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

[PATCH] ASoC: wm8962: Do not access WM8962_GPIO_BASE
by Fabio Estevam 03 Aug '20

03 Aug '20

According to the WM8962 datasheet, there is no register at address 0x200. WM8962_GPIO_BASE is just a base address for the GPIO registers and not a real register, so remove it from wm8962_readable_register(). Also, Register 515 (WM8962_GPIO_BASE + 3) does not exist, so skip its access. This fixes the following errors: wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Hi, There is still one more soc_component_read_no_lock error left on register 48. I can get rid of it with the below change: --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -151,6 +151,7 @@ static const struct reg_default wm8962_reg[] = { { 40, 0x0000 }, /* R40 - SPKOUTL volume */ { 41, 0x0000 }, /* R41 - SPKOUTR volume */ + { 48, 0x0000 }, /* R48 - Additional control(4) */ { 49, 0x0010 }, /* R49 - Class D Control 1 */ { 51, 0x0003 }, /* R51 - Class D Control 2 */ @@ -841,7 +842,6 @@ static bool wm8962_readable_register(struct device *dev, unsigned int reg) case WM8962_SPKOUTL_VOLUME: case WM8962_SPKOUTR_VOLUME: case WM8962_THERMAL_SHUTDOWN_STATUS: - case WM8962_ADDITIONAL_CONTROL_4: case WM8962_CLASS_D_CONTROL_1: case WM8962_CLASS_D_CONTROL_2: case WM8962_CLOCKING_4: I haven't submitted it yet because I don't know if this is the correct approach. Please advise. Thanks sound/soc/codecs/wm8962.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index df8cdc71357d..8159a3866cde 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -956,7 +956,6 @@ static bool wm8962_readable_register(struct device *dev, unsigned int reg) case WM8962_EQ39: case WM8962_EQ40: case WM8962_EQ41: - case WM8962_GPIO_BASE: case WM8962_GPIO_2: case WM8962_GPIO_3: case WM8962_GPIO_5: @@ -3437,7 +3436,13 @@ static int wm8962_probe(struct snd_soc_component *component) /* Save boards having to disable DMIC when not in use */ dmicclk = false; dmicdat = false; - for (i = 0; i < WM8962_MAX_GPIO; i++) { + for (i = 1; i < WM8962_MAX_GPIO; i++) { + /* + * Register 515 (WM8962_GPIO_BASE + 3) does not exist, + * so skip its access + */ + if (i == 3) + continue; switch (snd_soc_component_read(component, WM8962_GPIO_BASE + i) & WM8962_GP2_FN_MASK) { case WM8962_GPIO_FN_DMICCLK: -- 2.17.1

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[PATCH v3 0/2] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 03 Aug '20

03 Aug '20

Note: - The machine driver patch depends on LPASS patch series so it is not ready to be merged now. ASoC: qcom: Add support for SC7180 lpass variant https://patchwork.kernel.org/cover/11678133/ - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Thanks for the review! Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (1): ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 113 ++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 244 ++++++++++++++++++ 4 files changed, 371 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.163.g6104cc2f0b6-goog

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Re: 回复: [PATCH] ALSA: seq: KASAN: use-after-free Read in delete_and_unsubscribe_port
by Takashi Iwai 03 Aug '20

03 Aug '20

On Mon, 03 Aug 2020 03:35:05 +0200, Zhang, Qiang wrote: > > >Thanks for the patch. But I'm afraid that this change would break the > >existing behavior and might have a bad side-effect. > > >It's likely the same issue as reported in another syzkaller report > >("KASAN: invalid-free in snd_seq_port_disconnect"), and Hillf's patch > >below should covert this as well. Could you check whether it works? > > yes It's should same issue, add mutex lock in odev_ioctl, ensure serialization. > however, it should not be necessary to mutually exclusive with open and close. That's a big-hammer approach indeed, but it should be more reasonable in this case. It makes the patch shorter and simpler, while the OSS sequencer is an ancient interface that wasn't considered much for the concurrency, and this might also cover the case where the access to another sequencer object that is being to be closed. So, it'd be great if you can confirm that the patch actually works. Then we can brush up and merge it for 5.9-rc1. thanks, Takashi > > > > >thanks, > > >Takashi > > >--- > >--- a/sound/core/seq/oss/seq_oss.c > >+++ b/sound/core/seq/oss/seq_oss.c > >@@ -167,11 +167,17 @@ odev_write(struct file *file, const char > >static long > >odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg) > >{ > >+ long rc; > > struct seq_oss_devinfo *dp; > >+ > >+ mutex_lock(&register_mutex); > > dp = file->private_data; > > if (snd_BUG_ON(!dp)) > >- return -ENXIO; > >- return snd_seq_oss_ioctl(dp, cmd, arg); > >+ rc = -ENXIO; > >+ else > >+ rc = snd_seq_oss_ioctl(dp, cmd, arg); > >+ mutex_unlock(&register_mutex); > >+ return rc; > >} > > >#ifdef CONFIG_COMPAT

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[PATCH 1/3] ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
by Connor McAdams 03 Aug '20

03 Aug '20

When the ZxR headphone gain control was added, the ca0132_switch_get function was not updated, which meant that the changes to the control state were not saved when entering/exiting alsamixer. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 19c575fd28a1..2cbe01d59c16 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -5748,6 +5748,11 @@ static int ca0132_switch_get(struct snd_kcontrol *kcontrol, return 0; } + if (nid == ZXR_HEADPHONE_GAIN) { + *valp = spec->zxr_gain_set; + return 0; + } + return 0; } -- 2.20.1

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[PATCH v2] ASoC: fsl-asoc-card: Remove fsl_asoc_card_set_bias_level function
by Shengjiu Wang 03 Aug '20

03 Aug '20

With this case: aplay -Dhw:x 16khz.wav 24khz.wav There is sound distortion for 24khz.wav. The reason is that setting PLL of WM8962 with set_bias_level function, the bias level is not changed when 24khz.wav is played, then the PLL won't be reset, the clock is not correct, so distortion happens. The resolution of this issue is to remove fsl_asoc_card_set_bias_level. Move PLL configuration to hw_params and hw_free. After removing fsl_asoc_card_set_bias_level, also test WM8960 case, it can work. Fixes: 708b4351f08c ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - replace is_stream_in_use with streams - add "out" error handler in hw_params() sound/soc/fsl/fsl-asoc-card.c | 155 ++++++++++++++++------------------ 1 file changed, 71 insertions(+), 84 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4848ba61d083..b6a2a527dc04 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -73,6 +73,7 @@ struct cpu_priv { * @codec_priv: CODEC private data * @cpu_priv: CPU private data * @card: ASoC card structure + * @streams: Mask of current active streams * @sample_rate: Current sample rate * @sample_format: Current sample format * @asrc_rate: ASRC sample rate used by Back-Ends @@ -89,6 +90,7 @@ struct fsl_asoc_card_priv { struct codec_priv codec_priv; struct cpu_priv cpu_priv; struct snd_soc_card card; + u8 streams; u32 sample_rate; snd_pcm_format_t sample_format; u32 asrc_rate; @@ -151,21 +153,17 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct codec_priv *codec_priv = &priv->codec_priv; struct cpu_priv *cpu_priv = &priv->cpu_priv; struct device *dev = rtd->card->dev; + unsigned int pll_out; int ret; priv->sample_rate = params_rate(params); priv->sample_format = params_format(params); + priv->streams |= BIT(substream->stream); - /* - * If codec-dai is DAI Master and all configurations are already in the - * set_bias_level(), bypass the remaining settings in hw_params(). - * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. - */ - if ((priv->card.set_bias_level && - priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || - fsl_asoc_card_is_ac97(priv)) + if (fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -174,7 +172,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->sysclk_dir[tx]); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set sysclk for cpu dai\n"); - return ret; + goto out; } if (cpu_priv->slot_width) { @@ -182,6 +180,69 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->slot_width); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set TDM slot for cpu dai\n"); + goto out; + } + } + + /* Specific configuration for PLL */ + if (codec_priv->pll_id && codec_priv->fll_id) { + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, + codec_priv->mclk_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + goto out; + } + + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + goto out; + } + } + + return 0; + +out: + priv->streams &= ~BIT(substream->stream); + return ret; +} + +static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->streams &= ~BIT(substream->stream); + + if (!priv->streams && codec_priv->pll_id && + codec_priv->fll_id) { + /* Force freq to be 0 to avoid error message in codec */ + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->mclk_id, + 0, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, 0, 0, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to stop FLL: %d\n", ret); return ret; } } @@ -191,6 +252,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, static const struct snd_soc_ops fsl_asoc_card_ops = { .hw_params = fsl_asoc_card_hw_params, + .hw_free = fsl_asoc_card_hw_free, }; static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -254,75 +316,6 @@ static struct snd_soc_dai_link fsl_asoc_card_dai[] = { }, }; -static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct codec_priv *codec_priv = &priv->codec_priv; - struct device *dev = card->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = asoc_rtd_to_codec(rtd, 0); - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level != SND_SOC_BIAS_STANDBY) - break; - - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, - codec_priv->mclk_id, - codec_priv->mclk_freq, pll_out); - if (ret) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, - pll_out, SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level != SND_SOC_BIAS_PREPARE) - break; - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, - codec_priv->mclk_freq, - SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to switch away from FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0); - if (ret) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - break; - - default: - break; - } - - return 0; -} - static int fsl_asoc_card_audmux_init(struct device_node *np, struct fsl_asoc_card_priv *priv) { @@ -611,7 +604,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; - priv->card.set_bias_level = NULL; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; @@ -628,26 +620,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { codec_dai_name = "wm8962"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { codec_dai_name = "wm8960-hifi"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_AC97; priv->card.dapm_routes = audio_map_ac97; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97); } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) { codec_dai_name = "fsl-mqs-dai"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_NB_NF; @@ -657,7 +645,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) { codec_dai_name = "wm8524-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0; -- 2.27.0

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[PATCH] ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
by Huacai Chen 02 Aug '20

02 Aug '20

There are several Loongson-3 based laptops produced by CZC or Lemote, they use alc269/alc662 codecs and need specific pin-tables, this patch add their pin-tables. Signed-off-by: Huacai Chen <chenhc(a)lemote.com> --- sound/pci/hda/patch_realtek.c | 114 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 114 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 29f5878..65c75c6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6166,6 +6166,11 @@ enum { ALC289_FIXUP_ASUS_GA502, ALC256_FIXUP_ACER_MIC_NO_PRESENCE, ALC285_FIXUP_HP_GPIO_AMP_INIT, + ALC269_FIXUP_CZC_B20, + ALC269_FIXUP_CZC_TMI, + ALC269_FIXUP_CZC_L101, + ALC269_FIXUP_LEMOTE_A1802, + ALC269_FIXUP_LEMOTE_A190X, }; static const struct hda_fixup alc269_fixups[] = { @@ -7404,6 +7409,89 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_GPIO_LED }, + [ALC269_FIXUP_CZC_B20] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x411111f0 }, + { 0x14, 0x90170110 }, /* speaker */ + { 0x15, 0x032f1020 }, /* HP out */ + { 0x17, 0x411111f0 }, + { 0x18, 0x03ab1040 }, /* mic */ + { 0x19, 0xb7a7013f }, + { 0x1a, 0x0181305f }, + { 0x1b, 0x411111f0 }, + { 0x1d, 0x411111f0 }, + { 0x1e, 0x411111f0 }, + { } + }, + .chain_id = ALC269_FIXUP_DMIC, + }, + [ALC269_FIXUP_CZC_TMI] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x4000c000 }, + { 0x14, 0x90170110 }, /* speaker */ + { 0x15, 0x0421401f }, /* HP out */ + { 0x17, 0x411111f0 }, + { 0x18, 0x04a19020 }, /* mic */ + { 0x19, 0x411111f0 }, + { 0x1a, 0x411111f0 }, + { 0x1b, 0x411111f0 }, + { 0x1d, 0x40448505 }, + { 0x1e, 0x411111f0 }, + { 0x20, 0x8000ffff }, + { } + }, + .chain_id =ALC269_FIXUP_DMIC, + }, + [ALC269_FIXUP_CZC_L101] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x40000000 }, + { 0x14, 0x01014010 }, /* speaker */ + { 0x15, 0x411111f0 }, /* HP out */ + { 0x16, 0x411111f0 }, + { 0x18, 0x01a19020 }, /* mic */ + { 0x19, 0x02a19021 }, + { 0x1a, 0x0181302f }, + { 0x1b, 0x0221401f }, + { 0x1c, 0x411111f0 }, + { 0x1d, 0x4044c601 }, + { 0x1e, 0x411111f0 }, + { } + }, + .chain_id =ALC269_FIXUP_DMIC, + }, + [ALC269_FIXUP_LEMOTE_A1802] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x40000000 }, + { 0x14, 0x90170110 }, /* speaker */ + { 0x17, 0x411111f0 }, + { 0x18, 0x03a19040 }, /* mic1 */ + { 0x19, 0x90a70130 }, /* mic2 */ + { 0x1a, 0x411111f0 }, + { 0x1b, 0x411111f0 }, + { 0x1d, 0x40489d2d }, + { 0x1e, 0x411111f0 }, + { 0x20, 0x0003ffff }, + { 0x21, 0x03214020 }, + { } + }, + .chain_id = ALC269_FIXUP_DMIC, + }, + [ALC269_FIXUP_LEMOTE_A190X] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x15, 0x0121401f }, /* HP out */ + { 0x18, 0x01a19c20 }, /* rear mic */ + { 0x19, 0x99a3092f }, /* front mic */ + { 0x1b, 0x0201401f }, /* front lineout */ + { } + }, + .chain_id = ALC269_FIXUP_DMIC, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7693,9 +7781,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), + SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), + SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), + SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), + SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), + SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), #if 0 /* Below is a quirk table taken from the old code. @@ -8951,6 +9044,7 @@ enum { ALC662_FIXUP_LED_GPIO1, ALC662_FIXUP_IDEAPAD, ALC272_FIXUP_MARIO, + ALC662_FIXUP_CZC_ET26, ALC662_FIXUP_CZC_P10T, ALC662_FIXUP_SKU_IGNORE, ALC662_FIXUP_HP_RP5800, @@ -9020,6 +9114,25 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc272_fixup_mario, }, + [ALC662_FIXUP_CZC_ET26] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x403cc000}, + {0x14, 0x90170110}, /* speaker */ + {0x15, 0x411111f0}, + {0x16, 0x411111f0}, + {0x18, 0x01a19030}, /* mic */ + {0x19, 0x90a7013f}, /* int-mic */ + {0x1a, 0x01014020}, + {0x1b, 0x0121401f}, + {0x1c, 0x411111f0}, + {0x1d, 0x411111f0}, + {0x1e, 0x40478e35}, + {} + }, + .chained = true, + .chain_id = ALC662_FIXUP_SKU_IGNORE + }, [ALC662_FIXUP_CZC_P10T] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -9403,6 +9516,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO), SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68), SND_PCI_QUIRK(0x1b0a, 0x01b8, "ACER Veriton", ALC662_FIXUP_ACER_VERITON), + SND_PCI_QUIRK(0x1b35, 0x1234, "CZC ET26", ALC662_FIXUP_CZC_ET26), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS), -- 2.7.0

2 2

[PATCH] ALSA: seq: KASAN: use-after-free Read in delete_and_unsubscribe_port
by qiang.zhang＠windriver.com 01 Aug '20

01 Aug '20

From: Zhang Qiang <qiang.zhang(a)windriver.com> There is a potential race window,when a task acquire "src->list_mutex" write sem,traverse the linked list to find "subs" objects through parameter "info" in snd_seq_port_disconnect and then release this write sem, before this task acquire write sem again,this write sem may be acquired by another task, and get the same "subs" object through the same "info" before, it could happen "use-after-free" later, so a simple solution is to delete the object from the linked list when it is found. BUG: KASAN: use-after-free in list_empty include/linux/list.h:282 [inline] BUG: KASAN: use-after-free in delete_and_unsubscribe_port+0x8b/0x450 sound/core/seq/seq_ports.c:530 Read of size 8 at addr ffff888098523060 by task syz-executor.0/7202 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1f0/0x31e lib/dump_stack.c:118 print_address_description+0x66/0x5a0 mm/kasan/report.c:383 __kasan_report mm/kasan/report.c:513 [inline] kasan_report+0x132/0x1d0 mm/kasan/report.c:530 list_empty include/linux/list.h:282 [inline] delete_and_unsubscribe_port+0x8b/0x450 sound/core/seq/seq_ports.c:530 snd_seq_port_disconnect+0x568/0x610 sound/core/seq/seq_ports.c:612 snd_seq_ioctl_unsubscribe_port+0x349/0x6c0 sound/core/seq/seq_clientmgr.c:1525 snd_seq_oss_midi_close+0x397/0x620 sound/core/seq/oss/seq_oss_midi.c:405 snd_seq_oss_synth_reset+0x335/0x8b0 sound/core/seq/oss/seq_oss_synth.c:406 snd_seq_oss_reset+0x5b/0x250 sound/core/seq/oss/seq_oss_init.c:435 snd_seq_oss_ioctl+0x5c2/0x1090 sound/core/seq/oss/seq_oss_ioctl.c:93 odev_ioctl+0x51/0x70 sound/core/seq/oss/seq_oss.c:174 vfs_ioctl fs/ioctl.c:48 [inline] ksys_ioctl fs/ioctl.c:753 [inline] __do_sys_ioctl fs/ioctl.c:762 [inline] __se_sys_ioctl+0xf9/0x160 fs/ioctl.c:760 do_syscall_64+0x73/0xe0 arch/x86/entry/common.c:384 entry_SYSCALL_64_after_hwframe+0x44/0xa9 Allocated by task 7202: save_stack mm/kasan/common.c:48 [inline] set_track mm/kasan/common.c:56 [inline] __kasan_kmalloc+0x103/0x140 mm/kasan/common.c:494 kmem_cache_alloc_trace+0x234/0x300 mm/slab.c:3551 kmalloc include/linux/slab.h:555 [inline] kzalloc include/linux/slab.h:669 [inline] snd_seq_port_connect+0x66/0x460 sound/core/seq/seq_ports.c:553 snd_seq_ioctl_subscribe_port+0x349/0x6c0 sound/core/seq/seq_clientmgr.c:1484 snd_seq_oss_midi_open+0x4db/0x830 sound/core/seq/oss/seq_oss_midi.c:364 snd_seq_oss_synth_setup_midi+0x108/0x510 sound/core/seq/oss/seq_oss_synth.c:269 snd_seq_oss_open+0x899/0xe90 sound/core/seq/oss/seq_oss_init.c:261 odev_open+0x5e/0x90 sound/core/seq/oss/seq_oss.c:125 chrdev_open+0x498/0x580 fs/char_dev.c:414 do_dentry_open+0x813/0x1070 fs/open.c:828 do_open fs/namei.c:3243 [inline] path_openat+0x278d/0x37f0 fs/namei.c:3360 do_filp_open+0x191/0x3a0 fs/namei.c:3387 do_sys_openat2+0x463/0x770 fs/open.c:1179 do_sys_open fs/open.c:1195 [inline] __do_sys_openat fs/open.c:1209 [inline] __se_sys_openat fs/open.c:1204 [inline] __x64_sys_openat+0x1c8/0x1f0 fs/open.c:1204 do_syscall_64+0x73/0xe0 arch/x86/entry/common.c:384 entry_SYSCALL_64_after_hwframe+0x44/0xa9 Freed by task 7203: save_stack mm/kasan/common.c:48 [inline] set_track mm/kasan/common.c:56 [inline] kasan_set_free_info mm/kasan/common.c:316 [inline] __kasan_slab_free+0x114/0x170 mm/kasan/common.c:455 __cache_free mm/slab.c:3426 [inline] kfree+0x10a/0x220 mm/slab.c:3757 snd_seq_port_disconnect+0x570/0x610 sound/core/seq/seq_ports.c:614 snd_seq_ioctl_unsubscribe_port+0x349/0x6c0 sound/core/seq/seq_clientmgr.c:1525 snd_seq_oss_midi_close+0x397/0x620 sound/core/seq/oss/seq_oss_midi.c:405 snd_seq_oss_synth_reset+0x335/0x8b0 sound/core/seq/oss/seq_oss_synth.c:406 snd_seq_oss_reset+0x5b/0x250 sound/core/seq/oss/seq_oss_init.c:435 snd_seq_oss_ioctl+0x5c2/0x1090 sound/core/seq/oss/seq_oss_ioctl.c:93 odev_ioctl+0x51/0x70 sound/core/seq/oss/seq_oss.c:174 vfs_ioctl fs/ioctl.c:48 [inline] ksys_ioctl fs/ioctl.c:753 [inline] __do_sys_ioctl fs/ioctl.c:762 [inline] __se_sys_ioctl+0xf9/0x160 fs/ioctl.c:760 do_syscall_64+0x73/0xe0 arch/x86/entry/common.c:384 entry_SYSCALL_64_after_hwframe+0x44/0xa9 The buggy address belongs to the object at ffff888098523000 which belongs to the cache kmalloc-128 of size 128 The buggy address is located 96 bytes inside of 128-byte region [ffff888098523000, ffff888098523080) The buggy address belongs to the page: page:ffffea00026148c0 refcount:1 mapcount:0 mapping:0000000000000000 index:0x0 flags: 0xfffe0000000200(slab) raw: 00fffe0000000200 ffffea0002613988 ffffea000262c648 ffff8880aa400700 raw: 0000000000000000 ffff888098523000 0000000100000010 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff888098522f00: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff888098522f80: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc >ffff888098523000: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff888098523080: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff888098523100: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== Reported-by: syzbot+1a54a94bd32716796edd(a)syzkaller.appspotmail.com Signed-off-by: Zhang Qiang <qiang.zhang(a)windriver.com> --- sound/core/seq/seq_ports.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 83be6b982a87..9675d3fc146e 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -533,8 +533,7 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client, grp->exclusive = 0; write_unlock_irq(&grp->list_lock); - if (!empty) - unsubscribe_port(client, port, grp, &subs->info, ack); + unsubscribe_port(client, port, grp, &subs->info, ack); up_write(&grp->list_mutex); } @@ -599,6 +598,7 @@ int snd_seq_port_disconnect(struct snd_seq_client *connector, list_for_each_entry(subs, &src->list_head, src_list) { if (match_subs_info(info, &subs->info)) { atomic_dec(&subs->ref_count); /* mark as not ready */ + list_del_init(&subs->src_list); err = 0; break; } -- 2.26.2

2 1

[PATCH] ALSA: docs: fix typo
by Julia Lawall 01 Aug '20

01 Aug '20

GFP_KRENEL -> GFP_KERNEL Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- Documentation/sound/kernel-api/writing-an-alsa-driver.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index fa49688..aa9d5ab 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -3579,7 +3579,7 @@ dependent on the bus. For normal devices, pass the device pointer ``SNDRV_DMA_TYPE_DEV`` type. For the continuous buffer unrelated to the bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type. You can pass NULL to the device pointer in that case, which is the -default mode implying to allocate with ``GFP_KRENEL`` flag. +default mode implying to allocate with ``GFP_KERNEL`` flag. If you need a different GFP flag, you can pass it by encoding the flag into the device pointer via a special macro :c:func:`snd_dma_continuous_data()`.

2 1

[PATCH] ALSA: doc: use correct config variable name
by Julia Lawall 01 Aug '20

01 Aug '20

CONFIG_PCM_XRUN_DEBUG should be CONFIG_SND_PCM_XRUN_DEBUG Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- Documentation/sound/designs/procfile.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/designs/procfile.rst b/Documentation/sound/designs/procfile.rst index 29a4668..e9f7e0c 100644 --- a/Documentation/sound/designs/procfile.rst +++ b/Documentation/sound/designs/procfile.rst @@ -91,7 +91,7 @@ PCM Proc Files ``card*/pcm*/xrun_debug`` This file appears when ``CONFIG_SND_DEBUG=y`` and - ``CONFIG_PCM_XRUN_DEBUG=y``. + ``CONFIG_SND_PCM_XRUN_DEBUG=y``. This shows the status of xrun (= buffer overrun/xrun) and invalid PCM position debug/check of ALSA PCM middle layer. It takes an integer value, can be changed by writing to this

2 1