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

[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

1 0

[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

1 0

[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

1 0

[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

1 0

[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

1 0

[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