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

[PATCH] ALSA: pcm: oss: Fix a few more UBSAN fixes
by Takashi Iwai 18 Dec '20

18 Dec '20

There are a few places that call round{up|down}_pow_of_two() with the value zero, and this causes undefined behavior warnings. Avoid calling those macros if such a nonsense value is passed; it's a minor optimization as well, as we handle it as either an error or a value to be skipped, instead. Reported-by: syzbot+33ef0b6639a8d2d42b4c(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index de1917484647..142fc751a847 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -693,6 +693,8 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, oss_buffer_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL)) * oss_frame_size; + if (!oss_buffer_size) + return -EINVAL; oss_buffer_size = rounddown_pow_of_two(oss_buffer_size); if (atomic_read(&substream->mmap_count)) { if (oss_buffer_size > runtime->oss.mmap_bytes) @@ -728,17 +730,21 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, min_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - min_period_size *= oss_frame_size; - min_period_size = roundup_pow_of_two(min_period_size); - if (oss_period_size < min_period_size) - oss_period_size = min_period_size; + if (min_period_size) { + min_period_size *= oss_frame_size; + min_period_size = roundup_pow_of_two(min_period_size); + if (oss_period_size < min_period_size) + oss_period_size = min_period_size; + } max_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - max_period_size *= oss_frame_size; - max_period_size = rounddown_pow_of_two(max_period_size); - if (oss_period_size > max_period_size) - oss_period_size = max_period_size; + if (max_period_size) { + max_period_size *= oss_frame_size; + max_period_size = rounddown_pow_of_two(max_period_size); + if (oss_period_size > max_period_size) + oss_period_size = max_period_size; + } oss_periods = oss_buffer_size / oss_period_size; -- 2.26.2

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[PATCH] ALSA: pcm: Remove snd_pcm_lib_preallocate_dma_free()
by Lars-Peter Clausen 18 Dec '20

18 Dec '20

Since commit d4cfb30fce03 ("ALSA: pcm: Set per-card upper limit of PCM buffer allocations") snd_pcm_lib_preallocate_dma_free() is a single line function that has one caller, which is another single line function. Clean this up a bit and remove snd_pcm_lib_preallocate_dma_free() and directly call do_free_pages() from snd_pcm_lib_preallocate_free(). This is a bit less boilerplate. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/core/pcm_memory.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 4f03ba8ed0ae..ee6e9c5eec45 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -89,14 +89,6 @@ static int preallocate_pcm_pages(struct snd_pcm_substream *substream, size_t siz return 0; } -/* - * release the preallocated buffer if not yet done. - */ -static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream) -{ - do_free_pages(substream->pcm->card, &substream->dma_buffer); -} - /** * snd_pcm_lib_preallocate_free - release the preallocated buffer of the specified substream. * @substream: the pcm substream instance @@ -105,7 +97,7 @@ static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream */ void snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream) { - snd_pcm_lib_preallocate_dma_free(substream); + do_free_pages(substream->pcm->card, &substream->dma_buffer); } /** -- 2.20.1

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[PATCH 0/2] ALSA: Clear PCM buffers more properly
by Takashi Iwai 18 Dec '20

18 Dec '20

Hi, here are two small fix patches for PCM buffer allocations. Basically those address two things that have been discussed in the recent thread [1]: * Clear the remaining bytes in PCM buffer pages that are exposed via mmap * Assure the page aligned allocations for genalloc This doesn't cover the cases where non-standard PCM buffer allocation is used. Those should handle the buffer clearance by themselves. Takashi [1] https://lore.kernel.org/r/05c824e5-0c33-4182-26fa-b116a42b10d6@metafoo.de === Takashi Iwai (2): ALSA: memalloc: Align buffer allocations in page size ALSA: pcm: Clear the full allocated memory at hw_params sound/core/memalloc.c | 1 + sound/core/pcm_native.c | 9 +++++++-- 2 files changed, 8 insertions(+), 2 deletions(-) -- 2.26.2

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[PATCH] ALSA: usb-audio: Disable sample read check if firmware doesn't give back
by Takashi Iwai 18 Dec '20

18 Dec '20

Some buggy firmware don't give the current sample rate but leaves zero. Handle this case more gracefully without warning but just skip the current rate verification from the next time. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index e940dcee792b..31051f2be46d 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -534,6 +534,12 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, } crate = data[0] | (data[1] << 8) | (data[2] << 16); + if (!crate) { + dev_info(&dev->dev, "failed to read current rate; disabling the check\n"); + chip->sample_rate_read_error = 3; /* three strikes, see above */ + return 0; + } + if (crate != rate) { dev_warn(&dev->dev, "current rate %d is different from the runtime rate %d\n", crate, rate); // runtime->rate = crate; -- 2.26.2

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Re: [PATCH 06/18] arm64: dts: renesas: beacon: Configure Audio CODEC clocks
by Geert Uytterhoeven 18 Dec '20

18 Dec '20

Hi Adam, CC alsa-devel On Sun, Dec 13, 2020 at 7:38 PM Adam Ford <aford173(a)gmail.com> wrote: > With the newly added configurable clock options, the audio CODEC can > configure the mclk automatically. Add the reference to the versaclock. > Since the devices on I2C5 can communicate at 400KHz, let's also increase > that too > > Signed-off-by: Adam Ford <aford173(a)gmail.com> Thanks for your patch! > --- a/arch/arm64/boot/dts/renesas/beacon-renesom-baseboard.dtsi > +++ b/arch/arm64/boot/dts/renesas/beacon-renesom-baseboard.dtsi > @@ -424,13 +424,15 @@ &i2c0 { > > &i2c5 { > status = "okay"; > - clock-frequency = <100000>; > + clock-frequency = <400000>; > pinctrl-0 = <&i2c5_pins>; > pinctrl-names = "default"; > > codec: wm8962@1a { > compatible = "wlf,wm8962"; > reg = <0x1a>; > + clocks = <&versaclock6_bb 3>; > + clock-names = "mclk"; While the driver does get the (nameless) clock, the DT bindings lack any mention of a clocks property. It would be good to update the bindings. Note that arch/arm/boot/dts/imx6-logicpd-baseboard.dtsi and arch/arm64/boot/dts/freescale/imx8mm-beacon-baseboard.dtsi (both by your hand) use "xclk" instead of "mclk"? > DCVDD-supply = <&reg_audio>; > DBVDD-supply = <&reg_audio>; > AVDD-supply = <&reg_audio>; Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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Move SDM845 out of Qualcomm folder and add C360
by GitHub pull_request - opened 18 Dec '20

18 Dec '20

alsa-project/alsa-ucm-conf pull request #70 was opened from Srinivas-Kandagatla: With driver_name fix in sdm845 machine driver https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… /proc/asound/cards get populated correctly with proper driver name and card long name. With this in place we should be able to rearrange sdm845 directory to be inline with others and reuse some of the common code. Without this all the sdm845 based machines will get a generic ucm description which will not work, one of such instance is Lenovo C360 Laptop. This patchset also add support to Lenovo C360 support Thanks, srini Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/70 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/70.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: meson: axg-tdmin: fix axg skew offset
by Jerome Brunet 18 Dec '20

18 Dec '20

The signal captured on from tdm decoder of the AXG SoC is incorrect. It appears amplified. The skew offset of the decoder is wrong. Setting the skew offset to 3, like the g12 and sm1 SoCs, solves and gives correct data. Fixes: 13a22e6a98f8 ("ASoC: meson: add tdm input driver") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdmin.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c index 88ed95ae886b..b4faf9d5c1aa 100644 --- a/sound/soc/meson/axg-tdmin.c +++ b/sound/soc/meson/axg-tdmin.c @@ -224,15 +224,6 @@ static const struct axg_tdm_formatter_ops axg_tdmin_ops = { }; static const struct axg_tdm_formatter_driver axg_tdmin_drv = { - .component_drv = &axg_tdmin_component_drv, - .regmap_cfg = &axg_tdmin_regmap_cfg, - .ops = &axg_tdmin_ops, - .quirks = &(const struct axg_tdm_formatter_hw) { - .skew_offset = 2, - }, -}; - -static const struct axg_tdm_formatter_driver g12a_tdmin_drv = { .component_drv = &axg_tdmin_component_drv, .regmap_cfg = &axg_tdmin_regmap_cfg, .ops = &axg_tdmin_ops, @@ -247,10 +238,10 @@ static const struct of_device_id axg_tdmin_of_match[] = { .data = &axg_tdmin_drv, }, { .compatible = "amlogic,g12a-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, { .compatible = "amlogic,sm1-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, {} }; MODULE_DEVICE_TABLE(of, axg_tdmin_of_match); -- 2.29.2

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[PATCH] ASoC: meson: axg-tdm-interface: fix loopback
by Jerome Brunet 18 Dec '20

18 Dec '20

When the axg-tdm-interface was introduced, the backend DAI was marked as an endpoint when DPCM was walking the DAPM graph to find a its BE. It is no longer the case since this commit 8dd26dff00c0 ("ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks") Because of this, when DPCM finds a BE it does everything it needs on the DAIs but it won't power up the widgets between the FE and the BE if there is no actual endpoint after the BE. On meson-axg HWs, the loopback is a special DAI of the tdm-interface BE. It is only linked to the dummy codec since there no actual HW after it. >From the DAPM perspective, the DAI has no endpoint. Because of this, the TDM decoder, which is a widget between the FE and BE is not powered up. >From the user perspective, everything seems fine but no data is produced. Connecting the Loopback DAI to a dummy DAPM endpoint solves the problem. Fixes: 8dd26dff00c0 ("ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks") Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-interface.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index c8664ab80d45..87cac440b369 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -467,8 +467,20 @@ static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component, return ret; } +static const struct snd_soc_dapm_widget axg_tdm_iface_dapm_widgets[] = { + SND_SOC_DAPM_SIGGEN("Playback Signal"), +}; + +static const struct snd_soc_dapm_route axg_tdm_iface_dapm_routes[] = { + { "Loopback", NULL, "Playback Signal" }, +}; + static const struct snd_soc_component_driver axg_tdm_iface_component_drv = { - .set_bias_level = axg_tdm_iface_set_bias_level, + .dapm_widgets = axg_tdm_iface_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdm_iface_dapm_widgets), + .dapm_routes = axg_tdm_iface_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdm_iface_dapm_routes), + .set_bias_level = axg_tdm_iface_set_bias_level, }; static const struct of_device_id axg_tdm_iface_of_match[] = { -- 2.29.2

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[PATCH] ASoC: Intel: haswell: Add missing pm_ops
by Cezary Rojewski 18 Dec '20

18 Dec '20

haswell machine board is missing pm_ops what prevents it from undergoing suspend-resume procedure successfully. Assign default snd_soc_pm_ops so this is no longer the case. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/boards/haswell.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index c55d1239e705..c763bfeb1f38 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -189,6 +189,7 @@ static struct platform_driver haswell_audio = { .probe = haswell_audio_probe, .driver = { .name = "haswell-audio", + .pm = &snd_soc_pm_ops, }, }; -- 2.17.1

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[PATCH] ASoC: rt711: mutex between calibration and power state changes
by shumingf＠realtek.com 18 Dec '20

18 Dec '20

From: Shuming Fan <shumingf(a)realtek.com> To avoid calibration time-out, this patch adds the mutex between calibration and power state changes Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 5771c02c3459..85f744184a60 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -462,6 +462,8 @@ static int rt711_set_amp_gain_put(struct snd_kcontrol *kcontrol, unsigned int read_ll, read_rl; int i; + mutex_lock(&rt711->calibrate_mutex); + /* Can't use update bit function, so read the original value first */ addr_h = mc->reg; addr_l = mc->rreg; @@ -547,6 +549,8 @@ static int rt711_set_amp_gain_put(struct snd_kcontrol *kcontrol, 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); return 0; } @@ -859,9 +863,11 @@ static int rt711_set_bias_level(struct snd_soc_component *component, break; case SND_SOC_BIAS_STANDBY: + mutex_lock(&rt711->calibrate_mutex); regmap_write(rt711->regmap, RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + mutex_unlock(&rt711->calibrate_mutex); break; default: -- 2.29.0

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