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

[PATCH AUTOSEL 4.19 010/206] ASoC: kirkwood: fix IRQ error handling
by Sasha Levin 18 Sep '20

18 Sep '20

From: Russell King <rmk+kernel(a)armlinux.org.uk> [ Upstream commit 175fc928198236037174e5c5c066fe3c4691903e ] Propagate the error code from request_irq(), rather than returning -EBUSY. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> Link: https://lore.kernel.org/r/E1iNIqh-0000tW-EZ@rmk-PC.armlinux.org.uk Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index c6a58520d377a..255cc45905b81 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -136,7 +136,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, "kirkwood-i2s", priv); if (err) - return -EBUSY; + return err; /* * Enable Error interrupts. We're only ack'ing them but -- 2.25.1

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[PATCH AUTOSEL 5.4 328/330] ALSA: hda: Workaround for spurious wakeups on some Intel platforms
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit a6630529aecb5a3e84370c376ed658e892e6261e ] We've received a regression report on Intel HD-audio controller that wakes up immediately after S3 suspend. The bisection leads to the commit c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed"). This commit replaces the system-suspend to use pm_runtime_force_suspend() instead of the direct call of __azx_runtime_suspend(). However, by some really mysterious reason, pm_runtime_force_suspend() causes a spurious wakeup (although it calls the same __azx_runtime_suspend() internally). As an ugly workaround for now, revert the behavior to call __azx_runtime_suspend() and __azx_runtime_resume() for those old Intel platforms that may exhibit such a problem, while keeping the new standard pm_runtime_force_suspend() and pm_runtime_force_resume() pair for the remaining chips. Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208649 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200727164443.4233-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 17 ++++++++++++++--- 2 files changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 82e26442724ba..a356fb0e57738 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -41,7 +41,7 @@ /* 24 unused */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -/* 27 unused */ +#define AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP (1 << 27) /* Workaround for spurious wakeups after suspend */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 754e4d1a86b57..590ea262f2e20 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -295,7 +295,8 @@ enum { /* PCH for HSW/BDW; with runtime PM */ /* no i915 binding for this as HSW/BDW has another controller for HDMI */ #define AZX_DCAPS_INTEL_PCH \ - (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME) + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) /* HSW HDMI */ #define AZX_DCAPS_INTEL_HASWELL \ @@ -1026,7 +1027,14 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - pm_runtime_force_suspend(dev); + /* An ugly workaround: direct call of __azx_runtime_suspend() and + * __azx_runtime_resume() for old Intel platforms that suffer from + * spurious wakeups after S3 suspend + */ + if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) + __azx_runtime_suspend(chip); + else + pm_runtime_force_suspend(dev); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1054,7 +1062,10 @@ static int azx_resume(struct device *dev) if (azx_acquire_irq(chip, 1) < 0) return -EIO; - pm_runtime_force_resume(dev); + if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) + __azx_runtime_resume(chip, false); + else + pm_runtime_force_resume(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); -- 2.25.1

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[PATCH AUTOSEL 5.4 327/330] ALSA: hda: Always use jackpoll helper for jack update after resume
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 8d6762af302d69f76fa788a277a56a9d9cd275d5 ] HD-audio codec driver applies a tricky procedure to forcibly perform the runtime resume by mimicking the usage count even if the device has been runtime-suspended beforehand. This was needed to assure to trigger the jack detection update after the system resume. And recently we also applied the similar logic to the HD-audio controller side. However this seems leading to some inconsistency, and eventually PCI controller gets screwed up. This patch is an attempt to fix and clean up those behavior: instead of the tricky runtime resume procedure, the existing jackpoll work is scheduled when such a forced codec resume is required. The jackpoll work will power up the codec, and this alone should suffice for the jack status update in usual cases. If the extra polling is requested (by checking codec->jackpoll_interval), the manual update is invoked after that, and the codec is powered down again. Also, we filter the spurious wake up of the codec from the controller runtime resume by checking codec->relaxed_resume flag. If this flag is set, basically we don't need to wake up explicitly, but it's supposed to be done via the audio component notifier. Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed") Link: https://lore.kernel.org/r/20200422203744.26299-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_codec.c | 28 +++++++++++++++++----------- sound/pci/hda/hda_intel.c | 17 ++--------------- 2 files changed, 19 insertions(+), 26 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 12da263fb02ba..6da296def283e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -641,8 +641,18 @@ static void hda_jackpoll_work(struct work_struct *work) struct hda_codec *codec = container_of(work, struct hda_codec, jackpoll_work.work); - snd_hda_jack_set_dirty_all(codec); - snd_hda_jack_poll_all(codec); + /* for non-polling trigger: we need nothing if already powered on */ + if (!codec->jackpoll_interval && snd_hdac_is_power_on(&codec->core)) + return; + + /* the power-up/down sequence triggers the runtime resume */ + snd_hda_power_up_pm(codec); + /* update jacks manually if polling is required, too */ + if (codec->jackpoll_interval) { + snd_hda_jack_set_dirty_all(codec); + snd_hda_jack_poll_all(codec); + } + snd_hda_power_down_pm(codec); if (!codec->jackpoll_interval) return; @@ -2958,18 +2968,14 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = hda_codec_need_resume(codec); int ret; - /* The get/put pair below enforces the runtime resume even if the - * device hasn't been used at suspend time. This trick is needed to - * update the jack state change during the sleep. - */ - if (forced_resume) - pm_runtime_get_noresume(dev); ret = pm_runtime_force_resume(dev); - if (forced_resume) - pm_runtime_put(dev); + /* schedule jackpoll work for jack detection update */ + if (codec->jackpoll_interval || + (pm_runtime_suspended(dev) && hda_codec_need_resume(codec))) + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); return ret; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a6e8aaa091c7d..754e4d1a86b57 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1002,7 +1002,8 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt) if (status && from_rt) { list_for_each_codec(codec, &chip->bus) - if (status & (1 << codec->addr)) + if (!codec->relaxed_resume && + (status & (1 << codec->addr))) schedule_delayed_work(&codec->jackpoll_work, codec->jackpoll_interval); } @@ -1041,9 +1042,7 @@ static int azx_suspend(struct device *dev) static int azx_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); - struct hda_codec *codec; struct azx *chip; - bool forced_resume = false; if (!azx_is_pm_ready(card)) return 0; @@ -1055,19 +1054,7 @@ static int azx_resume(struct device *dev) if (azx_acquire_irq(chip, 1) < 0) return -EIO; - /* check for the forced resume */ - list_for_each_codec(codec, &chip->bus) { - if (hda_codec_need_resume(codec)) { - forced_resume = true; - break; - } - } - - if (forced_resume) - pm_runtime_get_noresume(dev); pm_runtime_force_resume(dev); - if (forced_resume) - pm_runtime_put(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); -- 2.25.1

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[PATCH AUTOSEL 5.4 299/330] ASoC: img-i2s-out: Fix runtime PM imbalance on error
by Sasha Levin 18 Sep '20

18 Sep '20

From: Dinghao Liu <dinghao.liu(a)zju.edu.cn> [ Upstream commit 65bd91dd6957390c42a0491b9622cf31a2cdb140 ] pm_runtime_get_sync() increments the runtime PM usage counter even the call returns an error code. Thus a pairing decrement is needed on the error handling path to keep the counter balanced. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> Link: https://lore.kernel.org/r/20200529012230.5863-1-dinghao.liu@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/img/img-i2s-out.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index 4b18534096336..9c4212f2f7269 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -347,8 +347,10 @@ static int img_i2s_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) chan_control_mask = IMG_I2S_OUT_CHAN_CTL_CLKT_MASK; ret = pm_runtime_get_sync(i2s->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(i2s->dev); return ret; + } img_i2s_out_disable(i2s); @@ -488,8 +490,10 @@ static int img_i2s_out_probe(struct platform_device *pdev) goto err_pm_disable; } ret = pm_runtime_get_sync(&pdev->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(&pdev->dev); goto err_suspend; + } reg = IMG_I2S_OUT_CTL_FRM_SIZE_MASK; img_i2s_out_writel(i2s, reg, IMG_I2S_OUT_CTL); -- 2.25.1

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[PATCH AUTOSEL 5.4 267/330] ALSA: hda: Fix potential race in unsol event handler
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c637fa151259c0f74665fde7cba5b7eac1417ae5 ] The unsol event handling code has a loop retrieving the read/write indices and the arrays without locking while the append to the array may happen concurrently. This may lead to some inconsistency. Although there hasn't been any proof of this bad results, it's still safer to protect the racy accesses. This patch adds the spinlock protection around the unsol handling loop for addressing it. Here we take bus->reg_lock as the writer side snd_hdac_bus_queue_event() is also protected by that lock. Link: https://lore.kernel.org/r/20200516062556.30951-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_bus.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 8f19876244ebe..53be2cac98e7c 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -158,6 +158,7 @@ static void snd_hdac_bus_process_unsol_events(struct work_struct *work) struct hdac_driver *drv; unsigned int rp, caddr, res; + spin_lock_irq(&bus->reg_lock); while (bus->unsol_rp != bus->unsol_wp) { rp = (bus->unsol_rp + 1) % HDA_UNSOL_QUEUE_SIZE; bus->unsol_rp = rp; @@ -169,10 +170,13 @@ static void snd_hdac_bus_process_unsol_events(struct work_struct *work) codec = bus->caddr_tbl[caddr & 0x0f]; if (!codec || !codec->dev.driver) continue; + spin_unlock_irq(&bus->reg_lock); drv = drv_to_hdac_driver(codec->dev.driver); if (drv->unsol_event) drv->unsol_event(codec, res); + spin_lock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); } /** -- 2.25.1

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[PATCH AUTOSEL 5.4 234/330] ALSA: hda: Skip controller resume if not needed
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c4c8dd6ef807663e42a5f04ea77cd62029eb99fa ] The HD-audio controller does system-suspend and resume operations by directly calling its helpers __azx_runtime_suspend() and __azx_runtime_resume(). However, in general, we don't have to resume always the device fully at the system resume; typically, if a device has been runtime-suspended, we can leave it to runtime resume. Usually for achieving this, the driver would call pm_runtime_force_suspend() and pm_runtime_force_resume() pairs in the system suspend and resume ops. Unfortunately, this doesn't work for the resume path in our case. For handling the jack detection at the system resume, a child codec device may need the (literally) forcibly resume even if it's been runtime-suspended, and for that, the controller device must be also resumed even if it's been suspended. This patch is an attempt to improve the situation. It replaces the direct __azx_runtime_suspend()/_resume() calls with with pm_runtime_force_suspend() and pm_runtime_force_resume() with a slight trick as we've done for the codec side. More exactly: - azx_has_pm_runtime() check is dropped from azx_runtime_suspend() and azx_runtime_resume(), so that it can be properly executed from the system-suspend/resume path - The WAKEEN handling depends on the card's power state now; it's set and cleared only for the runtime-suspend - azx_resume() checks whether any codec may need the forcible resume beforehand. If the forcible resume is required, it does temporary PM refcount up/down for actually triggering the runtime resume. - A new helper function, hda_codec_need_resume(), is introduced for checking whether the codec needs a forcible runtime-resume, and the existing code is rewritten with that. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-6-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/hda_codec.h | 5 +++++ sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_intel.c | 38 +++++++++++++++++++++++++++----------- 3 files changed, 33 insertions(+), 12 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 9a0393cf024c2..65c056ce91128 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -494,6 +494,11 @@ void snd_hda_update_power_acct(struct hda_codec *codec); static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {} #endif +static inline bool hda_codec_need_resume(struct hda_codec *codec) +{ + return !codec->relaxed_resume && codec->jacktbl.used; +} + #ifdef CONFIG_SND_HDA_PATCH_LOADER /* * patch firmware diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 103011e7285a3..12da263fb02ba 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2958,7 +2958,7 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used; + bool forced_resume = hda_codec_need_resume(codec); int ret; /* The get/put pair below enforces the runtime resume even if the diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7353d2ec359ae..a6e8aaa091c7d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1025,7 +1025,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - __azx_runtime_suspend(chip); + pm_runtime_force_suspend(dev); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1041,7 +1041,9 @@ static int azx_suspend(struct device *dev) static int azx_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); + struct hda_codec *codec; struct azx *chip; + bool forced_resume = false; if (!azx_is_pm_ready(card)) return 0; @@ -1052,7 +1054,20 @@ static int azx_resume(struct device *dev) chip->msi = 0; if (azx_acquire_irq(chip, 1) < 0) return -EIO; - __azx_runtime_resume(chip, false); + + /* check for the forced resume */ + list_for_each_codec(codec, &chip->bus) { + if (hda_codec_need_resume(codec)) { + forced_resume = true; + break; + } + } + + if (forced_resume) + pm_runtime_get_noresume(dev); + pm_runtime_force_resume(dev); + if (forced_resume) + pm_runtime_put(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); @@ -1099,12 +1114,12 @@ static int azx_runtime_suspend(struct device *dev) if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - if (!azx_has_pm_runtime(chip)) - return 0; /* enable controller wake up event */ - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | - STATESTS_INT_MASK); + if (snd_power_get_state(card) == SNDRV_CTL_POWER_D0) { + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + } __azx_runtime_suspend(chip); trace_azx_runtime_suspend(chip); @@ -1115,17 +1130,18 @@ static int azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; + bool from_rt = snd_power_get_state(card) == SNDRV_CTL_POWER_D0; if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - if (!azx_has_pm_runtime(chip)) - return 0; - __azx_runtime_resume(chip, true); + __azx_runtime_resume(chip, from_rt); /* disable controller Wake Up event*/ - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & - ~STATESTS_INT_MASK); + if (from_rt) { + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & + ~STATESTS_INT_MASK); + } trace_azx_runtime_resume(chip); return 0; -- 2.25.1

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[PATCH AUTOSEL 5.4 220/330] ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor
by Sasha Levin 18 Sep '20

18 Sep '20

From: Andreas Steinmetz <ast(a)domdv.de> [ Upstream commit 5c6cd7021a05a02fcf37f360592d7c18d4d807fb ] The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra endpoint descriptor. The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00 As the code in snd_usbmidi_get_ms_info() looks only at the first extra descriptor to find USB_DT_CS_ENDPOINT the device as such is recognized but there is neither input nor output configured. The patch iterates through the extra descriptors to find the proper one. With this patch the device is correctly configured. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> Link: https://lore.kernel.org/r/1c3b431a86f69e1d60745b6110cdb93c299f120b.camel@do… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/midi.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 0cb4142b05f64..bc9068b616bb9 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1827,6 +1827,28 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, return 0; } +static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor( + struct usb_host_endpoint *hostep) +{ + unsigned char *extra = hostep->extra; + int extralen = hostep->extralen; + + while (extralen > 3) { + struct usb_ms_endpoint_descriptor *ms_ep = + (struct usb_ms_endpoint_descriptor *)extra; + + if (ms_ep->bLength > 3 && + ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT && + ms_ep->bDescriptorSubtype == UAC_MS_GENERAL) + return ms_ep; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + /* * Returns MIDIStreaming device capabilities. */ @@ -1864,11 +1886,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; - if (hostep->extralen < 4 || - ms_ep->bLength < 4 || - ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) + ms_ep = find_usb_ms_endpoint_descriptor(hostep); + if (!ms_ep) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { -- 2.25.1

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[PATCH AUTOSEL 5.4 149/330] ASoC: SOF: ipc: check ipc return value before data copy
by Sasha Levin 18 Sep '20

18 Sep '20

From: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> [ Upstream commit 1919b42ca4ad75a2397081164661af3ce5a7b8f4 ] In tx_wait_done the ipc payload is copied before the DSP transaction error code is checked. This might lead to corrupted data in kernel side even though the error would be handled later. It is also pointless to copy the data in case of error. So change the order of error check and copy. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200228231850.9226-3-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/ipc.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index e7b1a80e2a14c..f38f651da2246 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -215,15 +215,17 @@ static int tx_wait_done(struct snd_sof_ipc *ipc, struct snd_sof_ipc_msg *msg, snd_sof_trace_notify_for_error(ipc->sdev); ret = -ETIMEDOUT; } else { - /* copy the data returned from DSP */ ret = msg->reply_error; - if (msg->reply_size) - memcpy(reply_data, msg->reply_data, msg->reply_size); - if (ret < 0) + if (ret < 0) { dev_err(sdev->dev, "error: ipc error for 0x%x size %zu\n", hdr->cmd, msg->reply_size); - else + } else { ipc_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd); + if (msg->reply_size) + /* copy the data returned from DSP */ + memcpy(reply_data, msg->reply_data, + msg->reply_size); + } } return ret; -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 131/330] ALSA: usb-audio: Don't create a mixer element with bogus volume range
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit e9a0ef0b5ddcbc0d56c65aefc0f18d16e6f71207 ] Some USB-audio descriptors provide a bogus volume range (e.g. volume min and max are identical), which confuses user-space. This patch makes the driver skipping such a control element. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206221 Link: https://lore.kernel.org/r/20200214144928.23628-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 9079c380228fc..8aa96ed0b1b56 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1684,6 +1684,16 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, /* get min/max values */ get_min_max_with_quirks(cval, 0, kctl); + /* skip a bogus volume range */ + if (cval->max <= cval->min) { + usb_audio_dbg(mixer->chip, + "[%d] FU [%s] skipped due to invalid volume\n", + cval->head.id, kctl->id.name); + snd_ctl_free_one(kctl); + return; + } + + if (control == UAC_FU_VOLUME) { check_mapped_dB(map, cval); if (cval->dBmin < cval->dBmax || !cval->initialized) { -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 116/330] ALSA: hda: Clear RIRB status before reading WP
by Sasha Levin 18 Sep '20

18 Sep '20

From: Mohan Kumar <mkumard(a)nvidia.com> [ Upstream commit 6d011d5057ff88ee556c000ac6fe0be23bdfcd72 ] RIRB interrupt status getting cleared after the write pointer is read causes a race condition, where last response(s) into RIRB may remain unserviced by IRQ, eventually causing azx_rirb_get_response to fall back to polling mode. Clearing the RIRB interrupt status ahead of write pointer access ensures that this condition is avoided. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> Signed-off-by: Viswanath L <viswanathl(a)nvidia.com> Link: https://lore.kernel.org/r/1580983853-351-1-git-send-email-viswanathl@nvidia… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_controller.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 76b507058cb4d..5e6081750bd9b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1159,16 +1159,23 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) if (snd_hdac_bus_handle_stream_irq(bus, status, stream_update)) active = true; - /* clear rirb int */ status = azx_readb(chip, RIRBSTS); if (status & RIRB_INT_MASK) { + /* + * Clearing the interrupt status here ensures that no + * interrupt gets masked after the RIRB wp is read in + * snd_hdac_bus_update_rirb. This avoids a possible + * race condition where codec response in RIRB may + * remain unserviced by IRQ, eventually falling back + * to polling mode in azx_rirb_get_response. + */ + azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); active = true; if (status & RIRB_INT_RESPONSE) { if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) udelay(80); snd_hdac_bus_update_rirb(bus); } - azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } } while (active && ++repeat < 10); -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 077/330] ALSA: hda: enable regmap internal locking
by Sasha Levin 18 Sep '20

18 Sep '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 8e85def5723eccea30ebf22645673692ab8cb3e2 ] This reverts commit 42ec336f1f9d ("ALSA: hda: Disable regmap internal locking"). Without regmap locking, there is a race between snd_hda_codec_amp_init() and PM callbacks issuing regcache_sync(). This was caught by following kernel warning trace: <4> [358.080081] WARNING: CPU: 2 PID: 4157 at drivers/base/regmap/regcache.c:498 regcache_cache_only+0xf5/0x130 [...] <4> [358.080148] Call Trace: <4> [358.080158] snd_hda_codec_amp_init+0x4e/0x100 [snd_hda_codec] <4> [358.080169] snd_hda_codec_amp_init_stereo+0x40/0x80 [snd_hda_codec] Suggested-by: Takashi Iwai <tiwai(a)suse.de> BugLink: https://gitlab.freedesktop.org/drm/intel/issues/592 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200108180856.5194-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_regmap.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 2596a881186fa..49780399c2849 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -363,7 +363,6 @@ static const struct regmap_config hda_regmap_cfg = { .reg_write = hda_reg_write, .use_single_read = true, .use_single_write = true, - .disable_locking = true, }; /** -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 065/330] ASoC: max98090: remove msleep in PLL unlocked workaround
by Sasha Levin 18 Sep '20

18 Sep '20

From: Tzung-Bi Shih <tzungbi(a)google.com> [ Upstream commit acb874a7c049ec49d8fc66c893170fb42c01bdf7 ] It was observed Baytrail-based chromebooks could cause continuous PLL unlocked when using playback stream and capture stream simultaneously. Specifically, starting a capture stream after started a playback stream. As a result, the audio data could corrupt or turn completely silent. As the datasheet suggested, the maximum PLL lock time should be 7 msec. The workaround resets the codec softly by toggling SHDN off and on if PLL failed to lock for 10 msec. Notably, there is no suggested hold time for SHDN off. On Baytrail-based chromebooks, it would easily happen continuous PLL unlocked if there is a 10 msec delay between SHDN off and on. Removes the msleep(). Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20191122073114.219945-2-tzungbi@google.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 45da2b51543e7..6b9d326e11b07 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2112,10 +2112,16 @@ static void max98090_pll_work(struct max98090_priv *max98090) dev_info_ratelimited(component->dev, "PLL unlocked\n"); + /* + * As the datasheet suggested, the maximum PLL lock time should be + * 7 msec. The workaround resets the codec softly by toggling SHDN + * off and on if PLL failed to lock for 10 msec. Notably, there is + * no suggested hold time for SHDN off. + */ + /* Toggle shutdown OFF then ON */ snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, 0); - msleep(10); snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, M98090_SHDNN_MASK); -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 019/330] ASoC: kirkwood: fix IRQ error handling
by Sasha Levin 18 Sep '20

18 Sep '20

From: Russell King <rmk+kernel(a)armlinux.org.uk> [ Upstream commit 175fc928198236037174e5c5c066fe3c4691903e ] Propagate the error code from request_irq(), rather than returning -EBUSY. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> Link: https://lore.kernel.org/r/E1iNIqh-0000tW-EZ@rmk-PC.armlinux.org.uk Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 6f69f314f2c2a..d2d5c25bf5502 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -132,7 +132,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, "kirkwood-i2s", priv); if (err) - return -EBUSY; + return err; /* * Enable Error interrupts. We're only ack'ing them but -- 2.25.1

1 0

[PATCH v2] ASoC: dt-bindings: correct interrupt flags in examples
by Krzysztof Kozlowski 17 Sep '20

17 Sep '20

GPIO_ACTIVE_x flags are not correct in the context of interrupt flags. These are simple defines so they could be used in DTS but they will not have the same meaning: 1. GPIO_ACTIVE_HIGH = 0 = IRQ_TYPE_NONE 2. GPIO_ACTIVE_LOW = 1 = IRQ_TYPE_EDGE_RISING Correct the interrupt flags, assuming the author of the code wanted same logical behavior behind the name "ACTIVE_xxx", this is: ACTIVE_HIGH => IRQ_TYPE_LEVEL_HIGH Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes since v1: 1. Add acks --- Documentation/devicetree/bindings/sound/max98090.txt | 2 +- Documentation/devicetree/bindings/sound/rt5640.txt | 2 +- Documentation/devicetree/bindings/sound/rt5659.txt | 2 +- Documentation/devicetree/bindings/sound/rt5665.txt | 2 +- Documentation/devicetree/bindings/sound/rt5668.txt | 2 +- Documentation/devicetree/bindings/sound/rt5677.txt | 2 +- Documentation/devicetree/bindings/sound/rt5682.txt | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index 7e1bbd5c27fd..39d640294c62 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -55,5 +55,5 @@ audio-codec@10 { compatible = "maxim,max98090"; reg = <0x10>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(H, 4) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(H, 4) IRQ_TYPE_LEVEL_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index e40e4893eed8..ff1228713f7e 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -88,7 +88,7 @@ rt5640 { compatible = "realtek,rt5640"; reg = <0x1c>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5659.txt b/Documentation/devicetree/bindings/sound/rt5659.txt index 1766e0543fc5..56788f50b6cf 100644 --- a/Documentation/devicetree/bindings/sound/rt5659.txt +++ b/Documentation/devicetree/bindings/sound/rt5659.txt @@ -72,7 +72,7 @@ rt5659 { compatible = "realtek,rt5659"; reg = <0x1b>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5665.txt b/Documentation/devicetree/bindings/sound/rt5665.txt index 8df170506986..f6ca96b4ce98 100644 --- a/Documentation/devicetree/bindings/sound/rt5665.txt +++ b/Documentation/devicetree/bindings/sound/rt5665.txt @@ -62,7 +62,7 @@ rt5659 { compatible = "realtek,rt5665"; reg = <0x1b>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5668.txt b/Documentation/devicetree/bindings/sound/rt5668.txt index c88b96e7764b..a2b7e9a2f2f3 100644 --- a/Documentation/devicetree/bindings/sound/rt5668.txt +++ b/Documentation/devicetree/bindings/sound/rt5668.txt @@ -41,7 +41,7 @@ rt5668 { compatible = "realtek,rt5668b"; reg = <0x1a>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index 1b3c13d206ff..da2430099181 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -64,7 +64,7 @@ rt5677 { compatible = "realtek,rt5677"; reg = <0x2c>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; gpio-controller; #gpio-cells = <2>; diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index ade1ece8b45f..707fa98d1310 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -58,7 +58,7 @@ rt5682 { compatible = "realtek,rt5682i"; reg = <0x1a>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; -- 2.17.1

2 1

Re: [trivial PATCH] treewide: Convert switch/case fallthrough; to break;
by Jacob Keller 17 Sep '20

17 Sep '20

On 9/9/2020 1:55 PM, Keith Busch wrote: > On Wed, Sep 09, 2020 at 01:06:39PM -0700, Joe Perches wrote: >> diff --git a/crypto/tcrypt.c b/crypto/tcrypt.c >> index eea0f453cfb6..8aac5bc60f4c 100644 >> --- a/crypto/tcrypt.c >> +++ b/crypto/tcrypt.c >> @@ -2464,7 +2464,7 @@ static int do_test(const char *alg, u32 type, u32 mask, int m, u32 num_mb) >> test_hash_speed("streebog512", sec, >> generic_hash_speed_template); >> if (mode > 300 && mode < 400) break; >> - fallthrough; >> + break; >> case 399: >> break; > > Just imho, this change makes the preceding 'if' look even more > pointless. Maybe the fallthrough was a deliberate choice? Not that my > opinion matters here as I don't know this module, but it looked a bit > odd to me. > Yea this does look very odd..

1 0

[PATCH v2 0/3] ASoC: fsl_sai: update the register list
by Shengjiu Wang 17 Sep '20

17 Sep '20

As sai ip is upgraded, so update sai register list. Shengjiu Wang (3): ASoC: fsl_sai: Add new added registers and new bit definition ASoC: fsl_sai: Add fsl_sai_check_version function ASoC: fsl_sai: Set MCLK input or output direction changes in v2: - update commit message for first commit - Add acked-by Nicolin sound/soc/fsl/fsl_sai.c | 77 ++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_sai.h | 87 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 164 insertions(+) -- 2.27.0

3 5

Re: [PATCH] ASoC: fsl_audmix: make clock and output src write only
by Mark Brown 17 Sep '20

17 Sep '20

On Mon, 14 Sep 2020 20:24:34 +0300, Viorel Suman (OSS) wrote: > "alsactl -f state.conf store/restore" sequence fails because setting > "mixing clock source" and "output source" requires active TDM clock > being started for configuration propagation. Make these two controls > write only so that their values are not stored at "alsactl store". Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: fsl_audmix: make clock and output src write only commit: 944c517b8c838832a166f1c89afbf8724f4a6b49 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

[PATCH] ASoC: dt-bindings: aries-wm8994: Match compatibles with enum
by Krzysztof Kozlowski 17 Sep '20

17 Sep '20

The common pattern for enumerating compatibles is enum, not oneOf. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- .../bindings/sound/samsung,aries-wm8994.yaml | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml index 902a0b66628e..1c6947294825 100644 --- a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -11,12 +11,11 @@ maintainers: properties: compatible: - oneOf: - - const: samsung,aries-wm8994 - description: With FM radio and modem master - - - const: samsung,fascinate4g-wm8994 - description: Without FM radio and modem slave + enum: + # With FM radio and modem master + - samsung,aries-wm8994 + # Without FM radio and modem slave + - samsung,fascinate4g-wm8994 model: $ref: /schemas/types.yaml#/definitions/string -- 2.17.1

3 2

[PATCH 1/6] ASoC: tlv320adcx140: Idle the device while writing registers
by Dan Murphy 17 Sep '20

17 Sep '20

It was observed that if the device was active and register writes were performed there were some unwanted behaviors particularly when writing the word length and some filter options. So when writing to the device the device should be placed in sleep mode and then exit sleep mode once the register update is complete. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 43 ++++++++++++++++++++++++-------- sound/soc/codecs/tlv320adcx140.h | 2 ++ 2 files changed, 34 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 8efe20605f9b..3909c1cf52be 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -30,7 +30,7 @@ struct adcx140_priv { struct regmap *regmap; struct device *dev; - int micbias_vg; + bool micbias_vg; unsigned int dai_fmt; unsigned int tdm_delay; @@ -614,11 +614,26 @@ static int adcx140_reset(struct adcx140_priv *adcx140) return ret; } +static void adcx140_pwr_ctrl(struct adcx140_priv *adcx140, bool power_state) +{ + int pwr_ctrl = 0; + + if (power_state) + pwr_ctrl = ADCX140_PWR_CFG_ADC_PDZ | ADCX140_PWR_CFG_PLL_PDZ; + + if (adcx140->micbias_vg && power_state) + pwr_ctrl |= ADCX140_PWR_CFG_BIAS_PDZ; + + regmap_update_bits(adcx140->regmap, ADCX140_PWR_CFG, + ADCX140_PWR_CTRL_MSK, pwr_ctrl); +} + static int adcx140_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); u8 data = 0; switch (params_width(params)) { @@ -640,9 +655,13 @@ static int adcx140_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + adcx140_pwr_ctrl(adcx140, false); + snd_soc_component_update_bits(component, ADCX140_ASI_CFG0, ADCX140_WORD_LEN_MSK, data); + adcx140_pwr_ctrl(adcx140, true); + return 0; } @@ -709,6 +728,8 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, adcx140->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + adcx140_pwr_ctrl(adcx140, false); + snd_soc_component_update_bits(component, ADCX140_ASI_CFG0, ADCX140_FSYNCINV_BIT | ADCX140_BCLKINV_BIT | @@ -721,6 +742,7 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, snd_soc_component_update_bits(component, ADCX140_ASI_CFG1, ADCX140_TX_OFFSET_MASK, offset); + adcx140_pwr_ctrl(adcx140, true); return 0; } @@ -818,12 +840,11 @@ static int adcx140_codec_probe(struct snd_soc_component *component) ret = device_property_read_u32(adcx140->dev, "ti,mic-bias-source", &bias_source); - if (ret) + if (ret || bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { bias_source = ADCX140_MIC_BIAS_VAL_VREF; - - if (bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { - dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); - return -EINVAL; + adcx140->micbias_vg = false; + } else { + adcx140->micbias_vg = true; } ret = device_property_read_u32(adcx140->dev, "ti,vref-source", @@ -906,6 +927,8 @@ static int adcx140_codec_probe(struct snd_soc_component *component) ADCX140_MIC_BIAS_VREF_MSK, bias_cfg); if (ret) dev_err(adcx140->dev, "setting MIC bias failed %d\n", ret); + + adcx140_pwr_ctrl(adcx140, true); out: return ret; } @@ -914,21 +937,19 @@ static int adcx140_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); - int pwr_cfg = 0; switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - pwr_cfg = ADCX140_PWR_CFG_BIAS_PDZ | ADCX140_PWR_CFG_PLL_PDZ | - ADCX140_PWR_CFG_ADC_PDZ; + adcx140_pwr_ctrl(adcx140, true); break; case SND_SOC_BIAS_OFF: - pwr_cfg = 0x0; + adcx140_pwr_ctrl(adcx140, false); break; } - return regmap_write(adcx140->regmap, ADCX140_PWR_CFG, pwr_cfg); + return 0; } static const struct snd_soc_component_driver soc_codec_driver_adcx140 = { diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index eedbc1d7221f..94c6d1fd2977 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -123,6 +123,7 @@ #define ADCX140_MIC_BIAS_VREF_1375V 2 #define ADCX140_MIC_BIAS_VREF_MSK GENMASK(1, 0) +#define ADCX140_PWR_CTRL_MSK GENMASK(7, 5) #define ADCX140_PWR_CFG_BIAS_PDZ BIT(7) #define ADCX140_PWR_CFG_ADC_PDZ BIT(6) #define ADCX140_PWR_CFG_PLL_PDZ BIT(5) @@ -145,4 +146,5 @@ #define ADCX140_GPO_CFG_MAX 4 #define ADCX140_GPO_DRV_MAX 5 + #endif /* _TLV320ADCX140_ */ -- 2.28.0

2 9

[PATCH] ASoC: qcom: lpass-sc7180: Add MODULE_DEVICE_TABLE
by Douglas Anderson 17 Sep '20

17 Sep '20

The lpass-sc7180 driver can be built as a module but is lacking a MODULE_DEVICE_TABLE. This means it won't auto-load. Fix this oversight. Fixes: 24caf8d9eb10 ("ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio") Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- sound/soc/qcom/lpass-sc7180.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 167bf2cbc556..a8a3d8f8f567 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -200,6 +200,7 @@ static const struct of_device_id sc7180_lpass_cpu_device_id[] = { {.compatible = "qcom,sc7180-lpass-cpu", .data = &sc7180_data}, {} }; +MODULE_DEVICE_TABLE(of, sc7180_lpass_cpu_device_id); static struct platform_driver sc7180_lpass_cpu_platform_driver = { .driver = { -- 2.28.0.618.gf4bc123cb7-goog

2 1

[PATCH] ASoC: dt-bindings: Correct interrupt flags in examples
by Krzysztof Kozlowski 17 Sep '20

17 Sep '20

GPIO_ACTIVE_x flags are not correct in the context of interrupt flags. These are simple defines so they could be used in DTS but they will not have the same meaning: 1. GPIO_ACTIVE_HIGH = 0 = IRQ_TYPE_NONE 2. GPIO_ACTIVE_LOW = 1 = IRQ_TYPE_EDGE_RISING Correct the interrupt flags, assuming the author of the code wanted some logical behavior behind the name "ACTIVE_xxx", this is: ACTIVE_HIGH => IRQ_TYPE_LEVEL_HIGH Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Documentation/devicetree/bindings/sound/max98090.txt | 2 +- Documentation/devicetree/bindings/sound/rt5640.txt | 2 +- Documentation/devicetree/bindings/sound/rt5659.txt | 2 +- Documentation/devicetree/bindings/sound/rt5665.txt | 2 +- Documentation/devicetree/bindings/sound/rt5668.txt | 2 +- Documentation/devicetree/bindings/sound/rt5677.txt | 2 +- Documentation/devicetree/bindings/sound/rt5682.txt | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index 7e1bbd5c27fd..39d640294c62 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -55,5 +55,5 @@ audio-codec@10 { compatible = "maxim,max98090"; reg = <0x10>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(H, 4) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(H, 4) IRQ_TYPE_LEVEL_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index e40e4893eed8..ff1228713f7e 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -88,7 +88,7 @@ rt5640 { compatible = "realtek,rt5640"; reg = <0x1c>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5659.txt b/Documentation/devicetree/bindings/sound/rt5659.txt index 1766e0543fc5..56788f50b6cf 100644 --- a/Documentation/devicetree/bindings/sound/rt5659.txt +++ b/Documentation/devicetree/bindings/sound/rt5659.txt @@ -72,7 +72,7 @@ rt5659 { compatible = "realtek,rt5659"; reg = <0x1b>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5665.txt b/Documentation/devicetree/bindings/sound/rt5665.txt index 8df170506986..f6ca96b4ce98 100644 --- a/Documentation/devicetree/bindings/sound/rt5665.txt +++ b/Documentation/devicetree/bindings/sound/rt5665.txt @@ -62,7 +62,7 @@ rt5659 { compatible = "realtek,rt5665"; reg = <0x1b>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5668.txt b/Documentation/devicetree/bindings/sound/rt5668.txt index c88b96e7764b..a2b7e9a2f2f3 100644 --- a/Documentation/devicetree/bindings/sound/rt5668.txt +++ b/Documentation/devicetree/bindings/sound/rt5668.txt @@ -41,7 +41,7 @@ rt5668 { compatible = "realtek,rt5668b"; reg = <0x1a>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index 1b3c13d206ff..da2430099181 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -64,7 +64,7 @@ rt5677 { compatible = "realtek,rt5677"; reg = <0x2c>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; gpio-controller; #gpio-cells = <2>; diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index ade1ece8b45f..707fa98d1310 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -58,7 +58,7 @@ rt5682 { compatible = "realtek,rt5682i"; reg = <0x1a>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; -- 2.17.1

3 2

[PATCH] ASoC: tlv320adcx140: Fix digital gain range
by Camel Guo 17 Sep '20

17 Sep '20

From: Camel Guo <camelg(a)axis.com> According to its datasheet, the digital gain should be -100 dB when CHx_DVOL is 1 and 27 dB when CHx_DVOL is 255. But with the current dig_vol_tlv, "Digital CHx Out Volume" shows 27.5 dB if CHx_DVOL is 255 and -95.5 dB if CHx_DVOL is 1. This commit fixes this bug. Fixes: 689c7655b50c ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family") Signed-off-by: Camel Guo <camelg(a)axis.com> --- sound/soc/codecs/tlv320adcx140.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 6d456aa269ad..13ece039e9fe 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -161,7 +161,7 @@ static const struct regmap_config adcx140_i2c_regmap = { }; /* Digital Volume control. From -100 to 27 dB in 0.5 dB steps */ -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10000, 50, 0); +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10050, 50, 0); /* ADC gain. From 0 to 42 dB in 1 dB steps */ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); -- 2.20.1

2 1

[PATCH v3 1/2] ASoC: dt-bindings: ak4458: Add dsd-path property
by Shengjiu Wang 17 Sep '20

17 Sep '20

Add "dsd-path" property, which is used for ak4497 codec to select the DSD input pin. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/ak4458.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/ak4458.txt b/Documentation/devicetree/bindings/sound/ak4458.txt index 8f6c84f21468..0416c14895d6 100644 --- a/Documentation/devicetree/bindings/sound/ak4458.txt +++ b/Documentation/devicetree/bindings/sound/ak4458.txt @@ -12,6 +12,9 @@ Optional properties: - mute-gpios: A GPIO specifier for the soft mute pin - AVDD-supply: Analog power supply - DVDD-supply: Digital power supply +- dsd-path: Select DSD input pins for ak4497 + 0: select #16, #17, #19 pins + 1: select #3, #4, #5 pins Example: -- 2.27.0

2 2

[PATCH 0/2] ASoC: q6afe: add clocks support
by Srinivas Kandagatla 17 Sep '20

17 Sep '20

q6afe already exposes clocks using apis, but not as proper clock controller driver. This patch puts those clocks in to a proper clock controller so that other drivers that depend on those clocks can be properly expressed. Srinivas Kandagatla (2): ASoC: q6afe: dt-bindings: add q6afe clock bindings ASoC: q6afe-clocks: add q6afe clock controller .../devicetree/bindings/sound/qcom,q6afe.txt | 23 ++ include/dt-bindings/sound/qcom,q6afe.h | 74 ++++- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-clocks.c | 270 ++++++++++++++++++ 5 files changed, 371 insertions(+), 1 deletion(-) create mode 100644 sound/soc/qcom/qdsp6/q6afe-clocks.c -- 2.21.0

2 3

[PATCH 0/8] ASoC: SOF: small fixes for 5.10
by Kai Vehmanen 17 Sep '20

17 Sep '20

Series that adds debug support for IMX platforms, more details to FW version information, adds missing -EACCESS handling to pm_runtime_get_sync() calls and a set of minor cosmetic, trace verbosity and coding style issues. Guennadi Liakhovetski (3): ASoC: SOF: (cosmetic) remove redundant "ret" variable uses ASoC: SOF: remove several superfluous type-casts ASoC: SOF: fix range checks Iulian Olaru (1): ASoC: SOF: imx: Add debug support for imx platforms Karol Trzcinski (1): ASoC: SOF: Add `src_hash` to `sof_ipc_fw_version` structure Pierre-Louis Bossart (3): ASoC: SOF: debug: update test for pm_runtime_get_sync() ASoC: SOF: control: update test for pm_runtime_get_sync() ASoC: SOF: Intel: hda: reduce verbosity of boot error logs include/sound/sof/info.h | 4 +- sound/soc/sof/control.c | 62 +++++++++++++-------------- sound/soc/sof/debug.c | 2 +- sound/soc/sof/imx/Kconfig | 8 ++++ sound/soc/sof/imx/Makefile | 3 ++ sound/soc/sof/imx/imx-common.c | 72 ++++++++++++++++++++++++++++++++ sound/soc/sof/imx/imx-common.h | 16 +++++++ sound/soc/sof/imx/imx8.c | 23 +++++++++- sound/soc/sof/imx/imx8m.c | 17 +++++++- sound/soc/sof/intel/hda-loader.c | 16 +++---- sound/soc/sof/intel/hda.c | 12 ++++-- sound/soc/sof/intel/hda.h | 2 + sound/soc/sof/sof-audio.c | 6 +-- sound/soc/sof/sof-priv.h | 8 ++++ sound/soc/sof/topology.c | 44 ++++++++++--------- 15 files changed, 226 insertions(+), 69 deletions(-) create mode 100644 sound/soc/sof/imx/imx-common.c create mode 100644 sound/soc/sof/imx/imx-common.h -- 2.27.0

2 9

[PATCH] ASoC: SOF: Intel: Add support for tgl-h
by Kai Vehmanen 17 Sep '20

17 Sep '20

From: Rander Wang <rander.wang(a)intel.com> SOF will support tgl-h and tgl-lp in different FW binaries due to hardware difference, so create another dev_desc entry with FW name of sof-tgl-h.ri and dsp_desc named tglh_chip_info for tgl-h. Fixes: c8d2e2bfaeffa ("ASoC: SOF: Intel: add PCI IDs for ICL-H and TGL-H") Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/tgl.c | 16 ++++++++++++++++ sound/soc/sof/sof-pci-dev.c | 18 +++++++++++++++++- 3 files changed, 34 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index f0f8f95c082b..659868da4068 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -737,6 +737,7 @@ extern const struct sof_intel_dsp_desc cnl_chip_info; extern const struct sof_intel_dsp_desc skl_chip_info; extern const struct sof_intel_dsp_desc icl_chip_info; extern const struct sof_intel_dsp_desc tgl_chip_info; +extern const struct sof_intel_dsp_desc tglh_chip_info; extern const struct sof_intel_dsp_desc ehl_chip_info; extern const struct sof_intel_dsp_desc jsl_chip_info; diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index f8d04fd66ceb..0278b67de1ec 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -135,3 +135,19 @@ const struct sof_intel_dsp_desc tgl_chip_info = { .ssp_base_offset = CNL_SSP_BASE_OFFSET, }; EXPORT_SYMBOL_NS(tgl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); + +const struct sof_intel_dsp_desc tglh_chip_info = { + /* Tigerlake-H */ + .cores_num = 2, + .init_core_mask = 1, + .host_managed_cores_mask = BIT(0), + .ipc_req = CNL_DSP_REG_HIPCIDR, + .ipc_req_mask = CNL_DSP_REG_HIPCIDR_BUSY, + .ipc_ack = CNL_DSP_REG_HIPCIDA, + .ipc_ack_mask = CNL_DSP_REG_HIPCIDA_DONE, + .ipc_ctl = CNL_DSP_REG_HIPCCTL, + .rom_init_timeout = 300, + .ssp_count = ICL_SSP_COUNT, + .ssp_base_offset = CNL_SSP_BASE_OFFSET, +}; +EXPORT_SYMBOL_NS(tglh_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 3ec380945466..8c53f6935417 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -223,6 +223,22 @@ static const struct sof_dev_desc tgl_desc = { .nocodec_tplg_filename = "sof-tgl-nocodec.tplg", .ops = &sof_tgl_ops, }; + +static const struct sof_dev_desc tglh_desc = { + .machines = snd_soc_acpi_intel_tgl_machines, + .alt_machines = snd_soc_acpi_intel_tgl_sdw_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .resindex_dma_base = -1, + .chip_info = &tglh_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .default_fw_filename = "sof-tgl-h.ri", + .nocodec_tplg_filename = "sof-tgl-nocodec.tplg", + .ops = &sof_tgl_ops, +}; #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) @@ -457,7 +473,7 @@ static const struct pci_device_id sof_pci_ids[] = { { PCI_DEVICE(0x8086, 0xa0c8), /* TGL-LP */ .driver_data = (unsigned long)&tgl_desc}, { PCI_DEVICE(0x8086, 0x43c8), /* TGL-H */ - .driver_data = (unsigned long)&tgl_desc}, + .driver_data = (unsigned long)&tglh_desc}, #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) -- 2.27.0

2 1

[PATCH 1/2] ASoC: codec: tlv320aic32x4: fix missing aic32x4_disable_regulators() in error path
by Matthias Schiffer 17 Sep '20

17 Sep '20

The regulators need to be disabled in the aic32x4_register_clocks() failure case as well. Fixes: 9d4befff5a95 ("ASoC: codec: tlv3204: Moving GPIO reset and add ADC reset") Signed-off-by: Matthias Schiffer <matthias.schiffer(a)ew.tq-group.com> --- sound/soc/codecs/tlv320aic32x4.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 8dcea566b375..a45fb496082c 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1230,8 +1230,7 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) &soc_component_dev_aic32x4, &aic32x4_dai, 1); if (ret) { dev_err(dev, "Failed to register component\n"); - aic32x4_disable_regulators(aic32x4); - return ret; + goto err_disable_regulators; } if (gpio_is_valid(aic32x4->rstn_gpio)) { @@ -1242,9 +1241,14 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) ret = aic32x4_register_clocks(dev, aic32x4->mclk_name); if (ret) - return ret; + goto err_disable_regulators; return 0; + +err_disable_regulators: + aic32x4_disable_regulators(aic32x4); + + return ret; } EXPORT_SYMBOL(aic32x4_probe); -- 2.17.1

2 2

[PATCH] ASoC: topology: disable size checks for bytes_ext controls if needed
by Kai Vehmanen 17 Sep '20

17 Sep '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> When CONFIG_SND_CTL_VALIDATION is set, accesses to extended bytes control generate spurious error messages when the size exceeds 512 bytes, such as [ 11.224223] sof_sdw sof_sdw: control 2:0:0:EQIIR5.0 eqiir_coef_5:0: invalid count 1024 In addition the error check returns -EINVAL which has the nasty side effect of preventing applications accessing controls from working, e.g. root@plb:~# alsamixer cannot load mixer controls: Invalid argument It's agreed that the control interface has been abused since 2014, but forcing a check should not prevent existing solutions from working. This patch skips the checks conditionally if CONFIG_SND_CTL_VALIDATION is set and the byte array provided by topology is > 512. This preserves the checks for all other cases. Fixes: 1a3232d2f61d2 ('ASoC: topology: Add support for TLV bytes controls') BugLink: https://github.com/thesofproject/linux/issues/2430 Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Jaska Uimonen <jaska.uimonen(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/soc-topology.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 28faaf58326e..63086fa8b861 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -592,6 +592,17 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, k->info = snd_soc_bytes_info_ext; k->tlv.c = snd_soc_bytes_tlv_callback; + /* + * When a topology-based implementation abuses the + * control interface and uses bytes_ext controls of + * more than 512 bytes, we need to disable the size + * checks, otherwise accesses to such controls will + * return an -EINVAL error and prevent the card from + * being configured. + */ + if (IS_ENABLED(CONFIG_SND_CTL_VALIDATION) && sbe->max > 512) + k->access |= SNDRV_CTL_ELEM_ACCESS_SKIP_CHECK; + ext_ops = tplg->bytes_ext_ops; num_ops = tplg->bytes_ext_ops_count; for (i = 0; i < num_ops; i++) { -- 2.27.0

2 1

[PATCH v5 0/5] Qualcomm's lpass-hdmi ASoC driver to support audio over dp port
by Srinivasa Rao Mandadapu 17 Sep '20

17 Sep '20

These patches are to support audio over DP port on Qualcomm's SC7180 LPASS Asoc. It includes machine driver, cpu driver, platform driver updates for HDMI path support, device tree documention, lpass variant structure optimization and configuration changes. These patches depends on the DP patch series https://patchwork.kernel.org/project/dri-devel/list/?series=332029 Changes Since v4: -- Updated with single compatible node for both I2S and HDMI. Changes Since v3: -- Removed id in lpass variant structure and used snd_soc_dai_driver id. Changes Since v2: -- Audio buffer size(i.e. LPASS_PLATFORM_BUFFER_SIZE) in lpass-platform.c increased. Changes Since v1: -- Commit messages are updated -- Addressed Rob Herring review comments V Sujith Kumar Reddy (5): ASoC: Add sc7180-lpass binding header hdmi define ASoC: dt-bindings: Add dt binding for lpass hdmi Asoc:qcom:lpass-cpu:Update dts property read API ASoC: qcom: Add support for lpass hdmi driver ASoC: qcom: sc7180: Add support for audio over DP .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 74 ++- include/dt-bindings/sound/sc7180-lpass.h | 1 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-cpu.c | 39 +- sound/soc/qcom/lpass-hdmi.c | 596 +++++++++++++++++++++ sound/soc/qcom/lpass-hdmi.h | 129 +++++ sound/soc/qcom/lpass-lpaif-reg.h | 52 +- sound/soc/qcom/lpass-platform.c | 468 ++++++++++++---- sound/soc/qcom/lpass-sc7180.c | 116 +++- sound/soc/qcom/lpass.h | 119 +++- 11 files changed, 1447 insertions(+), 154 deletions(-) create mode 100644 sound/soc/qcom/lpass-hdmi.c create mode 100644 sound/soc/qcom/lpass-hdmi.h -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

1 5

[PATCH] ASoC: rt715: Add module parameter to fix dmic pop sound issue.
by Pierre-Louis Bossart 17 Sep '20

17 Sep '20

From: Jack Yu <jack.yu(a)realtek.com> Add module parameter "power_up_delay" to fix pop noise on capture. The power_up_delay value is set with a default value of 400ms, smaller values are not recommended. BugLink: https://github.com/thesofproject/linux/issues/1969 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/rt715.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 099c8bd20006..0cf10dec1e3b 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -37,6 +37,10 @@ #include "rt715.h" +static int power_up_delay = 400; +module_param(power_up_delay, int, 0444); +MODULE_PARM_DESC(power_up_delay, "RT715 power up delay time in ms"); + static int rt715_index_write(struct regmap *regmap, unsigned int reg, unsigned int value) { @@ -498,6 +502,7 @@ static int rt715_set_bias_level(struct snd_soc_component *component, regmap_write(rt715->regmap, RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + msleep(power_up_delay); } break; -- 2.25.1

2 4

[PATCH 0/2] soundwire: sysfs: expose device number and status
by Pierre-Louis Bossart 17 Sep '20

17 Sep '20

This patchset combines three contributions: Srinivas Kandagalta suggested creating a device when it's detected on the bus but not described in platform firmware. I suggested adding the device number and status to show the difference with 'ghost' devices, described in firmware but not physically present. Vinod suggested a simpler way to report the status. I did not keep Vinod's patch separate since it was using the same group attribute as the other properties, which prevents status and device number from being reported if there is no firmware and no driver. These patches provide a nice addition for integrators/tests and were tested on Qualcomm and Intel platforms. Thanks Srinivas for the initial idea! Pierre-Louis Bossart (1): soundwire: sysfs: add slave status and device number before probe Srinivas Kandagatla (1): soundwire: bus: add enumerated Slave device to device list .../ABI/testing/sysfs-bus-soundwire-slave | 18 ++++++ drivers/soundwire/bus.c | 9 +++ drivers/soundwire/bus.h | 2 + drivers/soundwire/bus_type.c | 9 +++ drivers/soundwire/slave.c | 6 +- drivers/soundwire/sysfs_local.h | 4 ++ drivers/soundwire/sysfs_slave.c | 64 ++++++++++++++++++- 7 files changed, 109 insertions(+), 3 deletions(-) -- 2.25.1

2 5

[PATCH v2 0/2] soundwire: use {u32|u16}p_replace_bits
by Vinod Koul 17 Sep '20

17 Sep '20

This is v2 of the series to fix usage of FIELD_PREP and use {u32|u16}p_replace_bits. v1 used u32_replace_bits but looks like it might be removed so use the {u32|u16}p_replace_bits() instead. Also this helped to find that one instance in intel driver should have used u16p_replace_bits() so that is fixed too Vinod Koul (2): soundwire: cadence: use u32p_replace_bits soundwire: intel: use {u32|u16}p_replace_bits drivers/soundwire/cadence_master.c | 21 ++++++++++----------- drivers/soundwire/intel.c | 14 +++++++------- 2 files changed, 17 insertions(+), 18 deletions(-) -- 2.26.2

1 2

Re: [oss-drivers] [trivial PATCH] treewide: Convert switch/case fallthrough; to break;
by Simon Horman 17 Sep '20

17 Sep '20

On Wed, Sep 09, 2020 at 01:06:39PM -0700, Joe Perches wrote: > fallthrough to a separate case/default label break; isn't very readable. > > Convert pseudo-keyword fallthrough; statements to a simple break; when > the next label is case or default and the only statement in the next > label block is break; > > Found using: > > $ grep-2.5.4 -rP --include=*.[ch] -n "fallthrough;(\s*(case\s+\w+|default)\s*:\s*){1,7}break;" * > > Miscellanea: > > o Move or coalesce a couple label blocks above a default: block. > > Signed-off-by: Joe Perches <joe(a)perches.com> ... > diff --git a/drivers/net/ethernet/netronome/nfp/nfpcore/nfp6000_pcie.c b/drivers/net/ethernet/netronome/nfp/nfpcore/nfp6000_pcie.c > index 252fe06f58aa..1d5b87079104 100644 > --- a/drivers/net/ethernet/netronome/nfp/nfpcore/nfp6000_pcie.c > +++ b/drivers/net/ethernet/netronome/nfp/nfpcore/nfp6000_pcie.c > @@ -345,7 +345,7 @@ static int matching_bar(struct nfp_bar *bar, u32 tgt, u32 act, u32 tok, > baract = NFP_CPP_ACTION_RW; > if (act == 0) > act = NFP_CPP_ACTION_RW; > - fallthrough; > + break; > case NFP_PCIE_BAR_PCIE2CPP_MapType_FIXED: > break; > default: This is a cascading fall-through handling all map types. I don't think this change improves readability. ...

1 0

"aplay -D oss -l" still reads sound cards through ALSA, not OSS
by GitHub issues - edited 17 Sep '20

17 Sep '20

alsa-project/alsa-utils issue #56 was edited from yurivict: I don't have any ALSA devices, only OSS devices. When the OSS PCM is selected ```aplay``` still tries to read ALSA devices: ``` $ /usr/bin/aplay -D oss -l aplay: device_list:272: no soundcards found... ``` ``` $ /usr/bin/aplay -L null Discard all samples (playback) or generate zero samples (capture) oss Open Sound System ``` But ```/usr/bin/aplay x.wav``` plays it through the OSS card. aplay should list OSS cards when -D is selected. The ALSA interface should also list OSS cards when no ALSA cards are present but it doesn't. alsa-lib-1.1.8 alsa-plugins-oss-1.1.6 alsa-utils-1.1.8 Issue URL : https://github.com/alsa-project/alsa-utils/issues/56 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

"aplay -D oss -l" still reads sound cards through ALSA, not OSS
by GitHub issues - opened 17 Sep '20

17 Sep '20

alsa-project/alsa-utils issue #56 was opened from yurivict: I don't have any ALSA devices, only OSS devices. When the OSS PCM is selected ```aplay``` still tries to read ALSA devices: ``` $ /usr/bin/aplay -D oss -l aplay: device_list:272: no soundcards found... ``` ``` $ /usr/bin/aplay -L null Discard all samples (playback) or generate zero samples (capture) oss Open Sound System ``` alsa-lib-1.1.8 alsa-plugins-oss-1.1.6 alsa-utils-1.1.8 Issue URL : https://github.com/alsa-project/alsa-utils/issues/56 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/3] ASoC: fsl_sai: update the register list
by Shengjiu Wang 17 Sep '20

17 Sep '20

As sai ip is upgraded, so update sai register list. Shengjiu Wang (3): ASoC: fsl_sai: Add new added registers and new bit definition ASoC: fsl_sai: Add fsl_sai_check_version function ASoC: fsl_sai: Set MCLK input or output direction sound/soc/fsl/fsl_sai.c | 77 ++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_sai.h | 87 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 164 insertions(+) -- 2.27.0

3 9

Re: [PATCH] ASoC: fsl_audmix: make clock and output src write only
by Nicolin Chen 17 Sep '20

17 Sep '20

On Mon, Sep 14, 2020 at 08:24:34PM +0300, Viorel Suman (OSS) wrote: > From: Viorel Suman <viorel.suman(a)nxp.com> > > "alsactl -f state.conf store/restore" sequence fails because setting > "mixing clock source" and "output source" requires active TDM clock > being started for configuration propagation. Make these two controls > write only so that their values are not stored at "alsactl store". > > Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> > --- > sound/soc/fsl/fsl_audmix.c | 16 ++++++++++++---- > 1 file changed, 12 insertions(+), 4 deletions(-) > > diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c > index a447baf..7ad5925 100644 > --- a/sound/soc/fsl/fsl_audmix.c > +++ b/sound/soc/fsl/fsl_audmix.c > @@ -199,10 +199,18 @@ static int fsl_audmix_put_out_src(struct snd_kcontrol *kcontrol, > > static const struct snd_kcontrol_new fsl_audmix_snd_controls[] = { > /* FSL_AUDMIX_CTR controls */ > - SOC_ENUM_EXT("Mixing Clock Source", fsl_audmix_enum[0], > - snd_soc_get_enum_double, fsl_audmix_put_mix_clk_src), > - SOC_ENUM_EXT("Output Source", fsl_audmix_enum[1], > - snd_soc_get_enum_double, fsl_audmix_put_out_src), > + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .name = "Mixing Clock Source", > + .info = snd_soc_info_enum_double, > + .access = SNDRV_CTL_ELEM_ACCESS_WRITE, > + .put = fsl_audmix_put_mix_clk_src, > + .private_value = (unsigned long)&fsl_audmix_enum[0] }, > + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .name = "Output Source", > + .info = snd_soc_info_enum_double, > + .access = SNDRV_CTL_ELEM_ACCESS_WRITE, > + .put = fsl_audmix_put_out_src, > + .private_value = (unsigned long)&fsl_audmix_enum[1] }, > SOC_ENUM("Output Width", fsl_audmix_enum[2]), > SOC_ENUM("Frame Rate Diff Error", fsl_audmix_enum[3]), > SOC_ENUM("Clock Freq Diff Error", fsl_audmix_enum[4]), > -- > 2.7.4 >

1 0

[PATCH v3 0/3] soundwire: qcom: fix IP version v1.5.1 support
by Srinivas Kandagatla 16 Sep '20

16 Sep '20

While testing Qualcomm soundwire controller version 1.5.1, found two issue, Firstly the frame shape information configured vs the bus parameters are out of sync. secondly some ports on this ip version require block packing mode support. With this patchset I was able to test 2 WSA speakers! Also I found a regression due to move to REG_FIELD, which patch 1 fixes it! thanks, srini Changes since v2: - updated to use u32_replace_bits(). - wrapped up in single lines as suggested by vkoul. Srinivas Kandagatla (3): soundwire: qcom: clear BIT FIELDs before value set. soundwire: qcom: add support to block packing mode soundwire: qcom: get max rows and cols info from compatible drivers/soundwire/qcom.c | 70 +++++++++++++++++++++++++++++----------- 1 file changed, 51 insertions(+), 19 deletions(-) -- 2.21.0

1 3

[PATCH v2 0/3] soundwire: qcom: fix IP version v1.5.1 support
by Srinivas Kandagatla 16 Sep '20

16 Sep '20

While testing Qualcomm soundwire controller version 1.5.1, found two issue, Firstly the frame shape information configured vs the bus parameters are out of sync. secondly some ports on this ip version require block packing mode support. With this patchset I was able to test 2 WSA speakers! Also I found a regression due to move to REG_FIELD, which patch 1 fixes it! thanks, srini Changes since v1: - rebased on top of REG_FILED patch or soundwire-next branch - udated qcom_swrm_data to use u32 instead of int as suggested by VKoul Srinivas Kandagatla (3): soundwire: qcom: clear BIT FIELDs before value set. soundwire: qcom: add support to block packing mode soundwire: qcom: get max rows and cols info from compatible drivers/soundwire/qcom.c | 76 ++++++++++++++++++++++++++++++---------- 1 file changed, 58 insertions(+), 18 deletions(-) -- 2.21.0

3 10

[PATCH 1/2] soundwire: cadence: use u32_replace_bits
by Vinod Koul 16 Sep '20

16 Sep '20

FIELD_PREP() does not replace the bits so it is not apt in case where we modify a register. Use u32_replace_bits() instead. Fixes: 3cf25d63b1b9 ("soundwire: cadence: use FIELD_{GET|PREP}") Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- drivers/soundwire/cadence_master.c | 23 ++++++++++++----------- 1 file changed, 12 insertions(+), 11 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index f1efe5beead6..bb293e7010f3 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -1278,9 +1278,9 @@ static int cdns_port_params(struct sdw_bus *bus, dpn_config = cdns_readl(cdns, dpn_config_off); - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_WL, (p_params->bps - 1)); - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_PORT_FLOW, p_params->flow_mode); - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_PORT_DAT, p_params->data_mode); + dpn_config = u32_replace_bits(dpn_config, (p_params->bps - 1), CDNS_DPN_CONFIG_WL); + dpn_config = u32_replace_bits(dpn_config, p_params->flow_mode, CDNS_DPN_CONFIG_PORT_FLOW); + dpn_config = u32_replace_bits(dpn_config, p_params->data_mode, CDNS_DPN_CONFIG_PORT_DAT); cdns_writel(cdns, dpn_config_off, dpn_config); @@ -1316,18 +1316,19 @@ static int cdns_transport_params(struct sdw_bus *bus, } dpn_config = cdns_readl(cdns, dpn_config_off); - - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_BGC, t_params->blk_grp_ctrl); - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_BPM, t_params->blk_pkg_mode); + dpn_config = u32_replace_bits(dpn_config, t_params->blk_grp_ctrl, CDNS_DPN_CONFIG_BGC); + dpn_config = u32_replace_bits(dpn_config, t_params->blk_pkg_mode, CDNS_DPN_CONFIG_BPM); cdns_writel(cdns, dpn_config_off, dpn_config); - dpn_offsetctrl |= FIELD_PREP(CDNS_DPN_OFFSET_CTRL_1, t_params->offset1); - dpn_offsetctrl |= FIELD_PREP(CDNS_DPN_OFFSET_CTRL_2, t_params->offset2); + dpn_offsetctrl = u32_replace_bits(dpn_offsetctrl, + t_params->offset1, CDNS_DPN_OFFSET_CTRL_1); + dpn_offsetctrl = u32_replace_bits(dpn_offsetctrl, + t_params->offset2, CDNS_DPN_OFFSET_CTRL_2); cdns_writel(cdns, dpn_offsetctrl_off, dpn_offsetctrl); - dpn_hctrl |= FIELD_PREP(CDNS_DPN_HCTRL_HSTART, t_params->hstart); - dpn_hctrl |= FIELD_PREP(CDNS_DPN_HCTRL_HSTOP, t_params->hstop); - dpn_hctrl |= FIELD_PREP(CDNS_DPN_HCTRL_LCTRL, t_params->lane_ctrl); + dpn_hctrl = u32_replace_bits(dpn_hctrl, t_params->hstart, CDNS_DPN_HCTRL_HSTART); + dpn_hctrl = u32_replace_bits(dpn_hctrl, t_params->hstop, CDNS_DPN_HCTRL_HSTOP); + dpn_hctrl = u32_replace_bits(dpn_hctrl, t_params->lane_ctrl, CDNS_DPN_HCTRL_LCTRL); cdns_writel(cdns, dpn_hctrl_off, dpn_hctrl); cdns_writel(cdns, dpn_samplectrl_off, (t_params->sample_interval - 1)); -- 2.26.2

1 1

aplay: playing mono audio file alteres pulseaudio default Master left or right volume level
by GitHub issues - edited 16 Sep '20

16 Sep '20

alsa-project/alsa-utils issue #55 was edited from maxmbed: ### Description I am encountering an unexpected behaviour with pulse audio master volume in my application. If I set for example left channel to 100% and right channel 0%, and play mono audio file (i.e. audio file with 1 channel only), then I would expect to hear only the audio file on left channel. The actual behaviour in the example of above is when playing the audio file, the Master volume right channel will be automatically adjusted to 50%. If play again it increases to 75% and so on. This issue happens also with other R/L volume level, the case of above is just an example. **Important to note the case does not happen with stereo audio file** where audio sounds on left only channel without altering the volume level of right channel as we can expect. ### Steps to repeat issue: (a) Set mixer Lch to 100%, Rch to 0% ``` $ amixer -D default sset Master 65535,0 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 65535 [100%] [on] Front Right: Playback 0 [0%] [on] ``` (b) Play mono audio file ``` $ aplay -D default ringtone.wav Playing WAVE 'ringtone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono ``` ### Request (1) Is it by design expected ? (2) Is there a way to prevent this phenomenon (command option) ? Issue URL : https://github.com/alsa-project/alsa-utils/issues/55 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplay: playing mono audio file alteres pulseaudio default Master left or right volume level
by GitHub issues - edited 16 Sep '20

16 Sep '20

alsa-project/alsa-utils issue #55 was edited from maxmbed: ### Description I am encountering an unexpected behaviour with pulse audio master volume in my application. If I set for example left channel to 100% and right channel 0%, and play mono audio file (i.e. audio file with 1 channel only), then I would expect to hear only the audio file on left channel. The actual behaviour in the example of above is when playing the audio file, the Master volume right channel will be automatically adjusted to 50%. If play again it increases to 75% and so on. This issue happens also with other R/L volume level, the case of above is just an example. **Important to note the case does not happen with stereo audio file** where audio sounds on left only channel without altering the volume level as we can expect. ### Steps to repeat issue: (a) Set mixer Lch to 100%, Rch to 0% ``` $ amixer -D default sset Master 65535,0 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 65535 [100%] [on] Front Right: Playback 0 [0%] [on] ``` (b) Play mono audio file ``` $ aplay -D default ringtone.wav Playing WAVE 'ringtone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono ``` ### Request (1) Is it by design expected ? (2) Is there a way to prevent this phenomenon (command option) ? Issue URL : https://github.com/alsa-project/alsa-utils/issues/55 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2 0/3] regmap: add SoundWire 1.2 MBQ support
by Pierre-Louis Bossart 16 Sep '20

16 Sep '20

In preparation of the upstream contribution of SDCA (SoundWire Device Class for Audio) ASoC codec drivers [1] [2], add regmap support SoundWire 1.2 MBQ support. The MBQ (Multi-Byte Quantity) registers need to be handled in a different way from regular 8-bit SoundWire registers, their main application is going to be for volume/gain controls. The second patch was initially suggested for inclusion in the SoundWire tree, and was modified to add more background information on SDCA in the commit message as requested by Vinod Koul. Changes since v1: Rebased on regmap tree (conflict with SPI stuff). Removed mod_devicetable.h header Removed -EOPNOTSUPP error codes, use -ENOTSUPP Added long description of SDCA Used FIELD_PREP/GET as suggested by Vinod Koul Added Bard Liao's Acked-by tag. Pierre-Louis Bossart (3): regmap: sdw: add required header files soundwire: SDCA: add helper macro to access controls regmap: sdw: add support for SoundWire 1.2 MBQ drivers/base/regmap/Kconfig | 6 +- drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw-mbq.c | 101 ++++++++++++++++++++++++ drivers/base/regmap/regmap-sdw.c | 2 + include/linux/regmap.h | 21 +++++ include/linux/soundwire/sdw_registers.h | 33 ++++++++ 6 files changed, 163 insertions(+), 1 deletion(-) create mode 100644 drivers/base/regmap/regmap-sdw-mbq.c base-commit: d17343b87da422a59d99a3ed130573dbeb96c582 -- 2.25.1

2 17

aplay: playing mono audio file alteres pulseaudio default Master left or right volume level
by GitHub issues - edited 16 Sep '20

16 Sep '20

alsa-project/alsa-utils issue #55 was edited from maxmbed: ### Description I am encountering an unexpected behaviour with pulse audio master volume in my application. If I set for example left channel to 100% and right channel 0%, and play mono audio file (i.e. audio file with 1 channel only), then I would expect to hear only the audio file on left channel. The actual behaviour in the example of above is when playing the audio file, the Master volume right channel will be automatically adjusted to 50%. If play again it increases to 75% and so on. This issue happens also with other R/L volume level, the case of above is just an example. **Important to note the case does not happen with stereo audio file** where audio sounds on left only channel as we can expect. ### Steps to repeat issue: (a) Set mixer Lch to 100%, Rch to 0% ``` $ amixer -D default sset Master 65535,0 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 65535 [100%] [on] Front Right: Playback 0 [0%] [on] ``` (b) Play mono audio file ``` $ aplay -D default ringtone.wav Playing WAVE 'ringtone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono ``` ### Request (1) Is it by design expected ? (2) Is there a way to prevent this phenomenon (command option) ? Issue URL : https://github.com/alsa-project/alsa-utils/issues/55 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplay: playing mono audio file alteres pulseaudio default Master left or right volume level
by GitHub issues - edited 16 Sep '20

16 Sep '20

alsa-project/alsa-utils issue #55 was edited from maxmbed: ### Description I am encountering an unexpected behaviour with pulse audio master volume in my application. If I set for example left channel to 100% and right channel 0%, and play mono audio file (i.e. audio file with 1 channel only), then I would expect to hear only the audio file on left channel. The actual behaviour in the example of above is when playing the audio file, the Master volume right channel will be automatically adjusted to 50%. If play again it increases to 75% and so on. This issue happens also with other R/L volume level, the case of above is just an example. **Important to note the case does not happen with stereo audio file** where audio sounds on left only channel as we can expected. ### Steps to repeat issue: (a) Set mixer Lch to 100%, Rch to 0% ``` $ amixer -D default sset Master 65535,0 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 65535 [100%] [on] Front Right: Playback 0 [0%] [on] ``` (b) Play mono audio file ``` $ aplay -D default ringtone.wav Playing WAVE 'ringtone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono ``` ### Request (1) Is it by design expected ? (2) Is there a way to prevent this phenomenon (command option) ? Issue URL : https://github.com/alsa-project/alsa-utils/issues/55 Repository URL: https://github.com/alsa-project/alsa-utils

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aplay: playing mono audio file alteres pulseaudio default Master left or right volume level
by GitHub issues - opened 16 Sep '20

16 Sep '20

alsa-project/alsa-utils issue #55 was opened from maxmbed: ### Description I am encountering an expected behaviour with pulse audio master volume in my application. If I set for example left channel to 100% and right channel 0%, and play mono audio file (i.e. audio file with 1 channel only), then I would expect to hear only the audio file on left channel. The actual behaviour in the example of above is when playing the audio file, the Master volume right channel will be automatically adjusted to 50%. If play again it increases to 75% and so on. This issue happens also with other R/L volume level, the case of above is just an example. **Important to note the case does not happen with stereo audio file** where audio sounds on left only channel as we can expected. ### Steps to repeat issue: (a) Set mixer Lch to 100%, Rch to 0% ``` $ amixer -D default sset Master 65535,0 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 65535 [100%] [on] Front Right: Playback 0 [0%] [on] ``` (b) Play mono audio file ``` $ aplay -D default ringtone.wav Playing WAVE 'ringtone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono ``` ### Request (1) Is it by design expected ? (2) Is there a way to prevent this phenomenon (command option) ? Issue URL : https://github.com/alsa-project/alsa-utils/issues/55 Repository URL: https://github.com/alsa-project/alsa-utils

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Re: [PATCH] fsl: imx-audmix: Use devm_kcalloc() in imx_audmix_probe()
by Markus Elfring 16 Sep '20

16 Sep '20

> A multiplication for the size determination of a memory allocation > indicated that an array data structure should be processed. > Thus use the corresponding function "devm_kcalloc". I find the previous patch subject inappropriate. Would you like to choose an alternative? Regards, Markus

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Re: [PATCH v2 3/3] ASoC: tlv320adcx140: Add proper support for master mode
by Dan Murphy 16 Sep '20

16 Sep '20

Camel On 9/16/20 2:52 AM, Camel Guo wrote: > Please forget about this patch since Dan will upload a similar one. > > @Dan, see my comment below. > I have cc'd you on my patchset and as you can see I did not add any of the master mode programming only setting of the master mode bits. So this patch is still viable it's just the master mode bit programming that needs to be looked at. Dan

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[PATCH v3 00/11] USB: new USB control message helper functions
by Greg Kroah-Hartman 16 Sep '20

16 Sep '20

In a recent discussion about a USB networking bug found by syzbot, and fixed by Himadri Pandya, the design of the existing usb_control_msg() call was brought up as not being the "nicest" thing to use by Dmitry Vyukov: https://lore.kernel.org/r/CACT4Y+YbDODLRFn8M5QcY4CazhpeCaunJnP_udXtAs0rYoAS… The function makes it hard to get right, in that it will return the number of bytes sent/received, but almost no one checks to see if a short read/write happens. With a malicious, or broken, USB device, this can cause drivers to act on data that they did not anticipate, and sometimes copy internal kernel data out to userspace. So let's fix this up by creating two new functions, usb_control_msg_send() and usb_control_msg_recv(). These functions either complete the full transation, or they return an error, a short send/recv is now an error. They also accept data off of the stack, saving individual drivers the pain of having to constantly allocate memory on their own for tiny messages, thereby saving overall kernel code space. The api also does not require a raw USB "pipe" to be sent to the function, as we know the direction, so just pass in the endpoint number instead, again making it easier on the USB driver author to use. This series first takes a helper function out of the sound core for verifying USB endpoints to be able to use internally, and then adds the new functions, converts over some internal USB code to use them, and then starts to clean up some drivers using these new functions, as an example of the savings that can happen by using these functions. Thanks to Dmitry and Himadri for the idea on how to do all of this. greg k-h ----- Changes from v2: - add Andy's reviewed-by: to patch 3 - remove unneeded change in usb_enable_link_state() in hub.c in patch 4 Changes from v1: - added acks from Takashi Iwai - dropped changes to one function in patch 04 thanks to review from Alan Stern - typo fix in comment in patch 01 - added new patch 11 to remove some unneeded checks in the sound drivers for endpoint statuses that would always be true. Greg Kroah-Hartman (11): USB: move snd_usb_pipe_sanity_check into the USB core USB: add usb_control_msg_send() and usb_control_msg_recv() USB: core: message.c: use usb_control_msg_send() in a few places USB: core: hub.c: use usb_control_msg_send() in a few places USB: legousbtower: use usb_control_msg_recv() sound: usx2y: move to use usb_control_msg_send() sound: 6fire: move to use usb_control_msg_send() and usb_control_msg_recv() sound: line6: move to use usb_control_msg_send() and usb_control_msg_recv() sound: hiface: move to use usb_control_msg_send() Bluetooth: ath3k: use usb_control_msg_send() and usb_control_msg_recv() ALSA: remove calls to usb_pipe_type_check for control endpoints drivers/bluetooth/ath3k.c | 90 +++++------------ drivers/usb/core/hub.c | 99 ++++++++---------- drivers/usb/core/message.c | 171 ++++++++++++++++++++++++++++---- drivers/usb/core/urb.c | 31 ++++-- drivers/usb/misc/legousbtower.c | 60 ++++------- include/linux/usb.h | 7 ++ sound/usb/6fire/firmware.c | 38 +++---- sound/usb/helper.c | 16 +-- sound/usb/helper.h | 1 - sound/usb/hiface/pcm.c | 14 ++- sound/usb/line6/driver.c | 69 +++++-------- sound/usb/line6/podhd.c | 17 ++-- sound/usb/line6/toneport.c | 8 +- sound/usb/mixer_scarlett_gen2.c | 2 +- sound/usb/quirks.c | 12 +-- sound/usb/usx2y/us122l.c | 42 ++------ 16 files changed, 336 insertions(+), 341 deletions(-) -- 2.28.0

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[PATCH 01/10] dt-bindings: arm: samsung: pmu: Use unevaluatedProperties
by Krzysztof Kozlowski 16 Sep '20

16 Sep '20

Additional properties actually might appear (e.g. assigned-clocks) so use unevaluatedProperties to fix dtbs_check warnings like: arch/arm64/boot/dts/exynos/exynos5433-tm2.dt.yaml: system-controller@105c0000: 'assigned-clock-parents', 'assigned-clocks' do not match any of the regexes: 'pinctrl-[0-9]+' Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Documentation/devicetree/bindings/arm/samsung/pmu.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/arm/samsung/pmu.yaml b/Documentation/devicetree/bindings/arm/samsung/pmu.yaml index 686c13c14e32..30ff2da81416 100644 --- a/Documentation/devicetree/bindings/arm/samsung/pmu.yaml +++ b/Documentation/devicetree/bindings/arm/samsung/pmu.yaml @@ -86,7 +86,7 @@ required: - compatible - reg -additionalProperties: false +unevaluatedProperties: false allOf: - if: -- 2.17.1

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