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

[PATCH AUTOSEL 5.4 087/130] ALSA: rawmidi: Access runtime->avail always in spinlock
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 88a06d6fd6b369d88cec46c62db3e2604a2f50d5 ] The runtime->avail field may be accessed concurrently while some places refer to it without taking the runtime->lock spinlock, as detected by KCSAN. Usually this isn't a big problem, but for consistency and safety, we should take the spinlock at each place referencing this field. Reported-by: syzbot+a23a6f1215c84756577c(a)syzkaller.appspotmail.com Reported-by: syzbot+3d367d1df1d2b67f5c19(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20201206083527.21163-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/rawmidi.c | 49 +++++++++++++++++++++++++++++++------------- 1 file changed, 35 insertions(+), 14 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 94db4683cfaff..6a3543b8455fc 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -72,11 +72,21 @@ static inline unsigned short snd_rawmidi_file_flags(struct file *file) } } -static inline int snd_rawmidi_ready(struct snd_rawmidi_substream *substream) +static inline bool __snd_rawmidi_ready(struct snd_rawmidi_runtime *runtime) +{ + return runtime->avail >= runtime->avail_min; +} + +static bool snd_rawmidi_ready(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime = substream->runtime; + unsigned long flags; + bool ready; - return runtime->avail >= runtime->avail_min; + spin_lock_irqsave(&runtime->lock, flags); + ready = __snd_rawmidi_ready(runtime); + spin_unlock_irqrestore(&runtime->lock, flags); + return ready; } static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substream, @@ -945,7 +955,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, if (result > 0) { if (runtime->event) schedule_work(&runtime->event_work); - else if (snd_rawmidi_ready(substream)) + else if (__snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } spin_unlock_irqrestore(&runtime->lock, flags); @@ -1024,7 +1034,7 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun result = 0; while (count > 0) { spin_lock_irq(&runtime->lock); - while (!snd_rawmidi_ready(substream)) { + while (!__snd_rawmidi_ready(runtime)) { wait_queue_entry_t wait; if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) { @@ -1041,9 +1051,11 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_read1(substream, @@ -1181,7 +1193,7 @@ int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int coun runtime->avail += count; substream->bytes += count; if (count > 0) { - if (runtime->drain || snd_rawmidi_ready(substream)) + if (runtime->drain || __snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } return count; @@ -1370,9 +1382,11 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail && !timeout) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail && !timeout) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_write1(substream, buf, NULL, count); @@ -1452,6 +1466,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *substream; struct snd_rawmidi_runtime *runtime; + unsigned long buffer_size, avail, xruns; rmidi = entry->private_data; snd_iprintf(buffer, "%s\n\n", rmidi->name); @@ -1470,13 +1485,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Mode : %s\n" " Buffer size : %lu\n" " Avail : %lu\n", runtime->oss ? "OSS compatible" : "native", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail); + buffer_size, avail); } } } @@ -1494,13 +1512,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + xruns = runtime->xruns; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Buffer size : %lu\n" " Avail : %lu\n" " Overruns : %lu\n", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail, - (unsigned long) runtime->xruns); + buffer_size, avail, xruns); } } } -- 2.27.0

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[PATCH AUTOSEL 5.4 056/130] ALSA: usb-audio: Don't call usb_set_interface() at trigger callback
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4974b7950929e4a28d4eaee48e4ad07f168ac132 ] The PCM trigger callback is atomic, hence we must not call a function like usb_set_interface() there. Calling it from there would lead to a kernel Oops. Fix it by moving the usb_set_interface() call to set_sync_endpoint(). Also, apply the snd_usb_set_interface_quirk() for consistency, too. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/pcm.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 49ad4e7bb70b5..87389ab69b5ee 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -232,21 +232,6 @@ static int start_endpoints(struct snd_usb_substream *subs) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; - if (subs->data_endpoint->iface != subs->sync_endpoint->iface || - subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { - err = usb_set_interface(subs->dev, - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting); - if (err < 0) { - clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); - dev_err(&subs->dev->dev, - "%d:%d: cannot set interface (%d)\n", - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting, err); - return -EIO; - } - } - dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; @@ -512,6 +497,19 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, subs->data_endpoint->sync_master = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + if (err < 0) + return err; + dev_dbg(&dev->dev, "setting usb interface %d:%d\n", + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + snd_usb_set_interface_quirk(dev); + } + return 0; } -- 2.27.0

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[PATCH AUTOSEL 5.4 055/130] ALSA: usb-audio: Handle discrete rates properly in hw constraints
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit bc4e94aa8e72e79598e63a0b73febdcd8aeb541f ] In the current code, when the device provides the discrete sample rate tables with unusual sample rates, the driver tries to gather the whole values from the audioformat entries and create a hw-constraint rule to restrict with this single rate list. This is rather inefficient and may overlook the rates that are associated only with the certain audioformat entries. This patch improves the hw constraint setup by rewriting the existing hw_rule_rate(). The discrete sample rates (identified by rate_table and nr_rates of format entry) are checked in the existing hw_rule_rate() instead of extra rules; in the case of discrete rates, the function compares with each rate table entry and calculates the min/max values from there. For the contiguous rates, the behavior doesn't change. Along with it, snd_usb_pcm_check_knot() and snb_usb_substream rate_list field become superfluous, thus those are dropped. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 1 - sound/usb/pcm.c | 73 ++++++++++------------------------------------ sound/usb/stream.c | 1 - 3 files changed, 15 insertions(+), 60 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index d8ec5caf464de..d619e5e77a305 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -153,7 +153,6 @@ struct snd_usb_substream { u64 formats; /* format bitmasks (all or'ed) */ unsigned int num_formats; /* number of supported audio formats (list) */ struct list_head fmt_list; /* format list */ - struct snd_pcm_hw_constraint_list rate_list; /* limited rates */ spinlock_t lock; int last_frame_number; /* stored frame number */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 1a5e555002b2b..49ad4e7bb70b5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1034,27 +1034,31 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_usb_substream *subs = rule->private; struct audioformat *fp; struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - unsigned int rmin, rmax; + unsigned int rmin, rmax, r; int changed; + int i; hwc_debug("hw_rule_rate: (%d,%d)\n", it->min, it->max); - changed = 0; - rmin = rmax = 0; + rmin = UINT_MAX; + rmax = 0; list_for_each_entry(fp, &subs->fmt_list, list) { if (!hw_check_valid_format(subs, params, fp)) continue; - if (changed++) { - if (rmin > fp->rate_min) - rmin = fp->rate_min; - if (rmax < fp->rate_max) - rmax = fp->rate_max; + if (fp->rate_table && fp->nr_rates) { + for (i = 0; i < fp->nr_rates; i++) { + r = fp->rate_table[i]; + if (!snd_interval_test(it, r)) + continue; + rmin = min(rmin, r); + rmax = max(rmax, r); + } } else { - rmin = fp->rate_min; - rmax = fp->rate_max; + rmin = min(rmin, fp->rate_min); + rmax = max(rmax, fp->rate_max); } } - if (!changed) { + if (rmin > rmax) { hwc_debug(" --> get empty\n"); it->empty = 1; return -EINVAL; @@ -1200,50 +1204,6 @@ static int hw_rule_period_time(struct snd_pcm_hw_params *params, return changed; } -/* - * If the device supports unusual bit rates, does the request meet these? - */ -static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, - struct snd_usb_substream *subs) -{ - struct audioformat *fp; - int *rate_list; - int count = 0, needs_knot = 0; - int err; - - kfree(subs->rate_list.list); - subs->rate_list.list = NULL; - - list_for_each_entry(fp, &subs->fmt_list, list) { - if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) - return 0; - count += fp->nr_rates; - if (fp->rates & SNDRV_PCM_RATE_KNOT) - needs_knot = 1; - } - if (!needs_knot) - return 0; - - subs->rate_list.list = rate_list = - kmalloc_array(count, sizeof(int), GFP_KERNEL); - if (!subs->rate_list.list) - return -ENOMEM; - subs->rate_list.count = count; - subs->rate_list.mask = 0; - count = 0; - list_for_each_entry(fp, &subs->fmt_list, list) { - int i; - for (i = 0; i < fp->nr_rates; i++) - rate_list[count++] = fp->rate_table[i]; - } - err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - &subs->rate_list); - if (err < 0) - return err; - - return 0; -} - /* * set up the runtime hardware information. @@ -1333,9 +1293,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (err < 0) return err; } - err = snd_usb_pcm_check_knot(runtime, subs); - if (err < 0) - return err; return snd_usb_autoresume(subs->stream->chip); } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index d01edd5da6cf8..49c1b8a208582 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -47,7 +47,6 @@ static void free_substream(struct snd_usb_substream *subs) return; /* not initialized */ list_for_each_entry_safe(fp, n, &subs->fmt_list, list) audioformat_free(fp); - kfree(subs->rate_list.list); kfree(subs->str_pd); snd_media_stream_delete(subs); } -- 2.27.0

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[PATCH AUTOSEL 5.4 051/130] ASoC: SOF: IPC: fix implicit type overflow
by Sasha Levin 23 Dec '20

23 Dec '20

From: Curtis Malainey <cujomalainey(a)chromium.org> [ Upstream commit 7c1d0e554a359cca77bfabd2a29b06f5322d172d ] Implicit values may have a length of 15bits (s16) so we need to declare the proper size so we don't get undefined behaviour. This appears to be arch and compiler dependent. This commit is to keep the headers aligned between the firmware and kernel. UBSan discovered this bug in the firmware. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20201120144025.2166023-1-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/sof/header.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index 10f00c08dbb7a..aae673b2bb5e2 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -30,12 +30,12 @@ /* Global Message - Generic */ #define SOF_GLB_TYPE_SHIFT 28 -#define SOF_GLB_TYPE_MASK (0xf << SOF_GLB_TYPE_SHIFT) +#define SOF_GLB_TYPE_MASK (0xfL << SOF_GLB_TYPE_SHIFT) #define SOF_GLB_TYPE(x) ((x) << SOF_GLB_TYPE_SHIFT) /* Command Message - Generic */ #define SOF_CMD_TYPE_SHIFT 16 -#define SOF_CMD_TYPE_MASK (0xfff << SOF_CMD_TYPE_SHIFT) +#define SOF_CMD_TYPE_MASK (0xfffL << SOF_CMD_TYPE_SHIFT) #define SOF_CMD_TYPE(x) ((x) << SOF_CMD_TYPE_SHIFT) /* Global Message Types */ -- 2.27.0

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[PATCH AUTOSEL 5.10 070/217] ASoC: Intel: sof_sdw: add quirk for new TigerLake-SDCA device
by Sasha Levin 23 Dec '20

23 Dec '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 488cdbd8931fe4bc7f374a8b429e81d0e4b7ac76 ] Add quirks for jack detection, rt715 DAI and number of speakers. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Link: https://lore.kernel.org/r/20201111214318.150529-2-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index b29946eb43551..ca968901ac96f 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -48,6 +48,16 @@ static int sof_sdw_quirk_cb(const struct dmi_system_id *id) } static const struct dmi_system_id sof_sdw_quirk_table[] = { + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A32") + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX | + SOF_SDW_FOUR_SPK), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.27.0

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[PATCH AUTOSEL 5.10 066/217] ASoC: Fix vaud18 power leakage of mt6359
by Sasha Levin 23 Dec '20

23 Dec '20

From: "Shane.Chien" <shane.chien(a)mediatek.com> [ Upstream commit 64a70744b77898a15d7a5b2b4dc0fa9523a75cde ] vaud18 is power of mt6359 audio path. It should only enable when audio is used, instead of in boot up stage. Once mt6359 audio path is enabled or disabled, vaud18 is controlled by regulator supply widget "LDO_VAUD18". Due to vaud18 is controlled by regulator dapm macro instead of regmap, the macro MT6359_LDO_VAUD18_CON0 and variable avdd_reg is no used and removed from mt6359.h. Signed-off-by: Shane.Chien <shane.chien(a)mediatek.com> Link: https://lore.kernel.org/r/1604975492-6142-2-git-send-email-shane.chien@medi… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/mt6359.c | 25 +------------------------ sound/soc/codecs/mt6359.h | 8 -------- 2 files changed, 1 insertion(+), 32 deletions(-) diff --git a/sound/soc/codecs/mt6359.c b/sound/soc/codecs/mt6359.c index 81aafb553bdd9..64b26ab969ed8 100644 --- a/sound/soc/codecs/mt6359.c +++ b/sound/soc/codecs/mt6359.c @@ -1833,9 +1833,7 @@ static const struct snd_soc_dapm_widget mt6359_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY_S("CLK_BUF", SUPPLY_SEQ_CLK_BUF, MT6359_DCXO_CW12, RG_XO_AUDIO_EN_M_SFT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY_S("LDO_VAUD18", SUPPLY_SEQ_LDO_VAUD18, - MT6359_LDO_VAUD18_CON0, - RG_LDO_VAUD18_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("LDO_VAUD18", 0, 0), SND_SOC_DAPM_SUPPLY_S("AUDGLB", SUPPLY_SEQ_AUD_GLB, MT6359_AUDDEC_ANA_CON13, RG_AUDGLB_PWRDN_VA32_SFT, 1, NULL, 0), @@ -2697,20 +2695,6 @@ static int mt6359_platform_driver_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, priv); priv->dev = &pdev->dev; - priv->avdd_reg = devm_regulator_get(&pdev->dev, "vaud18"); - if (IS_ERR(priv->avdd_reg)) { - dev_err(&pdev->dev, "%s(), have no vaud18 supply: %ld", - __func__, PTR_ERR(priv->avdd_reg)); - return PTR_ERR(priv->avdd_reg); - } - - ret = regulator_enable(priv->avdd_reg); - if (ret) { - dev_err(&pdev->dev, "%s(), failed to enable regulator!\n", - __func__); - return ret; - } - ret = mt6359_parse_dt(priv); if (ret) { dev_warn(&pdev->dev, "%s() failed to parse dts\n", __func__); @@ -2731,13 +2715,6 @@ static int mt6359_platform_driver_remove(struct platform_device *pdev) dev_dbg(&pdev->dev, "%s(), dev name %s\n", __func__, dev_name(&pdev->dev)); - ret = regulator_disable(priv->avdd_reg); - if (ret) { - dev_err(&pdev->dev, "%s(), failed to disable regulator!\n", - __func__); - return ret; - } - return 0; } diff --git a/sound/soc/codecs/mt6359.h b/sound/soc/codecs/mt6359.h index 3792e534a91b6..e0aebf639e461 100644 --- a/sound/soc/codecs/mt6359.h +++ b/sound/soc/codecs/mt6359.h @@ -135,11 +135,6 @@ /* MT6359_DCXO_CW12 */ #define RG_XO_AUDIO_EN_M_SFT 13 -/* LDO_VAUD18_CON0 */ -#define RG_LDO_VAUD18_EN_SFT 0 -#define RG_LDO_VAUD18_EN_MASK 0x1 -#define RG_LDO_VAUD18_EN_MASK_SFT (0x1 << 0) - /* AUD_TOP_CKPDN_CON0 */ #define RG_VOW13M_CK_PDN_SFT 13 #define RG_VOW13M_CK_PDN_MASK 0x1 @@ -2132,7 +2127,6 @@ #define MT6359_DCXO_CW11 0x7a6 #define MT6359_DCXO_CW12 0x7a8 -#define MT6359_LDO_VAUD18_CON0 0x1c98 #define MT6359_GPIO_MODE0 0xcc #define MT6359_GPIO_MODE0_SET 0xce @@ -2469,7 +2463,6 @@ enum { enum { /* common */ SUPPLY_SEQ_CLK_BUF, - SUPPLY_SEQ_LDO_VAUD18, SUPPLY_SEQ_AUD_GLB, SUPPLY_SEQ_HP_PULL_DOWN, SUPPLY_SEQ_CLKSQ, @@ -2629,7 +2622,6 @@ struct mt6359_priv { int hp_gain_ctl; int hp_hifi_mode; int mtkaif_protocol; - struct regulator *avdd_reg; }; #define CODEC_MT6359_NAME "mtk-codec-mt6359" -- 2.27.0

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[PATCH AUTOSEL 5.10 013/217] ASoC: intel: sof_rt5682: Add quirk for Dooly
by Sasha Levin 23 Dec '20

23 Dec '20

From: Brent Lu <brent.lu(a)intel.com> [ Upstream commit bdd088ce5bfd32b95ab1bd90b49405e7c1f1fff5 ] This DMI product family string of this board is "Google_Hatch" so the DMI quirk will take place. However, this board is using rt1015 speaker amp instead of max98357a specified in the quirk. Therefore, we need an new DMI quirk for this board. Signed-off-by: Brent Lu <brent.lu(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20201030170559.20370-3-brent.lu@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index ddbb9fe7cc06b..c6c33910bc354 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -99,6 +99,24 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(1)), }, + { + /* + * Dooly is hatch family but using rt1015 amp so it + * requires a quirk before "Google_Hatch". + */ + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HP"), + DMI_MATCH(DMI_PRODUCT_NAME, "Dooly"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1015_SPEAKER_AMP_PRESENT | + SOF_RT1015_SPEAKER_AMP_100FS | + SOF_RT5682_SSP_AMP(1)), + }, { .callback = sof_rt5682_quirk_cb, .matches = { -- 2.27.0

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UBSAN: shift-out-of-bounds in snd_usbmidi_get_ms_info
by syzbot 22 Dec '20

22 Dec '20

Hello, syzbot found the following issue on: HEAD commit: 8653b778 Merge tag 'clk-for-linus' of git://git.kernel.org.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=109531a3500000 kernel config: https://syzkaller.appspot.com/x/.config?x=98408202fed1f636 dashboard link: https://syzkaller.appspot.com/bug?extid=92e45ae45543f89e8c88 compiler: gcc (GCC) 10.1.0-syz 20200507 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=1421ec47500000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=12bfa077500000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+92e45ae45543f89e8c88(a)syzkaller.appspotmail.com usb 1-1: config 0 interface 0 altsetting 0 bulk endpoint 0x8A has invalid maxpacket 31 usb 1-1: New USB device found, idVendor=324b, idProduct=5963, bcdDevice= 0.d2 usb 1-1: New USB device strings: Mfr=0, Product=0, SerialNumber=0 usb 1-1: config 0 descriptor?? usb 1-1: MIDIStreaming interface descriptor not found ================================================================================ UBSAN: shift-out-of-bounds in sound/usb/midi.c:1928:8 shift exponent 244 is too large for 32-bit type 'int' CPU: 1 PID: 17 Comm: kworker/1:0 Not tainted 5.10.0-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:79 [inline] dump_stack+0x107/0x163 lib/dump_stack.c:120 ubsan_epilogue+0xb/0x5a lib/ubsan.c:148 __ubsan_handle_shift_out_of_bounds.cold+0xb1/0x181 lib/ubsan.c:395 snd_usbmidi_get_ms_info.cold+0x83/0xd4 sound/usb/midi.c:1928 __snd_usbmidi_create+0x380/0x2320 sound/usb/midi.c:2404 snd_usb_create_quirk+0xa5/0xe0 sound/usb/quirks.c:562 usb_audio_probe+0x623/0x2ab0 sound/usb/card.c:795 --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

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[PATCH 1/2] ALSA: hda/realtek: Apply jack fixup for Quanta NL3
by Chris Chiu 22 Dec '20

22 Dec '20

The Quanta NL3 laptop has both a headphone output jack and a headset jack, on the right edge of the chassis. The pin information suggests that both of these are at the Front. The PulseAudio is confused to differentiate them so one of the jack can neither get the jack sense working nor the audio output. The ALC269_FIXUP_LIFEBOOK chained with ALC269_FIXUP_QUANTA_MUTE can help to differentiate 2 jacks and get the 'Auto-Mute Mode' working correctly. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8616c5624870..bb1010d78717 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8013,6 +8013,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x152d, 0x1082, "Quanta NL3", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x1558, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1401, "Clevo L140[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.20.1

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[PATCH] pcm: Fix a typo in SND_PCM_AUDIO_TSTAMP_TYPE_LAST definition
by Takashi Iwai 22 Dec '20

22 Dec '20

It was wrongly defined with SNDRV_ prefix. Fix it. Fixes: ea02dbd20a53 ("pcm: Add snd_pcm_audio_tstamp_type_t constants") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/pcm.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/pcm.h b/include/pcm.h index cf1eea8bbc9b..e300b951c076 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -361,7 +361,7 @@ typedef enum _snd_pcm_audio_tstamp_type { SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /**< link time reported by sample or wallclock counter, not reset on startup */ SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /**< link time estimated indirectly */ SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /**< link time synchronized with system time */ - SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED + SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED } snd_pcm_audio_tstamp_type_t; typedef struct _snd_pcm_audio_tstamp_config { -- 2.26.2

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