May 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: tlv320adcx140: Fix warnings when using W=1
by Dan Murphy 27 May '20

27 May '20

Fix the warnings when using the W=1 compiler flag. sound/soc/codecs/tlv320adcx140.c: In function ‘adcx140_reset’: sound/soc/codecs/tlv320adcx140.c:570:6: warning: variable ‘ret’ set but not used [-Wunused-but-set-variable] 570 | int ret = 0; | ^~~ This was set but only used in case where the reset GPIO is not defined. Have the function return the value of ret. sound/soc/codecs/tlv320adcx140.c: In function ‘adcx140_codec_probe’: sound/soc/codecs/tlv320adcx140.c:778:18: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] 778 | if (bias_source < ADCX140_MIC_BIAS_VAL_VREF || | ^ sound/soc/codecs/tlv320adcx140.c:789:18: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] 789 | if (vref_source < ADCX140_MIC_BIAS_VREF_275V || This condition will not occur since if the dt property is not set then the *_source variable is set to the default value. So there is no way that *_source can be less then 0. Which is what each #define is set to. The code just needs to make sure that the dt property is not out of the upper bounds. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 472d759ba8a3..2fe0df3b7550 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -582,7 +582,7 @@ static int adcx140_reset(struct adcx140_priv *adcx140) /* 8.4.2: wait >= 10 ms after entering sleep mode. */ usleep_range(10000, 100000); - return 0; + return ret; } static int adcx140_hw_params(struct snd_pcm_substream *substream, @@ -772,8 +772,7 @@ static int adcx140_codec_probe(struct snd_soc_component *component) if (ret) bias_source = ADCX140_MIC_BIAS_VAL_VREF; - if (bias_source < ADCX140_MIC_BIAS_VAL_VREF || - bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { + if (bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); return -EINVAL; } @@ -783,8 +782,7 @@ static int adcx140_codec_probe(struct snd_soc_component *component) if (ret) vref_source = ADCX140_MIC_BIAS_VREF_275V; - if (vref_source < ADCX140_MIC_BIAS_VREF_275V || - vref_source > ADCX140_MIC_BIAS_VREF_1375V) { + if (vref_source > ADCX140_MIC_BIAS_VREF_1375V) { dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); return -EINVAL; } -- 2.26.2

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[PATCH] ASoC: wm8962: Fix runtime PM imbalance on error
by Dinghao Liu 27 May '20

27 May '20

pm_runtime_get_sync() increments the runtime PM usage counter even the call returns an error code. Thus a pairing decrement is needed on the error handling path to keep the counter balanced. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/soc/codecs/wm8962.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d9d59f45833f..bb806b195d10 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2881,6 +2881,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s ret = pm_runtime_get_sync(component->dev); if (ret < 0) { + pm_runtime_put_noidle(component->dev); dev_err(component->dev, "Failed to resume device: %d\n", ret); return ret; } @@ -3013,6 +3014,7 @@ static irqreturn_t wm8962_irq(int irq, void *data) ret = pm_runtime_get_sync(dev); if (ret < 0) { + pm_runtime_put_noidle(dev); dev_err(dev, "Failed to resume: %d\n", ret); return IRQ_NONE; } -- 2.17.1

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[PATCH 0/8] ASoC: SOF: finalize Baytrail/CherryTrail support
by Pierre-Louis Bossart 27 May '20

27 May '20

This long-overdue patchset adds missing system suspend/resume support and hardens the IPC to solve module load/unload issues on specific devices such as Cyan Chromebook. With this series SOF is finally iso-feature with the legacy driver. Thanks to Ranjani and Keyon for the basic patches and Enric for testing. The last part needed for Baytrail/Cherrytrail is simplification of the driver/card names and turn-key UCM support (on-going work with Jaroslav). Keyon Jie (1): ASoC: SOF: Intel: BYT: harden IPC initialization and handling Pierre-Louis Bossart (3): ASoC: Intel: bytcr_rt5640/51: remove .ignore_suspend ASoC: Intel: byt/cht: add .pm_ops ASoC: SOF: Intel: BYT: mask BUSY or DONE interrupts in handler Ranjani Sridharan (4): ASoC: SOF: Intel: byt: Add PM callbacks ASoC: SOF: pm: handle resume on legacy Intel platforms ASoC: SOF: ipc: ignore DSP replies received when they are not expected ASoC: SOF: Intel: BYT: add .remove op sound/soc/intel/boards/bytcht_cx2072x.c | 3 + sound/soc/intel/boards/bytcht_da7213.c | 3 + sound/soc/intel/boards/bytcht_es8316.c | 3 + sound/soc/intel/boards/bytcr_rt5640.c | 7 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 3 + sound/soc/intel/boards/cht_bsw_nau8824.c | 3 + sound/soc/intel/boards/cht_bsw_rt5645.c | 3 + sound/soc/intel/boards/cht_bsw_rt5672.c | 3 + sound/soc/sof/intel/byt.c | 116 ++++++++++++++----- sound/soc/sof/ipc.c | 9 +- sound/soc/sof/pm.c | 8 +- sound/soc/sof/sof-priv.h | 2 +- 13 files changed, 122 insertions(+), 47 deletions(-) base-commit: 0d71a5cf691a8226151ceeb79fb872925f053df5 -- 2.20.1

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[PATCH] ASoC: dapm: Move dai_link widgets to runtime to fix use after free
by Charles Keepax 27 May '20

27 May '20

The newly added CODEC to CODEC DAI link widget pointers in snd_soc_dai_link are better placed in snd_soc_pcm_runtime. snd_soc_dai_link is really intended for static configuration of the DAI, and the runtime for dynamic data. The snd_soc_dai_link structures are not destroyed if the card is unbound. The widgets are cleared up on unbind, however if the card is rebound as the snd_soc_dai_link structures are reused these pointers will be left at their old values, causing access to freed memory. Fixes: 595571cca4de ("ASoC: dapm: Fix regression introducing multiple copies of DAI widgets") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/soc.h | 6 +++--- sound/soc/soc-dapm.c | 12 ++++++------ 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a7fa642601081..ba07cdfcdcee4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -801,9 +801,6 @@ struct snd_soc_dai_link { const struct snd_soc_pcm_stream *params; unsigned int num_params; - struct snd_soc_dapm_widget *playback_widget; - struct snd_soc_dapm_widget *capture_widget; - unsigned int dai_fmt; /* format to set on init */ enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ @@ -1169,6 +1166,9 @@ struct snd_soc_pcm_runtime { unsigned int num_codecs; unsigned int num_cpus; + struct snd_soc_dapm_widget *playback_widget; + struct snd_soc_dapm_widget *capture_widget; + struct delayed_work delayed_work; void (*close_delayed_work_func)(struct snd_soc_pcm_runtime *rtd); #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a4de3e4bc2eff..fc40f283a2480 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4340,16 +4340,16 @@ static void dapm_connect_dai_pair(struct snd_soc_card *card, codec = codec_dai->playback_widget; if (playback_cpu && codec) { - if (dai_link->params && !dai_link->playback_widget) { + if (dai_link->params && !rtd->playback_widget) { substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; dai = snd_soc_dapm_new_dai(card, substream, "playback"); if (IS_ERR(dai)) goto capture; - dai_link->playback_widget = dai; + rtd->playback_widget = dai; } dapm_connect_dai_routes(&card->dapm, cpu_dai, playback_cpu, - dai_link->playback_widget, + rtd->playback_widget, codec_dai, codec); } @@ -4358,16 +4358,16 @@ static void dapm_connect_dai_pair(struct snd_soc_card *card, codec = codec_dai->capture_widget; if (codec && capture_cpu) { - if (dai_link->params && !dai_link->capture_widget) { + if (dai_link->params && !rtd->capture_widget) { substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; dai = snd_soc_dapm_new_dai(card, substream, "capture"); if (IS_ERR(dai)) return; - dai_link->capture_widget = dai; + rtd->capture_widget = dai; } dapm_connect_dai_routes(&card->dapm, codec_dai, codec, - dai_link->capture_widget, + rtd->capture_widget, cpu_dai, capture_cpu); } } -- 2.11.0

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Re: [PATCH 1/2] dt-bindings: ASoC: renesas,rsnd: Add r8a7742 support
by Mark Brown 27 May '20

27 May '20

On Wed, May 27, 2020 at 12:32:28PM +0100, Lad, Prabhakar wrote: > On Wed, May 27, 2020 at 12:25 PM Mark Brown <broonie(a)kernel.org> wrote: > > > + - "renesas,rcar_sound-r8a7742" (RZ/G1H) > > I'd expect a matching patch adding this compatible to the driver. > The Renesas R-Car sound for RZ/G1H is identical to the R-Car Gen2 > family. So no driver change is needed and the fallback compatible > value "renesas,rcar_sound-gen2" will be used in the SOC DT. Then the patch to add the compatible string will be trivial.

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Add new codec supported for ALC287
by Kailang 27 May '20

27 May '20

Hi Takashi, Attach patch was new codec supported. Many Thanks. BR, Kailang

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Alsa Midi Driver Question/Issue
by liebrecht＠grossmann-venter.com 27 May '20

27 May '20

This is not really a straight forward user question and thereby I post it in the developer forum. My Problem: I use Alsa/Jack/Pulseaudio to run quite an extensive audio interface to DAW with involved Jack routings of both Audio and Midi. The system runs like greased lightning and I can do things on Linux that I could never do on windows period. The problem is that there exist applicatio9ns that is not ALSA/JACK Midi friendly. One such Application is Bitwig for Linux 1) Bitwig absolutely requires to have exclusive access to a USB pad Controller in order to recognise the controller. 2) We all know that the Alsa Midi driver forwards ALL Midi devices it finds to Jack where the aj2midid daemon provides a bridge to jack and the midi devices are then elegantly routed with jack to jack enabled programs. The alsa midi and jack bridge is a thing of beauty. 3) However Bitwig to be able to use a midi pad controller DEMANDS that you DISABLE alsa midi driver before it recognizes the midi controller. 4) This is very unfortunate and very W$ndish approach to devices. 5) So at the moment I can either use Bitwig with a pad controller and absolutely no other midi through Alsa/Jack will work, or my alsa/jack routings work but I cannot use Bitwig with a Midi Controller. My Question 1) Is it in any way possible to configure the Alsa Midi driver tgo ignore a specific Midi device in order that my Alsa/Jack routings can be used in conjunction with Bitwig using its pad-controller. 2) To summarize my question, I need to somehow blacklist the midi controller to not be touched or accessed ihn any way. Basically totally forget about its existence. 3) I am sure this must be possible as there are a lot of W$ndish style programs like this developed to need exclusive access to a midi device. 4) I tried to convince the developers of Bitwig to make Bitwig Jack aware, but it is seemingly not going to happen. That would have been the right way. I sincerely hope you will have a recommendation I can work on and impliment. Thank you.

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[PATCH] ALSA: usb-audio: Clean up quirk entries with macros
by Takashi Iwai 27 May '20

27 May '20

Introduced a couple of macros to simplify some quirk entries. Just cosmetic, no functional changes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks-table.h | 93 ++++++++++++++++-------------------------------- 1 file changed, 30 insertions(+), 63 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index eb89902a83be..5236f4d015c4 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -25,6 +25,21 @@ .idProduct = prod, \ .bInterfaceClass = USB_CLASS_VENDOR_SPEC +#define QUIRK_RENAME_DEVICE(_vendor, _device) \ + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \ + .vendor_name = _vendor, \ + .product_name = _device, \ + .ifnum = QUIRK_NO_INTERFACE \ + } + +#define QUIRK_DEVICE_PROFILE(_vendor, _device, _profile) \ + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \ + .vendor_name = _vendor, \ + .product_name = _device, \ + .profile_name = _profile, \ + .ifnum = QUIRK_NO_INTERFACE \ + } + /* FTDI devices */ { USB_DEVICE(0x0403, 0xb8d8), @@ -61,20 +76,12 @@ /* Creative/E-Mu devices */ { USB_DEVICE(0x041e, 0x3010), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Creative Labs", - .product_name = "Sound Blaster MP3+", - .ifnum = QUIRK_NO_INTERFACE - } + QUIRK_RENAME_DEVICE("Creative Labs", "Sound Blaster MP3+") }, /* Creative/Toshiba Multimedia Center SB-0500 */ { USB_DEVICE(0x041e, 0x3048), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Toshiba", - .product_name = "SB-0500", - .ifnum = QUIRK_NO_INTERFACE - } + QUIRK_RENAME_DEVICE("Toshiba", "SB-0500") }, { /* E-Mu 0202 USB */ @@ -207,11 +214,7 @@ .idProduct = 0x0990, .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Logitech, Inc.", - .product_name = "QuickCam Pro 9000", - .ifnum = QUIRK_NO_INTERFACE - } + QUIRK_RENAME_DEVICE("Logitech, Inc.", "QuickCam Pro 9000") }, /* @@ -2596,11 +2599,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x0ccd, 0x0028), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "TerraTec", - .product_name = "Aureon5.1MkII", - .ifnum = QUIRK_NO_INTERFACE - } + QUIRK_RENAME_DEVICE("TerraTec", "Aureon5.1MkII") }, { USB_DEVICE(0x0ccd, 0x0035), @@ -2615,19 +2614,11 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Stanton/N2IT Final Scratch v1 device ('Scratchamp') */ { USB_DEVICE(0x103d, 0x0100), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Stanton", - .product_name = "ScratchAmp", - .ifnum = QUIRK_NO_INTERFACE - } + QUIRK_RENAME_DEVICE("Stanton", "ScratchAmp") }, { USB_DEVICE(0x103d, 0x0101), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Stanton", - .product_name = "ScratchAmp", - .ifnum = QUIRK_NO_INTERFACE - } + QUIRK_RENAME_DEVICE("Stanton", "ScratchAmp") }, /* Novation EMS devices */ @@ -2788,11 +2779,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* aka. Serato Scratch Live DJ Box */ USB_DEVICE(0x13e5, 0x0001), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Rane", - .product_name = "SL-1", - .ifnum = QUIRK_NO_INTERFACE - } + QUIRK_RENAME_DEVICE("Rane", "SL-1") }, /* Native Instruments MK2 series */ @@ -3259,10 +3246,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), * is also used by the CM106 based cards, so make it unique. */ USB_DEVICE(0x0d8c, 0x0103), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .product_name = "Audio Advantage MicroII", - .ifnum = QUIRK_NO_INTERFACE - } + QUIRK_RENAME_DEVICE(NULL, "Audio Advantage MicroII") }, /* disabled due to regression for other devices; @@ -3368,12 +3352,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), /* Dell WD15 Dock */ { USB_DEVICE(0x0bda, 0x4014), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Dell", - .product_name = "WD15 Dock", - .profile_name = "Dell-WD15-Dock", - .ifnum = QUIRK_NO_INTERFACE - } + QUIRK_DEVICE_PROFILE("Dell", "WD15 Dock", "Dell-WD15-Dock") }, /* Dell WD19 Dock */ { @@ -3553,12 +3532,8 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), #define ALC1220_VB_DESKTOP(vend, prod) { \ USB_DEVICE(vend, prod), \ - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \ - .vendor_name = "Realtek", \ - .product_name = "ALC1220-VB-DT", \ - .profile_name = "Realtek-ALC1220-VB-Desktop", \ - .ifnum = QUIRK_NO_INTERFACE \ - } \ + QUIRK_DEVICE_PROFILE("Realtek", "ALC1220-VB-DT", \ + "Realtek-ALC1220-VB-Desktop") \ } ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ @@ -3574,21 +3549,13 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ */ { USB_DEVICE(0x0414, 0xa000), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Gigabyte", - .product_name = "Aorus Master Front Headphone", - .profile_name = "Gigabyte-Aorus-Master-Front-Headphone", - .ifnum = QUIRK_NO_INTERFACE - } + QUIRK_DEVICE_PROFILE("Gigabyte", "Aorus Master Front Headphone", + "Gigabyte-Aorus-Master-Front-Headphone") }, { USB_DEVICE(0x0414, 0xa001), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Gigabyte", - .product_name = "Aorus Master Main Audio", - .profile_name = "Gigabyte-Aorus-Master-Main-Audio", - .ifnum = QUIRK_NO_INTERFACE - } + QUIRK_DEVICE_PROFILE("Gigabyte", "Aorus Master Main Audio", + "Gigabyte-Aorus-Master-Main-Audio") }, #undef USB_DEVICE_VENDOR_SPEC -- 2.16.4

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[PATCH v2 0/2] ASoC topology header parsing refinement
by Keyon Jie 27 May '20

27 May '20

This small series is to optimize the header logging during the topology parsing. This is verified work fine on both SOF and SST drivers (bxt_rt298 with dfw_sst.bin). Change History: v2: - Change the internal used array to be 'static' to fix the issue reported by: kbuild test robot <lkp(a)intel.com> - Add testing coverage including Intel SST driver also. v1: - Initial version. Keyon Jie (2): ASoC: topology: refine and log the header in the correct pass ASoC: topology: remove the redundant pass checks sound/soc/soc-topology.c | 99 ++++++++++++++++++---------------------- 1 file changed, 44 insertions(+), 55 deletions(-) -- 2.25.1

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[PATCH] ASoC: Intel: sst: Fix runtime PM imbalance in sst_power_control
by Dinghao Liu 27 May '20

27 May '20

When sst_load_fw() returns an error code, a pairing runtime PM usage counter decrement is needed to keep the counter balanced. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/soc/intel/atom/sst/sst_drv_interface.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 762495385d5c..3897985b254f 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -150,6 +150,7 @@ static int sst_power_control(struct device *dev, bool state) if ((ctx->sst_state == SST_RESET) && (usage_count == 1)) { ret = sst_load_fw(ctx); if (ret) { + pm_runtime_put_sync(dev); dev_err(dev, "FW download fail %d\n", ret); sst_set_fw_state_locked(ctx, SST_RESET); ret = sst_pm_runtime_put(ctx); -- 2.17.1

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