_playback" }, + { "PGA", NULL, "DAC" }, + { "ClassD", NULL, "PGA" }, + { "OUTP", NULL, "ClassD" }, + { "OUTN", NULL, "ClassD" }, + { "VI ADC", NULL, "ClassD" }, + { "aif_capture", NULL, "VI ADC" }, +}; + +static int mt6660_component_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = -EINVAL; + + if (!strcmp(kcontrol->id.name, "Chip Rev")) { + ucontrol->value.integer.value[0] = chip->chip_rev & 0x0f; + ret = 0; + } + return ret; +} + +static const DECLARE_TLV_DB_SCALE(vol_ctl_tlv, -1155, 5, 0); + +static const struct snd_kcontrol_new mt6660_component_snd_controls[] = { + SOC_SINGLE_TLV("Digital Volume", MT6660_REG_VOL_CTRL, 0, 255, + 1, vol_ctl_tlv), + SOC_SINGLE("Hard Clip Switch", MT6660_REG_HCLIP_CTRL, 8, 1, 0), + SOC_SINGLE("Clip Switch", MT6660_REG_SPS_CTRL, 0, 1, 0), + SOC_SINGLE("Boost Mode", MT6660_REG_BST_CTRL, 0, 3, 0), + SOC_SINGLE("DRE Switch", MT6660_REG_DRE_CTRL, 0, 1, 0), + SOC_SINGLE("DC Protect Switch", MT6660_REG_DC_PROTECT_CTRL, 3, 1, 0), + SOC_SINGLE("Data Output Left Channel Selection", + MT6660_REG_DATAO_SEL, 3, 7, 0), + SOC_SINGLE("Data Output Right Channel Selection", + MT6660_REG_DATAO_SEL, 0, 7, 0), + SOC_SINGLE_EXT("T0 SEL", MT6660_REG_CALI_T0, 0, 7, 0, + snd_soc_get_volsw, NULL), + SOC_SINGLE_EXT("Chip Rev", MT6660_REG_DEVID, 8, 15, 0, + mt6660_component_get_volsw, NULL), +}; + +static int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) +{ + u8 reg_data; + int ret; + + ret = i2c_smbus_read_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL); + if (ret < 0) + return ret; + reg_data = (u8)ret; + if (on_off) + reg_data &= (~0x01); + else + reg_data |= 0x01; + return regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, reg_data); +} + +static int mt6660_component_probe(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "%s\n", __func__); + snd_soc_component_init_regmap(component, chip->regmap); + + return 0; +} + +static void mt6660_component_remove(struct snd_soc_component *component) +{ + dev_dbg(component->dev, "%s\n", __func__); + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver mt6660_component_driver = { + .probe = mt6660_component_probe, + .remove = mt6660_component_remove, + + .controls = mt6660_component_snd_controls, + .num_controls = ARRAY_SIZE(mt6660_component_snd_controls), + .dapm_widgets = mt6660_component_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt6660_component_dapm_widgets), + .dapm_routes = mt6660_component_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt6660_component_dapm_routes), + + .idle_bias_on = false, /* idle_bias_off = true */ +}; + +static int mt6660_component_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) +{ + int word_len = params_physical_width(hw_params); + int aud_bit = params_width(hw_params); + u16 reg_data = 0; + int ret; + + dev_dbg(dai->dev, "%s: ++\n", __func__); + dev_dbg(dai->dev, "format: 0x%08x\n", params_format(hw_params)); + dev_dbg(dai->dev, "rate: 0x%08x\n", params_rate(hw_params)); + dev_dbg(dai->dev, "word_len: %d, aud_bit: %d\n", word_len, aud_bit); + if (word_len > 32 || word_len < 16) { + dev_err(dai->dev, "not supported word length\n"); + return -ENOTSUPP; + } + switch (aud_bit) { + case 16: + reg_data = 3; + break; + case 18: + reg_data = 2; + break; + case 20: + reg_data = 1; + break; + case 24: + case 32: + reg_data = 0; + break; + default: + return -ENOTSUPP; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_SERIAL_CFG1, 0xc0, (reg_data << 6)); + if (ret < 0) { + dev_err(dai->dev, "config aud bit fail\n"); + return ret; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_TDM_CFG3, 0x3f0, word_len << 4); + if (ret < 0) { + dev_err(dai->dev, "config word len fail\n"); + return ret; + } + dev_dbg(dai->dev, "%s: --\n", __func__); + return 0; +} + +static const struct snd_soc_dai_ops mt6660_component_aif_ops = { + .hw_params = mt6660_component_aif_hw_params, +}; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_soc_dai_driver mt6660_codec_dai = { + .name = "mt6660-aif", + .playback = { + .stream_name = "aif_playback", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "aif_capture", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + /* dai properties */ + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, + /* dai operations */ + .ops = &mt6660_component_aif_ops, +}; + +static inline int _mt6660_chip_id_check(struct mt6660_chip *chip) +{ + u8 id[2]; + int ret; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, MT6660_REG_DEVID, 2, id); + if (ret < 0) + return ret; + ret = (id[0] << 8) + id[1]; + ret &= 0x0ff0; + if (ret != 0x00e0 && ret != 0x01e0) { + dev_err(chip->dev, "%s id(%x) not match\n", __func__, ret); + return -ENODEV; + } + return ret; +} + +static inline int _mt6660_chip_sw_reset(struct mt6660_chip *chip) +{ + int ret; + + /* turn on main pll first, then trigger reset */ + ret = regmap_write(chip->regmap, 0x03, 0x00); + if (ret < 0) + return ret; + ret = regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, 0x80); + if (ret < 0) + return ret; + msleep(30); + return 0; +} + +static inline int _mt6660_read_chip_revision(struct mt6660_chip *chip) +{ + u8 reg_data[2]; + int ret; + + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, MT6660_REG_DEVID, 2, reg_data); + if (ret < 0) { + dev_err(chip->dev, "get chip revision fail\n"); + return ret; + } + chip->chip_rev = reg_data[1]; + return 0; +} + +static int mt6660_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct mt6660_chip *chip = NULL; + int ret; + + dev_dbg(&client->dev, "%s\n", __func__); + chip = devm_kzalloc(&client->dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + chip->i2c = client; + chip->dev = &client->dev; + mutex_init(&chip->io_lock); + i2c_set_clientdata(client, chip); + + chip->regmap = devm_regmap_init(&client->dev, + NULL, chip, &mt6660_regmap_config); + if (IS_ERR(chip->regmap)) { + ret = PTR_ERR(chip->regmap); + dev_err(&client->dev, "failed to initialise regmap: %d\n", ret); + return ret; + } + + /* chip reset first */ + ret = _mt6660_chip_sw_reset(chip); + if (ret < 0) { + dev_err(chip->dev, "chip reset fail\n"); + goto probe_fail; + } + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 2 fail\n"); + goto probe_fail; + } + /* chip devid check */ + ret = _mt6660_chip_id_check(chip); + if (ret < 0) { + dev_err(chip->dev, "chip id check fail\n"); + goto probe_fail; + } + /* chip revision get */ + ret = _mt6660_read_chip_revision(chip); + if (ret < 0) { + dev_err(chip->dev, "read chip revision fail\n"); + goto probe_fail; + } + pm_runtime_set_active(chip->dev); + pm_runtime_enable(chip->dev); + + ret = devm_snd_soc_register_component(chip->dev, + &mt6660_component_driver, + &mt6660_codec_dai, 1); + return ret; +probe_fail: + _mt6660_chip_power_on(chip, 0); + mutex_destroy(&chip->io_lock); + return ret; +} + +static int mt6660_i2c_remove(struct i2c_client *client) +{ + struct mt6660_chip *chip = i2c_get_clientdata(client); + + pm_runtime_disable(chip->dev); + pm_runtime_set_suspended(chip->dev); + mutex_destroy(&chip->io_lock); + return 0; +} + +static int __maybe_unused mt6660_i2c_runtime_suspend(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "enter low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x01); +} + +static int __maybe_unused mt6660_i2c_runtime_resume(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "exit low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x00); +} + +static const struct dev_pm_ops mt6660_dev_pm_ops = { + SET_RUNTIME_PM_OPS(mt6660_i2c_runtime_suspend, + mt6660_i2c_runtime_resume, NULL) +}; + +static const struct of_device_id __maybe_unused mt6660_of_id[] = { + { .compatible = "mediatek,mt6660",}, + {}, +}; +MODULE_DEVICE_TABLE(of, mt6660_of_id); + +static const struct i2c_device_id mt6660_i2c_id[] = { + {"mt6660", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); + +static struct i2c_driver mt6660_i2c_driver = { + .driver = { + .name = "mt6660", + .of_match_table = of_match_ptr(mt6660_of_id), + .pm = &mt6660_dev_pm_ops, + }, + .probe = mt6660_i2c_probe, + .remove = mt6660_i2c_remove, + .id_table = mt6660_i2c_id, +}; +module_i2c_driver(mt6660_i2c_driver); + +MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com>"); +MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); +MODULE_LICENSE("GPL"); +MODULE_VERSION("1.0.7_G"); diff --git a/sound/soc/codecs/mt6660.h b/sound/soc/codecs/mt6660.h new file mode 100644 index 0000000..054a3c5 --- /dev/null +++ b/sound/soc/codecs/mt6660.h @@ -0,0 +1,77 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#ifndef __SND_SOC_MT6660_H +#define __SND_SOC_MT6660_H + +#include <linux/mutex.h> +#include <linux/regmap.h> + +#pragma pack(push, 1) +struct mt6660_platform_data { + u8 init_setting_num; + u32 *init_setting_addr; + u32 *init_setting_mask; + u32 *init_setting_val; +}; + +struct mt6660_chip { + struct i2c_client *i2c; + struct device *dev; + struct platform_device *param_dev; + struct mt6660_platform_data plat_data; + struct mutex io_lock; + struct regmap *regmap; + u16 chip_rev; +}; +#pragma pack(pop) + +#define MT6660_REG_DEVID (0x00) +#define MT6660_REG_SYSTEM_CTRL (0x03) +#define MT6660_REG_IRQ_STATUS1 (0x05) +#define MT6660_REG_ADDA_CLOCK (0x07) +#define MT6660_REG_SERIAL_CFG1 (0x10) +#define MT6660_REG_DATAO_SEL (0x12) +#define MT6660_REG_TDM_CFG3 (0x15) +#define MT6660_REG_HPF_CTRL (0x18) +#define MT6660_REG_HPF1_COEF (0x1A) +#define MT6660_REG_HPF2_COEF (0x1B) +#define MT6660_REG_PATH_BYPASS (0x1E) +#define MT6660_REG_WDT_CTRL (0x20) +#define MT6660_REG_HCLIP_CTRL (0x24) +#define MT6660_REG_VOL_CTRL (0x29) +#define MT6660_REG_SPS_CTRL (0x30) +#define MT6660_REG_SIGMAX (0x33) +#define MT6660_REG_CALI_T0 (0x3F) +#define MT6660_REG_BST_CTRL (0x40) +#define MT6660_REG_PROTECTION_CFG (0x46) +#define MT6660_REG_DA_GAIN (0x4c) +#define MT6660_REG_AUDIO_IN2_SEL (0x50) +#define MT6660_REG_SIG_GAIN (0x51) +#define MT6660_REG_PLL_CFG1 (0x60) +#define MT6660_REG_DRE_CTRL (0x68) +#define MT6660_REG_DRE_THDMODE (0x69) +#define MT6660_REG_DRE_CORASE (0x6B) +#define MT6660_REG_PWM_CTRL (0x70) +#define MT6660_REG_DC_PROTECT_CTRL (0x74) +#define MT6660_REG_ADC_USB_MODE (0x7c) +#define MT6660_REG_INTERNAL_CFG (0x88) +#define MT6660_REG_RESV0 (0x98) +#define MT6660_REG_RESV1 (0x99) +#define MT6660_REG_RESV2 (0x9A) +#define MT6660_REG_RESV3 (0x9B) +#define MT6660_REG_RESV6 (0xA2) +#define MT6660_REG_RESV7 (0xA3) +#define MT6660_REG_RESV10 (0xB0) +#define MT6660_REG_RESV11 (0xB1) +#define MT6660_REG_RESV16 (0xB6) +#define MT6660_REG_RESV17 (0xB7) +#define MT6660_REG_RESV19 (0xB9) +#define MT6660_REG_RESV21 (0xBB) +#define MT6660_REG_RESV23 (0xBD) +#define MT6660_REG_RESV31 (0xD3) +#define MT6660_REG_RESV40 (0xE0) + +#endif /* __SND_SOC_MT6660_H */ -- 2.7.4

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[alsa-devel] [PATCH v11 0/2] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 14 Jan '20

14 Jan '20

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 and Lenovo YOGA C630 Laptop based on SDM850 with WSA8815 speaker amplifiers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v10: - Add check in hw_free for prepare status Srinivas Kandagatla (2): dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 68 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1185 +++++++++++++++++ 4 files changed, 1265 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

3 6

[alsa-devel] [PATCH] ALSA: usb-audio: fix sync-ep altsetting sanity check
by Johan Hovold 14 Jan '20

14 Jan '20

The altsetting sanity check in set_sync_ep_implicit_fb_quirk() was checking for there to be at least one altsetting but then went on to access the second one, which may not exist. This could lead to random slab data being used to initialise the sync endpoint in snd_usb_add_endpoint(). Fixes: c75a8a7ae565 ("ALSA: snd-usb: add support for implicit feedback") Fixes: ca10a7ebdff1 ("ALSA: usb-audio: FT C400 sync playback EP to capture EP") Fixes: 5e35dc0338d8 ("ALSA: usb-audio: add implicit fb quirk for Behringer UFX1204") Fixes: 17f08b0d9aaf ("ALSA: usb-audio: add implicit fb quirk for Axe-Fx II") Fixes: 103e9625647a ("ALSA: usb-audio: simplify set_sync_ep_implicit_fb_quirk") Cc: stable <stable(a)vger.kernel.org> # 3.5 Signed-off-by: Johan Hovold <johan(a)kernel.org> --- sound/usb/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a11c8150af58..0e4eab96e23e 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -370,7 +370,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, add_sync_ep_from_ifnum: iface = usb_ifnum_to_if(dev, ifnum); - if (!iface || iface->num_altsetting == 0) + if (!iface || iface->num_altsetting < 2) return -EINVAL; alts = &iface->altsetting[1]; -- 2.24.1

2 1

[alsa-devel] [PATCH 0/2] ALSA: HDA: remove warnings with make W=1
by Pierre-Louis Bossart 14 Jan '20

14 Jan '20

Last series to make the sound subsystem report next to zero warnings (at least for the parts used by Intel). Pierre-Louis Bossart (2): ALSA: HDA: correct kernel-doc parameter descriptions ALSA: HDA: patch_hdmi: remove warnings with empty body sound/pci/hda/hda_codec.c | 1 + sound/pci/hda/hda_generic.c | 5 +++++ sound/pci/hda/hda_jack.c | 2 ++ sound/pci/hda/patch_ca0132.c | 1 + sound/pci/hda/patch_hdmi.c | 6 ++++-- 5 files changed, 13 insertions(+), 2 deletions(-) -- 2.20.1

2 4

[alsa-devel] [PATCH] ALSA: hda: Update kernel-doc function parameter descriptions
by Pierre-Louis Bossart 14 Jan '20

14 Jan '20

From: Keyon Jie <yang.jie(a)linux.intel.com> Make W=1 throws a lot of warnings, with multiple misalignments between function params and their descriptions. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/ext/hdac_ext_bus.c | 11 ++++++----- sound/hda/ext/hdac_ext_controller.c | 14 +++++++------- sound/hda/ext/hdac_ext_stream.c | 1 - sound/hda/hdac_bus.c | 3 +++ sound/hda/hdac_component.c | 1 + sound/hda/hdac_controller.c | 2 +- sound/hda/hdac_device.c | 12 ++++++++---- sound/hda/hdac_stream.c | 2 ++ 8 files changed, 28 insertions(+), 18 deletions(-) diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 242306d820ec..73bfa71845f6 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -19,10 +19,10 @@ MODULE_LICENSE("GPL v2"); /** * snd_hdac_ext_bus_init - initialize a HD-audio extended bus - * @ebus: the pointer to extended bus object + * @bus: the pointer to HDAC bus object * @dev: device pointer * @ops: bus verb operators - * default ops + * @ext_ops: operators used for ASoC HDA codec drivers * * Returns 0 if successful, or a negative error code. */ @@ -51,7 +51,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init); /** * snd_hdac_ext_bus_exit - clean up a HD-audio extended bus - * @ebus: the pointer to extended bus object + * @bus: the pointer to HDAC bus object */ void snd_hdac_ext_bus_exit(struct hdac_bus *bus) { @@ -67,8 +67,9 @@ static void default_release(struct device *dev) /** * snd_hdac_ext_bus_device_init - initialize the HDA extended codec base device - * @ebus: hdac extended bus to attach to + * @bus: hdac bus to attach to * @addr: codec address + * @hdev: hdac device to init * * Returns zero for success or a negative error code. */ @@ -114,7 +115,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_exit); /** * snd_hdac_ext_bus_device_remove - remove HD-audio extended codec base devices * - * @ebus: HD-audio extended bus + * @bus: the pointer to HDAC bus object */ void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus) { diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index cfab60d88c92..a684f0520b4b 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -28,7 +28,7 @@ /** * snd_hdac_ext_bus_ppcap_enable - enable/disable processing pipe capability - * @ebus: HD-audio extended core bus + * @bus: the pointer to HDAC bus object * @enable: flag to turn on/off the capability */ void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *bus, bool enable) @@ -50,7 +50,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable); /** * snd_hdac_ext_bus_ppcap_int_enable - ppcap interrupt enable/disable - * @ebus: HD-audio extended core bus + * @bus: the pointer to HDAC bus object * @enable: flag to enable/disable interrupt */ void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *bus, bool enable) @@ -77,7 +77,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable); /** * snd_hdac_ext_bus_get_ml_capabilities - get multilink capability - * @ebus: HD-audio extended core bus + * @bus: the pointer to HDAC bus object * * This will parse all links and read the mlink capabilities and add them * in hlink_list of extended hdac bus @@ -117,7 +117,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_get_ml_capabilities); /** * snd_hdac_link_free_all- free hdac extended link objects * - * @ebus: HD-audio ext core bus + * @bus: the pointer to HDAC bus object */ void snd_hdac_link_free_all(struct hdac_bus *bus) @@ -134,7 +134,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_link_free_all); /** * snd_hdac_ext_bus_get_link_index - get link based on codec name - * @ebus: HD-audio extended core bus + * @bus: the pointer to HDAC bus object * @codec_name: codec name */ struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus, @@ -211,7 +211,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down); /** * snd_hdac_ext_bus_link_power_up_all -power up all hda link - * @ebus: HD-audio extended bus + * @bus: the pointer to HDAC bus object */ int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus) { @@ -232,7 +232,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_up_all); /** * snd_hdac_ext_bus_link_power_down_all -power down all hda link - * @ebus: HD-audio extended bus + * @bus: the pointer to HDAC bus object */ int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus) { diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 6b1b4b834bae..c4d54a838773 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -530,7 +530,6 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_set_dpibr); /** * snd_hdac_ext_stream_set_lpib - sets the lpib value of a stream - * @bus: HD-audio core bus * @stream: hdac_ext_stream * @value: lpib value to set */ diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 48b227fff204..3fe62be1cbcc 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -22,6 +22,7 @@ static const struct hdac_bus_ops default_ops = { /** * snd_hdac_bus_init - initialize a HD-audio bas bus * @bus: the pointer to bus object + * @dev: device pointer * @ops: bus verb operators * * Returns 0 if successful, or a negative error code. @@ -64,6 +65,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_bus_exit); /** * snd_hdac_bus_exec_verb - execute a HD-audio verb on the given bus * @bus: bus object + * @addr: the HDAC device address * @cmd: HD-audio encoded verb * @res: pointer to store the response, NULL if performing asynchronously * @@ -84,6 +86,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_bus_exec_verb); /** * snd_hdac_bus_exec_verb_unlocked - unlocked version * @bus: bus object + * @addr: the HDAC device address * @cmd: HD-audio encoded verb * @res: pointer to store the response, NULL if performing asynchronously * diff --git a/sound/hda/hdac_component.c b/sound/hda/hdac_component.c index dfe7e755f594..89126c6fd216 100644 --- a/sound/hda/hdac_component.c +++ b/sound/hda/hdac_component.c @@ -262,6 +262,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_acomp_register_notifier); /** * snd_hdac_acomp_init - Initialize audio component * @bus: HDA core bus + * @aops: audio component ops * @match_master: match function for finding components * @extra_size: Extra bytes to allocate * diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index b856184af37f..bc4a8b606020 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -561,7 +561,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_bus_stop_chip); * snd_hdac_bus_handle_stream_irq - interrupt handler for streams * @bus: HD-audio core bus * @status: INTSTS register value - * @ask: callback to be called for woken streams + * @ack: callback to be called for woken streams * * Returns the bits of handled streams, or zero if no stream is handled. */ diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 7bcdb0e454f2..9a526aeef8da 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -135,7 +135,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_exit); /** * snd_hdac_device_register - register the hd-audio codec base device - * codec: the device to register + * @codec: the device to register */ int snd_hdac_device_register(struct hdac_device *codec) { @@ -158,7 +158,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_register); /** * snd_hdac_device_unregister - unregister the hd-audio codec base device - * codec: the device to unregister + * @codec: the device to unregister */ void snd_hdac_device_unregister(struct hdac_device *codec) { @@ -281,6 +281,10 @@ EXPORT_SYMBOL_GPL(snd_hdac_read); /** * _snd_hdac_read_parm - read a parmeter + * @codec: the codec object + * @nid: NID to read a parameter + * @parm: parameter to read + * @res: pointer to store the read value * * This function returns zero or an error unlike snd_hdac_read_parm(). */ @@ -1062,9 +1066,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_check_power_state); * snd_hdac_sync_power_state - wait until actual power state matches * with the target state * - * @hdac: the HDAC device + * @codec: the HDAC device * @nid: NID to send the command - * @target_state: target state to check for + * @power_state: target power state to wait for * * Return power state or PS_ERROR if codec rejects GET verb. */ diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index 682ed39f79b0..d01e69139164 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -590,7 +590,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_stream_timecounter_init); /** * snd_hdac_stream_sync_trigger - turn on/off stream sync register * @azx_dev: HD-audio core stream (master stream) + * @set: true = set, false = clear * @streams: bit flags of streams to sync + * @reg: the stream sync register address */ void snd_hdac_stream_sync_trigger(struct hdac_stream *azx_dev, bool set, unsigned int streams, unsigned int reg) -- 2.20.1

2 1

[alsa-devel] [PATCH] soundwire: intel: fix factor of two in MCLK handling
by Pierre-Louis Bossart 14 Jan '20

14 Jan '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> Somehow Intel folks were confused, the property is 2x what the mclk frequency actually is (checked the actual bus frequency with a scope) Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 0371d3d5501a..06ef3a3ac080 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -880,6 +880,9 @@ static int sdw_master_read_intel_prop(struct sdw_bus *bus) "intel-sdw-ip-clock", &prop->mclk_freq); + /* the values reported by BIOS are the 2x clock, not the bus clock */ + prop->mclk_freq /= 2; + fwnode_property_read_u32(link, "intel-quirk-mask", &quirk_mask); -- 2.20.1

2 1

[alsa-devel] [PATCH 0/2] soundwire: cadence: remove warnings
by Pierre-Louis Bossart 14 Jan '20

14 Jan '20

Fix warnings reported by cppcheck and make W=1 Pierre-Louis Bossart (2): soundwire: cadence: update kernel-doc parameter descriptions soundwire: cadence: remove useless variable incrementation drivers/soundwire/cadence_master.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) -- 2.20.1

2 3

[alsa-devel] [PATCH v6 0/2] soundwire: Add support to Qualcomm SoundWire master
by Srinivas Kandagatla 14 Jan '20

14 Jan '20

Thanks for reviewing the v5 patchset. Here is new patchset addressing all the comments from v5 This patchset adds support for Qualcomm SoundWire Master Controller found in most of Qualcomm SoCs and WCD audio codecs. This driver along with WCD934x codec and WSA881x Class-D Smart Speaker Amplifier drivers is tested on on DragonBoard DB845c based of SDM845 SoC and Lenovo YOGA C630 Laptop based on SDM850. SoundWire controller on SDM845 is integrated in WCD934x audio codec via SlimBus interface. Currently this driver is very minimal and only supports PDM. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Test and add PCM support. Thanks, srini Changes since v5: -No code changes. -Add extra comment in interrupt handler as suggested by Pierre Srinivas Kandagatla (2): dt-bindings: soundwire: add bindings for Qcom controller soundwire: qcom: add support for SoundWire controller .../bindings/soundwire/qcom,sdw.txt | 167 ++++ drivers/soundwire/Kconfig | 9 + drivers/soundwire/Makefile | 4 + drivers/soundwire/qcom.c | 861 ++++++++++++++++++ 4 files changed, 1041 insertions(+) create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 drivers/soundwire/qcom.c -- 2.21.0

2 3

[alsa-devel] [PATCH 0/6] soundwire: cadence: better logs and error handling
by Pierre-Louis Bossart 14 Jan '20

14 Jan '20

This is a stand-alone set of patches to improve error handling and provide better information to the platform integrator. As suggested by Vinod, these patches are shared first - with the risk that they are separated from the actual steps where they are needed. For reference, the complete set of 100+ patches required for SoundWire on Intel platforms is available here: https://github.com/thesofproject/linux/pull/1692 Pierre-Louis Bossart (4): soundwire: cadence_master: filter out bad interrupts soundwire: cadence_master: log more useful information during timeouts soundwire: cadence_master: handle multiple status reports per Slave soundwire: bus: check first if Slaves become UNATTACHED Rander Wang (2): soundwire: cadence_master: clear interrupt status before enabling interrupt soundwire: cadence_master: remove config update for interrupt setting drivers/soundwire/bus.c | 18 +++++++++ drivers/soundwire/cadence_master.c | 60 +++++++++++++++++++++++++----- 2 files changed, 68 insertions(+), 10 deletions(-) base-commit: b637124800a157c4df3699d1137d8533394f7678 -- 2.20.1

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[alsa-devel] MIDI 2.0 universal packet protocol
by GitHub issues - edited 14 Jan '20

14 Jan '20

alsa-project/alsa-lib issue #24 was edited from symdeb: Since MIDI 2.0 only support USB and not over DIN (for now), will MIDI 2.0 need a new or updated OS class driver for the 32/64 bit universal packet interface and device descriptor for devices ? Page 16 of the USB MIDI class driver spec defines support for 1,2 or 3 bytes transfers only, How will this have to be implemented by MIDI USB device manufacturers and well as on the Linux OSes side ? How will this impact Linux ALSA ? Would the first byte of the message with b7=0 not confuse the driver of running status ? Link universal packet format https://www.midi.org/articles-old/details-about-midi-2-0-midi-ci-profiles-a…) Link first MIDI 2.0 Ready product: Roland A-88MKII https://www.midi.org/articles-old/roland-announces Issue URL : https://github.com/alsa-project/alsa-lib/issues/24 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] MIDI 2.0 universal packet protocol
by GitHub issues - edited 14 Jan '20

14 Jan '20

alsa-project/alsa-lib issue #24 was edited from symdeb: Since MIDI 2.0 only support USB and not over DIN (for now), will MIDI 2.0 need a new or updated OS class driver for the 32/64 bit universal packet interface and device descriptor for devices ? Page 16 of the USB MIDI class driver spec defines support for 1,2 or 3 bytes transfers only, How will this have to be implemented by MIDI USB device manufacturers and well as on the Linux OSes side ? How will this impact Linux ALSA ? Would the first byte of the message with b7=0 not confuse the driver of running status ? Reference for packet format : [https://www.midi.org/articles-old/details-about-midi-2-0-midi-ci-profiles-a… Issue URL : https://github.com/alsa-project/alsa-lib/issues/24 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] MIDI 2.0 universal packet protocol
by GitHub issues - opened 14 Jan '20

14 Jan '20

alsa-project/alsa-lib issue #24 was opened from symdeb: Since MIDI 2.0 only support USB and non MIDI DIN (for now), will MIDI 2.0 need a new or updated OS class driver for the 32/64 bit universal packet interface and device descriptor for devices ? Page 16 of the USB MIDI class driver spec defines support for 1,2 or 3 bytes transfers only, How will this have to be implemented by MIDI USB device manufacturers and well as on the Linux OSes side ? How will this impact Linux ALSA ? Would the first byte of the message with b7=0 not confuse the driver of running status ? Reference for packet format : [https://www.midi.org/articles-old/details-about-midi-2-0-midi-ci-profiles-a… Issue URL : https://github.com/alsa-project/alsa-lib/issues/24 Repository URL: https://github.com/alsa-project/alsa-lib

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Re: [alsa-devel] Github MOTU PCI 324; Motu 2408; Motu 1224
by Takashi Sakamoto 14 Jan '20

14 Jan '20

Hi Yan, (CCed to alsa-devel) On Tue, Jan 14, 2020, at 02:07, Yannic Korten wrote: > Hey Mr. Takashi, > I'm new to Linux and new to Github. If you think this is spam or you > wanted to be in private: See this mail as spam - ignore and just delete > it. I'm sorry then. > But I have a problem and don't know whom to contact. It seems like you > have much experience with coding and the usage of ALSA (which seems to > be your project). > So, my setup is the following: > Ubuntu Studio 19.10 > MOTU 324 PCI Card (detected as 'Multimedia audio controller: Mark of > the Unicorn Inc PCI-324 Audiowire Interface (rev 0a)') > MOTU 1224 > MOTU 2408 > So I tried to install your files from github but it seems not to be > working. Either it's because it's just the support for some newer MOTU > devices or I'm to dumb. > Is it possible to get the above devices fully running on linux with > your "drivers"? Or are they not supporting? > Will it be possible to insert them to the code (easily)? Or is it way > to much expenditure for that old devices? > > Thanks for help > Greetings from Germany > Yan I'm an author/maintainer of ALSA firewire stack which includes snd-firewire-motu driver. The driver can handle MOTU's FireWire series for IEEE 1394 bus; e.g. MOTU 828mkII. Unfortunately, the driver is unavailable for your model. Your model is not compliant to IEEE 1394. It's for 'AudioWire', which is proprietary protocol developed by MOTU. Like 1394 OHCI card is for IEEE 1394 bus, PCI-324/424 ASIC card is for 'AudioWire'. The ASIC is designed by MOTU itself (Precisely S&S Research, Inc.) and the detail is closed. Although PCI-324/424 card has physical ports into which IEEE 1394 cables can connect, and actually AudioWire uses the same cable of IEEE 1394a, signals in the cable is completely different from IEEE 1394. In the summer of 2015, I bought AudioWire-based model (it's really cheep in used market) and attempted to connect it to my IEEE 1394 bus analyzer. As I expected, nothing can be detected. I guess that PCI-324/424 ASIC integrates a sort of computing core (e.g. ARM7TDMI) with PCI interface for AudioWire protocol. For this kind of device, people need to work for reverse-engineering by analyze MOTU driver's behaviour in memory space mapped to host system, and write new driver to emulate the behaviour. However, this is a kind of 'adventure' and no guarantee to achieve. On the other hand, it certainly consumes the batch of developer's resources, unless getting enough assists by vendors. This is a reason that you cannot see AudioWire compliant drivers as Open Source Software. Regards Takashi Sakamoto

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Re: [alsa-devel] [PATCH next] ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m
by Codrin.Ciubotariu＠microchip.com 13 Jan '20

13 Jan '20

On 13.01.2020 15:32, Chen Zhou wrote: > If CONFIG_SND_ATMEL_SOC_DMA=m, build error: > > sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': > (.text+0x7cd): undefined reference to `atmel_pcm_dma_platform_register' > > Function atmel_pcm_dma_platform_register is defined under > CONFIG SND_ATMEL_SOC_DMA, so select SND_ATMEL_SOC_DMA in > CONFIG SND_ATMEL_SOC_SSC, same to CONFIG_SND_ATMEL_SOC_PDC. Arnd sent a more complete patch for this issue: https://lkml.org/lkml/2019/10/1/620 It wasn't merged due to some integration issues. Could you please try it and let us know if it fixes your issue? Thanks and best regards, Codrin > > Reported-by: Hulk Robot <hulkci(a)huawei.com> > Signed-off-by: Chen Zhou <chenzhou10(a)huawei.com> > --- > sound/soc/atmel/Kconfig | 2 ++ > 1 file changed, 2 insertions(+) > > diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig > index f118c22..d1dc8e6 100644 > --- a/sound/soc/atmel/Kconfig > +++ b/sound/soc/atmel/Kconfig > @@ -19,6 +19,8 @@ config SND_ATMEL_SOC_DMA > > config SND_ATMEL_SOC_SSC > tristate > + select SND_ATMEL_SOC_DMA > + select SND_ATMEL_SOC_PDC > > config SND_ATMEL_SOC_SSC_PDC > tristate "SoC PCM DAI support for AT91 SSC controller using PDC" > -- > 2.7.4 >

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[alsa-devel] Applied "ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m" to the asoc tree
by Mark Brown 13 Jan '20

13 Jan '20

The patch ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8fea78029f5e6ed734ae1957bef23cfda1af4354 Mon Sep 17 00:00:00 2001 From: Chen Zhou <chenzhou10(a)huawei.com> Date: Mon, 13 Jan 2020 21:32:42 +0800 Subject: [PATCH] ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m If CONFIG_SND_ATMEL_SOC_DMA=m, build error: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': (.text+0x7cd): undefined reference to `atmel_pcm_dma_platform_register' Function atmel_pcm_dma_platform_register is defined under CONFIG SND_ATMEL_SOC_DMA, so select SND_ATMEL_SOC_DMA in CONFIG SND_ATMEL_SOC_SSC, same to CONFIG_SND_ATMEL_SOC_PDC. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Chen Zhou <chenzhou10(a)huawei.com> Link: https://lore.kernel.org/r/20200113133242.144550-1-chenzhou10@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/atmel/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index f118c229ed82..d1dc8e6366dc 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -19,6 +19,8 @@ config SND_ATMEL_SOC_DMA config SND_ATMEL_SOC_SSC tristate + select SND_ATMEL_SOC_DMA + select SND_ATMEL_SOC_PDC config SND_ATMEL_SOC_SSC_PDC tristate "SoC PCM DAI support for AT91 SSC controller using PDC" -- 2.20.1

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[alsa-devel] Applied "ASoC: rt1308-sdw: add rt1308 SdW amplifier driver" to the asoc tree
by Mark Brown 13 Jan '20

13 Jan '20

The patch ASoC: rt1308-sdw: add rt1308 SdW amplifier driver has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a87a6653a28c0d511b3fdf3d62302bed5f325e03 Mon Sep 17 00:00:00 2001 From: Shuming Fan <shumingf(a)realtek.com> Date: Fri, 10 Jan 2020 09:46:06 +0800 Subject: [PATCH] ASoC: rt1308-sdw: add rt1308 SdW amplifier driver This is the initial amplifier driver for rt1308-sdw. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20200110014606.17333-1-shumingf@realtek.com Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1308-sdw.c | 736 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1308-sdw.h | 169 ++++++++ 4 files changed, 913 insertions(+) create mode 100644 sound/soc/codecs/rt1308-sdw.c create mode 100644 sound/soc/codecs/rt1308-sdw.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cdfd912d5f8f..2f52e4cac96a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -168,6 +168,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT700_SDW if SOUNDWIRE select SND_SOC_RT711_SDW if SOUNDWIRE select SND_SOC_RT715_SDW if SOUNDWIRE + select SND_SOC_RT1308_SDW if SOUNDWIRE select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIMPLE_AMPLIFIER @@ -1005,6 +1006,11 @@ config SND_SOC_RT1305 config SND_SOC_RT1308 tristate +config SND_SOC_RT1308_SDW + tristate "Realtek RT1308 Codec - SDW" + depends on SOUNDWIRE + select REGMAP_SOUNDWIRE + config SND_SOC_RT5514 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ce285b33a806..66aa452f5aa2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -154,6 +154,7 @@ snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o snd-soc-rt1305-objs := rt1305.o snd-soc-rt1308-objs := rt1308.o +snd-soc-rt1308-sdw-objs := rt1308-sdw.o snd-soc-rt274-objs := rt274.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o @@ -448,6 +449,7 @@ obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o obj-$(CONFIG_SND_SOC_RT1308) += snd-soc-rt1308.o +obj-$(CONFIG_SND_SOC_RT1308_SDW) += snd-soc-rt1308-sdw.o obj-$(CONFIG_SND_SOC_RT274) += snd-soc-rt274.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c new file mode 100644 index 000000000000..8a03dbfe7906 --- /dev/null +++ b/sound/soc/codecs/rt1308-sdw.c @@ -0,0 +1,736 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rt1308-sdw.c -- rt1308 ALSA SoC audio driver +// +// Copyright(c) 2019 Realtek Semiconductor Corp. +// +// +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/pm_runtime.h> +#include <linux/mod_devicetable.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "rt1308.h" +#include "rt1308-sdw.h" + +static bool rt1308_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x00e0: + case 0x00f0: + case 0x2f01 ... 0x2f07: + case 0x3000 ... 0x3001: + case 0x3004 ... 0x3005: + case 0x3008: + case 0x300a: + case 0xc000 ... 0xcff3: + return true; + default: + return false; + } +} + +static bool rt1308_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2f01 ... 0x2f07: + case 0x3000 ... 0x3001: + case 0x3004 ... 0x3005: + case 0x3008: + case 0x300a: + case 0xc000: + return true; + default: + return false; + } +} + +static const struct regmap_config rt1308_sdw_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt1308_readable_register, + .volatile_reg = rt1308_volatile_register, + .max_register = 0xcfff, + .reg_defaults = rt1308_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rt1308_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +/* Bus clock frequency */ +#define RT1308_CLK_FREQ_9600000HZ 9600000 +#define RT1308_CLK_FREQ_12000000HZ 12000000 +#define RT1308_CLK_FREQ_6000000HZ 6000000 +#define RT1308_CLK_FREQ_4800000HZ 4800000 +#define RT1308_CLK_FREQ_2400000HZ 2400000 +#define RT1308_CLK_FREQ_12288000HZ 12288000 + +static int rt1308_clock_config(struct device *dev) +{ + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev); + unsigned int clk_freq, value; + + clk_freq = (rt1308->params.curr_dr_freq >> 1); + + switch (clk_freq) { + case RT1308_CLK_FREQ_12000000HZ: + value = 0x0; + break; + case RT1308_CLK_FREQ_6000000HZ: + value = 0x1; + break; + case RT1308_CLK_FREQ_9600000HZ: + value = 0x2; + break; + case RT1308_CLK_FREQ_4800000HZ: + value = 0x3; + break; + case RT1308_CLK_FREQ_2400000HZ: + value = 0x4; + break; + case RT1308_CLK_FREQ_12288000HZ: + value = 0x5; + break; + default: + return -EINVAL; + } + + regmap_write(rt1308->regmap, 0xe0, value); + regmap_write(rt1308->regmap, 0xf0, value); + + dev_dbg(dev, "%s complete, clk_freq=%d\n", __func__, clk_freq); + + return 0; +} + +static int rt1308_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval, i, num_of_ports = 1; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x00; /* BITMAP: 00010100 (not enable yet) */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + /* for sink */ + nval = hweight32(prop->sink_ports); + num_of_ports += nval; + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), + GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* Allocate port_ready based on num_of_ports */ + slave->port_ready = devm_kcalloc(&slave->dev, num_of_ports, + sizeof(*slave->port_ready), + GFP_KERNEL); + if (!slave->port_ready) + return -ENOMEM; + + /* Initialize completion */ + for (i = 0; i < num_of_ports; i++) + init_completion(&slave->port_ready[i]); + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + dev_dbg(&slave->dev, "%s\n", __func__); + + return 0; +} + +static int rt1308_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev); + int ret = 0; + unsigned int efuse_m_btl_l, efuse_m_btl_r, tmp; + unsigned int efuse_c_btl_l, efuse_c_btl_r; + + if (rt1308->hw_init) + return 0; + + ret = rt1308_read_prop(slave); + if (ret < 0) + goto _io_init_err_; + + if (rt1308->first_hw_init) { + regcache_cache_only(rt1308->regmap, false); + regcache_cache_bypass(rt1308->regmap, true); + } + + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + if (!rt1308->first_hw_init) { + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + /* sw reset */ + regmap_write(rt1308->regmap, RT1308_SDW_RESET, 0); + + /* read efuse */ + regmap_write(rt1308->regmap, 0xc360, 0x01); + regmap_write(rt1308->regmap, 0xc361, 0x80); + regmap_write(rt1308->regmap, 0xc7f0, 0x04); + regmap_write(rt1308->regmap, 0xc7f1, 0xfe); + msleep(100); + regmap_write(rt1308->regmap, 0xc7f0, 0x44); + msleep(20); + regmap_write(rt1308->regmap, 0xc240, 0x10); + + regmap_read(rt1308->regmap, 0xc861, &tmp); + efuse_m_btl_l = tmp; + regmap_read(rt1308->regmap, 0xc860, &tmp); + efuse_m_btl_l = efuse_m_btl_l | (tmp << 8); + regmap_read(rt1308->regmap, 0xc863, &tmp); + efuse_c_btl_l = tmp; + regmap_read(rt1308->regmap, 0xc862, &tmp); + efuse_c_btl_l = efuse_c_btl_l | (tmp << 8); + regmap_read(rt1308->regmap, 0xc871, &tmp); + efuse_m_btl_r = tmp; + regmap_read(rt1308->regmap, 0xc870, &tmp); + efuse_m_btl_r = efuse_m_btl_r | (tmp << 8); + regmap_read(rt1308->regmap, 0xc873, &tmp); + efuse_c_btl_r = tmp; + regmap_read(rt1308->regmap, 0xc872, &tmp); + efuse_c_btl_r = efuse_c_btl_r | (tmp << 8); + dev_info(&slave->dev, "%s m_btl_l=0x%x, m_btl_r=0x%x\n", __func__, + efuse_m_btl_l, efuse_m_btl_r); + dev_info(&slave->dev, "%s c_btl_l=0x%x, c_btl_r=0x%x\n", __func__, + efuse_c_btl_l, efuse_c_btl_r); + + /* initial settings */ + regmap_write(rt1308->regmap, 0xc103, 0xc0); + regmap_write(rt1308->regmap, 0xc030, 0x17); + regmap_write(rt1308->regmap, 0xc031, 0x81); + regmap_write(rt1308->regmap, 0xc032, 0x26); + regmap_write(rt1308->regmap, 0xc040, 0x80); + regmap_write(rt1308->regmap, 0xc041, 0x80); + regmap_write(rt1308->regmap, 0xc042, 0x06); + regmap_write(rt1308->regmap, 0xc052, 0x0a); + regmap_write(rt1308->regmap, 0xc080, 0x0a); + regmap_write(rt1308->regmap, 0xc060, 0x02); + regmap_write(rt1308->regmap, 0xc061, 0x75); + regmap_write(rt1308->regmap, 0xc062, 0x05); + regmap_write(rt1308->regmap, 0xc171, 0x07); + regmap_write(rt1308->regmap, 0xc173, 0x0d); + regmap_write(rt1308->regmap, 0xc311, 0x7f); + regmap_write(rt1308->regmap, 0xc900, 0x90); + regmap_write(rt1308->regmap, 0xc1a0, 0x84); + regmap_write(rt1308->regmap, 0xc1a1, 0x01); + regmap_write(rt1308->regmap, 0xc360, 0x78); + regmap_write(rt1308->regmap, 0xc361, 0x87); + regmap_write(rt1308->regmap, 0xc0a1, 0x71); + regmap_write(rt1308->regmap, 0xc210, 0x00); + regmap_write(rt1308->regmap, 0xc070, 0x00); + regmap_write(rt1308->regmap, 0xc100, 0xd7); + regmap_write(rt1308->regmap, 0xc101, 0xd7); + regmap_write(rt1308->regmap, 0xc300, 0x09); + + if (rt1308->first_hw_init) { + regcache_cache_bypass(rt1308->regmap, false); + regcache_mark_dirty(rt1308->regmap); + } else + rt1308->first_hw_init = true; + + /* Mark Slave initialization complete */ + rt1308->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + +_io_init_err_: + return ret; +} + +static int rt1308_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(&slave->dev); + + /* Update the status */ + rt1308->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + rt1308->hw_init = false; + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt1308->hw_init || rt1308->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt1308_io_init(&slave->dev, slave); +} + +static int rt1308_bus_config(struct sdw_slave *slave, + struct sdw_bus_params *params) +{ + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(&slave->dev); + int ret; + + memcpy(&rt1308->params, params, sizeof(*params)); + + ret = rt1308_clock_config(&slave->dev); + if (ret < 0) + dev_err(&slave->dev, "Invalid clk config"); + + return ret; +} + +static int rt1308_interrupt_callback(struct sdw_slave *slave, + struct sdw_slave_intr_status *status) +{ + dev_dbg(&slave->dev, + "%s control_port_stat=%x", __func__, status->control_port); + + return 0; +} + +static int rt1308_classd_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); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(30); + snd_soc_component_update_bits(component, + RT1308_SDW_OFFSET | (RT1308_POWER_STATUS << 4), + 0x3, 0x3); + msleep(40); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_component_update_bits(component, + RT1308_SDW_OFFSET | (RT1308_POWER_STATUS << 4), + 0x3, 0); + usleep_range(150000, 200000); + break; + + default: + break; + } + + return 0; +} + +static const char * const rt1308_rx_data_ch_select[] = { + "LR", + "LL", + "RL", + "RR", +}; + +static SOC_ENUM_SINGLE_DECL(rt1308_rx_data_ch_enum, + RT1308_SDW_OFFSET | (RT1308_DATA_PATH << 4), 0, + rt1308_rx_data_ch_select); + +static const struct snd_kcontrol_new rt1308_snd_controls[] = { + + /* I2S Data Channel Selection */ + SOC_ENUM("RX Channel Select", rt1308_rx_data_ch_enum), +}; + +static const struct snd_kcontrol_new rt1308_sto_dac_l = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", + RT1308_SDW_OFFSET_BYTE3 | (RT1308_DAC_SET << 4), + RT1308_DVOL_MUTE_L_EN_SFT, 1, 1); + +static const struct snd_kcontrol_new rt1308_sto_dac_r = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", + RT1308_SDW_OFFSET_BYTE3 | (RT1308_DAC_SET << 4), + RT1308_DVOL_MUTE_R_EN_SFT, 1, 1); + +static const struct snd_soc_dapm_widget rt1308_dapm_widgets[] = { + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "DP1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Supply Widgets */ + SND_SOC_DAPM_SUPPLY("MBIAS20U", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ALDO", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DBG", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DACL", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLK25M", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_R", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_L", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Power", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 3, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DLDO", + RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("VREF", + RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIXER_R", + RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIXER_L", + RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MBIAS4U", + RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("PLL2_LDO", + RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2B", + RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2F", + RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2F2", + RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2B2", + RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 0, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SWITCH("DAC L", SND_SOC_NOPM, 0, 0, &rt1308_sto_dac_l), + SND_SOC_DAPM_SWITCH("DAC R", SND_SOC_NOPM, 0, 0, &rt1308_sto_dac_r), + + /* Output Lines */ + SND_SOC_DAPM_PGA_E("CLASS D", SND_SOC_NOPM, 0, 0, NULL, 0, + rt1308_classd_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), +}; + +static const struct snd_soc_dapm_route rt1308_dapm_routes[] = { + + { "DAC", NULL, "AIF1RX" }, + + { "DAC", NULL, "MBIAS20U" }, + { "DAC", NULL, "ALDO" }, + { "DAC", NULL, "DBG" }, + { "DAC", NULL, "DACL" }, + { "DAC", NULL, "CLK25M" }, + { "DAC", NULL, "ADC_R" }, + { "DAC", NULL, "ADC_L" }, + { "DAC", NULL, "DLDO" }, + { "DAC", NULL, "VREF" }, + { "DAC", NULL, "MIXER_R" }, + { "DAC", NULL, "MIXER_L" }, + { "DAC", NULL, "MBIAS4U" }, + { "DAC", NULL, "PLL2_LDO" }, + { "DAC", NULL, "PLL2B" }, + { "DAC", NULL, "PLL2F" }, + { "DAC", NULL, "PLL2F2" }, + { "DAC", NULL, "PLL2B2" }, + + { "DAC L", "Switch", "DAC" }, + { "DAC R", "Switch", "DAC" }, + { "DAC L", NULL, "DAC Power" }, + { "DAC R", NULL, "DAC Power" }, + + { "CLASS D", NULL, "DAC L" }, + { "CLASS D", NULL, "DAC R" }, + { "SPOL", NULL, "CLASS D" }, + { "SPOR", NULL, "CLASS D" }, +}; + +static int rt1308_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + struct sdw_stream_data *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = (struct sdw_stream_runtime *)sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void rt1308_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1308_sdw_priv *rt1308 = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int retval, port, num_channels; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!rt1308->sdw_slave) + return -EINVAL; + + /* SoundWire specific configuration */ + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port = 1; + } else { + return -EINVAL; + } + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = (1 << (num_channels)) - 1; + port_config.num = port; + + retval = sdw_stream_add_slave(rt1308->sdw_slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + return retval; +} + +static int rt1308_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1308_sdw_priv *rt1308 = + snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt1308->sdw_slave) + return -EINVAL; + + sdw_stream_remove_slave(rt1308->sdw_slave, stream->sdw_stream); + return 0; +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static struct sdw_slave_ops rt1308_slave_ops = { + .read_prop = rt1308_read_prop, + .interrupt_callback = rt1308_interrupt_callback, + .update_status = rt1308_update_status, + .bus_config = rt1308_bus_config, +}; + +static const struct snd_soc_component_driver soc_component_sdw_rt1308 = { + .controls = rt1308_snd_controls, + .num_controls = ARRAY_SIZE(rt1308_snd_controls), + .dapm_widgets = rt1308_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt1308_dapm_widgets), + .dapm_routes = rt1308_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt1308_dapm_routes), +}; + +static const struct snd_soc_dai_ops rt1308_aif_dai_ops = { + .hw_params = rt1308_sdw_hw_params, + .hw_free = rt1308_sdw_pcm_hw_free, + .set_sdw_stream = rt1308_set_sdw_stream, + .shutdown = rt1308_sdw_shutdown, +}; + +#define RT1308_STEREO_RATES SNDRV_PCM_RATE_48000 +#define RT1308_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver rt1308_sdw_dai[] = { + { + .name = "rt1308-aif", + .playback = { + .stream_name = "DP1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT1308_STEREO_RATES, + .formats = RT1308_FORMATS, + }, + .ops = &rt1308_aif_dai_ops, + }, +}; + +static int rt1308_sdw_init(struct device *dev, struct regmap *regmap, + struct sdw_slave *slave) +{ + struct rt1308_sdw_priv *rt1308; + int ret; + + rt1308 = devm_kzalloc(dev, sizeof(*rt1308), GFP_KERNEL); + if (!rt1308) + return -ENOMEM; + + dev_set_drvdata(dev, rt1308); + rt1308->sdw_slave = slave; + rt1308->regmap = regmap; + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt1308->hw_init = false; + rt1308->first_hw_init = false; + + ret = devm_snd_soc_register_component(dev, + &soc_component_sdw_rt1308, + rt1308_sdw_dai, + ARRAY_SIZE(rt1308_sdw_dai)); + + dev_dbg(&slave->dev, "%s\n", __func__); + + return ret; +} + +static int rt1308_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap; + + /* Assign ops */ + slave->ops = &rt1308_slave_ops; + + /* Regmap Initialization */ + regmap = devm_regmap_init_sdw(slave, &rt1308_sdw_regmap); + if (!regmap) + return -EINVAL; + + rt1308_sdw_init(&slave->dev, regmap, slave); + + return 0; +} + +static const struct sdw_device_id rt1308_id[] = { + SDW_SLAVE_ENTRY(0x025d, 0x1308, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt1308_id); + +static int rt1308_dev_suspend(struct device *dev) +{ + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev); + + if (!rt1308->hw_init) + return 0; + + regcache_cache_only(rt1308->regmap, true); + + return 0; +} + +#define RT1308_PROBE_TIMEOUT 2000 + +static int rt1308_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt1308->hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT1308_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt1308->regmap, false); + regcache_sync_region(rt1308->regmap, 0xc000, 0xcfff); + + return 0; +} + +static const struct dev_pm_ops rt1308_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt1308_dev_suspend, rt1308_dev_resume) + SET_RUNTIME_PM_OPS(rt1308_dev_suspend, rt1308_dev_resume, NULL) +}; + +static struct sdw_driver rt1308_sdw_driver = { + .driver = { + .name = "rt1308", + .owner = THIS_MODULE, + .pm = &rt1308_pm, + }, + .probe = rt1308_sdw_probe, + .ops = &rt1308_slave_ops, + .id_table = rt1308_id, +}; +module_sdw_driver(rt1308_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT1308 driver SDW"); +MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h new file mode 100644 index 000000000000..c9341e70d6cf --- /dev/null +++ b/sound/soc/codecs/rt1308-sdw.h @@ -0,0 +1,169 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * rt1308-sdw.h -- RT1308 ALSA SoC audio driver header + * + * Copyright(c) 2019 Realtek Semiconductor Corp. + */ + +#ifndef __RT1308_SDW_H__ +#define __RT1308_SDW_H__ + +static const struct reg_default rt1308_reg_defaults[] = { + { 0x0000, 0x00 }, + { 0x0001, 0x00 }, + { 0x0002, 0x00 }, + { 0x0003, 0x00 }, + { 0x0004, 0x00 }, + { 0x0005, 0x01 }, + { 0x0020, 0x00 }, + { 0x0022, 0x00 }, + { 0x0023, 0x00 }, + { 0x0024, 0x00 }, + { 0x0025, 0x00 }, + { 0x0026, 0x00 }, + { 0x0030, 0x00 }, + { 0x0032, 0x00 }, + { 0x0033, 0x00 }, + { 0x0034, 0x00 }, + { 0x0035, 0x00 }, + { 0x0036, 0x00 }, + { 0x0040, 0x00 }, + { 0x0041, 0x00 }, + { 0x0042, 0x00 }, + { 0x0043, 0x00 }, + { 0x0044, 0x20 }, + { 0x0045, 0x01 }, + { 0x0046, 0x01 }, + { 0x0048, 0x00 }, + { 0x0049, 0x00 }, + { 0x0050, 0x20 }, + { 0x0051, 0x02 }, + { 0x0052, 0x5D }, + { 0x0053, 0x13 }, + { 0x0054, 0x08 }, + { 0x0055, 0x00 }, + { 0x0060, 0x00 }, + { 0x0070, 0x00 }, + { 0x00E0, 0x00 }, + { 0x00F0, 0x00 }, + { 0x0100, 0x00 }, + { 0x0101, 0x00 }, + { 0x0102, 0x20 }, + { 0x0103, 0x00 }, + { 0x0104, 0x00 }, + { 0x0105, 0x03 }, + { 0x0120, 0x00 }, + { 0x0122, 0x00 }, + { 0x0123, 0x00 }, + { 0x0124, 0x00 }, + { 0x0125, 0x00 }, + { 0x0126, 0x00 }, + { 0x0127, 0x00 }, + { 0x0130, 0x00 }, + { 0x0132, 0x00 }, + { 0x0133, 0x00 }, + { 0x0134, 0x00 }, + { 0x0135, 0x00 }, + { 0x0136, 0x00 }, + { 0x0137, 0x00 }, + { 0x0200, 0x00 }, + { 0x0201, 0x00 }, + { 0x0202, 0x00 }, + { 0x0203, 0x00 }, + { 0x0204, 0x00 }, + { 0x0205, 0x03 }, + { 0x0220, 0x00 }, + { 0x0222, 0x00 }, + { 0x0223, 0x00 }, + { 0x0224, 0x00 }, + { 0x0225, 0x00 }, + { 0x0226, 0x00 }, + { 0x0227, 0x00 }, + { 0x0230, 0x00 }, + { 0x0232, 0x00 }, + { 0x0233, 0x00 }, + { 0x0234, 0x00 }, + { 0x0235, 0x00 }, + { 0x0236, 0x00 }, + { 0x0237, 0x00 }, + { 0x0400, 0x00 }, + { 0x0401, 0x00 }, + { 0x0402, 0x00 }, + { 0x0403, 0x00 }, + { 0x0404, 0x00 }, + { 0x0405, 0x03 }, + { 0x0420, 0x00 }, + { 0x0422, 0x00 }, + { 0x0423, 0x00 }, + { 0x0424, 0x00 }, + { 0x0425, 0x00 }, + { 0x0426, 0x00 }, + { 0x0427, 0x00 }, + { 0x0430, 0x00 }, + { 0x0432, 0x00 }, + { 0x0433, 0x00 }, + { 0x0434, 0x00 }, + { 0x0435, 0x00 }, + { 0x0436, 0x00 }, + { 0x0437, 0x00 }, + { 0x0f00, 0x00 }, + { 0x0f01, 0x00 }, + { 0x0f02, 0x00 }, + { 0x0f03, 0x00 }, + { 0x0f04, 0x00 }, + { 0x0f05, 0x00 }, + { 0x0f20, 0x00 }, + { 0x0f22, 0x00 }, + { 0x0f23, 0x00 }, + { 0x0f24, 0x00 }, + { 0x0f25, 0x00 }, + { 0x0f26, 0x00 }, + { 0x0f27, 0x00 }, + { 0x0f30, 0x00 }, + { 0x0f32, 0x00 }, + { 0x0f33, 0x00 }, + { 0x0f34, 0x00 }, + { 0x0f35, 0x00 }, + { 0x0f36, 0x00 }, + { 0x0f37, 0x00 }, + { 0x2f01, 0x01 }, + { 0x2f02, 0x09 }, + { 0x2f03, 0x00 }, + { 0x2f04, 0x0f }, + { 0x2f05, 0x0b }, + { 0x2f06, 0x01 }, + { 0x2f07, 0x8e }, + { 0x3000, 0x00 }, + { 0x3001, 0x00 }, + { 0x3004, 0x01 }, + { 0x3005, 0x23 }, + { 0x3008, 0x02 }, + { 0x300a, 0x00 }, + { 0xc003 | (RT1308_DAC_SET << 4), 0x00 }, + { 0xc001 | (RT1308_POWER << 4), 0x00 }, + { 0xc002 | (RT1308_POWER << 4), 0x00 }, +}; + +#define RT1308_SDW_OFFSET 0xc000 +#define RT1308_SDW_OFFSET_BYTE0 0xc000 +#define RT1308_SDW_OFFSET_BYTE1 0xc001 +#define RT1308_SDW_OFFSET_BYTE2 0xc002 +#define RT1308_SDW_OFFSET_BYTE3 0xc003 + +#define RT1308_SDW_RESET (RT1308_SDW_OFFSET | (RT1308_RESET << 4)) + +struct rt1308_sdw_priv { + struct snd_soc_component *component; + struct regmap *regmap; + struct sdw_slave *sdw_slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +#endif /* __RT1308_SDW_H__ */ -- 2.20.1

1 0

[alsa-devel] [PATCH v5] ASoC: Add MediaTek MT6660 Speaker Amp Driver
by Jeff Chang 13 Jan '20

13 Jan '20

From: Jeff Chang <jeff_chang(a)richtek.com> The MT6660 is a boosted BTL class-D amplifier with V/I sensing. A built-in DC-DC step-up converter is used to provide efficient power for class-D amplifier with multi-level class-G operation. The digital audio interface supports I2S, left-justified, right-justified, TDM and DSP A/B format for audio in with a data out used for chip information like voltage sense and current sense, which are able to be monitored via DATAO through proper Signed-off-by: Jeff Chang <jeff_chang(a)richtek.com> --- Documentation/devicetree/bindings/sound/mt6660.txt | 53 ++ sound/soc/codecs/Kconfig | 11 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6660.c | 642 +++++++++++++++++++++ sound/soc/codecs/mt6660.h | 77 +++ 5 files changed, 785 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6660.txt create mode 100644 sound/soc/codecs/mt6660.c create mode 100644 sound/soc/codecs/mt6660.h changelogs between v5 & v4 - remove redundant initialization. - use packed structures. - remove useless switch case. - take care of memory allocation failures. changelogs between v4 & v3 - remove unnecessary kcontrols. - modify copy right header. - use dev_dbg instead of dev_info. - add necessary debug message. - add DT binding documentation. - add space before } at every table. changelogs between v3 & v2 - modify MT6660 Kconfig, remove unnecessary selection. - remove my own debug io interface. use standard regmap for debugging. - remove regmap volatile ops, we do not use cache. - remove component io read/write function, use snd_soc_component_init_regmap. - remove init setting write code. Using parsing dts to set them. - remove unnecessary pr_info log message. - remove mt6660_component_put_volsw. Using snd_soc_get_volsw. diff --git a/Documentation/devicetree/bindings/sound/mt6660.txt b/Documentation/devicetree/bindings/sound/mt6660.txt new file mode 100644 index 0000000..2a1736b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt6660.txt @@ -0,0 +1,53 @@ +MT6660 MediaTek Speaker Amplifier + +This device supports I2C mode only. + +Required properties: + + - compatible : "mediatek,mt6660" + + - reg : The I2C slave address + +Optional properties: + + - rt,init_setting_num : The initial register setting element number. + + - rt,init_setting_addr : the addreses array for INIT Setting table. + + - rt,init_setting_mask : the mask array for INIT Setting table. + + - rt,init_setting_val : the value array for INIT Setting table. + +Example: + + mt6660@34 { + status = "ok"; + compatible = "mediatek,mt6660"; + reg = <0x34>; + rt,init_setting_num = <26>; + rt,init_setting_addr = + <0x20 0x30 0x50 0xB1 + 0xD3 0xE0 0x98 0xB9 + 0xB7 0xB6 0x6B 0x07 + 0xBB 0x69 0xBD 0x70 + 0x7C 0x46 0x1A 0x1B + 0x51 0xA2 0x33 0x4C + 0x15 0x68>; + rt,init_setting_mask = + <0x80 0x01 0x1c 0x0c + 0x03 0x01 0x44 0xff + 0x7777 0x07 0xe0 0xff + 0xff 0xff 0xffff 0xff + 0xff 0xff 0xffffffff 0xffffffff + 0xff 0xff 0xffff 0xffff + 0x1800 0x1f>; + rt,init_setting_val = + <0x00 0x00 0x04 0x00 + 0x03 0x00 0x04 0x82 + 0x7273 0x03 0x20 0x70 + 0x20 0x40 0x17f8 0x15 + 0x00 0x1d 0x7fdb7ffe 0x7fdb7ffe + 0x58 0xce 0x7fff 0x0116 + 0x0800 0x07>; + }; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 229cc89..f135fbb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_MT6358 if MTK_PMIC_WRAP + select SND_SOC_MT6660 if I2C select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8822 if I2C @@ -1465,6 +1466,16 @@ config SND_SOC_MT6358 Enable support for the platform which uses MT6358 as external codec device. +config SND_SOC_MT6660 + tristate "Mediatek MT6660 Speaker Amplifier" + depends on I2C + help + MediaTek MT6660 is a smart power amplifier which contain + speaker protection, multi-band DRC, equalizer functions. + Select N if you don't have MT6660 on board. + Select M to build this as module. + + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c498373..2b6814c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -119,6 +119,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o +snd-soc-mt6660-objs := mt6660.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -403,6 +404,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o +obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c new file mode 100644 index 0000000..5f79044 --- /dev/null +++ b/sound/soc/codecs/mt6660.c @@ -0,0 +1,642 @@ +// SPDX-License-Identifier: GPL-2.0 // + +// Copyright (c) 2019 MediaTek Inc. + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/version.h> +#include <linux/err.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/pcm_params.h> +#include <linux/debugfs.h> + +#include "mt6660.h" + +#pragma pack(push, 1) +struct codec_reg_val { + u32 addr; + u32 mask; + u32 data; +}; + +struct reg_size_table { + u32 addr; + u8 size; +}; +#pragma pack(pop) + +static const struct reg_size_table mt6660_reg_size_table[] = { + { MT6660_REG_HPF1_COEF, 4 }, + { MT6660_REG_HPF2_COEF, 4 }, + { MT6660_REG_TDM_CFG3, 2 }, + { MT6660_REG_RESV17, 2 }, + { MT6660_REG_RESV23, 2 }, + { MT6660_REG_SIGMAX, 2 }, + { MT6660_REG_DEVID, 2 }, + { MT6660_REG_HCLIP_CTRL, 2 }, + { MT6660_REG_DA_GAIN, 2 }, +}; + +static int mt6660_get_reg_size(uint32_t addr) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(mt6660_reg_size_table); i++) { + if (mt6660_reg_size_table[i].addr == addr) + return mt6660_reg_size_table[i].size; + } + return 1; +} + +static int mt6660_reg_write(void *context, unsigned int reg, unsigned int val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + u8 reg_data[4]; + int i, ret; + + for (i = 0; i < size; i++) + reg_data[size - i - 1] = (val >> (8 * i)) & 0xff; + + ret = i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); + return ret; +} + +static int mt6660_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + int i, ret; + u8 data[4]; + u32 reg_data = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, reg, size, data); + if (ret < 0) + return ret; + for (i = 0; i < size; i++) { + reg_data <<= 8; + reg_data |= data[i]; + } + *val = reg_data; + return 0; +} + +static struct regmap_config mt6660_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .reg_write = mt6660_reg_write, + .reg_read = mt6660_reg_read, +}; + +static int mt6660_codec_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + + if (event == SND_SOC_DAPM_POST_PMU) + usleep_range(1000, 1100); + return 0; +} + +static int mt6660_codec_classd_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); + int ret; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dev_dbg(component->dev, + "%s: before classd turn on\n", __func__); + /* config to adaptive mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x03); + if (ret < 0) { + dev_err(component->dev, "config mode adaptive fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMU: + /* voltage sensing enable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x04); + if (ret < 0) { + dev_err(component->dev, + "enable voltage sensing fail\n"); + return ret; + } + dev_dbg(component->dev, "Amp on\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + dev_dbg(component->dev, "Amp off\n"); + /* voltage sensing disable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x00); + if (ret < 0) { + dev_err(component->dev, + "disable voltage sensing fail\n"); + return ret; + } + /* pop-noise improvement 1 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x10); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 1 fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, + "%s: after classd turn off\n", __func__); + /* pop-noise improvement 2 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x00); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 2 fail\n"); + return ret; + } + /* config to off mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x00); + if (ret < 0) { + dev_err(component->dev, "config mode off fail\n"); + return ret; + } + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget mt6660_component_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("DAC", NULL, MT6660_REG_PLL_CFG1, + 0, 1, mt6660_codec_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC("VI ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("ClassD", MT6660_REG_SYSTEM_CTRL, 2, 0, + NULL, 0, mt6660_codec_classd_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("OUTP"), + SND_SOC_DAPM_OUTPUT("OUTN"), +}; + +static const struct snd_soc_dapm_route mt6660_component_dapm_routes[] = { + { "DAC", NULL, "aif_playback" }, + { "PGA", NULL, "DAC" }, + { "ClassD", NULL, "PGA" }, + { "OUTP", NULL, "ClassD" }, + { "OUTN", NULL, "ClassD" }, + { "VI ADC", NULL, "ClassD" }, + { "aif_capture", NULL, "VI ADC" }, +}; + +static int mt6660_component_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = -EINVAL; + + if (!strcmp(kcontrol->id.name, "Chip Rev")) { + ucontrol->value.integer.value[0] = chip->chip_rev & 0x0f; + ret = 0; + } + return ret; +} + +static const DECLARE_TLV_DB_SCALE(vol_ctl_tlv, -1155, 5, 0); + +static const struct snd_kcontrol_new mt6660_component_snd_controls[] = { + SOC_SINGLE_TLV("Digital Volume", MT6660_REG_VOL_CTRL, 0, 255, + 1, vol_ctl_tlv), + SOC_SINGLE("Hard Clip Switch", MT6660_REG_HCLIP_CTRL, 8, 1, 0), + SOC_SINGLE("Clip Switch", MT6660_REG_SPS_CTRL, 0, 1, 0), + SOC_SINGLE("Boost Mode", MT6660_REG_BST_CTRL, 0, 3, 0), + SOC_SINGLE("DRE Switch", MT6660_REG_DRE_CTRL, 0, 1, 0), + SOC_SINGLE("DC Protect Switch", MT6660_REG_DC_PROTECT_CTRL, 3, 1, 0), + SOC_SINGLE("Data Output Left Channel Selection", + MT6660_REG_DATAO_SEL, 3, 7, 0), + SOC_SINGLE("Data Output Right Channel Selection", + MT6660_REG_DATAO_SEL, 0, 7, 0), + SOC_SINGLE_EXT("T0 SEL", MT6660_REG_CALI_T0, 0, 7, 0, + snd_soc_get_volsw, NULL), + SOC_SINGLE_EXT("Chip Rev", MT6660_REG_DEVID, 8, 15, 0, + mt6660_component_get_volsw, NULL), +}; + +static int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) +{ + u8 reg_data; + int ret; + + ret = i2c_smbus_read_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL); + if (ret < 0) + return ret; + reg_data = (u8)ret; + if (on_off) + reg_data &= (~0x01); + else + reg_data |= 0x01; + return regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, reg_data); +} + +static int mt6660_apply_plat_data(struct mt6660_chip *chip, + struct snd_soc_component *component) +{ + size_t i; + int num = chip->plat_data.init_setting_num; + int ret; + + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "%s power on failed\n", __func__); + return ret; + } + + for (i = 0; i < num; i++) { + ret = snd_soc_component_update_bits(component, + chip->plat_data.init_setting_addr[i], + chip->plat_data.init_setting_mask[i], + chip->plat_data.init_setting_val[i]); + if (ret < 0) + return ret; + } + ret = _mt6660_chip_power_on(chip, 0); + if (ret < 0) { + dev_err(chip->dev, "%s power on failed\n", __func__); + return ret; + } + return 0; +} + +static int mt6660_component_probe(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret; + + dev_dbg(component->dev, "%s\n", __func__); + snd_soc_component_init_regmap(component, chip->regmap); + + ret = mt6660_apply_plat_data(chip, component); + if (ret < 0) + dev_err(chip->dev, "mt6660 apply plat data failed\n"); + + return ret; +} + +static void mt6660_component_remove(struct snd_soc_component *component) +{ + dev_dbg(component->dev, "%s\n", __func__); + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver mt6660_component_driver = { + .probe = mt6660_component_probe, + .remove = mt6660_component_remove, + + .controls = mt6660_component_snd_controls, + .num_controls = ARRAY_SIZE(mt6660_component_snd_controls), + .dapm_widgets = mt6660_component_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt6660_component_dapm_widgets), + .dapm_routes = mt6660_component_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt6660_component_dapm_routes), + + .idle_bias_on = false, /* idle_bias_off = true */ +}; + +static int mt6660_component_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) +{ + int word_len = params_physical_width(hw_params); + int aud_bit = params_width(hw_params); + u16 reg_data = 0; + int ret; + + dev_dbg(dai->dev, "%s: ++\n", __func__); + dev_dbg(dai->dev, "format: 0x%08x\n", params_format(hw_params)); + dev_dbg(dai->dev, "rate: 0x%08x\n", params_rate(hw_params)); + dev_dbg(dai->dev, "word_len: %d, aud_bit: %d\n", word_len, aud_bit); + if (word_len > 32 || word_len < 16) { + dev_err(dai->dev, "not supported word length\n"); + return -ENOTSUPP; + } + switch (aud_bit) { + case 16: + reg_data = 3; + break; + case 18: + reg_data = 2; + break; + case 20: + reg_data = 1; + break; + case 24: + case 32: + reg_data = 0; + break; + default: + return -ENOTSUPP; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_SERIAL_CFG1, 0xc0, (reg_data << 6)); + if (ret < 0) { + dev_err(dai->dev, "config aud bit fail\n"); + return ret; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_TDM_CFG3, 0x3f0, word_len << 4); + if (ret < 0) { + dev_err(dai->dev, "config word len fail\n"); + return ret; + } + dev_dbg(dai->dev, "%s: --\n", __func__); + return 0; +} + +static const struct snd_soc_dai_ops mt6660_component_aif_ops = { + .hw_params = mt6660_component_aif_hw_params, +}; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_soc_dai_driver mt6660_codec_dai = { + .name = "mt6660-aif", + .playback = { + .stream_name = "aif_playback", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "aif_capture", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + /* dai properties */ + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, + /* dai operations */ + .ops = &mt6660_component_aif_ops, +}; + +static inline int _mt6660_chip_id_check(struct mt6660_chip *chip) +{ + u8 id[2]; + int ret; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, MT6660_REG_DEVID, 2, id); + if (ret < 0) + return ret; + ret = (id[0] << 8) + id[1]; + ret &= 0x0ff0; + if (ret != 0x00e0 && ret != 0x01e0) { + dev_err(chip->dev, "%s id(%x) not match\n", __func__, ret); + return -ENODEV; + } + return ret; +} + +static inline int _mt6660_chip_sw_reset(struct mt6660_chip *chip) +{ + int ret; + + /* turn on main pll first, then trigger reset */ + ret = regmap_write(chip->regmap, 0x03, 0x00); + if (ret < 0) + return ret; + ret = regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, 0x80); + if (ret < 0) + return ret; + msleep(30); + return 0; +} + +static inline int _mt6660_read_chip_revision(struct mt6660_chip *chip) +{ + u8 reg_data[2]; + int ret; + + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, MT6660_REG_DEVID, 2, reg_data); + if (ret < 0) { + dev_err(chip->dev, "get chip revision fail\n"); + return ret; + } + chip->chip_rev = reg_data[1]; + return 0; +} + +static int mt6660_parse_dt(struct mt6660_chip *chip, struct device *dev) +{ + struct device_node *np = dev->of_node; + u32 val; + size_t i; + + if (!np) { + dev_err(dev, "no device node\n"); + return -EINVAL; + } + + if (of_property_read_u32(np, "rt,init_setting_num", &val)) { + dev_err(dev, "no init setting\n"); + chip->plat_data.init_setting_num = 0; + } else { + chip->plat_data.init_setting_num = val; + } + + if (chip->plat_data.init_setting_num) { + chip->plat_data.init_setting_addr = + devm_kzalloc(dev, sizeof(u32) * + chip->plat_data.init_setting_num, GFP_KERNEL); + if (!chip->plat_data.init_setting_addr) { + dev_err(dev, "%s addr memory alloc failed\n", __func__); + return -ENOMEM; + } + chip->plat_data.init_setting_mask = + devm_kzalloc(dev, sizeof(u32) * + chip->plat_data.init_setting_num, GFP_KERNEL); + if (!chip->plat_data.init_setting_mask) { + dev_err(dev, "%s mask memory alloc failed\n", __func__); + return -ENOMEM; + } + chip->plat_data.init_setting_val = + devm_kzalloc(dev, sizeof(u32) * + chip->plat_data.init_setting_num, GFP_KERNEL); + if (!chip->plat_data.init_setting_val) { + dev_err(dev, "%s val memory alloc failed\n", __func__); + return -ENOMEM; + } + + if (of_property_read_u32_array(np, "rt,init_setting_addr", + chip->plat_data.init_setting_addr, + chip->plat_data.init_setting_num)) { + dev_err(dev, "no init setting addr\n"); + } + if (of_property_read_u32_array(np, "rt,init_setting_mask", + chip->plat_data.init_setting_mask, + chip->plat_data.init_setting_num)) { + dev_err(dev, "no init setting mask\n"); + } + if (of_property_read_u32_array(np, "rt,init_setting_val", + chip->plat_data.init_setting_val, + chip->plat_data.init_setting_num)) { + dev_err(dev, "no init setting val\n"); + } + } + + dev_dbg(dev, "%s, init stting table, num = %d\n", __func__, + chip->plat_data.init_setting_num); + for (i = 0; i < chip->plat_data.init_setting_num; i++) { + dev_dbg(dev, "0x%02x, 0x%08x, 0x%08x\n", + chip->plat_data.init_setting_addr[i], + chip->plat_data.init_setting_mask[i], + chip->plat_data.init_setting_val[i]); + } + return 0; +} + +static int mt6660_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct mt6660_chip *chip = NULL; + int ret; + + dev_dbg(&client->dev, "%s\n", __func__); + chip = devm_kzalloc(&client->dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + chip->i2c = client; + chip->dev = &client->dev; + mutex_init(&chip->io_lock); + i2c_set_clientdata(client, chip); + + ret = mt6660_parse_dt(chip, &client->dev); + if (ret < 0) { + dev_err(&client->dev, "parsing dts failed\n"); + return ret; + } + + chip->regmap = devm_regmap_init(&client->dev, + NULL, chip, &mt6660_regmap_config); + if (IS_ERR(chip->regmap)) { + ret = PTR_ERR(chip->regmap); + dev_err(&client->dev, "failed to initialise regmap: %d\n", ret); + return ret; + } + + /* chip reset first */ + ret = _mt6660_chip_sw_reset(chip); + if (ret < 0) { + dev_err(chip->dev, "chip reset fail\n"); + goto probe_fail; + } + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 2 fail\n"); + goto probe_fail; + } + /* chip devid check */ + ret = _mt6660_chip_id_check(chip); + if (ret < 0) { + dev_err(chip->dev, "chip id check fail\n"); + goto probe_fail; + } + /* chip revision get */ + ret = _mt6660_read_chip_revision(chip); + if (ret < 0) { + dev_err(chip->dev, "read chip revision fail\n"); + goto probe_fail; + } + pm_runtime_set_active(chip->dev); + pm_runtime_enable(chip->dev); + + ret = devm_snd_soc_register_component(chip->dev, + &mt6660_component_driver, + &mt6660_codec_dai, 1); + return ret; +probe_fail: + _mt6660_chip_power_on(chip, 0); + mutex_destroy(&chip->io_lock); + return ret; +} + +static int mt6660_i2c_remove(struct i2c_client *client) +{ + struct mt6660_chip *chip = i2c_get_clientdata(client); + + pm_runtime_disable(chip->dev); + pm_runtime_set_suspended(chip->dev); + mutex_destroy(&chip->io_lock); + return 0; +} + +static int __maybe_unused mt6660_i2c_runtime_suspend(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "enter low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x01); +} + +static int __maybe_unused mt6660_i2c_runtime_resume(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "exit low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x00); +} + +static const struct dev_pm_ops mt6660_dev_pm_ops = { + SET_RUNTIME_PM_OPS(mt6660_i2c_runtime_suspend, + mt6660_i2c_runtime_resume, NULL) +}; + +static const struct of_device_id __maybe_unused mt6660_of_id[] = { + { .compatible = "mediatek,mt6660",}, + {}, +}; +MODULE_DEVICE_TABLE(of, mt6660_of_id); + +static const struct i2c_device_id mt6660_i2c_id[] = { + {"mt6660", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); + +static struct i2c_driver mt6660_i2c_driver = { + .driver = { + .name = "mt6660", + .of_match_table = of_match_ptr(mt6660_of_id), + .pm = &mt6660_dev_pm_ops, + }, + .probe = mt6660_i2c_probe, + .remove = mt6660_i2c_remove, + .id_table = mt6660_i2c_id, +}; +module_i2c_driver(mt6660_i2c_driver); + +MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com>"); +MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); +MODULE_LICENSE("GPL"); +MODULE_VERSION("1.0.7_G"); diff --git a/sound/soc/codecs/mt6660.h b/sound/soc/codecs/mt6660.h new file mode 100644 index 0000000..054a3c5 --- /dev/null +++ b/sound/soc/codecs/mt6660.h @@ -0,0 +1,77 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#ifndef __SND_SOC_MT6660_H +#define __SND_SOC_MT6660_H + +#include <linux/mutex.h> +#include <linux/regmap.h> + +#pragma pack(push, 1) +struct mt6660_platform_data { + u8 init_setting_num; + u32 *init_setting_addr; + u32 *init_setting_mask; + u32 *init_setting_val; +}; + +struct mt6660_chip { + struct i2c_client *i2c; + struct device *dev; + struct platform_device *param_dev; + struct mt6660_platform_data plat_data; + struct mutex io_lock; + struct regmap *regmap; + u16 chip_rev; +}; +#pragma pack(pop) + +#define MT6660_REG_DEVID (0x00) +#define MT6660_REG_SYSTEM_CTRL (0x03) +#define MT6660_REG_IRQ_STATUS1 (0x05) +#define MT6660_REG_ADDA_CLOCK (0x07) +#define MT6660_REG_SERIAL_CFG1 (0x10) +#define MT6660_REG_DATAO_SEL (0x12) +#define MT6660_REG_TDM_CFG3 (0x15) +#define MT6660_REG_HPF_CTRL (0x18) +#define MT6660_REG_HPF1_COEF (0x1A) +#define MT6660_REG_HPF2_COEF (0x1B) +#define MT6660_REG_PATH_BYPASS (0x1E) +#define MT6660_REG_WDT_CTRL (0x20) +#define MT6660_REG_HCLIP_CTRL (0x24) +#define MT6660_REG_VOL_CTRL (0x29) +#define MT6660_REG_SPS_CTRL (0x30) +#define MT6660_REG_SIGMAX (0x33) +#define MT6660_REG_CALI_T0 (0x3F) +#define MT6660_REG_BST_CTRL (0x40) +#define MT6660_REG_PROTECTION_CFG (0x46) +#define MT6660_REG_DA_GAIN (0x4c) +#define MT6660_REG_AUDIO_IN2_SEL (0x50) +#define MT6660_REG_SIG_GAIN (0x51) +#define MT6660_REG_PLL_CFG1 (0x60) +#define MT6660_REG_DRE_CTRL (0x68) +#define MT6660_REG_DRE_THDMODE (0x69) +#define MT6660_REG_DRE_CORASE (0x6B) +#define MT6660_REG_PWM_CTRL (0x70) +#define MT6660_REG_DC_PROTECT_CTRL (0x74) +#define MT6660_REG_ADC_USB_MODE (0x7c) +#define MT6660_REG_INTERNAL_CFG (0x88) +#define MT6660_REG_RESV0 (0x98) +#define MT6660_REG_RESV1 (0x99) +#define MT6660_REG_RESV2 (0x9A) +#define MT6660_REG_RESV3 (0x9B) +#define MT6660_REG_RESV6 (0xA2) +#define MT6660_REG_RESV7 (0xA3) +#define MT6660_REG_RESV10 (0xB0) +#define MT6660_REG_RESV11 (0xB1) +#define MT6660_REG_RESV16 (0xB6) +#define MT6660_REG_RESV17 (0xB7) +#define MT6660_REG_RESV19 (0xB9) +#define MT6660_REG_RESV21 (0xBB) +#define MT6660_REG_RESV23 (0xBD) +#define MT6660_REG_RESV31 (0xD3) +#define MT6660_REG_RESV40 (0xE0) + +#endif /* __SND_SOC_MT6660_H */ -- 2.7.4

2 1

Re: [alsa-devel] Bug report for changes in ucm2 with chtrt5645 card on Lenovo Miix 320
by Jaroslav Kysela 13 Jan '20

13 Jan '20

Dne 13. 01. 20 v 15:59 Tim Schumacher napsal(a): > 'alsaucm set _verb HiFi' gives > > ALSA lib utils.c:261:(uc_mgr_config_load) could not open configuration file /usr/share/alsa/ucm2/HDA Intel PCH/HDA Intel PCH.conf > alsaucm: unable to obtain card list: No such file or directory > > but that seems unrelated to the problem because it's the same > output whether I delete LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf > and the sound works or I try your patch. Ok, try 'alsaucm -c hw:0 set _verb HiFi'. Jaroslav > > - Tim > > On 2020-01-13, Jaroslav Kysela wrote: >> Dne 13. 01. 20 v 15:27 Tim Schumacher napsal(a): >>> The patch doesn't help unfortunately. Deleting >>> LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf still works with the >>> patch installed. Can I help debug this somehow? >> >> If you have latest alsa-utils, the command 'alsaucm dump text' should work, >> otherwise try 'alsaucm set _verb HiFi'. >> >> Jaroslav >> >>> >>> - Tim >>> >>> On 2020-01-13, Jaroslav Kysela wrote: >>>> Dne 09. 01. 20 v 23:10 Tim Schumacher napsal(a): >>>>> Hi alsa devs, >>>>> >>>>> I'm on Arch Linux on a rather uncommon laptop (Lenovo Miix 320) and my sound >>>>> stopped working on the upgrade from alsa-lib 1.1.9 to 1.2.1. wabbits on >>>>> IRC helped me track down the problem to a change in the >>>>> /usr/share/alsa/ucm (now /usr/share/alsa/ucm2) files. It seems like >>>>> /usr/share/alsa/ucm2/chtrt5645/HiFi-dmic2.conf and a symlink at >>>>> /usr/share/alsa/ucm2/chtrt5645/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf >>>>> were specifically added for this laptop but in the contrary they break the >>>>> sound for me. >>>>> >>>>> After the upgrade to alsa-lib 1.2.1 the sound is not working. The commands play no sound: >>>>> >>>>> pasuspender -- speaker-test --nloops=1 --channels=2 --test=wav --device=hw:0,0 >>>>> pasuspender -- speaker-test --nloops=1 --channels=2 --test=wav --device=hw:0,1 >>>>> >>>>> And this is the alsa-info.sh output http://alsa-project.org/db/?f=f883910a5c5101b4b1ea4202d1fe84ccd139f796 >>>>> >>>>> After deleting the /usr/share/alsa/ucm2/chtrt5645/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf >>>>> symlink both commands from above play sound and this is the alsa-info.sh >>>>> output http://alsa-project.org/db/?f=e759eb9118a191b6c3b8c021fed58abc9cf95076 >>>> >>>> Thanks. I forgot to fix the file path in chtrt5645/chtrt5645-dmic2.conf . >>>> Could you test this patch? >>>> >>>> https://github.com/alsa-project/alsa-ucm-conf/commit/74f2a0f0884df7b9f2d08d… >>>> >>>> Jaroslav >>>> >>>> -- >>>> Jaroslav Kysela <perex(a)perex.cz> >>>> Linux Sound Maintainer; ALSA Project; Red Hat, Inc. >>>> >> >> >> -- >> Jaroslav Kysela <perex(a)perex.cz> >> Linux Sound Maintainer; ALSA Project; Red Hat, Inc. >> -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

[alsa-devel] [PATCH 0/8] ASOC: SOF: Intel: improvements for Broadwell, HDA, OSS
by Pierre-Louis Bossart 13 Jan '20

13 Jan '20

This patchset provides general improvements for SOF on Intel platforms. First we have good support for SOF on Broadwell now (DMA and stability issues seem to be being us). The legacy driver remains the recommendation until UCM and topologies are provided downstream, but this is promising. Kai also fixed a couple of user-reported bugs for HDaudio/HDMI support when external graphics are enabled, and OSS emulation. Additional work is needed for the case where the Intel internal graphics solution is disabled. Also included a change on error logs which apparently made some users nervous. Bard liao (1): ASoC: SOF: Intel: lower print level to dbg if we will reinit DSP Kai Vehmanen (3): ASoC: SOF: fix PCM playback through ALSA OSS emulation ASoC: SOF: Intel: fix HDA codec driver probe with multiple controllers ASoC: hdac_hda: Fix error in driver removal after failed probe Pan Xiuli (1): ASoC: Intel: broadwell: change cpu_dai and platform components for SOF Pierre-Louis Bossart (3): ASoC: Intel: bdw-rt5677: fix Kconfig dependencies ASoC: Intel: bdw-rt5677: change cpu_dai and platform components for SOF ASoC: Intel: bdw-rt5650: change cpu_dai and platform components for SOF sound/soc/codecs/hdac_hda.c | 4 ++- sound/soc/intel/boards/Kconfig | 3 ++ sound/soc/intel/boards/bdw-rt5650.c | 10 +++++- sound/soc/intel/boards/bdw-rt5677.c | 10 +++++- sound/soc/intel/boards/broadwell.c | 10 +++++- sound/soc/sof/intel/hda-codec.c | 19 +++++++---- sound/soc/sof/intel/hda-loader.c | 6 ++-- sound/soc/sof/pcm.c | 53 +++++++++++++++++------------ 8 files changed, 79 insertions(+), 36 deletions(-) -- 2.20.1

2 15

[alsa-devel] [PATCH -next] ASoC: rt711: remove unused including <linux/version.h>
by YueHaibing 13 Jan '20

13 Jan '20

Remove including <linux/version.h> that don't need it. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/rt711.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 3bebba7a63be..2daed7692a3b 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -8,7 +8,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h>

2 1

[alsa-devel] [PATCH] ASoC: txx9: Remove unused rtd variable
by Paul Burton 13 Jan '20

13 Jan '20

Commit a857e073ffc6 ("ASoC: txx9: txx9aclc: remove snd_pcm_ops") removed the last use of the rtd variable but didn't remove its definition, leading to the following warning/error for MIPS rbtx49xx_defconfig builds: sound/soc/txx9/txx9aclc.c: In function 'txx9aclc_pcm_hw_params': sound/soc/txx9/txx9aclc.c:54:30: error: unused variable 'rtd' [-Werror=unused-variable] struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); ^~~ Resolve this by removing the unused variable. Signed-off-by: Paul Burton <paulburton(a)kernel.org> Fixes: a857e073ffc6 ("ASoC: txx9: txx9aclc: remove snd_pcm_ops") Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/txx9/txx9aclc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 33c78d33e5a1..9a55926ebf07 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -51,7 +51,6 @@ static int txx9aclc_pcm_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct txx9aclc_dmadata *dmadata = runtime->private_data; int ret; -- 2.24.1

3 2

[alsa-devel] [PATCH] ASoC: rt715: add RT715 codec driver
by jack.yu＠realtek.com 13 Jan '20

13 Jan '20

From: Jack Yu <jack.yu(a)realtek.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt715-sdw.c | 613 ++++++++++++++++++++++++ sound/soc/codecs/rt715-sdw.h | 337 ++++++++++++++ sound/soc/codecs/rt715.c | 873 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt715.h | 221 +++++++++ 6 files changed, 2056 insertions(+) create mode 100644 sound/soc/codecs/rt715-sdw.c create mode 100644 sound/soc/codecs/rt715-sdw.h create mode 100644 sound/soc/codecs/rt715.c create mode 100644 sound/soc/codecs/rt715.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 65b81888ca3d..cb27ab37a8db 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -166,6 +166,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_RT5682 if I2C select SND_SOC_RT711_SDW if SOUNDWIRE + select SND_SOC_RT715_SDW if SOUNDWIRE select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIMPLE_AMPLIFIER @@ -1069,6 +1070,15 @@ config SND_SOC_RT711_SDW select SND_SOC_RT711 select REGMAP_SOUNDWIRE +config SND_SOC_RT715 + tristate + +config SND_SOC_RT715_SDW + tristate "Realtek RT715 Codec - SDW" + depends on SOUNDWIRE + select SND_SOC_RT715 + select REGMAP_SOUNDWIRE + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f4dfe033d120..b1ac7461d7a0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -174,6 +174,7 @@ snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o snd-soc-rt5682-objs := rt5682.o snd-soc-rt711-objs := rt711.o rt711-sdw.o +snd-soc-rt715-objs := rt715.o rt715-sdw.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -467,6 +468,7 @@ obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o obj-$(CONFIG_SND_SOC_RT711) += snd-soc-rt711.o +obj-$(CONFIG_SND_SOC_RT715) += snd-soc-rt715.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c new file mode 100644 index 000000000000..c35591fd281b --- /dev/null +++ b/sound/soc/codecs/rt715-sdw.c @@ -0,0 +1,613 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * rt715-sdw.c -- rt715 ALSA SoC audio driver + * + * Copyright(c) 2019 Realtek Semiconductor Corp. + * + * ALC715 ASoC Codec Driver based Intel Dummy SdW codec driver + * + */ +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/mod_devicetable.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include "rt715.h" +#include "rt715-sdw.h" + +static bool rt715_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x00e0 ... 0x00e5: + case 0x00ee ... 0x00ef: + case 0x00f0 ... 0x00f5: + case 0x00fe ... 0x00ff: + case 0x02e0: + case 0x02f0: + case 0x04e0: + case 0x04f0: + case 0x06e0: + case 0x06f0: + case 0x2000 ... 0x2016: + case 0x201a ... 0x2027: + case 0x2029 ... 0x202a: + case 0x202d ... 0x2034: + case 0x2200 ... 0x2204: + case 0x2206 ... 0x2212: + case 0x2220 ... 0x2223: + case 0x2230 ... 0x2239: + case 0x22f0 ... 0x22f3: + case 0x3122: + case 0x3123: + case 0x3124: + case 0x3125: + case 0x3607: + case 0x3608: + case 0x3609: + case 0x3610: + case 0x3611: + case 0x3627: + case 0x3712: + case 0x3713: + case 0x3718: + case 0x3719: + case 0x371a: + case 0x371b: + case 0x371d: + case 0x3729: + case 0x385e: + case 0x3859: + case 0x4c12: + case 0x4c13: + case 0x4c1d: + case 0x4c29: + case 0x4d12: + case 0x4d13: + case 0x4d1d: + case 0x4d29: + case 0x4e12: + case 0x4e13: + case 0x4e1d: + case 0x4e29: + case 0x4f12: + case 0x4f13: + case 0x4f1d: + case 0x4f29: + case 0x7207: + case 0x7208: + case 0x7209: + case 0x7227: + case 0x7307: + case 0x7308: + case 0x7309: + case 0x7312: + case 0x7313: + case 0x7318: + case 0x7319: + case 0x731a: + case 0x731b: + case 0x731d: + case 0x7327: + case 0x7329: + case 0x8287: + case 0x8288: + case 0x8289: + case 0x82a7: + case 0x8387: + case 0x8388: + case 0x8389: + case 0x8392: + case 0x8393: + case 0x8398: + case 0x8399: + case 0x839a: + case 0x839b: + case 0x839d: + case 0x83a7: + case 0x83a9: + case 0x752039: + return true; + default: + return false; + } +} + +static bool rt715_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x00e5: + case 0x00f0: + case 0x00f3: + case 0x00f5: + case 0x2009: + case 0x2016: + case 0x201b: + case 0x201c: + case 0x201d: + case 0x201f: + case 0x2023: + case 0x2230: + case 0x200b ... 0x200e: /* i2c read */ + case 0x2012 ... 0x2015: /* HD-A read */ + case 0x202d ... 0x202f: /* BRA */ + case 0x2201 ... 0x2212: /* i2c debug */ + case 0x2220 ... 0x2223: /* decoded HD-A */ + return true; + default: + return false; + } +} + +static int rt715_sdw_read(void *context, unsigned int reg, unsigned int *val) +{ + struct device *dev = context; + struct rt715_priv *rt715 = dev_get_drvdata(dev); + unsigned int sdw_data_3, sdw_data_2, sdw_data_1, sdw_data_0; + unsigned int reg2 = 0, reg3 = 0, reg4 = 0, mask, nid, val2; + unsigned int is_hda_reg = 1, is_index_reg = 0; + int ret; + + if (reg > 0xffff) + is_index_reg = 1; + + mask = reg & 0xf000; + + if (is_index_reg) { /* index registers */ + val2 = reg & 0xff; + reg = reg >> 8; + nid = reg & 0xff; + ret = regmap_write(rt715->sdw_regmap, reg, 0); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt715->sdw_regmap, reg2, val2); + if (ret < 0) + return ret; + + reg3 = RT715_PRIV_DATA_R_H | nid; + ret = regmap_write(rt715->sdw_regmap, reg3, + ((*val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg4 = reg3 + 0x1000; + reg4 |= 0x80; + ret = regmap_write(rt715->sdw_regmap, reg4, (*val & 0xff)); + if (ret < 0) + return ret; + } else if (mask == 0x3000) { + reg += 0x8000; + ret = regmap_write(rt715->sdw_regmap, reg, *val); + if (ret < 0) + return ret; + } else if (mask == 0x7000) { + reg += 0x2000; + reg |= 0x800; + ret = regmap_write(rt715->sdw_regmap, reg, + ((*val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt715->sdw_regmap, reg2, (*val & 0xff)); + if (ret < 0) + return ret; + } else if ((reg & 0xff00) == 0x8300) { /* for R channel */ + reg2 = reg - 0x1000; + reg2 &= ~0x80; + ret = regmap_write(rt715->sdw_regmap, reg2, + ((*val >> 8) & 0xff)); + if (ret < 0) + return ret; + ret = regmap_write(rt715->sdw_regmap, reg, (*val & 0xff)); + if (ret < 0) + return ret; + } else if (mask == 0x9000) { + ret = regmap_write(rt715->sdw_regmap, reg, + ((*val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt715->sdw_regmap, reg2, (*val & 0xff)); + if (ret < 0) + return ret; + } else if (mask == 0xb000) { + ret = regmap_write(rt715->sdw_regmap, reg, *val); + if (ret < 0) + return ret; + } else { + ret = regmap_read(rt715->sdw_regmap, reg, val); + if (ret < 0) + return ret; + is_hda_reg = 0; + } + + if (is_hda_reg || is_index_reg) { + sdw_data_3 = 0; + sdw_data_2 = 0; + sdw_data_1 = 0; + sdw_data_0 = 0; + ret = regmap_read(rt715->sdw_regmap, RT715_READ_HDA_3, + &sdw_data_3); + if (ret < 0) + return ret; + ret = regmap_read(rt715->sdw_regmap, RT715_READ_HDA_2, + &sdw_data_2); + if (ret < 0) + return ret; + ret = regmap_read(rt715->sdw_regmap, RT715_READ_HDA_1, + &sdw_data_1); + if (ret < 0) + return ret; + ret = regmap_read(rt715->sdw_regmap, RT715_READ_HDA_0, + &sdw_data_0); + if (ret < 0) + return ret; + *val = ((sdw_data_3 & 0xff) << 24) | + ((sdw_data_2 & 0xff) << 16) | + ((sdw_data_1 & 0xff) << 8) | (sdw_data_0 & 0xff); + } + + if (is_hda_reg == 0) + dev_dbg(dev, "[%s] %04x => %08x\n", __func__, reg, *val); + else if (is_index_reg) + dev_dbg(dev, "[%s] %04x %04x %04x %04x => %08x\n", __func__, + reg, reg2, reg3, reg4, *val); + else + dev_dbg(dev, "[%s] %04x %04x => %08x\n", + __func__, reg, reg2, *val); + + return 0; +} + +static int rt715_sdw_write(void *context, unsigned int reg, unsigned int val) +{ + struct device *dev = context; + struct rt715_priv *rt715 = dev_get_drvdata(dev); + unsigned int reg2 = 0, reg3, reg4, nid, mask, val2; + unsigned int is_index_reg = 0; + int ret; + + if (reg > 0xffff) + is_index_reg = 1; + + mask = reg & 0xf000; + + if (is_index_reg) { /* index registers */ + val2 = reg & 0xff; + reg = reg >> 8; + nid = reg & 0xff; + ret = regmap_write(rt715->sdw_regmap, reg, 0); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt715->sdw_regmap, reg2, val2); + if (ret < 0) + return ret; + + reg3 = RT715_PRIV_DATA_W_H | nid; + ret = regmap_write(rt715->sdw_regmap, reg3, + ((val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg4 = reg3 + 0x1000; + reg4 |= 0x80; + ret = regmap_write(rt715->sdw_regmap, reg4, (val & 0xff)); + if (ret < 0) + return ret; + is_index_reg = 1; + } else if (reg < 0x4fff) { + ret = regmap_write(rt715->sdw_regmap, reg, val); + if (ret < 0) + return ret; + } else if (reg == RT715_FUNC_RESET) { + ret = regmap_write(rt715->sdw_regmap, reg, val); + if (ret < 0) + return ret; + } else if (mask == 0x7000) { + ret = regmap_write(rt715->sdw_regmap, reg, + ((val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt715->sdw_regmap, reg2, (val & 0xff)); + if (ret < 0) + return ret; + } else if ((reg & 0xff00) == 0x8300) { /* for R channel */ + reg2 = reg - 0x1000; + reg2 &= ~0x80; + ret = regmap_write(rt715->sdw_regmap, reg2, + ((val >> 8) & 0xff)); + if (ret < 0) + return ret; + ret = regmap_write(rt715->sdw_regmap, reg, (val & 0xff)); + if (ret < 0) + return ret; + } + + if (reg2 == 0) + dev_dbg(dev, "[%s] %04x <= %04x\n", __func__, reg, val); + else if (is_index_reg) + dev_dbg(dev, "[%s] %04x %04x %04x %04x <= %04x %04x\n", + __func__, reg, reg2, reg3, reg4, val2, val); + else + dev_dbg(dev, "[%s] %04x %04x <= %04x\n", + __func__, reg, reg2, val); + + return 0; +} + +static const struct regmap_config rt715_regmap = { + .reg_bits = 24, + .val_bits = 32, + .readable_reg = rt715_readable_register, /* Readable registers */ + .volatile_reg = rt715_volatile_register, /* volatile register */ + .max_register = 0x752039, /* Maximum number of register */ + .reg_defaults = rt715_reg_defaults, /* Defaults */ + .num_reg_defaults = ARRAY_SIZE(rt715_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, + .reg_read = rt715_sdw_read, + .reg_write = rt715_sdw_write, +}; + +static const struct regmap_config rt715_sdw_regmap = { + .name = "sdw", + .reg_bits = 32, /* Total register space for SDW */ + .val_bits = 8, /* Total number of bits in register */ + .max_register = 0xff01, /* Maximum number of register */ + .cache_type = REGCACHE_NONE, + .use_single_read = true, + .use_single_write = true, +}; + +int hda_to_sdw(unsigned int nid, unsigned int verb, unsigned int payload, + unsigned int *sdw_addr_h, unsigned int *sdw_data_h, + unsigned int *sdw_addr_l, unsigned int *sdw_data_l) +{ + unsigned int offset_h, offset_l, e_verb; + + if (((verb & 0xff) != 0) || verb == 0xf00) { /* 12 bits command */ + if (verb == 0x7ff) /* special case */ + offset_h = 0; + else + offset_h = 0x3000; + + if (verb & 0x800) /* get command */ + e_verb = (verb - 0xf00) | 0x80; + else /* set command */ + e_verb = (verb - 0x700); + + *sdw_data_h = payload; /* 7 bits payload */ + *sdw_addr_l = *sdw_data_l = 0; + } else { /* 4 bits command */ + if ((verb & 0x800) == 0x800) { /* read */ + offset_h = 0x9000; + offset_l = 0xa000; + } else { /* write */ + offset_h = 0x7000; + offset_l = 0x8000; + } + e_verb = verb >> 8; + *sdw_data_h = (payload >> 8); /* 16 bits payload [15:8] */ + *sdw_addr_l = (e_verb << 8) | nid | 0x80; /* 0x80: valid bit */ + *sdw_addr_l += offset_l; + *sdw_data_l = payload & 0xff; + } + + *sdw_addr_h = (e_verb << 8) | nid; + *sdw_addr_h += offset_h; + + return 0; +} +EXPORT_SYMBOL(hda_to_sdw); + +static int rt715_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt715_priv *rt715 = dev_get_drvdata(&slave->dev); + + /* Update the status */ + rt715->status = status; + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt715->hw_init || rt715->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt715_io_init(&slave->dev, slave); +} + +static int rt715_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval, i, num_of_ports = 1; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->paging_support = false; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x50;/* BITMAP: 01010000 */ + prop->sink_ports = 0x0; /* BITMAP: 00000000 */ + + nval = hweight32(prop->source_ports); + num_of_ports += nval; + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), + GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + dpn = prop->src_dpn_prop; + i = 0; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + num_of_ports += nval; + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), + GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + dpn = prop->sink_dpn_prop; + i = 0; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* Allocate port_ready based on num_of_ports */ + slave->port_ready = devm_kcalloc(&slave->dev, num_of_ports, + sizeof(*slave->port_ready), + GFP_KERNEL); + if (!slave->port_ready) + return -ENOMEM; + + /* Initialize completion */ + for (i = 0; i < num_of_ports; i++) + init_completion(&slave->port_ready[i]); + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + /* wake-up event */ + prop->wake_capable = 1; + + return 0; +} + +static int rt715_bus_config(struct sdw_slave *slave, + struct sdw_bus_params *params) +{ + struct rt715_priv *rt715 = dev_get_drvdata(&slave->dev); + int ret; + + memcpy(&rt715->params, params, sizeof(*params)); + + ret = rt715_clock_config(&slave->dev); + if (ret < 0) + dev_err(&slave->dev, "Invalid clk config"); + + return 0; +} + +static struct sdw_slave_ops rt715_slave_ops = { + .read_prop = rt715_read_prop, + .update_status = rt715_update_status, + .bus_config = rt715_bus_config, +}; + +static int rt715_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *sdw_regmap, *regmap; + + /* Assign ops */ + slave->ops = &rt715_slave_ops; + + /* Regmap Initialization */ + sdw_regmap = devm_regmap_init_sdw(slave, &rt715_sdw_regmap); + if (!sdw_regmap) + return -EINVAL; + + regmap = devm_regmap_init(&slave->dev, NULL, &slave->dev, + &rt715_regmap); + if (!regmap) + return -EINVAL; + + rt715_init(&slave->dev, sdw_regmap, regmap, slave); + + return 0; +} + +static const struct sdw_device_id rt715_id[] = { + SDW_SLAVE_ENTRY(0x025d, 0x715, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt715_id); + +static int rt715_dev_suspend(struct device *dev) +{ + struct rt715_priv *rt715 = dev_get_drvdata(dev); + + if (!rt715->hw_init) + return 0; + + regcache_cache_only(rt715->regmap, true); + + return 0; +} + +#define RT715_PROBE_TIMEOUT 2000 + +static int rt715_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = to_sdw_slave_device(dev); + struct rt715_priv *rt715 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt715->hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT715_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt715->regmap, false); + regcache_sync_region(rt715->regmap, 0x3000, 0x8fff); + regcache_sync_region(rt715->regmap, 0x752039, 0x752039); + + return 0; +} + +static const struct dev_pm_ops rt715_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt715_dev_suspend, rt715_dev_resume) + SET_RUNTIME_PM_OPS(rt715_dev_suspend, rt715_dev_resume, NULL) +}; + +static struct sdw_driver rt715_sdw_driver = { + .driver = { + .name = "rt715", + .owner = THIS_MODULE, + .pm = &rt715_pm, + }, + .probe = rt715_sdw_probe, + .ops = &rt715_slave_ops, + .id_table = rt715_id, +}; +module_sdw_driver(rt715_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT715 driver SDW"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt715-sdw.h b/sound/soc/codecs/rt715-sdw.h new file mode 100644 index 000000000000..5d7661e335ae --- /dev/null +++ b/sound/soc/codecs/rt715-sdw.h @@ -0,0 +1,337 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * rt715-sdw.h -- RT715 ALSA SoC audio driver header + * + * Copyright(c) 2019 Realtek Semiconductor Corp. + */ + +#ifndef __RT715_SDW_H__ +#define __RT715_SDW_H__ + +static const struct reg_default rt715_reg_defaults[] = { + { 0x0000, 0x00 }, + { 0x0001, 0x00 }, + { 0x0002, 0x00 }, + { 0x0003, 0x00 }, + { 0x0004, 0x00 }, + { 0x0005, 0x01 }, + { 0x0020, 0x00 }, + { 0x0022, 0x00 }, + { 0x0023, 0x00 }, + { 0x0024, 0x00 }, + { 0x0025, 0x00 }, + { 0x0026, 0x00 }, + { 0x0030, 0x00 }, + { 0x0032, 0x00 }, + { 0x0033, 0x00 }, + { 0x0034, 0x00 }, + { 0x0035, 0x00 }, + { 0x0036, 0x00 }, + { 0x0040, 0x00 }, + { 0x0041, 0x00 }, + { 0x0042, 0x00 }, + { 0x0043, 0x00 }, + { 0x0044, 0x20 }, + { 0x0045, 0x01 }, + { 0x0046, 0x00 }, + { 0x0050, 0x20 }, + { 0x0051, 0x02 }, + { 0x0052, 0x5d }, + { 0x0053, 0x07 }, + { 0x0054, 0x15 }, + { 0x0055, 0x00 }, + { 0x0060, 0x00 }, + { 0x0070, 0x00 }, + { 0x0080, 0x00 }, + { 0x0088, 0x10 }, + { 0x00e0, 0x00 }, + { 0x00e1, 0x00 }, + { 0x00e2, 0x00 }, + { 0x00e3, 0x00 }, + { 0x00e4, 0x00 }, + { 0x00e5, 0x00 }, + { 0x00ee, 0x00 }, + { 0x00ef, 0x00 }, + { 0x00f0, 0x00 }, + { 0x00f1, 0x00 }, + { 0x00f2, 0x00 }, + { 0x00f3, 0x00 }, + { 0x00f4, 0x00 }, + { 0x00f5, 0x00 }, + { 0x00fe, 0x00 }, + { 0x00ff, 0x00 }, + { 0x0200, 0x00 }, + { 0x0201, 0x00 }, + { 0x0202, 0x20 }, + { 0x0203, 0x00 }, + { 0x0204, 0x00 }, + { 0x0205, 0x03 }, + { 0x0220, 0x00 }, + { 0x0221, 0x00 }, + { 0x0222, 0x00 }, + { 0x0223, 0x00 }, + { 0x0224, 0x00 }, + { 0x0225, 0x00 }, + { 0x0226, 0x00 }, + { 0x0227, 0x00 }, + { 0x0230, 0x00 }, + { 0x0231, 0x00 }, + { 0x0232, 0x00 }, + { 0x0233, 0x00 }, + { 0x0234, 0x00 }, + { 0x0235, 0x00 }, + { 0x0236, 0x00 }, + { 0x0237, 0x00 }, + { 0x02e0, 0x00 }, + { 0x02f0, 0x00 }, + { 0x0400, 0x00 }, + { 0x0401, 0x00 }, + { 0x0402, 0x20 }, + { 0x0403, 0x00 }, + { 0x0404, 0x00 }, + { 0x0405, 0x0f }, + { 0x0420, 0x00 }, + { 0x0421, 0x00 }, + { 0x0422, 0x00 }, + { 0x0423, 0x00 }, + { 0x0424, 0x00 }, + { 0x0425, 0x00 }, + { 0x0426, 0x00 }, + { 0x0427, 0x00 }, + { 0x0430, 0x00 }, + { 0x0431, 0x00 }, + { 0x0432, 0x00 }, + { 0x0433, 0x00 }, + { 0x0434, 0x00 }, + { 0x0435, 0x00 }, + { 0x0436, 0x00 }, + { 0x0437, 0x00 }, + { 0x04e0, 0x00 }, + { 0x04f0, 0x00 }, + { 0x0600, 0x00 }, + { 0x0601, 0x00 }, + { 0x0602, 0x20 }, + { 0x0603, 0x00 }, + { 0x0604, 0x00 }, + { 0x0605, 0xff }, + { 0x0620, 0x00 }, + { 0x0621, 0x00 }, + { 0x0622, 0x00 }, + { 0x0623, 0x00 }, + { 0x0624, 0x00 }, + { 0x0625, 0x00 }, + { 0x0626, 0x00 }, + { 0x0627, 0x00 }, + { 0x0630, 0x00 }, + { 0x0631, 0x00 }, + { 0x0632, 0x00 }, + { 0x0633, 0x00 }, + { 0x0634, 0x00 }, + { 0x0635, 0x00 }, + { 0x0636, 0x00 }, + { 0x0637, 0x00 }, + { 0x06e0, 0x00 }, + { 0x06f0, 0x00 }, + { 0x0f00, 0x00 }, + { 0x0f01, 0x00 }, + { 0x0f02, 0x00 }, + { 0x0f03, 0x00 }, + { 0x0f04, 0x00 }, + { 0x0f05, 0xff }, + { 0x0f06, 0x00 }, + { 0x0f07, 0x00 }, + { 0x0f08, 0x00 }, + { 0x0f09, 0x00 }, + { 0x0f0a, 0x00 }, + { 0x0f0b, 0x00 }, + { 0x0f0c, 0x00 }, + { 0x0f0d, 0x00 }, + { 0x0f0e, 0x00 }, + { 0x0f0f, 0x00 }, + { 0x0f10, 0x00 }, + { 0x0f11, 0x00 }, + { 0x0f12, 0x00 }, + { 0x0f13, 0x00 }, + { 0x0f14, 0x00 }, + { 0x0f15, 0x00 }, + { 0x0f16, 0x00 }, + { 0x0f17, 0x00 }, + { 0x0f18, 0x00 }, + { 0x0f19, 0x00 }, + { 0x0f1a, 0x00 }, + { 0x0f1b, 0x00 }, + { 0x0f1c, 0x00 }, + { 0x0f1d, 0x00 }, + { 0x0f1e, 0x00 }, + { 0x0f1f, 0x00 }, + { 0x0f20, 0x00 }, + { 0x0f21, 0x00 }, + { 0x0f22, 0x00 }, + { 0x0f23, 0x00 }, + { 0x0f24, 0x00 }, + { 0x0f25, 0x00 }, + { 0x0f26, 0x00 }, + { 0x0f27, 0x00 }, + { 0x0f30, 0x00 }, + { 0x0f31, 0x00 }, + { 0x0f32, 0x00 }, + { 0x0f33, 0x00 }, + { 0x0f34, 0x00 }, + { 0x0f35, 0x00 }, + { 0x0f36, 0x00 }, + { 0x0f37, 0x00 }, + { 0x2000, 0x00 }, + { 0x2001, 0x00 }, + { 0x2002, 0x00 }, + { 0x2003, 0x00 }, + { 0x2004, 0x00 }, + { 0x2005, 0x00 }, + { 0x2006, 0x00 }, + { 0x2007, 0x00 }, + { 0x2008, 0x00 }, + { 0x2009, 0x03 }, + { 0x200a, 0x00 }, + { 0x200b, 0x00 }, + { 0x200c, 0x00 }, + { 0x200d, 0x00 }, + { 0x200e, 0x00 }, + { 0x200f, 0x10 }, + { 0x2010, 0x00 }, + { 0x2011, 0x00 }, + { 0x2012, 0x00 }, + { 0x2013, 0x00 }, + { 0x2014, 0x00 }, + { 0x2015, 0x00 }, + { 0x2016, 0x00 }, + { 0x201a, 0x00 }, + { 0x201b, 0x00 }, + { 0x201c, 0x00 }, + { 0x201d, 0x00 }, + { 0x201e, 0x00 }, + { 0x201f, 0x00 }, + { 0x2020, 0x00 }, + { 0x2021, 0x00 }, + { 0x2022, 0x00 }, + { 0x2023, 0x00 }, + { 0x2024, 0x00 }, + { 0x2025, 0x01 }, + { 0x2026, 0x00 }, + { 0x2027, 0x00 }, + { 0x2029, 0x00 }, + { 0x202a, 0x00 }, + { 0x202d, 0x00 }, + { 0x202e, 0x00 }, + { 0x202f, 0x00 }, + { 0x2030, 0x00 }, + { 0x2031, 0x00 }, + { 0x2032, 0x00 }, + { 0x2033, 0x00 }, + { 0x2034, 0x00 }, + { 0x2200, 0x00 }, + { 0x2201, 0x00 }, + { 0x2202, 0x00 }, + { 0x2203, 0x00 }, + { 0x2204, 0x00 }, + { 0x2206, 0x00 }, + { 0x2207, 0x00 }, + { 0x2208, 0x00 }, + { 0x2209, 0x00 }, + { 0x220a, 0x00 }, + { 0x220b, 0x00 }, + { 0x220c, 0x00 }, + { 0x220d, 0x00 }, + { 0x220e, 0x00 }, + { 0x220f, 0x00 }, + { 0x2210, 0x00 }, + { 0x2211, 0x00 }, + { 0x2212, 0x00 }, + { 0x2220, 0x00 }, + { 0x2221, 0x00 }, + { 0x2222, 0x00 }, + { 0x2223, 0x00 }, + { 0x2230, 0x00 }, + { 0x2231, 0x0f }, + { 0x2232, 0x00 }, + { 0x2233, 0x00 }, + { 0x2234, 0x00 }, + { 0x2235, 0x00 }, + { 0x2236, 0x00 }, + { 0x2237, 0x00 }, + { 0x2238, 0x00 }, + { 0x2239, 0x00 }, + { 0x22f0, 0x00 }, + { 0x22f1, 0x00 }, + { 0x22f2, 0x00 }, + { 0x22f3, 0x00 }, + { 0x3122, 0x02 }, + { 0x3123, 0x03 }, + { 0x3124, 0x00 }, + { 0x3125, 0x01 }, + { 0x3607, 0x00 }, + { 0x3608, 0x00 }, + { 0x3609, 0x00 }, + { 0x3610, 0x00 }, + { 0x3611, 0x00 }, + { 0x3627, 0x00 }, + { 0x3712, 0x00 }, + { 0x3713, 0x00 }, + { 0x3718, 0x00 }, + { 0x3719, 0x00 }, + { 0x371a, 0x00 }, + { 0x371b, 0x00 }, + { 0x371d, 0x00 }, + { 0x3729, 0x00 }, + { 0x385e, 0x00 }, + { 0x3859, 0x00 }, + { 0x4c12, 0x411111f0 }, + { 0x4c13, 0x411111f0 }, + { 0x4c1d, 0x411111f0 }, + { 0x4c29, 0x411111f0 }, + { 0x4d12, 0x411111f0 }, + { 0x4d13, 0x411111f0 }, + { 0x4d1d, 0x411111f0 }, + { 0x4d29, 0x411111f0 }, + { 0x4e12, 0x411111f0 }, + { 0x4e13, 0x411111f0 }, + { 0x4e1d, 0x411111f0 }, + { 0x4e29, 0x411111f0 }, + { 0x4f12, 0x411111f0 }, + { 0x4f13, 0x411111f0 }, + { 0x4f1d, 0x411111f0 }, + { 0x4f29, 0x411111f0 }, + { 0x7207, 0x00 }, + { 0x8287, 0x00 }, + { 0x7208, 0x00 }, + { 0x8288, 0x00 }, + { 0x7209, 0x00 }, + { 0x8289, 0x00 }, + { 0x7227, 0x00 }, + { 0x82a7, 0x00 }, + { 0x7307, 0x97 }, + { 0x8387, 0x97 }, + { 0x7308, 0x97 }, + { 0x8388, 0x97 }, + { 0x7309, 0x97 }, + { 0x8389, 0x97 }, + { 0x7312, 0x00 }, + { 0x8392, 0x00 }, + { 0x7313, 0x00 }, + { 0x8393, 0x00 }, + { 0x7318, 0x00 }, + { 0x8398, 0x00 }, + { 0x7319, 0x00 }, + { 0x8399, 0x00 }, + { 0x731a, 0x00 }, + { 0x839a, 0x00 }, + { 0x731b, 0x00 }, + { 0x839b, 0x00 }, + { 0x731d, 0x00 }, + { 0x839d, 0x00 }, + { 0x7327, 0x97 }, + { 0x83a7, 0x97 }, + { 0x7329, 0x00 }, + { 0x83a9, 0x00 }, + { 0x752039, 0xa500 }, +}; + +#endif /* __RT715_H__ */ diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c new file mode 100644 index 000000000000..5c6f05b8d8ab --- /dev/null +++ b/sound/soc/codecs/rt715.c @@ -0,0 +1,873 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * rt715.c -- rt715 ALSA SoC audio driver + * + * Copyright(c) 2019 Realtek Semiconductor Corp. + * + * ALC715 ASoC Codec Driver based Intel Dummy SdW codec driver + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/pm.h> +#include <linux/soundwire/sdw.h> +#include <linux/gpio.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/hda_verbs.h> + +#include "rt715.h" + +static int rt715_index_write(struct regmap *regmap, unsigned int reg, + unsigned int value) +{ + int ret; + unsigned int addr = ((RT715_PRIV_INDEX_W_H) << 8) | reg; + + ret = regmap_write(regmap, addr, value); + if (ret < 0) { + pr_err("Failed to set private value: %08x <= %04x %d\n", ret, + addr, value); + } + + return ret; +} + +static void rt715_get_gain(struct rt715_priv *rt715, unsigned int addr_h, + unsigned int addr_l, unsigned int val_h, + unsigned int *r_val, unsigned int *l_val) +{ + int ret; + /* R Channel */ + *r_val = (val_h << 8); + ret = regmap_read(rt715->regmap, addr_l, r_val); + if (ret < 0) + pr_err("Failed to get R channel gain.\n"); + + /* L Channel */ + val_h |= 0x20; + *l_val = (val_h << 8); + ret = regmap_read(rt715->regmap, addr_h, l_val); + if (ret < 0) + pr_err("Failed to get L channel gain.\n"); +} + +/* For Verb-Set Amplifier Gain (Verb ID = 3h) */ +static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int addr_h, addr_l, val_h, val_ll, val_lr; + unsigned int read_ll, read_rl; + int i; + + /* Can't use update bit function, so read the original value first */ + addr_h = mc->reg; + addr_l = mc->rreg; + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + /* L Channel */ + if (mc->invert) { + /* for mute */ + val_ll = (mc->max - ucontrol->value.integer.value[0]) << 7; + /* keep gain */ + read_ll = read_ll & 0x7f; + val_ll |= read_ll; + } else { + /* for gain */ + val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); + if (val_ll > mc->max) + val_ll = mc->max; + /* keep mute status */ + read_ll = read_ll & 0x80; + val_ll |= read_ll; + } + + /* R Channel */ + if (mc->invert) { + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + /* for mute */ + val_lr = (mc->max - ucontrol->value.integer.value[1]) << 7; + /* keep gain */ + read_rl = read_rl & 0x7f; + val_lr |= read_rl; + } else { + /* for gain */ + val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); + if (val_lr > mc->max) + val_lr = mc->max; + /* keep mute status */ + read_rl = read_rl & 0x80; + val_lr |= read_rl; + } + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << mc->shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8 | val_ll)); + regmap_write(rt715->regmap, addr_l, + (val_h << 8 | val_ll)); + } else { + /* Lch*/ + val_h = (1 << mc->shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8 | val_ll)); + /* Rch */ + val_h = (1 << mc->shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8 | val_lr)); + } + /* check result */ + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } + /* D0:power on state, D3: power saving mode */ + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + return 0; +} + +static int rt715_set_amp_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int addr_h, addr_l, val_h; + unsigned int read_ll, read_rl; + + addr_h = mc->reg; + addr_l = mc->rreg; + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (mc->invert) { + /* for mute status */ + read_ll = !((read_ll & 0x80) >> RT715_MUTE_SFT); + read_rl = !((read_rl & 0x80) >> RT715_MUTE_SFT); + } else { + /* for gain */ + read_ll = read_ll & 0x7f; + read_rl = read_rl & 0x7f; + } + ucontrol->value.integer.value[0] = read_ll; + ucontrol->value.integer.value[1] = read_rl; + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +#define SOC_DOUBLE_R_EXT(xname, reg_left, reg_right, xshift, xmax, xinvert,\ + xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ + xmax, xinvert) } + +static const struct snd_kcontrol_new rt715_snd_controls[] = { + /* Capture switch */ + SOC_DOUBLE_R_EXT("ADC 07 Capture Switch", RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 1, 1, + rt715_set_amp_gain_get, rt715_set_amp_gain_put), + SOC_DOUBLE_R_EXT("ADC 08 Capture Switch", RT715_SET_GAIN_LINE_ADC_H, + RT715_SET_GAIN_LINE_ADC_L, RT715_DIR_IN_SFT, 1, 1, + rt715_set_amp_gain_get, rt715_set_amp_gain_put), + SOC_DOUBLE_R_EXT("ADC 09 Capture Switch", RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC_L, RT715_DIR_IN_SFT, 1, 1, + rt715_set_amp_gain_get, rt715_set_amp_gain_put), + SOC_DOUBLE_R_EXT("ADC 27 Capture Switch", RT715_SET_GAIN_MIX_ADC2_H, + RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 1, 1, + rt715_set_amp_gain_get, rt715_set_amp_gain_put), + /* Volume Control */ + SOC_DOUBLE_R_EXT_TLV("ADC 07 Capture Volume", RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + in_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("ADC 08 Capture Volume", RT715_SET_GAIN_LINE_ADC_H, + RT715_SET_GAIN_LINE_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + in_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("ADC 09 Capture Volume", RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + in_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("ADC 27 Capture Volume", RT715_SET_GAIN_MIX_ADC2_H, + RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 0x3f, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + in_vol_tlv), + /* MIC Boost Control */ + SOC_DOUBLE_R_EXT_TLV("DMIC1 Boost", RT715_SET_GAIN_DMIC1_H, + RT715_SET_GAIN_DMIC1_L, RT715_DIR_IN_SFT, 3, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + mic_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("DMIC2 Boost", RT715_SET_GAIN_DMIC2_H, + RT715_SET_GAIN_DMIC2_L, RT715_DIR_IN_SFT, 3, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + mic_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("DMIC3 Boost", RT715_SET_GAIN_DMIC3_H, + RT715_SET_GAIN_DMIC3_L, RT715_DIR_IN_SFT, 3, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + mic_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("DMIC4 Boost", RT715_SET_GAIN_DMIC4_H, + RT715_SET_GAIN_DMIC4_L, RT715_DIR_IN_SFT, 3, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + mic_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("MIC1 Boost", RT715_SET_GAIN_MIC1_H, + RT715_SET_GAIN_MIC1_L, RT715_DIR_IN_SFT, 3, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + mic_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("MIC2 Boost", RT715_SET_GAIN_MIC2_H, + RT715_SET_GAIN_MIC2_L, RT715_DIR_IN_SFT, 3, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + mic_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("LINE1 Boost", RT715_SET_GAIN_LINE1_H, + RT715_SET_GAIN_LINE1_L, RT715_DIR_IN_SFT, 3, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + mic_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("LINE2 Boost", RT715_SET_GAIN_LINE2_H, + RT715_SET_GAIN_LINE2_L, RT715_DIR_IN_SFT, 3, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + mic_vol_tlv), +}; + +static int rt715_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int reg, val; + int ret; + + /* nid = e->reg, vid = 0xf01 */ + reg = RT715_VERB_SET_CONNECT_SEL | e->reg; + ret = regmap_read(rt715->regmap, reg, &val); + if (ret < 0) { + dev_err(component->dev, "%s: sdw read failed: %d\n", + __func__, ret); + return ret; + } + + /* + * The first two indices of ADC Mux 24/25 are routed to the same + * hardware source. ie, ADC Mux 24 0/1 will both connect to MIC2. + * To have a unique set of inputs, we skip the index1 of the muxes. + */ + if ((e->reg == RT715_MUX_IN3 || e->reg == RT715_MUX_IN4) && (val > 0)) + val -= 1; + ucontrol->value.enumerated.item[0] = val; + + return 0; +} + +static int rt715_mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int val, val2 = 0, change, reg; + int ret; + + if (item[0] >= e->items) + return -EINVAL; + + /* Verb ID = 0x701h, nid = e->reg */ + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + + reg = RT715_VERB_SET_CONNECT_SEL | e->reg; + ret = regmap_read(rt715->regmap, reg, &val2); + if (ret < 0) { + dev_err(component->dev, "%s: sdw read failed: %d\n", + __func__, ret); + return ret; + } + + if (val == val2) + change = 0; + else + change = 1; + + if (change) { + reg = RT715_VERB_SET_CONNECT_SEL | e->reg; + regmap_write(rt715->regmap, reg, val); + } + + snd_soc_dapm_mux_update_power(dapm, kcontrol, + item[0], e, NULL); + + return change; +} + +static const char * const adc_22_23_mux_text[] = { + "MIC1", + "MIC2", + "LINE1", + "LINE2", + "DMIC1", + "DMIC2", + "DMIC3", + "DMIC4", +}; + +/** + * Due to mux design for nid 24 (MUX_IN3)/25 (MUX_IN4), connection index 0 and + * 1 will be connected to the same dmic source, therefore we skip index 1 to + * avoid misunderstanding on usage of dapm routing. + */ +static const unsigned int rt715_adc_24_25_values[] = { + 0, + 2, + 3, + 4, + 5, +}; + +static const char * const adc_24_mux_text[] = { + "MIC2", + "DMIC1", + "DMIC2", + "DMIC3", + "DMIC4", +}; + +static const char * const adc_25_mux_text[] = { + "MIC1", + "DMIC1", + "DMIC2", + "DMIC3", + "DMIC4", +}; + +static SOC_ENUM_SINGLE_DECL( + rt715_adc22_enum, RT715_MUX_IN1, 0, adc_22_23_mux_text); + +static SOC_ENUM_SINGLE_DECL( + rt715_adc23_enum, RT715_MUX_IN2, 0, adc_22_23_mux_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(rt715_adc24_enum, + RT715_MUX_IN3, 0, 0xf, + adc_24_mux_text, rt715_adc_24_25_values); +static SOC_VALUE_ENUM_SINGLE_DECL(rt715_adc25_enum, + RT715_MUX_IN4, 0, 0xf, + adc_25_mux_text, rt715_adc_24_25_values); + +static const struct snd_kcontrol_new rt715_adc22_mux = + SOC_DAPM_ENUM_EXT("ADC 22 Mux", rt715_adc22_enum, + rt715_mux_get, rt715_mux_put); + +static const struct snd_kcontrol_new rt715_adc23_mux = + SOC_DAPM_ENUM_EXT("ADC 23 Mux", rt715_adc23_enum, + rt715_mux_get, rt715_mux_put); + +static const struct snd_kcontrol_new rt715_adc24_mux = + SOC_DAPM_ENUM_EXT("ADC 24 Mux", rt715_adc24_enum, + rt715_mux_get, rt715_mux_put); + +static const struct snd_kcontrol_new rt715_adc25_mux = + SOC_DAPM_ENUM_EXT("ADC 25 Mux", rt715_adc25_enum, + rt715_mux_get, rt715_mux_put); + +static const struct snd_soc_dapm_widget rt715_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("DMIC3"), + SND_SOC_DAPM_INPUT("DMIC4"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("LINE2"), + SND_SOC_DAPM_ADC("ADC 07", NULL, RT715_SET_STREAMID_MIC_ADC, 4, 0), + SND_SOC_DAPM_ADC("ADC 08", NULL, RT715_SET_STREAMID_LINE_ADC, 4, 0), + SND_SOC_DAPM_ADC("ADC 09", NULL, RT715_SET_STREAMID_MIX_ADC, 4, 0), + SND_SOC_DAPM_ADC("ADC 27", NULL, RT715_SET_STREAMID_MIX_ADC2, 4, 0), + SND_SOC_DAPM_MUX("ADC 22 Mux", SND_SOC_NOPM, 0, 0, + &rt715_adc22_mux), + SND_SOC_DAPM_MUX("ADC 23 Mux", SND_SOC_NOPM, 0, 0, + &rt715_adc23_mux), + SND_SOC_DAPM_MUX("ADC 24 Mux", SND_SOC_NOPM, 0, 0, + &rt715_adc24_mux), + SND_SOC_DAPM_MUX("ADC 25 Mux", SND_SOC_NOPM, 0, 0, + &rt715_adc25_mux), + SND_SOC_DAPM_AIF_OUT("DP4TX", "DP4 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP6TX", "DP6 Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route rt715_audio_map[] = { + {"DP6TX", NULL, "ADC 09"}, + {"DP6TX", NULL, "ADC 08"}, + {"DP4TX", NULL, "ADC 07"}, + {"DP4TX", NULL, "ADC 27"}, + {"ADC 09", NULL, "ADC 22 Mux"}, + {"ADC 08", NULL, "ADC 23 Mux"}, + {"ADC 07", NULL, "ADC 24 Mux"}, + {"ADC 27", NULL, "ADC 25 Mux"}, + {"ADC 22 Mux", "MIC1", "MIC1"}, + {"ADC 22 Mux", "MIC2", "MIC2"}, + {"ADC 22 Mux", "LINE1", "LINE1"}, + {"ADC 22 Mux", "LINE2", "LINE2"}, + {"ADC 22 Mux", "DMIC1", "DMIC1"}, + {"ADC 22 Mux", "DMIC2", "DMIC2"}, + {"ADC 22 Mux", "DMIC3", "DMIC3"}, + {"ADC 22 Mux", "DMIC4", "DMIC4"}, + {"ADC 23 Mux", "MIC1", "MIC1"}, + {"ADC 23 Mux", "MIC2", "MIC2"}, + {"ADC 23 Mux", "LINE1", "LINE1"}, + {"ADC 23 Mux", "LINE2", "LINE2"}, + {"ADC 23 Mux", "DMIC1", "DMIC1"}, + {"ADC 23 Mux", "DMIC2", "DMIC2"}, + {"ADC 23 Mux", "DMIC3", "DMIC3"}, + {"ADC 23 Mux", "DMIC4", "DMIC4"}, + {"ADC 24 Mux", "MIC2", "MIC2"}, + {"ADC 24 Mux", "DMIC1", "DMIC1"}, + {"ADC 24 Mux", "DMIC2", "DMIC2"}, + {"ADC 24 Mux", "DMIC3", "DMIC3"}, + {"ADC 24 Mux", "DMIC4", "DMIC4"}, + {"ADC 25 Mux", "MIC1", "MIC1"}, + {"ADC 25 Mux", "DMIC1", "DMIC1"}, + {"ADC 25 Mux", "DMIC2", "DMIC2"}, + {"ADC 25 Mux", "DMIC3", "DMIC3"}, + {"ADC 25 Mux", "DMIC4", "DMIC4"}, +}; + +static int rt715_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, + AC_PWRST_D0); + } + break; + + case SND_SOC_BIAS_STANDBY: + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, + AC_PWRST_D3); + break; + + default: + break; + } + dapm->bias_level = level; + return 0; +} + +static const struct snd_soc_component_driver soc_codec_dev_rt715 = { + .set_bias_level = rt715_set_bias_level, + .controls = rt715_snd_controls, + .num_controls = ARRAY_SIZE(rt715_snd_controls), + .dapm_widgets = rt715_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt715_dapm_widgets), + .dapm_routes = rt715_audio_map, + .num_dapm_routes = ARRAY_SIZE(rt715_audio_map), +}; + +static int rt715_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + + struct sdw_stream_data *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = (struct sdw_stream_runtime *)sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void rt715_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) + +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int rt715_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int retval, port, num_channels; + unsigned int val = 0; + + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!rt715->slave) + return -EINVAL; + + switch (dai->id) { + case RT715_AIF1: + direction = SDW_DATA_DIR_TX; + port = 6; + rt715_index_write(rt715->regmap, RT715_SDW_INPUT_SEL, 0xa500); + break; + case RT715_AIF2: + direction = SDW_DATA_DIR_TX; + port = 4; + rt715_index_write(rt715->regmap, RT715_SDW_INPUT_SEL, 0xa000); + break; + default: + dev_err(component->dev, "Invalid DAI id %d\n", dai->id); + return -EINVAL; + } + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = (1 << (num_channels)) - 1; + port_config.num = port; + + retval = sdw_stream_add_slave(rt715->slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + /* bit 15 Stream Type 0:PCM 1:Non-PCM, should always be PCM */ + case 44100: + val |= 0x40 << 8; + break; + case 48000: + val |= 0x0 << 8; + break; + default: + dev_err(component->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(component->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 8: + break; + case 16: + val |= (0x1 << 4); + break; + case 20: + val |= (0x2 << 4); + break; + case 24: + val |= (0x3 << 4); + break; + case 32: + val |= (0x4 << 4); + break; + default: + return -EINVAL; + } + + regmap_write(rt715->regmap, RT715_MIC_ADC_FORMAT_H, val); + regmap_write(rt715->regmap, RT715_MIC_LINE_FORMAT_H, val); + regmap_write(rt715->regmap, RT715_MIX_ADC_FORMAT_H, val); + regmap_write(rt715->regmap, RT715_MIX_ADC2_FORMAT_H, val); + + return retval; +} + +static int rt715_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt715->slave) + return -EINVAL; + + sdw_stream_remove_slave(rt715->slave, stream->sdw_stream); + return 0; +} + +#define RT715_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT715_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static struct snd_soc_dai_ops rt715_ops = { + .hw_params = rt715_pcm_hw_params, + .hw_free = rt715_pcm_hw_free, + .set_sdw_stream = rt715_set_sdw_stream, + .shutdown = rt715_shutdown, +}; + +static struct snd_soc_dai_driver rt715_dai[] = { + { + .name = "rt715-aif1", + .id = RT715_AIF1, + .capture = { + .stream_name = "DP6 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT715_STEREO_RATES, + .formats = RT715_FORMATS, + }, + .ops = &rt715_ops, + }, + { + .name = "rt715-aif2", + .id = RT715_AIF2, + .capture = { + .stream_name = "DP4 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT715_STEREO_RATES, + .formats = RT715_FORMATS, + }, + .ops = &rt715_ops, + }, +}; + +/* Bus clock frequency */ +#define RT715_CLK_FREQ_9600000HZ 9600000 +#define RT715_CLK_FREQ_12000000HZ 12000000 +#define RT715_CLK_FREQ_6000000HZ 6000000 +#define RT715_CLK_FREQ_4800000HZ 4800000 +#define RT715_CLK_FREQ_2400000HZ 2400000 +#define RT715_CLK_FREQ_12288000HZ 12288000 + +int rt715_clock_config(struct device *dev) +{ + struct rt715_priv *rt715 = dev_get_drvdata(dev); + unsigned int clk_freq, value; + + clk_freq = (rt715->params.curr_dr_freq >> 1); + + switch (clk_freq) { + case RT715_CLK_FREQ_12000000HZ: + value = 0x0; + break; + case RT715_CLK_FREQ_6000000HZ: + value = 0x1; + break; + case RT715_CLK_FREQ_9600000HZ: + value = 0x2; + break; + case RT715_CLK_FREQ_4800000HZ: + value = 0x3; + break; + case RT715_CLK_FREQ_2400000HZ: + value = 0x4; + break; + case RT715_CLK_FREQ_12288000HZ: + value = 0x5; + break; + default: + return -EINVAL; + } + + regmap_write(rt715->regmap, 0xe0, value); + regmap_write(rt715->regmap, 0xf0, value); + + return 0; +} + +int rt715_init(struct device *dev, struct regmap *sdw_regmap, + struct regmap *regmap, struct sdw_slave *slave) +{ + struct rt715_priv *rt715; + int ret; + + rt715 = devm_kzalloc(dev, sizeof(*rt715), GFP_KERNEL); + if (!rt715) + return -ENOMEM; + + dev_set_drvdata(dev, rt715); + rt715->slave = slave; + rt715->regmap = regmap; + rt715->sdw_regmap = sdw_regmap; + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt715->hw_init = false; + rt715->first_hw_init = false; + + ret = devm_snd_soc_register_component(dev, + &soc_codec_dev_rt715, + rt715_dai, + ARRAY_SIZE(rt715_dai)); + + return ret; +} + +int rt715_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt715_priv *rt715 = dev_get_drvdata(dev); + + if (rt715->hw_init) + return 0; + + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + if (!rt715->first_hw_init) { + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + /* Mute nid=08h/09h */ + regmap_write(rt715->regmap, RT715_SET_GAIN_LINE_ADC_H, 0xb080); + regmap_write(rt715->regmap, RT715_SET_GAIN_MIX_ADC_H, 0xb080); + /* Mute nid=07h/27h */ + regmap_write(rt715->regmap, RT715_SET_GAIN_MIC_ADC_H, 0xb080); + regmap_write(rt715->regmap, RT715_SET_GAIN_MIX_ADC2_H, 0xb080); + + /* Set Pin Widget */ + regmap_write(rt715->regmap, RT715_SET_PIN_DMIC1, 0x20); + regmap_write(rt715->regmap, RT715_SET_PIN_DMIC2, 0x20); + regmap_write(rt715->regmap, RT715_SET_PIN_DMIC3, 0x20); + regmap_write(rt715->regmap, RT715_SET_PIN_DMIC4, 0x20); + /* Set Converter Stream */ + regmap_write(rt715->regmap, RT715_SET_STREAMID_LINE_ADC, 0x10); + regmap_write(rt715->regmap, RT715_SET_STREAMID_MIX_ADC, 0x10); + regmap_write(rt715->regmap, RT715_SET_STREAMID_MIC_ADC, 0x10); + regmap_write(rt715->regmap, RT715_SET_STREAMID_MIX_ADC2, 0x10); + /* Set Configuration Default */ + regmap_write(rt715->regmap, RT715_SET_DMIC1_CONFIG_DEFAULT1, 0xd0); + regmap_write(rt715->regmap, RT715_SET_DMIC1_CONFIG_DEFAULT2, 0x11); + regmap_write(rt715->regmap, RT715_SET_DMIC1_CONFIG_DEFAULT3, 0xa1); + regmap_write(rt715->regmap, RT715_SET_DMIC1_CONFIG_DEFAULT4, 0x81); + regmap_write(rt715->regmap, RT715_SET_DMIC2_CONFIG_DEFAULT1, 0xd1); + regmap_write(rt715->regmap, RT715_SET_DMIC2_CONFIG_DEFAULT2, 0x11); + regmap_write(rt715->regmap, RT715_SET_DMIC2_CONFIG_DEFAULT3, 0xa1); + regmap_write(rt715->regmap, RT715_SET_DMIC2_CONFIG_DEFAULT4, 0x81); + regmap_write(rt715->regmap, RT715_SET_DMIC3_CONFIG_DEFAULT1, 0xd0); + regmap_write(rt715->regmap, RT715_SET_DMIC3_CONFIG_DEFAULT2, 0x11); + regmap_write(rt715->regmap, RT715_SET_DMIC3_CONFIG_DEFAULT3, 0xa1); + regmap_write(rt715->regmap, RT715_SET_DMIC3_CONFIG_DEFAULT4, 0x81); + regmap_write(rt715->regmap, RT715_SET_DMIC4_CONFIG_DEFAULT1, 0xd1); + regmap_write(rt715->regmap, RT715_SET_DMIC4_CONFIG_DEFAULT2, 0x11); + regmap_write(rt715->regmap, RT715_SET_DMIC4_CONFIG_DEFAULT3, 0xa1); + regmap_write(rt715->regmap, RT715_SET_DMIC4_CONFIG_DEFAULT4, 0x81); + + /* Finish Initial Settings, set power to D3 */ + regmap_write(rt715->regmap, RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + + if (rt715->first_hw_init) + regcache_mark_dirty(rt715->regmap); + else + rt715->first_hw_init = true; + + /* Mark Slave initialization complete */ + rt715->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + return 0; +} + +MODULE_DESCRIPTION("ASoC rt715 driver"); +MODULE_DESCRIPTION("ASoC rt715 driver SDW"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt715.h b/sound/soc/codecs/rt715.h new file mode 100644 index 000000000000..df0f24f9bc0c --- /dev/null +++ b/sound/soc/codecs/rt715.h @@ -0,0 +1,221 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * rt715.h -- RT715 ALSA SoC audio driver header + * + * Copyright(c) 2019 Realtek Semiconductor Corp. + */ + +#ifndef __RT715_H__ +#define __RT715_H__ + +#include <linux/regulator/consumer.h> + +struct rt715_priv { + struct regmap *regmap; + struct regmap *sdw_regmap; + struct snd_soc_codec *codec; + struct sdw_slave *slave; + int dbg_nid; + int dbg_vid; + int dbg_payload; + enum sdw_slave_status status; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +/* NID */ +#define RT715_AUDIO_FUNCTION_GROUP 0x01 +#define RT715_MIC_ADC 0x07 +#define RT715_LINE_ADC 0x08 +#define RT715_MIX_ADC 0x09 +#define RT715_DMIC1 0x12 +#define RT715_DMIC2 0x13 +#define RT715_MIC1 0x18 +#define RT715_MIC2 0x19 +#define RT715_LINE1 0x1a +#define RT715_LINE2 0x1b +#define RT715_DMIC3 0x1d +#define RT715_DMIC4 0x29 +#define RT715_VENDOR_REGISTERS 0x20 +#define RT715_MUX_IN1 0x22 +#define RT715_MUX_IN2 0x23 +#define RT715_MUX_IN3 0x24 +#define RT715_MUX_IN4 0x25 +#define RT715_MIX_ADC2 0x27 +#define RT715_INLINE_CMD 0x55 + +/* Index (NID:20h) */ +#define RT715_SDW_INPUT_SEL 0x39 +#define RT715_EXT_DMIC_CLK_CTRL2 0x54 + +/* Verb */ +#define RT715_VERB_SET_CONNECT_SEL 0x3100 +#define RT715_VERB_GET_CONNECT_SEL 0xb100 +#define RT715_VERB_SET_EAPD_BTLENABLE 0x3c00 +#define RT715_VERB_SET_POWER_STATE 0x3500 +#define RT715_VERB_SET_CHANNEL_STREAMID 0x3600 +#define RT715_VERB_SET_PIN_WIDGET_CONTROL 0x3700 +#define RT715_VERB_SET_CONFIG_DEFAULT1 0x4c00 +#define RT715_VERB_SET_CONFIG_DEFAULT2 0x4d00 +#define RT715_VERB_SET_CONFIG_DEFAULT3 0x4e00 +#define RT715_VERB_SET_CONFIG_DEFAULT4 0x4f00 +#define RT715_VERB_SET_UNSOLICITED_ENABLE 0x3800 +#define RT715_SET_AMP_GAIN_MUTE_H 0x7300 +#define RT715_SET_AMP_GAIN_MUTE_L 0x8380 +#define RT715_READ_HDA_3 0x2012 +#define RT715_READ_HDA_2 0x2013 +#define RT715_READ_HDA_1 0x2014 +#define RT715_READ_HDA_0 0x2015 +#define RT715_PRIV_INDEX_W_H 0x7520 +#define RT715_PRIV_INDEX_W_L 0x85a0 +#define RT715_PRIV_DATA_W_H 0x7420 +#define RT715_PRIV_DATA_W_L 0x84a0 +#define RT715_PRIV_INDEX_R_H 0x9d20 +#define RT715_PRIV_INDEX_R_L 0xada0 +#define RT715_PRIV_DATA_R_H 0x9c20 +#define RT715_PRIV_DATA_R_L 0xaca0 +#define RT715_MIC_ADC_FORMAT_H 0x7207 +#define RT715_MIC_ADC_FORMAT_L 0x8287 +#define RT715_MIC_LINE_FORMAT_H 0x7208 +#define RT715_MIC_LINE_FORMAT_L 0x8288 +#define RT715_MIX_ADC_FORMAT_H 0x7209 +#define RT715_MIX_ADC_FORMAT_L 0x8289 +#define RT715_MIX_ADC2_FORMAT_H 0x7227 +#define RT715_MIX_ADC2_FORMAT_L 0x82a7 +#define RT715_FUNC_RESET 0xff01 + +#define RT715_SET_AUDIO_POWER_STATE\ + (RT715_VERB_SET_POWER_STATE | RT715_AUDIO_FUNCTION_GROUP) +#define RT715_SET_PIN_DMIC1\ + (RT715_VERB_SET_PIN_WIDGET_CONTROL | RT715_DMIC1) +#define RT715_SET_PIN_DMIC2\ + (RT715_VERB_SET_PIN_WIDGET_CONTROL | RT715_DMIC2) +#define RT715_SET_PIN_DMIC3\ + (RT715_VERB_SET_PIN_WIDGET_CONTROL | RT715_DMIC3) +#define RT715_SET_PIN_DMIC4\ + (RT715_VERB_SET_PIN_WIDGET_CONTROL | RT715_DMIC4) +#define RT715_SET_PIN_MIC1\ + (RT715_VERB_SET_PIN_WIDGET_CONTROL | RT715_MIC1) +#define RT715_SET_PIN_MIC2\ + (RT715_VERB_SET_PIN_WIDGET_CONTROL | RT715_MIC2) +#define RT715_SET_PIN_LINE1\ + (RT715_VERB_SET_PIN_WIDGET_CONTROL | RT715_LINE1) +#define RT715_SET_PIN_LINE2\ + (RT715_VERB_SET_PIN_WIDGET_CONTROL | RT715_LINE2) +#define RT715_SET_MIC1_UNSOLICITED_ENABLE\ + (RT715_VERB_SET_UNSOLICITED_ENABLE | RT715_MIC1) +#define RT715_SET_MIC2_UNSOLICITED_ENABLE\ + (RT715_VERB_SET_UNSOLICITED_ENABLE | RT715_MIC2) +#define RT715_SET_STREAMID_MIC_ADC\ + (RT715_VERB_SET_CHANNEL_STREAMID | RT715_MIC_ADC) +#define RT715_SET_STREAMID_LINE_ADC\ + (RT715_VERB_SET_CHANNEL_STREAMID | RT715_LINE_ADC) +#define RT715_SET_STREAMID_MIX_ADC\ + (RT715_VERB_SET_CHANNEL_STREAMID | RT715_MIX_ADC) +#define RT715_SET_STREAMID_MIX_ADC2\ + (RT715_VERB_SET_CHANNEL_STREAMID | RT715_MIX_ADC2) +#define RT715_SET_GAIN_MIC_ADC_L\ + (RT715_SET_AMP_GAIN_MUTE_L | RT715_MIC_ADC) +#define RT715_SET_GAIN_MIC_ADC_H\ + (RT715_SET_AMP_GAIN_MUTE_H | RT715_MIC_ADC) +#define RT715_SET_GAIN_LINE_ADC_L\ + (RT715_SET_AMP_GAIN_MUTE_L | RT715_LINE_ADC) +#define RT715_SET_GAIN_LINE_ADC_H\ + (RT715_SET_AMP_GAIN_MUTE_H | RT715_LINE_ADC) +#define RT715_SET_GAIN_MIX_ADC_L\ + (RT715_SET_AMP_GAIN_MUTE_L | RT715_MIX_ADC) +#define RT715_SET_GAIN_MIX_ADC_H\ + (RT715_SET_AMP_GAIN_MUTE_H | RT715_MIX_ADC) +#define RT715_SET_GAIN_MIX_ADC2_L\ + (RT715_SET_AMP_GAIN_MUTE_L | RT715_MIX_ADC2) +#define RT715_SET_GAIN_MIX_ADC2_H\ + (RT715_SET_AMP_GAIN_MUTE_H | RT715_MIX_ADC2) +#define RT715_SET_GAIN_DMIC1_L\ + (RT715_SET_AMP_GAIN_MUTE_L | RT715_DMIC1) +#define RT715_SET_GAIN_DMIC1_H\ + (RT715_SET_AMP_GAIN_MUTE_H | RT715_DMIC1) +#define RT715_SET_GAIN_DMIC2_L\ + (RT715_SET_AMP_GAIN_MUTE_L | RT715_DMIC2) +#define RT715_SET_GAIN_DMIC2_H\ + (RT715_SET_AMP_GAIN_MUTE_H | RT715_DMIC2) +#define RT715_SET_GAIN_DMIC3_L\ + (RT715_SET_AMP_GAIN_MUTE_L | RT715_DMIC3) +#define RT715_SET_GAIN_DMIC3_H\ + (RT715_SET_AMP_GAIN_MUTE_H | RT715_DMIC3) +#define RT715_SET_GAIN_DMIC4_L\ + (RT715_SET_AMP_GAIN_MUTE_L | RT715_DMIC4) +#define RT715_SET_GAIN_DMIC4_H\ + (RT715_SET_AMP_GAIN_MUTE_H | RT715_DMIC4) +#define RT715_SET_GAIN_MIC1_L\ + (RT715_SET_AMP_GAIN_MUTE_L | RT715_MIC1) +#define RT715_SET_GAIN_MIC1_H\ + (RT715_SET_AMP_GAIN_MUTE_H | RT715_MIC1) +#define RT715_SET_GAIN_MIC2_L\ + (RT715_SET_AMP_GAIN_MUTE_L | RT715_MIC2) +#define RT715_SET_GAIN_MIC2_H\ + (RT715_SET_AMP_GAIN_MUTE_H | RT715_MIC2) +#define RT715_SET_GAIN_LINE1_L\ + (RT715_SET_AMP_GAIN_MUTE_L | RT715_LINE1) +#define RT715_SET_GAIN_LINE1_H\ + (RT715_SET_AMP_GAIN_MUTE_H | RT715_LINE1) +#define RT715_SET_GAIN_LINE2_L\ + (RT715_SET_AMP_GAIN_MUTE_L | RT715_LINE2) +#define RT715_SET_GAIN_LINE2_H\ + (RT715_SET_AMP_GAIN_MUTE_H | RT715_LINE2) +#define RT715_SET_DMIC1_CONFIG_DEFAULT1\ + (RT715_VERB_SET_CONFIG_DEFAULT1 | RT715_DMIC1) +#define RT715_SET_DMIC2_CONFIG_DEFAULT1\ + (RT715_VERB_SET_CONFIG_DEFAULT1 | RT715_DMIC2) +#define RT715_SET_DMIC1_CONFIG_DEFAULT2\ + (RT715_VERB_SET_CONFIG_DEFAULT2 | RT715_DMIC1) +#define RT715_SET_DMIC2_CONFIG_DEFAULT2\ + (RT715_VERB_SET_CONFIG_DEFAULT2 | RT715_DMIC2) +#define RT715_SET_DMIC1_CONFIG_DEFAULT3\ + (RT715_VERB_SET_CONFIG_DEFAULT3 | RT715_DMIC1) +#define RT715_SET_DMIC2_CONFIG_DEFAULT3\ + (RT715_VERB_SET_CONFIG_DEFAULT3 | RT715_DMIC2) +#define RT715_SET_DMIC1_CONFIG_DEFAULT4\ + (RT715_VERB_SET_CONFIG_DEFAULT4 | RT715_DMIC1) +#define RT715_SET_DMIC2_CONFIG_DEFAULT4\ + (RT715_VERB_SET_CONFIG_DEFAULT4 | RT715_DMIC2) +#define RT715_SET_DMIC3_CONFIG_DEFAULT1\ + (RT715_VERB_SET_CONFIG_DEFAULT1 | RT715_DMIC3) +#define RT715_SET_DMIC4_CONFIG_DEFAULT1\ + (RT715_VERB_SET_CONFIG_DEFAULT1 | RT715_DMIC4) +#define RT715_SET_DMIC3_CONFIG_DEFAULT2\ + (RT715_VERB_SET_CONFIG_DEFAULT2 | RT715_DMIC3) +#define RT715_SET_DMIC4_CONFIG_DEFAULT2\ + (RT715_VERB_SET_CONFIG_DEFAULT2 | RT715_DMIC4) +#define RT715_SET_DMIC3_CONFIG_DEFAULT3\ + (RT715_VERB_SET_CONFIG_DEFAULT3 | RT715_DMIC3) +#define RT715_SET_DMIC4_CONFIG_DEFAULT3\ + (RT715_VERB_SET_CONFIG_DEFAULT3 | RT715_DMIC4) +#define RT715_SET_DMIC3_CONFIG_DEFAULT4\ + (RT715_VERB_SET_CONFIG_DEFAULT4 | RT715_DMIC3) +#define RT715_SET_DMIC4_CONFIG_DEFAULT4\ + (RT715_VERB_SET_CONFIG_DEFAULT4 | RT715_DMIC4) + +#define RT715_MUTE_SFT 7 +#define RT715_DIR_IN_SFT 6 +#define RT715_DIR_OUT_SFT 7 + +enum { + RT715_AIF1, + RT715_AIF2, + RT715_AIFS, +}; + +int rt715_io_init(struct device *dev, struct sdw_slave *slave); +int rt715_init(struct device *dev, struct regmap *sdw_regmap, + struct regmap *regmap, struct sdw_slave *slave); + +int hda_to_sdw(unsigned int nid, unsigned int verb, unsigned int payload, + unsigned int *sdw_addr_h, unsigned int *sdw_data_h, + unsigned int *sdw_addr_l, unsigned int *sdw_data_l); +int rt715_clock_config(struct device *dev); +#endif /* __RT715_H__ */ -- 2.24.1

2 1

[alsa-devel] Applied "ASoC: SOF: Intel: fix HDA codec driver probe with multiple controllers" to the asoc tree
by Mark Brown 13 Jan '20

13 Jan '20

The patch ASoC: SOF: Intel: fix HDA codec driver probe with multiple controllers has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 2c63bea714780f8e1fc9cb7bc10deda26fada25b Mon Sep 17 00:00:00 2001 From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Date: Fri, 10 Jan 2020 17:57:50 -0600 Subject: [PATCH] ASoC: SOF: Intel: fix HDA codec driver probe with multiple controllers In case system has multiple HDA controllers, it can happen that same HDA codec driver is used for codecs of multiple controllers. In this case, SOF may fail to probe the HDA driver and SOF initialization fails. SOF HDA code currently relies that a call to request_module() will also run device matching logic to attach driver to the codec instance. However if driver for another HDA controller was already loaded and it already loaded the HDA codec driver, this breaks current logic in SOF. In this case the request_module() SOF does becomes a no-op and HDA Codec driver is not attached to the codec instance sitting on the HDA bus SOF is controlling. Typical scenario would be a system with both external and internal GPUs, with driver of the external GPU loaded first. Fix this by adding similar logic as is used in legacy HDA driver where an explicit device_attach() call is done after request_module(). Also add logic to propagate errors reported by device_attach() back to caller. This also works in the case where drivers are not built as modules. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200110235751.3404-8-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-codec.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 827f84a0722e..fbfa225d1c5a 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -24,19 +24,18 @@ #define IDISP_VID_INTEL 0x80860000 /* load the legacy HDA codec driver */ -#ifdef MODULE -static void hda_codec_load_module(struct hda_codec *codec) +static int hda_codec_load_module(struct hda_codec *codec) { +#ifdef MODULE char alias[MODULE_NAME_LEN]; const char *module = alias; snd_hdac_codec_modalias(&codec->core, alias, sizeof(alias)); dev_dbg(&codec->core.dev, "loading codec module: %s\n", module); request_module(module); -} -#else -static void hda_codec_load_module(struct hda_codec *codec) {} #endif + return device_attach(hda_codec_dev(codec)); +} /* enable controller wake up event for all codecs with jack connectors */ void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) @@ -129,10 +128,16 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address) if ((mach_params && mach_params->common_hdmi_codec_drv) || (resp & 0xFFFF0000) != IDISP_VID_INTEL) { hdev->type = HDA_DEV_LEGACY; - hda_codec_load_module(&hda_priv->codec); + ret = hda_codec_load_module(&hda_priv->codec); + /* + * handle ret==0 (no driver bound) as an error, but pass + * other return codes without modification + */ + if (ret == 0) + ret = -ENOENT; } - return 0; + return ret; #else hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL); if (!hdev) -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: sti: fix possible sleep-in-atomic" to the asoc tree
by Mark Brown 13 Jan '20

13 Jan '20

The patch ASoC: sti: fix possible sleep-in-atomic has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ce780a47c3c01e1e179d0792df6b853a913928f1 Mon Sep 17 00:00:00 2001 From: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Date: Mon, 13 Jan 2020 11:04:00 +0100 Subject: [PATCH] ASoC: sti: fix possible sleep-in-atomic Change mutex and spinlock management to avoid sleep in atomic issue. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Link: https://lore.kernel.org/r/20200113100400.30472-1-arnaud.pouliquen@st.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sti/uniperif_player.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 48ea915b24ba..2ed92c990b97 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -226,7 +226,6 @@ static void uni_player_set_channel_status(struct uniperif *player, * sampling frequency. If no sample rate is already specified, then * set one. */ - mutex_lock(&player->ctrl_lock); if (runtime) { switch (runtime->rate) { case 22050: @@ -303,7 +302,6 @@ static void uni_player_set_channel_status(struct uniperif *player, player->stream_settings.iec958.status[3 + (n * 4)] << 24; SET_UNIPERIF_CHANNEL_STA_REGN(player, n, status); } - mutex_unlock(&player->ctrl_lock); /* Update the channel status */ if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) @@ -365,8 +363,10 @@ static int uni_player_prepare_iec958(struct uniperif *player, SET_UNIPERIF_CTRL_ZERO_STUFF_HW(player); + mutex_lock(&player->ctrl_lock); /* Update the channel status */ uni_player_set_channel_status(player, runtime); + mutex_unlock(&player->ctrl_lock); /* Clear the user validity user bits */ SET_UNIPERIF_USER_VALIDITY_VALIDITY_LR(player, 0); @@ -598,7 +598,6 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, iec958->status[1] = ucontrol->value.iec958.status[1]; iec958->status[2] = ucontrol->value.iec958.status[2]; iec958->status[3] = ucontrol->value.iec958.status[3]; - mutex_unlock(&player->ctrl_lock); spin_lock_irqsave(&player->irq_lock, flags); if (player->substream && player->substream->runtime) @@ -608,6 +607,8 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, uni_player_set_channel_status(player, NULL); spin_unlock_irqrestore(&player->irq_lock, flags); + mutex_unlock(&player->ctrl_lock); + return 0; } -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: cros_ec_codec: Make the device acpi compatible
by Yu-Hsuan Hsu 13 Jan '20

13 Jan '20

Add ACPI entry for cros_ec_codec. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/cros_ec_codec.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index c81bbef2367a0f..6a24f570c5e86f 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -10,6 +10,7 @@ #include <crypto/hash.h> #include <crypto/sha.h> +#include <linux/acpi.h> #include <linux/delay.h> #include <linux/device.h> #include <linux/io.h> @@ -1047,10 +1048,17 @@ static const struct of_device_id cros_ec_codec_of_match[] = { MODULE_DEVICE_TABLE(of, cros_ec_codec_of_match); #endif +static const struct acpi_device_id cros_ec_codec_acpi_id[] = { + { "GOOG0013", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, cros_ec_codec_acpi_id); + static struct platform_driver cros_ec_codec_platform_driver = { .driver = { .name = "cros-ec-codec", .of_match_table = of_match_ptr(cros_ec_codec_of_match), + .acpi_match_table = ACPI_PTR(cros_ec_codec_acpi_id), }, .probe = cros_ec_codec_platform_probe, }; -- 2.25.0.rc1.283.g88dfdc4193-goog

2 1

[alsa-devel] [PATCH] ASoC: msm8916-wcd-digital: Reset RX interpolation path after use
by Stephan Gerhold 13 Jan '20

13 Jan '20

For some reason, attempting to route audio through QDSP6 on MSM8916 causes the RX interpolation path to get "stuck" after playing audio a few times. In this situation, the analog codec part is still working, but the RX path in the digital codec stops working, so you only hear the analog parts powering up. After a reboot everything works again. So far I was not able to reproduce the problem when using lpass-cpu. The downstream kernel driver avoids this by resetting the RX interpolation path after use. In mainline we do something similar for the TX decimator (LPASS_CDC_CLK_TX_RESET_B1_CTL), but the interpolator reset (LPASS_CDC_CLK_RX_RESET_CTL) got lost when the msm8916-wcd driver was split into analog and digital. Fix this problem by adding the reset to msm8916_wcd_digital_enable_interpolator(). Fixes: 150db8c5afa1 ("ASoC: codecs: Add msm8916-wcd digital codec") Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Tested on msm8916-samsung-a5u: - qdsp6 does no longer stop working after playing audio a few times - lpass-cpu is still working fine (no difference) --- sound/soc/codecs/msm8916-wcd-digital.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 58b2468fb2a7..09fccacadd6b 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -586,6 +586,12 @@ static int msm8916_wcd_digital_enable_interpolator( snd_soc_component_write(component, rx_gain_reg[w->shift], snd_soc_component_read32(component, rx_gain_reg[w->shift])); break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, LPASS_CDC_CLK_RX_RESET_CTL, + 1 << w->shift, 1 << w->shift); + snd_soc_component_update_bits(component, LPASS_CDC_CLK_RX_RESET_CTL, + 1 << w->shift, 0x0); + break; } return 0; } -- 2.24.1

3 4

[alsa-devel] [PATCH for-5.6] ALSA: hda: Manage concurrent reg access more properly
by Takashi Iwai 13 Jan '20

13 Jan '20

In the commit 8e85def5723e ("ALSA: hda: enable regmap internal locking"), we re-enabled the regmap lock due to the reported regression that showed the possible concurrent accesses. It was a temporary workaround, and there are still a few opened races even after the revert. In this patch, we cover those still opened windows with a proper mutex lock and disable the regmap internal lock again. First off, the patch introduces a new snd_hdac_device.regmap_lock mutex that is applied for each snd_hdac_regmap_*() call, including read, write and update helpers. The mutex is applied carefully so that it won't block the self-power-up procedure in the helper function. Also, this assures the protection for the accesses without regmap, too. The snd_hdac_regmap_update_raw() is refactored to use the standard regmap_update_bits_check() function instead of the open-code. The non-regmap case is still open-coded but it's an easy part. The all read and write operations are in the single mutex protection, so it's now race-free. In addition, a couple of new helper functions are added: snd_hdac_regmap_update_raw_once() and snd_hdac_regmap_sync(). Both are called from HD-audio legacy driver. The former is to initialize the given verb bits but only once when it's not initialized yet. Due to this condition, the function invokes regcache_cache_only(), and it's now performed inside the regmap_lock (formerly it was racy) too. The latter function is for simply invoking regcache_sync() inside the regmap_lock, which is called from the codec resume call path. Along with that, the HD-audio codec driver code is slightly modified / simplified to adapt those new functions. And finally, snd_hdac_regmap_read_raw(), *_write_raw(), etc are rewritten with the helper macro. It's just for simplification because the code logic is identical among all those functions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_regmap.h | 3 + include/sound/hdaudio.h | 1 + sound/hda/hdac_device.c | 1 + sound/hda/hdac_regmap.c | 142 +++++++++++++++++++++++++++++++----------- sound/pci/hda/hda_codec.c | 30 ++++----- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/hda_local.h | 2 + sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/hda/patch_realtek.c | 4 +- sound/pci/hda/patch_via.c | 2 +- 10 files changed, 135 insertions(+), 54 deletions(-) diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index 5141f8ffbb12..4c1b9bebbd60 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -24,6 +24,9 @@ int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, unsigned int val); int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg, unsigned int mask, unsigned int val); +int snd_hdac_regmap_update_raw_once(struct hdac_device *codec, unsigned int reg, + unsigned int mask, unsigned int val); +void snd_hdac_regmap_sync(struct hdac_device *codec); /** * snd_hdac_regmap_encode_verb - encode the verb to a pseudo register diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index bc2f77a6f17b..541ca99b154b 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -86,6 +86,7 @@ struct hdac_device { /* regmap */ struct regmap *regmap; + struct mutex regmap_lock; struct snd_array vendor_verbs; bool lazy_cache:1; /* don't wake up for writes */ bool caps_overwriting:1; /* caps overwrite being in process */ diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b4f8725f5ddf..7bcdb0e454f2 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -57,6 +57,7 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, codec->addr = addr; codec->type = HDA_DEV_CORE; mutex_init(&codec->widget_lock); + mutex_init(&codec->regmap_lock); pm_runtime_set_active(&codec->dev); pm_runtime_get_noresume(&codec->dev); atomic_set(&codec->in_pm, 0); diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 0c8188a48a00..d75f31eb9d78 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -363,6 +363,7 @@ static const struct regmap_config hda_regmap_cfg = { .reg_write = hda_reg_write, .use_single_read = true, .use_single_write = true, + .disable_locking = true, }; /** @@ -425,12 +426,29 @@ EXPORT_SYMBOL_GPL(snd_hdac_regmap_add_vendor_verb); static int reg_raw_write(struct hdac_device *codec, unsigned int reg, unsigned int val) { + int err; + + mutex_lock(&codec->regmap_lock); if (!codec->regmap) - return hda_reg_write(codec, reg, val); + err = hda_reg_write(codec, reg, val); else - return regmap_write(codec->regmap, reg, val); + err = regmap_write(codec->regmap, reg, val); + mutex_unlock(&codec->regmap_lock); + return err; } +/* a helper macro to call @func_call; retry with power-up if failed */ +#define CALL_RAW_FUNC(codec, func_call) \ + ({ \ + int _err = func_call; \ + if (_err == -EAGAIN) { \ + _err = snd_hdac_power_up_pm(codec); \ + if (_err >= 0) \ + _err = func_call; \ + snd_hdac_power_down_pm(codec); \ + } \ + _err;}) + /** * snd_hdac_regmap_write_raw - write a pseudo register with power mgmt * @codec: the codec object @@ -442,42 +460,29 @@ static int reg_raw_write(struct hdac_device *codec, unsigned int reg, int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, unsigned int val) { - int err; - - err = reg_raw_write(codec, reg, val); - if (err == -EAGAIN) { - err = snd_hdac_power_up_pm(codec); - if (err >= 0) - err = reg_raw_write(codec, reg, val); - snd_hdac_power_down_pm(codec); - } - return err; + return CALL_RAW_FUNC(codec, reg_raw_write(codec, reg, val)); } EXPORT_SYMBOL_GPL(snd_hdac_regmap_write_raw); static int reg_raw_read(struct hdac_device *codec, unsigned int reg, unsigned int *val, bool uncached) { + int err; + + mutex_lock(&codec->regmap_lock); if (uncached || !codec->regmap) - return hda_reg_read(codec, reg, val); + err = hda_reg_read(codec, reg, val); else - return regmap_read(codec->regmap, reg, val); + err = regmap_read(codec->regmap, reg, val); + mutex_unlock(&codec->regmap_lock); + return err; } static int __snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg, unsigned int *val, bool uncached) { - int err; - - err = reg_raw_read(codec, reg, val, uncached); - if (err == -EAGAIN) { - err = snd_hdac_power_up_pm(codec); - if (err >= 0) - err = reg_raw_read(codec, reg, val, uncached); - snd_hdac_power_down_pm(codec); - } - return err; + return CALL_RAW_FUNC(codec, reg_raw_read(codec, reg, val, uncached)); } /** @@ -504,6 +509,35 @@ int snd_hdac_regmap_read_raw_uncached(struct hdac_device *codec, return __snd_hdac_regmap_read_raw(codec, reg, val, true); } +static int reg_raw_update(struct hdac_device *codec, unsigned int reg, + unsigned int mask, unsigned int val) +{ + unsigned int orig; + bool change; + int err; + + mutex_lock(&codec->regmap_lock); + if (codec->regmap) { + err = regmap_update_bits_check(codec->regmap, reg, mask, val, + &change); + if (!err) + err = change ? 1 : 0; + } else { + err = hda_reg_read(codec, reg, &orig); + if (!err) { + val &= mask; + val |= orig & ~mask; + if (val != orig) { + err = hda_reg_write(codec, reg, val); + if (!err) + err = 1; + } + } + } + mutex_unlock(&codec->regmap_lock); + return err; +} + /** * snd_hdac_regmap_update_raw - update a pseudo register with power mgmt * @codec: the codec object @@ -515,20 +549,58 @@ int snd_hdac_regmap_read_raw_uncached(struct hdac_device *codec, */ int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg, unsigned int mask, unsigned int val) +{ + return CALL_RAW_FUNC(codec, reg_raw_update(codec, reg, mask, val)); +} +EXPORT_SYMBOL_GPL(snd_hdac_regmap_update_raw); + +static int reg_raw_update_once(struct hdac_device *codec, unsigned int reg, + unsigned int mask, unsigned int val) { unsigned int orig; int err; - val &= mask; - err = snd_hdac_regmap_read_raw(codec, reg, &orig); - if (err < 0) - return err; - val |= orig & ~mask; - if (val == orig) - return 0; - err = snd_hdac_regmap_write_raw(codec, reg, val); + if (!codec->regmap) + return reg_raw_update(codec, reg, mask, val); + + mutex_lock(&codec->regmap_lock); + regcache_cache_only(codec->regmap, true); + err = regmap_read(codec->regmap, reg, &orig); + regcache_cache_only(codec->regmap, false); if (err < 0) - return err; - return 1; + err = regmap_update_bits(codec->regmap, reg, mask, val); + mutex_unlock(&codec->regmap_lock); + return err; } -EXPORT_SYMBOL_GPL(snd_hdac_regmap_update_raw); + +/** + * snd_hdac_regmap_update_raw_once - initialize the register value only once + * @codec: the codec object + * @reg: pseudo register + * @mask: bit mask to update + * @val: value to update + * + * Performs the update of the register bits only once when the register + * hasn't been initialized yet. Used in HD-audio legacy driver. + * Returns zero if successful or a negative error code + */ +int snd_hdac_regmap_update_raw_once(struct hdac_device *codec, unsigned int reg, + unsigned int mask, unsigned int val) +{ + return CALL_RAW_FUNC(codec, reg_raw_update_once(codec, reg, mask, val)); +} +EXPORT_SYMBOL_GPL(snd_hdac_regmap_update_raw_once); + +/** + * snd_hdac_regmap_sync - sync out the cached values for PM resume + * @codec: the codec object + */ +void snd_hdac_regmap_sync(struct hdac_device *codec) +{ + if (codec->regmap) { + mutex_lock(&codec->regmap_lock); + regcache_sync(codec->regmap); + mutex_unlock(&codec->regmap_lock); + } +} +EXPORT_SYMBOL_GPL(snd_hdac_regmap_sync); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4377b2aba835..83aa4c1015d2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1267,6 +1267,18 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, } EXPORT_SYMBOL_GPL(snd_hda_override_amp_caps); +static unsigned int encode_amp(struct hda_codec *codec, hda_nid_t nid, + int ch, int dir, int idx) +{ + unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx); + + /* enable fake mute if no h/w mute but min=mute */ + if ((query_amp_caps(codec, nid, dir) & + (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) == AC_AMPCAP_MIN_MUTE) + cmd |= AC_AMP_FAKE_MUTE; + return cmd; +} + /** * snd_hda_codec_amp_update - update the AMP mono value * @codec: HD-audio codec @@ -1282,12 +1294,8 @@ EXPORT_SYMBOL_GPL(snd_hda_override_amp_caps); int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int dir, int idx, int mask, int val) { - unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx); + unsigned int cmd = encode_amp(codec, nid, ch, dir, idx); - /* enable fake mute if no h/w mute but min=mute */ - if ((query_amp_caps(codec, nid, dir) & - (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) == AC_AMPCAP_MIN_MUTE) - cmd |= AC_AMP_FAKE_MUTE; return snd_hdac_regmap_update_raw(&codec->core, cmd, mask, val); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update); @@ -1335,16 +1343,11 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_stereo); int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch, int dir, int idx, int mask, int val) { - int orig; + unsigned int cmd = encode_amp(codec, nid, ch, dir, idx); if (!codec->core.regmap) return -EINVAL; - regcache_cache_only(codec->core.regmap, true); - orig = snd_hda_codec_amp_read(codec, nid, ch, dir, idx); - regcache_cache_only(codec->core.regmap, false); - if (orig >= 0) - return 0; - return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, mask, val); + return snd_hdac_regmap_update_raw_once(&codec->core, cmd, mask, val); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init); @@ -2905,8 +2908,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) else { if (codec->patch_ops.init) codec->patch_ops.init(codec); - if (codec->core.regmap) - regcache_sync(codec->core.regmap); + snd_hda_regmap_sync(codec); } if (codec->jackpoll_interval) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index fc001c64ef20..6815f9dc8545 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -6027,7 +6027,7 @@ int snd_hda_gen_init(struct hda_codec *codec) /* call init functions of standard auto-mute helpers */ update_automute_all(codec); - regcache_sync(codec->core.regmap); + snd_hda_regmap_sync(codec); if (spec->vmaster_mute.sw_kctl && spec->vmaster_mute.hook) snd_hda_sync_vmaster_hook(&spec->vmaster_mute); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 3942e1b528d8..3dca65d79b02 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -138,6 +138,8 @@ int snd_hda_codec_reset(struct hda_codec *codec); void snd_hda_codec_register(struct hda_codec *codec); void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec); +#define snd_hda_regmap_sync(codec) snd_hdac_regmap_sync(&(codec)->core) + enum { HDA_VMUTE_OFF, HDA_VMUTE_ON, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3beb842817ff..3a18fa4f8c21 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2404,7 +2404,7 @@ static int generic_hdmi_resume(struct hda_codec *codec) int pin_idx; codec->patch_ops.init(codec); - regcache_sync(codec->core.regmap); + snd_hda_regmap_sync(codec); for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 181e15072692..d88a001ba8cf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -907,7 +907,7 @@ static int alc_resume(struct hda_codec *codec) if (!spec->no_depop_delay) msleep(150); /* to avoid pop noise */ codec->patch_ops.init(codec); - regcache_sync(codec->core.regmap); + snd_hda_regmap_sync(codec); hda_call_check_power_status(codec, 0x01); return 0; } @@ -3638,7 +3638,7 @@ static int alc269_resume(struct hda_codec *codec) msleep(200); } - regcache_sync(codec->core.regmap); + snd_hda_regmap_sync(codec); hda_call_check_power_status(codec, 0x01); /* on some machine, the BIOS will clear the codec gpio data when enter diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index b40d01e01832..7ef8f3105cdb 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -396,7 +396,7 @@ static int via_resume(struct hda_codec *codec) /* some delay here to make jack detection working (bko#98921) */ msleep(10); codec->patch_ops.init(codec); - regcache_sync(codec->core.regmap); + snd_hda_regmap_sync(codec); return 0; } #endif -- 2.16.4

2 2

[alsa-devel] [PATCH v2] asoc: sti: fix possible sleep-in-atomic
by Arnaud Pouliquen 13 Jan '20

13 Jan '20

Change mutex and spinlock management to avoid sleep in atomic issue. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- V1 to V2 suppress unexpected [INTERNAL REVIEW] tag in subject sound/soc/sti/uniperif_player.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 48ea915b24ba..2ed92c990b97 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -226,7 +226,6 @@ static void uni_player_set_channel_status(struct uniperif *player, * sampling frequency. If no sample rate is already specified, then * set one. */ - mutex_lock(&player->ctrl_lock); if (runtime) { switch (runtime->rate) { case 22050: @@ -303,7 +302,6 @@ static void uni_player_set_channel_status(struct uniperif *player, player->stream_settings.iec958.status[3 + (n * 4)] << 24; SET_UNIPERIF_CHANNEL_STA_REGN(player, n, status); } - mutex_unlock(&player->ctrl_lock); /* Update the channel status */ if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) @@ -365,8 +363,10 @@ static int uni_player_prepare_iec958(struct uniperif *player, SET_UNIPERIF_CTRL_ZERO_STUFF_HW(player); + mutex_lock(&player->ctrl_lock); /* Update the channel status */ uni_player_set_channel_status(player, runtime); + mutex_unlock(&player->ctrl_lock); /* Clear the user validity user bits */ SET_UNIPERIF_USER_VALIDITY_VALIDITY_LR(player, 0); @@ -598,7 +598,6 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, iec958->status[1] = ucontrol->value.iec958.status[1]; iec958->status[2] = ucontrol->value.iec958.status[2]; iec958->status[3] = ucontrol->value.iec958.status[3]; - mutex_unlock(&player->ctrl_lock); spin_lock_irqsave(&player->irq_lock, flags); if (player->substream && player->substream->runtime) @@ -608,6 +607,8 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, uni_player_set_channel_status(player, NULL); spin_unlock_irqrestore(&player->irq_lock, flags); + mutex_unlock(&player->ctrl_lock); + return 0; } -- 2.17.1

1 0

[alsa-devel] [PATCH v3] ALSA: usb-audio: Add boot quirk for MOTU M Series
by Alexander Tsoy 13 Jan '20

13 Jan '20

Add delay to make sure that audio urbs are not sent too early. Otherwise the device hangs. Windows driver makes ~2s delay, so use about the same time delay value. snd_usb_apply_boot_quirk() is called 3 times for my MOTU M4, which is an overkill. Thus a quirk that is called only once is implemented. Also send two vendor-specific control messages before and after the delay. This behaviour is blindly copied from the Windows driver. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- v3 changes: fixed identations once more sound/usb/card.c | 4 ++++ sound/usb/quirks.c | 38 ++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 5 +++++ 3 files changed, 47 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 9f743ebae615..2f582ac7cf78 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -600,6 +600,10 @@ static int usb_audio_probe(struct usb_interface *intf, } } if (! chip) { + err = snd_usb_apply_boot_quirk_once(dev, intf, quirk, id); + if (err < 0) + return err; + /* it's a fresh one. * now look for an empty slot and create a new card instance */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 82184036437b..0464df785f2e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1113,6 +1113,31 @@ static int snd_usb_motu_microbookii_boot_quirk(struct usb_device *dev) return err; } +static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev) +{ + int ret; + + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x0, 0, NULL, 0, 1000); + + if (ret < 0) + return ret; + + msleep(2000); + + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x20, 0, NULL, 0, 1000); + + if (ret < 0) + return ret; + + return 0; +} + /* * Setup quirks */ @@ -1297,6 +1322,19 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return 0; } +int snd_usb_apply_boot_quirk_once(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk, + unsigned int id) +{ + switch (id) { + case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + return snd_usb_motu_m_series_boot_quirk(dev); + } + + return 0; +} + /* * check if the device uses big-endian samples */ diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index a80e0ddd0736..df0355843a4c 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -20,6 +20,11 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, const struct snd_usb_audio_quirk *quirk, unsigned int usb_id); +int snd_usb_apply_boot_quirk_once(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk, + unsigned int usb_id); + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, struct audioformat *fmt); -- 2.24.1

2 1

[alsa-devel] [PATCH 0/3] ALSA: dice: add support for Alesis MasterControl
by Takashi Sakamoto 13 Jan '20

13 Jan '20

Hi, This patchset includes some fixes to support Alesis MasterControl. Like Alesis iO 14/26, MasterControl supports two ports for tx isoc communication (PCM/MIDI capture) and one port for rx isoc communication (PCM/MIDI playback) as well. Takashi Sakamoto (3): dice: fix fallback from protocol extension into limited functionality dice: loosen stream format check for MIDI conformant data channel dice: add support for Alesis MasterControl sound/firewire/dice/dice-alesis.c | 24 ++++++++++++++++++++++++ sound/firewire/dice/dice-extension.c | 5 ++++- sound/firewire/dice/dice-stream.c | 9 +++------ sound/firewire/dice/dice.c | 8 ++++++++ sound/firewire/dice/dice.h | 1 + 5 files changed, 40 insertions(+), 7 deletions(-) -- 2.20.1

2 6

[alsa-devel] [PATCH 0/3] ALSA: oxfw: fixes for Stanton SCS.1d
by Takashi Sakamoto 13 Jan '20

13 Jan '20

Hi, ALSA oxfw driver includes functionality of former snd-scs1x implements. However, since it was merged in Linux v4.5, the driver can't add sound card instance for Stanton SCS.1d due to its quirk. This patchset fixes the regression. Takashi Sakamoto (3): ALSA: oxfw: use ENXIO for not-supported cases ALSA: oxfw: don't add MIDI/PCM interface when packet streaming is unavailable ALSA: oxfw: fix for Stanton SCS.1d sound/firewire/oxfw/oxfw-command.c | 6 +-- sound/firewire/oxfw/oxfw-stream.c | 82 +++++++++++++++++------------- sound/firewire/oxfw/oxfw.c | 39 +++++++------- sound/firewire/oxfw/oxfw.h | 1 + 4 files changed, 74 insertions(+), 54 deletions(-) -- 2.20.1

2 6

[alsa-devel] [PATCH] ALSA: firewire-tascam: fix corruption due to spin lock without restoration in SoftIRQ context
by Takashi Sakamoto 13 Jan '20

13 Jan '20

ALSA firewire-tascam driver can bring corruption due to spin lock without restoration of IRQ flag in SoftIRQ context. This commit fixes the bug. Cc: Scott Bahling <sbahling(a)suse.com> Cc: <stable(a)vger.kernel.org> # v4.21 Fixes: d7167422433c ("ALSA: firewire-tascam: queue events for change of control surface") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/tascam/amdtp-tascam.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c index e80bb84c43f6..f823a2ab3544 100644 --- a/sound/firewire/tascam/amdtp-tascam.c +++ b/sound/firewire/tascam/amdtp-tascam.c @@ -157,14 +157,15 @@ static void read_status_messages(struct amdtp_stream *s, if ((before ^ after) & mask) { struct snd_firewire_tascam_change *entry = &tscm->queue[tscm->push_pos]; + unsigned long flag; - spin_lock_irq(&tscm->lock); + spin_lock_irqsave(&tscm->lock, flag); entry->index = index; entry->before = before; entry->after = after; if (++tscm->push_pos >= SND_TSCM_QUEUE_COUNT) tscm->push_pos = 0; - spin_unlock_irq(&tscm->lock); + spin_unlock_irqrestore(&tscm->lock, flag); wake_up(&tscm->hwdep_wait); } -- 2.20.1

2 1

[alsa-devel] Quote strings containing [ or ] when saving an alsa config
by GitHub pull_request - opened 13 Jan '20

13 Jan '20

alsa-project/alsa-lib pull request #23 was opened from Bertware: This pull request will fix issue #22. When saving the configuration for a control, values containing square brackets should be quoted. Please review and test as this code hasn't been tested yet. Note: does the '+' character also need quotes? If so, now might be a good time to add it to this list. Request URL : https://github.com/alsa-project/alsa-lib/pull/23 Patch URL : https://github.com/alsa-project/alsa-lib/pull/23.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH] ALSA: cmipci: Fix possible a data race in snd_cmipci_interrupt()
by Jia-Ju Bai 13 Jan '20

13 Jan '20

The functions snd_cmipci_interrupt() and snd_cmipci_capture_trigger() may be concurrently executed. The function snd_cmipci_capture_trigger() calls snd_cmipci_pcm_trigger(). In snd_cmipci_pcm_trigger(), the variable rec->running is written with holding a spinlock cm->reg_lock. But in snd_cmipci_interrupt(), the identical variable cm->channel[0].running or cm->channel[1].running is read without holding this spinlock. Thus, a possible data race may occur. To fix this data race, in snd_cmipci_interrupt(), the variables cm->channel[0].running and cm->channel[1].running are read with holding the spinlock cm->reg_lock. This data race is found by the runtime testing of our tool DILP-2. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/pci/cmipci.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index dd9d62e2b633..f791152ec48f 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -1430,7 +1430,7 @@ static int snd_cmipci_capture_spdif_hw_free(struct snd_pcm_substream *subs) static irqreturn_t snd_cmipci_interrupt(int irq, void *dev_id) { struct cmipci *cm = dev_id; - unsigned int status, mask = 0; + unsigned int run_flag0, run_flag1, status, mask = 0; /* fastpath out, to ease interrupt sharing */ status = snd_cmipci_read(cm, CM_REG_INT_STATUS); @@ -1445,15 +1445,17 @@ static irqreturn_t snd_cmipci_interrupt(int irq, void *dev_id) mask |= CM_CH1_INT_EN; snd_cmipci_clear_bit(cm, CM_REG_INT_HLDCLR, mask); snd_cmipci_set_bit(cm, CM_REG_INT_HLDCLR, mask); + run_flag0 = cm->channel[0].running; + run_flag1 = cm->channel[1].running; spin_unlock(&cm->reg_lock); if (cm->rmidi && (status & CM_UARTINT)) snd_mpu401_uart_interrupt(irq, cm->rmidi->private_data); if (cm->pcm) { - if ((status & CM_CHINT0) && cm->channel[0].running) + if ((status & CM_CHINT0) && run_flag0) snd_pcm_period_elapsed(cm->channel[0].substream); - if ((status & CM_CHINT1) && cm->channel[1].running) + if ((status & CM_CHINT1) && run_flag1) snd_pcm_period_elapsed(cm->channel[1].substream); } return IRQ_HANDLED; -- 2.17.1

2 3

[alsa-devel] [PATCH 0/6] soundwire: stream: fix state machines and transitions
by Pierre-Louis Bossart 13 Jan '20

13 Jan '20

The existing stream support works fine with simple cases, but does not map well with ALSA transitions for underflows/resume where prepare() can be called multiple times. Concurrency with multiple devices per links or multiple streams enabled on the same link also needs to be fixed. These patches are the result of hours of validation on the Intel side and should benefit other implementations since there is nothing hardware-specific. The Intel-specific changes being reviewed do depend on those stream changes though to be functional. Bard Liao (1): soundwire: stream: only prepare stream when it is configured. Pierre-Louis Bossart (3): soundwire: stream: remove redundant pr_err traces soundwire: stream: update state machine and add state checks soundwire: stream: do not update parameters during DISABLED-PREPARED transition Rander Wang (2): soundwire: stream: fix support for multiple Slaves on the same link soundwire: stream: don't program ports for a stream that has not been prepared Documentation/driver-api/soundwire/stream.rst | 63 ++++++++---- drivers/soundwire/stream.c | 97 +++++++++++++++---- 2 files changed, 124 insertions(+), 36 deletions(-) base-commit: 09f6a72d014386939d21899921dd379006471a4b -- 2.20.1

2 15

[alsa-devel] [PATCH v2] ALSA: usb-audio: Add support for Presonus Studio 1810c
by mickflemm＠gmail.com 12 Jan '20

12 Jan '20

This patch adds support for Presonus Studio 1810c, a usb interface that's UAC2 compliant with a few quirks and a few extra hw-specific controls. I've tested all 3 altsettings and the added switch controls and they work as expected. More infos on the card: https://www.presonus.com/products/Studio-1810c Note that this work is based on packet inspection with usbmon. I just wanted to get this card to work for using it on our open-source radio station: https://github.com/UoC-Radio v2 address issues reported by Takashi: * Properly get/set enum type controls * Prevent race condition on switch_get/set * Various control naming changes * Various coding style fixes Signed-off-by: Nick Kossifidis <mickflemm(a)gmail.com> --- sound/usb/Makefile | 1 + sound/usb/format.c | 17 ++ sound/usb/mixer_quirks.c | 5 + sound/usb/mixer_s1810c.c | 595 +++++++++++++++++++++++++++++++++++++++ sound/usb/mixer_s1810c.h | 7 + sound/usb/quirks.c | 36 +++ 6 files changed, 661 insertions(+) create mode 100644 sound/usb/mixer_s1810c.c create mode 100644 sound/usb/mixer_s1810c.h diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 78edd7d2f..56031026b 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -13,6 +13,7 @@ snd-usb-audio-objs := card.o \ mixer_scarlett.o \ mixer_scarlett_gen2.o \ mixer_us16x08.o \ + mixer_s1810c.o \ pcm.o \ power.o \ proc.o \ diff --git a/sound/usb/format.c b/sound/usb/format.c index d79db7130..edf3f2a55 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -262,6 +262,23 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, } for (rate = min; rate <= max; rate += res) { + + /* + * Presonus Studio 1810c anounces invalid + * sampling rates for its streams. + */ + if (chip->usb_id == USB_ID(0x0194f, 0x010c) && + ((rate > 48000 && fp->altsetting == 1) || + ((rate < 88200 || rate > 96000) + && fp->altsetting == 2) || + ((rate < 176400 || rate > 192000) + && fp->altsetting == 3))) { + if (res) + continue; + else + break; + } + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 39e27ae6c..71e705fab 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -34,6 +34,7 @@ #include "mixer_scarlett.h" #include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" +#include "mixer_s1810c.h" #include "helper.h" struct std_mono_table { @@ -2277,6 +2278,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fd4): /* RME */ err = snd_rme_controls_create(mixer); break; + + case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */ + err = snd_sc1810_init_mixer(mixer); + break; } return err; diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c new file mode 100644 index 000000000..ae3929e1c --- /dev/null +++ b/sound/usb/mixer_s1810c.c @@ -0,0 +1,595 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Presonus Studio 1810c driver for ALSA + * Copyright (C) 2019 Nick Kossifidis <mickflemm(a)gmail.com> + * + * Based on reverse engineering of the communication protocol + * between the windows driver / Univeral Control (UC) program + * and the device, through usbmon. + * + * For now this bypasses the mixer, with all channels split, + * so that the software can mix with greater flexibility. + * It also adds controls for the 4 buttons on the front of + * the device. + */ + +#include <linux/usb.h> +#include <linux/usb/audio-v2.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/control.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "mixer_quirks.h" +#include "helper.h" + +#define SC1810C_CMD_REQ 160 +#define SC1810C_CMD_REQTYPE \ + (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT) +#define SC1810C_CMD_F1 0x50617269 +#define SC1810C_CMD_F2 0x14 + +/* + * DISCLAIMER: These are just guesses based on the + * dumps I got. + * + * It seems like a selects between + * device (0), mixer (0x64) and output (0x65) + * + * For mixer (0x64): + * * b selects an input channel (see below). + * * c selects an output channel pair (see below). + * * d selects left (0) or right (1) of that pair. + * * e 0-> disconnect, 0x01000000-> connect, + * 0x0109-> used for stereo-linking channels, + * e is also used for setting volume levels + * in which case b is also set so I guess + * this way it is possible to set the volume + * level from the specified input to the + * specified output. + * + * IN Channels: + * 0 - 7 Mic/Inst/Line (Analog inputs) + * 8 - 9 S/PDIF + * 10 - 17 ADAT + * 18 - 35 DAW (Inputs from the host) + * + * OUT Channels (pairs): + * 0 -> Main out + * 1 -> Line1/2 + * 2 -> Line3/4 + * 3 -> S/PDIF + * 4 -> ADAT? + * + * For device (0): + * * b and c are not used, at least not on the + * dumps I got. + * * d sets the control id to be modified + * (see below). + * * e sets the setting for that control. + * (so for the switches I was interested + * in it's 0/1) + * + * For output (0x65): + * * b is the output channel (see above). + * * c is zero. + * * e I guess the same as with mixer except 0x0109 + * which I didn't see in my dumps. + * + * The two fixed fields have the same values for + * mixer and output but a different set for device. + */ +struct s1810c_ctl_packet { + u32 a; + u32 b; + u32 fixed1; + u32 fixed2; + u32 c; + u32 d; + u32 e; +}; + +#define SC1810C_CTL_LINE_SW 0 +#define SC1810C_CTL_MUTE_SW 1 +#define SC1810C_CTL_AB_SW 3 +#define SC1810C_CTL_48V_SW 4 + +#define SC1810C_SET_STATE_REQ 161 +#define SC1810C_SET_STATE_REQTYPE SC1810C_CMD_REQTYPE +#define SC1810C_SET_STATE_F1 0x64656D73 +#define SC1810C_SET_STATE_F2 0xF4 + +#define SC1810C_GET_STATE_REQ 162 +#define SC1810C_GET_STATE_REQTYPE \ + (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN) +#define SC1810C_GET_STATE_F1 SC1810C_SET_STATE_F1 +#define SC1810C_GET_STATE_F2 SC1810C_SET_STATE_F2 + +#define SC1810C_STATE_F1_IDX 2 +#define SC1810C_STATE_F2_IDX 3 + +/* + * This packet includes mixer volumes and + * various other fields, it's an extended + * version of ctl_packet, with a and b + * being zero and different f1/f2. + */ +struct s1810c_state_packet { + u32 fields[63]; +}; + +#define SC1810C_STATE_48V_SW 58 +#define SC1810C_STATE_LINE_SW 59 +#define SC1810C_STATE_MUTE_SW 60 +#define SC1810C_STATE_AB_SW 62 + +struct s1810_mixer_state { + uint16_t seqnum; + struct mutex usb_mutex; + struct mutex data_mutex; +}; + +static int +snd_s1810c_send_ctl_packet(struct usb_device *dev, u32 a, + u32 b, u32 c, u32 d, u32 e) +{ + struct s1810c_ctl_packet pkt = { 0 }; + int ret = 0; + + pkt.fixed1 = SC1810C_CMD_F1; + pkt.fixed2 = SC1810C_CMD_F2; + + pkt.a = a; + pkt.b = b; + pkt.c = c; + pkt.d = d; + /* + * Value for settings 0/1 for this + * output channel is always 0 (probably because + * there is no ADAT output on 1810c) + */ + pkt.e = (c == 4) ? 0 : e; + + ret = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + SC1810C_CMD_REQ, + SC1810C_CMD_REQTYPE, 0, 0, &pkt, sizeof(pkt)); + if (ret < 0) { + dev_warn(&dev->dev, "could not send ctl packet\n"); + return ret; + } + return 0; +} + +/* + * When opening Universal Control the program periodicaly + * sends and receives state packets for syncinc state between + * the device and the host. + * + * Note that if we send only the request to get data back we'll + * get an error, we need to first send an empty state packet and + * then ask to receive a filled. Their seqnumbers must also match. + */ +static int +snd_sc1810c_get_status_field(struct usb_device *dev, + u32 *field, int field_idx, uint16_t *seqnum) +{ + struct s1810c_state_packet pkt_out = { 0 }; + struct s1810c_state_packet pkt_in = { 0 }; + int ret = 0; + + pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1; + pkt_out.fields[SC1810C_STATE_F2_IDX] = SC1810C_SET_STATE_F2; + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + SC1810C_SET_STATE_REQ, + SC1810C_SET_STATE_REQTYPE, + (*seqnum), 0, &pkt_out, sizeof(pkt_out)); + if (ret < 0) { + dev_warn(&dev->dev, "could not send state packet (%d)\n", ret); + return ret; + } + + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + SC1810C_GET_STATE_REQ, + SC1810C_GET_STATE_REQTYPE, + (*seqnum), 0, &pkt_in, sizeof(pkt_in)); + if (ret < 0) { + dev_warn(&dev->dev, "could not get state field %u (%d)\n", + field_idx, ret); + return ret; + } + + (*field) = pkt_in.fields[field_idx]; + (*seqnum)++; + return 0; +} + +/* + * This is what I got when bypassing the mixer with + * all channels split. I'm not 100% sure of what's going + * on, I could probably clean this up based on my observations + * but I prefer to keep the same behavior as the windows driver. + */ +static int snd_s1810c_init_mixer_maps(struct snd_usb_audio *chip) +{ + u32 a, b, c, e, n, off; + struct usb_device *dev = chip->dev; + + /* Set initial volume levels ? */ + a = 0x64; + e = 0xbc; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off, c = 0; b < 18 + off; b++) { + /* This channel to all outputs ? */ + for (c = 0; c <= 8; c++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + /* This channel to main output (again) */ + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + /* + * I noticed on UC that DAW channels have different + * initial volumes (they can go both above and below + * 0db), so this makes sense. + */ + e = 0xb53bf0; + } + + /* Connect analog outputs ? */ + a = 0x65; + e = 0x01000000; + for (b = 1; b < 3; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + snd_s1810c_send_ctl_packet(dev, a, 0, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 0, 0, 1, e); + + /* Set initial volume levels for S/PDIF mappings ? */ + a = 0x64; + e = 0xbc; + c = 3; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off; b < 18 + off; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + e = 0xb53bf0; + } + + /* Connect S/PDIF output ? */ + a = 0x65; + e = 0x01000000; + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + /* Connect all outputs (again) ? */ + a = 0x65; + e = 0x01000000; + for (b = 0; b < 4; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + + /* Basic routing to get sound out of the device */ + a = 0x64; + e = 0x01000000; + for (c = 0; c < 4; c++) { + for (b = 0; b < 36; b++) { + if ((c == 0 && b == 18) || /* DAW1/2 -> Main */ + (c == 1 && b == 20) || /* DAW3/4 -> Line3/4 */ + (c == 2 && b == 22) || /* DAW4/5 -> Line5/4 */ + (c == 3 && b == 24)) { /* DAW5/6 -> S/PDIF */ + /* Left */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0); + b++; + /* Right */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } else { + /* Leave the rest disconnected */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0); + } + } + } + + /* Set initial volume levels for S/PDIF (again) ? */ + a = 0x64; + e = 0xbc; + c = 3; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off; b < 18 + off; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + e = 0xb53bf0; + } + + /* Connect S/PDIF outputs (again) ? */ + a = 0x65; + e = 0x01000000; + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + /* Again ? */ + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + return 0; +} + +/* + * Sync state with the device and retrieve the requested field, + * whose index is specified in (kctl->private_value & 0xFF), + * from the received fields array. + */ +static int +snd_s1810c_get_switch_state(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl, u32 *state) +{ + struct snd_usb_audio *chip = mixer->chip; + struct s1810_mixer_state *private = mixer->private_data; + u32 field = 0; + u32 ctl_idx = (u32) (kctl->private_value & 0xFF); + int ret = 0; + + mutex_lock(&private->usb_mutex); + ret = snd_sc1810c_get_status_field(chip->dev, &field, + ctl_idx, &private->seqnum); + if (ret < 0) + goto unlock; + + *state = field; + unlock: + mutex_unlock(&private->usb_mutex); + return ret ? ret : 0; +} + +/* + * Send a control packet to the device for the control id + * specified in (kctl->private_value >> 8) with value + * specified in (kctl->private_value >> 16). + */ +static int +snd_s1810c_set_switch_state(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl) +{ + struct snd_usb_audio *chip = mixer->chip; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_id = (pval >> 8) & 0xFF; + u32 ctl_val = (pval >> 16) & 0x1; + int ret = 0; + + mutex_lock(&private->usb_mutex); + ret = snd_s1810c_send_ctl_packet(chip->dev, 0, 0, 0, ctl_id, ctl_val); + mutex_unlock(&private->usb_mutex); + return ret; +} + +/* Generic get/set/init functions for switch controls */ + +static int +snd_s1810c_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ctl_elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_idx = pval & 0xFF; + u32 state = 0; + int ret = 0; + + mutex_lock(&private->data_mutex); + ret = snd_s1810c_get_switch_state(mixer, kctl, &state); + if (ret < 0) + goto unlock; + + switch (ctl_idx) { + case SC1810C_STATE_LINE_SW: + case SC1810C_STATE_AB_SW: + ctl_elem->value.enumerated.item[0] = (int)state; + break; + default: + ctl_elem->value.integer.value[0] = (long)state; + } + + unlock: + mutex_unlock(&private->data_mutex); + return (ret < 0) ? ret : 0; +} + +static int +snd_s1810c_switch_set(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ctl_elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_idx = pval & 0xFF; + u32 curval = 0; + u32 newval = 0; + int ret = 0; + + mutex_lock(&private->data_mutex); + ret = snd_s1810c_get_switch_state(mixer, kctl, &curval); + if (ret < 0) + goto unlock; + + switch (ctl_idx) { + case SC1810C_STATE_LINE_SW: + case SC1810C_STATE_AB_SW: + newval = (u32) ctl_elem->value.enumerated.item[0]; + break; + default: + newval = (u32) ctl_elem->value.integer.value[0]; + } + + if (curval == newval) + goto unlock; + + kctl->private_value &= ~(0x1 << 16); + kctl->private_value |= (unsigned int)(newval & 0x1) << 16; + ret = snd_s1810c_set_switch_state(mixer, kctl); + + unlock: + mutex_unlock(&private->data_mutex); + return (ret < 0) ? 0 : 1; +} + +static int +snd_s1810c_switch_init(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *new_kctl) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + + elem = kzalloc(sizeof(struct usb_mixer_elem_info), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = 0; + elem->head.id = 0; + elem->channels = 1; + + kctl = snd_ctl_new1(new_kctl, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + return snd_usb_mixer_add_control(&elem->head, kctl); +} + +static int +snd_s1810c_line_sw_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "Preamp On (Mic/Inst)", + "Preamp Off (Line in)" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static const struct snd_kcontrol_new snd_s1810c_line_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line 1/2 Source Type Switch", + .info = snd_s1810c_line_sw_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_LINE_SW | SC1810C_CTL_LINE_SW << 8) +}; + +static const struct snd_kcontrol_new snd_s1810c_mute_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mute Main Out Switch", + .info = snd_ctl_boolean_mono_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_MUTE_SW | SC1810C_CTL_MUTE_SW << 8) +}; + +static const struct snd_kcontrol_new snd_s1810c_48v_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "48V Phantom Power On Mic Inputs Switch", + .info = snd_ctl_boolean_mono_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_48V_SW | SC1810C_CTL_48V_SW << 8) +}; + +static int +snd_s1810c_ab_sw_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "1/2", + "3/4" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static const struct snd_kcontrol_new snd_s1810c_ab_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone 1 Source Route", + .info = snd_s1810c_ab_sw_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_AB_SW | SC1810C_CTL_AB_SW << 8) +}; + +static void snd_sc1810_mixer_state_free(struct usb_mixer_interface *mixer) +{ + struct s1810_mixer_state *private = mixer->private_data; + kfree(private); + mixer->private_data = NULL; +} + +/* Entry point, called from mixer_quirks.c */ +int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer) +{ + struct s1810_mixer_state *private = NULL; + struct snd_usb_audio *chip = mixer->chip; + struct usb_device *dev = chip->dev; + int ret = 0; + + /* Run this only once */ + if (!list_empty(&chip->mixer_list)) + return 0; + + dev_info(&dev->dev, + "Presonus Studio 1810c, device_setup: %u\n", chip->setup); + if (chip->setup == 1) + dev_info(&dev->dev, "(8out/18in @ 48KHz)\n"); + else if (chip->setup == 2) + dev_info(&dev->dev, "(6out/8in @ 192KHz)\n"); + else + dev_info(&dev->dev, "(8out/14in @ 96KHz)\n"); + + ret = snd_s1810c_init_mixer_maps(chip); + if (ret < 0) + return ret; + + private = kzalloc(sizeof(struct s1810_mixer_state), GFP_KERNEL); + if (!private) + return -ENOMEM; + + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + + mixer->private_data = private; + mixer->private_free = snd_sc1810_mixer_state_free; + + private->seqnum = 1; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_line_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_mute_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_48v_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw); + if (ret < 0) + return ret; + return ret; +} diff --git a/sound/usb/mixer_s1810c.h b/sound/usb/mixer_s1810c.h new file mode 100644 index 000000000..a79a3743c --- /dev/null +++ b/sound/usb/mixer_s1810c.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Presonus Studio 1810c driver for ALSA + * Copyright (C) 2019 Nick Kossifidis <mickflemm(a)gmail.com> + */ + +int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a81c20664..9bbdea882 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1227,6 +1227,38 @@ static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip, return 0; /* keep this altsetting */ } +static int s1810c_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno) +{ + /* + * Altno settings: + * + * Playback (Interface 1): + * 1: 6 Analog + 2 S/PDIF + * 2: 6 Analog + 2 S/PDIF + * 3: 6 Analog + * + * Capture (Interface 2): + * 1: 8 Analog + 2 S/PDIF + 8 ADAT + * 2: 8 Analog + 2 S/PDIF + 4 ADAT + * 3: 8 Analog + */ + + /* + * I'll leave 2 as the default one and + * use device_setup to switch to the + * other two. + */ + if ((chip->setup == 0 || chip->setup > 2) && altno != 2) + return 1; + else if (chip->setup == 1 && altno != 1) + return 1; + else if (chip->setup == 2 && altno != 3) + return 1; + + return 0; +} + int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, int iface, int altno) @@ -1240,6 +1272,10 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, /* fasttrackpro usb: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0763, 0x2012)) return fasttrackpro_skip_setting_quirk(chip, iface, altno); + /* presonus studio 1810c: skip altsets incompatible with device_setup */ + if (chip->usb_id == USB_ID(0x0194f, 0x010c)) + return s1810c_skip_setting_quirk(chip, iface, altno); + return 0; } -- 2.23.0

3 2

[alsa-devel] [PATCH v2] ALSA: usb-audio: Add boot quirk for MOTU M Series
by Alexander Tsoy 12 Jan '20

12 Jan '20

Add delay to make sure that audio urbs are not sent too early. Otherwise the device hangs. Windows driver makes ~2s delay, so use about the same time delay value. snd_usb_apply_boot_quirk() is called 3 times for my MOTU M4, which is an overkill. Thus a quirk that is called only once is implemented. Also send two vendor-specific control messages before and after the delay. This behaviour is blindly copied from the Windows driver. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- v2 changes: indentation fixes sound/usb/card.c | 4 ++++ sound/usb/quirks.c | 38 ++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 5 +++++ 3 files changed, 47 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 9f743ebae615..2f582ac7cf78 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -600,6 +600,10 @@ static int usb_audio_probe(struct usb_interface *intf, } } if (! chip) { + err = snd_usb_apply_boot_quirk_once(dev, intf, quirk, id); + if (err < 0) + return err; + /* it's a fresh one. * now look for an empty slot and create a new card instance */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 82184036437b..0464df785f2e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1113,6 +1113,31 @@ static int snd_usb_motu_microbookii_boot_quirk(struct usb_device *dev) return err; } +static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev) +{ + int ret; + + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x0, 0, NULL, 0, 1000); + + if (ret < 0) + return ret; + + msleep(2000); + + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x20, 0, NULL, 0, 1000); + + if (ret < 0) + return ret; + + return 0; +} + /* * Setup quirks */ @@ -1297,6 +1322,19 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return 0; } +int snd_usb_apply_boot_quirk_once(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk, + unsigned int id) +{ + switch (id) { + case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + return snd_usb_motu_m_series_boot_quirk(dev); + } + + return 0; +} + /* * check if the device uses big-endian samples */ diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index a80e0ddd0736..b46b108769e8 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -20,6 +20,11 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, const struct snd_usb_audio_quirk *quirk, unsigned int usb_id); +int snd_usb_apply_boot_quirk_once(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk, + unsigned int usb_id); + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, struct audioformat *fmt); -- 2.24.1

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Add boot quirk for MOTU M Series
by Alexander Tsoy 12 Jan '20

12 Jan '20

Add delay to make sure that audio urbs are not sent too early. Otherwise the device hangs. Windows driver makes ~2s delay, so use about the same time delay value. snd_usb_apply_boot_quirk() is called 3 times for my MOTU M4, which is an overkill. Thus a quirk that is called only once is implemented. Also send two vendor-specific control messages before and after the delay. This behaviour is blindly copied from the Windows driver. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/card.c | 4 ++++ sound/usb/quirks.c | 38 ++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 5 +++++ 3 files changed, 47 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 9f743ebae615..2f582ac7cf78 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -600,6 +600,10 @@ static int usb_audio_probe(struct usb_interface *intf, } } if (! chip) { + err = snd_usb_apply_boot_quirk_once(dev, intf, quirk, id); + if (err < 0) + return err; + /* it's a fresh one. * now look for an empty slot and create a new card instance */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 82184036437b..c93c53126243 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1113,6 +1113,31 @@ static int snd_usb_motu_microbookii_boot_quirk(struct usb_device *dev) return err; } +static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev) +{ + int ret; + + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x0, 0, NULL, 0, 1000); + + if (ret < 0) + return ret; + + msleep(2000); + + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x20, 0, NULL, 0, 1000); + + if (ret < 0) + return ret; + + return 0; +} + /* * Setup quirks */ @@ -1297,6 +1322,19 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return 0; } +int snd_usb_apply_boot_quirk_once(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk, + unsigned int id) +{ + switch (id) { + case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + return snd_usb_motu_m_series_boot_quirk(dev); + } + + return 0; +} + /* * check if the device uses big-endian samples */ diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index a80e0ddd0736..fa83a98ad7bb 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -20,6 +20,11 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, const struct snd_usb_audio_quirk *quirk, unsigned int usb_id); +int snd_usb_apply_boot_quirk_once(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk, + unsigned int usb_id); + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, struct audioformat *fmt); -- 2.24.1

2 1

[alsa-devel] [PATCH 0/2] ALSA: fix compiler warnings
by Pierre-Louis Bossart 12 Jan '20

12 Jan '20

2 additional issues reported by GCC W=1. the first one looks like a bug, the second one more of a style issue. Fixes for ASoC will be provided separately after review/test. Pierre-Louis Bossart (2): ALSA: HDA: patch_realtek: fix empty macro usage in if block ALSA: usb: update old-style static const declaration sound/pci/hda/patch_realtek.c | 4 ++-- sound/usb/mixer_quirks.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) -- 2.20.1

2 4

[alsa-devel] [PATCH] ALSA: timer: fix nsec/sec initialization confusion
by Pierre-Louis Bossart 12 Jan '20

12 Jan '20

GCC reports a warning with W=1: sound/core/timer.c: In function ‘snd_timer_user_read’: sound/core/timer.c:2219:19: warning: initialized field overwritten [-Woverride-init] 2219 | .tstamp_sec = tread->tstamp_nsec, | ^~~~~ sound/core/timer.c:2219:19: note: (near initialization for ‘(anonymous).tstamp_sec’) Assigning nsec values to sec fields is problematic in general, even more so when the initial goal was to survive the 2030 timer armageddon. Fix by using the proper field in the initialization Cc: Baolin Wang <baolin.wang(a)linaro.org> Cc: Arnd Bergmann <arnd(a)arndb.de> Fixes: 07094ae6f9527 ("ALSA: Avoid using timespec for struct snd_timer_tread") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/core/timer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 8835ff91a893..d9f85f2d66a3 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -2216,7 +2216,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, tread32 = (struct snd_timer_tread32) { .event = tread->event, .tstamp_sec = tread->tstamp_sec, - .tstamp_sec = tread->tstamp_nsec, + .tstamp_nsec = tread->tstamp_nsec, .val = tread->val, }; -- 2.20.1

3 2

[alsa-devel] alsa-gobject: ctl: fix wrong operation for values in enumerated type of element
by GitHub pull_request - opened 12 Jan '20

12 Jan '20

alsa-project/alsa-gobject pull request #4 was opened from takaswie: The 'snd_ctl_elem_value' structure includes union member for values. Although it includes 'enumerated' member for values in enumerated type of element, current implementation of alsactl library uses 'integer' member to pick up the values. As a result, in any 64 bit System V ABI, the library fails to get/set values for the enumerated type of element. This commit fixes the bug. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-gobject/pull/4 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/4.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[alsa-devel] When storing settings which contain a [ or ], the setting is not quoted.
by GitHub issues - edited 10 Jan '20

10 Jan '20

alsa-project/alsa-lib issue #22 was edited from Bertware: I am using a Raspberry Pi 1 B+ with a Hifiberry DAC+ADC Pro HAT. This HAT contains a PCM5122 Stereo DAC and an ADC. Now to the issue: The ADC has parameters such as `VINR1[SE]` and `VINR2[SE] + VINR1[SE]`. When using `alsactl store` or`alsactl restore`, restoring will fail due to an invalid file. `ALSA lib conf.c:1887:(_snd_config_load_with_include) _toplevel_:354:30:Unexpected char` This error points to the parameter `VINR1[SE]`, which is not surrounded by quotes. The other parameters, which also contain other special characters such as spaces, are surrounded by quotes. An excerpt of the file is included below. This clearly illustrates which names were encapsulated between quotes. ``` control.28 { iface MIXER name 'ADC Right Capture Source' value VINR1[SE] comment { access 'read write' type ENUMERATED count 1 item.0 'No Select' item.1 VINR1[SE] item.2 VINR2[SE] item.3 'VINR2[SE] + VINR1[SE]' item.4 VINR3[SE] item.5 'VINR3[SE] + VINR1[SE]' item.6 'VINR3[SE] + VINR2[SE]' item.7 'VINR3[SE] + VINR2[SE] + VINR1[SE]' item.8 VINR4[SE] item.9 'VINR4[SE] + VINR1[SE]' item.10 'VINR4[SE] + VINR2[SE]' item.11 'VINR4[SE] + VINR2[SE] + VINR1[SE]' item.12 'VINR4[SE] + VINR3[SE]' item.13 'VINR4[SE] + VINR3[SE] + VINR1[SE]' item.14 'VINR4[SE] + VINR3[SE] + VINR2[SE]' item.15 'VINR4[SE] + VINR3[SE] + VINR2[SE] + VINR1[SE]' item.16 '{VIN2P, VIN2M}[DIFF]' item.17 '{VIN3P, VIN3M}[DIFF]' item.18 '{VIN2P, VIN2M}[DIFF] + {VIN3P, VIN3M}[DIFF]' } } ``` Manually fixing this file by adding '' around the items which miss quotes fixes the issue, and allows restoring it. **How this leads to alsa-lib and where the issue is located** Creation of the output data and writing it to a file is handled by the save_state method in alsactl: https://github.com/alsa-project/alsa-utils/blob/master/alsactl/state.c#L1539 This code in turn calls snd_config_save in the alsa-lib project: https://github.com/alsa-project/alsa-lib/blob/b20b400e2f598c86abaf697d66396… Then further to string_print: https://github.com/alsa-project/alsa-lib/blob/b20b400e2f598c86abaf697d66396… And then in this loop which decides which characters need to be quoted: https://github.com/alsa-project/alsa-lib/blob/b20b400e2f598c86abaf697d66396… **Proposed solution** The characters `[` and `]` should be a reason to put text between quotes. These characters should be added to the switch statement in print_string here: https://github.com/alsa-project/alsa-lib/blob/b20b400e2f598c86abaf697d66396… Regards, Bert Issue URL : https://github.com/alsa-project/alsa-lib/issues/22 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] When storing settings which contain a [ or ], the setting is not quoted.
by GitHub issues - edited 10 Jan '20

10 Jan '20

alsa-project/alsa-lib issue #22 was edited from Bertware: I am using a Raspberry Pi 1 B+ with a Hifiberry DAC+ADC Pro HAT. This HAT contains a PCM5122 Stereo DAC and an ADC. Now to the issue: The ADC has parameters such as `VINR1[SE]` and `VINR2[SE] + VINR1[SE]`. When using `alsactl store` or`alsactl restore`, restoring will fail due to an invalid file. `ALSA lib conf.c:1887:(_snd_config_load_with_include) _toplevel_:354:30:Unexpected char` This error points to the parameter `VINR1[SE]`, which is not surrounded by quotes. The other parameters, which also contain other special characters such as spaces, are surrounded by quotes. An excerpt of the file is included below. This clearly illustrates which names were encapsulated between quotes. ``` control.28 { iface MIXER name 'ADC Right Capture Source' value VINR1[SE] comment { access 'read write' type ENUMERATED count 1 item.0 'No Select' item.1 VINR1[SE] item.2 VINR2[SE] item.3 'VINR2[SE] + VINR1[SE]' item.4 VINR3[SE] item.5 'VINR3[SE] + VINR1[SE]' item.6 'VINR3[SE] + VINR2[SE]' item.7 'VINR3[SE] + VINR2[SE] + VINR1[SE]' item.8 VINR4[SE] item.9 'VINR4[SE] + VINR1[SE]' item.10 'VINR4[SE] + VINR2[SE]' item.11 'VINR4[SE] + VINR2[SE] + VINR1[SE]' item.12 'VINR4[SE] + VINR3[SE]' item.13 'VINR4[SE] + VINR3[SE] + VINR1[SE]' item.14 'VINR4[SE] + VINR3[SE] + VINR2[SE]' item.15 'VINR4[SE] + VINR3[SE] + VINR2[SE] + VINR1[SE]' item.16 '{VIN2P, VIN2M}[DIFF]' item.17 '{VIN3P, VIN3M}[DIFF]' item.18 '{VIN2P, VIN2M}[DIFF] + {VIN3P, VIN3M}[DIFF]' } } ``` Creation of the output data and writing it to a file is handled by the save_state method in alsactl: https://github.com/alsa-project/alsa-utils/blob/master/alsactl/state.c#L1539 This code in turn calls snd_config_save in the alsa-lib project: https://github.com/alsa-project/alsa-lib/blob/b20b400e2f598c86abaf697d66396… Then further to string_print: https://github.com/alsa-project/alsa-lib/blob/b20b400e2f598c86abaf697d66396… And then in this loop which decides which characters need to be quoted: https://github.com/alsa-project/alsa-lib/blob/b20b400e2f598c86abaf697d66396… *Proposed solution* The characters `[` and `]` should be a reason to put text between quotes. These characters should be added to the switch statement in print_string here: https://github.com/alsa-project/alsa-lib/blob/b20b400e2f598c86abaf697d66396… Regards, Bert Issue URL : https://github.com/alsa-project/alsa-lib/issues/22 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] When storing settings which contain a [ or ], the setting is not quoted.
by GitHub issues - opened 10 Jan '20

10 Jan '20

alsa-project/alsa-lib issue #22 was opened from Bertware: I am using a Raspberry Pi 1 B+ with a Hifiberry DAC+ADC Pro HAT. This hat contains a PCM5122 Stereo DAC and an ADC. Now to the issue: The ADC has parameters such as `VINR1[SE]` and `VINR2[SE] + VINR1[SE]`. When using `alsactl store` or`alsactl restore`, restoring will fail due to an invalid file. `ALSA lib conf.c:1887:(_snd_config_load_with_include) _toplevel_:354:30:Unexpected char` This error points to the parameter `VINR1[SE]`, which is not surrounded by quotes. The other parameters, which also contain other special characters such as spaces, are surrounded by quotes. An excerpt of the file is included below. This clearly illustrates which names were encapsulated between quotes. ``` control.28 { iface MIXER name 'ADC Right Capture Source' value VINR1[SE] comment { access 'read write' type ENUMERATED count 1 item.0 'No Select' item.1 VINR1[SE] item.2 VINR2[SE] item.3 'VINR2[SE] + VINR1[SE]' item.4 VINR3[SE] item.5 'VINR3[SE] + VINR1[SE]' item.6 'VINR3[SE] + VINR2[SE]' item.7 'VINR3[SE] + VINR2[SE] + VINR1[SE]' item.8 VINR4[SE] item.9 'VINR4[SE] + VINR1[SE]' item.10 'VINR4[SE] + VINR2[SE]' item.11 'VINR4[SE] + VINR2[SE] + VINR1[SE]' item.12 'VINR4[SE] + VINR3[SE]' item.13 'VINR4[SE] + VINR3[SE] + VINR1[SE]' item.14 'VINR4[SE] + VINR3[SE] + VINR2[SE]' item.15 'VINR4[SE] + VINR3[SE] + VINR2[SE] + VINR1[SE]' item.16 '{VIN2P, VIN2M}[DIFF]' item.17 '{VIN3P, VIN3M}[DIFF]' item.18 '{VIN2P, VIN2M}[DIFF] + {VIN3P, VIN3M}[DIFF]' } } ``` Creation of the output data and writing it to a file is handled by the save_state method in alsactl: https://github.com/alsa-project/alsa-utils/blob/master/alsactl/state.c#L1539 This code in turn calls snd_config_save in the alsa-lib project: https://github.com/alsa-project/alsa-lib/blob/b20b400e2f598c86abaf697d66396… Then further to string_print: https://github.com/alsa-project/alsa-lib/blob/b20b400e2f598c86abaf697d66396… And then in this loop which decides which characters need to be quoted: https://github.com/alsa-project/alsa-lib/blob/b20b400e2f598c86abaf697d66396… *Proposed solution* The characters `[` and `]` should be a reason to put text between quotes. These characters should be added to the switch statement in print_string here: https://github.com/alsa-project/alsa-lib/blob/b20b400e2f598c86abaf697d66396… Regards, Bert Issue URL : https://github.com/alsa-project/alsa-lib/issues/22 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH v5 0/2] soundwire: Add support to Qualcomm SoundWire master
by Srinivas Kandagatla 10 Jan '20

10 Jan '20

Thanks for reviewing the v4 patchset. Here is new patchset addressing all the comments from v3 This patchset adds support for Qualcomm SoundWire Master Controller found in most of Qualcomm SoCs and WCD audio codecs. This driver along with WCD934x codec and WSA881x Class-D Smart Speaker Amplifier drivers is tested on on DragonBoard DB845c based of SDM845 SoC and Lenovo YOGA C630 Laptop based on SDM850. SoundWire controller on SDM845 is integrated in WCD934x audio codec via SlimBus interface. Currently this driver is very minimal and only supports PDM. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Test and add PCM support. Thanks, srini Changes since v4: - moved stream handling to codec as there is a strong hw requirements on port and PA enable sequence on codec side - removed dummy runtime pm - cleaned up code as suggested by Pierre Srinivas Kandagatla (2): dt-bindings: soundwire: add bindings for Qcom controller soundwire: qcom: add support for SoundWire controller .../bindings/soundwire/qcom,sdw.txt | 167 ++++ drivers/soundwire/Kconfig | 9 + drivers/soundwire/Makefile | 4 + drivers/soundwire/qcom.c | 856 ++++++++++++++++++ 4 files changed, 1036 insertions(+) create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 drivers/soundwire/qcom.c -- 2.21.0

2 6

[alsa-devel] ALSA does not detect the built-in microphone on HP EliteBook 850 G6
by lazarenkovegor 10 Jan '20

10 Jan '20

Hello! ALSA does not detect the built-in microphone on HP EliteBook 850 G6 with Ubuntu 19.10 Kenrel 5.3.0-27-generic I tried to specify the following "options snd-hda-intel model" in /etc/modprobe.d/alsa-base.conf : - *laptop-amic*,* laptop-dmic* - a recording device has appeared but it does not work. LED of MicMute on hardware button is always ON - *auto*, *3stack*,* hp-gpio4*,* hp-gpio-led*, *hp-dock-pins*, *hp-dock-gpio-mic*, *hp-9480m*, *alc280-hp-headset*, *alc221-hp-mic*, *hp-rp5800 *- there is no audio recorder in the system. LED of MicMute on hardware button is always ON - *hp-gpio2-hotkey *- there is no audio recorder in the system. LED of MicMute on hardware button is always OFF Can you help set up ALSA to work with the specified laptop? If necessary, I can compile and install the necessary patches and send work logs.

1 0

[alsa-devel] Applied "ASoC: max98090: Drop incorrectly applied duplicate commit" to the asoc tree
by Mark Brown 10 Jan '20

10 Jan '20

The patch ASoC: max98090: Drop incorrectly applied duplicate commit has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a84188eced6109983af54f9435a26d21eac3f8cc Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Fri, 10 Jan 2020 13:24:02 +0000 Subject: [PATCH] ASoC: max98090: Drop incorrectly applied duplicate commit This reverts commit 4e93c1294f4b051 (ASoC: max98090: fix incorrect helper in max98090_dapm_put_enum_double()) which was misapplied. Reported-by: Marek Szyprowski <m.szyprowski(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ba0e3ba162f8..454cb8e5b0a1 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -98,7 +98,7 @@ static int max98090_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = - snd_soc_dapm_kcontrol_component(kcontrol); + snd_soc_kcontrol_component(kcontrol); struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); int ret; -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: stm32: dfsdm: fix 16 bits record
by Olivier Moysan 10 Jan '20

10 Jan '20

In stm32_afsdm_pcm_cb function, the transfer size is provided in bytes. However, samples are copied as 16 bits words from iio buffer. Divide by two the transfer size, to copy the right number of samples. Fixes: 1e7f6e1c69f0 ("ASoC: stm32: dfsdm: add 16 bits audio record support") Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_adfsdm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 807fee1eac66..51407a21c440 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -153,13 +153,13 @@ static const struct snd_soc_component_driver stm32_adfsdm_dai_component = { .name = "stm32_dfsdm_audio", }; -static void memcpy_32to16(void *dest, const void *src, size_t n) +static void stm32_memcpy_32to16(void *dest, const void *src, size_t n) { unsigned int i = 0; u16 *d = (u16 *)dest, *s = (u16 *)src; s++; - for (i = n; i > 0; i--) { + for (i = n >> 1; i > 0; i--) { *d++ = *s++; s++; } @@ -186,8 +186,8 @@ static int stm32_afsdm_pcm_cb(const void *data, size_t size, void *private) if ((priv->pos + src_size) > buff_size) { if (format == SNDRV_PCM_FORMAT_S16_LE) - memcpy_32to16(&pcm_buff[priv->pos], src_buff, - buff_size - priv->pos); + stm32_memcpy_32to16(&pcm_buff[priv->pos], src_buff, + buff_size - priv->pos); else memcpy(&pcm_buff[priv->pos], src_buff, buff_size - priv->pos); @@ -196,8 +196,8 @@ static int stm32_afsdm_pcm_cb(const void *data, size_t size, void *private) } if (format == SNDRV_PCM_FORMAT_S16_LE) - memcpy_32to16(&pcm_buff[priv->pos], - &src_buff[src_size - cur_size], cur_size); + stm32_memcpy_32to16(&pcm_buff[priv->pos], + &src_buff[src_size - cur_size], cur_size); else memcpy(&pcm_buff[priv->pos], &src_buff[src_size - cur_size], cur_size); -- 2.17.1

2 1

[alsa-devel] [PATCH v3 resend 0/8] ASoC: soc-core cleanup step8
by Kuninori Morimoto 10 Jan '20

10 Jan '20

Hi Mark These are resend of v3 of soc-core cleanup step8. The diff from v2 are - add Reported-by, Reviewed-By for each patches - [2/8] is added - [8/8] is added These are for-v5.6 branch, but it will have conflict if you merged to for-v5.5 branch. These are based on for-5.5 + for-5.6 branch. Kuninori Morimoto (8): ASoC: soc-core: remove snd_soc_rtdcom_list ASoC: soc-core: rename snd_soc_rtdcom_add() to snd_soc_rtd_add_component() ASoC: soc-core: care .ignore_suspend for Component suspend ASoC: soc-core: remove duplicate pinctrl operation when suspend ASoC: soc-core: do pinctrl_pm_select_xxx() as component ASoC: soc-core: add snd_soc_close_delayed_work() ASoC: soc-dapm: add snd_soc_dapm_stream_stop() ASoC: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP include/sound/soc-dapm.h | 1 + include/sound/soc.h | 19 +++--- sound/soc/soc-component.c | 33 +++++---- sound/soc/soc-compress.c | 124 ++++++++++------------------------ sound/soc/soc-core.c | 167 +++++++++++++++++++++++----------------------- sound/soc/soc-dapm.c | 23 +++++++ sound/soc/soc-pcm.c | 115 ++++++++----------------------- 7 files changed, 191 insertions(+), 291 deletions(-) -- 2.7.4

2 16

[alsa-devel] [PATCH 1/2] ASoC: rt700: add rt700 codec driver
by shumingf＠realtek.com 10 Jan '20

10 Jan '20

From: Shuming Fan <shumingf(a)realtek.com> This is the initial codec driver for rt700. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt700-sdw.c | 551 +++++++++++++++ sound/soc/codecs/rt700-sdw.h | 335 +++++++++ sound/soc/codecs/rt700.c | 1238 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt700.h | 174 +++++ 6 files changed, 2310 insertions(+) create mode 100644 sound/soc/codecs/rt700-sdw.c create mode 100644 sound/soc/codecs/rt700-sdw.h create mode 100644 sound/soc/codecs/rt700.c create mode 100644 sound/soc/codecs/rt700.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 65b81888ca3d..7761aeec4d07 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -165,6 +165,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_RT5682 if I2C + select SND_SOC_RT700_SDW if SOUNDWIRE select SND_SOC_RT711_SDW if SOUNDWIRE select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE @@ -1060,6 +1061,15 @@ config SND_SOC_RT5677_SPI config SND_SOC_RT5682 tristate +config SND_SOC_RT700 + tristate + +config SND_SOC_RT700_SDW + tristate "Realtek RT700 Codec - SDW" + depends on SOUNDWIRE + select SND_SOC_RT700 + select REGMAP_SOUNDWIRE + config SND_SOC_RT711 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f4dfe033d120..288b7af354a0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -173,6 +173,7 @@ snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o snd-soc-rt5682-objs := rt5682.o +snd-soc-rt700-objs := rt700.o rt700-sdw.o snd-soc-rt711-objs := rt711.o rt711-sdw.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -466,6 +467,7 @@ obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o +obj-$(CONFIG_SND_SOC_RT700) += snd-soc-rt700.o obj-$(CONFIG_SND_SOC_RT711) += snd-soc-rt711.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c new file mode 100644 index 000000000000..314103601af3 --- /dev/null +++ b/sound/soc/codecs/rt700-sdw.c @@ -0,0 +1,551 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rt700-sdw.c -- rt700 ALSA SoC audio driver +// +// Copyright(c) 2019 Realtek Semiconductor Corp. +// +// + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/mod_devicetable.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include "rt700.h" +#include "rt700-sdw.h" + +static bool rt700_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x00e0: + case 0x00f0: + case 0x2000 ... 0x200e: + case 0x2012 ... 0x2016: + case 0x201a ... 0x2027: + case 0x2029 ... 0x202a: + case 0x202d ... 0x2034: + case 0x2200 ... 0x2204: + case 0x2206 ... 0x2212: + case 0x2220 ... 0x2223: + case 0x2230 ... 0x2231: + case 0x3000 ... 0x3fff: + case 0x7000 ... 0x7fff: + case 0x8300 ... 0x83ff: + case 0x9c00 ... 0x9cff: + case 0xb900 ... 0xb9ff: + case 0x75201a: + case 0x752045: + case 0x752046: + case 0x752048: + case 0x75204a: + case 0x75206b: + case 0x752080: + case 0x752081: + return true; + default: + return false; + } +} + +static bool rt700_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2009: + case 0x2016: + case 0x201b: + case 0x201c: + case 0x201d: + case 0x201f: + case 0x2021: + case 0x2023: + case 0x2230: + case 0x200b ... 0x200e: /* i2c read */ + case 0x2012 ... 0x2015: /* HD-A read */ + case 0x202d ... 0x202f: /* BRA */ + case 0x2201 ... 0x2212: /* i2c debug */ + case 0x2220 ... 0x2223: /* decoded HD-A */ + case 0x9c00 ... 0x9cff: + case 0xb900 ... 0xb9ff: + case 0xff01: + case 0x75201a: + case 0x752046: + case 0x752080: + case 0x752081: + return true; + default: + return false; + } +} + +static int rt700_sdw_read(void *context, unsigned int reg, unsigned int *val) +{ + struct device *dev = context; + struct rt700_priv *rt700 = dev_get_drvdata(dev); + unsigned int sdw_data_3, sdw_data_2, sdw_data_1, sdw_data_0; + unsigned int reg2 = 0, reg3 = 0, reg4 = 0, mask, nid, val2; + unsigned int is_hda_reg = 1, is_index_reg = 0; + int ret; + + if (reg > 0xffff) + is_index_reg = 1; + + mask = reg & 0xf000; + + if (is_index_reg) { /* index registers */ + val2 = reg & 0xff; + reg = reg >> 8; + nid = reg & 0xff; + ret = regmap_write(rt700->sdw_regmap, reg, 0); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt700->sdw_regmap, reg2, val2); + if (ret < 0) + return ret; + + reg3 = RT700_PRIV_DATA_R_H | nid; + ret = regmap_write(rt700->sdw_regmap, + reg3, ((*val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg4 = reg3 + 0x1000; + reg4 |= 0x80; + ret = regmap_write(rt700->sdw_regmap, reg4, (*val & 0xff)); + if (ret < 0) + return ret; + } else if (mask == 0x3000) { + reg += 0x8000; + ret = regmap_write(rt700->sdw_regmap, reg, *val); + if (ret < 0) + return ret; + } else if (mask == 0x7000) { + reg += 0x2000; + reg |= 0x800; + ret = regmap_write(rt700->sdw_regmap, + reg, ((*val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt700->sdw_regmap, reg2, (*val & 0xff)); + if (ret < 0) + return ret; + } else if ((reg & 0xff00) == 0x8300) { /* for R channel */ + reg2 = reg - 0x1000; + reg2 &= ~0x80; + ret = regmap_write(rt700->sdw_regmap, + reg2, ((*val >> 8) & 0xff)); + if (ret < 0) + return ret; + ret = regmap_write(rt700->sdw_regmap, reg, (*val & 0xff)); + if (ret < 0) + return ret; + } else if (mask == 0x9000) { + ret = regmap_write(rt700->sdw_regmap, + reg, ((*val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt700->sdw_regmap, reg2, (*val & 0xff)); + if (ret < 0) + return ret; + } else if (mask == 0xb000) { + ret = regmap_write(rt700->sdw_regmap, reg, *val); + if (ret < 0) + return ret; + } else { + ret = regmap_read(rt700->sdw_regmap, reg, val); + if (ret < 0) + return ret; + is_hda_reg = 0; + } + + if (is_hda_reg || is_index_reg) { + sdw_data_3 = 0; + sdw_data_2 = 0; + sdw_data_1 = 0; + sdw_data_0 = 0; + ret = regmap_read(rt700->sdw_regmap, + RT700_READ_HDA_3, &sdw_data_3); + if (ret < 0) + return ret; + ret = regmap_read(rt700->sdw_regmap, + RT700_READ_HDA_2, &sdw_data_2); + if (ret < 0) + return ret; + ret = regmap_read(rt700->sdw_regmap, + RT700_READ_HDA_1, &sdw_data_1); + if (ret < 0) + return ret; + ret = regmap_read(rt700->sdw_regmap, + RT700_READ_HDA_0, &sdw_data_0); + if (ret < 0) + return ret; + *val = ((sdw_data_3 & 0xff) << 24) | + ((sdw_data_2 & 0xff) << 16) | + ((sdw_data_1 & 0xff) << 8) | (sdw_data_0 & 0xff); + } + + if (is_hda_reg == 0) + dev_dbg(dev, "[%s] %04x => %08x\n", __func__, reg, *val); + else if (is_index_reg) + dev_dbg(dev, "[%s] %04x %04x %04x %04x => %08x\n", + __func__, reg, reg2, reg3, reg4, *val); + else + dev_dbg(dev, "[%s] %04x %04x => %08x\n", + __func__, reg, reg2, *val); + + return 0; +} + +static int rt700_sdw_write(void *context, unsigned int reg, unsigned int val) +{ + struct device *dev = context; + struct rt700_priv *rt700 = dev_get_drvdata(dev); + unsigned int reg2 = 0, reg3, reg4, nid, mask, val2; + unsigned int is_index_reg = 0; + int ret; + + if (reg > 0xffff) + is_index_reg = 1; + + mask = reg & 0xf000; + + if (is_index_reg) { /* index registers */ + val2 = reg & 0xff; + reg = reg >> 8; + nid = reg & 0xff; + ret = regmap_write(rt700->sdw_regmap, reg, 0); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt700->sdw_regmap, reg2, val2); + if (ret < 0) + return ret; + + reg3 = RT700_PRIV_DATA_W_H | nid; + ret = regmap_write(rt700->sdw_regmap, + reg3, ((val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg4 = reg3 + 0x1000; + reg4 |= 0x80; + ret = regmap_write(rt700->sdw_regmap, reg4, (val & 0xff)); + if (ret < 0) + return ret; + is_index_reg = 1; + } else if (reg < 0x4fff) { + ret = regmap_write(rt700->sdw_regmap, reg, val); + if (ret < 0) + return ret; + } else if (reg == 0xff01) { + ret = regmap_write(rt700->sdw_regmap, reg, val); + if (ret < 0) + return ret; + } else if (mask == 0x7000) { + ret = regmap_write(rt700->sdw_regmap, + reg, ((val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt700->sdw_regmap, reg2, (val & 0xff)); + if (ret < 0) + return ret; + } else if ((reg & 0xff00) == 0x8300) { /* for R channel */ + reg2 = reg - 0x1000; + reg2 &= ~0x80; + ret = regmap_write(rt700->sdw_regmap, + reg2, ((val >> 8) & 0xff)); + if (ret < 0) + return ret; + ret = regmap_write(rt700->sdw_regmap, reg, (val & 0xff)); + if (ret < 0) + return ret; + } + + if (reg2 == 0) + dev_dbg(dev, "[%s] %04x <= %04x\n", __func__, reg, val); + else if (is_index_reg) + dev_dbg(dev, "[%s] %04x %04x %04x %04x <= %04x %04x\n", + __func__, reg, reg2, reg3, reg4, val2, val); + else + dev_dbg(dev, "[%s] %04x %04x <= %04x\n", + __func__, reg, reg2, val); + + return 0; +} + +static const struct regmap_config rt700_regmap = { + .reg_bits = 24, + .val_bits = 32, + .readable_reg = rt700_readable_register, + .volatile_reg = rt700_volatile_register, + .max_register = 0x755800, + .reg_defaults = rt700_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rt700_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, + .reg_read = rt700_sdw_read, + .reg_write = rt700_sdw_write, +}; + +static const struct regmap_config rt700_sdw_regmap = { + .name = "sdw", + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt700_readable_register, + .max_register = 0xff01, + .cache_type = REGCACHE_NONE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt700_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt700_priv *rt700 = dev_get_drvdata(&slave->dev); + + /* Update the status */ + rt700->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + rt700->hw_init = false; + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt700->hw_init || rt700->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt700_io_init(&slave->dev, slave); +} + +static int rt700_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval, i, num_of_ports = 1; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->paging_support = false; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x14; /* BITMAP: 00010100 */ + prop->sink_ports = 0xA; /* BITMAP: 00001010 */ + + nval = hweight32(prop->source_ports); + num_of_ports += nval; + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), + GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + num_of_ports += nval; + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), + GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* Allocate port_ready based on num_of_ports */ + slave->port_ready = devm_kcalloc(&slave->dev, num_of_ports, + sizeof(*slave->port_ready), + GFP_KERNEL); + if (!slave->port_ready) + return -ENOMEM; + + /* Initialize completion */ + for (i = 0; i < num_of_ports; i++) + init_completion(&slave->port_ready[i]); + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + /* wake-up event */ + prop->wake_capable = 1; + + return 0; +} + +static int rt700_bus_config(struct sdw_slave *slave, + struct sdw_bus_params *params) +{ + struct rt700_priv *rt700 = dev_get_drvdata(&slave->dev); + int ret; + + memcpy(&rt700->params, params, sizeof(*params)); + + ret = rt700_clock_config(&slave->dev); + if (ret < 0) + dev_err(&slave->dev, "Invalid clk config"); + + return ret; +} + +static int rt700_interrupt_callback(struct sdw_slave *slave, + struct sdw_slave_intr_status *status) +{ + struct rt700_priv *rt700 = dev_get_drvdata(&slave->dev); + + dev_dbg(&slave->dev, + "%s control_port_stat=%x", __func__, status->control_port); + + if (status->control_port & 0x4) { + mod_delayed_work(system_power_efficient_wq, + &rt700->jack_detect_work, msecs_to_jiffies(250)); + } + + return 0; +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static struct sdw_slave_ops rt700_slave_ops = { + .read_prop = rt700_read_prop, + .interrupt_callback = rt700_interrupt_callback, + .update_status = rt700_update_status, + .bus_config = rt700_bus_config, +}; + +static int rt700_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *sdw_regmap, *regmap; + + /* Assign ops */ + slave->ops = &rt700_slave_ops; + + /* Regmap Initialization */ + sdw_regmap = devm_regmap_init_sdw(slave, &rt700_sdw_regmap); + if (!sdw_regmap) + return -EINVAL; + + regmap = devm_regmap_init(&slave->dev, NULL, + &slave->dev, &rt700_regmap); + if (!regmap) + return -EINVAL; + + rt700_init(&slave->dev, sdw_regmap, regmap, slave); + + return 0; +} + +static int rt700_sdw_remove(struct sdw_slave *slave) +{ + struct rt700_priv *rt700 = dev_get_drvdata(&slave->dev); + + if (rt700 && rt700->hw_init) { + cancel_delayed_work(&rt700->jack_detect_work); + cancel_delayed_work(&rt700->jack_btn_check_work); + } + + return 0; +} + +static const struct sdw_device_id rt700_id[] = { + SDW_SLAVE_ENTRY(0x025d, 0x700, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt700_id); + +static int rt700_dev_suspend(struct device *dev) +{ + struct rt700_priv *rt700 = dev_get_drvdata(dev); + + if (!rt700->hw_init) + return 0; + + regcache_cache_only(rt700->regmap, true); + + return 0; +} + +#define RT700_PROBE_TIMEOUT 2000 + +static int rt700_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt700_priv *rt700 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt700->hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT700_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt700->regmap, false); + regcache_sync_region(rt700->regmap, 0x3000, 0x8fff); + regcache_sync_region(rt700->regmap, 0x752010, 0x75206b); + + return 0; +} + +static const struct dev_pm_ops rt700_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt700_dev_suspend, rt700_dev_resume) + SET_RUNTIME_PM_OPS(rt700_dev_suspend, rt700_dev_resume, NULL) +}; + +static struct sdw_driver rt700_sdw_driver = { + .driver = { + .name = "rt700", + .owner = THIS_MODULE, + .pm = &rt700_pm, + }, + .probe = rt700_sdw_probe, + .remove = rt700_sdw_remove, + .ops = &rt700_slave_ops, + .id_table = rt700_id, +}; +module_sdw_driver(rt700_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT700 driver SDW"); +MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt700-sdw.h b/sound/soc/codecs/rt700-sdw.h new file mode 100644 index 000000000000..4ad0dcfd16fd --- /dev/null +++ b/sound/soc/codecs/rt700-sdw.h @@ -0,0 +1,335 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * rt700-sdw.h -- RT700 ALSA SoC audio driver header + * + * Copyright(c) 2019 Realtek Semiconductor Corp. + */ + +#ifndef __RT700_SDW_H__ +#define __RT700_SDW_H__ + +static const struct reg_default rt700_reg_defaults[] = { + { 0x0000, 0x0000 }, + { 0x0001, 0x0000 }, + { 0x0002, 0x0000 }, + { 0x0003, 0x0000 }, + { 0x0004, 0x0000 }, + { 0x0005, 0x0001 }, + { 0x0020, 0x0000 }, + { 0x0022, 0x0000 }, + { 0x0023, 0x0000 }, + { 0x0024, 0x0000 }, + { 0x0025, 0x0000 }, + { 0x0026, 0x0000 }, + { 0x0030, 0x0000 }, + { 0x0032, 0x0000 }, + { 0x0033, 0x0000 }, + { 0x0034, 0x0000 }, + { 0x0035, 0x0000 }, + { 0x0036, 0x0000 }, + { 0x0040, 0x0000 }, + { 0x0041, 0x0000 }, + { 0x0042, 0x0000 }, + { 0x0043, 0x0000 }, + { 0x0044, 0x0020 }, + { 0x0045, 0x0001 }, + { 0x0046, 0x0000 }, + { 0x0050, 0x0000 }, + { 0x0051, 0x0000 }, + { 0x0052, 0x0000 }, + { 0x0053, 0x0000 }, + { 0x0054, 0x0000 }, + { 0x0055, 0x0000 }, + { 0x0060, 0x0000 }, + { 0x0070, 0x0000 }, + { 0x00e0, 0x0000 }, + { 0x00f0, 0x0000 }, + { 0x0100, 0x0000 }, + { 0x0101, 0x0000 }, + { 0x0102, 0x0000 }, + { 0x0103, 0x0000 }, + { 0x0104, 0x0000 }, + { 0x0105, 0x0000 }, + { 0x0120, 0x0000 }, + { 0x0121, 0x0000 }, + { 0x0122, 0x0000 }, + { 0x0123, 0x0000 }, + { 0x0124, 0x0000 }, + { 0x0125, 0x0000 }, + { 0x0126, 0x0000 }, + { 0x0127, 0x0000 }, + { 0x0130, 0x0000 }, + { 0x0131, 0x0000 }, + { 0x0132, 0x0000 }, + { 0x0133, 0x0000 }, + { 0x0134, 0x0000 }, + { 0x0135, 0x0000 }, + { 0x0136, 0x0000 }, + { 0x0137, 0x0000 }, + { 0x0200, 0x0000 }, + { 0x0201, 0x0000 }, + { 0x0202, 0x0000 }, + { 0x0203, 0x0000 }, + { 0x0204, 0x0000 }, + { 0x0205, 0x0000 }, + { 0x0220, 0x0000 }, + { 0x0221, 0x0000 }, + { 0x0222, 0x0000 }, + { 0x0223, 0x0000 }, + { 0x0224, 0x0000 }, + { 0x0225, 0x0000 }, + { 0x0226, 0x0000 }, + { 0x0227, 0x0000 }, + { 0x0230, 0x0000 }, + { 0x0231, 0x0000 }, + { 0x0232, 0x0000 }, + { 0x0233, 0x0000 }, + { 0x0234, 0x0000 }, + { 0x0235, 0x0000 }, + { 0x0236, 0x0000 }, + { 0x0237, 0x0000 }, + { 0x0300, 0x0000 }, + { 0x0301, 0x0000 }, + { 0x0302, 0x0000 }, + { 0x0303, 0x0000 }, + { 0x0304, 0x0000 }, + { 0x0305, 0x0000 }, + { 0x0320, 0x0000 }, + { 0x0321, 0x0000 }, + { 0x0322, 0x0000 }, + { 0x0323, 0x0000 }, + { 0x0324, 0x0000 }, + { 0x0325, 0x0000 }, + { 0x0326, 0x0000 }, + { 0x0327, 0x0000 }, + { 0x0330, 0x0000 }, + { 0x0331, 0x0000 }, + { 0x0332, 0x0000 }, + { 0x0333, 0x0000 }, + { 0x0334, 0x0000 }, + { 0x0335, 0x0000 }, + { 0x0336, 0x0000 }, + { 0x0337, 0x0000 }, + { 0x0400, 0x0000 }, + { 0x0401, 0x0000 }, + { 0x0402, 0x0000 }, + { 0x0403, 0x0000 }, + { 0x0404, 0x0000 }, + { 0x0405, 0x0000 }, + { 0x0420, 0x0000 }, + { 0x0421, 0x0000 }, + { 0x0422, 0x0000 }, + { 0x0423, 0x0000 }, + { 0x0424, 0x0000 }, + { 0x0425, 0x0000 }, + { 0x0426, 0x0000 }, + { 0x0427, 0x0000 }, + { 0x0430, 0x0000 }, + { 0x0431, 0x0000 }, + { 0x0432, 0x0000 }, + { 0x0433, 0x0000 }, + { 0x0434, 0x0000 }, + { 0x0435, 0x0000 }, + { 0x0436, 0x0000 }, + { 0x0437, 0x0000 }, + { 0x0500, 0x0000 }, + { 0x0501, 0x0000 }, + { 0x0502, 0x0000 }, + { 0x0503, 0x0000 }, + { 0x0504, 0x0000 }, + { 0x0505, 0x0000 }, + { 0x0520, 0x0000 }, + { 0x0521, 0x0000 }, + { 0x0522, 0x0000 }, + { 0x0523, 0x0000 }, + { 0x0524, 0x0000 }, + { 0x0525, 0x0000 }, + { 0x0526, 0x0000 }, + { 0x0527, 0x0000 }, + { 0x0530, 0x0000 }, + { 0x0531, 0x0000 }, + { 0x0532, 0x0000 }, + { 0x0533, 0x0000 }, + { 0x0534, 0x0000 }, + { 0x0535, 0x0000 }, + { 0x0536, 0x0000 }, + { 0x0537, 0x0000 }, + { 0x0600, 0x0000 }, + { 0x0601, 0x0000 }, + { 0x0602, 0x0000 }, + { 0x0603, 0x0000 }, + { 0x0604, 0x0000 }, + { 0x0605, 0x0000 }, + { 0x0620, 0x0000 }, + { 0x0621, 0x0000 }, + { 0x0622, 0x0000 }, + { 0x0623, 0x0000 }, + { 0x0624, 0x0000 }, + { 0x0625, 0x0000 }, + { 0x0626, 0x0000 }, + { 0x0627, 0x0000 }, + { 0x0630, 0x0000 }, + { 0x0631, 0x0000 }, + { 0x0632, 0x0000 }, + { 0x0633, 0x0000 }, + { 0x0634, 0x0000 }, + { 0x0635, 0x0000 }, + { 0x0636, 0x0000 }, + { 0x0637, 0x0000 }, + { 0x0700, 0x0000 }, + { 0x0701, 0x0000 }, + { 0x0702, 0x0000 }, + { 0x0703, 0x0000 }, + { 0x0704, 0x0000 }, + { 0x0705, 0x0000 }, + { 0x0720, 0x0000 }, + { 0x0721, 0x0000 }, + { 0x0722, 0x0000 }, + { 0x0723, 0x0000 }, + { 0x0724, 0x0000 }, + { 0x0725, 0x0000 }, + { 0x0726, 0x0000 }, + { 0x0727, 0x0000 }, + { 0x0730, 0x0000 }, + { 0x0731, 0x0000 }, + { 0x0732, 0x0000 }, + { 0x0733, 0x0000 }, + { 0x0734, 0x0000 }, + { 0x0735, 0x0000 }, + { 0x0736, 0x0000 }, + { 0x0737, 0x0000 }, + { 0x0800, 0x0000 }, + { 0x0801, 0x0000 }, + { 0x0802, 0x0000 }, + { 0x0803, 0x0000 }, + { 0x0804, 0x0000 }, + { 0x0805, 0x0000 }, + { 0x0820, 0x0000 }, + { 0x0821, 0x0000 }, + { 0x0822, 0x0000 }, + { 0x0823, 0x0000 }, + { 0x0824, 0x0000 }, + { 0x0825, 0x0000 }, + { 0x0826, 0x0000 }, + { 0x0827, 0x0000 }, + { 0x0830, 0x0000 }, + { 0x0831, 0x0000 }, + { 0x0832, 0x0000 }, + { 0x0833, 0x0000 }, + { 0x0834, 0x0000 }, + { 0x0835, 0x0000 }, + { 0x0836, 0x0000 }, + { 0x0837, 0x0000 }, + { 0x0f00, 0x0000 }, + { 0x0f01, 0x0000 }, + { 0x0f02, 0x0000 }, + { 0x0f03, 0x0000 }, + { 0x0f04, 0x0000 }, + { 0x0f05, 0x0000 }, + { 0x0f20, 0x0000 }, + { 0x0f21, 0x0000 }, + { 0x0f22, 0x0000 }, + { 0x0f23, 0x0000 }, + { 0x0f24, 0x0000 }, + { 0x0f25, 0x0000 }, + { 0x0f26, 0x0000 }, + { 0x0f27, 0x0000 }, + { 0x0f30, 0x0000 }, + { 0x0f31, 0x0000 }, + { 0x0f32, 0x0000 }, + { 0x0f33, 0x0000 }, + { 0x0f34, 0x0000 }, + { 0x0f35, 0x0000 }, + { 0x0f36, 0x0000 }, + { 0x0f37, 0x0000 }, + { 0x2000, 0x0000 }, + { 0x2001, 0x0000 }, + { 0x2002, 0x0000 }, + { 0x2003, 0x0000 }, + { 0x2004, 0x0000 }, + { 0x2005, 0x0000 }, + { 0x2006, 0x0000 }, + { 0x2007, 0x0000 }, + { 0x2008, 0x0000 }, + { 0x2009, 0x0003 }, + { 0x200a, 0x0003 }, + { 0x200b, 0x0000 }, + { 0x200c, 0x0000 }, + { 0x200d, 0x0000 }, + { 0x200e, 0x0000 }, + { 0x2012, 0x0000 }, + { 0x2013, 0x0000 }, + { 0x2014, 0x0000 }, + { 0x2015, 0x0000 }, + { 0x2016, 0x0000 }, + { 0x201a, 0x0000 }, + { 0x201b, 0x0000 }, + { 0x201c, 0x0000 }, + { 0x201d, 0x0000 }, + { 0x201e, 0x0000 }, + { 0x201f, 0x0000 }, + { 0x2020, 0x0000 }, + { 0x2021, 0x0000 }, + { 0x2022, 0x0000 }, + { 0x2023, 0x0000 }, + { 0x2024, 0x0000 }, + { 0x2025, 0x0002 }, + { 0x2026, 0x0000 }, + { 0x2027, 0x0000 }, + { 0x2029, 0x0000 }, + { 0x202a, 0x0000 }, + { 0x202d, 0x0000 }, + { 0x202e, 0x0000 }, + { 0x202f, 0x0000 }, + { 0x2030, 0x0000 }, + { 0x2031, 0x0000 }, + { 0x2032, 0x0000 }, + { 0x2033, 0x0000 }, + { 0x2034, 0x0000 }, + { 0x2200, 0x0000 }, + { 0x2201, 0x0000 }, + { 0x2202, 0x0000 }, + { 0x2203, 0x0000 }, + { 0x2204, 0x0000 }, + { 0x2206, 0x0000 }, + { 0x2207, 0x0000 }, + { 0x2208, 0x0000 }, + { 0x2209, 0x0000 }, + { 0x220a, 0x0000 }, + { 0x220b, 0x0000 }, + { 0x220c, 0x0000 }, + { 0x220d, 0x0000 }, + { 0x220e, 0x0000 }, + { 0x220f, 0x0000 }, + { 0x2211, 0x0000 }, + { 0x2212, 0x0000 }, + { 0x2220, 0x0000 }, + { 0x2221, 0x0000 }, + { 0x2222, 0x0000 }, + { 0x2223, 0x0000 }, + { 0x2230, 0x0000 }, + { 0x2231, 0x0000 }, + { 0x3121, 0x0001 }, + { 0x3122, 0x0000 }, + { 0x3123, 0x0000 }, + { 0x7303, 0x0057 }, + { 0x7303, 0x0057 }, + { 0x8383, 0x0057 }, + { 0x7308, 0x0097 }, + { 0x8388, 0x0097 }, + { 0x7309, 0x0097 }, + { 0x8389, 0x0097 }, + { 0x7312, 0x0000 }, + { 0x8392, 0x0000 }, + { 0x7313, 0x0000 }, + { 0x8393, 0x0000 }, + { 0x7319, 0x0000 }, + { 0x8399, 0x0000 }, + { 0x75201a, 0x8003 }, + { 0x752045, 0x5289 }, + { 0x752048, 0xd049 }, + { 0x75204a, 0xa83b }, + { 0x75206b, 0x5064 }, +}; + +#endif /* __RT700_H__ */ diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c new file mode 100644 index 000000000000..b1830c1ebf8a --- /dev/null +++ b/sound/soc/codecs/rt700.c @@ -0,0 +1,1238 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rt700.c -- rt700 ALSA SoC audio driver +// +// Copyright(c) 2019 Realtek Semiconductor Corp. +// +// + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm_runtime.h> +#include <linux/pm.h> +#include <linux/soundwire/sdw.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/hda_verbs.h> +#include <sound/jack.h> + +#include "rt700.h" + +static int rt700_index_write(struct regmap *regmap, + unsigned int reg, unsigned int value) +{ + int ret; + unsigned int addr = (RT700_PRIV_INDEX_W_H << 8) | reg; + + ret = regmap_write(regmap, addr, value); + if (ret < 0) + pr_err("Failed to set private value: %06x <= %04x ret=%d\n", + addr, value, ret); + + return ret; +} + +static int rt700_index_read(struct regmap *regmap, + unsigned int reg, unsigned int *value) +{ + int ret; + unsigned int addr = (RT700_PRIV_INDEX_W_H << 8) | reg; + + *value = 0; + ret = regmap_read(regmap, addr, value); + if (ret < 0) + pr_err("Failed to get private value: %06x => %04x ret=%d\n", + addr, *value, ret); + + return ret; +} + +static unsigned int rt700_button_detect(struct rt700_priv *rt700) +{ + unsigned int btn_type = 0, val80, val81; + int ret; + + ret = rt700_index_read(rt700->regmap, RT700_IRQ_FLAG_TABLE1, &val80); + if (ret < 0) + goto read_error; + ret = rt700_index_read(rt700->regmap, RT700_IRQ_FLAG_TABLE2, &val81); + if (ret < 0) + goto read_error; + + val80 &= 0x0381; + val81 &= 0xff00; + + switch (val80) { + case 0x0200: + case 0x0100: + case 0x0080: + btn_type |= SND_JACK_BTN_0; + break; + case 0x0001: + btn_type |= SND_JACK_BTN_3; + break; + } + switch (val81) { + case 0x8000: + case 0x4000: + case 0x2000: + btn_type |= SND_JACK_BTN_1; + break; + case 0x1000: + case 0x0800: + case 0x0400: + btn_type |= SND_JACK_BTN_2; + break; + case 0x0200: + case 0x0100: + btn_type |= SND_JACK_BTN_3; + break; + } +read_error: + return btn_type; +} + +static int rt700_headset_detect(struct rt700_priv *rt700) +{ + unsigned int buf, loop = 0; + int ret; + unsigned int jack_status = 0, reg; + + ret = rt700_index_read(rt700->regmap, + RT700_COMBO_JACK_AUTO_CTL2, &buf); + if (ret < 0) + goto io_error; + + while (loop < 500 && + (buf & RT700_COMBOJACK_AUTO_DET_STATUS) == 0) { + loop++; + + usleep_range(9000, 10000); + ret = rt700_index_read(rt700->regmap, + RT700_COMBO_JACK_AUTO_CTL2, &buf); + if (ret < 0) + goto io_error; + + reg = RT700_VERB_GET_PIN_SENSE | RT700_HP_OUT; + ret = regmap_read(rt700->regmap, reg, &jack_status); + if ((jack_status & (1 << 31)) == 0) + goto remove_error; + } + + if (loop >= 500) + goto to_error; + + if (buf & RT700_COMBOJACK_AUTO_DET_TRS) + rt700->jack_type = SND_JACK_HEADPHONE; + else if ((buf & RT700_COMBOJACK_AUTO_DET_CTIA) || + (buf & RT700_COMBOJACK_AUTO_DET_OMTP)) + rt700->jack_type = SND_JACK_HEADSET; + + return 0; + +to_error: + ret = -ETIMEDOUT; + pr_err_ratelimited("Time-out error in %s\n", __func__); + return ret; +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); + return ret; +remove_error: + pr_err_ratelimited("Jack removal in %s\n", __func__); + return -ENODEV; +} + +static void rt700_jack_detect_handler(struct work_struct *work) +{ + struct rt700_priv *rt700 = + container_of(work, struct rt700_priv, jack_detect_work.work); + int btn_type = 0, ret; + unsigned int jack_status = 0, reg; + + if (!rt700->hs_jack) + return; + + if (!rt700->component->card->instantiated) + return; + + reg = RT700_VERB_GET_PIN_SENSE | RT700_HP_OUT; + ret = regmap_read(rt700->regmap, reg, &jack_status); + if (ret < 0) + goto io_error; + + /* pin attached */ + if (jack_status & (1 << 31)) { + /* jack in */ + if (rt700->jack_type == 0) { + ret = rt700_headset_detect(rt700); + if (ret < 0) + return; + if (rt700->jack_type == SND_JACK_HEADSET) + btn_type = rt700_button_detect(rt700); + } else if (rt700->jack_type == SND_JACK_HEADSET) { + /* jack is already in, report button event */ + btn_type = rt700_button_detect(rt700); + } + } else { + /* jack out */ + rt700->jack_type = 0; + } + + dev_dbg(&rt700->slave->dev, + "in %s, jack_type=0x%x\n", __func__, rt700->jack_type); + dev_dbg(&rt700->slave->dev, + "in %s, btn_type=0x%x\n", __func__, btn_type); + + snd_soc_jack_report(rt700->hs_jack, rt700->jack_type | btn_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (btn_type) { + /* button released */ + snd_soc_jack_report(rt700->hs_jack, rt700->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + mod_delayed_work(system_power_efficient_wq, + &rt700->jack_btn_check_work, msecs_to_jiffies(200)); + } + + return; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); +} + +static void rt700_btn_check_handler(struct work_struct *work) +{ + struct rt700_priv *rt700 = container_of(work, struct rt700_priv, + jack_btn_check_work.work); + int btn_type = 0, ret; + unsigned int jack_status = 0, reg; + + reg = RT700_VERB_GET_PIN_SENSE | RT700_HP_OUT; + ret = regmap_read(rt700->regmap, reg, &jack_status); + if (ret < 0) + goto io_error; + + /* pin attached */ + if (jack_status & (1 << 31)) { + if (rt700->jack_type == SND_JACK_HEADSET) { + /* jack is already in, report button event */ + btn_type = rt700_button_detect(rt700); + } + } else { + rt700->jack_type = 0; + } + + /* cbj comparator */ + ret = rt700_index_read(rt700->regmap, RT700_COMBO_JACK_AUTO_CTL2, &reg); + if (ret < 0) + goto io_error; + + if ((reg & 0xf0) == 0xf0) + btn_type = 0; + + dev_dbg(&rt700->slave->dev, + "%s, btn_type=0x%x\n", __func__, btn_type); + snd_soc_jack_report(rt700->hs_jack, rt700->jack_type | btn_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (btn_type) { + /* button released */ + snd_soc_jack_report(rt700->hs_jack, rt700->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + mod_delayed_work(system_power_efficient_wq, + &rt700->jack_btn_check_work, msecs_to_jiffies(200)); + } + + return; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); +} + +static void rt700_jack_init(struct rt700_priv *rt700) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(rt700->component); + + /* power on */ + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt700->regmap, + RT700_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + if (rt700->hs_jack) { + /* Enable Jack Detection */ + regmap_write(rt700->regmap, + RT700_SET_MIC2_UNSOLICITED_ENABLE, 0x82); + regmap_write(rt700->regmap, + RT700_SET_HP_UNSOLICITED_ENABLE, 0x81); + regmap_write(rt700->regmap, + RT700_SET_INLINE_UNSOLICITED_ENABLE, 0x83); + rt700_index_write(rt700->regmap, 0x10, 0x2420); + rt700_index_write(rt700->regmap, 0x19, 0x2e11); + + dev_dbg(&rt700->slave->dev, "in %s enable\n", __func__); + + mod_delayed_work(system_power_efficient_wq, + &rt700->jack_detect_work, msecs_to_jiffies(250)); + } else { + regmap_write(rt700->regmap, + RT700_SET_MIC2_UNSOLICITED_ENABLE, 0x00); + regmap_write(rt700->regmap, + RT700_SET_HP_UNSOLICITED_ENABLE, 0x00); + regmap_write(rt700->regmap, + RT700_SET_INLINE_UNSOLICITED_ENABLE, 0x00); + + dev_dbg(&rt700->slave->dev, "in %s disable\n", __func__); + } + + /* power off */ + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt700->regmap, + RT700_SET_AUDIO_POWER_STATE, AC_PWRST_D3); +} + +static int rt700_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + + rt700->hs_jack = hs_jack; + + if (!rt700->hw_init) { + dev_dbg(&rt700->slave->dev, + "%s hw_init not ready yet\n", __func__); + return 0; + } + + rt700_jack_init(rt700); + + return 0; +} + +static void rt700_get_gain(struct rt700_priv *rt700, unsigned int addr_h, + unsigned int addr_l, unsigned int val_h, + unsigned int *r_val, unsigned int *l_val) +{ + /* R Channel */ + *r_val = (val_h << 8); + regmap_read(rt700->regmap, addr_l, r_val); + + /* L Channel */ + val_h |= 0x20; + *l_val = (val_h << 8); + regmap_read(rt700->regmap, addr_h, l_val); +} + +/* For Verb-Set Amplifier Gain (Verb ID = 3h) */ +static int rt700_set_amp_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + unsigned int addr_h, addr_l, val_h, val_ll, val_lr; + unsigned int read_ll, read_rl; + int i; + + /* Can't use update bit function, so read the original value first */ + addr_h = mc->reg; + addr_l = mc->rreg; + if (mc->shift == RT700_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt700_get_gain(rt700, addr_h, addr_l, val_h, &read_rl, &read_ll); + + /* L Channel */ + if (mc->invert) { + /* for mute */ + val_ll = (mc->max - ucontrol->value.integer.value[0]) << 7; + /* keep gain */ + read_ll = read_ll & 0x7f; + val_ll |= read_ll; + } else { + /* for gain */ + val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); + if (val_ll > mc->max) + val_ll = mc->max; + /* keep mute status */ + read_ll = read_ll & 0x80; + val_ll |= read_ll; + } + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt700->regmap, + RT700_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* R Channel */ + if (mc->invert) { + /* for mute */ + val_lr = (mc->max - ucontrol->value.integer.value[1]) << 7; + /* keep gain */ + read_rl = read_rl & 0x7f; + val_lr |= read_rl; + } else { + /* for gain */ + val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); + if (val_lr > mc->max) + val_lr = mc->max; + /* keep mute status */ + read_rl = read_rl & 0x80; + val_lr |= read_rl; + } + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << mc->shift) | (3 << 4); + regmap_write(rt700->regmap, + addr_h, (val_h << 8 | val_ll)); + regmap_write(rt700->regmap, + addr_l, (val_h << 8 | val_ll)); + } else { + /* Lch*/ + val_h = (1 << mc->shift) | (1 << 5); + regmap_write(rt700->regmap, + addr_h, (val_h << 8 | val_ll)); + + /* Rch */ + val_h = (1 << mc->shift) | (1 << 4); + regmap_write(rt700->regmap, + addr_l, (val_h << 8 | val_lr)); + } + /* check result */ + if (mc->shift == RT700_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt700_get_gain(rt700, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt700->regmap, + RT700_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + return 0; +} + +static int rt700_set_amp_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int addr_h, addr_l, val_h; + unsigned int read_ll, read_rl; + + addr_h = mc->reg; + addr_l = mc->rreg; + if (mc->shift == RT700_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt700_get_gain(rt700, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (mc->invert) { + /* for mute status */ + read_ll = !((read_ll & 0x80) >> RT700_MUTE_SFT); + read_rl = !((read_rl & 0x80) >> RT700_MUTE_SFT); + } else { + /* for gain */ + read_ll = read_ll & 0x7f; + read_rl = read_rl & 0x7f; + } + ucontrol->value.integer.value[0] = read_ll; + ucontrol->value.integer.value[1] = read_rl; + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt700_snd_controls[] = { + SOC_DOUBLE_R_EXT_TLV("DAC Front Playback Volume", + RT700_SET_GAIN_DAC1_H, RT700_SET_GAIN_DAC1_L, + RT700_DIR_OUT_SFT, 0x57, 0, + rt700_set_amp_gain_get, rt700_set_amp_gain_put, out_vol_tlv), + SOC_DOUBLE_R_EXT("ADC 08 Capture Switch", + RT700_SET_GAIN_ADC2_H, RT700_SET_GAIN_ADC2_L, + RT700_DIR_IN_SFT, 1, 1, + rt700_set_amp_gain_get, rt700_set_amp_gain_put), + SOC_DOUBLE_R_EXT("ADC 09 Capture Switch", + RT700_SET_GAIN_ADC1_H, RT700_SET_GAIN_ADC1_L, + RT700_DIR_IN_SFT, 1, 1, + rt700_set_amp_gain_get, rt700_set_amp_gain_put), + SOC_DOUBLE_R_EXT_TLV("ADC 08 Capture Volume", + RT700_SET_GAIN_ADC2_H, RT700_SET_GAIN_ADC2_L, + RT700_DIR_IN_SFT, 0x3f, 0, + rt700_set_amp_gain_get, rt700_set_amp_gain_put, in_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("ADC 09 Capture Volume", + RT700_SET_GAIN_ADC1_H, RT700_SET_GAIN_ADC1_L, + RT700_DIR_IN_SFT, 0x3f, 0, + rt700_set_amp_gain_get, rt700_set_amp_gain_put, in_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("AMIC Volume", + RT700_SET_GAIN_AMIC_H, RT700_SET_GAIN_AMIC_L, + RT700_DIR_IN_SFT, 3, 0, + rt700_set_amp_gain_get, rt700_set_amp_gain_put, mic_vol_tlv), +}; + +static int rt700_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + unsigned int reg, val = 0, nid; + int ret; + + if (strstr(ucontrol->id.name, "HPO Mux")) + nid = RT700_HP_OUT; + else if (strstr(ucontrol->id.name, "ADC 22 Mux")) + nid = RT700_MIXER_IN1; + else if (strstr(ucontrol->id.name, "ADC 23 Mux")) + nid = RT700_MIXER_IN2; + else + return -EINVAL; + + /* vid = 0xf01 */ + reg = RT700_VERB_SET_CONNECT_SEL | nid; + ret = regmap_read(rt700->regmap, reg, &val); + if (ret < 0) + return ret; + + ucontrol->value.enumerated.item[0] = val; + + return 0; +} + +static int rt700_mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int val, val2 = 0, change, reg, nid; + int ret; + + if (item[0] >= e->items) + return -EINVAL; + + if (strstr(ucontrol->id.name, "HPO Mux")) + nid = RT700_HP_OUT; + else if (strstr(ucontrol->id.name, "ADC 22 Mux")) + nid = RT700_MIXER_IN1; + else if (strstr(ucontrol->id.name, "ADC 23 Mux")) + nid = RT700_MIXER_IN2; + else + return -EINVAL; + + /* Verb ID = 0x701h */ + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + + reg = RT700_VERB_SET_CONNECT_SEL | nid; + ret = regmap_read(rt700->regmap, reg, &val2); + if (ret < 0) + return ret; + + if (val == val2) + change = 0; + else + change = 1; + + if (change) { + reg = RT700_VERB_SET_CONNECT_SEL | nid; + regmap_write(rt700->regmap, reg, val); + } + + snd_soc_dapm_mux_update_power(dapm, kcontrol, + item[0], e, NULL); + + return change; +} + +static const char * const adc_mux_text[] = { + "MIC2", + "LINE1", + "LINE2", + "DMIC", +}; + +static SOC_ENUM_SINGLE_DECL( + rt700_adc22_enum, SND_SOC_NOPM, 0, adc_mux_text); + +static SOC_ENUM_SINGLE_DECL( + rt700_adc23_enum, SND_SOC_NOPM, 0, adc_mux_text); + +static const struct snd_kcontrol_new rt700_adc22_mux = + SOC_DAPM_ENUM_EXT("ADC 22 Mux", rt700_adc22_enum, + rt700_mux_get, rt700_mux_put); + +static const struct snd_kcontrol_new rt700_adc23_mux = + SOC_DAPM_ENUM_EXT("ADC 23 Mux", rt700_adc23_enum, + rt700_mux_get, rt700_mux_put); + +static const char * const out_mux_text[] = { + "Front", + "Surround", +}; + +static SOC_ENUM_SINGLE_DECL( + rt700_hp_enum, SND_SOC_NOPM, 0, out_mux_text); + +static const struct snd_kcontrol_new rt700_hp_mux = + SOC_DAPM_ENUM_EXT("HP Mux", rt700_hp_enum, + rt700_mux_get, rt700_mux_put); + +static int rt700_dac_front_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 rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt700->regmap, + RT700_SET_STREAMID_DAC1, 0x10); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt700->regmap, + RT700_SET_STREAMID_DAC1, 0x00); + break; + } + return 0; +} + +static int rt700_dac_surround_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 rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt700->regmap, + RT700_SET_STREAMID_DAC2, 0x10); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt700->regmap, + RT700_SET_STREAMID_DAC2, 0x00); + break; + } + return 0; +} + +static int rt700_adc_09_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 rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt700->regmap, + RT700_SET_STREAMID_ADC1, 0x10); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt700->regmap, + RT700_SET_STREAMID_ADC1, 0x00); + break; + } + return 0; +} + +static int rt700_adc_08_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 rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt700->regmap, + RT700_SET_STREAMID_ADC2, 0x10); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt700->regmap, + RT700_SET_STREAMID_ADC2, 0x00); + break; + } + return 0; +} + +static int rt700_hpo_mux_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 rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + unsigned int val_h = (1 << RT700_DIR_OUT_SFT) | (0x3 << 4); + unsigned int val_l; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + val_l = 0x00; + regmap_write(rt700->regmap, + RT700_SET_GAIN_HP_H, (val_h << 8 | val_l)); + break; + case SND_SOC_DAPM_PRE_PMD: + val_l = (1 << RT700_MUTE_SFT); + regmap_write(rt700->regmap, + RT700_SET_GAIN_HP_H, (val_h << 8 | val_l)); + usleep_range(50000, 55000); + break; + } + return 0; +} + +static int rt700_spk_pga_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 rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + unsigned int val_h = (1 << RT700_DIR_OUT_SFT) | (0x3 << 4); + unsigned int val_l; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + val_l = 0x00; + regmap_write(rt700->regmap, + RT700_SET_GAIN_SPK_H, (val_h << 8 | val_l)); + break; + case SND_SOC_DAPM_PRE_PMD: + val_l = (1 << RT700_MUTE_SFT); + regmap_write(rt700->regmap, + RT700_SET_GAIN_SPK_H, (val_h << 8 | val_l)); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget rt700_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("HP"), + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("LINE2"), + SND_SOC_DAPM_DAC_E("DAC Front", NULL, SND_SOC_NOPM, 0, 0, + rt700_dac_front_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC Surround", NULL, SND_SOC_NOPM, 0, 0, + rt700_dac_surround_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MUX_E("HPO Mux", SND_SOC_NOPM, 0, 0, &rt700_hp_mux, + rt700_hpo_mux_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("SPK PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + rt700_spk_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("ADC 09", NULL, SND_SOC_NOPM, 0, 0, + rt700_adc_09_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("ADC 08", NULL, SND_SOC_NOPM, 0, 0, + rt700_adc_08_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MUX("ADC 22 Mux", SND_SOC_NOPM, 0, 0, + &rt700_adc22_mux), + SND_SOC_DAPM_MUX("ADC 23 Mux", SND_SOC_NOPM, 0, 0, + &rt700_adc23_mux), + SND_SOC_DAPM_AIF_IN("DP1RX", "DP1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DP3RX", "DP3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP2TX", "DP2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP4TX", "DP4 Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route rt700_audio_map[] = { + {"DAC Front", NULL, "DP1RX"}, + {"DAC Surround", NULL, "DP3RX"}, + {"DP2TX", NULL, "ADC 09"}, + {"DP4TX", NULL, "ADC 08"}, + {"ADC 09", NULL, "ADC 22 Mux"}, + {"ADC 08", NULL, "ADC 23 Mux"}, + {"ADC 22 Mux", "DMIC", "DMIC1"}, + {"ADC 22 Mux", "LINE1", "LINE1"}, + {"ADC 22 Mux", "LINE2", "LINE2"}, + {"ADC 22 Mux", "MIC2", "MIC2"}, + {"ADC 23 Mux", "DMIC", "DMIC2"}, + {"ADC 23 Mux", "LINE1", "LINE1"}, + {"ADC 23 Mux", "LINE2", "LINE2"}, + {"ADC 23 Mux", "MIC2", "MIC2"}, + {"HPO Mux", "Front", "DAC Front"}, + {"HPO Mux", "Surround", "DAC Surround"}, + {"HP", NULL, "HPO Mux"}, + {"SPK PGA", NULL, "DAC Front"}, + {"SPK", NULL, "SPK PGA"}, +}; + +static int rt700_probe(struct snd_soc_component *component) +{ + struct rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + + rt700->component = component; + + return 0; +} + +static int rt700_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + regmap_write(rt700->regmap, + RT700_SET_AUDIO_POWER_STATE, + AC_PWRST_D0); + } + break; + + case SND_SOC_BIAS_STANDBY: + regmap_write(rt700->regmap, + RT700_SET_AUDIO_POWER_STATE, + AC_PWRST_D3); + break; + + default: + break; + } + dapm->bias_level = level; + return 0; +} + +static const struct snd_soc_component_driver soc_codec_dev_rt700 = { + .probe = rt700_probe, + .set_bias_level = rt700_set_bias_level, + .controls = rt700_snd_controls, + .num_controls = ARRAY_SIZE(rt700_snd_controls), + .dapm_widgets = rt700_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt700_dapm_widgets), + .dapm_routes = rt700_audio_map, + .num_dapm_routes = ARRAY_SIZE(rt700_audio_map), + .set_jack = rt700_set_jack_detect, +}; + +static int rt700_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + struct sdw_stream_data *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = (struct sdw_stream_runtime *)sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void rt700_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int rt700_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int retval, port, num_channels; + unsigned int val = 0; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!rt700->slave) + return -EINVAL; + + /* SoundWire specific configuration */ + /* This code assumes port 1 for playback and port 2 for capture */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port = 1; + } else { + direction = SDW_DATA_DIR_TX; + port = 2; + } + + switch (dai->id) { + case RT700_AIF1: + break; + case RT700_AIF2: + port += 2; + break; + default: + dev_err(component->dev, "Invalid DAI id %d\n", dai->id); + return -EINVAL; + } + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = (1 << (num_channels)) - 1; + port_config.num = port; + + retval = sdw_stream_add_slave(rt700->slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(component->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 8: + break; + case 16: + val |= (0x1 << 4); + break; + case 20: + val |= (0x2 << 4); + break; + case 24: + val |= (0x3 << 4); + break; + case 32: + val |= (0x4 << 4); + break; + default: + return -EINVAL; + } + + /* 48Khz */ + regmap_write(rt700->regmap, RT700_DAC_FORMAT_H, val); + regmap_write(rt700->regmap, RT700_ADC_FORMAT_H, val); + + return retval; +} + +static int rt700_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt700->slave) + return -EINVAL; + + sdw_stream_remove_slave(rt700->slave, stream->sdw_stream); + return 0; +} + +#define RT700_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT700_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static struct snd_soc_dai_ops rt700_ops = { + .hw_params = rt700_pcm_hw_params, + .hw_free = rt700_pcm_hw_free, + .set_sdw_stream = rt700_set_sdw_stream, + .shutdown = rt700_shutdown, +}; + +static struct snd_soc_dai_driver rt700_dai[] = { + { + .name = "rt700-aif1", + .id = RT700_AIF1, + .playback = { + .stream_name = "DP1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT700_STEREO_RATES, + .formats = RT700_FORMATS, + }, + .capture = { + .stream_name = "DP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT700_STEREO_RATES, + .formats = RT700_FORMATS, + }, + .ops = &rt700_ops, + }, + { + .name = "rt700-aif2", + .id = RT700_AIF2, + .playback = { + .stream_name = "DP3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT700_STEREO_RATES, + .formats = RT700_FORMATS, + }, + .capture = { + .stream_name = "DP4 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT700_STEREO_RATES, + .formats = RT700_FORMATS, + }, + .ops = &rt700_ops, + }, +}; + +/* Bus clock frequency */ +#define RT700_CLK_FREQ_9600000HZ 9600000 +#define RT700_CLK_FREQ_12000000HZ 12000000 +#define RT700_CLK_FREQ_6000000HZ 6000000 +#define RT700_CLK_FREQ_4800000HZ 4800000 +#define RT700_CLK_FREQ_2400000HZ 2400000 +#define RT700_CLK_FREQ_12288000HZ 12288000 + +int rt700_clock_config(struct device *dev) +{ + struct rt700_priv *rt700 = dev_get_drvdata(dev); + unsigned int clk_freq, value; + + clk_freq = (rt700->params.curr_dr_freq >> 1); + + switch (clk_freq) { + case RT700_CLK_FREQ_12000000HZ: + value = 0x0; + break; + case RT700_CLK_FREQ_6000000HZ: + value = 0x1; + break; + case RT700_CLK_FREQ_9600000HZ: + value = 0x2; + break; + case RT700_CLK_FREQ_4800000HZ: + value = 0x3; + break; + case RT700_CLK_FREQ_2400000HZ: + value = 0x4; + break; + case RT700_CLK_FREQ_12288000HZ: + value = 0x5; + break; + default: + return -EINVAL; + } + + regmap_write(rt700->regmap, 0xe0, value); + regmap_write(rt700->regmap, 0xf0, value); + + dev_dbg(dev, "%s complete, clk_freq=%d\n", __func__, clk_freq); + + return 0; +} + +int rt700_init(struct device *dev, struct regmap *sdw_regmap, + struct regmap *regmap, struct sdw_slave *slave) + +{ + struct rt700_priv *rt700; + int ret; + + rt700 = devm_kzalloc(dev, sizeof(*rt700), GFP_KERNEL); + if (!rt700) + return -ENOMEM; + + dev_set_drvdata(dev, rt700); + rt700->slave = slave; + rt700->sdw_regmap = sdw_regmap; + rt700->regmap = regmap; + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt700->hw_init = false; + rt700->first_hw_init = false; + + ret = devm_snd_soc_register_component(dev, + &soc_codec_dev_rt700, + rt700_dai, + ARRAY_SIZE(rt700_dai)); + + dev_dbg(&slave->dev, "%s\n", __func__); + + return ret; +} + +int rt700_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt700_priv *rt700 = dev_get_drvdata(dev); + + if (rt700->hw_init) + return 0; + + if (rt700->first_hw_init) { + regcache_cache_only(rt700->regmap, false); + regcache_cache_bypass(rt700->regmap, true); + } + + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + if (!rt700->first_hw_init) { + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + /* reset */ + regmap_write(rt700->regmap, 0xff01, 0x0000); + regmap_write(rt700->regmap, 0x7520, 0x001a); + regmap_write(rt700->regmap, 0x7420, 0xc003); + + /* power on */ + regmap_write(rt700->regmap, RT700_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + /* Set Pin Widget */ + regmap_write(rt700->regmap, RT700_SET_PIN_HP, 0x40); + regmap_write(rt700->regmap, RT700_SET_PIN_SPK, 0x40); + regmap_write(rt700->regmap, RT700_SET_EAPD_SPK, RT700_EAPD_HIGH); + regmap_write(rt700->regmap, RT700_SET_PIN_DMIC1, 0x20); + regmap_write(rt700->regmap, RT700_SET_PIN_DMIC2, 0x20); + regmap_write(rt700->regmap, RT700_SET_PIN_MIC2, 0x20); + + /* Set Configuration Default */ + regmap_write(rt700->regmap, 0x4f12, 0x91); + regmap_write(rt700->regmap, 0x4e12, 0xd6); + regmap_write(rt700->regmap, 0x4d12, 0x11); + regmap_write(rt700->regmap, 0x4c12, 0x20); + regmap_write(rt700->regmap, 0x4f13, 0x91); + regmap_write(rt700->regmap, 0x4e13, 0xd6); + regmap_write(rt700->regmap, 0x4d13, 0x11); + regmap_write(rt700->regmap, 0x4c13, 0x21); + + regmap_write(rt700->regmap, 0x4f19, 0x02); + regmap_write(rt700->regmap, 0x4e19, 0xa1); + regmap_write(rt700->regmap, 0x4d19, 0x90); + regmap_write(rt700->regmap, 0x4c19, 0x80); + + /* Enable Line2 */ + regmap_write(rt700->regmap, 0x371b, 0x40); + regmap_write(rt700->regmap, 0x731b, 0xb0); + regmap_write(rt700->regmap, 0x839b, 0x00); + + /* Set index */ + rt700_index_write(rt700->regmap, 0x4a, 0x201b); + rt700_index_write(rt700->regmap, 0x45, 0x5089); + rt700_index_write(rt700->regmap, 0x6b, 0x5064); + rt700_index_write(rt700->regmap, 0x48, 0xd249); + + /* Finish Initial Settings, set power to D3 */ + regmap_write(rt700->regmap, RT700_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + + if (!rt700->first_hw_init) { + INIT_DELAYED_WORK(&rt700->jack_detect_work, + rt700_jack_detect_handler); + INIT_DELAYED_WORK(&rt700->jack_btn_check_work, + rt700_btn_check_handler); + } + + /* + * if set_jack callback occurred early than io_init, + * we set up the jack detection function now + */ + if (rt700->hs_jack) + rt700_jack_init(rt700); + + if (rt700->first_hw_init) { + regcache_cache_bypass(rt700->regmap, false); + regcache_mark_dirty(rt700->regmap); + } else + rt700->first_hw_init = true; + + /* Mark Slave initialization complete */ + rt700->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + + return 0; +} + +MODULE_DESCRIPTION("ASoC RT700 driver SDW"); +MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt700.h b/sound/soc/codecs/rt700.h new file mode 100644 index 000000000000..794ee2e29051 --- /dev/null +++ b/sound/soc/codecs/rt700.h @@ -0,0 +1,174 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * rt700.h -- RT700 ALSA SoC audio driver header + * + * Copyright(c) 2019 Realtek Semiconductor Corp. + */ + +#ifndef __RT700_H__ +#define __RT700_H__ + +extern const struct dev_pm_ops rt700_runtime_pm; + +struct rt700_priv { + struct snd_soc_component *component; + struct regmap *regmap; + struct regmap *sdw_regmap; + struct sdw_slave *slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; + struct snd_soc_jack *hs_jack; + struct delayed_work jack_detect_work; + struct delayed_work jack_btn_check_work; + int jack_type; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +/* NID */ +#define RT700_AUDIO_FUNCTION_GROUP 0x01 +#define RT700_DAC_OUT1 0x02 +#define RT700_DAC_OUT2 0x03 +#define RT700_ADC_IN1 0x09 +#define RT700_ADC_IN2 0x08 +#define RT700_DMIC1 0x12 +#define RT700_DMIC2 0x13 +#define RT700_SPK_OUT 0x14 +#define RT700_MIC2 0x19 +#define RT700_LINE1 0x1a +#define RT700_LINE2 0x1b +#define RT700_BEEP 0x1d +#define RT700_SPDIF 0x1e +#define RT700_VENDOR_REGISTERS 0x20 +#define RT700_HP_OUT 0x21 +#define RT700_MIXER_IN1 0x22 +#define RT700_MIXER_IN2 0x23 +#define RT700_INLINE_CMD 0x55 + +/* Index (NID:20h) */ +#define RT700_DAC_DC_CALI_CTL1 0x00 +#define RT700_PARA_VERB_CTL 0x1a +#define RT700_COMBO_JACK_AUTO_CTL1 0x45 +#define RT700_COMBO_JACK_AUTO_CTL2 0x46 +#define RT700_INLINE_CMD_CTL 0x48 +#define RT700_DIGITAL_MISC_CTRL4 0x4a +#define RT700_VREFOUT_CTL 0x6b +#define RT700_FSM_CTL 0x6f +#define RT700_IRQ_FLAG_TABLE1 0x80 +#define RT700_IRQ_FLAG_TABLE2 0x81 +#define RT700_IRQ_FLAG_TABLE3 0x82 + +/* Verb */ +#define RT700_VERB_SET_CONNECT_SEL 0x3100 +#define RT700_VERB_SET_EAPD_BTLENABLE 0x3c00 +#define RT700_VERB_GET_CONNECT_SEL 0xb100 +#define RT700_VERB_SET_POWER_STATE 0x3500 +#define RT700_VERB_SET_CHANNEL_STREAMID 0x3600 +#define RT700_VERB_SET_PIN_WIDGET_CONTROL 0x3700 +#define RT700_VERB_SET_UNSOLICITED_ENABLE 0x3800 +#define RT700_SET_AMP_GAIN_MUTE_H 0x7300 +#define RT700_SET_AMP_GAIN_MUTE_L 0x8380 +#define RT700_VERB_GET_PIN_SENSE 0xb900 + +#define RT700_READ_HDA_3 0x2012 +#define RT700_READ_HDA_2 0x2013 +#define RT700_READ_HDA_1 0x2014 +#define RT700_READ_HDA_0 0x2015 +#define RT700_PRIV_INDEX_W_H 0x7520 +#define RT700_PRIV_INDEX_W_L 0x85a0 +#define RT700_PRIV_DATA_W_H 0x7420 +#define RT700_PRIV_DATA_W_L 0x84a0 +#define RT700_PRIV_INDEX_R_H 0x9d20 +#define RT700_PRIV_INDEX_R_L 0xada0 +#define RT700_PRIV_DATA_R_H 0x9c20 +#define RT700_PRIV_DATA_R_L 0xaca0 +#define RT700_DAC_FORMAT_H 0x7203 +#define RT700_DAC_FORMAT_L 0x8283 +#define RT700_ADC_FORMAT_H 0x7209 +#define RT700_ADC_FORMAT_L 0x8289 +#define RT700_SET_AUDIO_POWER_STATE\ + (RT700_VERB_SET_POWER_STATE | RT700_AUDIO_FUNCTION_GROUP) +#define RT700_SET_PIN_DMIC1\ + (RT700_VERB_SET_PIN_WIDGET_CONTROL | RT700_DMIC1) +#define RT700_SET_PIN_DMIC2\ + (RT700_VERB_SET_PIN_WIDGET_CONTROL | RT700_DMIC2) +#define RT700_SET_PIN_SPK\ + (RT700_VERB_SET_PIN_WIDGET_CONTROL | RT700_SPK_OUT) +#define RT700_SET_PIN_HP\ + (RT700_VERB_SET_PIN_WIDGET_CONTROL | RT700_HP_OUT) +#define RT700_SET_PIN_MIC2\ + (RT700_VERB_SET_PIN_WIDGET_CONTROL | RT700_MIC2) +#define RT700_SET_PIN_LINE1\ + (RT700_VERB_SET_PIN_WIDGET_CONTROL | RT700_LINE1) +#define RT700_SET_PIN_LINE2\ + (RT700_VERB_SET_PIN_WIDGET_CONTROL | RT700_LINE2) +#define RT700_SET_MIC2_UNSOLICITED_ENABLE\ + (RT700_VERB_SET_UNSOLICITED_ENABLE | RT700_MIC2) +#define RT700_SET_HP_UNSOLICITED_ENABLE\ + (RT700_VERB_SET_UNSOLICITED_ENABLE | RT700_HP_OUT) +#define RT700_SET_INLINE_UNSOLICITED_ENABLE\ + (RT700_VERB_SET_UNSOLICITED_ENABLE | RT700_INLINE_CMD) +#define RT700_SET_STREAMID_DAC1\ + (RT700_VERB_SET_CHANNEL_STREAMID | RT700_DAC_OUT1) +#define RT700_SET_STREAMID_DAC2\ + (RT700_VERB_SET_CHANNEL_STREAMID | RT700_DAC_OUT2) +#define RT700_SET_STREAMID_ADC1\ + (RT700_VERB_SET_CHANNEL_STREAMID | RT700_ADC_IN1) +#define RT700_SET_STREAMID_ADC2\ + (RT700_VERB_SET_CHANNEL_STREAMID | RT700_ADC_IN2) +#define RT700_SET_GAIN_DAC1_L\ + (RT700_SET_AMP_GAIN_MUTE_L | RT700_DAC_OUT1) +#define RT700_SET_GAIN_DAC1_H\ + (RT700_SET_AMP_GAIN_MUTE_H | RT700_DAC_OUT1) +#define RT700_SET_GAIN_ADC1_L\ + (RT700_SET_AMP_GAIN_MUTE_L | RT700_ADC_IN1) +#define RT700_SET_GAIN_ADC1_H\ + (RT700_SET_AMP_GAIN_MUTE_H | RT700_ADC_IN1) +#define RT700_SET_GAIN_ADC2_L\ + (RT700_SET_AMP_GAIN_MUTE_L | RT700_ADC_IN2) +#define RT700_SET_GAIN_ADC2_H\ + (RT700_SET_AMP_GAIN_MUTE_H | RT700_ADC_IN2) +#define RT700_SET_GAIN_AMIC_L\ + (RT700_SET_AMP_GAIN_MUTE_L | RT700_MIC2) +#define RT700_SET_GAIN_AMIC_H\ + (RT700_SET_AMP_GAIN_MUTE_H | RT700_MIC2) +#define RT700_SET_GAIN_HP_L\ + (RT700_SET_AMP_GAIN_MUTE_L | RT700_HP_OUT) +#define RT700_SET_GAIN_HP_H\ + (RT700_SET_AMP_GAIN_MUTE_H | RT700_HP_OUT) +#define RT700_SET_GAIN_SPK_L\ + (RT700_SET_AMP_GAIN_MUTE_L | RT700_SPK_OUT) +#define RT700_SET_GAIN_SPK_H\ + (RT700_SET_AMP_GAIN_MUTE_H | RT700_SPK_OUT) +#define RT700_SET_EAPD_SPK\ + (RT700_VERB_SET_EAPD_BTLENABLE | RT700_SPK_OUT) + +/* combo jack auto switch control 2 (0x46)(NID:20h) */ +#define RT700_COMBOJACK_AUTO_DET_STATUS (0x1 << 11) +#define RT700_COMBOJACK_AUTO_DET_TRS (0x1 << 10) +#define RT700_COMBOJACK_AUTO_DET_CTIA (0x1 << 9) +#define RT700_COMBOJACK_AUTO_DET_OMTP (0x1 << 8) + +#define RT700_EAPD_HIGH 0x2 +#define RT700_EAPD_LOW 0x0 +#define RT700_MUTE_SFT 7 +#define RT700_DIR_IN_SFT 6 +#define RT700_DIR_OUT_SFT 7 + +enum { + RT700_AIF1, + RT700_AIF2, + RT700_AIFS, +}; + +int rt700_io_init(struct device *dev, struct sdw_slave *slave); +int rt700_init(struct device *dev, struct regmap *sdw_regmap, + struct regmap *regmap, struct sdw_slave *slave); + +int rt700_jack_detect(struct rt700_priv *rt700, bool *hp, bool *mic); +int rt700_clock_config(struct device *dev); +#endif /* __RT700_H__ */ -- 2.24.1

3 2

[alsa-devel] [PATCH 1/2] ASoC: max98090: fix incorrect helper in max98090_dapm_put_enum_double()
by Marek Szyprowski 10 Jan '20

10 Jan '20

Commit 62d5ae4cafb7 ("ASoC: max98090: save and restore SHDN when changing sensitive registers") extended the code for handling "LTENL Mux", "LTENR Mux", "LBENL Mux" and "LBENR Mux" controls by adding a custom max98090_dapm_put_enum_double() function to them. However that function used incorrect helper to get its component object. Fix this by using the proper snd_soc_dapm_* helper. This fixes the following NULL pointer exception observed on Exynos4412-based Odroid U3 board: 8<--- cut here --- Unable to handle kernel NULL pointer dereference at virtual address 000000b0 pgd = (ptrval) [000000b0] *pgd=00000000 Internal error: Oops: 5 [#1] PREEMPT SMP ARM Modules linked in: CPU: 0 PID: 1104 Comm: alsactl Not tainted 5.5.0-rc5-next-20200107 #166 Hardware name: SAMSUNG EXYNOS (Flattened Device Tree) PC is at __mutex_lock+0x54/0xb18 LR is at ___might_sleep+0x3c/0x2e0 ... Process alsactl (pid: 1104, stack limit = 0x(ptrval)) ... [<c0b49630>] (__mutex_lock) from [<c0b4a110>] (mutex_lock_nested+0x1c/0x24) [<c0b4a110>] (mutex_lock_nested) from [<c0839b3c>] (max98090_shdn_save+0x1c/0x28) [<c0839b3c>] (max98090_shdn_save) from [<c083a4f8>] (max98090_dapm_put_enum_double+0x20/0x40) [<c083a4f8>] (max98090_dapm_put_enum_double) from [<c080d0e8>] (snd_ctl_ioctl+0x190/0xbb8) [<c080d0e8>] (snd_ctl_ioctl) from [<c02cafec>] (ksys_ioctl+0x470/0xaf8) [<c02cafec>] (ksys_ioctl) from [<c0101000>] (ret_fast_syscall+0x0/0x28) ... ---[ end trace 0e93f0580f4b9241 ]--- Fixes: 62d5ae4cafb7 ("ASoC: max98090: save and restore SHDN when changing sensitive registers") Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 4c7b16d557e2..c01ce4a3f86d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -82,7 +82,7 @@ static int max98090_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = - snd_soc_kcontrol_component(kcontrol); + snd_soc_dapm_kcontrol_component(kcontrol); struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); int ret; -- 2.17.1

3 10

[alsa-devel] [PATCH v6 1/2] PCI: Add a helper to check Power Resource Requirements _PR3 existence
by Kai-Heng Feng 10 Jan '20

10 Jan '20

A driver may want to know the existence of _PR3, to choose different runtime suspend behavior. A user will be add in next patch. This is mostly the same as nouveau_pr3_present(). Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v6: - Only define the function when CONFIG_ACPI is set. v5: - Add wording suggestion from Bjorn. v4: - Let caller to find its upstream port device. drivers/pci/pci.c | 18 ++++++++++++++++++ include/linux/pci.h | 2 ++ 2 files changed, 20 insertions(+) diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c index e7982af9a5d8..1df99d9e350e 100644 --- a/drivers/pci/pci.c +++ b/drivers/pci/pci.c @@ -5856,6 +5856,24 @@ int pci_set_vga_state(struct pci_dev *dev, bool decode, return 0; } +#ifdef CONFIG_ACPI +bool pci_pr3_present(struct pci_dev *pdev) +{ + struct acpi_device *adev; + + if (acpi_disabled) + return false; + + adev = ACPI_COMPANION(&pdev->dev); + if (!adev) + return false; + + return adev->power.flags.power_resources && + acpi_has_method(adev->handle, "_PR3"); +} +EXPORT_SYMBOL_GPL(pci_pr3_present); +#endif + /** * pci_add_dma_alias - Add a DMA devfn alias for a device * @dev: the PCI device for which alias is added diff --git a/include/linux/pci.h b/include/linux/pci.h index f9088c89a534..1d15c5d49cdd 100644 --- a/include/linux/pci.h +++ b/include/linux/pci.h @@ -2310,9 +2310,11 @@ struct irq_domain *pci_host_bridge_acpi_msi_domain(struct pci_bus *bus); void pci_msi_register_fwnode_provider(struct fwnode_handle *(*fn)(struct device *)); +bool pci_pr3_present(struct pci_dev *pdev); #else static inline struct irq_domain * pci_host_bridge_acpi_msi_domain(struct pci_bus *bus) { return NULL; } +static bool pci_pr3_present(struct pci_dev *pdev) { return false; } #endif #ifdef CONFIG_EEH -- 2.17.1

3 8

[alsa-devel] Bug report for changes in ucm2 with chtrt5645 card on Lenovo Miix 320
by Tim Schumacher 10 Jan '20

10 Jan '20

Hi alsa devs, I'm on Arch Linux on a rather uncommon laptop (Lenovo Miix 320) and my sound stopped working on the upgrade from alsa-lib 1.1.9 to 1.2.1. wabbits on IRC helped me track down the problem to a change in the /usr/share/alsa/ucm (now /usr/share/alsa/ucm2) files. It seems like /usr/share/alsa/ucm2/chtrt5645/HiFi-dmic2.conf and a symlink at /usr/share/alsa/ucm2/chtrt5645/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf were specifically added for this laptop but in the contrary they break the sound for me. After the upgrade to alsa-lib 1.2.1 the sound is not working. The commands play no sound: pasuspender -- speaker-test --nloops=1 --channels=2 --test=wav --device=hw:0,0 pasuspender -- speaker-test --nloops=1 --channels=2 --test=wav --device=hw:0,1 And this is the alsa-info.sh output http://alsa-project.org/db/?f=f883910a5c5101b4b1ea4202d1fe84ccd139f796 After deleting the /usr/share/alsa/ucm2/chtrt5645/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf symlink both commands from above play sound and this is the alsa-info.sh output http://alsa-project.org/db/?f=e759eb9118a191b6c3b8c021fed58abc9cf95076 Thanks. - Tim

1 0

[alsa-devel] [PATCH 2/2] ASoC: rt1308-sdw: add rt1308 SdW amplifier driver
by shumingf＠realtek.com 10 Jan '20

10 Jan '20

From: Shuming Fan <shumingf(a)realtek.com> This is the initial amplifier driver for rt1308-sdw. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1308-sdw.c | 736 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1308-sdw.h | 169 ++++++++ 4 files changed, 913 insertions(+) create mode 100644 sound/soc/codecs/rt1308-sdw.c create mode 100644 sound/soc/codecs/rt1308-sdw.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7761aeec4d07..1de31f88a324 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -167,6 +167,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5682 if I2C select SND_SOC_RT700_SDW if SOUNDWIRE select SND_SOC_RT711_SDW if SOUNDWIRE + select SND_SOC_RT1308_SDW if SOUNDWIRE select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIMPLE_AMPLIFIER @@ -1004,6 +1005,11 @@ config SND_SOC_RT1305 config SND_SOC_RT1308 tristate +config SND_SOC_RT1308_SDW + tristate "Realtek RT1308 Codec - SDW" + depends on SOUNDWIRE + select REGMAP_SOUNDWIRE + config SND_SOC_RT5514 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 288b7af354a0..d49138640e0e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -154,6 +154,7 @@ snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o snd-soc-rt1305-objs := rt1305.o snd-soc-rt1308-objs := rt1308.o +snd-soc-rt1308-sdw-objs := rt1308-sdw.o snd-soc-rt274-objs := rt274.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o @@ -447,6 +448,7 @@ obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o obj-$(CONFIG_SND_SOC_RT1308) += snd-soc-rt1308.o +obj-$(CONFIG_SND_SOC_RT1308_SDW) += snd-soc-rt1308-sdw.o obj-$(CONFIG_SND_SOC_RT274) += snd-soc-rt274.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c new file mode 100644 index 000000000000..8a03dbfe7906 --- /dev/null +++ b/sound/soc/codecs/rt1308-sdw.c @@ -0,0 +1,736 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rt1308-sdw.c -- rt1308 ALSA SoC audio driver +// +// Copyright(c) 2019 Realtek Semiconductor Corp. +// +// +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/pm_runtime.h> +#include <linux/mod_devicetable.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "rt1308.h" +#include "rt1308-sdw.h" + +static bool rt1308_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x00e0: + case 0x00f0: + case 0x2f01 ... 0x2f07: + case 0x3000 ... 0x3001: + case 0x3004 ... 0x3005: + case 0x3008: + case 0x300a: + case 0xc000 ... 0xcff3: + return true; + default: + return false; + } +} + +static bool rt1308_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2f01 ... 0x2f07: + case 0x3000 ... 0x3001: + case 0x3004 ... 0x3005: + case 0x3008: + case 0x300a: + case 0xc000: + return true; + default: + return false; + } +} + +static const struct regmap_config rt1308_sdw_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt1308_readable_register, + .volatile_reg = rt1308_volatile_register, + .max_register = 0xcfff, + .reg_defaults = rt1308_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rt1308_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +/* Bus clock frequency */ +#define RT1308_CLK_FREQ_9600000HZ 9600000 +#define RT1308_CLK_FREQ_12000000HZ 12000000 +#define RT1308_CLK_FREQ_6000000HZ 6000000 +#define RT1308_CLK_FREQ_4800000HZ 4800000 +#define RT1308_CLK_FREQ_2400000HZ 2400000 +#define RT1308_CLK_FREQ_12288000HZ 12288000 + +static int rt1308_clock_config(struct device *dev) +{ + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev); + unsigned int clk_freq, value; + + clk_freq = (rt1308->params.curr_dr_freq >> 1); + + switch (clk_freq) { + case RT1308_CLK_FREQ_12000000HZ: + value = 0x0; + break; + case RT1308_CLK_FREQ_6000000HZ: + value = 0x1; + break; + case RT1308_CLK_FREQ_9600000HZ: + value = 0x2; + break; + case RT1308_CLK_FREQ_4800000HZ: + value = 0x3; + break; + case RT1308_CLK_FREQ_2400000HZ: + value = 0x4; + break; + case RT1308_CLK_FREQ_12288000HZ: + value = 0x5; + break; + default: + return -EINVAL; + } + + regmap_write(rt1308->regmap, 0xe0, value); + regmap_write(rt1308->regmap, 0xf0, value); + + dev_dbg(dev, "%s complete, clk_freq=%d\n", __func__, clk_freq); + + return 0; +} + +static int rt1308_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval, i, num_of_ports = 1; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x00; /* BITMAP: 00010100 (not enable yet) */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + /* for sink */ + nval = hweight32(prop->sink_ports); + num_of_ports += nval; + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), + GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* Allocate port_ready based on num_of_ports */ + slave->port_ready = devm_kcalloc(&slave->dev, num_of_ports, + sizeof(*slave->port_ready), + GFP_KERNEL); + if (!slave->port_ready) + return -ENOMEM; + + /* Initialize completion */ + for (i = 0; i < num_of_ports; i++) + init_completion(&slave->port_ready[i]); + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + dev_dbg(&slave->dev, "%s\n", __func__); + + return 0; +} + +static int rt1308_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev); + int ret = 0; + unsigned int efuse_m_btl_l, efuse_m_btl_r, tmp; + unsigned int efuse_c_btl_l, efuse_c_btl_r; + + if (rt1308->hw_init) + return 0; + + ret = rt1308_read_prop(slave); + if (ret < 0) + goto _io_init_err_; + + if (rt1308->first_hw_init) { + regcache_cache_only(rt1308->regmap, false); + regcache_cache_bypass(rt1308->regmap, true); + } + + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + if (!rt1308->first_hw_init) { + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + /* sw reset */ + regmap_write(rt1308->regmap, RT1308_SDW_RESET, 0); + + /* read efuse */ + regmap_write(rt1308->regmap, 0xc360, 0x01); + regmap_write(rt1308->regmap, 0xc361, 0x80); + regmap_write(rt1308->regmap, 0xc7f0, 0x04); + regmap_write(rt1308->regmap, 0xc7f1, 0xfe); + msleep(100); + regmap_write(rt1308->regmap, 0xc7f0, 0x44); + msleep(20); + regmap_write(rt1308->regmap, 0xc240, 0x10); + + regmap_read(rt1308->regmap, 0xc861, &tmp); + efuse_m_btl_l = tmp; + regmap_read(rt1308->regmap, 0xc860, &tmp); + efuse_m_btl_l = efuse_m_btl_l | (tmp << 8); + regmap_read(rt1308->regmap, 0xc863, &tmp); + efuse_c_btl_l = tmp; + regmap_read(rt1308->regmap, 0xc862, &tmp); + efuse_c_btl_l = efuse_c_btl_l | (tmp << 8); + regmap_read(rt1308->regmap, 0xc871, &tmp); + efuse_m_btl_r = tmp; + regmap_read(rt1308->regmap, 0xc870, &tmp); + efuse_m_btl_r = efuse_m_btl_r | (tmp << 8); + regmap_read(rt1308->regmap, 0xc873, &tmp); + efuse_c_btl_r = tmp; + regmap_read(rt1308->regmap, 0xc872, &tmp); + efuse_c_btl_r = efuse_c_btl_r | (tmp << 8); + dev_info(&slave->dev, "%s m_btl_l=0x%x, m_btl_r=0x%x\n", __func__, + efuse_m_btl_l, efuse_m_btl_r); + dev_info(&slave->dev, "%s c_btl_l=0x%x, c_btl_r=0x%x\n", __func__, + efuse_c_btl_l, efuse_c_btl_r); + + /* initial settings */ + regmap_write(rt1308->regmap, 0xc103, 0xc0); + regmap_write(rt1308->regmap, 0xc030, 0x17); + regmap_write(rt1308->regmap, 0xc031, 0x81); + regmap_write(rt1308->regmap, 0xc032, 0x26); + regmap_write(rt1308->regmap, 0xc040, 0x80); + regmap_write(rt1308->regmap, 0xc041, 0x80); + regmap_write(rt1308->regmap, 0xc042, 0x06); + regmap_write(rt1308->regmap, 0xc052, 0x0a); + regmap_write(rt1308->regmap, 0xc080, 0x0a); + regmap_write(rt1308->regmap, 0xc060, 0x02); + regmap_write(rt1308->regmap, 0xc061, 0x75); + regmap_write(rt1308->regmap, 0xc062, 0x05); + regmap_write(rt1308->regmap, 0xc171, 0x07); + regmap_write(rt1308->regmap, 0xc173, 0x0d); + regmap_write(rt1308->regmap, 0xc311, 0x7f); + regmap_write(rt1308->regmap, 0xc900, 0x90); + regmap_write(rt1308->regmap, 0xc1a0, 0x84); + regmap_write(rt1308->regmap, 0xc1a1, 0x01); + regmap_write(rt1308->regmap, 0xc360, 0x78); + regmap_write(rt1308->regmap, 0xc361, 0x87); + regmap_write(rt1308->regmap, 0xc0a1, 0x71); + regmap_write(rt1308->regmap, 0xc210, 0x00); + regmap_write(rt1308->regmap, 0xc070, 0x00); + regmap_write(rt1308->regmap, 0xc100, 0xd7); + regmap_write(rt1308->regmap, 0xc101, 0xd7); + regmap_write(rt1308->regmap, 0xc300, 0x09); + + if (rt1308->first_hw_init) { + regcache_cache_bypass(rt1308->regmap, false); + regcache_mark_dirty(rt1308->regmap); + } else + rt1308->first_hw_init = true; + + /* Mark Slave initialization complete */ + rt1308->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + +_io_init_err_: + return ret; +} + +static int rt1308_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(&slave->dev); + + /* Update the status */ + rt1308->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + rt1308->hw_init = false; + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt1308->hw_init || rt1308->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt1308_io_init(&slave->dev, slave); +} + +static int rt1308_bus_config(struct sdw_slave *slave, + struct sdw_bus_params *params) +{ + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(&slave->dev); + int ret; + + memcpy(&rt1308->params, params, sizeof(*params)); + + ret = rt1308_clock_config(&slave->dev); + if (ret < 0) + dev_err(&slave->dev, "Invalid clk config"); + + return ret; +} + +static int rt1308_interrupt_callback(struct sdw_slave *slave, + struct sdw_slave_intr_status *status) +{ + dev_dbg(&slave->dev, + "%s control_port_stat=%x", __func__, status->control_port); + + return 0; +} + +static int rt1308_classd_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); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(30); + snd_soc_component_update_bits(component, + RT1308_SDW_OFFSET | (RT1308_POWER_STATUS << 4), + 0x3, 0x3); + msleep(40); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_component_update_bits(component, + RT1308_SDW_OFFSET | (RT1308_POWER_STATUS << 4), + 0x3, 0); + usleep_range(150000, 200000); + break; + + default: + break; + } + + return 0; +} + +static const char * const rt1308_rx_data_ch_select[] = { + "LR", + "LL", + "RL", + "RR", +}; + +static SOC_ENUM_SINGLE_DECL(rt1308_rx_data_ch_enum, + RT1308_SDW_OFFSET | (RT1308_DATA_PATH << 4), 0, + rt1308_rx_data_ch_select); + +static const struct snd_kcontrol_new rt1308_snd_controls[] = { + + /* I2S Data Channel Selection */ + SOC_ENUM("RX Channel Select", rt1308_rx_data_ch_enum), +}; + +static const struct snd_kcontrol_new rt1308_sto_dac_l = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", + RT1308_SDW_OFFSET_BYTE3 | (RT1308_DAC_SET << 4), + RT1308_DVOL_MUTE_L_EN_SFT, 1, 1); + +static const struct snd_kcontrol_new rt1308_sto_dac_r = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", + RT1308_SDW_OFFSET_BYTE3 | (RT1308_DAC_SET << 4), + RT1308_DVOL_MUTE_R_EN_SFT, 1, 1); + +static const struct snd_soc_dapm_widget rt1308_dapm_widgets[] = { + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "DP1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Supply Widgets */ + SND_SOC_DAPM_SUPPLY("MBIAS20U", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ALDO", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DBG", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DACL", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLK25M", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_R", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_L", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Power", + RT1308_SDW_OFFSET | (RT1308_POWER << 4), 3, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DLDO", + RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("VREF", + RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIXER_R", + RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIXER_L", + RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MBIAS4U", + RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("PLL2_LDO", + RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2B", + RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2F", + RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2F2", + RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2B2", + RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 0, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SWITCH("DAC L", SND_SOC_NOPM, 0, 0, &rt1308_sto_dac_l), + SND_SOC_DAPM_SWITCH("DAC R", SND_SOC_NOPM, 0, 0, &rt1308_sto_dac_r), + + /* Output Lines */ + SND_SOC_DAPM_PGA_E("CLASS D", SND_SOC_NOPM, 0, 0, NULL, 0, + rt1308_classd_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), +}; + +static const struct snd_soc_dapm_route rt1308_dapm_routes[] = { + + { "DAC", NULL, "AIF1RX" }, + + { "DAC", NULL, "MBIAS20U" }, + { "DAC", NULL, "ALDO" }, + { "DAC", NULL, "DBG" }, + { "DAC", NULL, "DACL" }, + { "DAC", NULL, "CLK25M" }, + { "DAC", NULL, "ADC_R" }, + { "DAC", NULL, "ADC_L" }, + { "DAC", NULL, "DLDO" }, + { "DAC", NULL, "VREF" }, + { "DAC", NULL, "MIXER_R" }, + { "DAC", NULL, "MIXER_L" }, + { "DAC", NULL, "MBIAS4U" }, + { "DAC", NULL, "PLL2_LDO" }, + { "DAC", NULL, "PLL2B" }, + { "DAC", NULL, "PLL2F" }, + { "DAC", NULL, "PLL2F2" }, + { "DAC", NULL, "PLL2B2" }, + + { "DAC L", "Switch", "DAC" }, + { "DAC R", "Switch", "DAC" }, + { "DAC L", NULL, "DAC Power" }, + { "DAC R", NULL, "DAC Power" }, + + { "CLASS D", NULL, "DAC L" }, + { "CLASS D", NULL, "DAC R" }, + { "SPOL", NULL, "CLASS D" }, + { "SPOR", NULL, "CLASS D" }, +}; + +static int rt1308_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + struct sdw_stream_data *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = (struct sdw_stream_runtime *)sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void rt1308_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1308_sdw_priv *rt1308 = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int retval, port, num_channels; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!rt1308->sdw_slave) + return -EINVAL; + + /* SoundWire specific configuration */ + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port = 1; + } else { + return -EINVAL; + } + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = (1 << (num_channels)) - 1; + port_config.num = port; + + retval = sdw_stream_add_slave(rt1308->sdw_slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + return retval; +} + +static int rt1308_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1308_sdw_priv *rt1308 = + snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt1308->sdw_slave) + return -EINVAL; + + sdw_stream_remove_slave(rt1308->sdw_slave, stream->sdw_stream); + return 0; +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static struct sdw_slave_ops rt1308_slave_ops = { + .read_prop = rt1308_read_prop, + .interrupt_callback = rt1308_interrupt_callback, + .update_status = rt1308_update_status, + .bus_config = rt1308_bus_config, +}; + +static const struct snd_soc_component_driver soc_component_sdw_rt1308 = { + .controls = rt1308_snd_controls, + .num_controls = ARRAY_SIZE(rt1308_snd_controls), + .dapm_widgets = rt1308_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt1308_dapm_widgets), + .dapm_routes = rt1308_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt1308_dapm_routes), +}; + +static const struct snd_soc_dai_ops rt1308_aif_dai_ops = { + .hw_params = rt1308_sdw_hw_params, + .hw_free = rt1308_sdw_pcm_hw_free, + .set_sdw_stream = rt1308_set_sdw_stream, + .shutdown = rt1308_sdw_shutdown, +}; + +#define RT1308_STEREO_RATES SNDRV_PCM_RATE_48000 +#define RT1308_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver rt1308_sdw_dai[] = { + { + .name = "rt1308-aif", + .playback = { + .stream_name = "DP1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT1308_STEREO_RATES, + .formats = RT1308_FORMATS, + }, + .ops = &rt1308_aif_dai_ops, + }, +}; + +static int rt1308_sdw_init(struct device *dev, struct regmap *regmap, + struct sdw_slave *slave) +{ + struct rt1308_sdw_priv *rt1308; + int ret; + + rt1308 = devm_kzalloc(dev, sizeof(*rt1308), GFP_KERNEL); + if (!rt1308) + return -ENOMEM; + + dev_set_drvdata(dev, rt1308); + rt1308->sdw_slave = slave; + rt1308->regmap = regmap; + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt1308->hw_init = false; + rt1308->first_hw_init = false; + + ret = devm_snd_soc_register_component(dev, + &soc_component_sdw_rt1308, + rt1308_sdw_dai, + ARRAY_SIZE(rt1308_sdw_dai)); + + dev_dbg(&slave->dev, "%s\n", __func__); + + return ret; +} + +static int rt1308_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap; + + /* Assign ops */ + slave->ops = &rt1308_slave_ops; + + /* Regmap Initialization */ + regmap = devm_regmap_init_sdw(slave, &rt1308_sdw_regmap); + if (!regmap) + return -EINVAL; + + rt1308_sdw_init(&slave->dev, regmap, slave); + + return 0; +} + +static const struct sdw_device_id rt1308_id[] = { + SDW_SLAVE_ENTRY(0x025d, 0x1308, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt1308_id); + +static int rt1308_dev_suspend(struct device *dev) +{ + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev); + + if (!rt1308->hw_init) + return 0; + + regcache_cache_only(rt1308->regmap, true); + + return 0; +} + +#define RT1308_PROBE_TIMEOUT 2000 + +static int rt1308_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt1308->hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT1308_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt1308->regmap, false); + regcache_sync_region(rt1308->regmap, 0xc000, 0xcfff); + + return 0; +} + +static const struct dev_pm_ops rt1308_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt1308_dev_suspend, rt1308_dev_resume) + SET_RUNTIME_PM_OPS(rt1308_dev_suspend, rt1308_dev_resume, NULL) +}; + +static struct sdw_driver rt1308_sdw_driver = { + .driver = { + .name = "rt1308", + .owner = THIS_MODULE, + .pm = &rt1308_pm, + }, + .probe = rt1308_sdw_probe, + .ops = &rt1308_slave_ops, + .id_table = rt1308_id, +}; +module_sdw_driver(rt1308_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT1308 driver SDW"); +MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h new file mode 100644 index 000000000000..c9341e70d6cf --- /dev/null +++ b/sound/soc/codecs/rt1308-sdw.h @@ -0,0 +1,169 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * rt1308-sdw.h -- RT1308 ALSA SoC audio driver header + * + * Copyright(c) 2019 Realtek Semiconductor Corp. + */ + +#ifndef __RT1308_SDW_H__ +#define __RT1308_SDW_H__ + +static const struct reg_default rt1308_reg_defaults[] = { + { 0x0000, 0x00 }, + { 0x0001, 0x00 }, + { 0x0002, 0x00 }, + { 0x0003, 0x00 }, + { 0x0004, 0x00 }, + { 0x0005, 0x01 }, + { 0x0020, 0x00 }, + { 0x0022, 0x00 }, + { 0x0023, 0x00 }, + { 0x0024, 0x00 }, + { 0x0025, 0x00 }, + { 0x0026, 0x00 }, + { 0x0030, 0x00 }, + { 0x0032, 0x00 }, + { 0x0033, 0x00 }, + { 0x0034, 0x00 }, + { 0x0035, 0x00 }, + { 0x0036, 0x00 }, + { 0x0040, 0x00 }, + { 0x0041, 0x00 }, + { 0x0042, 0x00 }, + { 0x0043, 0x00 }, + { 0x0044, 0x20 }, + { 0x0045, 0x01 }, + { 0x0046, 0x01 }, + { 0x0048, 0x00 }, + { 0x0049, 0x00 }, + { 0x0050, 0x20 }, + { 0x0051, 0x02 }, + { 0x0052, 0x5D }, + { 0x0053, 0x13 }, + { 0x0054, 0x08 }, + { 0x0055, 0x00 }, + { 0x0060, 0x00 }, + { 0x0070, 0x00 }, + { 0x00E0, 0x00 }, + { 0x00F0, 0x00 }, + { 0x0100, 0x00 }, + { 0x0101, 0x00 }, + { 0x0102, 0x20 }, + { 0x0103, 0x00 }, + { 0x0104, 0x00 }, + { 0x0105, 0x03 }, + { 0x0120, 0x00 }, + { 0x0122, 0x00 }, + { 0x0123, 0x00 }, + { 0x0124, 0x00 }, + { 0x0125, 0x00 }, + { 0x0126, 0x00 }, + { 0x0127, 0x00 }, + { 0x0130, 0x00 }, + { 0x0132, 0x00 }, + { 0x0133, 0x00 }, + { 0x0134, 0x00 }, + { 0x0135, 0x00 }, + { 0x0136, 0x00 }, + { 0x0137, 0x00 }, + { 0x0200, 0x00 }, + { 0x0201, 0x00 }, + { 0x0202, 0x00 }, + { 0x0203, 0x00 }, + { 0x0204, 0x00 }, + { 0x0205, 0x03 }, + { 0x0220, 0x00 }, + { 0x0222, 0x00 }, + { 0x0223, 0x00 }, + { 0x0224, 0x00 }, + { 0x0225, 0x00 }, + { 0x0226, 0x00 }, + { 0x0227, 0x00 }, + { 0x0230, 0x00 }, + { 0x0232, 0x00 }, + { 0x0233, 0x00 }, + { 0x0234, 0x00 }, + { 0x0235, 0x00 }, + { 0x0236, 0x00 }, + { 0x0237, 0x00 }, + { 0x0400, 0x00 }, + { 0x0401, 0x00 }, + { 0x0402, 0x00 }, + { 0x0403, 0x00 }, + { 0x0404, 0x00 }, + { 0x0405, 0x03 }, + { 0x0420, 0x00 }, + { 0x0422, 0x00 }, + { 0x0423, 0x00 }, + { 0x0424, 0x00 }, + { 0x0425, 0x00 }, + { 0x0426, 0x00 }, + { 0x0427, 0x00 }, + { 0x0430, 0x00 }, + { 0x0432, 0x00 }, + { 0x0433, 0x00 }, + { 0x0434, 0x00 }, + { 0x0435, 0x00 }, + { 0x0436, 0x00 }, + { 0x0437, 0x00 }, + { 0x0f00, 0x00 }, + { 0x0f01, 0x00 }, + { 0x0f02, 0x00 }, + { 0x0f03, 0x00 }, + { 0x0f04, 0x00 }, + { 0x0f05, 0x00 }, + { 0x0f20, 0x00 }, + { 0x0f22, 0x00 }, + { 0x0f23, 0x00 }, + { 0x0f24, 0x00 }, + { 0x0f25, 0x00 }, + { 0x0f26, 0x00 }, + { 0x0f27, 0x00 }, + { 0x0f30, 0x00 }, + { 0x0f32, 0x00 }, + { 0x0f33, 0x00 }, + { 0x0f34, 0x00 }, + { 0x0f35, 0x00 }, + { 0x0f36, 0x00 }, + { 0x0f37, 0x00 }, + { 0x2f01, 0x01 }, + { 0x2f02, 0x09 }, + { 0x2f03, 0x00 }, + { 0x2f04, 0x0f }, + { 0x2f05, 0x0b }, + { 0x2f06, 0x01 }, + { 0x2f07, 0x8e }, + { 0x3000, 0x00 }, + { 0x3001, 0x00 }, + { 0x3004, 0x01 }, + { 0x3005, 0x23 }, + { 0x3008, 0x02 }, + { 0x300a, 0x00 }, + { 0xc003 | (RT1308_DAC_SET << 4), 0x00 }, + { 0xc001 | (RT1308_POWER << 4), 0x00 }, + { 0xc002 | (RT1308_POWER << 4), 0x00 }, +}; + +#define RT1308_SDW_OFFSET 0xc000 +#define RT1308_SDW_OFFSET_BYTE0 0xc000 +#define RT1308_SDW_OFFSET_BYTE1 0xc001 +#define RT1308_SDW_OFFSET_BYTE2 0xc002 +#define RT1308_SDW_OFFSET_BYTE3 0xc003 + +#define RT1308_SDW_RESET (RT1308_SDW_OFFSET | (RT1308_RESET << 4)) + +struct rt1308_sdw_priv { + struct snd_soc_component *component; + struct regmap *regmap; + struct sdw_slave *sdw_slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +#endif /* __RT1308_SDW_H__ */ -- 2.24.1

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[alsa-devel] [PATCH v1] ASoC: tlv320aic32x4: handle regmap_read error gracefully
by Peter Seiderer 09 Jan '20

09 Jan '20

Fixes: [ 5.169310] Division by zero in kernel. [ 5.200998] CPU: 1 PID: 1 Comm: swapper/0 Not tainted 5.3.18-20191021-1+ #14 [ 5.203049] cdc_acm 2-1.6:1.0: ttyACM0: USB ACM device [ 5.208198] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 5.220084] Backtrace: [ 5.222628] [<8010f60c>] (dump_backtrace) from [<8010f9a8>] (show_stack+0x20/0x24) [ 5.230348] r7:810a1c6c r6:00000000 r5:60000013 r4:810a1c6c [ 5.236097] [<8010f988>] (show_stack) from [<809e06a0>] (dump_stack+0xac/0xd8) [ 5.243469] [<809e05f4>] (dump_stack) from [<8010f780>] (__div0+0x24/0x28) [ 5.250431] r9:8111adc8 r8:ae631180 r7:aebd27c0 r6:ae631e40 r5:00000000 r4:81006508 [ 5.258325] [<8010f75c>] (__div0) from [<809de7ac>] (Ldiv0+0x8/0x10) [ 5.264841] [<8085c7e0>] (clk_aic32x4_div_recalc_rate) from [<805ba70c>] (__clk_register+0x2f8/0x7e4) [ 5.274141] r5:80dd065c r4:ae6bd480 [ 5.277869] [<805ba414>] (__clk_register) from [<805bace0>] (devm_clk_register+0x58/0x8c) [ 5.286130] r10:81006508 r9:810946d4 r8:00000000 r7:ae8de1c0 r6:ae631ac0 r5:ae631e40 [ 5.294103] r4:ae8d8020 [ 5.296724] [<805bac88>] (devm_clk_register) from [<8085cea8>] (aic32x4_register_clocks+0x120/0x14c) [ 5.306004] r7:ae8de1c0 r6:ae8d8020 r5:ae631e40 r4:810946c0 [ 5.311818] [<8085cd88>] (aic32x4_register_clocks) from [<8085bf60>] (aic32x4_probe+0x94/0x468) [ 5.320602] r10:81094730 r9:00000000 r8:af361fc0 r7:bfd6d040 r6:00000000 r5:ae8d8020 [ 5.328574] r4:af361e40 [ 5.331195] [<8085becc>] (aic32x4_probe) from [<8085cf60>] (aic32x4_i2c_probe+0x6c/0x88) [ 5.339434] r8:00000000 r7:ae8d8000 r6:81094730 r5:ae8d8000 r4:81006508 [ 5.346288] [<8085cef4>] (aic32x4_i2c_probe) from [<807554b0>] (i2c_device_probe+0x2ac/0x2f0) [ 5.354894] r5:8085cef4 r4:ae8d8020 [ 5.358625] [<80755204>] (i2c_device_probe) from [<80678e34>] (really_probe+0x11c/0x428) [ 5.366802] r9:00000000 r8:810b3e78 r7:00000000 r6:8111e020 r5:ae8d8020 r4:8111e01c [ 5.374694] [<80678d18>] (really_probe) from [<80679388>] (driver_probe_device+0x88/0x1e0) [ 5.383106] r10:80f63860 r9:ffffe000 r8:ffffe000 r7:80679794 r6:81094730 r5:81094730 [ 5.391080] r4:ae8d8020 [ 5.393702] [<80679300>] (driver_probe_device) from [<8067978c>] (device_driver_attach+0x68/0x70) [ 5.402724] r9:ffffe000 r8:ffffe000 r7:80679794 r6:81094730 r5:00000000 r4:ae8d8020 [ 5.410555] [<80679724>] (device_driver_attach) from [<80679858>] (__driver_attach+0xc4/0x164) [ 5.419313] r7:80679794 r6:ae8d8020 r5:81094730 r4:00000000 [ 5.425123] [<80679794>] (__driver_attach) from [<80676a14>] (bus_for_each_dev+0x84/0xc4) [ 5.433384] r7:80679794 r6:81094730 r5:81006508 r4:ae8dc0c0 [ 5.439192] [<80676990>] (bus_for_each_dev) from [<80678668>] (driver_attach+0x2c/0x30) [ 5.447279] r7:00000000 r6:af361500 r5:8107fd94 r4:81094730 [ 5.453087] [<8067863c>] (driver_attach) from [<80677fc4>] (bus_add_driver+0x1d0/0x210) [ 5.461240] [<80677df4>] (bus_add_driver) from [<80679f34>] (driver_register+0x84/0x118) [ 5.469414] r7:00000000 r6:80f4ac9c r5:81006508 r4:81094730 [ 5.475224] [<80679eb0>] (driver_register) from [<80755dfc>] (i2c_register_driver+0x4c/0xb8) [ 5.483807] r5:81006508 r4:81094714 [ 5.487472] [<80755db0>] (i2c_register_driver) from [<80f4acc0>] (aic32x4_i2c_driver_init+0x24/0x28) [ 5.496750] r5:81006508 r4:810a7180 [ 5.500415] [<80f4ac9c>] (aic32x4_i2c_driver_init) from [<80103288>] (do_one_initcall+0x64/0x2d0) [ 5.509442] [<80103224>] (do_one_initcall) from [<80f014a8>] (kernel_init_freeable+0x300/0x390) [ 5.518287] r8:810c7300 r7:810c7300 r6:00000007 r5:80f920c4 r4:80f63840 [ 5.525079] [<80f011a8>] (kernel_init_freeable) from [<809f892c>] (kernel_init+0x18/0x124) [ 5.533490] r10:00000000 r9:00000000 r8:00000000 r7:00000000 r6:00000000 r5:809f8914 [ 5.541461] r4:00000000 [ 5.544084] [<809f8914>] (kernel_init) from [<801010b4>] (ret_from_fork+0x14/0x20) [ 5.551800] Exception stack(0xaf115fb0 to 0xaf115ff8) [ 5.556935] 5fa0: 00000000 00000000 00000000 00000000 [ 5.565262] 5fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 5.573522] 5fe0: 00000000 00000000 00000000 00000000 00000013 00000000 [ 5.580283] r5:809f8914 r4:00000000 Signed-off-by: Peter Seiderer <ps.report(a)gmx.net> --- sound/soc/codecs/tlv320aic32x4-clk.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 156c153c12ab..7a82e3448780 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -338,7 +338,8 @@ static unsigned long clk_aic32x4_div_recalc_rate(struct clk_hw *hw, unsigned int val; - regmap_read(div->regmap, div->reg, &val); + if (regmap_read(div->regmap, div->reg, &val)) + return 0; return DIV_ROUND_UP(parent_rate, val & AIC32X4_DIV_MASK); } -- 2.24.0

2 5

[alsa-devel] [PATCH] ASoC: stm32: sai: fix possible circular locking
by Olivier Moysan 09 Jan '20

09 Jan '20

In current driver, locks can be taken as follows: - Register access: take a lock on regmap config and then on clock. - Master clock provider: take a lock on clock and then on regmap config. This can lead to the circular locking summarized below. Remove peripheral clock management through regmap framework, and manage peripheral clock in driver instead. On register access, lock on clock is taken first, which allows to avoid possible locking issue. [ 6696.561513] ====================================================== [ 6696.567670] WARNING: possible circular locking dependency detected [ 6696.573842] 4.19.49 #866 Not tainted [ 6696.577397] ------------------------------------------------------ [ 6696.583566] pulseaudio/6439 is trying to acquire lock: [ 6696.588697] 87b0a25b (enable_lock){..-.}, at: clk_enable_lock+0x64/0x128 [ 6696.595377] [ 6696.595377] but task is already holding lock: [ 6696.601197] d858f825 (stm32_sai_sub:1342:(sai->regmap_config)->lock){....} ... [ 6696.812513] Possible unsafe locking scenario: [ 6696.812513] [ 6696.818418] CPU0 CPU1 [ 6696.822935] ---- ---- [ 6696.827451] lock(stm32_sai_sub:1342:(sai->regmap_config)->lock); [ 6696.833618] lock(enable_lock); [ 6696.839350] lock(stm32_sai_sub:1342: (sai->regmap_config)->lock); [ 6696.848035] lock(enable_lock); Fixes: 03e78a242a15 ("ASoC: stm32: sai: add h7 support") Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 194 ++++++++++++++++++++++++---------- 1 file changed, 140 insertions(+), 54 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 48e629ac2d88..30bcd5d3a32a 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -184,6 +184,56 @@ static bool stm32_sai_sub_writeable_reg(struct device *dev, unsigned int reg) } } +static int stm32_sai_sub_reg_up(struct stm32_sai_sub_data *sai, + unsigned int reg, unsigned int mask, + unsigned int val) +{ + int ret; + + ret = clk_enable(sai->pdata->pclk); + if (ret < 0) + return ret; + + ret = regmap_update_bits(sai->regmap, reg, mask, val); + + clk_disable(sai->pdata->pclk); + + return ret; +} + +static int stm32_sai_sub_reg_wr(struct stm32_sai_sub_data *sai, + unsigned int reg, unsigned int mask, + unsigned int val) +{ + int ret; + + ret = clk_enable(sai->pdata->pclk); + if (ret < 0) + return ret; + + ret = regmap_write_bits(sai->regmap, reg, mask, val); + + clk_disable(sai->pdata->pclk); + + return ret; +} + +static int stm32_sai_sub_reg_rd(struct stm32_sai_sub_data *sai, + unsigned int reg, unsigned int *val) +{ + int ret; + + ret = clk_enable(sai->pdata->pclk); + if (ret < 0) + return ret; + + ret = regmap_read(sai->regmap, reg, val); + + clk_disable(sai->pdata->pclk); + + return ret; +} + static const struct regmap_config stm32_sai_sub_regmap_config_f4 = { .reg_bits = 32, .reg_stride = 4, @@ -295,7 +345,7 @@ static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai, mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version)); cr1 = SAI_XCR1_MCKDIV_SET(div); - ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1); + ret = stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, mask, cr1); if (ret < 0) dev_err(&sai->pdev->dev, "Failed to update CR1 register\n"); @@ -372,8 +422,8 @@ static int stm32_sai_mclk_enable(struct clk_hw *hw) dev_dbg(&sai->pdev->dev, "Enable master clock\n"); - return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, - SAI_XCR1_MCKEN, SAI_XCR1_MCKEN); + return stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, + SAI_XCR1_MCKEN, SAI_XCR1_MCKEN); } static void stm32_sai_mclk_disable(struct clk_hw *hw) @@ -383,7 +433,7 @@ static void stm32_sai_mclk_disable(struct clk_hw *hw) dev_dbg(&sai->pdev->dev, "Disable master clock\n"); - regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0); + stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0); } static const struct clk_ops mclk_ops = { @@ -446,15 +496,15 @@ static irqreturn_t stm32_sai_isr(int irq, void *devid) unsigned int sr, imr, flags; snd_pcm_state_t status = SNDRV_PCM_STATE_RUNNING; - regmap_read(sai->regmap, STM_SAI_IMR_REGX, &imr); - regmap_read(sai->regmap, STM_SAI_SR_REGX, &sr); + stm32_sai_sub_reg_rd(sai, STM_SAI_IMR_REGX, &imr); + stm32_sai_sub_reg_rd(sai, STM_SAI_SR_REGX, &sr); flags = sr & imr; if (!flags) return IRQ_NONE; - regmap_write_bits(sai->regmap, STM_SAI_CLRFR_REGX, SAI_XCLRFR_MASK, - SAI_XCLRFR_MASK); + stm32_sai_sub_reg_wr(sai, STM_SAI_CLRFR_REGX, SAI_XCLRFR_MASK, + SAI_XCLRFR_MASK); if (!sai->substream) { dev_err(&pdev->dev, "Device stopped. Spurious IRQ 0x%x\n", sr); @@ -503,8 +553,8 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, int ret; if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) { - ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, - SAI_XCR1_NODIV, + ret = stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, + SAI_XCR1_NODIV, freq ? 0 : SAI_XCR1_NODIV); if (ret < 0) return ret; @@ -583,7 +633,7 @@ static int stm32_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, slotr_mask |= SAI_XSLOTR_SLOTEN_MASK; - regmap_update_bits(sai->regmap, STM_SAI_SLOTR_REGX, slotr_mask, slotr); + stm32_sai_sub_reg_up(sai, STM_SAI_SLOTR_REGX, slotr_mask, slotr); sai->slot_width = slot_width; sai->slots = slots; @@ -665,7 +715,7 @@ static int stm32_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) cr1_mask |= SAI_XCR1_CKSTR; frcr_mask |= SAI_XFRCR_FSPOL; - regmap_update_bits(sai->regmap, STM_SAI_FRCR_REGX, frcr_mask, frcr); + stm32_sai_sub_reg_up(sai, STM_SAI_FRCR_REGX, frcr_mask, frcr); /* DAI clock master masks */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -693,7 +743,7 @@ static int stm32_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) cr1_mask |= SAI_XCR1_SLAVE; conf_update: - ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, cr1_mask, cr1); + ret = stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, cr1_mask, cr1); if (ret < 0) { dev_err(cpu_dai->dev, "Failed to update CR1 register\n"); return ret; @@ -730,12 +780,12 @@ static int stm32_sai_startup(struct snd_pcm_substream *substream, } /* Enable ITs */ - regmap_write_bits(sai->regmap, STM_SAI_CLRFR_REGX, - SAI_XCLRFR_MASK, SAI_XCLRFR_MASK); + stm32_sai_sub_reg_wr(sai, STM_SAI_CLRFR_REGX, + SAI_XCLRFR_MASK, SAI_XCLRFR_MASK); imr = SAI_XIMR_OVRUDRIE; if (STM_SAI_IS_CAPTURE(sai)) { - regmap_read(sai->regmap, STM_SAI_CR2_REGX, &cr2); + stm32_sai_sub_reg_rd(sai, STM_SAI_CR2_REGX, &cr2); if (cr2 & SAI_XCR2_MUTECNT_MASK) imr |= SAI_XIMR_MUTEDETIE; } @@ -745,8 +795,8 @@ static int stm32_sai_startup(struct snd_pcm_substream *substream, else imr |= SAI_XIMR_AFSDETIE | SAI_XIMR_LFSDETIE; - regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX, - SAI_XIMR_MASK, imr); + stm32_sai_sub_reg_up(sai, STM_SAI_IMR_REGX, + SAI_XIMR_MASK, imr); return 0; } @@ -763,10 +813,10 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, * SAI fifo threshold is set to half fifo, to keep enough space * for DMA incoming bursts. */ - regmap_write_bits(sai->regmap, STM_SAI_CR2_REGX, - SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK, - SAI_XCR2_FFLUSH | - SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); + stm32_sai_sub_reg_wr(sai, STM_SAI_CR2_REGX, + SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK, + SAI_XCR2_FFLUSH | + SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); /* DS bits in CR1 not set for SPDIF (size forced to 24 bits).*/ if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { @@ -795,7 +845,7 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, if ((sai->slots == 2) && (params_channels(params) == 1)) cr1 |= SAI_XCR1_MONO; - ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, cr1_mask, cr1); + ret = stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, cr1_mask, cr1); if (ret < 0) { dev_err(cpu_dai->dev, "Failed to update CR1 register\n"); return ret; @@ -809,7 +859,7 @@ static int stm32_sai_set_slots(struct snd_soc_dai *cpu_dai) struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int slotr, slot_sz; - regmap_read(sai->regmap, STM_SAI_SLOTR_REGX, &slotr); + stm32_sai_sub_reg_rd(sai, STM_SAI_SLOTR_REGX, &slotr); /* * If SLOTSZ is set to auto in SLOTR, align slot width on data size @@ -831,16 +881,16 @@ static int stm32_sai_set_slots(struct snd_soc_dai *cpu_dai) sai->slots = 2; /* The number of slots in the audio frame is equal to NBSLOT[3:0] + 1*/ - regmap_update_bits(sai->regmap, STM_SAI_SLOTR_REGX, - SAI_XSLOTR_NBSLOT_MASK, - SAI_XSLOTR_NBSLOT_SET((sai->slots - 1))); + stm32_sai_sub_reg_up(sai, STM_SAI_SLOTR_REGX, + SAI_XSLOTR_NBSLOT_MASK, + SAI_XSLOTR_NBSLOT_SET((sai->slots - 1))); /* Set default slots mask if not already set from DT */ if (!(slotr & SAI_XSLOTR_SLOTEN_MASK)) { sai->slot_mask = (1 << sai->slots) - 1; - regmap_update_bits(sai->regmap, - STM_SAI_SLOTR_REGX, SAI_XSLOTR_SLOTEN_MASK, - SAI_XSLOTR_SLOTEN_SET(sai->slot_mask)); + stm32_sai_sub_reg_up(sai, + STM_SAI_SLOTR_REGX, SAI_XSLOTR_SLOTEN_MASK, + SAI_XSLOTR_SLOTEN_SET(sai->slot_mask)); } dev_dbg(cpu_dai->dev, "Slots %d, slot width %d\n", @@ -870,14 +920,14 @@ static void stm32_sai_set_frame(struct snd_soc_dai *cpu_dai) dev_dbg(cpu_dai->dev, "Frame length %d, frame active %d\n", sai->fs_length, fs_active); - regmap_update_bits(sai->regmap, STM_SAI_FRCR_REGX, frcr_mask, frcr); + stm32_sai_sub_reg_up(sai, STM_SAI_FRCR_REGX, frcr_mask, frcr); if ((sai->fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_LSB) { offset = sai->slot_width - sai->data_size; - regmap_update_bits(sai->regmap, STM_SAI_SLOTR_REGX, - SAI_XSLOTR_FBOFF_MASK, - SAI_XSLOTR_FBOFF_SET(offset)); + stm32_sai_sub_reg_up(sai, STM_SAI_SLOTR_REGX, + SAI_XSLOTR_FBOFF_MASK, + SAI_XSLOTR_FBOFF_SET(offset)); } } @@ -994,9 +1044,9 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, return -EINVAL; } - regmap_update_bits(sai->regmap, - STM_SAI_CR1_REGX, - SAI_XCR1_OSR, cr1); + stm32_sai_sub_reg_up(sai, + STM_SAI_CR1_REGX, + SAI_XCR1_OSR, cr1); div = stm32_sai_get_clk_div(sai, sai_clk_rate, sai->mclk_rate); @@ -1058,12 +1108,12 @@ static int stm32_sai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: dev_dbg(cpu_dai->dev, "Enable DMA and SAI\n"); - regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, - SAI_XCR1_DMAEN, SAI_XCR1_DMAEN); + stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, + SAI_XCR1_DMAEN, SAI_XCR1_DMAEN); /* Enable SAI */ - ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, - SAI_XCR1_SAIEN, SAI_XCR1_SAIEN); + ret = stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, + SAI_XCR1_SAIEN, SAI_XCR1_SAIEN); if (ret < 0) dev_err(cpu_dai->dev, "Failed to update CR1 register\n"); break; @@ -1072,16 +1122,16 @@ static int stm32_sai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: dev_dbg(cpu_dai->dev, "Disable DMA and SAI\n"); - regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX, - SAI_XIMR_MASK, 0); + stm32_sai_sub_reg_up(sai, STM_SAI_IMR_REGX, + SAI_XIMR_MASK, 0); - regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, - SAI_XCR1_SAIEN, - (unsigned int)~SAI_XCR1_SAIEN); + stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, + SAI_XCR1_SAIEN, + (unsigned int)~SAI_XCR1_SAIEN); - ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, - SAI_XCR1_DMAEN, - (unsigned int)~SAI_XCR1_DMAEN); + ret = stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, + SAI_XCR1_DMAEN, + (unsigned int)~SAI_XCR1_DMAEN); if (ret < 0) dev_err(cpu_dai->dev, "Failed to update CR1 register\n"); @@ -1101,7 +1151,7 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); unsigned long flags; - regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX, SAI_XIMR_MASK, 0); + stm32_sai_sub_reg_up(sai, STM_SAI_IMR_REGX, SAI_XIMR_MASK, 0); clk_disable_unprepare(sai->sai_ck); @@ -1169,7 +1219,7 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) cr1_mask |= SAI_XCR1_SYNCEN_MASK; cr1 |= SAI_XCR1_SYNCEN_SET(sai->sync); - return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, cr1_mask, cr1); + return stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, cr1_mask, cr1); } static const struct snd_soc_dai_ops stm32_sai_pcm_dai_ops = { @@ -1322,8 +1372,13 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, if (STM_SAI_HAS_PDM(sai) && STM_SAI_IS_SUB_A(sai)) sai->regmap_config = &stm32_sai_sub_regmap_config_h7; - sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "sai_ck", - base, sai->regmap_config); + /* + * Do not manage peripheral clock through regmap framework as this + * can lead to circular locking issue with sai master clock provider. + * Manage peripheral clock directly in driver instead. + */ + sai->regmap = devm_regmap_init_mmio(&pdev->dev, base, + sai->regmap_config); if (IS_ERR(sai->regmap)) { dev_err(&pdev->dev, "Failed to initialize MMIO\n"); return PTR_ERR(sai->regmap); @@ -1420,6 +1475,10 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, return PTR_ERR(sai->sai_ck); } + ret = clk_prepare(sai->pdata->pclk); + if (ret < 0) + return ret; + if (STM_SAI_IS_F4(sai->pdata)) return 0; @@ -1501,22 +1560,48 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return 0; } +static int stm32_sai_sub_remove(struct platform_device *pdev) +{ + struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev); + + clk_unprepare(sai->pdata->pclk); + + return 0; +} + #ifdef CONFIG_PM_SLEEP static int stm32_sai_sub_suspend(struct device *dev) { struct stm32_sai_sub_data *sai = dev_get_drvdata(dev); + int ret; + + ret = clk_enable(sai->pdata->pclk); + if (ret < 0) + return ret; regcache_cache_only(sai->regmap, true); regcache_mark_dirty(sai->regmap); + + clk_disable(sai->pdata->pclk); + return 0; } static int stm32_sai_sub_resume(struct device *dev) { struct stm32_sai_sub_data *sai = dev_get_drvdata(dev); + int ret; + + ret = clk_enable(sai->pdata->pclk); + if (ret < 0) + return ret; regcache_cache_only(sai->regmap, false); - return regcache_sync(sai->regmap); + ret = regcache_sync(sai->regmap); + + clk_disable(sai->pdata->pclk); + + return ret; } #endif /* CONFIG_PM_SLEEP */ @@ -1531,6 +1616,7 @@ static struct platform_driver stm32_sai_sub_driver = { .pm = &stm32_sai_sub_pm_ops, }, .probe = stm32_sai_sub_probe, + .remove = stm32_sai_sub_remove, }; module_platform_driver(stm32_sai_sub_driver); -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: rt711: add rt711 codec driver
by shumingf＠realtek.com 09 Jan '20

09 Jan '20

From: Shuming Fan <shumingf(a)realtek.com> This is the initial codec driver for rt711. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt711-sdw.c | 552 +++++++++++++++ sound/soc/codecs/rt711-sdw.h | 281 ++++++++ sound/soc/codecs/rt711.c | 1293 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt711.h | 227 ++++++ 6 files changed, 2365 insertions(+) create mode 100644 sound/soc/codecs/rt711-sdw.c create mode 100644 sound/soc/codecs/rt711-sdw.h create mode 100644 sound/soc/codecs/rt711.c create mode 100644 sound/soc/codecs/rt711.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 146682049007..f58c6d35e47a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -165,6 +165,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_RT5682 if I2C + select SND_SOC_RT711_SDW if SOUNDWIRE select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIMPLE_AMPLIFIER @@ -1058,6 +1059,15 @@ config SND_SOC_RT5677_SPI config SND_SOC_RT5682 tristate +config SND_SOC_RT711 + tristate + +config SND_SOC_RT711_SDW + tristate "Realtek RT711 Codec - SDW" + depends on SOUNDWIRE + select SND_SOC_RT711 + select REGMAP_SOUNDWIRE + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0290fb389835..3fb6d0c04f5d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -173,6 +173,7 @@ snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o snd-soc-rt5682-objs := rt5682.o +snd-soc-rt711-objs := rt711.o rt711-sdw.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -464,6 +465,7 @@ obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o +obj-$(CONFIG_SND_SOC_RT711) += snd-soc-rt711.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c new file mode 100644 index 000000000000..e28dc84ede5b --- /dev/null +++ b/sound/soc/codecs/rt711-sdw.c @@ -0,0 +1,552 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rt711-sdw.c -- rt711 ALSA SoC audio driver +// +// Copyright(c) 2019 Realtek Semiconductor Corp. +// +// + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/mod_devicetable.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include "rt711.h" +#include "rt711-sdw.h" + +static bool rt711_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x00e0: + case 0x00f0: + case 0x2012 ... 0x2016: + case 0x201a ... 0x2027: + case 0x2029 ... 0x202a: + case 0x202d ... 0x2034: + case 0x2201 ... 0x2204: + case 0x2206 ... 0x2212: + case 0x2220 ... 0x2223: + case 0x2230 ... 0x2239: + case 0x2f01 ... 0x2f0f: + case 0x3000 ... 0x3fff: + case 0x7000 ... 0x7fff: + case 0x8300 ... 0x83ff: + case 0x9c00 ... 0x9cff: + case 0xb900 ... 0xb9ff: + case 0x752009: + case 0x752011: + case 0x75201a: + case 0x752045: + case 0x752046: + case 0x752048: + case 0x75204a: + case 0x75206b: + case 0x75206f: + case 0x752080: + case 0x752081: + case 0x752091: + case 0x755800: + return true; + default: + return false; + } +} + +static bool rt711_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2016: + case 0x201b: + case 0x201c: + case 0x201d: + case 0x201f: + case 0x2021: + case 0x2023: + case 0x2230: + case 0x2012 ... 0x2015: /* HD-A read */ + case 0x202d ... 0x202f: /* BRA */ + case 0x2201 ... 0x2212: /* i2c debug */ + case 0x2220 ... 0x2223: /* decoded HD-A */ + case 0x9c00 ... 0x9cff: + case 0xb900 ... 0xb9ff: + case 0xff01: + case 0x75201a: + case 0x752046: + case 0x752080: + case 0x752081: + case 0x755800: + return true; + default: + return false; + } +} + +static int rt711_sdw_read(void *context, unsigned int reg, unsigned int *val) +{ + struct device *dev = context; + struct rt711_priv *rt711 = dev_get_drvdata(dev); + unsigned int sdw_data_3, sdw_data_2, sdw_data_1, sdw_data_0; + unsigned int reg2 = 0, reg3 = 0, reg4 = 0, mask, nid, val2; + unsigned int is_hda_reg = 1, is_index_reg = 0; + int ret; + + if (reg > 0xffff) + is_index_reg = 1; + + mask = reg & 0xf000; + + if (is_index_reg) { /* index registers */ + val2 = reg & 0xff; + reg = reg >> 8; + nid = reg & 0xff; + ret = regmap_write(rt711->sdw_regmap, reg, 0); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt711->sdw_regmap, reg2, val2); + if (ret < 0) + return ret; + + reg3 = RT711_PRIV_DATA_R_H | nid; + ret = regmap_write(rt711->sdw_regmap, + reg3, ((*val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg4 = reg3 + 0x1000; + reg4 |= 0x80; + ret = regmap_write(rt711->sdw_regmap, reg4, (*val & 0xff)); + if (ret < 0) + return ret; + } else if (mask == 0x3000) { + reg += 0x8000; + ret = regmap_write(rt711->sdw_regmap, reg, *val); + if (ret < 0) + return ret; + } else if (mask == 0x7000) { + reg += 0x2000; + reg |= 0x800; + ret = regmap_write(rt711->sdw_regmap, + reg, ((*val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt711->sdw_regmap, reg2, (*val & 0xff)); + if (ret < 0) + return ret; + } else if ((reg & 0xff00) == 0x8300) { /* for R channel */ + reg2 = reg - 0x1000; + reg2 &= ~0x80; + ret = regmap_write(rt711->sdw_regmap, + reg2, ((*val >> 8) & 0xff)); + if (ret < 0) + return ret; + ret = regmap_write(rt711->sdw_regmap, reg, (*val & 0xff)); + if (ret < 0) + return ret; + } else if (mask == 0x9000) { + ret = regmap_write(rt711->sdw_regmap, + reg, ((*val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt711->sdw_regmap, reg2, (*val & 0xff)); + if (ret < 0) + return ret; + } else if (mask == 0xb000) { + ret = regmap_write(rt711->sdw_regmap, reg, *val); + if (ret < 0) + return ret; + } else { + ret = regmap_read(rt711->sdw_regmap, reg, val); + if (ret < 0) + return ret; + is_hda_reg = 0; + } + + if (is_hda_reg || is_index_reg) { + sdw_data_3 = 0; + sdw_data_2 = 0; + sdw_data_1 = 0; + sdw_data_0 = 0; + ret = regmap_read(rt711->sdw_regmap, + RT711_READ_HDA_3, &sdw_data_3); + if (ret < 0) + return ret; + ret = regmap_read(rt711->sdw_regmap, + RT711_READ_HDA_2, &sdw_data_2); + if (ret < 0) + return ret; + ret = regmap_read(rt711->sdw_regmap, + RT711_READ_HDA_1, &sdw_data_1); + if (ret < 0) + return ret; + ret = regmap_read(rt711->sdw_regmap, + RT711_READ_HDA_0, &sdw_data_0); + if (ret < 0) + return ret; + *val = ((sdw_data_3 & 0xff) << 24) | + ((sdw_data_2 & 0xff) << 16) | + ((sdw_data_1 & 0xff) << 8) | (sdw_data_0 & 0xff); + } + + if (is_hda_reg == 0) + dev_dbg(dev, "[%s] %04x => %08x\n", __func__, reg, *val); + else if (is_index_reg) + dev_dbg(dev, "[%s] %04x %04x %04x %04x => %08x\n", + __func__, reg, reg2, reg3, reg4, *val); + else + dev_dbg(dev, "[%s] %04x %04x => %08x\n", + __func__, reg, reg2, *val); + + return 0; +} + +static int rt711_sdw_write(void *context, unsigned int reg, unsigned int val) +{ + struct device *dev = context; + struct rt711_priv *rt711 = dev_get_drvdata(dev); + unsigned int reg2 = 0, reg3, reg4, nid, mask, val2; + unsigned int is_index_reg = 0; + int ret; + + if (reg > 0xffff) + is_index_reg = 1; + + mask = reg & 0xf000; + + if (is_index_reg) { /* index registers */ + val2 = reg & 0xff; + reg = reg >> 8; + nid = reg & 0xff; + ret = regmap_write(rt711->sdw_regmap, reg, 0); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt711->sdw_regmap, reg2, val2); + if (ret < 0) + return ret; + + reg3 = RT711_PRIV_DATA_W_H | nid; + ret = regmap_write(rt711->sdw_regmap, + reg3, ((val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg4 = reg3 + 0x1000; + reg4 |= 0x80; + ret = regmap_write(rt711->sdw_regmap, reg4, (val & 0xff)); + if (ret < 0) + return ret; + is_index_reg = 1; + } else if (reg < 0x4fff) { + ret = regmap_write(rt711->sdw_regmap, reg, val); + if (ret < 0) + return ret; + } else if (reg == RT711_FUNC_RESET) { + ret = regmap_write(rt711->sdw_regmap, reg, val); + if (ret < 0) + return ret; + } else if (mask == 0x7000) { + ret = regmap_write(rt711->sdw_regmap, + reg, ((val >> 8) & 0xff)); + if (ret < 0) + return ret; + reg2 = reg + 0x1000; + reg2 |= 0x80; + ret = regmap_write(rt711->sdw_regmap, reg2, (val & 0xff)); + if (ret < 0) + return ret; + } else if ((reg & 0xff00) == 0x8300) { /* for R channel */ + reg2 = reg - 0x1000; + reg2 &= ~0x80; + ret = regmap_write(rt711->sdw_regmap, + reg2, ((val >> 8) & 0xff)); + if (ret < 0) + return ret; + ret = regmap_write(rt711->sdw_regmap, reg, (val & 0xff)); + if (ret < 0) + return ret; + } + + if (reg2 == 0) + dev_dbg(dev, "[%s] %04x <= %04x\n", __func__, reg, val); + else if (is_index_reg) + dev_dbg(dev, "[%s] %04x %04x %04x %04x <= %04x %04x\n", + __func__, reg, reg2, reg3, reg4, val2, val); + else + dev_dbg(dev, "[%s] %04x %04x <= %04x\n", + __func__, reg, reg2, val); + + return 0; +} + +static const struct regmap_config rt711_regmap = { + .reg_bits = 24, + .val_bits = 32, + .readable_reg = rt711_readable_register, + .volatile_reg = rt711_volatile_register, + .max_register = 0x755800, + .reg_defaults = rt711_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rt711_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, + .reg_read = rt711_sdw_read, + .reg_write = rt711_sdw_write, +}; + +static const struct regmap_config rt711_sdw_regmap = { + .name = "sdw", + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt711_readable_register, + .max_register = 0xff01, + .cache_type = REGCACHE_NONE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt711_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt711_priv *rt711 = dev_get_drvdata(&slave->dev); + + /* Update the status */ + rt711->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + rt711->hw_init = false; + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt711->hw_init || rt711->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt711_io_init(&slave->dev, slave); +} + +static int rt711_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval, i, num_of_ports = 1; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->paging_support = false; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x14; /* BITMAP: 00010100 */ + prop->sink_ports = 0x8; /* BITMAP: 00001000 */ + + nval = hweight32(prop->source_ports); + num_of_ports += nval; + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), + GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + num_of_ports += nval; + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), + GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* Allocate port_ready based on num_of_ports */ + slave->port_ready = devm_kcalloc(&slave->dev, num_of_ports, + sizeof(*slave->port_ready), + GFP_KERNEL); + if (!slave->port_ready) + return -ENOMEM; + + /* Initialize completion */ + for (i = 0; i < num_of_ports; i++) + init_completion(&slave->port_ready[i]); + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + /* wake-up event */ + prop->wake_capable = 1; + + return 0; +} + +static int rt711_bus_config(struct sdw_slave *slave, + struct sdw_bus_params *params) +{ + struct rt711_priv *rt711 = dev_get_drvdata(&slave->dev); + int ret; + + memcpy(&rt711->params, params, sizeof(*params)); + + ret = rt711_clock_config(&slave->dev); + if (ret < 0) + dev_err(&slave->dev, "Invalid clk config"); + + return ret; +} + +static int rt711_interrupt_callback(struct sdw_slave *slave, + struct sdw_slave_intr_status *status) +{ + struct rt711_priv *rt711 = dev_get_drvdata(&slave->dev); + + dev_dbg(&slave->dev, + "%s control_port_stat=%x", __func__, status->control_port); + + if (status->control_port & 0x4) { + mod_delayed_work(system_power_efficient_wq, + &rt711->jack_detect_work, msecs_to_jiffies(250)); + } + + return 0; +} + +static struct sdw_slave_ops rt711_slave_ops = { + .read_prop = rt711_read_prop, + .interrupt_callback = rt711_interrupt_callback, + .update_status = rt711_update_status, + .bus_config = rt711_bus_config, +}; + +static int rt711_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *sdw_regmap, *regmap; + + /* Assign ops */ + slave->ops = &rt711_slave_ops; + + /* Regmap Initialization */ + sdw_regmap = devm_regmap_init_sdw(slave, &rt711_sdw_regmap); + if (!sdw_regmap) + return -EINVAL; + + regmap = devm_regmap_init(&slave->dev, NULL, + &slave->dev, &rt711_regmap); + if (!regmap) + return -EINVAL; + + rt711_init(&slave->dev, sdw_regmap, regmap, slave); + + return 0; +} + +static int rt711_sdw_remove(struct sdw_slave *slave) +{ + struct rt711_priv *rt711 = dev_get_drvdata(&slave->dev); + + if (rt711 && rt711->hw_init) { + cancel_delayed_work(&rt711->jack_detect_work); + cancel_delayed_work(&rt711->jack_btn_check_work); + cancel_work_sync(&rt711->calibration_work); + } + + return 0; +} + +static const struct sdw_device_id rt711_id[] = { + SDW_SLAVE_ENTRY(0x025d, 0x711, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt711_id); + +static int rt711_dev_suspend(struct device *dev) +{ + struct rt711_priv *rt711 = dev_get_drvdata(dev); + + if (!rt711->hw_init) + return 0; + + regcache_cache_only(rt711->regmap, true); + + return 0; +} + +#define RT711_PROBE_TIMEOUT 2000 + +static int rt711_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt711_priv *rt711 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt711->hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT711_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt711->regmap, false); + regcache_sync_region(rt711->regmap, 0x3000, 0x8fff); + regcache_sync_region(rt711->regmap, 0x752009, 0x752091); + + return 0; +} + +static const struct dev_pm_ops rt711_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt711_dev_suspend, rt711_dev_resume) + SET_RUNTIME_PM_OPS(rt711_dev_suspend, rt711_dev_resume, NULL) +}; + +static struct sdw_driver rt711_sdw_driver = { + .driver = { + .name = "rt711", + .owner = THIS_MODULE, + .pm = &rt711_pm, + }, + .probe = rt711_sdw_probe, + .remove = rt711_sdw_remove, + .ops = &rt711_slave_ops, + .id_table = rt711_id, +}; +module_sdw_driver(rt711_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT711 SDW driver"); +MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt711-sdw.h b/sound/soc/codecs/rt711-sdw.h new file mode 100644 index 000000000000..43b2b984b29c --- /dev/null +++ b/sound/soc/codecs/rt711-sdw.h @@ -0,0 +1,281 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * rt711-sdw.h -- RT711 ALSA SoC audio driver header + * + * Copyright(c) 2019 Realtek Semiconductor Corp. + */ + +#ifndef __RT711_SDW_H__ +#define __RT711_SDW_H__ + +static const struct reg_default rt711_reg_defaults[] = { + { 0x0000, 0x00 }, + { 0x0001, 0x00 }, + { 0x0002, 0x00 }, + { 0x0003, 0x00 }, + { 0x0004, 0x00 }, + { 0x0005, 0x01 }, + { 0x0020, 0x00 }, + { 0x0022, 0x00 }, + { 0x0023, 0x00 }, + { 0x0024, 0x00 }, + { 0x0025, 0x00 }, + { 0x0026, 0x00 }, + { 0x0030, 0x00 }, + { 0x0032, 0x00 }, + { 0x0033, 0x00 }, + { 0x0034, 0x00 }, + { 0x0035, 0x00 }, + { 0x0036, 0x00 }, + { 0x0040, 0x00 }, + { 0x0041, 0x00 }, + { 0x0042, 0x00 }, + { 0x0043, 0x00 }, + { 0x0044, 0x20 }, + { 0x0045, 0x01 }, + { 0x0046, 0x01 }, + { 0x0050, 0x20 }, + { 0x0051, 0x02 }, + { 0x0052, 0x5d }, + { 0x0053, 0x07 }, + { 0x0054, 0x11 }, + { 0x0055, 0x00 }, + { 0x0060, 0x00 }, + { 0x0070, 0x00 }, + { 0x0080, 0xc0 }, + { 0x0088, 0x00 }, + { 0x00e0, 0x00 }, + { 0x00e1, 0x00 }, + { 0x00e2, 0x00 }, + { 0x00e3, 0x00 }, + { 0x00e5, 0x00 }, + { 0x00ee, 0x00 }, + { 0x00ef, 0x00 }, + { 0x00f0, 0x00 }, + { 0x00f1, 0x00 }, + { 0x00f2, 0x00 }, + { 0x00f3, 0x00 }, + { 0x00f4, 0x00 }, + { 0x00f5, 0x00 }, + { 0x00fe, 0x00 }, + { 0x00ff, 0x00 }, + { 0x0100, 0x00 }, + { 0x0101, 0x00 }, + { 0x0102, 0x00 }, + { 0x0103, 0x00 }, + { 0x0104, 0x00 }, + { 0x0105, 0x00 }, + { 0x0120, 0x00 }, + { 0x0122, 0x00 }, + { 0x0123, 0x00 }, + { 0x0124, 0x00 }, + { 0x0125, 0x00 }, + { 0x0126, 0x00 }, + { 0x0127, 0x00 }, + { 0x0130, 0x00 }, + { 0x0132, 0x00 }, + { 0x0133, 0x00 }, + { 0x0134, 0x00 }, + { 0x0135, 0x00 }, + { 0x0136, 0x00 }, + { 0x0137, 0x00 }, + { 0x0200, 0x00 }, + { 0x0201, 0x00 }, + { 0x0202, 0x00 }, + { 0x0203, 0x00 }, + { 0x0204, 0x00 }, + { 0x0205, 0x03 }, + { 0x0220, 0x00 }, + { 0x0222, 0x00 }, + { 0x0223, 0x00 }, + { 0x0224, 0x00 }, + { 0x0225, 0x00 }, + { 0x0226, 0x00 }, + { 0x0227, 0x00 }, + { 0x0230, 0x00 }, + { 0x0232, 0x00 }, + { 0x0233, 0x00 }, + { 0x0234, 0x00 }, + { 0x0235, 0x00 }, + { 0x0236, 0x00 }, + { 0x0237, 0x00 }, + { 0x0300, 0x00 }, + { 0x0301, 0x00 }, + { 0x0302, 0x20 }, + { 0x0303, 0x00 }, + { 0x0304, 0x00 }, + { 0x0305, 0x03 }, + { 0x0320, 0x00 }, + { 0x0322, 0x00 }, + { 0x0323, 0x00 }, + { 0x0324, 0x00 }, + { 0x0325, 0x00 }, + { 0x0326, 0x00 }, + { 0x0327, 0x00 }, + { 0x0330, 0x00 }, + { 0x0332, 0x00 }, + { 0x0333, 0x00 }, + { 0x0334, 0x00 }, + { 0x0335, 0x00 }, + { 0x0336, 0x00 }, + { 0x0337, 0x00 }, + { 0x0400, 0x00 }, + { 0x0401, 0x00 }, + { 0x0402, 0x00 }, + { 0x0403, 0x00 }, + { 0x0404, 0x00 }, + { 0x0405, 0x03 }, + { 0x0420, 0x00 }, + { 0x0422, 0x00 }, + { 0x0423, 0x00 }, + { 0x0424, 0x00 }, + { 0x0425, 0x00 }, + { 0x0426, 0x00 }, + { 0x0427, 0x00 }, + { 0x0430, 0x00 }, + { 0x0432, 0x00 }, + { 0x0433, 0x00 }, + { 0x0434, 0x00 }, + { 0x0435, 0x00 }, + { 0x0436, 0x00 }, + { 0x0437, 0x00 }, + { 0x0f00, 0x00 }, + { 0x0f01, 0x00 }, + { 0x0f02, 0x20 }, + { 0x0f03, 0x00 }, + { 0x0f04, 0x00 }, + { 0x0f05, 0x03 }, + { 0x0f06, 0x00 }, + { 0x0f07, 0x00 }, + { 0x0f08, 0x00 }, + { 0x0f09, 0x00 }, + { 0x0f10, 0x00 }, + { 0x0f11, 0x00 }, + { 0x0f12, 0x00 }, + { 0x0f13, 0x00 }, + { 0x0f14, 0x00 }, + { 0x0f15, 0x00 }, + { 0x0f16, 0x00 }, + { 0x0f17, 0x00 }, + { 0x0f18, 0x00 }, + { 0x0f19, 0x00 }, + { 0x0f1a, 0x00 }, + { 0x0f1b, 0x00 }, + { 0x0f1c, 0x00 }, + { 0x0f1d, 0x00 }, + { 0x0f1e, 0x00 }, + { 0x0f1f, 0x00 }, + { 0x0f20, 0x00 }, + { 0x0f22, 0x00 }, + { 0x0f23, 0x00 }, + { 0x0f24, 0x00 }, + { 0x0f25, 0x00 }, + { 0x0f26, 0x00 }, + { 0x0f27, 0x00 }, + { 0x0f30, 0x00 }, + { 0x0f32, 0x00 }, + { 0x0f33, 0x00 }, + { 0x0f34, 0x00 }, + { 0x0f35, 0x00 }, + { 0x0f36, 0x00 }, + { 0x0f37, 0x00 }, + { 0x2012, 0x00 }, + { 0x2013, 0x00 }, + { 0x2014, 0x00 }, + { 0x2015, 0x00 }, + { 0x2016, 0x00 }, + { 0x201a, 0x00 }, + { 0x201b, 0x00 }, + { 0x201c, 0x0c }, + { 0x201d, 0x00 }, + { 0x201e, 0x00 }, + { 0x201f, 0x00 }, + { 0x2020, 0x00 }, + { 0x2021, 0x00 }, + { 0x2022, 0x00 }, + { 0x2023, 0x00 }, + { 0x2024, 0x00 }, + { 0x2025, 0x01 }, + { 0x2026, 0x00 }, + { 0x2027, 0x00 }, + { 0x2029, 0x00 }, + { 0x202a, 0x00 }, + { 0x202d, 0x00 }, + { 0x202e, 0x00 }, + { 0x202f, 0x00 }, + { 0x2030, 0x00 }, + { 0x2031, 0x00 }, + { 0x2032, 0x00 }, + { 0x2033, 0x00 }, + { 0x2034, 0x00 }, + { 0x2201, 0xc7 }, + { 0x2202, 0x0c }, + { 0x2203, 0x22 }, + { 0x2204, 0x04 }, + { 0x2206, 0x00 }, + { 0x2207, 0x00 }, + { 0x2208, 0x00 }, + { 0x2209, 0x00 }, + { 0x220a, 0x00 }, + { 0x220b, 0x00 }, + { 0x220c, 0x00 }, + { 0x220d, 0x04 }, + { 0x220e, 0x00 }, + { 0x220f, 0x00 }, + { 0x2211, 0x01 }, + { 0x2212, 0x00 }, + { 0x2220, 0x00 }, + { 0x2221, 0x00 }, + { 0x2222, 0x00 }, + { 0x2223, 0x00 }, + { 0x2230, 0x00 }, + { 0x2231, 0x2f }, + { 0x2232, 0x80 }, + { 0x2233, 0x00 }, + { 0x2234, 0x00 }, + { 0x2235, 0x00 }, + { 0x2236, 0x00 }, + { 0x2237, 0x00 }, + { 0x2238, 0x00 }, + { 0x2239, 0x00 }, + { 0x2f01, 0x00 }, + { 0x2f02, 0x09 }, + { 0x2f03, 0x00 }, + { 0x2f04, 0x00 }, + { 0x2f05, 0x0b }, + { 0x2f06, 0x01 }, + { 0x2f07, 0xcf }, + { 0x2f08, 0x00 }, + { 0x2f09, 0x00 }, + { 0x2f0a, 0x00 }, + { 0x2f0b, 0x00 }, + { 0x2f0c, 0x00 }, + { 0x2f0d, 0x00 }, + { 0x2f0e, 0x00 }, + { 0x2f0f, 0x00 }, + { 0x3122, 0x00 }, + { 0x3123, 0x00 }, + { 0x7303, 0x57 }, + { 0x8383, 0x57 }, + { 0x7308, 0x97 }, + { 0x8388, 0x97 }, + { 0x7309, 0x97 }, + { 0x8389, 0x97 }, + { 0x7312, 0x00 }, + { 0x8392, 0x00 }, + { 0x7313, 0x00 }, + { 0x8393, 0x00 }, + { 0x7319, 0x00 }, + { 0x8399, 0x00 }, + { 0x752009, 0x1029 }, + { 0x752011, 0x007a }, + { 0x75201a, 0x8003 }, + { 0x752045, 0x5289 }, + { 0x752048, 0xd049 }, + { 0x75204a, 0xa83b }, + { 0x75206b, 0x5064 }, + { 0x75206f, 0x058b }, + { 0x752091, 0x0000 }, +}; + +#endif /* __RT711_SDW_H__ */ diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c new file mode 100644 index 000000000000..3bebba7a63be --- /dev/null +++ b/sound/soc/codecs/rt711.c @@ -0,0 +1,1293 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rt711.c -- rt711 ALSA SoC audio driver +// +// Copyright(c) 2019 Realtek Semiconductor Corp. +// +// + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm_runtime.h> +#include <linux/pm.h> +#include <linux/soundwire/sdw.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/hda_verbs.h> +#include <sound/jack.h> + +#include "rt711.h" + +static int rt711_index_write(struct regmap *regmap, + unsigned int nid, unsigned int reg, unsigned int value) +{ + int ret; + unsigned int addr = ((RT711_PRIV_INDEX_W_H | nid) << 8) | reg; + + ret = regmap_write(regmap, addr, value); + if (ret < 0) + pr_err("Failed to set private value: %06x <= %04x ret=%d\n", + addr, value, ret); + + return ret; +} + +static int rt711_index_read(struct regmap *regmap, + unsigned int nid, unsigned int reg, unsigned int *value) +{ + int ret; + unsigned int addr = ((RT711_PRIV_INDEX_W_H | nid) << 8) | reg; + + *value = 0; + ret = regmap_read(regmap, addr, value); + if (ret < 0) + pr_err("Failed to get private value: %06x => %04x ret=%d\n", + addr, *value, ret); + + return ret; +} + +static int rt711_index_update_bits(struct regmap *regmap, unsigned int nid, + unsigned int reg, unsigned int mask, unsigned int val) +{ + unsigned int tmp, orig; + int ret; + + ret = rt711_index_read(regmap, nid, reg, &orig); + if (ret < 0) + return ret; + + tmp = orig & ~mask; + tmp |= val & mask; + + return rt711_index_write(regmap, nid, reg, tmp); +} + +static void rt711_reset(struct regmap *regmap) +{ + regmap_write(regmap, RT711_FUNC_RESET, 0); + rt711_index_update_bits(regmap, RT711_VENDOR_REG, + RT711_PARA_VERB_CTL, RT711_HIDDEN_REG_SW_RESET, + RT711_HIDDEN_REG_SW_RESET); +} + +static int rt711_calibration(struct rt711_priv *rt711) +{ + unsigned int val, loop = 0; + struct device *dev; + struct regmap *regmap = rt711->regmap; + int ret = 0; + + mutex_lock(&rt711->calibrate_mutex); + regmap_write(rt711->regmap, + RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + dev = regmap_get_device(regmap); + + /* Calibration manual mode */ + rt711_index_update_bits(regmap, RT711_VENDOR_REG, RT711_FSM_CTL, + 0xf, 0x0); + + /* trigger */ + rt711_index_update_bits(regmap, RT711_VENDOR_CALI, + RT711_DAC_DC_CALI_CTL1, RT711_DAC_DC_CALI_TRIGGER, + RT711_DAC_DC_CALI_TRIGGER); + + /* wait for calibration process */ + rt711_index_read(regmap, RT711_VENDOR_CALI, + RT711_DAC_DC_CALI_CTL1, &val); + + while (val & RT711_DAC_DC_CALI_TRIGGER) { + if (loop >= 500) { + pr_err("%s, calibration time-out!\n", + __func__); + ret = -ETIMEDOUT; + break; + } + loop++; + + usleep_range(10000, 11000); + rt711_index_read(regmap, RT711_VENDOR_CALI, + RT711_DAC_DC_CALI_CTL1, &val); + } + + /* depop mode */ + rt711_index_update_bits(regmap, RT711_VENDOR_REG, + RT711_FSM_CTL, 0xf, RT711_DEPOP_CTL); + + regmap_write(rt711->regmap, + RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + mutex_unlock(&rt711->calibrate_mutex); + + dev_dbg(dev, "%s calibration complete, ret=%d\n", __func__, ret); + return ret; +} + +static unsigned int rt711_button_detect(struct rt711_priv *rt711) +{ + unsigned int btn_type = 0, val80, val81; + int ret; + + ret = rt711_index_read(rt711->regmap, RT711_VENDOR_REG, + RT711_IRQ_FLAG_TABLE1, &val80); + if (ret < 0) + goto read_error; + ret = rt711_index_read(rt711->regmap, RT711_VENDOR_REG, + RT711_IRQ_FLAG_TABLE2, &val81); + if (ret < 0) + goto read_error; + + val80 &= 0x0381; + val81 &= 0xff00; + + switch (val80) { + case 0x0200: + case 0x0100: + case 0x0080: + btn_type |= SND_JACK_BTN_0; + break; + case 0x0001: + btn_type |= SND_JACK_BTN_3; + break; + } + switch (val81) { + case 0x8000: + case 0x4000: + case 0x2000: + btn_type |= SND_JACK_BTN_1; + break; + case 0x1000: + case 0x0800: + case 0x0400: + btn_type |= SND_JACK_BTN_2; + break; + case 0x0200: + case 0x0100: + btn_type |= SND_JACK_BTN_3; + break; + } +read_error: + return btn_type; +} + +static int rt711_headset_detect(struct rt711_priv *rt711) +{ + unsigned int buf, loop = 0; + int ret; + unsigned int jack_status = 0, reg; + + ret = rt711_index_read(rt711->regmap, RT711_VENDOR_REG, + RT711_COMBO_JACK_AUTO_CTL2, &buf); + if (ret < 0) + goto io_error; + + while (loop < 500 && + (buf & RT711_COMBOJACK_AUTO_DET_STATUS) == 0) { + loop++; + + usleep_range(9000, 10000); + ret = rt711_index_read(rt711->regmap, RT711_VENDOR_REG, + RT711_COMBO_JACK_AUTO_CTL2, &buf); + if (ret < 0) + goto io_error; + + reg = RT711_VERB_GET_PIN_SENSE | RT711_HP_OUT; + ret = regmap_read(rt711->regmap, reg, &jack_status); + if (ret < 0) + goto io_error; + if ((jack_status & (1 << 31)) == 0) + goto remove_error; + } + + if (loop >= 500) + goto to_error; + + if (buf & RT711_COMBOJACK_AUTO_DET_TRS) + rt711->jack_type = SND_JACK_HEADPHONE; + else if ((buf & RT711_COMBOJACK_AUTO_DET_CTIA) || + (buf & RT711_COMBOJACK_AUTO_DET_OMTP)) + rt711->jack_type = SND_JACK_HEADSET; + + return 0; + +to_error: + ret = -ETIMEDOUT; + pr_err_ratelimited("Time-out error in %s\n", __func__); + return ret; +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); + return ret; +remove_error: + pr_err_ratelimited("Jack removal in %s\n", __func__); + return -ENODEV; +} + +static void rt711_jack_detect_handler(struct work_struct *work) +{ + struct rt711_priv *rt711 = + container_of(work, struct rt711_priv, jack_detect_work.work); + int btn_type = 0, ret; + unsigned int jack_status = 0, reg; + + if (!rt711->hs_jack) + return; + + if (!rt711->component->card->instantiated) + return; + + reg = RT711_VERB_GET_PIN_SENSE | RT711_HP_OUT; + ret = regmap_read(rt711->regmap, reg, &jack_status); + if (ret < 0) + goto io_error; + + /* pin attached */ + if (jack_status & (1 << 31)) { + /* jack in */ + if (rt711->jack_type == 0) { + ret = rt711_headset_detect(rt711); + if (ret < 0) + return; + if (rt711->jack_type == SND_JACK_HEADSET) + btn_type = rt711_button_detect(rt711); + } else if (rt711->jack_type == SND_JACK_HEADSET) { + /* jack is already in, report button event */ + btn_type = rt711_button_detect(rt711); + } + } else { + /* jack out */ + rt711->jack_type = 0; + } + + dev_dbg(&rt711->slave->dev, + "in %s, jack_type=0x%x\n", __func__, rt711->jack_type); + dev_dbg(&rt711->slave->dev, + "in %s, btn_type=0x%x\n", __func__, btn_type); + + snd_soc_jack_report(rt711->hs_jack, rt711->jack_type | btn_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (btn_type) { + /* button released */ + snd_soc_jack_report(rt711->hs_jack, rt711->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + mod_delayed_work(system_power_efficient_wq, + &rt711->jack_btn_check_work, msecs_to_jiffies(200)); + } + + return; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); +} + +static void rt711_btn_check_handler(struct work_struct *work) +{ + struct rt711_priv *rt711 = container_of(work, struct rt711_priv, + jack_btn_check_work.work); + int btn_type = 0, ret; + unsigned int jack_status = 0, reg; + + reg = RT711_VERB_GET_PIN_SENSE | RT711_HP_OUT; + ret = regmap_read(rt711->regmap, reg, &jack_status); + if (ret < 0) + goto io_error; + + /* pin attached */ + if (jack_status & (1 << 31)) { + if (rt711->jack_type == SND_JACK_HEADSET) { + /* jack is already in, report button event */ + btn_type = rt711_button_detect(rt711); + } + } else { + rt711->jack_type = 0; + } + + /* cbj comparator */ + ret = rt711_index_read(rt711->regmap, RT711_VENDOR_REG, + RT711_COMBO_JACK_AUTO_CTL2, &reg); + if (ret < 0) + goto io_error; + + if ((reg & 0xf0) == 0xf0) + btn_type = 0; + + dev_dbg(&rt711->slave->dev, + "%s, btn_type=0x%x\n", __func__, btn_type); + snd_soc_jack_report(rt711->hs_jack, rt711->jack_type | btn_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (btn_type) { + /* button released */ + snd_soc_jack_report(rt711->hs_jack, rt711->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + mod_delayed_work(system_power_efficient_wq, + &rt711->jack_btn_check_work, msecs_to_jiffies(200)); + } + + return; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); +} + +static void rt711_jack_init(struct rt711_priv *rt711) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(rt711->component); + + mutex_lock(&rt711->calibrate_mutex); + /* power on */ + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt711->regmap, + RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + if (rt711->hs_jack) { + /* unsolicited response & IRQ control */ + regmap_write(rt711->regmap, + RT711_SET_MIC2_UNSOLICITED_ENABLE, 0x82); + regmap_write(rt711->regmap, + RT711_SET_HP_UNSOLICITED_ENABLE, 0x81); + regmap_write(rt711->regmap, + RT711_SET_INLINE_UNSOLICITED_ENABLE, 0x83); + rt711_index_write(rt711->regmap, RT711_VENDOR_REG, + 0x10, 0x2420); + rt711_index_write(rt711->regmap, RT711_VENDOR_REG, + 0x19, 0x2e11); + + switch (rt711->jd_src) { + case RT711_JD1: + /* default settings was already for JD1 */ + break; + case RT711_JD2: + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_JD_CTL2, RT711_JD2_2PORT_200K_DECODE_HP | + RT711_HP_JD_SEL_JD2, + RT711_JD2_2PORT_200K_DECODE_HP | + RT711_HP_JD_SEL_JD2); + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_CC_DET1, + RT711_HP_JD_FINAL_RESULT_CTL_JD12, + RT711_HP_JD_FINAL_RESULT_CTL_JD12); + break; + default: + dev_warn(rt711->component->dev, "Wrong JD source\n"); + break; + } + + dev_dbg(&rt711->slave->dev, "in %s enable\n", __func__); + + mod_delayed_work(system_power_efficient_wq, + &rt711->jack_detect_work, msecs_to_jiffies(250)); + } else { + regmap_write(rt711->regmap, + RT711_SET_MIC2_UNSOLICITED_ENABLE, 0x00); + regmap_write(rt711->regmap, + RT711_SET_HP_UNSOLICITED_ENABLE, 0x00); + regmap_write(rt711->regmap, + RT711_SET_INLINE_UNSOLICITED_ENABLE, 0x00); + + dev_dbg(&rt711->slave->dev, "in %s disable\n", __func__); + } + + /* power off */ + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt711->regmap, + RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + mutex_unlock(&rt711->calibrate_mutex); +} + +static int rt711_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + + rt711->hs_jack = hs_jack; + + if (!rt711->hw_init) { + dev_dbg(&rt711->slave->dev, + "%s hw_init not ready yet\n", __func__); + return 0; + } + + rt711_jack_init(rt711); + + return 0; +} + +static void rt711_get_gain(struct rt711_priv *rt711, unsigned int addr_h, + unsigned int addr_l, unsigned int val_h, + unsigned int *r_val, unsigned int *l_val) +{ + /* R Channel */ + *r_val = (val_h << 8); + regmap_read(rt711->regmap, addr_l, r_val); + + /* L Channel */ + val_h |= 0x20; + *l_val = (val_h << 8); + regmap_read(rt711->regmap, addr_h, l_val); +} + +/* For Verb-Set Amplifier Gain (Verb ID = 3h) */ +static int rt711_set_amp_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned int addr_h, addr_l, val_h, val_ll, val_lr; + unsigned int read_ll, read_rl; + int i; + + /* Can't use update bit function, so read the original value first */ + addr_h = mc->reg; + addr_l = mc->rreg; + if (mc->shift == RT711_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt711_get_gain(rt711, addr_h, addr_l, val_h, &read_rl, &read_ll); + + /* L Channel */ + if (mc->invert) { + /* for mute/unmute */ + val_ll = (mc->max - ucontrol->value.integer.value[0]) + << RT711_MUTE_SFT; + /* keep gain */ + read_ll = read_ll & 0x7f; + val_ll |= read_ll; + } else { + /* for gain */ + val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); + if (val_ll > mc->max) + val_ll = mc->max; + /* keep mute status */ + read_ll = read_ll & (1 << RT711_MUTE_SFT); + val_ll |= read_ll; + } + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt711->regmap, + RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* R Channel */ + if (mc->invert) { + /* for mute/unmute */ + val_lr = (mc->max - ucontrol->value.integer.value[1]) + << RT711_MUTE_SFT; + /* keep gain */ + read_rl = read_rl & 0x7f; + val_lr |= read_rl; + } else { + /* for gain */ + val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); + if (val_lr > mc->max) + val_lr = mc->max; + /* keep mute status */ + read_rl = read_rl & (1 << RT711_MUTE_SFT); + val_lr |= read_rl; + } + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << mc->shift) | (3 << 4); + regmap_write(rt711->regmap, + addr_h, (val_h << 8 | val_ll)); + regmap_write(rt711->regmap, + addr_l, (val_h << 8 | val_ll)); + } else { + /* Lch*/ + val_h = (1 << mc->shift) | (1 << 5); + regmap_write(rt711->regmap, + addr_h, (val_h << 8 | val_ll)); + + /* Rch */ + val_h = (1 << mc->shift) | (1 << 4); + regmap_write(rt711->regmap, + addr_l, (val_h << 8 | val_lr)); + } + /* check result */ + if (mc->shift == RT711_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt711_get_gain(rt711, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt711->regmap, + RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + return 0; +} + +static int rt711_set_amp_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int addr_h, addr_l, val_h; + unsigned int read_ll, read_rl; + + /* switch to get command */ + addr_h = mc->reg; + addr_l = mc->rreg; + if (mc->shift == RT711_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt711_get_gain(rt711, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (mc->invert) { + /* mute/unmute for switch controls */ + read_ll = !((read_ll & 0x80) >> RT711_MUTE_SFT); + read_rl = !((read_rl & 0x80) >> RT711_MUTE_SFT); + } else { + /* for gain volume controls */ + read_ll = read_ll & 0x7f; + read_rl = read_rl & 0x7f; + } + ucontrol->value.integer.value[0] = read_ll; + ucontrol->value.integer.value[1] = read_rl; + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt711_snd_controls[] = { + SOC_DOUBLE_R_EXT_TLV("DAC Surr Playback Volume", + RT711_SET_GAIN_DAC2_H, RT711_SET_GAIN_DAC2_L, + RT711_DIR_OUT_SFT, 0x57, 0, + rt711_set_amp_gain_get, rt711_set_amp_gain_put, out_vol_tlv), + SOC_DOUBLE_R_EXT("ADC 08 Capture Switch", + RT711_SET_GAIN_ADC2_H, RT711_SET_GAIN_ADC2_L, + RT711_DIR_IN_SFT, 1, 1, + rt711_set_amp_gain_get, rt711_set_amp_gain_put), + SOC_DOUBLE_R_EXT("ADC 09 Capture Switch", + RT711_SET_GAIN_ADC1_H, RT711_SET_GAIN_ADC1_L, + RT711_DIR_IN_SFT, 1, 1, + rt711_set_amp_gain_get, rt711_set_amp_gain_put), + SOC_DOUBLE_R_EXT_TLV("ADC 08 Capture Volume", + RT711_SET_GAIN_ADC2_H, RT711_SET_GAIN_ADC2_L, + RT711_DIR_IN_SFT, 0x3f, 0, + rt711_set_amp_gain_get, rt711_set_amp_gain_put, in_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("ADC 09 Capture Volume", + RT711_SET_GAIN_ADC1_H, RT711_SET_GAIN_ADC1_L, + RT711_DIR_IN_SFT, 0x3f, 0, + rt711_set_amp_gain_get, rt711_set_amp_gain_put, in_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("AMIC Volume", + RT711_SET_GAIN_AMIC_H, RT711_SET_GAIN_AMIC_L, + RT711_DIR_IN_SFT, 3, 0, + rt711_set_amp_gain_get, rt711_set_amp_gain_put, mic_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("DMIC1 Volume", + RT711_SET_GAIN_DMIC1_H, RT711_SET_GAIN_DMIC1_L, + RT711_DIR_IN_SFT, 3, 0, + rt711_set_amp_gain_get, rt711_set_amp_gain_put, mic_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("DMIC2 Volume", + RT711_SET_GAIN_DMIC2_H, RT711_SET_GAIN_DMIC2_L, + RT711_DIR_IN_SFT, 3, 0, + rt711_set_amp_gain_get, rt711_set_amp_gain_put, mic_vol_tlv), +}; + +static int rt711_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned int reg, val = 0, nid; + int ret; + + if (strstr(ucontrol->id.name, "ADC 22 Mux")) + nid = RT711_MIXER_IN1; + else if (strstr(ucontrol->id.name, "ADC 23 Mux")) + nid = RT711_MIXER_IN2; + else + return -EINVAL; + + /* vid = 0xf01 */ + reg = RT711_VERB_SET_CONNECT_SEL | nid; + ret = regmap_read(rt711->regmap, reg, &val); + if (ret < 0) { + dev_err(component->dev, "%s: sdw read failed: %d\n", + __func__, ret); + return ret; + } + + ucontrol->value.enumerated.item[0] = val; + + return 0; +} + +static int rt711_mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int val, val2 = 0, change, reg, nid; + int ret; + + if (item[0] >= e->items) + return -EINVAL; + + if (strstr(ucontrol->id.name, "ADC 22 Mux")) + nid = RT711_MIXER_IN1; + else if (strstr(ucontrol->id.name, "ADC 23 Mux")) + nid = RT711_MIXER_IN2; + else + return -EINVAL; + + /* Verb ID = 0x701h */ + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + + reg = RT711_VERB_SET_CONNECT_SEL | nid; + ret = regmap_read(rt711->regmap, reg, &val2); + if (ret < 0) { + dev_err(component->dev, "%s: sdw read failed: %d\n", + __func__, ret); + return ret; + } + + if (val == val2) + change = 0; + else + change = 1; + + if (change) { + reg = RT711_VERB_SET_CONNECT_SEL | nid; + regmap_write(rt711->regmap, reg, val); + } + + snd_soc_dapm_mux_update_power(dapm, kcontrol, + item[0], e, NULL); + + return change; +} + +static const char * const adc_mux_text[] = { + "MIC2", + "LINE1", + "LINE2", + "DMIC", +}; + +static SOC_ENUM_SINGLE_DECL( + rt711_adc22_enum, SND_SOC_NOPM, 0, adc_mux_text); + +static SOC_ENUM_SINGLE_DECL( + rt711_adc23_enum, SND_SOC_NOPM, 0, adc_mux_text); + +static const struct snd_kcontrol_new rt711_adc22_mux = + SOC_DAPM_ENUM_EXT("ADC 22 Mux", rt711_adc22_enum, + rt711_mux_get, rt711_mux_put); + +static const struct snd_kcontrol_new rt711_adc23_mux = + SOC_DAPM_ENUM_EXT("ADC 23 Mux", rt711_adc23_enum, + rt711_mux_get, rt711_mux_put); + +static int rt711_dac_surround_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 rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned int val_h = (1 << RT711_DIR_OUT_SFT) | (0x3 << 4); + unsigned int val_l; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt711->regmap, + RT711_SET_STREAMID_DAC2, 0x10); + + val_l = 0x00; + regmap_write(rt711->regmap, + RT711_SET_GAIN_HP_H, (val_h << 8 | val_l)); + break; + case SND_SOC_DAPM_PRE_PMD: + val_l = (1 << RT711_MUTE_SFT); + regmap_write(rt711->regmap, + RT711_SET_GAIN_HP_H, (val_h << 8 | val_l)); + usleep_range(50000, 55000); + + regmap_write(rt711->regmap, + RT711_SET_STREAMID_DAC2, 0x00); + break; + } + return 0; +} + +static int rt711_adc_09_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 rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt711->regmap, + RT711_SET_STREAMID_ADC1, 0x10); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt711->regmap, + RT711_SET_STREAMID_ADC1, 0x00); + break; + } + return 0; +} + +static int rt711_adc_08_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 rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt711->regmap, + RT711_SET_STREAMID_ADC2, 0x10); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt711->regmap, + RT711_SET_STREAMID_ADC2, 0x00); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget rt711_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("HP"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("LINE2"), + + SND_SOC_DAPM_DAC_E("DAC Surround", NULL, SND_SOC_NOPM, 0, 0, + rt711_dac_surround_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("ADC 09", NULL, SND_SOC_NOPM, 0, 0, + rt711_adc_09_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("ADC 08", NULL, SND_SOC_NOPM, 0, 0, + rt711_adc_08_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MUX("ADC 22 Mux", SND_SOC_NOPM, 0, 0, + &rt711_adc22_mux), + SND_SOC_DAPM_MUX("ADC 23 Mux", SND_SOC_NOPM, 0, 0, + &rt711_adc23_mux), + + SND_SOC_DAPM_AIF_IN("DP3RX", "DP3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP2TX", "DP2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP4TX", "DP4 Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route rt711_audio_map[] = { + {"DAC Surround", NULL, "DP3RX"}, + {"DP2TX", NULL, "ADC 09"}, + {"DP4TX", NULL, "ADC 08"}, + + {"ADC 09", NULL, "ADC 22 Mux"}, + {"ADC 08", NULL, "ADC 23 Mux"}, + {"ADC 22 Mux", "DMIC", "DMIC1"}, + {"ADC 22 Mux", "LINE1", "LINE1"}, + {"ADC 22 Mux", "LINE2", "LINE2"}, + {"ADC 22 Mux", "MIC2", "MIC2"}, + {"ADC 23 Mux", "DMIC", "DMIC2"}, + {"ADC 23 Mux", "LINE1", "LINE1"}, + {"ADC 23 Mux", "LINE2", "LINE2"}, + {"ADC 23 Mux", "MIC2", "MIC2"}, + + {"HP", NULL, "DAC Surround"}, +}; + +static int rt711_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + regmap_write(rt711->regmap, + RT711_SET_AUDIO_POWER_STATE, + AC_PWRST_D0); + } + break; + + case SND_SOC_BIAS_STANDBY: + regmap_write(rt711->regmap, + RT711_SET_AUDIO_POWER_STATE, + AC_PWRST_D3); + break; + + default: + break; + } + + return 0; +} + +static int rt711_parse_dt(struct rt711_priv *rt711, struct device *dev) +{ + device_property_read_u32(dev, "realtek,jd-src", + &rt711->jd_src); + + return 0; +} + +static int rt711_probe(struct snd_soc_component *component) +{ + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + + rt711_parse_dt(rt711, &rt711->slave->dev); + rt711->component = component; + + return 0; +} + +static const struct snd_soc_component_driver soc_codec_dev_rt711 = { + .probe = rt711_probe, + .set_bias_level = rt711_set_bias_level, + .controls = rt711_snd_controls, + .num_controls = ARRAY_SIZE(rt711_snd_controls), + .dapm_widgets = rt711_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt711_dapm_widgets), + .dapm_routes = rt711_audio_map, + .num_dapm_routes = ARRAY_SIZE(rt711_audio_map), + .set_jack = rt711_set_jack_detect, +}; + +static int rt711_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + struct sdw_stream_data *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = (struct sdw_stream_runtime *)sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void rt711_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int rt711_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int retval, port, num_channels; + unsigned int val = 0; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!rt711->slave) + return -EINVAL; + + /* SoundWire specific configuration */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port = 3; + } else { + direction = SDW_DATA_DIR_TX; + if (dai->id == RT711_AIF1) + port = 4; + else if (dai->id == RT711_AIF2) + port = 2; + else + return -EINVAL; + } + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = (1 << (num_channels)) - 1; + port_config.num = port; + + retval = sdw_stream_add_slave(rt711->slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(component->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 8: + break; + case 16: + val |= (0x1 << 4); + break; + case 20: + val |= (0x2 << 4); + break; + case 24: + val |= (0x3 << 4); + break; + case 32: + val |= (0x4 << 4); + break; + default: + return -EINVAL; + } + + /* 48Khz */ + regmap_write(rt711->regmap, RT711_DAC_FORMAT_H, val); + regmap_write(rt711->regmap, RT711_ADC1_FORMAT_H, val); + regmap_write(rt711->regmap, RT711_ADC2_FORMAT_H, val); + + return retval; +} + +static int rt711_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt711->slave) + return -EINVAL; + + sdw_stream_remove_slave(rt711->slave, stream->sdw_stream); + return 0; +} + +#define RT711_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static struct snd_soc_dai_ops rt711_ops = { + .hw_params = rt711_pcm_hw_params, + .hw_free = rt711_pcm_hw_free, + .set_sdw_stream = rt711_set_sdw_stream, + .shutdown = rt711_shutdown, +}; + +static struct snd_soc_dai_driver rt711_dai[] = { + { + .name = "rt711-aif1", + .id = RT711_AIF1, + .playback = { + .stream_name = "DP3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT711_STEREO_RATES, + .formats = RT711_FORMATS, + }, + .capture = { + .stream_name = "DP4 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT711_STEREO_RATES, + .formats = RT711_FORMATS, + }, + .ops = &rt711_ops, + }, + { + .name = "rt711-aif2", + .id = RT711_AIF2, + .capture = { + .stream_name = "DP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT711_STEREO_RATES, + .formats = RT711_FORMATS, + }, + .ops = &rt711_ops, + } +}; + +/* Bus clock frequency */ +#define RT711_CLK_FREQ_9600000HZ 9600000 +#define RT711_CLK_FREQ_12000000HZ 12000000 +#define RT711_CLK_FREQ_6000000HZ 6000000 +#define RT711_CLK_FREQ_4800000HZ 4800000 +#define RT711_CLK_FREQ_2400000HZ 2400000 +#define RT711_CLK_FREQ_12288000HZ 12288000 + +int rt711_clock_config(struct device *dev) +{ + struct rt711_priv *rt711 = dev_get_drvdata(dev); + unsigned int clk_freq, value; + + clk_freq = (rt711->params.curr_dr_freq >> 1); + + switch (clk_freq) { + case RT711_CLK_FREQ_12000000HZ: + value = 0x0; + break; + case RT711_CLK_FREQ_6000000HZ: + value = 0x1; + break; + case RT711_CLK_FREQ_9600000HZ: + value = 0x2; + break; + case RT711_CLK_FREQ_4800000HZ: + value = 0x3; + break; + case RT711_CLK_FREQ_2400000HZ: + value = 0x4; + break; + case RT711_CLK_FREQ_12288000HZ: + value = 0x5; + break; + default: + return -EINVAL; + } + + regmap_write(rt711->regmap, 0xe0, value); + regmap_write(rt711->regmap, 0xf0, value); + + dev_dbg(dev, "%s complete, clk_freq=%d\n", __func__, clk_freq); + + return 0; +} + +static void rt711_calibration_work(struct work_struct *work) +{ + struct rt711_priv *rt711 = + container_of(work, struct rt711_priv, calibration_work); + + rt711_calibration(rt711); +} + +int rt711_init(struct device *dev, struct regmap *sdw_regmap, + struct regmap *regmap, struct sdw_slave *slave) +{ + struct rt711_priv *rt711; + int ret; + + rt711 = devm_kzalloc(dev, sizeof(*rt711), GFP_KERNEL); + if (!rt711) + return -ENOMEM; + + dev_set_drvdata(dev, rt711); + rt711->slave = slave; + rt711->sdw_regmap = sdw_regmap; + rt711->regmap = regmap; + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt711->hw_init = false; + rt711->first_hw_init = false; + + /* JD source uses JD2 in default */ + rt711->jd_src = RT711_JD2; + + ret = devm_snd_soc_register_component(dev, + &soc_codec_dev_rt711, + rt711_dai, + ARRAY_SIZE(rt711_dai)); + + dev_dbg(&slave->dev, "%s\n", __func__); + + return ret; +} + +int rt711_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt711_priv *rt711 = dev_get_drvdata(dev); + + if (rt711->hw_init) + return 0; + + if (rt711->first_hw_init) { + regcache_cache_only(rt711->regmap, false); + regcache_cache_bypass(rt711->regmap, true); + } + + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + if (!rt711->first_hw_init) { + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + rt711_reset(rt711->regmap); + + /* power on */ + regmap_write(rt711->regmap, RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* Set Pin Widget */ + regmap_write(rt711->regmap, RT711_SET_PIN_MIC2, 0x25); + regmap_write(rt711->regmap, RT711_SET_PIN_HP, 0xc0); + regmap_write(rt711->regmap, RT711_SET_PIN_DMIC1, 0x20); + regmap_write(rt711->regmap, RT711_SET_PIN_DMIC2, 0x20); + regmap_write(rt711->regmap, RT711_SET_PIN_LINE1, 0x20); + regmap_write(rt711->regmap, RT711_SET_PIN_LINE2, 0x20); + + /* Mute HP/ADC1/ADC2 */ + regmap_write(rt711->regmap, RT711_SET_GAIN_HP_H, 0xa080); + regmap_write(rt711->regmap, RT711_SET_GAIN_HP_H, 0x9080); + regmap_write(rt711->regmap, RT711_SET_GAIN_ADC2_H, 0x6080); + regmap_write(rt711->regmap, RT711_SET_GAIN_ADC2_H, 0x5080); + regmap_write(rt711->regmap, RT711_SET_GAIN_ADC1_H, 0x6080); + regmap_write(rt711->regmap, RT711_SET_GAIN_ADC1_H, 0x5080); + + /* Set Configuration Default */ + regmap_write(rt711->regmap, 0x4f12, 0x91); + regmap_write(rt711->regmap, 0x4e12, 0xd6); + regmap_write(rt711->regmap, 0x4d12, 0x11); + regmap_write(rt711->regmap, 0x4c12, 0x20); + regmap_write(rt711->regmap, 0x4f13, 0x91); + regmap_write(rt711->regmap, 0x4e13, 0xd6); + regmap_write(rt711->regmap, 0x4d13, 0x11); + regmap_write(rt711->regmap, 0x4c13, 0x21); + regmap_write(rt711->regmap, 0x4c21, 0xf0); + regmap_write(rt711->regmap, 0x4d21, 0x11); + regmap_write(rt711->regmap, 0x4e21, 0x11); + regmap_write(rt711->regmap, 0x4f21, 0x01); + + /* Data port arrangement */ + rt711_index_write(rt711->regmap, RT711_VENDOR_REG, + RT711_TX_RX_MUX_CTL, 0x0154); + + /* Set index */ + rt711_index_write(rt711->regmap, RT711_VENDOR_REG, + RT711_DIGITAL_MISC_CTRL4, 0x201b); + rt711_index_write(rt711->regmap, RT711_VENDOR_REG, + RT711_COMBO_JACK_AUTO_CTL1, 0x5089); + rt711_index_write(rt711->regmap, RT711_VENDOR_REG, + RT711_VREFOUT_CTL, 0x5064); + rt711_index_write(rt711->regmap, RT711_VENDOR_REG, + RT711_INLINE_CMD_CTL, 0xd249); + + /* Finish Initial Settings, set power to D3 */ + regmap_write(rt711->regmap, RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + + if (rt711->first_hw_init) + rt711_calibration(rt711); + else { + INIT_DELAYED_WORK(&rt711->jack_detect_work, + rt711_jack_detect_handler); + INIT_DELAYED_WORK(&rt711->jack_btn_check_work, + rt711_btn_check_handler); + mutex_init(&rt711->calibrate_mutex); + INIT_WORK(&rt711->calibration_work, rt711_calibration_work); + schedule_work(&rt711->calibration_work); + } + + /* + * if set_jack callback occurred early than io_init, + * we set up the jack detection function now + */ + if (rt711->hs_jack) + rt711_jack_init(rt711); + + if (rt711->first_hw_init) { + regcache_cache_bypass(rt711->regmap, false); + regcache_mark_dirty(rt711->regmap); + } else + rt711->first_hw_init = true; + + /* Mark Slave initialization complete */ + rt711->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +MODULE_DESCRIPTION("ASoC RT711 SDW driver"); +MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt711.h b/sound/soc/codecs/rt711.h new file mode 100644 index 000000000000..ca0f581feec7 --- /dev/null +++ b/sound/soc/codecs/rt711.h @@ -0,0 +1,227 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * rt711.h -- RT711 ALSA SoC audio driver header + * + * Copyright(c) 2019 Realtek Semiconductor Corp. + */ + +#ifndef __RT711_H__ +#define __RT711_H__ + +extern const struct dev_pm_ops rt711_runtime_pm; + +struct rt711_priv { + struct regmap *regmap; + struct regmap *sdw_regmap; + struct snd_soc_component *component; + struct sdw_slave *slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; + struct snd_soc_jack *hs_jack; + struct delayed_work jack_detect_work; + struct delayed_work jack_btn_check_work; + struct work_struct calibration_work; + struct mutex calibrate_mutex; /* for headset calibration */ + int jack_type, jd_src; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +/* NID */ +#define RT711_AUDIO_FUNCTION_GROUP 0x01 +#define RT711_DAC_OUT2 0x03 +#define RT711_ADC_IN1 0x09 +#define RT711_ADC_IN2 0x08 +#define RT711_DMIC1 0x12 +#define RT711_DMIC2 0x13 +#define RT711_MIC2 0x19 +#define RT711_LINE1 0x1a +#define RT711_LINE2 0x1b +#define RT711_BEEP 0x1d +#define RT711_VENDOR_REG 0x20 +#define RT711_HP_OUT 0x21 +#define RT711_MIXER_IN1 0x22 +#define RT711_MIXER_IN2 0x23 +#define RT711_INLINE_CMD 0x55 +#define RT711_VENDOR_CALI 0x58 +#define RT711_VENDOR_IMS_DRE 0x5b + +/* Index (NID:20h) */ +#define RT711_DAC_DC_CALI_CTL1 0x00 +#define RT711_JD_CTL2 0x09 +#define RT711_CC_DET1 0x11 +#define RT711_PARA_VERB_CTL 0x1a +#define RT711_COMBO_JACK_AUTO_CTL1 0x45 +#define RT711_COMBO_JACK_AUTO_CTL2 0x46 +#define RT711_INLINE_CMD_CTL 0x48 +#define RT711_DIGITAL_MISC_CTRL4 0x4a +#define RT711_VREFOUT_CTL 0x6b +#define RT711_FSM_CTL 0x6f +#define RT711_IRQ_FLAG_TABLE1 0x80 +#define RT711_IRQ_FLAG_TABLE2 0x81 +#define RT711_IRQ_FLAG_TABLE3 0x82 +#define RT711_TX_RX_MUX_CTL 0x91 + +/* Index (NID:5bh) */ +#define RT711_IMS_DIGITAL_CTL1 0x00 +#define RT711_HP_IMS_RESULT_L 0x20 +#define RT711_HP_IMS_RESULT_R 0x21 + +/* Verb */ +#define RT711_VERB_SET_CONNECT_SEL 0x3100 +#define RT711_VERB_SET_EAPD_BTLENABLE 0x3c00 +#define RT711_VERB_GET_CONNECT_SEL 0xb100 +#define RT711_VERB_SET_POWER_STATE 0x3500 +#define RT711_VERB_SET_CHANNEL_STREAMID 0x3600 +#define RT711_VERB_SET_PIN_WIDGET_CONTROL 0x3700 +#define RT711_VERB_SET_UNSOLICITED_ENABLE 0x3800 +#define RT711_SET_AMP_GAIN_MUTE_H 0x7300 +#define RT711_SET_AMP_GAIN_MUTE_L 0x8380 +#define RT711_VERB_GET_POWER_STATE 0xb500 +#define RT711_VERB_GET_CHANNEL_STREAMID 0xb600 +#define RT711_VERB_GET_PIN_SENSE 0xb900 +#define RT711_FUNC_RESET 0xff01 + +#define RT711_READ_HDA_3 0x2012 +#define RT711_READ_HDA_2 0x2013 +#define RT711_READ_HDA_1 0x2014 +#define RT711_READ_HDA_0 0x2015 +#define RT711_PRIV_INDEX_W_H 0x7500 +#define RT711_PRIV_INDEX_W_L 0x8580 +#define RT711_PRIV_DATA_W_H 0x7400 +#define RT711_PRIV_DATA_W_L 0x8480 +#define RT711_PRIV_INDEX_R_H 0x9d00 +#define RT711_PRIV_INDEX_R_L 0xad80 +#define RT711_PRIV_DATA_R_H 0x9c00 +#define RT711_PRIV_DATA_R_L 0xac80 +#define RT711_DAC_FORMAT_H 0x7203 +#define RT711_DAC_FORMAT_L 0x8283 +#define RT711_ADC1_FORMAT_H 0x7209 +#define RT711_ADC1_FORMAT_L 0x8289 +#define RT711_ADC2_FORMAT_H 0x7208 +#define RT711_ADC2_FORMAT_L 0x8288 + +#define RT711_SET_AUDIO_POWER_STATE\ + (RT711_VERB_SET_POWER_STATE | RT711_AUDIO_FUNCTION_GROUP) +#define RT711_GET_AUDIO_POWER_STATE\ + (RT711_VERB_GET_POWER_STATE | RT711_AUDIO_FUNCTION_GROUP) +#define RT711_SET_PIN_DMIC1\ + (RT711_VERB_SET_PIN_WIDGET_CONTROL | RT711_DMIC1) +#define RT711_SET_PIN_DMIC2\ + (RT711_VERB_SET_PIN_WIDGET_CONTROL | RT711_DMIC2) +#define RT711_SET_PIN_HP\ + (RT711_VERB_SET_PIN_WIDGET_CONTROL | RT711_HP_OUT) +#define RT711_SET_PIN_MIC2\ + (RT711_VERB_SET_PIN_WIDGET_CONTROL | RT711_MIC2) +#define RT711_SET_PIN_LINE1\ + (RT711_VERB_SET_PIN_WIDGET_CONTROL | RT711_LINE1) +#define RT711_SET_PIN_LINE2\ + (RT711_VERB_SET_PIN_WIDGET_CONTROL | RT711_LINE2) +#define RT711_SET_MIC2_UNSOLICITED_ENABLE\ + (RT711_VERB_SET_UNSOLICITED_ENABLE | RT711_MIC2) +#define RT711_SET_HP_UNSOLICITED_ENABLE\ + (RT711_VERB_SET_UNSOLICITED_ENABLE | RT711_HP_OUT) +#define RT711_SET_INLINE_UNSOLICITED_ENABLE\ + (RT711_VERB_SET_UNSOLICITED_ENABLE | RT711_INLINE_CMD) +#define RT711_SET_STREAMID_DAC2\ + (RT711_VERB_SET_CHANNEL_STREAMID | RT711_DAC_OUT2) +#define RT711_SET_STREAMID_ADC1\ + (RT711_VERB_SET_CHANNEL_STREAMID | RT711_ADC_IN1) +#define RT711_SET_STREAMID_ADC2\ + (RT711_VERB_SET_CHANNEL_STREAMID | RT711_ADC_IN2) +#define RT711_GET_STREAMID_DAC2\ + (RT711_VERB_GET_CHANNEL_STREAMID | RT711_DAC_OUT2) +#define RT711_GET_STREAMID_ADC1\ + (RT711_VERB_GET_CHANNEL_STREAMID | RT711_ADC_IN1) +#define RT711_GET_STREAMID_ADC2\ + (RT711_VERB_GET_CHANNEL_STREAMID | RT711_ADC_IN2) +#define RT711_SET_GAIN_DAC2_L\ + (RT711_SET_AMP_GAIN_MUTE_L | RT711_DAC_OUT2) +#define RT711_SET_GAIN_DAC2_H\ + (RT711_SET_AMP_GAIN_MUTE_H | RT711_DAC_OUT2) +#define RT711_SET_GAIN_ADC1_L\ + (RT711_SET_AMP_GAIN_MUTE_L | RT711_ADC_IN1) +#define RT711_SET_GAIN_ADC1_H\ + (RT711_SET_AMP_GAIN_MUTE_H | RT711_ADC_IN1) +#define RT711_SET_GAIN_ADC2_L\ + (RT711_SET_AMP_GAIN_MUTE_L | RT711_ADC_IN2) +#define RT711_SET_GAIN_ADC2_H\ + (RT711_SET_AMP_GAIN_MUTE_H | RT711_ADC_IN2) +#define RT711_SET_GAIN_AMIC_L\ + (RT711_SET_AMP_GAIN_MUTE_L | RT711_MIC2) +#define RT711_SET_GAIN_AMIC_H\ + (RT711_SET_AMP_GAIN_MUTE_H | RT711_MIC2) +#define RT711_SET_GAIN_DMIC1_L\ + (RT711_SET_AMP_GAIN_MUTE_L | RT711_DMIC1) +#define RT711_SET_GAIN_DMIC1_H\ + (RT711_SET_AMP_GAIN_MUTE_H | RT711_DMIC1) +#define RT711_SET_GAIN_DMIC2_L\ + (RT711_SET_AMP_GAIN_MUTE_L | RT711_DMIC2) +#define RT711_SET_GAIN_DMIC2_H\ + (RT711_SET_AMP_GAIN_MUTE_H | RT711_DMIC2) +#define RT711_SET_GAIN_HP_L\ + (RT711_SET_AMP_GAIN_MUTE_L | RT711_HP_OUT) +#define RT711_SET_GAIN_HP_H\ + (RT711_SET_AMP_GAIN_MUTE_H | RT711_HP_OUT) + +/* DAC DC offset calibration control-1 (0x00)(NID:20h) */ +#define RT711_DAC_DC_CALI_TRIGGER (0x1 << 15) + +/* jack detect control 2 (0x09)(NID:20h) */ +#define RT711_JD2_2PORT_200K_DECODE_HP (0x1 << 13) +#define RT711_HP_JD_SEL_JD1 (0x0 << 1) +#define RT711_HP_JD_SEL_JD2 (0x1 << 1) + +/* CC DET1 (0x11)(NID:20h) */ +#define RT711_HP_JD_FINAL_RESULT_CTL_JD12 (0x1 << 10) +#define RT711_HP_JD_FINAL_RESULT_CTL_CCDET (0x0 << 10) + +/* Parameter & Verb control (0x1a)(NID:20h) */ +#define RT711_HIDDEN_REG_SW_RESET (0x1 << 14) + +/* combo jack auto switch control 2 (0x46)(NID:20h) */ +#define RT711_COMBOJACK_AUTO_DET_STATUS (0x1 << 11) +#define RT711_COMBOJACK_AUTO_DET_TRS (0x1 << 10) +#define RT711_COMBOJACK_AUTO_DET_CTIA (0x1 << 9) +#define RT711_COMBOJACK_AUTO_DET_OMTP (0x1 << 8) + +/* FSM control (0x6f)(NID:20h) */ +#define RT711_CALI_CTL (0x0 << 0) +#define RT711_COMBOJACK_CTL (0x1 << 0) +#define RT711_IMS_CTL (0x2 << 0) +#define RT711_DEPOP_CTL (0x3 << 0) + +/* Impedance Sense Digital Control 1 (0x00)(NID:5bh) */ +#define RT711_TRIGGER_IMS (0x1 << 15) +#define RT711_IMS_EN (0x1 << 6) + +#define RT711_EAPD_HIGH 0x2 +#define RT711_EAPD_LOW 0x0 +#define RT711_MUTE_SFT 7 +/* set input/output mapping to payload[14][15] separately */ +#define RT711_DIR_IN_SFT 6 +#define RT711_DIR_OUT_SFT 7 + +enum { + RT711_AIF1, + RT711_AIF2, + RT711_AIFS, +}; + +enum rt711_jd_src { + RT711_JD_NULL, + RT711_JD1, + RT711_JD2 +}; + +int rt711_io_init(struct device *dev, struct sdw_slave *slave); +int rt711_init(struct device *dev, struct regmap *sdw_regmap, + struct regmap *regmap, struct sdw_slave *slave); + +int rt711_jack_detect(struct rt711_priv *rt711, bool *hp, bool *mic); +int rt711_clock_config(struct device *dev); +#endif /* __RT711_H__ */ -- 2.24.1

3 2

[alsa-devel] [PATCH v2] ASoC: max98090: save and restore SHDN when changing sensitive registers
by Tzung-Bi Shih 09 Jan '20

09 Jan '20

According to the datasheet, there are some registers can only be changed when SHDN is 0. Changing these settings during SHDN = 1 can compromise device stability and performance specifications. Saves SHDN before writing to these sensitive registers and restores SHDN afterward. Here is the register list codec driver of max98090 wants to change: M98090_REG_QUICK_SYSTEM_CLOCK 0x04 M98090_REG_QUICK_SAMPLE_RATE 0x05 M98090_REG_DAI_INTERFACE 0x06 M98090_REG_DAC_PATH 0x07 M98090_REG_MIC_DIRECT_TO_ADC 0x08 M98090_REG_LINE_TO_ADC 0x09 M98090_REG_ANALOG_MIC_LOOP 0x0A M98090_REG_ANALOG_LINE_LOOP 0x0B M98090_REG_SYSTEM_CLOCK 0x1B M98090_REG_CLOCK_MODE 0x1C M98090_REG_CLOCK_RATIO_NI_MSB 0x1D M98090_REG_CLOCK_RATIO_NI_LSB 0x1E M98090_REG_CLOCK_RATIO_MI_MSB 0x1F M98090_REG_CLOCK_RATIO_MI_LSB 0x20 M98090_REG_MASTER_MODE 0x21 M98090_REG_INTERFACE_FORMAT 0x22 M98090_REG_TDM_CONTROL 0x23 M98090_REG_TDM_FORMAT 0x24 M98090_REG_IO_CONFIGURATION 0x25 M98090_REG_FILTER_CONFIG 0x26 M98090_REG_INPUT_ENABLE 0x3E M98090_REG_OUTPUT_ENABLE 0x3F M98090_REG_BIAS_CONTROL 0x42 M98090_REG_DAC_CONTROL 0x43 M98090_REG_ADC_CONTROL 0x44 M98090_REG_DRC_TIMING 0x33 M98090_REG_DRC_COMPRESSOR 0x34 M98090_REG_DRC_EXPANDER 0x35 M98090_REG_DSP_FILTER_ENABLE 0x41 M98090_REG_EQUALIZER_BASE 0x46 M98090_REG_RECORD_BIQUAD_BASE 0xAF M98090_REG_DIGITAL_MIC_ENABLE 0x13 M98090_REG_DIGITAL_MIC_CONFIG 0x14 Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- This patch is a follow up fix for the question: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/157364.h… Changes from v1: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-November/158855.… - fix a typo in commit message - rebase to the latest for-next (a few line numbers changed) sound/soc/codecs/max98090.c | 433 ++++++++++++++++++++++++++---------- sound/soc/codecs/max98090.h | 3 +- 2 files changed, 312 insertions(+), 124 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index e46b6ada13b1..da23810f958e 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -5,24 +5,149 @@ * Copyright 2011-2012 Maxim Integrated Products */ +#include <linux/acpi.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/i2c.h> #include <linux/module.h> +#include <linux/mutex.h> #include <linux/of.h> #include <linux/pm.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/slab.h> -#include <linux/acpi.h> -#include <linux/clk.h> #include <sound/jack.h> +#include <sound/max98090.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/tlv.h> -#include <sound/max98090.h> #include "max98090.h" +static void max98090_shdn_save_locked(struct max98090_priv *max98090) +{ + int shdn = 0; + + /* saved_shdn, saved_count, SHDN are protected by card->dapm_mutex */ + regmap_read(max98090->regmap, M98090_REG_DEVICE_SHUTDOWN, &shdn); + max98090->saved_shdn |= shdn; + ++max98090->saved_count; + + if (shdn) + regmap_write(max98090->regmap, M98090_REG_DEVICE_SHUTDOWN, 0x0); +} + +static void max98090_shdn_restore_locked(struct max98090_priv *max98090) +{ + /* saved_shdn, saved_count, SHDN are protected by card->dapm_mutex */ + if (--max98090->saved_count == 0) { + if (max98090->saved_shdn) { + regmap_write(max98090->regmap, + M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK); + max98090->saved_shdn = 0; + } + } +} + +static void max98090_shdn_save(struct max98090_priv *max98090) +{ + mutex_lock(&max98090->component->card->dapm_mutex); + max98090_shdn_save_locked(max98090); +} + +static void max98090_shdn_restore(struct max98090_priv *max98090) +{ + max98090_shdn_restore_locked(max98090); + mutex_unlock(&max98090->component->card->dapm_mutex); +} + +static int max98090_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct max98090_priv *max98090 = + snd_soc_component_get_drvdata(component); + int ret; + + max98090_shdn_save(max98090); + ret = snd_soc_put_volsw(kcontrol, ucontrol); + max98090_shdn_restore(max98090); + + return ret; +} + +static int max98090_dapm_put_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct max98090_priv *max98090 = + snd_soc_component_get_drvdata(component); + int ret; + + max98090_shdn_save(max98090); + ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + max98090_shdn_restore(max98090); + + return ret; +} + +static int max98090_put_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct max98090_priv *max98090 = + snd_soc_component_get_drvdata(component); + int ret; + + max98090_shdn_save(max98090); + ret = snd_soc_put_enum_double(kcontrol, ucontrol); + max98090_shdn_restore(max98090); + + return ret; +} + +static int max98090_bytes_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct max98090_priv *max98090 = + snd_soc_component_get_drvdata(component); + int ret; + + max98090_shdn_save(max98090); + ret = snd_soc_bytes_put(kcontrol, ucontrol); + max98090_shdn_restore(max98090); + + return ret; +} + +static int max98090_dapm_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 max98090_priv *max98090 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_PRE_PMD: + max98090_shdn_save_locked(max98090); + break; + case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_POST_PMD: + max98090_shdn_restore_locked(max98090); + break; + } + + return 0; +} + /* Allows for sparsely populated register maps */ static const struct reg_default max98090_reg[] = { { 0x00, 0x00 }, /* 00 Software Reset */ @@ -506,10 +631,13 @@ static SOC_ENUM_SINGLE_DECL(max98090_adchp_enum, max98090_pwr_perf_text); static const struct snd_kcontrol_new max98090_snd_controls[] = { - SOC_ENUM("MIC Bias VCM Bandgap", max98090_vcmbandgap_enum), + SOC_ENUM_EXT("MIC Bias VCM Bandgap", max98090_vcmbandgap_enum, + snd_soc_get_enum_double, max98090_put_enum_double), - SOC_SINGLE("DMIC MIC Comp Filter Config", M98090_REG_DIGITAL_MIC_CONFIG, - M98090_DMIC_COMP_SHIFT, M98090_DMIC_COMP_NUM - 1, 0), + SOC_SINGLE_EXT("DMIC MIC Comp Filter Config", + M98090_REG_DIGITAL_MIC_CONFIG, + M98090_DMIC_COMP_SHIFT, M98090_DMIC_COMP_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), SOC_SINGLE_EXT_TLV("MIC1 Boost Volume", M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, @@ -564,24 +692,34 @@ static const struct snd_kcontrol_new max98090_snd_controls[] = { M98090_AVR_SHIFT, M98090_AVR_NUM - 1, 1, max98090_av_tlv), - SOC_ENUM("ADC Oversampling Rate", max98090_osr128_enum), - SOC_SINGLE("ADC Quantizer Dither", M98090_REG_ADC_CONTROL, - M98090_ADCDITHER_SHIFT, M98090_ADCDITHER_NUM - 1, 0), - SOC_ENUM("ADC High Performance Mode", max98090_adchp_enum), - - SOC_SINGLE("DAC Mono Mode", M98090_REG_IO_CONFIGURATION, - M98090_DMONO_SHIFT, M98090_DMONO_NUM - 1, 0), - SOC_SINGLE("SDIN Mode", M98090_REG_IO_CONFIGURATION, - M98090_SDIEN_SHIFT, M98090_SDIEN_NUM - 1, 0), - SOC_SINGLE("SDOUT Mode", M98090_REG_IO_CONFIGURATION, - M98090_SDOEN_SHIFT, M98090_SDOEN_NUM - 1, 0), - SOC_SINGLE("SDOUT Hi-Z Mode", M98090_REG_IO_CONFIGURATION, - M98090_HIZOFF_SHIFT, M98090_HIZOFF_NUM - 1, 1), - SOC_ENUM("Filter Mode", max98090_mode_enum), - SOC_SINGLE("Record Path DC Blocking", M98090_REG_FILTER_CONFIG, - M98090_AHPF_SHIFT, M98090_AHPF_NUM - 1, 0), - SOC_SINGLE("Playback Path DC Blocking", M98090_REG_FILTER_CONFIG, - M98090_DHPF_SHIFT, M98090_DHPF_NUM - 1, 0), + SOC_ENUM_EXT("ADC Oversampling Rate", max98090_osr128_enum, + snd_soc_get_enum_double, max98090_put_enum_double), + SOC_SINGLE_EXT("ADC Quantizer Dither", M98090_REG_ADC_CONTROL, + M98090_ADCDITHER_SHIFT, M98090_ADCDITHER_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), + SOC_ENUM_EXT("ADC High Performance Mode", max98090_adchp_enum, + snd_soc_get_enum_double, max98090_put_enum_double), + + SOC_SINGLE_EXT("DAC Mono Mode", M98090_REG_IO_CONFIGURATION, + M98090_DMONO_SHIFT, M98090_DMONO_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), + SOC_SINGLE_EXT("SDIN Mode", M98090_REG_IO_CONFIGURATION, + M98090_SDIEN_SHIFT, M98090_SDIEN_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), + SOC_SINGLE_EXT("SDOUT Mode", M98090_REG_IO_CONFIGURATION, + M98090_SDOEN_SHIFT, M98090_SDOEN_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), + SOC_SINGLE_EXT("SDOUT Hi-Z Mode", M98090_REG_IO_CONFIGURATION, + M98090_HIZOFF_SHIFT, M98090_HIZOFF_NUM - 1, 1, + snd_soc_get_volsw, max98090_put_volsw), + SOC_ENUM_EXT("Filter Mode", max98090_mode_enum, + snd_soc_get_enum_double, max98090_put_enum_double), + SOC_SINGLE_EXT("Record Path DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_AHPF_SHIFT, M98090_AHPF_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), + SOC_SINGLE_EXT("Playback Path DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_DHPF_SHIFT, M98090_DHPF_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), SOC_SINGLE_TLV("Digital BQ Volume", M98090_REG_ADC_BIQUAD_LEVEL, M98090_AVBQ_SHIFT, M98090_AVBQ_NUM - 1, 1, max98090_dv_tlv), SOC_SINGLE_EXT_TLV("Digital Sidetone Volume", @@ -594,13 +732,17 @@ static const struct snd_kcontrol_new max98090_snd_controls[] = { SOC_SINGLE_TLV("Digital Volume", M98090_REG_DAI_PLAYBACK_LEVEL, M98090_DV_SHIFT, M98090_DV_NUM - 1, 1, max98090_dv_tlv), - SND_SOC_BYTES("EQ Coefficients", M98090_REG_EQUALIZER_BASE, 105), - SOC_SINGLE("Digital EQ 3 Band Switch", M98090_REG_DSP_FILTER_ENABLE, - M98090_EQ3BANDEN_SHIFT, M98090_EQ3BANDEN_NUM - 1, 0), - SOC_SINGLE("Digital EQ 5 Band Switch", M98090_REG_DSP_FILTER_ENABLE, - M98090_EQ5BANDEN_SHIFT, M98090_EQ5BANDEN_NUM - 1, 0), - SOC_SINGLE("Digital EQ 7 Band Switch", M98090_REG_DSP_FILTER_ENABLE, - M98090_EQ7BANDEN_SHIFT, M98090_EQ7BANDEN_NUM - 1, 0), + SND_SOC_BYTES_E("EQ Coefficients", M98090_REG_EQUALIZER_BASE, 105, + snd_soc_bytes_get, max98090_bytes_put), + SOC_SINGLE_EXT("Digital EQ 3 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ3BANDEN_SHIFT, M98090_EQ3BANDEN_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), + SOC_SINGLE_EXT("Digital EQ 5 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ5BANDEN_SHIFT, M98090_EQ5BANDEN_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), + SOC_SINGLE_EXT("Digital EQ 7 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ7BANDEN_SHIFT, M98090_EQ7BANDEN_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), SOC_SINGLE("Digital EQ Clipping Detection", M98090_REG_DAI_PLAYBACK_LEVEL_EQ, M98090_EQCLPN_SHIFT, M98090_EQCLPN_NUM - 1, 1), @@ -608,25 +750,34 @@ static const struct snd_kcontrol_new max98090_snd_controls[] = { M98090_DVEQ_SHIFT, M98090_DVEQ_NUM - 1, 1, max98090_dv_tlv), - SOC_SINGLE("ALC Enable", M98090_REG_DRC_TIMING, - M98090_DRCEN_SHIFT, M98090_DRCEN_NUM - 1, 0), - SOC_ENUM("ALC Attack Time", max98090_drcatk_enum), - SOC_ENUM("ALC Release Time", max98090_drcrls_enum), + SOC_SINGLE_EXT("ALC Enable", M98090_REG_DRC_TIMING, + M98090_DRCEN_SHIFT, M98090_DRCEN_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), + SOC_ENUM_EXT("ALC Attack Time", max98090_drcatk_enum, + snd_soc_get_enum_double, max98090_put_enum_double), + SOC_ENUM_EXT("ALC Release Time", max98090_drcrls_enum, + snd_soc_get_enum_double, max98090_put_enum_double), SOC_SINGLE_TLV("ALC Make Up Volume", M98090_REG_DRC_GAIN, M98090_DRCG_SHIFT, M98090_DRCG_NUM - 1, 0, max98090_alcmakeup_tlv), - SOC_ENUM("ALC Compression Ratio", max98090_alccmp_enum), - SOC_ENUM("ALC Expansion Ratio", max98090_drcexp_enum), - SOC_SINGLE_TLV("ALC Compression Threshold Volume", + SOC_ENUM_EXT("ALC Compression Ratio", max98090_alccmp_enum, + snd_soc_get_enum_double, max98090_put_enum_double), + SOC_ENUM_EXT("ALC Expansion Ratio", max98090_drcexp_enum, + snd_soc_get_enum_double, max98090_put_enum_double), + SOC_SINGLE_EXT_TLV("ALC Compression Threshold Volume", M98090_REG_DRC_COMPRESSOR, M98090_DRCTHC_SHIFT, - M98090_DRCTHC_NUM - 1, 1, max98090_alccomp_tlv), - SOC_SINGLE_TLV("ALC Expansion Threshold Volume", + M98090_DRCTHC_NUM - 1, 1, + snd_soc_get_volsw, max98090_put_volsw, max98090_alccomp_tlv), + SOC_SINGLE_EXT_TLV("ALC Expansion Threshold Volume", M98090_REG_DRC_EXPANDER, M98090_DRCTHE_SHIFT, - M98090_DRCTHE_NUM - 1, 1, max98090_drcexp_tlv), + M98090_DRCTHE_NUM - 1, 1, + snd_soc_get_volsw, max98090_put_volsw, max98090_drcexp_tlv), - SOC_ENUM("DAC HP Playback Performance Mode", - max98090_dac_perfmode_enum), - SOC_ENUM("DAC High Performance Mode", max98090_dachp_enum), + SOC_ENUM_EXT("DAC HP Playback Performance Mode", + max98090_dac_perfmode_enum, + snd_soc_get_enum_double, max98090_put_enum_double), + SOC_ENUM_EXT("DAC High Performance Mode", max98090_dachp_enum, + snd_soc_get_enum_double, max98090_put_enum_double), SOC_SINGLE_TLV("Headphone Left Mixer Volume", M98090_REG_HP_CONTROL, M98090_MIXHPLG_SHIFT, @@ -684,9 +835,12 @@ static const struct snd_kcontrol_new max98090_snd_controls[] = { SOC_SINGLE("Volume Adjustment Smoothing", M98090_REG_LEVEL_CONTROL, M98090_VSENN_SHIFT, M98090_VSENN_NUM - 1, 1), - SND_SOC_BYTES("Biquad Coefficients", M98090_REG_RECORD_BIQUAD_BASE, 15), - SOC_SINGLE("Biquad Switch", M98090_REG_DSP_FILTER_ENABLE, - M98090_ADCBQEN_SHIFT, M98090_ADCBQEN_NUM - 1, 0), + SND_SOC_BYTES_E("Biquad Coefficients", + M98090_REG_RECORD_BIQUAD_BASE, 15, + snd_soc_bytes_get, max98090_bytes_put), + SOC_SINGLE_EXT("Biquad Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_ADCBQEN_SHIFT, M98090_ADCBQEN_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), }; static const struct snd_kcontrol_new max98091_snd_controls[] = { @@ -695,10 +849,12 @@ static const struct snd_kcontrol_new max98091_snd_controls[] = { M98090_DMIC34_ZEROPAD_SHIFT, M98090_DMIC34_ZEROPAD_NUM - 1, 0), - SOC_ENUM("Filter DMIC34 Mode", max98090_filter_dmic34mode_enum), - SOC_SINGLE("DMIC34 DC Blocking", M98090_REG_FILTER_CONFIG, + SOC_ENUM_EXT("Filter DMIC34 Mode", max98090_filter_dmic34mode_enum, + snd_soc_get_enum_double, max98090_put_enum_double), + SOC_SINGLE_EXT("DMIC34 DC Blocking", M98090_REG_FILTER_CONFIG, M98090_FLT_DMIC34HPF_SHIFT, - M98090_FLT_DMIC34HPF_NUM - 1, 0), + M98090_FLT_DMIC34HPF_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), SOC_SINGLE_TLV("DMIC3 Boost Volume", M98090_REG_DMIC3_VOLUME, M98090_DMIC_AV3G_SHIFT, M98090_DMIC_AV3G_NUM - 1, 0, @@ -716,8 +872,9 @@ static const struct snd_kcontrol_new max98091_snd_controls[] = { SND_SOC_BYTES("DMIC34 Biquad Coefficients", M98090_REG_DMIC34_BIQUAD_BASE, 15), - SOC_SINGLE("DMIC34 Biquad Switch", M98090_REG_DSP_FILTER_ENABLE, - M98090_DMIC34BQEN_SHIFT, M98090_DMIC34BQEN_NUM - 1, 0), + SOC_SINGLE_EXT("DMIC34 Biquad Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_DMIC34BQEN_SHIFT, M98090_DMIC34BQEN_NUM - 1, 0, + snd_soc_get_volsw, max98090_put_volsw), SOC_SINGLE_TLV("DMIC34 BQ PreAttenuation Volume", M98090_REG_DMIC34_BQ_PREATTEN, M98090_AV34BQ_SHIFT, @@ -771,19 +928,6 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, return 0; } -static int max98090_shdn_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 max98090_priv *max98090 = snd_soc_component_get_drvdata(component); - - if (event & SND_SOC_DAPM_POST_PMU) - max98090->shdn_pending = true; - - return 0; - -} - static const char *mic1_mux_text[] = { "IN12", "IN56" }; static SOC_ENUM_SINGLE_DECL(mic1_mux_enum, @@ -884,10 +1028,14 @@ static SOC_ENUM_SINGLE_DECL(ltenr_mux_enum, lten_mux_text); static const struct snd_kcontrol_new max98090_ltenl_mux = - SOC_DAPM_ENUM("LTENL Mux", ltenl_mux_enum); + SOC_DAPM_ENUM_EXT("LTENL Mux", ltenl_mux_enum, + snd_soc_dapm_get_enum_double, + max98090_dapm_put_enum_double); static const struct snd_kcontrol_new max98090_ltenr_mux = - SOC_DAPM_ENUM("LTENR Mux", ltenr_mux_enum); + SOC_DAPM_ENUM_EXT("LTENR Mux", ltenr_mux_enum, + snd_soc_dapm_get_enum_double, + max98090_dapm_put_enum_double); static const char *lben_mux_text[] = { "Normal", "Loopback" }; @@ -902,10 +1050,14 @@ static SOC_ENUM_SINGLE_DECL(lbenr_mux_enum, lben_mux_text); static const struct snd_kcontrol_new max98090_lbenl_mux = - SOC_DAPM_ENUM("LBENL Mux", lbenl_mux_enum); + SOC_DAPM_ENUM_EXT("LBENL Mux", lbenl_mux_enum, + snd_soc_dapm_get_enum_double, + max98090_dapm_put_enum_double); static const struct snd_kcontrol_new max98090_lbenr_mux = - SOC_DAPM_ENUM("LBENR Mux", lbenr_mux_enum); + SOC_DAPM_ENUM_EXT("LBENR Mux", lbenr_mux_enum, + snd_soc_dapm_get_enum_double, + max98090_dapm_put_enum_double); static const char *stenl_mux_text[] = { "Normal", "Sidetone Left" }; @@ -1072,21 +1224,25 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN56"), SND_SOC_DAPM_SUPPLY("MICBIAS", M98090_REG_INPUT_ENABLE, - M98090_MBEN_SHIFT, 0, NULL, 0), + M98090_MBEN_SHIFT, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), SND_SOC_DAPM_SUPPLY("SHDN", M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("SDIEN", M98090_REG_IO_CONFIGURATION, - M98090_SDIEN_SHIFT, 0, NULL, 0), + M98090_SDIEN_SHIFT, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), SND_SOC_DAPM_SUPPLY("SDOEN", M98090_REG_IO_CONFIGURATION, - M98090_SDOEN_SHIFT, 0, NULL, 0), + M98090_SDOEN_SHIFT, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), SND_SOC_DAPM_SUPPLY("DMICL_ENA", M98090_REG_DIGITAL_MIC_ENABLE, - M98090_DIGMICL_SHIFT, 0, max98090_shdn_event, - SND_SOC_DAPM_POST_PMU), + M98090_DIGMICL_SHIFT, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), SND_SOC_DAPM_SUPPLY("DMICR_ENA", M98090_REG_DIGITAL_MIC_ENABLE, - M98090_DIGMICR_SHIFT, 0, max98090_shdn_event, - SND_SOC_DAPM_POST_PMU), + M98090_DIGMICR_SHIFT, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), SND_SOC_DAPM_SUPPLY("AHPF", M98090_REG_FILTER_CONFIG, - M98090_AHPF_SHIFT, 0, NULL, 0), + M98090_AHPF_SHIFT, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), /* * Note: Sysclk and misc power supplies are taken care of by SHDN @@ -1116,10 +1272,12 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { &max98090_lineb_mixer_controls[0], ARRAY_SIZE(max98090_lineb_mixer_controls)), - SND_SOC_DAPM_PGA("LINEA Input", M98090_REG_INPUT_ENABLE, - M98090_LINEAEN_SHIFT, 0, NULL, 0), - SND_SOC_DAPM_PGA("LINEB Input", M98090_REG_INPUT_ENABLE, - M98090_LINEBEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("LINEA Input", M98090_REG_INPUT_ENABLE, + M98090_LINEAEN_SHIFT, 0, NULL, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_PGA_E("LINEB Input", M98090_REG_INPUT_ENABLE, + M98090_LINEBEN_SHIFT, 0, NULL, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), SND_SOC_DAPM_MIXER("Left ADC Mixer", SND_SOC_NOPM, 0, 0, &max98090_left_adc_mixer_controls[0], @@ -1130,11 +1288,11 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { ARRAY_SIZE(max98090_right_adc_mixer_controls)), SND_SOC_DAPM_ADC_E("ADCL", NULL, M98090_REG_INPUT_ENABLE, - M98090_ADLEN_SHIFT, 0, max98090_shdn_event, - SND_SOC_DAPM_POST_PMU), + M98090_ADLEN_SHIFT, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), SND_SOC_DAPM_ADC_E("ADCR", NULL, M98090_REG_INPUT_ENABLE, - M98090_ADREN_SHIFT, 0, max98090_shdn_event, - SND_SOC_DAPM_POST_PMU), + M98090_ADREN_SHIFT, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), SND_SOC_DAPM_AIF_OUT("AIFOUTL", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), @@ -1162,10 +1320,12 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("AIFINL", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIFINR", "HiFi Playback", 1, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DACL", NULL, M98090_REG_OUTPUT_ENABLE, - M98090_DALEN_SHIFT, 0), - SND_SOC_DAPM_DAC("DACR", NULL, M98090_REG_OUTPUT_ENABLE, - M98090_DAREN_SHIFT, 0), + SND_SOC_DAPM_DAC_E("DACL", NULL, M98090_REG_OUTPUT_ENABLE, + M98090_DALEN_SHIFT, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_DAC_E("DACR", NULL, M98090_REG_OUTPUT_ENABLE, + M98090_DAREN_SHIFT, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), SND_SOC_DAPM_MIXER("Left Headphone Mixer", SND_SOC_NOPM, 0, 0, &max98090_left_hp_mixer_controls[0], @@ -1200,20 +1360,26 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_MUX("MIXHPRSEL Mux", SND_SOC_NOPM, 0, 0, &max98090_mixhprsel_mux), - SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE, - M98090_HPLEN_SHIFT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP Right Out", M98090_REG_OUTPUT_ENABLE, - M98090_HPREN_SHIFT, 0, NULL, 0), - - SND_SOC_DAPM_PGA("SPK Left Out", M98090_REG_OUTPUT_ENABLE, - M98090_SPLEN_SHIFT, 0, NULL, 0), - SND_SOC_DAPM_PGA("SPK Right Out", M98090_REG_OUTPUT_ENABLE, - M98090_SPREN_SHIFT, 0, NULL, 0), - - SND_SOC_DAPM_PGA("RCV Left Out", M98090_REG_OUTPUT_ENABLE, - M98090_RCVLEN_SHIFT, 0, NULL, 0), - SND_SOC_DAPM_PGA("RCV Right Out", M98090_REG_OUTPUT_ENABLE, - M98090_RCVREN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("HP Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_HPLEN_SHIFT, 0, NULL, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_PGA_E("HP Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_HPREN_SHIFT, 0, NULL, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), + + SND_SOC_DAPM_PGA_E("SPK Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_SPLEN_SHIFT, 0, NULL, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_PGA_E("SPK Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_SPREN_SHIFT, 0, NULL, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), + + SND_SOC_DAPM_PGA_E("RCV Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_RCVLEN_SHIFT, 0, NULL, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_PGA_E("RCV Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_RCVREN_SHIFT, 0, NULL, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), @@ -1228,9 +1394,11 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DMIC4"), SND_SOC_DAPM_SUPPLY("DMIC3_ENA", M98090_REG_DIGITAL_MIC_ENABLE, - M98090_DIGMIC3_SHIFT, 0, NULL, 0), + M98090_DIGMIC3_SHIFT, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), SND_SOC_DAPM_SUPPLY("DMIC4_ENA", M98090_REG_DIGITAL_MIC_ENABLE, - M98090_DIGMIC4_SHIFT, 0, NULL, 0), + M98090_DIGMIC4_SHIFT, 0, max98090_dapm_event, + SND_SOC_DAPM_PRE_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD), }; static const struct snd_soc_dapm_route max98090_dapm_routes[] = { @@ -1501,6 +1669,11 @@ static void max98090_configure_bclk(struct snd_soc_component *component) return; } + /* + * Master mode: no need to save and restore SHDN for the following + * sensitive registers. + */ + /* Check for supported PCLK to LRCLK ratios */ for (i = 0; i < ARRAY_SIZE(pclk_rates); i++) { if ((pclk_rates[i] == max98090->sysclk) && @@ -1587,12 +1760,14 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: /* Set to slave mode PLL - MAS mode off */ + max98090_shdn_save(max98090); snd_soc_component_write(component, M98090_REG_CLOCK_RATIO_NI_MSB, 0x00); snd_soc_component_write(component, M98090_REG_CLOCK_RATIO_NI_LSB, 0x00); snd_soc_component_update_bits(component, M98090_REG_CLOCK_MODE, M98090_USE_M1_MASK, 0); + max98090_shdn_restore(max98090); max98090->master = false; break; case SND_SOC_DAIFMT_CBM_CFM: @@ -1618,7 +1793,9 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, dev_err(component->dev, "DAI clock mode unsupported"); return -EINVAL; } + max98090_shdn_save(max98090); snd_soc_component_write(component, M98090_REG_MASTER_MODE, regval); + max98090_shdn_restore(max98090); regval = 0; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -1663,8 +1840,10 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, if (max98090->tdm_slots > 1) regval ^= M98090_BCI_MASK; + max98090_shdn_save(max98090); snd_soc_component_write(component, M98090_REG_INTERFACE_FORMAT, regval); + max98090_shdn_restore(max98090); } return 0; @@ -1676,6 +1855,7 @@ static int max98090_set_tdm_slot(struct snd_soc_dai *codec_dai, struct snd_soc_component *component = codec_dai->component; struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); struct max98090_cdata *cdata; + cdata = &max98090->dai[0]; if (slots < 0 || slots > 4) @@ -1685,6 +1865,7 @@ static int max98090_set_tdm_slot(struct snd_soc_dai *codec_dai, max98090->tdm_width = slot_width; if (max98090->tdm_slots > 1) { + max98090_shdn_save(max98090); /* SLOTL SLOTR SLOTDLY */ snd_soc_component_write(component, M98090_REG_TDM_FORMAT, 0 << M98090_TDM_SLOTL_SHIFT | @@ -1695,6 +1876,7 @@ static int max98090_set_tdm_slot(struct snd_soc_dai *codec_dai, snd_soc_component_update_bits(component, M98090_REG_TDM_CONTROL, M98090_TDM_MASK, M98090_TDM_MASK); + max98090_shdn_restore(max98090); } /* @@ -1894,6 +2076,7 @@ static int max98090_configure_dmic(struct max98090_priv *max98090, dmic_freq = dmic_table[pclk_index].settings[micclk_index].freq; dmic_comp = dmic_table[pclk_index].settings[micclk_index].comp[i]; + max98090_shdn_save(max98090); regmap_update_bits(max98090->regmap, M98090_REG_DIGITAL_MIC_ENABLE, M98090_MICCLK_MASK, micclk_index << M98090_MICCLK_SHIFT); @@ -1902,6 +2085,7 @@ static int max98090_configure_dmic(struct max98090_priv *max98090, M98090_DMIC_COMP_MASK | M98090_DMIC_FREQ_MASK, dmic_comp << M98090_DMIC_COMP_SHIFT | dmic_freq << M98090_DMIC_FREQ_SHIFT); + max98090_shdn_restore(max98090); return 0; } @@ -1938,8 +2122,10 @@ static int max98090_dai_hw_params(struct snd_pcm_substream *substream, switch (params_width(params)) { case 16: + max98090_shdn_save(max98090); snd_soc_component_update_bits(component, M98090_REG_INTERFACE_FORMAT, M98090_WS_MASK, 0); + max98090_shdn_restore(max98090); break; default: return -EINVAL; @@ -1950,6 +2136,7 @@ static int max98090_dai_hw_params(struct snd_pcm_substream *substream, cdata->rate = max98090->lrclk; + max98090_shdn_save(max98090); /* Update filter mode */ if (max98090->lrclk < 24000) snd_soc_component_update_bits(component, M98090_REG_FILTER_CONFIG, @@ -1965,6 +2152,7 @@ static int max98090_dai_hw_params(struct snd_pcm_substream *substream, else snd_soc_component_update_bits(component, M98090_REG_FILTER_CONFIG, M98090_DHF_MASK, M98090_DHF_MASK); + max98090_shdn_restore(max98090); max98090_configure_dmic(max98090, max98090->dmic_freq, max98090->pclk, max98090->lrclk); @@ -1995,6 +2183,7 @@ static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, * 0x02 (when master clk is 20MHz to 40MHz).. * 0x03 (when master clk is 40MHz to 60MHz).. */ + max98090_shdn_save(max98090); if ((freq >= 10000000) && (freq <= 20000000)) { snd_soc_component_write(component, M98090_REG_SYSTEM_CLOCK, M98090_PSCLK_DIV1); @@ -2009,8 +2198,10 @@ static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, max98090->pclk = freq >> 2; } else { dev_err(component->dev, "Invalid master clock frequency\n"); + max98090_shdn_restore(max98090); return -EINVAL; } + max98090_shdn_restore(max98090); max98090->sysclk = freq; @@ -2122,10 +2313,12 @@ static void max98090_pll_work(struct max98090_priv *max98090) */ /* Toggle shutdown OFF then ON */ + mutex_lock(&component->card->dapm_mutex); snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, 0); snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, M98090_SHDNN_MASK); + mutex_unlock(&component->card->dapm_mutex); for (i = 0; i < 10; ++i) { /* Give PLL time to lock */ @@ -2448,7 +2641,12 @@ static int max98090_probe(struct snd_soc_component *component) */ snd_soc_component_read32(component, M98090_REG_DEVICE_STATUS); - /* High Performance is default */ + /* + * SHDN should be 0 at the point, no need to save/restore for the + * following registers. + * + * High Performance is default + */ snd_soc_component_update_bits(component, M98090_REG_DAC_CONTROL, M98090_DACHP_MASK, 1 << M98090_DACHP_SHIFT); @@ -2459,7 +2657,12 @@ static int max98090_probe(struct snd_soc_component *component) M98090_ADCHP_MASK, 1 << M98090_ADCHP_SHIFT); - /* Turn on VCM bandgap reference */ + /* + * SHDN should be 0 at the point, no need to save/restore for the + * following registers. + * + * Turn on VCM bandgap reference + */ snd_soc_component_write(component, M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_MASK); @@ -2491,25 +2694,9 @@ static void max98090_remove(struct snd_soc_component *component) max98090->component = NULL; } -static void max98090_seq_notifier(struct snd_soc_component *component, - enum snd_soc_dapm_type event, int subseq) -{ - struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); - - if (max98090->shdn_pending) { - snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, - M98090_SHDNN_MASK, 0); - msleep(40); - snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, - M98090_SHDNN_MASK, M98090_SHDNN_MASK); - max98090->shdn_pending = false; - } -} - static const struct snd_soc_component_driver soc_component_dev_max98090 = { .probe = max98090_probe, .remove = max98090_remove, - .seq_notifier = max98090_seq_notifier, .set_bias_level = max98090_set_bias_level, .idle_bias_on = 1, .use_pmdown_time = 1, diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index a197114b0dad..0a31708b7df7 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1539,7 +1539,8 @@ struct max98090_priv { unsigned int pa2en; unsigned int sidetone; bool master; - bool shdn_pending; + int saved_count; + int saved_shdn; }; int max98090_mic_detect(struct snd_soc_component *component, -- 2.24.0.432.g9d3f5f5b63-goog

3 20

[alsa-devel] CRAS UCMs V2
by GitHub pull_request - opened 09 Jan '20

09 Jan '20

alsa-project/alsa-lib pull request #21 was opened from cujomalainey: v2 changes Dropped Jack* as it looks like we may need to do some renaming internally. Dropped Gain controls as we are adjusting our definitions. Dropped disable software volume per comments and will look at removing internally Notes: Per request regarding CaptureChannelMap, is it work changing the format if the current format transfers directly to alsa plugin inputs? Apologies for the multi month delay between versions Request URL : https://github.com/alsa-project/alsa-lib/pull/21 Patch URL : https://github.com/alsa-project/alsa-lib/pull/21.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH v4] ASoC: Add MediaTek MT6660 Speaker Amp Driver
by Jeff Chang 09 Jan '20

09 Jan '20

From: Jeff Chang <jeff_chang(a)richtek.com> The MT6660 is a boosted BTL class-D amplifier with V/I sensing. A built-in DC-DC step-up converter is used to provide efficient power for class-D amplifier with multi-level class-G operation. The digital audio interface supports I2S, left-justified, right-justified, TDM and DSP A/B format for audio in with a data out used for chip information like voltage sense and current sense, which are able to be monitored via DATAO through proper Signed-off-by: Jeff Chang <jeff_chang(a)richtek.com> --- Documentation/devicetree/bindings/sound/mt6660.txt | 53 ++ sound/soc/codecs/Kconfig | 11 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6660.c | 628 +++++++++++++++++++++ sound/soc/codecs/mt6660.h | 75 +++ 5 files changed, 769 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6660.txt create mode 100644 sound/soc/codecs/mt6660.c create mode 100644 sound/soc/codecs/mt6660.h changelogs between v4 & v3 - remove unnecessary kcontrols. - modify copy right header. - use dev_dbg instead of dev_info. - add necessary debug message. - add DT binding documentation. - add space before } at every table. changelogs between v3 & v2 - modify MT6660 Kconfig, remove unnecessary selection. - remove my own debug io interface. use standard regmap for debugging. - remove regmap volatile ops, we do not use cache. - remove component io read/write function, use snd_soc_component_init_regmap. - remove init setting write code. Using parsing dts to set them. - remove unnecessary pr_info log message. - remove mt6660_component_put_volsw. Using snd_soc_get_volsw diff --git a/Documentation/devicetree/bindings/sound/mt6660.txt b/Documentation/devicetree/bindings/sound/mt6660.txt new file mode 100644 index 0000000..2a1736b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt6660.txt @@ -0,0 +1,53 @@ +MT6660 MediaTek Speaker Amplifier + +This device supports I2C mode only. + +Required properties: + + - compatible : "mediatek,mt6660" + + - reg : The I2C slave address + +Optional properties: + + - rt,init_setting_num : The initial register setting element number. + + - rt,init_setting_addr : the addreses array for INIT Setting table. + + - rt,init_setting_mask : the mask array for INIT Setting table. + + - rt,init_setting_val : the value array for INIT Setting table. + +Example: + + mt6660@34 { + status = "ok"; + compatible = "mediatek,mt6660"; + reg = <0x34>; + rt,init_setting_num = <26>; + rt,init_setting_addr = + <0x20 0x30 0x50 0xB1 + 0xD3 0xE0 0x98 0xB9 + 0xB7 0xB6 0x6B 0x07 + 0xBB 0x69 0xBD 0x70 + 0x7C 0x46 0x1A 0x1B + 0x51 0xA2 0x33 0x4C + 0x15 0x68>; + rt,init_setting_mask = + <0x80 0x01 0x1c 0x0c + 0x03 0x01 0x44 0xff + 0x7777 0x07 0xe0 0xff + 0xff 0xff 0xffff 0xff + 0xff 0xff 0xffffffff 0xffffffff + 0xff 0xff 0xffff 0xffff + 0x1800 0x1f>; + rt,init_setting_val = + <0x00 0x00 0x04 0x00 + 0x03 0x00 0x04 0x82 + 0x7273 0x03 0x20 0x70 + 0x20 0x40 0x17f8 0x15 + 0x00 0x1d 0x7fdb7ffe 0x7fdb7ffe + 0x58 0xce 0x7fff 0x0116 + 0x0800 0x07>; + }; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 229cc89..f135fbb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_MT6358 if MTK_PMIC_WRAP + select SND_SOC_MT6660 if I2C select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8822 if I2C @@ -1465,6 +1466,16 @@ config SND_SOC_MT6358 Enable support for the platform which uses MT6358 as external codec device. +config SND_SOC_MT6660 + tristate "Mediatek MT6660 Speaker Amplifier" + depends on I2C + help + MediaTek MT6660 is a smart power amplifier which contain + speaker protection, multi-band DRC, equalizer functions. + Select N if you don't have MT6660 on board. + Select M to build this as module. + + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c498373..2b6814c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -119,6 +119,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o +snd-soc-mt6660-objs := mt6660.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -403,6 +404,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o +obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c new file mode 100644 index 0000000..b8fc53b --- /dev/null +++ b/sound/soc/codecs/mt6660.c @@ -0,0 +1,628 @@ +// SPDX-License-Identifier: GPL-2.0 // + +// Copyright (c) 2019 MediaTek Inc. + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/version.h> +#include <linux/err.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/pcm_params.h> +#include <linux/debugfs.h> + +#include "mt6660.h" + +struct codec_reg_val { + u32 addr; + u32 mask; + u32 data; +}; + +struct reg_size_table { + u32 addr; + u8 size; +}; + +static const struct reg_size_table mt6660_reg_size_table[] = { + { MT6660_REG_HPF1_COEF, 4 }, + { MT6660_REG_HPF2_COEF, 4 }, + { MT6660_REG_TDM_CFG3, 2 }, + { MT6660_REG_RESV17, 2 }, + { MT6660_REG_RESV23, 2 }, + { MT6660_REG_SIGMAX, 2 }, + { MT6660_REG_DEVID, 2 }, + { MT6660_REG_HCLIP_CTRL, 2 }, + { MT6660_REG_DA_GAIN, 2 }, +}; + +static int mt6660_get_reg_size(uint32_t addr) +{ + int i = 0; + + for (i = 0; i < ARRAY_SIZE(mt6660_reg_size_table); i++) { + if (mt6660_reg_size_table[i].addr == addr) + return mt6660_reg_size_table[i].size; + } + return 1; +} + +static int mt6660_reg_write(void *context, unsigned int reg, unsigned int val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + u8 reg_data[4] = {0}; + int i = 0, ret = 0; + + for (i = 0; i < size; i++) + reg_data[size - i - 1] = (val >> (8 * i)) & 0xff; + + ret = i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); + if (ret < 0) + return ret; + return 0; +} + +static int mt6660_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + int i = 0, ret = 0; + u8 data[4] = {0}; + u32 reg_data = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, reg, size, data); + if (ret < 0) + return ret; + for (i = 0; i < size; i++) { + reg_data <<= 8; + reg_data |= data[i]; + } + *val = reg_data; + return 0; +} + +static struct regmap_config mt6660_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .reg_write = mt6660_reg_write, + .reg_read = mt6660_reg_read, +}; + +static int mt6660_codec_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + usleep_range(1000, 1100); + break; + } + return 0; +} + +static int mt6660_codec_classd_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); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dev_dbg(component->dev, + "%s: before classd turn on\n", __func__); + /* config to adaptive mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x03); + if (ret < 0) { + dev_err(component->dev, "config mode adaptive fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMU: + /* voltage sensing enable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x04); + if (ret < 0) { + dev_err(component->dev, + "enable voltage sensing fail\n"); + return ret; + } + dev_dbg(component->dev, "Amp on\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + dev_dbg(component->dev, "Amp off\n"); + /* voltage sensing disable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x00); + if (ret < 0) { + dev_err(component->dev, + "disable voltage sensing fail\n"); + return ret; + } + /* pop-noise improvement 1 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x10); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 1 fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, + "%s: after classd turn off\n", __func__); + /* pop-noise improvement 2 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x00); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 2 fail\n"); + return ret; + } + /* config to off mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x00); + if (ret < 0) { + dev_err(component->dev, "config mode off fail\n"); + return ret; + } + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget mt6660_component_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("DAC", NULL, MT6660_REG_PLL_CFG1, + 0, 1, mt6660_codec_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC("VI ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("ClassD", MT6660_REG_SYSTEM_CTRL, 2, 0, + NULL, 0, mt6660_codec_classd_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("OUTP"), + SND_SOC_DAPM_OUTPUT("OUTN"), +}; + +static const struct snd_soc_dapm_route mt6660_component_dapm_routes[] = { + { "DAC", NULL, "aif_playback" }, + { "PGA", NULL, "DAC" }, + { "ClassD", NULL, "PGA" }, + { "OUTP", NULL, "ClassD" }, + { "OUTN", NULL, "ClassD" }, + { "VI ADC", NULL, "ClassD" }, + { "aif_capture", NULL, "VI ADC" }, +}; + +static int mt6660_component_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = -EINVAL; + + if (!strcmp(kcontrol->id.name, "Chip Rev")) { + ucontrol->value.integer.value[0] = chip->chip_rev & 0x0f; + ret = 0; + } + return ret; +} + +static const DECLARE_TLV_DB_SCALE(vol_ctl_tlv, -1155, 5, 0); + +static const struct snd_kcontrol_new mt6660_component_snd_controls[] = { + SOC_SINGLE_TLV("Digital Volume", MT6660_REG_VOL_CTRL, 0, 255, + 1, vol_ctl_tlv), + SOC_SINGLE("Hard Clip Switch", MT6660_REG_HCLIP_CTRL, 8, 1, 0), + SOC_SINGLE("Clip Switch", MT6660_REG_SPS_CTRL, 0, 1, 0), + SOC_SINGLE("Boost Mode", MT6660_REG_BST_CTRL, 0, 3, 0), + SOC_SINGLE("DRE Switch", MT6660_REG_DRE_CTRL, 0, 1, 0), + SOC_SINGLE("DC Protect Switch", MT6660_REG_DC_PROTECT_CTRL, 3, 1, 0), + SOC_SINGLE("Data Output Left Channel Selection", + MT6660_REG_DATAO_SEL, 3, 7, 0), + SOC_SINGLE("Data Output Right Channel Selection", + MT6660_REG_DATAO_SEL, 0, 7, 0), + SOC_SINGLE_EXT("T0 SEL", MT6660_REG_CALI_T0, 0, 7, 0, + snd_soc_get_volsw, NULL), + SOC_SINGLE_EXT("Chip Rev", MT6660_REG_DEVID, 8, 15, 0, + mt6660_component_get_volsw, NULL), +}; + +static inline int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) +{ + u8 reg_data = 0; + int ret = 0; + + ret = i2c_smbus_read_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL); + if (ret < 0) + return ret; + reg_data = (u8)ret; + if (on_off) + reg_data &= (~0x01); + else + reg_data |= 0x01; + return regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, reg_data); +} + +static int mt6660_apply_plat_data(struct mt6660_chip *chip, + struct snd_soc_component *component) +{ + size_t i = 0; + int num = chip->plat_data.init_setting_num; + int ret = 0; + + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "%s power on failed\n", __func__); + return ret; + } + + for (i = 0; i < num; i++) { + ret = snd_soc_component_update_bits(component, + chip->plat_data.init_setting_addr[i], + chip->plat_data.init_setting_mask[i], + chip->plat_data.init_setting_val[i]); + if (ret < 0) + return ret; + } + ret = _mt6660_chip_power_on(chip, 0); + if (ret < 0) { + dev_err(chip->dev, "%s power on failed\n", __func__); + return ret; + } + return 0; +} + +static int mt6660_component_probe(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + + dev_dbg(component->dev, "%s\n", __func__); + snd_soc_component_init_regmap(component, chip->regmap); + + ret = mt6660_apply_plat_data(chip, component); + if (ret < 0) + dev_err(chip->dev, "mt6660 apply plat data failed\n"); + + return ret; +} + +static void mt6660_component_remove(struct snd_soc_component *component) +{ + dev_dbg(component->dev, "%s\n", __func__); + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver mt6660_component_driver = { + .probe = mt6660_component_probe, + .remove = mt6660_component_remove, + + .controls = mt6660_component_snd_controls, + .num_controls = ARRAY_SIZE(mt6660_component_snd_controls), + .dapm_widgets = mt6660_component_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt6660_component_dapm_widgets), + .dapm_routes = mt6660_component_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt6660_component_dapm_routes), + + .idle_bias_on = false, /* idle_bias_off = true */ +}; + +static int mt6660_component_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) +{ + int word_len = params_physical_width(hw_params); + int aud_bit = params_width(hw_params); + u16 reg_data = 0; + int ret = 0; + + dev_dbg(dai->dev, "%s: ++\n", __func__); + dev_dbg(dai->dev, "format: 0x%08x\n", params_format(hw_params)); + dev_dbg(dai->dev, "rate: 0x%08x\n", params_rate(hw_params)); + dev_dbg(dai->dev, "word_len: %d, aud_bit: %d\n", word_len, aud_bit); + if (word_len > 32 || word_len < 16) { + dev_err(dai->dev, "not supported word length\n"); + return -ENOTSUPP; + } + switch (aud_bit) { + case 16: + reg_data = 3; + break; + case 18: + reg_data = 2; + break; + case 20: + reg_data = 1; + break; + case 24: + case 32: + reg_data = 0; + break; + default: + return -ENOTSUPP; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_SERIAL_CFG1, 0xc0, (reg_data << 6)); + if (ret < 0) { + dev_err(dai->dev, "config aud bit fail\n"); + return ret; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_TDM_CFG3, 0x3f0, word_len << 4); + if (ret < 0) { + dev_err(dai->dev, "config word len fail\n"); + return ret; + } + dev_dbg(dai->dev, "%s: --\n", __func__); + return 0; +} + +static const struct snd_soc_dai_ops mt6660_component_aif_ops = { + .hw_params = mt6660_component_aif_hw_params, +}; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_soc_dai_driver mt6660_codec_dai = { + .name = "mt6660-aif", + .playback = { + .stream_name = "aif_playback", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "aif_capture", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + /* dai properties */ + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, + /* dai operations */ + .ops = &mt6660_component_aif_ops, +}; + +static inline int _mt6660_chip_id_check(struct mt6660_chip *chip) +{ + u8 id[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, MT6660_REG_DEVID, 2, id); + if (ret < 0) + return ret; + ret = (id[0] << 8) + id[1]; + ret &= 0x0ff0; + if (ret != 0x00e0 && ret != 0x01e0) { + dev_err(chip->dev, "%s id(%x) not match\n", __func__, ret); + return -ENODEV; + } + return ret; +} + +static inline int _mt6660_chip_sw_reset(struct mt6660_chip *chip) +{ + int ret; + + /* turn on main pll first, then trigger reset */ + ret = regmap_write(chip->regmap, 0x03, 0x00); + if (ret < 0) + return ret; + ret = regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, 0x80); + msleep(30); + return 0; +} + +static inline int _mt6660_read_chip_revision(struct mt6660_chip *chip) +{ + u8 reg_data[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, MT6660_REG_DEVID, 2, reg_data); + if (ret < 0) { + dev_err(chip->dev, "get chip revision fail\n"); + return ret; + } + chip->chip_rev = reg_data[1]; + return 0; +} + +static int mt6660_parse_dt(struct mt6660_chip *chip, struct device *dev) +{ + struct device_node *np = dev->of_node; + u32 val; + size_t i = 0; + + if (!np) { + dev_err(dev, "no device node\n"); + return -EINVAL; + } + + if (of_property_read_u32(np, "rt,init_setting_num", &val)) { + dev_err(dev, "no init setting\n"); + chip->plat_data.init_setting_num = 0; + } else { + chip->plat_data.init_setting_num = val; + } + + chip->plat_data.init_setting_addr = + devm_kzalloc(dev, sizeof(u32) * + chip->plat_data.init_setting_num, GFP_KERNEL); + chip->plat_data.init_setting_mask = + devm_kzalloc(dev, sizeof(u32) * + chip->plat_data.init_setting_num, GFP_KERNEL); + chip->plat_data.init_setting_val = + devm_kzalloc(dev, sizeof(u32) * + chip->plat_data.init_setting_num, GFP_KERNEL); + + if (of_property_read_u32_array(np, "rt,init_setting_addr", + chip->plat_data.init_setting_addr, + chip->plat_data.init_setting_num)) { + dev_err(dev, "no init setting addr\n"); + } + if (of_property_read_u32_array(np, "rt,init_setting_mask", + chip->plat_data.init_setting_mask, + chip->plat_data.init_setting_num)) { + dev_err(dev, "no init setting addr\n"); + } + if (of_property_read_u32_array(np, "rt,init_setting_val", + chip->plat_data.init_setting_val, + chip->plat_data.init_setting_num)) { + dev_err(dev, "no init setting addr\n"); + } + + dev_dbg(dev, "%s, init stting table, num = %d\n", __func__, + chip->plat_data.init_setting_num); + for (i = 0; i < chip->plat_data.init_setting_num; i++) { + dev_dbg(dev, "0x%02x, 0x%08x, 0x%08x\n", + chip->plat_data.init_setting_addr[i], + chip->plat_data.init_setting_mask[i], + chip->plat_data.init_setting_val[i]); + } + return 0; +} + +static int mt6660_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct mt6660_chip *chip = NULL; + int ret = 0; + + dev_dbg(&client->dev, "%s\n", __func__); + chip = devm_kzalloc(&client->dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + chip->i2c = client; + chip->dev = &client->dev; + mutex_init(&chip->io_lock); + i2c_set_clientdata(client, chip); + + ret = mt6660_parse_dt(chip, &client->dev); + if (ret < 0) { + dev_err(&client->dev, "parsing dts failed\n"); + return ret; + } + + chip->regmap = devm_regmap_init(&client->dev, + NULL, chip, &mt6660_regmap_config); + if (IS_ERR(chip->regmap)) { + ret = PTR_ERR(chip->regmap); + dev_err(&client->dev, "failed to initialise regmap: %d\n", ret); + return ret; + } + + /* chip reset first */ + ret = _mt6660_chip_sw_reset(chip); + if (ret < 0) { + dev_err(chip->dev, "chip reset fail\n"); + goto probe_fail; + } + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 2 fail\n"); + goto probe_fail; + } + /* chip devid check */ + ret = _mt6660_chip_id_check(chip); + if (ret < 0) { + dev_err(chip->dev, "chip id check fail\n"); + goto probe_fail; + } + /* chip revision get */ + ret = _mt6660_read_chip_revision(chip); + if (ret < 0) { + dev_err(chip->dev, "read chip revision fail\n"); + goto probe_fail; + } + pm_runtime_set_active(chip->dev); + pm_runtime_enable(chip->dev); + + ret = devm_snd_soc_register_component(chip->dev, + &mt6660_component_driver, + &mt6660_codec_dai, 1); + return ret; +probe_fail: + _mt6660_chip_power_on(chip, 0); + mutex_destroy(&chip->io_lock); + return ret; +} + +static int mt6660_i2c_remove(struct i2c_client *client) +{ + struct mt6660_chip *chip = i2c_get_clientdata(client); + + pm_runtime_disable(chip->dev); + pm_runtime_set_suspended(chip->dev); + mutex_destroy(&chip->io_lock); + return 0; +} + +static int __maybe_unused mt6660_i2c_runtime_suspend(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "enter low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x01); +} + +static int __maybe_unused mt6660_i2c_runtime_resume(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "exit low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x00); +} + +static const struct dev_pm_ops mt6660_dev_pm_ops = { + SET_RUNTIME_PM_OPS(mt6660_i2c_runtime_suspend, + mt6660_i2c_runtime_resume, NULL) +}; + +static const struct of_device_id __maybe_unused mt6660_of_id[] = { + { .compatible = "mediatek,mt6660",}, + {}, +}; +MODULE_DEVICE_TABLE(of, mt6660_of_id); + +static const struct i2c_device_id mt6660_i2c_id[] = { + {"mt6660", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); + +static struct i2c_driver mt6660_i2c_driver = { + .driver = { + .name = "mt6660", + .of_match_table = of_match_ptr(mt6660_of_id), + .pm = &mt6660_dev_pm_ops, + }, + .probe = mt6660_i2c_probe, + .remove = mt6660_i2c_remove, + .id_table = mt6660_i2c_id, +}; +module_i2c_driver(mt6660_i2c_driver); + +MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com>"); +MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); +MODULE_LICENSE("GPL"); +MODULE_VERSION("1.0.7_G"); diff --git a/sound/soc/codecs/mt6660.h b/sound/soc/codecs/mt6660.h new file mode 100644 index 0000000..6c40b40 --- /dev/null +++ b/sound/soc/codecs/mt6660.h @@ -0,0 +1,75 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#ifndef __SND_SOC_MT6660_H +#define __SND_SOC_MT6660_H + +#include <linux/mutex.h> +#include <linux/regmap.h> + +struct mt6660_platform_data { + u8 init_setting_num; + u32 *init_setting_addr; + u32 *init_setting_mask; + u32 *init_setting_val; +}; + +struct mt6660_chip { + struct i2c_client *i2c; + struct device *dev; + struct platform_device *param_dev; + struct mt6660_platform_data plat_data; + struct mutex io_lock; + struct regmap *regmap; + u16 chip_rev; +}; + +#define MT6660_REG_DEVID (0x00) +#define MT6660_REG_SYSTEM_CTRL (0x03) +#define MT6660_REG_IRQ_STATUS1 (0x05) +#define MT6660_REG_ADDA_CLOCK (0x07) +#define MT6660_REG_SERIAL_CFG1 (0x10) +#define MT6660_REG_DATAO_SEL (0x12) +#define MT6660_REG_TDM_CFG3 (0x15) +#define MT6660_REG_HPF_CTRL (0x18) +#define MT6660_REG_HPF1_COEF (0x1A) +#define MT6660_REG_HPF2_COEF (0x1B) +#define MT6660_REG_PATH_BYPASS (0x1E) +#define MT6660_REG_WDT_CTRL (0x20) +#define MT6660_REG_HCLIP_CTRL (0x24) +#define MT6660_REG_VOL_CTRL (0x29) +#define MT6660_REG_SPS_CTRL (0x30) +#define MT6660_REG_SIGMAX (0x33) +#define MT6660_REG_CALI_T0 (0x3F) +#define MT6660_REG_BST_CTRL (0x40) +#define MT6660_REG_PROTECTION_CFG (0x46) +#define MT6660_REG_DA_GAIN (0x4c) +#define MT6660_REG_AUDIO_IN2_SEL (0x50) +#define MT6660_REG_SIG_GAIN (0x51) +#define MT6660_REG_PLL_CFG1 (0x60) +#define MT6660_REG_DRE_CTRL (0x68) +#define MT6660_REG_DRE_THDMODE (0x69) +#define MT6660_REG_DRE_CORASE (0x6B) +#define MT6660_REG_PWM_CTRL (0x70) +#define MT6660_REG_DC_PROTECT_CTRL (0x74) +#define MT6660_REG_ADC_USB_MODE (0x7c) +#define MT6660_REG_INTERNAL_CFG (0x88) +#define MT6660_REG_RESV0 (0x98) +#define MT6660_REG_RESV1 (0x99) +#define MT6660_REG_RESV2 (0x9A) +#define MT6660_REG_RESV3 (0x9B) +#define MT6660_REG_RESV6 (0xA2) +#define MT6660_REG_RESV7 (0xA3) +#define MT6660_REG_RESV10 (0xB0) +#define MT6660_REG_RESV11 (0xB1) +#define MT6660_REG_RESV16 (0xB6) +#define MT6660_REG_RESV17 (0xB7) +#define MT6660_REG_RESV19 (0xB9) +#define MT6660_REG_RESV21 (0xBB) +#define MT6660_REG_RESV23 (0xBD) +#define MT6660_REG_RESV31 (0xD3) +#define MT6660_REG_RESV40 (0xE0) + +#endif /* __SND_SOC_MT6660_H */ -- 2.7.4

3 2

[alsa-devel] [PATCH for-5.6 0/2] drm/bridge: dw-hdmi: PCM API updates
by Takashi Iwai 09 Jan '20

09 Jan '20

Hi, this is a patch set for updating ALSA PCM API usages in dw-hdmi driver. I already tried to "fix" the driver some time ago but it was utterly wrong. So this is a combination of the revised patch and another cleanup patch. The first one is to change the buffer allocation mechanism in the driver to the manual allocation of the h/w buffer and the automatic allocation of PCM stream buffers via the new standard API. The significant change is that size of the h/w buffer isn't no longer controlled via ALSA preallocation proc file but rather via the new module option (if any). The second one is a oneliner patch just to remove the superfluous PCM ops. Both need the ALSA PCM core changes in 5.5-rc1, so please apply them on top of 5.5-rc1 or later. Or, just let me know if I should apply them through sound git tree. thanks, Takashi === Takashi Iwai (2): drm/bridge: dw-hdmi: Follow the standard ALSA memalloc way drm/bridge: dw-hdmi: Drop superfluous ioctl PCM ops .../gpu/drm/bridge/synopsys/dw-hdmi-ahb-audio.c | 49 ++++++++++++---------- 1 file changed, 26 insertions(+), 23 deletions(-) -- 2.16.4

3 5

[alsa-devel] Playback/Capture on a codec-to-codec link
by Gabriele Zampieri 09 Jan '20

09 Jan '20

Dear all, I'm working on a board that is equipped with a Bluetooth chip and a gsm phone, both connected to the CPU (AM3352) on 2 different pcm. I wrote the codecs and they are working (read as: I can play/record on both devices). What I want to achieve now is to route those pcm together and still being able to (at least) play audio on both. I tried alsaloop, but it eats too much CPU (80% +) even in half duplex loop (BT capture => GSM playback), and it was the only application running. In [1] I saw that I should be able to route the codec together in kernel space (to avoid data copy between user and kernel spaces). I did not tried that solution yet because it's not really clear to me how to implement it, any suggestions/examples? If I follow that solution, can I still use those PCM to playback my audio file (I see that I can disconnect that link through a mixer control, so this should be possible)? Again any suggestion is really appreciated. Best regards, Gabriele [1]: https://www.kernel.org/doc/html/latest/sound/soc/dpcm.html#hostless-pcm-str…

1 0

[alsa-devel] [PATCH] ALSA: hda: Rename back to dmic_detect option
by Takashi Iwai 09 Jan '20

09 Jan '20

We've got quite a few bug reports showing the SOF driver being loaded unintentionally recently, and the reason seems to be that users didn't know the module option change: with the recent kernel, a new option dsp_driver=1 has to be passed to a new module snd-intel-dspcfg instead of snd_hda_intel.dmic_detect=0 option. That is, actually there are two tricky things here: - We changed the whole detection in another module and another option semantics. - The existing option for skipping the DSP probe was also renamed. For avoiding the confusion and giving user more hint, this patch reverts the renamed option dsp_driver back to dmic_detect for snd-hda-intel module, and show the warning about the module option change when the non-default value is passed. Fixes: 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5b92f290cbb0..8ef223aa1e37 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -125,7 +125,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dsp_driver = 1; +static bool dmic_detect = 1; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -160,9 +160,10 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dsp_driver, bool, 0444); -MODULE_PARM_DESC(dsp_driver, "Allow DSP driver selection (bypass this driver) " - "(0=off, 1=on) (default=1)"); +module_param(dmic_detect, bool, 0444); +MODULE_PARM_DESC(dmic_detect, "Allow DSP driver selection (bypass this driver) " + "(0=off, 1=on) (default=1); " + "deprecated, use snd-intel-dspcfg.dsp_driver option instead"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2099,11 +2100,13 @@ static int azx_probe(struct pci_dev *pci, /* * stop probe if another Intel's DSP driver should be activated */ - if (dsp_driver) { + if (dmic_detect) { err = snd_intel_dsp_driver_probe(pci); if (err != SND_INTEL_DSP_DRIVER_ANY && err != SND_INTEL_DSP_DRIVER_LEGACY) return -ENODEV; + } else { + dev_warn(&pci->dev, "dmic_detect option is deprecated, pass snd-intel-dspcfg.dsp_driver=1 option instead\n"); } err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, -- 2.16.4

1 0

[alsa-devel] [PATCH][resend] ASoC: simple-card: switch to yaml base Documentation
by Kuninori Morimoto 09 Jan '20

09 Jan '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/simple-card.txt | 351 --------------- .../devicetree/bindings/sound/simple-card.yaml | 472 +++++++++++++++++++++ 2 files changed, 472 insertions(+), 351 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/simple-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-card.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt deleted file mode 100644 index 79954cd6..0000000 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ /dev/null @@ -1,351 +0,0 @@ -Simple-Card: - -Simple-Card specifies audio DAI connections of SoC <-> codec. - -Required properties: - -- compatible : "simple-audio-card" - -Optional properties: - -- simple-audio-card,name : User specified audio sound card name, one string - property. -- simple-audio-card,widgets : Please refer to widgets.txt. -- simple-audio-card,routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. -- simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec - mclk. When defined, mclk-fs property defined in - dai-link sub nodes are ignored. -- simple-audio-card,hp-det-gpio : Reference to GPIO that signals when - headphones are attached. -- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when - a microphone is attached. -- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such - as amplifiers, to be added to the sound card. -- simple-audio-card,pin-switches : List of strings containing the widget names for - which pin switches must be created. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section below. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - -- format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. -- mclk-fs : Multiplication factor between stream - rate and codec mclk, applied only for - the dai-link. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - -- dai-tdm-slot-num : Please refer to tdm-slot.txt. -- dai-tdm-slot-width : Please refer to tdm-slot.txt. -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) - If a clock is specified, it is - enabled with clk_prepare_enable() - in dai startup() and disabled with - clk_disable_unprepare() in dai - shutdown(). - If a clock is specified and a - multiplication factor is given with - mclk-fs, the clock will be set to the - calculated mclk frequency when the - stream starts. -- system-clock-direction-out : specifies clock direction as 'out' on - initialization. It is useful for some aCPUs with - fixed clocks. - -------------------------------------------- -Example 1 - single DAI link: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "VF610-Tower-Sound-Card"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dailink0_master>; - simple-audio-card,frame-master = <&dailink0_master>; - simple-audio-card,widgets = - "Microphone", "Microphone Jack", - "Headphone", "Headphone Jack", - "Speaker", "External Speaker"; - simple-audio-card,routing = - "MIC_IN", "Microphone Jack", - "Headphone Jack", "HP_OUT", - "External Speaker", "LINE_OUT"; - - simple-audio-card,cpu { - sound-dai = <&sh_fsi2 0>; - }; - - dailink0_master: simple-audio-card,codec { - sound-dai = <&ak4648>; - clocks = <&osc>; - }; -}; - -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; - }; -}; - -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; -}; - -------------------------------------------- -Example 2 - many DAI links: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "Cubox Audio"; - - simple-audio-card,dai-link@0 { /* I2S - HDMI */ - reg = <0>; - format = "i2s"; - cpu { - sound-dai = <&audio1 0>; - }; - codec { - sound-dai = <&tda998x 0>; - }; - }; - - simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ - reg = <1>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&tda998x 1>; - }; - }; - - simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ - reg = <2>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&spdif_codec>; - }; - }; -}; - -------------------------------------------- -Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec -through TPA6130A2 amplifier to headphones: -------------------------------------------- - -&i2c0 { - codec: tlv320dac3100@18 { - compatible = "ti,tlv320dac3100"; - ... - } - - amp: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - ... - } -} - -sound { - compatible = "simple-audio-card"; - ... - simple-audio-card,widgets = - "Headphone", "Headphone Jack"; - simple-audio-card,routing = - "Headphone Jack", "HPLEFT", - "Headphone Jack", "HPRIGHT", - "LEFTIN", "HPL", - "RIGHTIN", "HPR"; - simple-audio-card,aux-devs = <&amp>; - simple-audio-card,cpu { - sound-dai = <&ssi2>; - }; - simple-audio-card,codec { - sound-dai = <&codec>; - clocks = ... - }; -}; - -------------------------------------------- -Example 4. Sampling Rate Conversion -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - simple-audio-card,convert-rate = <48000>; - - simple-audio-card,prefix = "ak4642"; - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "DAI0 Capture", "ak4642 Capture"; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - system-clock-frequency = <11289600>; - }; -}; - -------------------------------------------- -Example 5. 2 CPU 1 Codec (Mixing) -------------------------------------------- -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dpcmcpu>; - simple-audio-card,frame-master = <&dpcmcpu>; - - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "ak4642 Playback", "DAI1 Playback"; - - dpcmcpu: cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - - cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - - codec { - prefix = "ak4642"; - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; -}; - -------------------------------------------- -Example 6 - many DAI links with DPCM: -------------------------------------------- - -CPU0 ------ ak4613 -CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ -CPU2 --/ /* DPCM 3ch/4ch */ -CPU3 --/ /* DPCM 5ch/6ch */ -CPU4 --/ /* DPCM 7ch/8ch */ -CPU5 ------ PCM3168A-c - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,routing = - "pcm3168a Playback", "DAI1 Playback", - "pcm3168a Playback", "DAI2 Playback", - "pcm3168a Playback", "DAI3 Playback", - "pcm3168a Playback", "DAI4 Playback"; - - simple-audio-card,dai-link@0 { - format = "left_j"; - bitclock-master = <&sndcpu0>; - frame-master = <&sndcpu0>; - - sndcpu0: cpu { - sound-dai = <&rcar_sound 0>; - }; - codec { - sound-dai = <&ak4613>; - }; - }; - simple-audio-card,dai-link@1 { - format = "i2s"; - bitclock-master = <&sndcpu1>; - frame-master = <&sndcpu1>; - - convert-channels = <8>; /* TDM Split */ - - sndcpu1: cpu@0 { - sound-dai = <&rcar_sound 1>; - }; - cpu@1 { - sound-dai = <&rcar_sound 2>; - }; - cpu@2 { - sound-dai = <&rcar_sound 3>; - }; - cpu@3 { - sound-dai = <&rcar_sound 4>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - dai-tdm-slot-num = <8>; - sound-dai = <&pcm3168a 0>; - }; - }; - simple-audio-card,dai-link@2 { - format = "i2s"; - bitclock-master = <&sndcpu2>; - frame-master = <&sndcpu2>; - - sndcpu2: cpu { - sound-dai = <&rcar_sound 5>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - sound-dai = <&pcm3168a 1>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml new file mode 100644 index 0000000..ac3975f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -0,0 +1,472 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Card Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +definitions: + + frame-master: + description: Indicates dai-link frame master. + $ref: /schemas/types.yaml#/definitions/phandle-array + + bitclock-master: + description: Indicates dai-link bit clock master + $ref: /schemas/types.yaml#/definitions/phandle-array + + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + + hp-det-gpio: + description: | + Reference to GPIO that signals when headphones are attached. + $ref: /schemas/types.yaml#/definitions/phandle + + aux-devs: + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + $ref: /schemas/types.yaml#/definitions/phandle-array + + mic-det-gpio: + description: | + Reference to GPIO that signals when a microphone is attached. + $ref: /schemas/types.yaml#/definitions/phandle + + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + + pin-switches: + description: the widget names for which pin switches must be created. + $ref: /schemas/types.yaml#/definitions/string-array + + format: + description: audio format. + oneOf: + - items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + + # common properties + mclk-fs: + $ref: "#/definitions/mclk-fs" + prefix: + $ref: "#/definitions/prefix" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + frame-master: + $ref: /schemas/types.yaml#/definitions/flag + bitclock-master: + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + clocks: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/phandle-array + system-clock-frequency: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/uint32 + system-clock-direction-out: + description: | + specifies clock direction as 'out' on initialization. + It is useful for some aCPUs with fixed clocks. + $ref: /schemas/types.yaml#/definitions/flag + required: + - sound-dai + +properties: + compatible: + contains: + enum: + - simple-audio-card + - simple-scu-audio-card + + "#address-cells": true + "#size-cells": true + + simple-audio-card,name: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/string + + simple-audio-card,widgets: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + simple-audio-card,routing: + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + +patternProperties: + "^simple-audio-card,cpu.*": + description: CPU node + $ref: "#/definitions/dai" + "^simple-audio-card,codec.*": + description: CPU node + $ref: "#/definitions/dai" + + # common properties + "^simple-audio-card,frame-master$": + $ref: "#/definitions/frame-master" + "^simple-audio-card,bitclock-master$": + $ref: "#/definitions/bitclock-master" + "^simple-audio-card,frame-inversion$": + $ref: "#/definitions/frame-inversion" + "^simple-audio-card,bitclock-inversion$": + $ref: "#/definitions/bitclock-inversion" + "^simple-audio-card,format$": + $ref: "#/definitions/format" + "^simple-audio-card,mclk-fs$": + $ref: "#/definitions/mclk-fs" + "^simple-audio-card,hp-det-gpio$": + $ref: "#/definitions/hp-det-gpio" + "^simple-audio-card,mic-det-gpio$": + $ref: "#/definitions/mic-det-gpio" + "^simple-audio-card,aux-devs$": + $ref: "#/definitions/aux-devs" + "^simple-audio-card,convert-rate$": + $ref: "#/definitions/convert-rate" + "^simple-audio-card,convert-channels$": + $ref: "#/definitions/convert-channels" + "^simple-audio-card,prefix$": + $ref: "#/definitions/prefix" + "^simple-audio-card,pin-switches$": + $ref: "#/definitions/pin-switches" + + "^simple-audio-card,dai-link": + description: | + Container for dai-link level properties and the CPU and CODEC sub-nodes. + This container may be omitted when the card has only one DAI link. + type: object + properties: + reg: + $ref: /schemas/types.yaml#/definitions/uint32 + + # common properties + frame-master: + $ref: "#/definitions/frame-master" + bitclock-master: + $ref: "#/definitions/bitclock-master" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + format: + $ref: "#/definitions/format" + mclk-fs: + $ref: "#/definitions/mclk-fs" + hp-det-gpio: + $ref: "#/definitions/hp-det-gpio" + mic-det-gpio: + $ref: "#/definitions/mic-det-gpio" + aux-devs: + $ref: "#/definitions/aux-devs" + convert-rate: + $ref: "#/definitions/convert-rate" + convert-channels: + $ref: "#/definitions/convert-channels" + prefix: + $ref: "#/definitions/prefix" + pin-switches: + $ref: "#/definitions/pin-switches" + + patternProperties: + "^cpu.*": + $ref: "#/definitions/dai" + "^codec.*": + $ref: "#/definitions/dai" + additionalProperties: false + +required: + - compatible + +additionalProperties: false + +examples: +#-------------------- +# single DAI link +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dailink0_master>; + simple-audio-card,frame-master = <&dailink0_master>; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + dailink0_master: simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + }; + }; + +#-------------------- +# Multi DAI links +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "Cubox Audio"; + + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,dai-link@0 { /* I2S - HDMI */ + reg = <0>; + format = "i2s"; + cpu { + sound-dai = <&audio0>; + }; + codec { + sound-dai = <&tda998x0>; + }; + }; + + simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ + reg = <1>; + cpu { + sound-dai = <&audio1>; + }; + codec { + sound-dai = <&tda998x1>; + }; + }; + + simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + reg = <2>; + cpu { + sound-dai = <&audio2>; + }; + codec { + sound-dai = <&spdif_codec>; + }; + }; + }; + +#-------------------- +# route audio from IMX6 SSI2 through TLV320DAC3100 codec +# through TPA6130A2 amplifier to headphones: +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&amp>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +#-------------------- +# Sampling Rate Conversion +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + simple-audio-card,convert-rate = <48000>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "DAI0 Capture", "ak4642 Capture"; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; + }; + +#-------------------- +# 2 CPU 1 Codec (Mixing) +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; + + simple-audio-card,convert-rate = <48000>; + simple-audio-card,convert-channels = <2>; + + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: simple-audio-card,cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + + simple-audio-card,cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + + simple-audio-card,codec { + prefix = "ak4642"; + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; + }; + +#-------------------- +# Multi DAI links with DPCM: +# +# CPU0 ------ ak4613 +# CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ +# CPU2 --/ /* DPCM 3ch/4ch */ +# CPU3 --/ /* DPCM 5ch/6ch */ +# CPU4 --/ /* DPCM 7ch/8ch */ +# CPU5 ------ PCM3168A-c +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,routing = + "pcm3168a Playback", "DAI1 Playback", + "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback"; + + simple-audio-card,dai-link@0 { + format = "left_j"; + bitclock-master = <&sndcpu0>; + frame-master = <&sndcpu0>; + + sndcpu0: cpu { + sound-dai = <&rcar_sound 0>; + }; + codec { + sound-dai = <&ak4613>; + }; + }; + + simple-audio-card,dai-link@1 { + format = "i2s"; + bitclock-master = <&sndcpu1>; + frame-master = <&sndcpu1>; + + convert-channels = <8>; /* TDM Split */ + + sndcpu1: cpu@0 { + sound-dai = <&rcar_sound 1>; + }; + cpu@1 { + sound-dai = <&rcar_sound 2>; + }; + cpu@2 { + sound-dai = <&rcar_sound 3>; + }; + cpu@3 { + sound-dai = <&rcar_sound 4>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + dai-tdm-slot-num = <8>; + sound-dai = <&pcm3168a 0>; + }; + }; + + simple-audio-card,dai-link@2 { + format = "i2s"; + bitclock-master = <&sndcpu2>; + frame-master = <&sndcpu2>; + + sndcpu2: cpu { + sound-dai = <&rcar_sound 5>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + sound-dai = <&pcm3168a 1>; + }; + }; + }; -- 2.7.4

2 2

[alsa-devel] MIDI 2.0 packet interface impact on ALSA & Linux
by happy musicmaker 09 Jan '20

09 Jan '20

What is the impact assessment so far done for MIDI 2.0 for ALSA & Linux ? Will a kernel change be needed for (USB) MIDI to support the MIDI 2.0 packet interface ? How about the ALSA sequencer API ? Any activities around this ? MIDI application based on Jack MIDI impact ? https://www.midi.org/articles-old/details-about-midi-2-0-midi-ci-profiles-a… https://www.midi.org/articles-old/midi-2-0-scope https://thefeaturestory.com/2019/12/05/artlogic-give-exclusive-preview-of-m… https://www.namm.org/thenammshow/2020/events/a3e-workshop-art-logic-midi-20… https://juce.com/discover/stories/introducing-midi-2-0 https://www.musictech.net/features/midi-2-0-what-mean-for-you/

1 0

[alsa-devel] [PATCH][resend] ASoC: renesas, rsnd: switch to yaml base Documentation
by Kuninori Morimoto 09 Jan '20

09 Jan '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. It is still keeping detail explanations at .txt Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/renesas,rsnd.txt | 518 -------------------- .../devicetree/bindings/sound/renesas,rsnd.yaml | 527 +++++++++++++++++++++ 2 files changed, 527 insertions(+), 518 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 797fd03..b731f16 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -253,521 +253,3 @@ This is example of TDM 6ch. Driver can automatically switches TDM <-> stereo mode in this case. see "Example: simple sound card for TDM" - -============================================= -Required properties: -============================================= - -- compatible : "renesas,rcar_sound-<soctype>", fallbacks - "renesas,rcar_sound-gen1" if generation1, and - "renesas,rcar_sound-gen2" if generation2 (or RZ/G1) - "renesas,rcar_sound-gen3" if generation3 (or RZ/G2) - Examples with soctypes are: - - "renesas,rcar_sound-r8a7743" (RZ/G1M) - - "renesas,rcar_sound-r8a7744" (RZ/G1N) - - "renesas,rcar_sound-r8a7745" (RZ/G1E) - - "renesas,rcar_sound-r8a77470" (RZ/G1C) - - "renesas,rcar_sound-r8a774a1" (RZ/G2M) - - "renesas,rcar_sound-r8a774b1" (RZ/G2N) - - "renesas,rcar_sound-r8a774c0" (RZ/G2E) - - "renesas,rcar_sound-r8a7778" (R-Car M1A) - - "renesas,rcar_sound-r8a7779" (R-Car H1) - - "renesas,rcar_sound-r8a7790" (R-Car H2) - - "renesas,rcar_sound-r8a7791" (R-Car M2-W) - - "renesas,rcar_sound-r8a7793" (R-Car M2-N) - - "renesas,rcar_sound-r8a7794" (R-Car E2) - - "renesas,rcar_sound-r8a7795" (R-Car H3) - - "renesas,rcar_sound-r8a7796" (R-Car M3-W) - - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - - "renesas,rcar_sound-r8a77990" (R-Car E3) - - "renesas,rcar_sound-r8a77995" (R-Car D3) -- reg : Should contain the register physical address. - required register is - SRU/ADG/SSI if generation1 - SRU/ADG/SSIU/SSI/AUDIO-DMAC-periperi if generation2/generation3 - Select extended AUDIO-DMAC-periperi address if SoC has it, - otherwise select normal AUDIO-DMAC-periperi address. -- reg-names : Should contain the register names. - scu/adg/ssi if generation1 - scu/adg/ssiu/ssi/audmapp if generation2/generation3 -- rcar_sound,ssi : Should contain SSI feature. - The number of SSI subnode should be same as HW. - see below for detail. -- rcar_sound,ssiu : Should contain SSIU feature. - The number of SSIU subnode should be same as HW. - see below for detail. -- rcar_sound,src : Should contain SRC feature. - The number of SRC subnode should be same as HW. - see below for detail. -- rcar_sound,ctu : Should contain CTU feature. - The number of CTU subnode should be same as HW. - see below for detail. -- rcar_sound,mix : Should contain MIX feature. - The number of MIX subnode should be same as HW. - see below for detail. -- rcar_sound,dvc : Should contain DVC feature. - The number of DVC subnode should be same as HW. - see below for detail. -- rcar_sound,dai : DAI contents. - The number of DAI subnode should be same as HW. - see below for detail. -- #sound-dai-cells : it must be 0 if your system is using single DAI - it must be 1 if your system is using multi DAI -- clocks : References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. -- clock-names : List of necessary clock names. - "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X", - "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i" - -Optional properties: -- #clock-cells : it must be 0 if your system has audio_clkout - it must be 1 if your system has audio_clkout0/1/2/3 -- clock-frequency : for all audio_clkout0/1/2/3 -- clkout-lr-asynchronous : boolean property. it indicates that audio_clkoutn - is asynchronizes with lr-clock. -- resets : References to SSI resets. -- reset-names : List of valid reset names. - "ssi-all", "ssi.X" - -SSI subnode properties: -- interrupts : Should contain SSI interrupt for PIO transfer -- shared-pin : if shared clock pin -- pio-transfer : use PIO transfer mode -- no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case -- dma : Should contain Audio DMAC entry -- dma-names : SSI case "rx" (=playback), "tx" (=capture) - Deprecated: see SSIU subnode properties - SSIU case "rxu" (=playback), "txu" (=capture) - -SSIU subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "rx" (=playback), "tx" (=capture) - -SRC subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "rx" (=playback), "tx" (=capture) - -DVC subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "tx" (=playback/capture) - -DAI subnode properties: -- playback : list of playback modules -- capture : list of capture modules - - -============================================= -Example: -============================================= - -rcar_sound: sound@ec500000 { - #sound-dai-cells = <1>; - compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2"; - reg = <0 0xec500000 0 0x1000>, /* SCU */ - <0 0xec5a0000 0 0x100>, /* ADG */ - <0 0xec540000 0 0x1000>, /* SSIU */ - <0 0xec541000 0 0x1280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ - reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; - - clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>, - <&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>, - <&mstp10_clks R8A7790_CLK_SSI7>, <&mstp10_clks R8A7790_CLK_SSI6>, - <&mstp10_clks R8A7790_CLK_SSI5>, <&mstp10_clks R8A7790_CLK_SSI4>, - <&mstp10_clks R8A7790_CLK_SSI3>, <&mstp10_clks R8A7790_CLK_SSI2>, - <&mstp10_clks R8A7790_CLK_SSI1>, <&mstp10_clks R8A7790_CLK_SSI0>, - <&mstp10_clks R8A7790_CLK_SCU_SRC9>, <&mstp10_clks R8A7790_CLK_SCU_SRC8>, - <&mstp10_clks R8A7790_CLK_SCU_SRC7>, <&mstp10_clks R8A7790_CLK_SCU_SRC6>, - <&mstp10_clks R8A7790_CLK_SCU_SRC5>, <&mstp10_clks R8A7790_CLK_SCU_SRC4>, - <&mstp10_clks R8A7790_CLK_SCU_SRC3>, <&mstp10_clks R8A7790_CLK_SCU_SRC2>, - <&mstp10_clks R8A7790_CLK_SCU_SRC1>, <&mstp10_clks R8A7790_CLK_SCU_SRC0>, - <&mstp10_clks R8A7790_CLK_SCU_DVC0>, <&mstp10_clks R8A7790_CLK_SCU_DVC1>, - <&audio_clk_a>, <&audio_clk_b>, <&audio_clk_c>, <&m2_clk>; - clock-names = "ssi-all", - "ssi.9", "ssi.8", "ssi.7", "ssi.6", "ssi.5", - "ssi.4", "ssi.3", "ssi.2", "ssi.1", "ssi.0", - "src.9", "src.8", "src.7", "src.6", "src.5", - "src.4", "src.3", "src.2", "src.1", "src.0", - "dvc.0", "dvc.1", - "clk_a", "clk_b", "clk_c", "clk_i"; - - rcar_sound,dvc { - dvc0: dvc-0 { - dmas = <&audma0 0xbc>; - dma-names = "tx"; - }; - dvc1: dvc-1 { - dmas = <&audma0 0xbe>; - dma-names = "tx"; - }; - }; - - rcar_sound,mix { - mix0: mix-0 { }; - mix1: mix-1 { }; - }; - - rcar_sound,ctu { - ctu00: ctu-0 { }; - ctu01: ctu-1 { }; - ctu02: ctu-2 { }; - ctu03: ctu-3 { }; - ctu10: ctu-4 { }; - ctu11: ctu-5 { }; - ctu12: ctu-6 { }; - ctu13: ctu-7 { }; - }; - - rcar_sound,src { - src0: src-0 { - interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x85>, <&audma1 0x9a>; - dma-names = "rx", "tx"; - }; - src1: src-1 { - interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x87>, <&audma1 0x9c>; - dma-names = "rx", "tx"; - }; - src2: src-2 { - interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x89>, <&audma1 0x9e>; - dma-names = "rx", "tx"; - }; - src3: src-3 { - interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8b>, <&audma1 0xa0>; - dma-names = "rx", "tx"; - }; - src4: src-4 { - interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8d>, <&audma1 0xb0>; - dma-names = "rx", "tx"; - }; - src5: src-5 { - interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8f>, <&audma1 0xb2>; - dma-names = "rx", "tx"; - }; - src6: src-6 { - interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x91>, <&audma1 0xb4>; - dma-names = "rx", "tx"; - }; - src7: src-7 { - interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x93>, <&audma1 0xb6>; - dma-names = "rx", "tx"; - }; - src8: src-8 { - interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x95>, <&audma1 0xb8>; - dma-names = "rx", "tx"; - }; - src9: src-9 { - interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x97>, <&audma1 0xba>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,ssiu { - ssiu00: ssiu-0 { - dmas = <&audma0 0x15>, <&audma1 0x16>; - dma-names = "rx", "tx"; - }; - ssiu01: ssiu-1 { - dmas = <&audma0 0x35>, <&audma1 0x36>; - dma-names = "rx", "tx"; - }; - - ... - - ssiu95: ssiu-49 { - dmas = <&audma0 0xA5>, <&audma1 0xA6>; - dma-names = "rx", "tx"; - }; - ssiu96: ssiu-50 { - dmas = <&audma0 0xA7>, <&audma1 0xA8>; - dma-names = "rx", "tx"; - }; - ssiu97: ssiu-51 { - dmas = <&audma0 0xA9>, <&audma1 0xAA>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,ssi { - ssi0: ssi-0 { - interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x01>, <&audma1 0x02>; - dma-names = "rx", "tx"; - }; - ssi1: ssi-1 { - interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x03>, <&audma1 0x04>; - dma-names = "rx", "tx"; - }; - - ... - - ssi8: ssi-8 { - interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x11>, <&audma1 0x12>; - dma-names = "rx", "tx"; - }; - ssi9: ssi-9 { - interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x13>, <&audma1 0x14>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,dai { - dai0 { - playback = <&ssi5 &src5>; - capture = <&ssi6>; - }; - dai1 { - playback = <&ssi3>; - }; - dai2 { - capture = <&ssi4>; - }; - dai3 { - playback = <&ssi7>; - }; - dai4 { - capture = <&ssi8>; - }; - }; -}; - -============================================= -Example: simple sound card -============================================= - - rsnd_ak4643: sound { - compatible = "simple-audio-card"; - - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; - }; - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &src2 &dvc0>; - capture = <&ssi1 &src3 &dvc1>; - }; - }; -}; - -&ssi1 { - shared-pin; -}; - -============================================= -Example: simple sound card for Asynchronous mode -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - /* - * SRC Asynchronous mode setting - * Playback: - * All input data will be converted to 48kHz - * Capture: - * Inputed 48kHz data will be converted to - * system specified Hz - */ - simple-audio-card,convert-rate = <48000>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; -}; - -============================================= -Example: simple sound card for channel convert -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - /* - * CTU setting - * All input data will be converted to 2ch - * as output data - */ - simple-audio-card,convert-channels = <2>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; -}; - -============================================= -Example: simple sound card for MIXer -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - simple-audio-card,cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - simple-audio-card,cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - simple-audio-card,codec { - ... - }; -}; - -&rcar_sound { - ... - rcar_sound,dai { - dai0 { - playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>; - }; - dai1 { - playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>; - }; - }; -}; - -============================================= -Example: simple sound card for TDM -============================================= - -rsnd_tdm: sound { - compatible = "simple-audio-card"; - - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - dai-tdm-slot-num = <6>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&xxx>; - }; -}; - -============================================= -Example: simple sound card for TDM Split -============================================= - -sound_card: sound { - compatible = "audio-graph-scu-card"; - prefix = "xxxx"; - routing = "xxxx Playback", "DAI0 Playback", - "xxxx Playback", "DAI1 Playback", - "xxxx Playback", "DAI2 Playback", - "xxxx Playback", "DAI3 Playback"; - convert-channels = <8>; /* TDM Split */ - - dais = <&rsnd_port0 /* playback ch1/ch2 */ - &rsnd_port1 /* playback ch3/ch4 */ - &rsnd_port2 /* playback ch5/ch6 */ - &rsnd_port3 /* playback ch7/ch8 */ - >; -}; - -audio-codec { - ... - port { - codec_0: endpoint@1 { - remote-endpoint = <&rsnd_ep0>; - }; - codec_1: endpoint@2 { - remote-endpoint = <&rsnd_ep1>; - }; - codec_2: endpoint@3 { - remote-endpoint = <&rsnd_ep2>; - }; - codec_3: endpoint@4 { - remote-endpoint = <&rsnd_ep3>; - }; - }; -}; - -&rcar_sound { - ... - ports { - rsnd_port0: port@0 { - rsnd_ep0: endpoint { - remote-endpoint = <&codec_0>; - ... - playback = <&ssiu30 &ssi3>; - }; - }; - rsnd_port1: port@1 { - rsnd_ep1: endpoint { - remote-endpoint = <&codec_1>; - ... - playback = <&ssiu31 &ssi3>; - }; - }; - rsnd_port2: port@2 { - rsnd_ep2: endpoint { - remote-endpoint = <&codec_2>; - ... - playback = <&ssiu32 &ssi3>; - }; - }; - rsnd_port3: port@3 { - rsnd_ep3: endpoint { - remote-endpoint = <&codec_3>; - ... - playback = <&ssiu33 &ssi3>; - }; - }; - }; -}; - -============================================= -Example: simple sound card for Multi channel -============================================= - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml new file mode 100644 index 0000000..5fb036c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -0,0 +1,527 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,rsnd.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas R-Car Sound Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + $nodename: + pattern: "^sound@.*" + + compatible: + oneOf: + # for Gen1 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7778 # R-Car M1A + - renesas,rcar_sound-r8a7779 # R-Car H1 + - enum: + - renesas,rcar_sound-gen1 + # for Gen2 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7743 # RZ/G1M + - renesas,rcar_sound-r8a7744 # RZ/G1N + - renesas,rcar_sound-r8a7745 # RZ/G1E + - renesas,rcar_sound-r8a77470 # RZ/G1C + - renesas,rcar_sound-r8a7790 # R-Car H2 + - renesas,rcar_sound-r8a7791 # R-Car M2-W + - renesas,rcar_sound-r8a7793 # R-Car M2-N + - renesas,rcar_sound-r8a7794 # R-Car E2 + - enum: + - renesas,rcar_sound-gen2 + # for Gen3 SoC + - items: + - enum: + - renesas,rcar_sound-r8a774a1 # RZ/G2M + - renesas,rcar_sound-r8a774b1 # RZ/G2N + - renesas,rcar_sound-r8a774c0 # RZ/G2E + - renesas,rcar_sound-r8a7795 # R-Car H3 + - renesas,rcar_sound-r8a7796 # R-Car M3-W + - renesas,rcar_sound-r8a77965 # R-Car M3-N + - renesas,rcar_sound-r8a77990 # R-Car E3 + - renesas,rcar_sound-r8a77995 # R-Car D3 + - enum: + - renesas,rcar_sound-gen3 + # for Generic + - items: + - enum: + - renesas,rcar_sound-gen1 + - renesas,rcar_sound-gen2 + - renesas,rcar_sound-gen3 + + reg: + minItems: 3 # Gen1 + maxItems: 5 # Gen2/Gen3 + + reg-names: + description: | + Should contain the register names. + scu/adg/ssi if generation1 + scu/adg/ssiu/ssi/audmapp if generation2/generation3 + minItems: 3 # Gen1 + maxItems: 5 # Gen2/Gen3 + allOf: + - items: + enum: + - scu + - adg + - ssiu + - ssi + - audmapp + + "#sound-dai-cells": + description: | + it must be 0 if your system is using single DAI + it must be 1 if your system is using multi DAIs + enum: [0, 1] + + "#clock-cells": + description: | + it must be 0 if your system has audio_clkout + it must be 1 if your system has audio_clkout0/1/2/3 + enum: [0, 1] + + clock-frequency: + description: for audio_clkout0/1/2/3 + $ref: /schemas/types.yaml#/definitions/uint32-array + + clkout-lr-asynchronous: + description: audio_clkoutn is asynchronizes with lr-clock. + $ref: /schemas/types.yaml#/definitions/flag + + power-domains: true + + resets: + description: References to SSI resets + $ref: /schemas/types.yaml#/definitions/phandle-array + + reset-names: + description: List of valid reset names. + allOf: + - items: + enum: + - ssi-all + - ssi.9 + - ssi.8 + - ssi.7 + - ssi.6 + - ssi.5 + - ssi.4 + - ssi.3 + - ssi.2 + - ssi.1 + - ssi.0 + + clocks: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. + + clock-names: + description: List of necessary clock names. + allOf: + - items: + enum: + - ssi-all + - ssi.9 + - ssi.8 + - ssi.7 + - ssi.6 + - ssi.5 + - ssi.4 + - ssi.3 + - ssi.2 + - ssi.1 + - ssi.0 + - src.9 + - src.8 + - src.7 + - src.6 + - src.5 + - src.4 + - src.3 + - src.2 + - src.1 + - src.0 + - mix.1 + - mix.0 + - ctu.1 + - ctu.0 + - dvc.1 + - dvc.0 + - clk_a + - clk_b + - clk_c + - clk_i + + # For OF-graph + port: + description: OF-Graph subnode + type: object + properties: + reg: + $ref: /schemas/types.yaml#/definitions/uint32 + endpoint: + type: object + properties: + remote-endpoint: + $ref: /schemas/types.yaml#/definitions/phandle-array + dai-format: + $ref: "simple-card.yaml#/definitions/format" + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + required: + - remote-endpoint + + # For multi OF-graph + ports: + description: multi OF-Graph subnode + type: object + patternProperties: + "port.*": + $ref: "#properties/port" + +patternProperties: + "^rcar_sound,dvc$": + description: DVC subnode. + type: object + patternProperties: + "dvc.*": + type: object + properties: + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + const: "tx" + required: + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,mix$": + description: MIX subnode. + type: object + patternProperties: + "mix.*": + type: object + additionalProperties: false + + "^rcar_sound,ctu$": + description: CTU subnode. + type: object + patternProperties: + "ctu.*": + type: object + additionalProperties: false + + "^rcar_sound,src$": + description: SRC subnode. + type: object + patternProperties: + "src.*": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + if: + properties: + status: + const: "ok" + then: + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,ssiu$": + description: SSIU subnode. + type: object + patternProperties: + "ssiu.*": + type: object + properties: + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + required: + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,ssi$": + description: SSI subnode. + type: object + patternProperties: + "ssi.*": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + - txu # if no ssiu node + - rxu # if no ssiu node + + shared-pin: + description: shared clock pin + $ref: /schemas/types.yaml#/definitions/flag + pio-transfer: + description: PIO transfer mode + $ref: /schemas/types.yaml#/definitions/flag + no-busif: + description: BUSIF is not used when [mem -> SSI] via DMA case + $ref: /schemas/types.yaml#/definitions/flag + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + # For DAI base + "^rcar_sound,dai$": + description: DAI subnode. + type: object + patternProperties: + "dai.*": + type: object + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + anyOf: + - required: + - playback + - required: + - capture + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + rcar_sound: sound@ec500000 { + #sound-dai-cells = <1>; + compatible = "renesas,rcar_sound-r8a7790", "renesas,rcar_sound-gen2"; + reg = <0 0xec500000 0 0x1000>, /* SCU */ + <0 0xec5a0000 0 0x100>, /* ADG */ + <0 0xec540000 0 0x1000>, /* SSIU */ + <0 0xec541000 0 0x1280>, /* SSI */ + <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; + + clocks = <&mstp10_clks 1005>, /* SSI-ALL */ + <&mstp10_clks 1006>, <&mstp10_clks 1007>, /* SSI9, SSI8 */ + <&mstp10_clks 1008>, <&mstp10_clks 1009>, /* SSI7, SSI6 */ + <&mstp10_clks 1010>, <&mstp10_clks 1011>, /* SSI5, SSI4 */ + <&mstp10_clks 1012>, <&mstp10_clks 1013>, /* SSI3, SSI2 */ + <&mstp10_clks 1014>, <&mstp10_clks 1015>, /* SSI1, SSI0 */ + <&mstp10_clks 1022>, <&mstp10_clks 1023>, /* SRC9, SRC8 */ + <&mstp10_clks 1024>, <&mstp10_clks 1025>, /* SRC7, SRC6 */ + <&mstp10_clks 1026>, <&mstp10_clks 1027>, /* SRC5, SRC4 */ + <&mstp10_clks 1028>, <&mstp10_clks 1029>, /* SRC3, SRC2 */ + <&mstp10_clks 1030>, <&mstp10_clks 1031>, /* SRC1, SRC0 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* DVC0, DVC1 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* CLKA, CLKB */ + <&mstp10_clks 1019>, <&mstp10_clks 1018>; /* CLKC, CLKI */ + + clock-names = "ssi-all", + "ssi.9", "ssi.8", + "ssi.7", "ssi.6", + "ssi.5", "ssi.4", + "ssi.3", "ssi.2", + "ssi.1", "ssi.0", + "src.9", "src.8", + "src.7", "src.6", + "src.5", "src.4", + "src.3", "src.2", + "src.1", "src.0", + "dvc.0", "dvc.1", + "clk_a", "clk_b", + "clk_c", "clk_i"; + + rcar_sound,dvc { + dvc0: dvc-0 { + dmas = <&audma0 0xbc>; + dma-names = "tx"; + }; + dvc1: dvc-1 { + dmas = <&audma0 0xbe>; + dma-names = "tx"; + }; + }; + + + rcar_sound,mix { + mix0: mix-0 { }; + mix1: mix-1 { }; + }; + + rcar_sound,ctu { + ctu00: ctu-0 { }; + ctu01: ctu-1 { }; + ctu02: ctu-2 { }; + ctu03: ctu-3 { }; + ctu10: ctu-4 { }; + ctu11: ctu-5 { }; + ctu12: ctu-6 { }; + ctu13: ctu-7 { }; + }; + + rcar_sound,src { + src0: src-0 { + status = "disabled"; + }; + src1: src-1 { + interrupts = <0 353 0>; + dmas = <&audma0 0x87>, <&audma1 0x9c>; + dma-names = "rx", "tx"; + }; + /* skip after src-2 */ + }; + + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + /* skip after ssiu-2 */ + }; + + rcar_sound,ssi { + ssi0: ssi-0 { + interrupts = <0 370 1>; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; + }; + ssi1: ssi-1 { + interrupts = <0 371 1>; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; + }; + /* skip other ssi-2 */ + }; + + /* DAI base */ + rcar_sound,dai { + dai0 { + playback = <&ssi5 &src5>; + capture = <&ssi6>; + }; + dai1 { + playback = <&ssi3>; + }; + dai2 { + capture = <&ssi4>; + }; + dai3 { + playback = <&ssi7>; + }; + dai4 { + capture = <&ssi8>; + }; + }; + + /* OF-Graph */ + port { + rsnd_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + + /* Multi OF-Graph */ + ports { + #address-cells = <1>; + #size-cells = <0>; + rsnd_port0: port@0 { + reg = <0>; + rsnd_endpoint0: endpoint { + remote-endpoint = <&codec_endpoint1>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + rsnd_port1: port@1 { + reg = <1>; + rsnd_endpoint1: endpoint { + remote-endpoint = <&codec_endpoint2>; + + dai-format = "i2s"; + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + }; + }; + + + codec { + ports { + port@0 { + codec_endpoint: endpoint { + remote-endpoint = <&rsnd_endpoint>; + }; + }; + port@1 { + codec_endpoint1: endpoint { + remote-endpoint = <&rsnd_endpoint0>; + }; + }; + port@2 { + codec_endpoint2: endpoint { + remote-endpoint = <&rsnd_endpoint1>; + }; + }; + }; + }; -- 2.7.4

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[alsa-devel] [PATCH] treewide: remove redundent IS_ERR() before error code check
by Masahiro Yamada 08 Jan '20

08 Jan '20

'PTR_ERR(p) == -E*' is a stronger condition than IS_ERR(p). Hence, IS_ERR(p) is unneeded. The semantic patch that generates this commit is as follows: // <smpl> @@ expression ptr; constant error_code; @@ -IS_ERR(ptr) && (PTR_ERR(ptr) == - error_code) +PTR_ERR(ptr) == - error_code // </smpl> Signed-off-by: Masahiro Yamada <masahiroy(a)kernel.org> --- crypto/af_alg.c | 2 +- drivers/acpi/scan.c | 2 +- drivers/char/hw_random/bcm2835-rng.c | 2 +- drivers/char/hw_random/omap-rng.c | 4 ++-- drivers/clk/clk.c | 2 +- drivers/dma/mv_xor_v2.c | 2 +- drivers/gpio/gpiolib-devres.c | 2 +- drivers/gpio/gpiolib-of.c | 8 ++++---- drivers/gpio/gpiolib.c | 2 +- drivers/i2c/busses/i2c-mv64xxx.c | 5 ++--- drivers/i2c/busses/i2c-synquacer.c | 2 +- drivers/mtd/ubi/build.c | 2 +- drivers/of/device.c | 2 +- drivers/pci/controller/pci-tegra.c | 2 +- drivers/phy/phy-core.c | 4 ++-- drivers/spi/spi-orion.c | 3 +-- drivers/video/fbdev/imxfb.c | 2 +- fs/ext4/super.c | 2 +- fs/f2fs/node.c | 2 +- fs/ocfs2/suballoc.c | 2 +- fs/sysfs/group.c | 2 +- net/core/dev.c | 2 +- net/core/filter.c | 2 +- net/xfrm/xfrm_policy.c | 2 +- sound/soc/codecs/ak4104.c | 3 +-- sound/soc/codecs/cs4270.c | 3 +-- sound/soc/codecs/tlv320aic32x4.c | 6 ++---- sound/soc/sunxi/sun4i-spdif.c | 2 +- 28 files changed, 35 insertions(+), 41 deletions(-) diff --git a/crypto/af_alg.c b/crypto/af_alg.c index 0dceaabc6321..0cc82bec0657 100644 --- a/crypto/af_alg.c +++ b/crypto/af_alg.c @@ -169,7 +169,7 @@ static int alg_bind(struct socket *sock, struct sockaddr *uaddr, int addr_len) sa->salg_name[sizeof(sa->salg_name) + addr_len - sizeof(*sa) - 1] = 0; type = alg_get_type(sa->salg_type); - if (IS_ERR(type) && PTR_ERR(type) == -ENOENT) { + if (PTR_ERR(type) == -ENOENT) { request_module("algif-%s", sa->salg_type); type = alg_get_type(sa->salg_type); } diff --git a/drivers/acpi/scan.c b/drivers/acpi/scan.c index 915650bf519f..6d3448895382 100644 --- a/drivers/acpi/scan.c +++ b/drivers/acpi/scan.c @@ -1462,7 +1462,7 @@ int acpi_dma_configure(struct device *dev, enum dev_dma_attr attr) iort_dma_setup(dev, &dma_addr, &size); iommu = iort_iommu_configure(dev); - if (IS_ERR(iommu) && PTR_ERR(iommu) == -EPROBE_DEFER) + if (PTR_ERR(iommu) == -EPROBE_DEFER) return -EPROBE_DEFER; arch_setup_dma_ops(dev, dma_addr, size, diff --git a/drivers/char/hw_random/bcm2835-rng.c b/drivers/char/hw_random/bcm2835-rng.c index d2a5791eb49f..cbf5eaea662c 100644 --- a/drivers/char/hw_random/bcm2835-rng.c +++ b/drivers/char/hw_random/bcm2835-rng.c @@ -157,7 +157,7 @@ static int bcm2835_rng_probe(struct platform_device *pdev) /* Clock is optional on most platforms */ priv->clk = devm_clk_get(dev, NULL); - if (IS_ERR(priv->clk) && PTR_ERR(priv->clk) == -EPROBE_DEFER) + if (PTR_ERR(priv->clk) == -EPROBE_DEFER) return -EPROBE_DEFER; priv->rng.name = pdev->name; diff --git a/drivers/char/hw_random/omap-rng.c b/drivers/char/hw_random/omap-rng.c index 0ed07d16ec8e..65952393e1bb 100644 --- a/drivers/char/hw_random/omap-rng.c +++ b/drivers/char/hw_random/omap-rng.c @@ -476,7 +476,7 @@ static int omap_rng_probe(struct platform_device *pdev) } priv->clk = devm_clk_get(&pdev->dev, NULL); - if (IS_ERR(priv->clk) && PTR_ERR(priv->clk) == -EPROBE_DEFER) + if (PTR_ERR(priv->clk) == -EPROBE_DEFER) return -EPROBE_DEFER; if (!IS_ERR(priv->clk)) { ret = clk_prepare_enable(priv->clk); @@ -488,7 +488,7 @@ static int omap_rng_probe(struct platform_device *pdev) } priv->clk_reg = devm_clk_get(&pdev->dev, "reg"); - if (IS_ERR(priv->clk_reg) && PTR_ERR(priv->clk_reg) == -EPROBE_DEFER) + if (PTR_ERR(priv->clk_reg) == -EPROBE_DEFER) return -EPROBE_DEFER; if (!IS_ERR(priv->clk_reg)) { ret = clk_prepare_enable(priv->clk_reg); diff --git a/drivers/clk/clk.c b/drivers/clk/clk.c index 6a11239ccde3..3c27699487c1 100644 --- a/drivers/clk/clk.c +++ b/drivers/clk/clk.c @@ -429,7 +429,7 @@ static void clk_core_fill_parent_index(struct clk_core *core, u8 index) parent = ERR_PTR(-EPROBE_DEFER); } else { parent = clk_core_get(core, index); - if (IS_ERR(parent) && PTR_ERR(parent) == -ENOENT && entry->name) + if (PTR_ERR(parent) == -ENOENT && entry->name) parent = clk_core_lookup(entry->name); } diff --git a/drivers/dma/mv_xor_v2.c b/drivers/dma/mv_xor_v2.c index e3850f04f676..157c959311ea 100644 --- a/drivers/dma/mv_xor_v2.c +++ b/drivers/dma/mv_xor_v2.c @@ -750,7 +750,7 @@ static int mv_xor_v2_probe(struct platform_device *pdev) } xor_dev->clk = devm_clk_get(&pdev->dev, NULL); - if (IS_ERR(xor_dev->clk) && PTR_ERR(xor_dev->clk) == -EPROBE_DEFER) { + if (PTR_ERR(xor_dev->clk) == -EPROBE_DEFER) { ret = EPROBE_DEFER; goto disable_reg_clk; } diff --git a/drivers/gpio/gpiolib-devres.c b/drivers/gpio/gpiolib-devres.c index 4421be09b960..72b6001c56ef 100644 --- a/drivers/gpio/gpiolib-devres.c +++ b/drivers/gpio/gpiolib-devres.c @@ -308,7 +308,7 @@ devm_gpiod_get_array_optional(struct device *dev, const char *con_id, struct gpio_descs *descs; descs = devm_gpiod_get_array(dev, con_id, flags); - if (IS_ERR(descs) && (PTR_ERR(descs) == -ENOENT)) + if (PTR_ERR(descs) == -ENOENT) return NULL; return descs; diff --git a/drivers/gpio/gpiolib-of.c b/drivers/gpio/gpiolib-of.c index b696e4598a24..3d3659cd72ef 100644 --- a/drivers/gpio/gpiolib-of.c +++ b/drivers/gpio/gpiolib-of.c @@ -505,24 +505,24 @@ struct gpio_desc *of_find_gpio(struct device *dev, const char *con_id, break; } - if (IS_ERR(desc) && PTR_ERR(desc) == -ENOENT) { + if (PTR_ERR(desc) == -ENOENT) { /* Special handling for SPI GPIOs if used */ desc = of_find_spi_gpio(dev, con_id, &of_flags); } - if (IS_ERR(desc) && PTR_ERR(desc) == -ENOENT) { + if (PTR_ERR(desc) == -ENOENT) { /* This quirk looks up flags and all */ desc = of_find_spi_cs_gpio(dev, con_id, idx, flags); if (!IS_ERR(desc)) return desc; } - if (IS_ERR(desc) && PTR_ERR(desc) == -ENOENT) { + if (PTR_ERR(desc) == -ENOENT) { /* Special handling for regulator GPIOs if used */ desc = of_find_regulator_gpio(dev, con_id, &of_flags); } - if (IS_ERR(desc) && PTR_ERR(desc) == -ENOENT) + if (PTR_ERR(desc) == -ENOENT) desc = of_find_arizona_gpio(dev, con_id, &of_flags); if (IS_ERR(desc)) diff --git a/drivers/gpio/gpiolib.c b/drivers/gpio/gpiolib.c index 78a16e42f222..20a6adf2128f 100644 --- a/drivers/gpio/gpiolib.c +++ b/drivers/gpio/gpiolib.c @@ -5067,7 +5067,7 @@ struct gpio_descs *__must_check gpiod_get_array_optional(struct device *dev, struct gpio_descs *descs; descs = gpiod_get_array(dev, con_id, flags); - if (IS_ERR(descs) && (PTR_ERR(descs) == -ENOENT)) + if (PTR_ERR(descs) == -ENOENT) return NULL; return descs; diff --git a/drivers/i2c/busses/i2c-mv64xxx.c b/drivers/i2c/busses/i2c-mv64xxx.c index a5a95ea5b81a..febb7c7ea72b 100644 --- a/drivers/i2c/busses/i2c-mv64xxx.c +++ b/drivers/i2c/busses/i2c-mv64xxx.c @@ -901,14 +901,13 @@ mv64xxx_i2c_probe(struct platform_device *pd) /* Not all platforms have clocks */ drv_data->clk = devm_clk_get(&pd->dev, NULL); - if (IS_ERR(drv_data->clk) && PTR_ERR(drv_data->clk) == -EPROBE_DEFER) + if (PTR_ERR(drv_data->clk) == -EPROBE_DEFER) return -EPROBE_DEFER; if (!IS_ERR(drv_data->clk)) clk_prepare_enable(drv_data->clk); drv_data->reg_clk = devm_clk_get(&pd->dev, "reg"); - if (IS_ERR(drv_data->reg_clk) && - PTR_ERR(drv_data->reg_clk) == -EPROBE_DEFER) + if (PTR_ERR(drv_data->reg_clk) == -EPROBE_DEFER) return -EPROBE_DEFER; if (!IS_ERR(drv_data->reg_clk)) clk_prepare_enable(drv_data->reg_clk); diff --git a/drivers/i2c/busses/i2c-synquacer.c b/drivers/i2c/busses/i2c-synquacer.c index 39762f0611b1..86026798b4f7 100644 --- a/drivers/i2c/busses/i2c-synquacer.c +++ b/drivers/i2c/busses/i2c-synquacer.c @@ -553,7 +553,7 @@ static int synquacer_i2c_probe(struct platform_device *pdev) &i2c->pclkrate); i2c->pclk = devm_clk_get(&pdev->dev, "pclk"); - if (IS_ERR(i2c->pclk) && PTR_ERR(i2c->pclk) == -EPROBE_DEFER) + if (PTR_ERR(i2c->pclk) == -EPROBE_DEFER) return -EPROBE_DEFER; if (!IS_ERR_OR_NULL(i2c->pclk)) { dev_dbg(&pdev->dev, "clock source %p\n", i2c->pclk); diff --git a/drivers/mtd/ubi/build.c b/drivers/mtd/ubi/build.c index d636bbe214cb..e58dda6c7075 100644 --- a/drivers/mtd/ubi/build.c +++ b/drivers/mtd/ubi/build.c @@ -1159,7 +1159,7 @@ static struct mtd_info * __init open_mtd_device(const char *mtd_dev) * MTD device name. */ mtd = get_mtd_device_nm(mtd_dev); - if (IS_ERR(mtd) && PTR_ERR(mtd) == -ENODEV) + if (PTR_ERR(mtd) == -ENODEV) /* Probably this is an MTD character device node path */ mtd = open_mtd_by_chdev(mtd_dev); } else diff --git a/drivers/of/device.c b/drivers/of/device.c index e9127db7b067..27203bfd0b22 100644 --- a/drivers/of/device.c +++ b/drivers/of/device.c @@ -161,7 +161,7 @@ int of_dma_configure(struct device *dev, struct device_node *np, bool force_dma) coherent ? " " : " not "); iommu = of_iommu_configure(dev, np); - if (IS_ERR(iommu) && PTR_ERR(iommu) == -EPROBE_DEFER) + if (PTR_ERR(iommu) == -EPROBE_DEFER) return -EPROBE_DEFER; dev_dbg(dev, "device is%sbehind an iommu\n", diff --git a/drivers/pci/controller/pci-tegra.c b/drivers/pci/controller/pci-tegra.c index 673a1725ef38..40928c4d261d 100644 --- a/drivers/pci/controller/pci-tegra.c +++ b/drivers/pci/controller/pci-tegra.c @@ -1406,7 +1406,7 @@ static struct phy *devm_of_phy_optional_get_index(struct device *dev, phy = devm_of_phy_get(dev, np, name); kfree(name); - if (IS_ERR(phy) && PTR_ERR(phy) == -ENODEV) + if (PTR_ERR(phy) == -ENODEV) phy = NULL; return phy; diff --git a/drivers/phy/phy-core.c b/drivers/phy/phy-core.c index b04f4fe85ac2..48d355cf0210 100644 --- a/drivers/phy/phy-core.c +++ b/drivers/phy/phy-core.c @@ -690,7 +690,7 @@ struct phy *phy_optional_get(struct device *dev, const char *string) { struct phy *phy = phy_get(dev, string); - if (IS_ERR(phy) && (PTR_ERR(phy) == -ENODEV)) + if (PTR_ERR(phy) == -ENODEV) phy = NULL; return phy; @@ -744,7 +744,7 @@ struct phy *devm_phy_optional_get(struct device *dev, const char *string) { struct phy *phy = devm_phy_get(dev, string); - if (IS_ERR(phy) && (PTR_ERR(phy) == -ENODEV)) + if (PTR_ERR(phy) == -ENODEV) phy = NULL; return phy; diff --git a/drivers/spi/spi-orion.c b/drivers/spi/spi-orion.c index c7266ef295fd..1f59beb7d27e 100644 --- a/drivers/spi/spi-orion.c +++ b/drivers/spi/spi-orion.c @@ -646,8 +646,7 @@ static int orion_spi_probe(struct platform_device *pdev) /* The following clock is only used by some SoCs */ spi->axi_clk = devm_clk_get(&pdev->dev, "axi"); - if (IS_ERR(spi->axi_clk) && - PTR_ERR(spi->axi_clk) == -EPROBE_DEFER) { + if (PTR_ERR(spi->axi_clk) == -EPROBE_DEFER) { status = -EPROBE_DEFER; goto out_rel_clk; } diff --git a/drivers/video/fbdev/imxfb.c b/drivers/video/fbdev/imxfb.c index b3286d1fa543..5fad6bf16322 100644 --- a/drivers/video/fbdev/imxfb.c +++ b/drivers/video/fbdev/imxfb.c @@ -1017,7 +1017,7 @@ static int imxfb_probe(struct platform_device *pdev) } fbi->lcd_pwr = devm_regulator_get(&pdev->dev, "lcd"); - if (IS_ERR(fbi->lcd_pwr) && (PTR_ERR(fbi->lcd_pwr) == -EPROBE_DEFER)) { + if (PTR_ERR(fbi->lcd_pwr) == -EPROBE_DEFER) { ret = -EPROBE_DEFER; goto failed_lcd; } diff --git a/fs/ext4/super.c b/fs/ext4/super.c index 2937a8873fe1..b20ac7a47425 100644 --- a/fs/ext4/super.c +++ b/fs/ext4/super.c @@ -5987,7 +5987,7 @@ static ssize_t ext4_quota_write(struct super_block *sb, int type, bh = ext4_bread(handle, inode, blk, EXT4_GET_BLOCKS_CREATE | EXT4_GET_BLOCKS_METADATA_NOFAIL); - } while (IS_ERR(bh) && (PTR_ERR(bh) == -ENOSPC) && + } while (PTR_ERR(bh) == -ENOSPC && ext4_should_retry_alloc(inode->i_sb, &retries)); if (IS_ERR(bh)) return PTR_ERR(bh); diff --git a/fs/f2fs/node.c b/fs/f2fs/node.c index 3314a0f3405e..9d02cdcdbb07 100644 --- a/fs/f2fs/node.c +++ b/fs/f2fs/node.c @@ -875,7 +875,7 @@ static int truncate_dnode(struct dnode_of_data *dn) /* get direct node */ page = f2fs_get_node_page(F2FS_I_SB(dn->inode), dn->nid); - if (IS_ERR(page) && PTR_ERR(page) == -ENOENT) + if (PTR_ERR(page) == -ENOENT) return 1; else if (IS_ERR(page)) return PTR_ERR(page); diff --git a/fs/ocfs2/suballoc.c b/fs/ocfs2/suballoc.c index 4180c3ef0a68..939df99d2dec 100644 --- a/fs/ocfs2/suballoc.c +++ b/fs/ocfs2/suballoc.c @@ -696,7 +696,7 @@ static int ocfs2_block_group_alloc(struct ocfs2_super *osb, bg_bh = ocfs2_block_group_alloc_contig(osb, handle, alloc_inode, ac, cl); - if (IS_ERR(bg_bh) && (PTR_ERR(bg_bh) == -ENOSPC)) + if (PTR_ERR(bg_bh) == -ENOSPC) bg_bh = ocfs2_block_group_alloc_discontig(handle, alloc_inode, ac, cl); diff --git a/fs/sysfs/group.c b/fs/sysfs/group.c index d41c21fef138..c4ab045926b7 100644 --- a/fs/sysfs/group.c +++ b/fs/sysfs/group.c @@ -449,7 +449,7 @@ int __compat_only_sysfs_link_entry_to_kobj(struct kobject *kobj, } link = kernfs_create_link(kobj->sd, target_name, entry); - if (IS_ERR(link) && PTR_ERR(link) == -EEXIST) + if (PTR_ERR(link) == -EEXIST) sysfs_warn_dup(kobj->sd, target_name); kernfs_put(entry); diff --git a/net/core/dev.c b/net/core/dev.c index 0ad39c87b7fd..6d849b6f5724 100644 --- a/net/core/dev.c +++ b/net/core/dev.c @@ -5723,7 +5723,7 @@ static enum gro_result dev_gro_receive(struct napi_struct *napi, struct sk_buff if (&ptype->list == head) goto normal; - if (IS_ERR(pp) && PTR_ERR(pp) == -EINPROGRESS) { + if (PTR_ERR(pp) == -EINPROGRESS) { ret = GRO_CONSUMED; goto ok; } diff --git a/net/core/filter.c b/net/core/filter.c index 28b3c258188c..687934f3aad7 100644 --- a/net/core/filter.c +++ b/net/core/filter.c @@ -1573,7 +1573,7 @@ int sk_reuseport_attach_bpf(u32 ufd, struct sock *sk) return -EPERM; prog = bpf_prog_get_type(ufd, BPF_PROG_TYPE_SOCKET_FILTER); - if (IS_ERR(prog) && PTR_ERR(prog) == -EINVAL) + if (PTR_ERR(prog) == -EINVAL) prog = bpf_prog_get_type(ufd, BPF_PROG_TYPE_SK_REUSEPORT); if (IS_ERR(prog)) return PTR_ERR(prog); diff --git a/net/xfrm/xfrm_policy.c b/net/xfrm/xfrm_policy.c index f2d1e573ea55..93b9e00cca33 100644 --- a/net/xfrm/xfrm_policy.c +++ b/net/xfrm/xfrm_policy.c @@ -3186,7 +3186,7 @@ struct dst_entry *xfrm_lookup_route(struct net *net, struct dst_entry *dst_orig, flags | XFRM_LOOKUP_QUEUE | XFRM_LOOKUP_KEEP_DST_REF); - if (IS_ERR(dst) && PTR_ERR(dst) == -EREMOTE) + if (PTR_ERR(dst) == -EREMOTE) return make_blackhole(net, dst_orig->ops->family, dst_orig); if (IS_ERR(dst)) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index e8c5fda82e08..979cfb165eed 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -295,8 +295,7 @@ static int ak4104_spi_probe(struct spi_device *spi) reset_gpiod = devm_gpiod_get_optional(&spi->dev, "reset", GPIOD_OUT_HIGH); - if (IS_ERR(reset_gpiod) && - PTR_ERR(reset_gpiod) == -EPROBE_DEFER) + if (PTR_ERR(reset_gpiod) == -EPROBE_DEFER) return -EPROBE_DEFER; /* read the 'reserved' register - according to the datasheet, it diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 793a14d58667..5f25b9f872bd 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -681,8 +681,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, reset_gpiod = devm_gpiod_get_optional(&i2c_client->dev, "reset", GPIOD_OUT_HIGH); - if (IS_ERR(reset_gpiod) && - PTR_ERR(reset_gpiod) == -EPROBE_DEFER) + if (PTR_ERR(reset_gpiod) == -EPROBE_DEFER) return -EPROBE_DEFER; cs4270->regmap = devm_regmap_init_i2c(i2c_client, &cs4270_regmap); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index b4e9a6c73f90..d087f3b20b1d 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1098,11 +1098,9 @@ static int aic32x4_setup_regulators(struct device *dev, return PTR_ERR(aic32x4->supply_av); } } else { - if (IS_ERR(aic32x4->supply_dv) && - PTR_ERR(aic32x4->supply_dv) == -EPROBE_DEFER) + if (PTR_ERR(aic32x4->supply_dv) == -EPROBE_DEFER) return -EPROBE_DEFER; - if (IS_ERR(aic32x4->supply_av) && - PTR_ERR(aic32x4->supply_av) == -EPROBE_DEFER) + if (PTR_ERR(aic32x4->supply_av) == -EPROBE_DEFER) return -EPROBE_DEFER; } diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index cbe598b0fb10..98a9fe645521 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -555,7 +555,7 @@ static int sun4i_spdif_probe(struct platform_device *pdev) if (quirks->has_reset) { host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); - if (IS_ERR(host->rst) && PTR_ERR(host->rst) == -EPROBE_DEFER) { + if (PTR_ERR(host->rst) == -EPROBE_DEFER) { ret = -EPROBE_DEFER; dev_err(&pdev->dev, "Failed to get reset: %d\n", ret); return ret; -- 2.17.1

7 7

[alsa-devel] [PATCH] ALSA: hda: enable regmap internal locking
by Kai Vehmanen 08 Jan '20

08 Jan '20

This reverts commit 42ec336f1f9d ("ALSA: hda: Disable regmap internal locking"). Without regmap locking, there is a race between snd_hda_codec_amp_init() and PM callbacks issuing regcache_sync(). This was caught by following kernel warning trace: <4> [358.080081] WARNING: CPU: 2 PID: 4157 at drivers/base/regmap/regcache.c:498 regcache_cache_only+0xf5/0x130 [...] <4> [358.080148] Call Trace: <4> [358.080158] snd_hda_codec_amp_init+0x4e/0x100 [snd_hda_codec] <4> [358.080169] snd_hda_codec_amp_init_stereo+0x40/0x80 [snd_hda_codec] Suggested-by: Takashi Iwai <tiwai(a)suse.de> BugLink: https://gitlab.freedesktop.org/drm/intel/issues/592 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/hdac_regmap.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 906b1e20bae0..286361ecd640 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -363,7 +363,6 @@ static const struct regmap_config hda_regmap_cfg = { .reg_write = hda_reg_write, .use_single_read = true, .use_single_write = true, - .disable_locking = true, }; /** -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: Fix NULL dereference at freeing
by Takashi Iwai 08 Jan '20

08 Jan '20

When an ASoC driver with pcm_destruct component ops is freed before the PCM object instantiation (e.g. deferring the probe), it hits an Oops at snd_soc_pcm_component_free() that calls the pcm_destruct ops unconditionally. Fix it by adding a NULL-check of rtd->pcm before calling callbacks. Fixes: c64bfc906600 ("ASoC: soc-core: add new pcm_construct/pcm_destruct") Reported-by: Matthias Brugger <matthias.bgg(a)gmail.com> Tested-by: Ben Ho <ben.ho(a)mediatek.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-component.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 9054558ce386..b94680fb26fa 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -539,6 +539,9 @@ void snd_soc_pcm_component_free(struct snd_soc_pcm_runtime *rtd) struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_component *component; + if (!rtd->pcm) + return; + for_each_rtd_components(rtd, rtdcom, component) if (component->driver->pcm_destruct) component->driver->pcm_destruct(component, rtd->pcm); -- 2.16.4

2 1

[alsa-devel] [PATCH 0/2] ASoC: SOF: Intel: update paths to reflect linux-firmware tree
by Pierre-Louis Bossart 08 Jan '20

08 Jan '20

Since we have unfortunately no API to query which firmware signature is required for a given platform, by default the firmware signed with the production key is selected (under intel/sof). DMI quirks are added to deal with platforms using the community key (under intel/sof/community) Pierre-Louis Bossart (2): ASoC: SOF: pci: change the default firmware path when the community key is used ASoC: SOF: loader: add dynamic debug trace sound/soc/sof/loader.c | 3 +++ sound/soc/sof/sof-pci-dev.c | 44 +++++++++++++++++++++++++++++++++++-- 2 files changed, 45 insertions(+), 2 deletions(-) -- 2.20.1

2 4

[alsa-devel] [PATCH] ASoC: gtm601: fix build warning
by Arnd Bergmann 08 Jan '20

08 Jan '20

The driver produces warnings without CONFIG_OF, and makes no sense without it either: sound/soc/codecs/gtm601.c:50:34: error: 'bm818_dai' defined but not used [-Werror=unused-variable] static struct snd_soc_dai_driver bm818_dai = { ^~~~~~~~~ sound/soc/codecs/gtm601.c:32:34: error: 'gtm601_dai' defined but not used [-Werror=unused-variable] static struct snd_soc_dai_driver gtm601_dai = { ^~~~~~~~~~ Remove the #ifdef check to avoid the warning. Fixes: 057a317a8d94 ("ASoC: gtm601: add Broadmobi bm818 sound profile") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/gtm601.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/gtm601.c b/sound/soc/codecs/gtm601.c index 7f05ebcb88d1..ae9e1c70ca57 100644 --- a/sound/soc/codecs/gtm601.c +++ b/sound/soc/codecs/gtm601.c @@ -87,14 +87,12 @@ static int gtm601_platform_probe(struct platform_device *pdev) (struct snd_soc_dai_driver *)dai_driver, 1); } -#if defined(CONFIG_OF) static const struct of_device_id gtm601_codec_of_match[] = { { .compatible = "option,gtm601", .data = (void *)&gtm601_dai }, { .compatible = "broadmobi,bm818", .data = (void *)&bm818_dai }, {}, }; MODULE_DEVICE_TABLE(of, gtm601_codec_of_match); -#endif static struct platform_driver gtm601_codec_driver = { .driver = { -- 2.20.0

2 1

[alsa-devel] [PATCH] ALSA: hda: Fix a typo in comments
by Takashi Iwai 08 Jan '20

08 Jan '20

Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_regmap.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 286361ecd640..0c8188a48a00 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -508,7 +508,7 @@ int snd_hdac_regmap_read_raw_uncached(struct hdac_device *codec, * snd_hdac_regmap_update_raw - update a pseudo register with power mgmt * @codec: the codec object * @reg: pseudo register - * @mask: bit mask to udpate + * @mask: bit mask to update * @val: value to update * * Returns zero if successful or a negative error code. -- 2.16.4

1 0

[alsa-devel] [PATCH -next] ALSA: pci: echoaudio: remove set but not used variable 'chip'
by YueHaibing 08 Jan '20

08 Jan '20

sound/pci/echoaudio/echoaudio.c: In function snd_echo_mixer_info: sound/pci/echoaudio/echoaudio.c:1233:20: warning: variable chip set but not used [-Wunused-but-set-variable] sound/pci/echoaudio/echoaudio.c: In function 'snd_echo_vmixer_info': sound/pci/echoaudio/echoaudio.c:1300:20: warning: variable 'chip' set but not used [-Wunused-but-set-variable] commit e67c3f0fd44c ("ALSA: pci: echoaudio: remove usage of dimen menber of elem_value structure") left behind this unused variable. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/pci/echoaudio/echoaudio.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 07fe0b3..0941a7a 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1230,9 +1230,6 @@ static const struct snd_kcontrol_new snd_echo_intput_nominal_level = { static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - struct echoaudio *chip; - - chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; @@ -1300,9 +1297,6 @@ static struct snd_kcontrol_new snd_echo_monitor_mixer = { static int snd_echo_vmixer_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - struct echoaudio *chip; - - chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; -- 2.7.4

2 1

Re: [alsa-devel] Tascam Coding Inquiry
by Takashi Sakamoto 08 Jan '20

08 Jan '20

Hi, On Wed, Jan 08, 2020 at 01:36:36AM -0700, The Dream Master Solo wrote: > Hello and good day to you. I am reaching out in regards to code I've > found online. I looks like you are the author. > [1]https://elixir.bootlin.com/linux/v5.5-rc5/source/sound/firewire/tascam > I am interested in using this code in creating a 64 bit driver for OSX > 10.13 or newer. I've just begun to learn coding a few weeks ago. Any > help that you can give me towards accomplishing this goal would be > greatly appreciated! It's my pleasure to help developers ;) You can freely utilize the idea written in the source of ALSA firewire-tascam driver with respect for the modern framework of copyright treaty and law. (This is one of my aim to publish the result of my reverse engineering work for the series of device; FW-1884, FW-1082, FW-1804 and FE-8.) I notice that it's better for you to have enough care of software licensing itself. The source of driver is licensed under GNU General Public License version 2.0 (GPLv2)[1]. Roughly speaking, when you just copy any code from the source of driver to your software and distribute your software to the others, your software should be licensed by GPLv2 or compatible ones because it includes lines in the source of driver. On the other hand, as long as you write your software by your hand at first place, you can freely license your software because the borrowing of idea is free from the license on the copyright treaty and law. Well, ALSA firewire-tascam driver uses controllers compliant to 1394 OHCI specification to communicate to the devices. For this purpose, the driver uses kernel implementation of Linux FireWire subsystem. I think that MacOS still includes the similar implementation and as a first step you start study for it. Additionally the specification of 1394 OHCI will help your work: * 1394 Open Host Controller Interface Specification Release 1.00 (October 20, 1997, Promoters of the 1394 Open HCI) * 1394 Open Host Controller Interface Specification Release 1.1 (January 6, 2000, Promoters of the 1394 Open HCI) [1] https://www.gnu.org/licenses/old-licenses/gpl-2.0.en.html Regards Takashi Sakamoto

1 0

[alsa-devel] Add quirk for Lenovo X1 Yoga
by Kailang 08 Jan '20

08 Jan '20

Hi Takashi, Attach patch was for Lenovo X1 Yoga. BR, Kailang

3 4

[alsa-devel] Set EAPD control to default for ALC222
by Kailang 08 Jan '20

08 Jan '20

Hi Takashi, Add Set EAPD control to default for ALC222. Many Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH] mixer: Fix memory leak for more than 16 file descriptor case
by David Fries 08 Jan '20

08 Jan '20

Signed-off-by: David Fries <David(a)Fries.net> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> --- Added signed off. I saw this when digging through the source code, wrote up a test case, and verified with valgrind. I don't have that many sound cards so I modified mixer.c to always take the conditional. alloca seemed like the least effort way to fix this. Other thoughts were to test pfds != spfds and free, but given how little bytes would be allocated putting on the stack seems reasonable. I also created the github pull request 20. https://github.com/alsa-project/alsa-lib/pull/20 src/mixer/mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 82056474..b1af9945 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -767,7 +767,7 @@ int snd_mixer_wait(snd_mixer_t *mixer, int timeout) if (count < 0) return count; if ((unsigned int) count > sizeof(spfds) / sizeof(spfds[0])) { - pfds = malloc(count * sizeof(*pfds)); + pfds = alloca(count * sizeof(*pfds)); if (!pfds) return -ENOMEM; err = snd_mixer_poll_descriptors(mixer, pfds, -- 2.11.0

2 1

[alsa-devel] [PATCH] ALSA: control: potential uninitialized return value
by Dan Carpenter 08 Jan '20

08 Jan '20

Smatch complains that "ret" might be uninitialized. Fixes: fbd3eb7f66c5 ("ALSA: control: Add verification for kctl accesses") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index d06033d418a8..aa0c0cf182af 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -896,7 +896,7 @@ static int sanity_check_elem_value(struct snd_card *card, u32 pattern) { size_t offset; - int i, ret; + int i, ret = 0; u32 *p; switch (info->type) { -- 2.11.0

2 1

[alsa-devel] [PATCH -next] ASoC: amd: acp3x: Fix return value check in acp3x_dai_probe()
by Wei Yongjun 08 Jan '20

08 Jan '20

In case of error, the function devm_ioremap() returns NULL pointer not ERR_PTR(). The IS_ERR() test in the return value check should be replaced with NULL test. Fixes: c9fe7db6e884 ("ASoC: amd: Refactoring of DAI from DMA driver") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/amd/raven/acp3x-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index d9b287b8396c..bf51cadf8682 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -321,8 +321,8 @@ static int acp3x_dai_probe(struct platform_device *pdev) } adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); - if (IS_ERR(adata->acp3x_base)) - return PTR_ERR(adata->acp3x_base); + if (!adata->acp3x_base) + return -ENOMEM; adata->i2s_irq = res->start; dev_set_drvdata(&pdev->dev, adata);

1 0

[alsa-devel] [PATCH 00/10] mfd: Add support for Analog Devices A2B transceiver
by Daniel Mack 08 Jan '20

08 Jan '20

This patch series adds support for Analog Device's AD242x A2B transceivers. https://www.analog.com/media/en/technical-documentation/user-guides/AD242x_… These transceivers are used to form an audio network by connecting the parts in a daisy-chain. On top of audio, the devices expose some other functions such as GPIO, programmable clock outputs and remote-side I2C bus master. The first node in the chain is called the master node, and all other devices are called slave nodes. Up to 15 such devices can be connected this way. The master device responds on two addresses on the I2C bus. The primary one is used to access all registers in the master node itself, the secondary is for accessing remote nodes after prior setup through the master node. In the driver stack, these details are hidden behind specific regmap configs. The driver stack is implemented as MFD core and companion drivers that can be registered as sub-devices in DT. Drivers for these sub-devices can be used for both master and slave nodes, as they just interface with the node's regmap. The master node is responsible for discovering all the slave nodes at probe time, and it needs to take the used audio and routing modes in each of the slave devices into account in order to pre-calculate the bus-timings correctly. Hence, this bus is not hot-pluggable. Transceivers can both receive and provide audio, and streams can be routed from one node to any other, including many others. The tricky bit is how to expose the audio routing in DT in a sane way. The way it is implemented here, the slave nodes specify the number of slots they each consume and generate, and which thereof they forward from one side to the other. This mimics the internal register structure and should allow for even exotic setups. Please let me know what you think and what could be improved. Thanks, Daniel Daniel Mack (10): dt-bindings: mfd: Add documentation for ad242x dt-bindings: i2c: Add documentation for ad242x i2c controllers dt-bindings: gpio: Add documentation for AD242x GPIO controllers dt-bindings: clock: Add documentation for AD242x clock providers dt-bindings: sound: Add documentation for AD242x codecs mfd: Add core driver for AD242x A2B transceivers i2c: Add driver for AD242x bus controller gpio: Add driver for AD242x GPIO controllers clk: Add support for AD242x clock output providers ASoC: Add codec component for AD242x nodes .../bindings/clock/adi,ad242x-clk.yaml | 32 + .../bindings/gpio/adi,ad242x-gpio.yaml | 65 ++ .../bindings/i2c/adi,ad242x-i2c.yaml | 31 + .../bindings/mfd/adi,ad242x-bus.yaml | 29 + .../bindings/mfd/adi,ad242x-master.yaml | 235 +++++++ .../bindings/mfd/adi,ad242x-slave.yaml | 108 ++++ .../bindings/sound/adi,ad242x-codec.yaml | 31 + drivers/clk/Kconfig | 6 + drivers/clk/Makefile | 1 + drivers/clk/clk-ad242x.c | 231 +++++++ drivers/gpio/Kconfig | 6 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-ad242x.c | 229 +++++++ drivers/i2c/busses/Kconfig | 10 + drivers/i2c/busses/Makefile | 1 + drivers/i2c/busses/i2c-ad242x.c | 178 +++++ drivers/mfd/Kconfig | 11 + drivers/mfd/Makefile | 1 + drivers/mfd/ad242x-bus.c | 42 ++ drivers/mfd/ad242x-master.c | 611 ++++++++++++++++++ drivers/mfd/ad242x-node.c | 262 ++++++++ drivers/mfd/ad242x-slave.c | 234 +++++++ include/dt-bindings/clock/adi,ad242x.h | 9 + include/linux/mfd/ad242x.h | 400 ++++++++++++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad242x.c | 338 ++++++++++ 27 files changed, 3109 insertions(+) create mode 100644 Documentation/devicetree/bindings/clock/adi,ad242x-clk.yaml create mode 100644 Documentation/devicetree/bindings/gpio/adi,ad242x-gpio.yaml create mode 100644 Documentation/devicetree/bindings/i2c/adi,ad242x-i2c.yaml create mode 100644 Documentation/devicetree/bindings/mfd/adi,ad242x-bus.yaml create mode 100644 Documentation/devicetree/bindings/mfd/adi,ad242x-master.yaml create mode 100644 Documentation/devicetree/bindings/mfd/adi,ad242x-slave.yaml create mode 100644 Documentation/devicetree/bindings/sound/adi,ad242x-codec.yaml create mode 100644 drivers/clk/clk-ad242x.c create mode 100644 drivers/gpio/gpio-ad242x.c create mode 100644 drivers/i2c/busses/i2c-ad242x.c create mode 100644 drivers/mfd/ad242x-bus.c create mode 100644 drivers/mfd/ad242x-master.c create mode 100644 drivers/mfd/ad242x-node.c create mode 100644 drivers/mfd/ad242x-slave.c create mode 100644 include/dt-bindings/clock/adi,ad242x.h create mode 100644 include/linux/mfd/ad242x.h create mode 100644 sound/soc/codecs/ad242x.c -- 2.23.0

8 35

[alsa-devel] [PATCH 0/7] ALSA: hda: Enable HDAudio compress
by Cezary Rojewski 07 Jan '20

07 Jan '20

This set of patches provides new interfaces - page allocation - and runtime flow adjustments - PM support - for compress operations. For HDA part, work has been done to account for compress streams when servicing IRQs, setting up BDLs and assigning DMAs. End goal is to make room for one of DSP debug features: data probing. It takes advantage of compress streams when extracting data from running audio pipeline. Initial review and development of probes can be found under: https://github.com/thesofproject/linux/pull/1276 with this very set of patches being separated and reviewed on: https://github.com/thesofproject/linux/pull/1571 Cezary Rojewski (7): ALSA: hda: Allow for compress stream to hdac_ext_stream assignment ALSA: hda: Prepare for compress stream support ALSA: hda: Interrupt servicing and BDL setup for compress streams ALSA: core: Expand DMA buffer information ALSA: core: Implement compress page allocation and free routines ASoC: compress: Add pm_runtime support ASoC: SOF: Intel: Account for compress streams when servicing IRQs include/sound/compress_driver.h | 40 ++++++++++++++++++------ include/sound/hdaudio.h | 2 ++ include/sound/hdaudio_ext.h | 2 ++ sound/core/compress_offload.c | 42 ++++++++++++++++++++++++++ sound/hda/ext/hdac_ext_stream.c | 46 +++++++++++++++++++++++++--- sound/hda/hdac_controller.c | 4 +-- sound/hda/hdac_stream.c | 52 ++++++++++++++++++++------------ sound/soc/soc-compress.c | 29 +++++++++++++++++- sound/soc/sof/intel/hda-stream.c | 26 ++++++++++++++-- 9 files changed, 205 insertions(+), 38 deletions(-) -- 2.17.1

4 18

[alsa-devel] [PATCH] mixer: Fix memory leak for more than 16 file descriptor case
by David Fries 07 Jan '20

07 Jan '20

--- I saw this when digging through the source code, wrote up a test case, and verified with valgrind. I don't have that many sound cards so I modified mixer.c to always take the conditional. alloca seemed like the least effort way to fix this. Other thoughts were to test pfds != spfds and free, but given how little bytes would be allocated putting on the stack seems reasonable. I also created the github pull request 20. https://github.com/alsa-project/alsa-lib/pull/20 src/mixer/mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 82056474..b1af9945 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -767,7 +767,7 @@ int snd_mixer_wait(snd_mixer_t *mixer, int timeout) if (count < 0) return count; if ((unsigned int) count > sizeof(spfds) / sizeof(spfds[0])) { - pfds = malloc(count * sizeof(*pfds)); + pfds = alloca(count * sizeof(*pfds)); if (!pfds) return -ENOMEM; err = snd_mixer_poll_descriptors(mixer, pfds, -- 2.11.0

2 2

[alsa-devel] [PATCH alsa-utils 00/11] alsa-info.sh: Improve output and fix file upload issues
by David Ward 07 Jan '20

07 Jan '20

David Ward (11): alsa-info.sh: Consolidate PCI device output alsa-info.sh: Read from /proc/modules and sort the result alsa-info.sh: Simplify iteration over cards when calling amixer alsa-info.sh: Use existing function to print ALSA configuration files alsa-info.sh: Exit script after writing information to stdout alsa-info.sh: Replace gauge with infobox for upload dialog alsa-info.sh: Remove progress spinner during upload without dialog alsa-info.sh: Condense nested commands for file upload alsa-info.sh: Condense nested commands for formatting upload result alsa-info.sh: Perform test for wget earlier alsa-info.sh: Warn after actual upload failure; do not ping server alsa-info/alsa-info.sh | 251 ++++++++++++++++++------------------------------- 1 file changed, 90 insertions(+), 161 deletions(-) -- 1.8.3.1

2 12

[alsa-devel] [PATCH] pinctrl: lochnagar: select GPIOLIB
by Arnd Bergmann 07 Jan '20

07 Jan '20

In a rare randconfig build I came across one configuration that does not enable CONFIG_GPIOLIB, which is needed by lochnagar: ERROR: "devm_gpiochip_add_data" [drivers/pinctrl/cirrus/pinctrl-lochnagar.ko] undefined! ERROR: "gpiochip_generic_free" [drivers/pinctrl/cirrus/pinctrl-lochnagar.ko] undefined! ERROR: "gpiochip_generic_request" [drivers/pinctrl/cirrus/pinctrl-lochnagar.ko] undefined! ERROR: "gpiochip_get_data" [drivers/pinctrl/cirrus/pinctrl-lochnagar.ko] undefined! Add another 'select' like all other pinctrl drivers have. Fixes: 0548448b719a ("pinctrl: lochnagar: Add support for the Cirrus Logic Lochnagar") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- I wonder if GPIOLIB should just become mandatory when enabling the pinctrl subsystem, or if there are still good reasons for leaving it disabled on any machine that uses CONFIG_PINCTRL. --- drivers/pinctrl/cirrus/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/pinctrl/cirrus/Kconfig b/drivers/pinctrl/cirrus/Kconfig index f1806fd781a0..530426a74f75 100644 --- a/drivers/pinctrl/cirrus/Kconfig +++ b/drivers/pinctrl/cirrus/Kconfig @@ -2,6 +2,7 @@ config PINCTRL_LOCHNAGAR tristate "Cirrus Logic Lochnagar pinctrl driver" depends on MFD_LOCHNAGAR + select GPIOLIB select PINMUX select PINCONF select GENERIC_PINCONF -- 2.20.0

4 6

[alsa-devel] [PATCH] ASoC: rt5677: add SPI_MASTER dependency
by Arnd Bergmann 07 Jan '20

07 Jan '20

When CONFIG_SPI is disabled, the newly added code for the DSP firmware loading fails to link: ERROR: "rt5677_spi_hotword_detected" [sound/soc/codecs/snd-soc-rt5677.ko] undefined! ERROR: "rt5677_spi_write" [sound/soc/codecs/snd-soc-rt5677.ko] undefined! Add a dependency to prevent this configuration. Note: the does not work with the DT probing, as there is no binding for the SPI half of the driver, but nothing seems to be using that with the mainline kernel anyway. Fixes: 461c623270e4 ("ASoC: rt5677: Load firmware via SPI using delayed work") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/Kconfig | 1 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/mediatek/Kconfig | 2 +- sound/soc/tegra/Kconfig | 2 +- 4 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 71b7286d14f2..8e9c461a84f8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1047,6 +1047,7 @@ config SND_SOC_RT5670 config SND_SOC_RT5677 tristate + depends on SPI_MASTER select REGMAP_I2C select REGMAP_IRQ diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index ef20316e83d1..da7f9111f3d3 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -34,6 +34,7 @@ if SND_SOC_INTEL_HASWELL || SND_SOC_SOF_BROADWELL config SND_SOC_INTEL_BDW_RT5677_MACH tristate "Broadwell with RT5677 codec" depends on I2C + depends on SPI_MASTER depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST depends on GPIOLIB || COMPILE_TEST depends on X86_INTEL_LPSS || COMPILE_TEST diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index a656d2014127..4699787c93ed 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -97,7 +97,7 @@ config SND_SOC_MT8173_RT5650_RT5514 config SND_SOC_MT8173_RT5650_RT5676 tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs" - depends on SND_SOC_MT8173 && I2C + depends on SND_SOC_MT8173 && I2C && SPI_MASTER select SND_SOC_RT5645 select SND_SOC_RT5677 select SND_SOC_HDMI_CODEC diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index addadc827b91..df36e84c0116 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -122,7 +122,7 @@ config SND_SOC_TEGRA_MAX98090 config SND_SOC_TEGRA_RT5677 tristate "SoC Audio support for Tegra boards using a RT5677 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on SND_SOC_TEGRA && I2C && GPIOLIB && SPI_MASTER select SND_SOC_RT5677 help Say Y or M here if you want to add support for SoC audio on Tegra -- 2.20.0

4 4

[alsa-devel] [PATCH v5 00/11] ASoC: Add support to WCD9340/WCD9341 codec
by Srinivas Kandagatla 07 Jan '20

07 Jan '20

This patchset adds support to Qualcomm WCD9340/WCD9341 Codec which is a standalone Hi-Fi audio codec IC. This codec supports both I2S/I2C and SLIMbus audio interfaces. On slimbus interface it supports two data lanes; 16 Tx ports and 8 Rx ports. It has Five DACs and seven dedicated interpolators, Multibutton headset control (MBHC), Active noise cancellation, Sidetone paths, MAD (mic activity detection) and codec processing engine. It supports Class-H differential earpiece out and stereo single ended headphones out. This codec also has integrated SoundWire controller. Patchset for this is already sent for review at https://patchwork.kernel.org/cover/11185769/ This patchset has been tested on SDM845 based DragonBoard DB845c and Lenovo Yoga C630 laptop with WSA881x smart speaker amplifiers via soundwire and 4 DMICs. gpio controller patch does not have any link dependency, it can go by its own. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. Thanks, srini Changes since v4: - updated microvolt bindings as suggested by Rob. - slimbus optional property patch already applied. Srinivas Kandagatla (11): ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec mfd: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add basic controls ASoC: wcd934x: add playback dapm widgets ASoC: wcd934x: add capture dapm widgets ASoC: wcd934x: add audio routings dt-bindings: gpio: wcd934x: Add bindings for gpio gpio: wcd934x: Add support to wcd934x gpio controller ASoC: qcom: dt-bindings: Add compatible for DB845c and Lenovo Yoga ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga .../bindings/gpio/qcom,wcd934x-gpio.yaml | 47 + .../devicetree/bindings/sound/qcom,sdm845.txt | 5 +- .../bindings/sound/qcom,wcd934x.yaml | 175 + drivers/gpio/Kconfig | 8 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-wcd934x.c | 104 + drivers/mfd/Kconfig | 12 + drivers/mfd/Makefile | 1 + drivers/mfd/wcd934x.c | 306 + include/linux/mfd/wcd934x/registers.h | 531 ++ include/linux/mfd/wcd934x/wcd934x.h | 31 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd934x.c | 5084 +++++++++++++++++ sound/soc/qcom/sdm845.c | 86 +- 15 files changed, 6401 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/gpio/qcom,wcd934x-gpio.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml create mode 100644 drivers/gpio/gpio-wcd934x.c create mode 100644 drivers/mfd/wcd934x.c create mode 100644 include/linux/mfd/wcd934x/registers.h create mode 100644 include/linux/mfd/wcd934x/wcd934x.h create mode 100644 sound/soc/codecs/wcd934x.c -- 2.21.0

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[alsa-devel] Right place for defining ACPI GPIO mapping for codec
by Yauhen Kharuzhy 06 Jan '20

06 Jan '20

Hello, I am working now to get sound working at Lenovo Yoga Book tablet. It is Intel CherryTrail-based device and has RT5677 sound codec. The RT5677 codec driver uses two GPIOs to reset and enable the codec, with names 'realtek,reset' and 'realtek,pow-ldo2'. The ACPI definition lacks a _DSD section with GPIO name->CRS ID definition, so I need to manually define this mapping somewhere using existing devm_acpi_dev_add_driver_gpios() method (various devices has various _CRS definitions order and content, so this is cannot be placed into rt5677.c codec driver). The most obvious place for this is ASoC machine driver for my device, but the codec driver is binded to the ACPI device and initializes before machine driver. Can somebody advice how to define such GPIOs mapping for codec in my case? The ACPI definition of sound device is below. Second I2C address definition is for TS3A277E jack detector. The first GPIO in the _CRS is codec reset, the second is pow-ldo2. Device (RTEK) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "10EC5677") // _HID: Hardware ID Name (_CID, "10EC5677") // _CID: Compatible ID Name (_DDN, "Realtek IIS Audio Codec") // _DDN: DOS Device Name Name (_SUB, "17AA7005") // _SUB: Subsystem ID Name (_UID, One) // _UID: Unique ID Name (_PR0, Package (0x01) // _PR0: Power Resources for D0 { CLK3 }) Name (CHAN, Package (0x02) { One, 0x0124F800 }) Method (_CRS, 0, NotSerialized) // _CRS: Current Resource Settings { Name (SBUF, ResourceTemplate () { I2cSerialBusV2 (0x002C, ControllerInitiated, 0x000186A0, AddressingMode7Bit, "\\_SB.PCI0.I2C1", 0x00, ResourceConsumer, , Exclusive, ) I2cSerialBusV2 (0x003B, ControllerInitiated, 0x000186A0, AddressingMode7Bit, "\\_SB.PCI0.I2C1", 0x00, ResourceConsumer, , Exclusive, ) GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.GPO3", 0x00, ResourceConsumer, , ) { // Pin list 0x0019 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.GPO3", 0x00, ResourceConsumer, , ) { // Pin list 0x0012 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.GPO3", 0x00, ResourceConsumer, , ) { // Pin list 0x0030 } GpioInt (Edge, ActiveLow, Exclusive, PullNone, 0x0000, "\\_SB.GPO0", 0x00, ResourceConsumer, , ) { // Pin list 0x005B } GpioInt (Edge, ActiveLow, Exclusive, PullNone, 0x0000, "\\_SB.GPO0", 0x00, ResourceConsumer, , ) { // Pin list 0x004D } SpiSerialBusV2 (0x0001, PolarityLow, FourWireMode, 0x08, ControllerInitiated, 0x003D0900, ClockPolarityHigh, ClockPhaseSecond, "\\_SB.PCI0.SPI1", 0x00, ResourceConsumer, , Exclusive, ) }) Return (SBUF) /* \_SB_.PCI0.I2C1.RTEK._CRS.SBUF */ } Method (_STA, 0, NotSerialized) // _STA: Status { Return (0x0F) } Method (_DIS, 0, NotSerialized) // _DIS: Disable Device { } } -- Yauhen Kharuzhy

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[alsa-devel] mixer: Fix memory leak for more than 16 file descriptor case
by GitHub pull_request - opened 06 Jan '20

06 Jan '20

alsa-project/alsa-lib pull request #20 was opened from dfries: I saw this when digging through the source code, wrote up a test case, and verified with valgrind. I don't have that many sound cards so I modified mixer.c to always take the conditional. alloca seemed like the least effort way to fix this. Other thoughts were to test pfds != spfds and free, but given how little bytes would be allocated putting on the stack seems reasonable. Request URL : https://github.com/alsa-project/alsa-lib/pull/20 Patch URL : https://github.com/alsa-project/alsa-lib/pull/20.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [PATCH v2] ASoC: rt5640: Fix NULL dereference on module unload
by Dmitry Osipenko 06 Jan '20

06 Jan '20

The rt5640->jack is NULL if jack is already disabled at the time of driver's module unloading. Signed-off-by: Dmitry Osipenko <digetx(a)gmail.com> --- Changelog: v2: - Addressed review comment from Takashi Iwai by making patch's diff a bit cleaner and simpler. - Addressed review comment from Mark Brown by removing noisy backtrace from the commit's message and moving it after --- in order to keep it discoverable in the ML archive. Backtrace: Unable to handle kernel NULL pointer dereference at virtual address 00000024 ... (rt5640_set_jack [snd_soc_rt5640]) (snd_soc_component_set_jack [snd_soc_core]) (soc_remove_component [snd_soc_core]) (soc_cleanup_card_resources [snd_soc_core]) (snd_soc_unregister_card [snd_soc_core]) (tegra_rt5640_remove [snd_soc_tegra_rt5640]) (platform_drv_remove) (device_release_driver_internal) (driver_detach) (bus_remove_driver) (tegra_rt5640_driver_exit (sys_delete_module) sound/soc/codecs/rt5640.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index adbae1f36a8a..747ca248bf10 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2432,6 +2432,13 @@ static void rt5640_disable_jack_detect(struct snd_soc_component *component) { struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); + /* + * soc_remove_component() force-disables jack and thus rt5640->jack + * could be NULL at the time of driver's module unloading. + */ + if (!rt5640->jack) + return; + disable_irq(rt5640->irq); rt5640_cancel_work(rt5640); -- 2.24.0

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[alsa-devel] [PATCH] ASoC: Intel: bytcht_es8316: Fix Irbis NB41 netbook quirk
by Hans de Goede 06 Jan '20

06 Jan '20

When a quirk for the Irbis NB41 netbook was added, to override the defaults for this device, I forgot to add/keep the BYT_CHT_ES8316_SSP0 part of the defaults, completely breaking audio on this netbook. This commit adds the BYT_CHT_ES8316_SSP0 flag to the Irbis NB41 netbook quirk, making audio work again. Cc: stable(a)vger.kernel.org Cc: russianneuromancer(a)ya.ru Fixes: aa2ba991c420 ("ASoC: Intel: bytcht_es8316: Add quirk for Irbis NB41 netbook") Reported-and-tested-by: russianneuromancer(a)ya.ru Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcht_es8316.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 46612331f5ea..54e97455d7f6 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -442,7 +442,8 @@ static const struct dmi_system_id byt_cht_es8316_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "IRBIS"), DMI_MATCH(DMI_PRODUCT_NAME, "NB41"), }, - .driver_data = (void *)(BYT_CHT_ES8316_INTMIC_IN2_MAP + .driver_data = (void *)(BYT_CHT_ES8316_SSP0 + | BYT_CHT_ES8316_INTMIC_IN2_MAP | BYT_CHT_ES8316_JD_INVERTED), }, { /* Teclast X98 Plus II */ -- 2.24.1

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[alsa-devel] [PATCH v3] ASoC: Intel: sof_rt5682: Ignore the speaker amp when there isn't one.
by Sam McNally 06 Jan '20

06 Jan '20

Some members of the Google_Hatch family include a rt5682 jack codec, but no speaker amplifier. This uses the same driver (sof_rt5682) as a combination of rt5682 jack codec and max98357a speaker amplifier. Within the sof_rt5682 driver, these cases are not currently distinguishable, relying on a DMI quirk to decide the configuration. This causes an incorrect configuration when only the rt5682 is present on a Google_Hatch device. For CML, the jack codec is used as the primary key when matching, with a possible speaker amplifier described in quirk_data. The two cases of interest are the second and third 10EC5682 entries in snd_soc_acpi_intel_cml_machines[]. The second entry matches the combination of rt5682 and max98357a, resulting in the quirk_data field in the snd_soc_acpi_mach being non-null, pointing at max98357a_spk_codecs, the snd_soc_acpi_codecs for the matched speaker amplifier. The third entry matches just the rt5682, resulting in a null quirk_data. The sof_rt5682 driver's DMI data matching identifies that a speaker amplifier is present for all Google_Hatch family devices. Detect cases where there is no speaker amplifier by checking for a null quirk_data in the snd_soc_acpi_mach and remove the speaker amplifier bit in that case. Signed-off-by: Sam McNally <sammc(a)chromium.org> --- Changes in v3: - Rewrote commit message to refer to correct snd_soc_acpi_intel_cml_machines[] entries and better describe the change Changes in v2: - Added details about the relevant ACPI matches to the description sound/soc/intel/boards/sof_rt5682.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index ad8a2b4bc709..8a13231dee15 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -603,6 +603,14 @@ static int sof_audio_probe(struct platform_device *pdev) dmi_check_system(sof_rt5682_quirk_table); + mach = (&pdev->dev)->platform_data; + + /* A speaker amp might not be present when the quirk claims one is. + * Detect this via whether the machine driver match includes quirk_data. + */ + if ((sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) + sof_rt5682_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -663,7 +671,6 @@ static int sof_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); sof_audio_card_rt5682.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_rt5682, -- 2.24.1.735.g03f4e72817-goog

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[alsa-devel] [PATCH] ASoC: bdw-rt5677: add spi driver compile switches
by Curtis Malainey 06 Jan '20

06 Jan '20

If the SPI driver is not compiled in then the RT5677_SPI driver will not be included as well which will cause the bdw-rt5677 driver to fail to probe since the DSP components are managed by the RT5677_SPI driver. The solution is to remove them when the driver is not present as part of the build. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> CC: Pan Xiuli <xiuli.pan(a)linux.intel.com> --- sound/soc/intel/boards/bdw-rt5677.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 2af8e5a62da84..d5ded2c545d2c 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -74,11 +74,13 @@ static const struct snd_soc_dapm_route bdw_rt5677_map[] = { /* CODEC BE connections */ {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +#if IS_ENABLED(CONFIG_SND_SOC_RT5677_SPI) {"DSP Capture", NULL, "DSP Buffer"}, /* DSP Clock Connections */ { "DSP Buffer", NULL, "SSP0 CODEC IN" }, { "SSP0 CODEC IN", NULL, "DSPTX" }, +#endif }; static const struct snd_kcontrol_new bdw_rt5677_controls[] = { @@ -319,6 +321,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { SND_SOC_DAILINK_REG(fe, dummy, platform), }, +#if IS_ENABLED(CONFIG_SND_SOC_RT5677_SPI) /* Non-DPCM links */ { .name = "Codec DSP", @@ -326,6 +329,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { .ops = &bdw_rt5677_dsp_ops, SND_SOC_DAILINK_REG(dsp), }, +#endif /* Back End DAI links */ { -- 2.24.1.735.g03f4e72817-goog

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[alsa-devel] [GIT PULL] ASoC fixes for v5.5
by Mark Brown 06 Jan '20

06 Jan '20

The following changes since commit 556672d75ff486e0b6786056da624131679e0576: ASoC: wm8962: fix lambda value (2019-12-16 11:47:28 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.5-rc5 for you to fetch changes up to ff3b57417012fcc963ec281f5705bed837e4b1ac: ASoC: Intel: boards: Fix compile-testing RT1011/RT5682 (2020-01-03 00:56:47 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.5 More fixes that have been collected, nothing super remarkable here - the few core fixes are mainly error handling related as are many of the driver fixes. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: Intel: boards: Fix compile-testing RT1011/RT5682 Chuhong Yuan (1): ASoC: fsl_audmix: add missed pm_runtime_disable Colin Ian King (1): ASoC: SOF: imx8: fix memory allocation failure check on priv->pd_dev Daniel Baluta (2): ASoC: soc-core: Set dpcm_playback / dpcm_capture ASoC: SOF: imx8: Fix dsp_box offset Dragos Tarcatu (1): ASoC: topology: Prevent use-after-free in snd_soc_get_pcm_runtime() Kai Vehmanen (1): ASoC: SOF: fix fault at driver unload after failed probe Olivier Moysan (3): ASoC: stm32: spdifrx: fix inconsistent lock state ASoC: stm32: spdifrx: fix race condition in irq handler ASoC: stm32: spdifrx: fix input pin state management Pierre-Louis Bossart (1): ASoC: SOF: Intel: hda: hda-dai: fix oops on hda_link .hw_free Takashi Iwai (1): ASoC: core: Fix access to uninitialized list heads sound/soc/fsl/fsl_audmix.c | 9 ++++++- sound/soc/intel/boards/cml_rt1011_rt5682.c | 1 - sound/soc/soc-core.c | 14 ++++++----- sound/soc/soc-topology.c | 6 ++--- sound/soc/sof/imx/imx8.c | 5 +++- sound/soc/sof/intel/hda-dai.c | 11 ++++++-- sound/soc/sof/ipc.c | 3 +++ sound/soc/stm/stm32_spdifrx.c | 40 +++++++++++++++++++----------- 8 files changed, 61 insertions(+), 28 deletions(-)

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[alsa-devel] [PATCH] slimbus: Use the correct style for SPDX License Identifier
by Nishad Kamdar 06 Jan '20

06 Jan '20

This patch corrects the SPDX License Identifier style in header file related to SLIMbus driver. For C header files Documentation/process/license-rules.rst mandates C-like comments (opposed to C source files where C++ style should be used). Changes made by using a script provided by Joe Perches here: https://lkml.org/lkml/2019/2/7/46. Suggested-by: Joe Perches <joe(a)perches.com> Signed-off-by: Nishad Kamdar <nishadkamdar(a)gmail.com> --- drivers/slimbus/slimbus.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index b2f013bfe42e..c73035915f1d 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0 +/* SPDX-License-Identifier: GPL-2.0 */ /* * Copyright (c) 2011-2017, The Linux Foundation */ -- 2.17.1

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