From 2a0435df963f996ca870a2ef1cbf1773dc0ea25a Mon Sep 17 00:00:00 2001 From: Stephan Gerhold Date: Thu, 7 Jan 2021 17:51:31 +0100 Subject: [PATCH 01/19] ASoC: hdmi-codec: Fix return value in hdmi_codec_set_jack() Sound is broken on the DragonBoard 410c (apq8016_sbc) since 5.10: hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: error at snd_soc_component_set_jack on hdmi-audio-codec.1.auto: -95 qcom-apq8016-sbc 7702000.sound: Failed to set jack: -95 ADV7533: ASoC: error at snd_soc_link_init on ADV7533: -95 hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: error at snd_soc_component_set_jack on hdmi-audio-codec.1.auto: -95 qcom-apq8016-sbc: probe of 7702000.sound failed with error -95 This happens because apq8016_sbc calls snd_soc_component_set_jack() on all codec DAIs and attempts to ignore failures with return code -ENOTSUPP. -ENOTSUPP is also excluded from error logging in soc_component_ret(). However, hdmi_codec_set_jack() returns -E*OP*NOTSUPP if jack detection is not supported, which is not handled in apq8016_sbc and soc_component_ret(). Make it return -ENOTSUPP instead to fix sound and silence the errors. Cc: Cheng-Yi Chiang Cc: Srinivas Kandagatla Fixes: 55c5cc63ab32 ("ASoC: hdmi-codec: Use set_jack ops to set jack") Signed-off-by: Stephan Gerhold Acked-by: Nicolin Chen Link: https://lore.kernel.org/r/20210107165131.2535-1-stephan@gerhold.net Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/fsl/imx-hdmi.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index d5fcc4db8284..0f3ac22f2cf8 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -717,7 +717,7 @@ static int hdmi_codec_set_jack(struct snd_soc_component *component, void *data) { struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); - int ret = -EOPNOTSUPP; + int ret = -ENOTSUPP; if (hcp->hcd.ops->hook_plugged_cb) { hcp->jack = jack; diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index ede4a9ad1054..dbbb7618351c 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -90,7 +90,7 @@ static int imx_hdmi_init(struct snd_soc_pcm_runtime *rtd) } ret = snd_soc_component_set_jack(component, &data->hdmi_jack, NULL); - if (ret && ret != -EOPNOTSUPP) { + if (ret && ret != -ENOTSUPP) { dev_err(card->dev, "Can't set HDMI Jack %d\n", ret); return ret; } From 0d38fd8d252446d39050578ea32ed89b9adeb202 Mon Sep 17 00:00:00 2001 From: Mauro Carvalho Chehab Date: Wed, 13 Jan 2021 12:15:43 +0100 Subject: [PATCH 02/19] MAINTAINERS: update references to stm32 audio bindings Changeset 81437cc3b0d9 ("Merge series "dt-bindings: stm32: convert audio dfsdm to json-schema" from Olivier Moysan :") removed bindings/sound/st,stm32-adfsdm.txt, as stm32-* audio bindings are now under: bindings/iio/adc/st,stm32-*.yaml. Update cross-references to them accordingly. Signed-off-by: Mauro Carvalho Chehab Link: https://lore.kernel.org/r/03950bbd5cf7bac10eaaff3725e283d3ec2538c5.1610536535.git.mchehab+huawei@kernel.org Signed-off-by: Mark Brown --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index a2359903fcc8..7d3270b3fd84 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -16964,7 +16964,7 @@ M: Olivier Moysan M: Arnaud Pouliquen L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Maintained -F: Documentation/devicetree/bindings/sound/st,stm32-*.txt +F: Documentation/devicetree/bindings/iio/adc/st,stm32-*.yaml F: sound/soc/stm/ STM32 TIMER/LPTIMER DRIVERS From bcd7059abc19e6ec5b2260dff6a008fb99c4eef9 Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Wed, 13 Jan 2021 02:11:23 +0800 Subject: [PATCH 03/19] ASoC: SOF: Intel: hda: Resume codec to do jack detection Instead of queueing jackpoll_work, runtime resume the codec to let it use different jack detection methods based on jackpoll_interval. This partially matches SOF driver's behavior with commit a6e7d0a4bdb0 ("ALSA: hda: fix jack detection with Realtek codecs when in D3"), the difference is SOF unconditionally resumes the codec. Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20210112181128.1229827-1-kai.heng.feng@canonical.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-codec.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 6875fa570c2c..df59c79cfdfc 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -93,8 +93,7 @@ void hda_codec_jack_check(struct snd_sof_dev *sdev) * has been recorded in STATESTS */ if (codec->jacktbl.used) - schedule_delayed_work(&codec->jackpoll_work, - codec->jackpoll_interval); + pm_request_resume(&codec->core.dev); } #else void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) {} From 31ba0c0776027896553bd8477baff7c8b5d95699 Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Wed, 13 Jan 2021 02:11:24 +0800 Subject: [PATCH 04/19] ASoC: SOF: Intel: hda: Modify existing helper to disable WAKEEN Modify hda_codec_jack_wake_enable() to also support disable WAKEEN. In addition, this patch also moves the WAKEEN disablement call out of hda_codec_jack_check() into hda_codec_jack_wake_enable(). This is a preparation for next patch. No functional change intended. Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20210112181128.1229827-2-kai.heng.feng@canonical.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-codec.c | 16 +++++++--------- sound/soc/sof/intel/hda-dsp.c | 6 ++++-- sound/soc/sof/intel/hda.h | 2 +- 3 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index df59c79cfdfc..b7e9931ead57 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -63,16 +63,18 @@ static int hda_codec_load_module(struct hda_codec *codec) } /* enable controller wake up event for all codecs with jack connectors */ -void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) +void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev, bool enable) { struct hda_bus *hbus = sof_to_hbus(sdev); struct hdac_bus *bus = sof_to_bus(sdev); struct hda_codec *codec; unsigned int mask = 0; - list_for_each_codec(codec, hbus) - if (codec->jacktbl.used) - mask |= BIT(codec->core.addr); + if (enable) { + list_for_each_codec(codec, hbus) + if (codec->jacktbl.used) + mask |= BIT(codec->core.addr); + } snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, mask); } @@ -81,12 +83,8 @@ void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) void hda_codec_jack_check(struct snd_sof_dev *sdev) { struct hda_bus *hbus = sof_to_hbus(sdev); - struct hdac_bus *bus = sof_to_bus(sdev); struct hda_codec *codec; - /* disable controller Wake Up event*/ - snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, 0); - list_for_each_codec(codec, hbus) /* * Wake up all jack-detecting codecs regardless whether an event @@ -96,7 +94,7 @@ void hda_codec_jack_check(struct snd_sof_dev *sdev) pm_request_resume(&codec->core.dev); } #else -void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) {} +void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev, bool enable) {} void hda_codec_jack_check(struct snd_sof_dev *sdev) {} #endif /* CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC */ EXPORT_SYMBOL_NS(hda_codec_jack_wake_enable, SND_SOC_SOF_HDA_AUDIO_CODEC); diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 2b001151fe37..7d00107cf3b2 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -617,7 +617,7 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) if (runtime_suspend) - hda_codec_jack_wake_enable(sdev); + hda_codec_jack_wake_enable(sdev, true); /* power down all hda link */ snd_hdac_ext_bus_link_power_down_all(bus); @@ -683,8 +683,10 @@ static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume) #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) /* check jack status */ - if (runtime_resume) + if (runtime_resume) { + hda_codec_jack_wake_enable(sdev, false); hda_codec_jack_check(sdev); + } /* turn off the links that were off before suspend */ list_for_each_entry(hlink, &bus->hlink_list, list) { diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 9ec8ae0fd649..a3b6f3e9121c 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -650,7 +650,7 @@ void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev); */ void hda_codec_probe_bus(struct snd_sof_dev *sdev, bool hda_codec_use_common_hdmi); -void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev); +void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev, bool enable); void hda_codec_jack_check(struct snd_sof_dev *sdev); #endif /* CONFIG_SND_SOC_SOF_HDA */ From ef4d764c99f792b725d4754a3628830f094f5c58 Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Wed, 13 Jan 2021 02:11:25 +0800 Subject: [PATCH 05/19] ASoC: SOF: Intel: hda: Avoid checking jack on system suspend System takes a very long time to suspend after commit 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization"): [ 90.065964] PM: suspend entry (s2idle) [ 90.067337] Filesystems sync: 0.001 seconds [ 90.185758] Freezing user space processes ... (elapsed 0.002 seconds) done. [ 90.188713] OOM killer disabled. [ 90.188714] Freezing remaining freezable tasks ... (elapsed 0.001 seconds) done. [ 90.190024] printk: Suspending console(s) (use no_console_suspend to debug) [ 90.904912] intel_pch_thermal 0000:00:12.0: CPU-PCH is cool [49C], continue to suspend [ 321.262505] snd_hda_codec_realtek ehdaudio0D0: Unable to sync register 0x2b8000. -5 [ 328.426919] snd_hda_codec_realtek ehdaudio0D0: Unable to sync register 0x2b8000. -5 [ 329.490933] ACPI: EC: interrupt blocked That commit keeps the codec suspended during the system suspend. However, mute/micmute LED will clear codec's direct-complete flag by dpm_clear_superiors_direct_complete(). This doesn't play well with SOF driver. When its runtime resume is called for system suspend, hda_codec_jack_check() schedules jackpoll_work which uses snd_hdac_is_power_on() to check whether codec is suspended. Because the direct-complete path isn't taken, pm_runtime_disable() isn't called so snd_hdac_is_power_on() returns false and jackpoll continues to run, and snd_hda_power_up_pm() cannot power up an already suspended codec in multiple attempts, causes the long delay on system suspend: if (dev->power.direct_complete) { if (pm_runtime_status_suspended(dev)) { pm_runtime_disable(dev); if (pm_runtime_status_suspended(dev)) { pm_dev_dbg(dev, state, "direct-complete "); goto Complete; } pm_runtime_enable(dev); } dev->power.direct_complete = false; } When direct-complete path is taken, snd_hdac_is_power_on() returns true and hda_jackpoll_work() is skipped by accident. So this is still not correct. If we were to use snd_hdac_is_power_on() in system PM path, pm_runtime_status_suspended() should be used instead of pm_runtime_suspended(), otherwise pm_runtime_{enable,disable}() may change the outcome of snd_hdac_is_power_on(). Because devices suspend in reverse order (i.e. child first), it doesn't make much sense to resume an already suspended codec from audio controller. So avoid the issue by making sure jackpoll isn't used in system PM process. Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization") Signed-off-by: Kai-Heng Feng Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20210112181128.1229827-3-kai.heng.feng@canonical.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 7d00107cf3b2..1c5e05b88a90 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -685,7 +685,8 @@ static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume) /* check jack status */ if (runtime_resume) { hda_codec_jack_wake_enable(sdev, false); - hda_codec_jack_check(sdev); + if (sdev->system_suspend_target == SOF_SUSPEND_NONE) + hda_codec_jack_check(sdev); } /* turn off the links that were off before suspend */ From 5e941fc033e411118fb3a7d9e0b97f8cf702cd39 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Wed, 13 Jan 2021 17:56:29 +0200 Subject: [PATCH 06/19] ALSA: hda: Add AlderLake-P PCI ID and HDMI codec vid Add HD Audio PCI ID and HDMI codec vendor ID for Intel AlderLake-P. Signed-off-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Guennadi Liakhovetski Link: https://lore.kernel.org/r/20210113155629.4097057-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e4dd2ff5e473..8d568277088a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2507,6 +2507,9 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-S */ { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Alderlake-P */ + { PCI_DEVICE(0x8086, 0x51c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 74d246a0dc6d..97adff0cbcab 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4346,6 +4346,7 @@ HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), From 9c25af250214e45f6d1c21ff6239a1ffeeedf20e Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Wed, 13 Jan 2021 17:07:15 +0200 Subject: [PATCH 07/19] ASoC: SOF: Intel: fix page fault at probe if i915 init fails The earlier commit to fix runtime PM in case i915 init fails, introduces a possibility to hit a page fault. snd_hdac_ext_bus_device_exit() is designed to be called from dev.release(). Calling it outside device reference counting, is not safe and may lead to calling the device_exit() function twice. Additionally, as part of ext_bus_device_init(), the device is also registered with snd_hdac_device_register(). Thus before calling device_exit(), the device must be removed from device hierarchy first. Fix the issue by rolling back init actions by calling hdac_device_unregister() and then releasing device with put_device(). This matches with existing code in hdac-ext module. To complete the fix, add handling for the case where hda_codec_load_module() returns -ENODEV, and clean up the hdac_ext resources also in this case. In future work, hdac-ext interface should be extended to allow clients more flexibility to handle the life-cycle of individual devices, beyond just the current snd_hdac_ext_bus_device_remove(), which removes all devices. BugLink: https://github.com/thesofproject/linux/issues/2646 Reported-by: Jaroslav Kysela Fixes: 6c63c954e1c5 ("ASoC: SOF: fix a runtime pm issue in SOF when HDMI codec doesn't work") Signed-off-by: Kai Vehmanen Reviewed-by: Rander Wang Reviewed-by: Libin Yang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20210113150715.3992635-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-codec.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index b7e9931ead57..6744318de612 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -153,7 +153,8 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (!hdev->bus->audio_component) { dev_dbg(sdev->dev, "iDisp hw present but no driver\n"); - goto error; + ret = -ENOENT; + goto out; } hda_priv->need_display_power = true; } @@ -170,24 +171,23 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, * other return codes without modification */ if (ret == 0) - goto error; + ret = -ENOENT; } - return ret; - -error: - snd_hdac_ext_bus_device_exit(hdev); - return -ENOENT; - +out: + if (ret < 0) { + snd_hdac_device_unregister(hdev); + put_device(&hdev->dev); + } #else hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL); if (!hdev) return -ENOMEM; ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev, HDA_DEV_ASOC); +#endif return ret; -#endif } /* Codec initialization */ From e4ea77f8e53f9accb9371fba34c189d0447ecce0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 11 Jan 2021 09:16:11 +0100 Subject: [PATCH 08/19] ALSA: usb-audio: Always apply the hw constraints for implicit fb sync Since the commit 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync"), we apply the hw constraints for the implicit feedback sync to make the secondary open aligned with the already opened stream setup. This change assumed that the secondary open is performed after the first stream has been already set up, and adds the hw constraints to sync with the first stream's parameters only when the EP setup for the first stream was confirmed at the open time. However, most of applications handling the full-duplex operations do open both playback and capture streams at first, then set up both streams. This results in skipping the additional hw constraints since the counter-part stream hasn't been set up yet at the open of the second stream, and it eventually leads to "incompatible EP" error in the end. This patch corrects the behavior by always applying the hw constraints for the implicit fb sync. The hw constraint rules are defined so that they check the sync EP dynamically at each invocation, instead. This covers the concurrent stream setups better and lets the hw refine calls resolving to the right configuration. Also this patch corrects a minor error that has existed in the debug print that isn't built as default. Fixes: 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync") Link: https://lore.kernel.org/r/20210111081611.12790-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 171 ++++++++++++++++++++++++++++++------------------ 1 file changed, 108 insertions(+), 63 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 56079901769f..f71965bf815f 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -663,7 +663,7 @@ static int hw_check_valid_format(struct snd_usb_substream *subs, check_fmts.bits[1] = (u32)(fp->formats >> 32); snd_mask_intersect(&check_fmts, fmts); if (snd_mask_empty(&check_fmts)) { - hwc_debug(" > check: no supported format %d\n", fp->format); + hwc_debug(" > check: no supported format 0x%llx\n", fp->formats); return 0; } /* check the channels */ @@ -775,24 +775,11 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, return apply_hw_params_minmax(it, rmin, rmax); } -static int hw_rule_format(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int apply_hw_params_format_bits(struct snd_mask *fmt, u64 fbits) { - struct snd_usb_substream *subs = rule->private; - const struct audioformat *fp; - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - u64 fbits; u32 oldbits[2]; int changed; - hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]); - fbits = 0; - list_for_each_entry(fp, &subs->fmt_list, list) { - if (!hw_check_valid_format(subs, params, fp)) - continue; - fbits |= fp->formats; - } - oldbits[0] = fmt->bits[0]; oldbits[1] = fmt->bits[1]; fmt->bits[0] &= (u32)fbits; @@ -806,6 +793,24 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, return changed; } +static int hw_rule_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct audioformat *fp; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + u64 fbits; + + hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]); + fbits = 0; + list_for_each_entry(fp, &subs->fmt_list, list) { + if (!hw_check_valid_format(subs, params, fp)) + continue; + fbits |= fp->formats; + } + return apply_hw_params_format_bits(fmt, fbits); +} + static int hw_rule_period_time(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -833,64 +838,92 @@ static int hw_rule_period_time(struct snd_pcm_hw_params *params, return apply_hw_params_minmax(it, pmin, UINT_MAX); } -/* apply PCM hw constraints from the concurrent sync EP */ -static int apply_hw_constraint_from_sync(struct snd_pcm_runtime *runtime, - struct snd_usb_substream *subs) +/* get the EP or the sync EP for implicit fb when it's already set up */ +static const struct snd_usb_endpoint * +get_sync_ep_from_substream(struct snd_usb_substream *subs) { struct snd_usb_audio *chip = subs->stream->chip; - struct snd_usb_endpoint *ep; const struct audioformat *fp; - int err; + const struct snd_usb_endpoint *ep; list_for_each_entry(fp, &subs->fmt_list, list) { ep = snd_usb_get_endpoint(chip, fp->endpoint); if (ep && ep->cur_rate) - goto found; + return ep; if (!fp->implicit_fb) continue; /* for the implicit fb, check the sync ep as well */ ep = snd_usb_get_endpoint(chip, fp->sync_ep); if (ep && ep->cur_rate) - goto found; + return ep; } - return 0; + return NULL; +} - found: - if (!find_format(&subs->fmt_list, ep->cur_format, ep->cur_rate, - ep->cur_channels, false, NULL)) { - usb_audio_dbg(chip, "EP 0x%x being used, but not applicable\n", - ep->ep_num); +/* additional hw constraints for implicit feedback mode */ +static int hw_rule_format_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + ep = get_sync_ep_from_substream(subs); + if (!ep) return 0; - } - usb_audio_dbg(chip, "EP 0x%x being used, using fixed params:\n", - ep->ep_num); - usb_audio_dbg(chip, "rate=%d, period_size=%d, periods=%d\n", - ep->cur_rate, ep->cur_period_frames, - ep->cur_buffer_periods); + hwc_debug("applying %s\n", __func__); + return apply_hw_params_format_bits(fmt, pcm_format_to_bits(ep->cur_format)); +} - runtime->hw.formats = subs->formats; - runtime->hw.rate_min = runtime->hw.rate_max = ep->cur_rate; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - runtime->hw.periods_min = runtime->hw.periods_max = - ep->cur_buffer_periods; +static int hw_rule_rate_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_interval *it; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_RATE, - -1); - if (err < 0) - return err; + ep = get_sync_ep_from_substream(subs); + if (!ep) + return 0; - err = snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - ep->cur_period_frames, - ep->cur_period_frames); - if (err < 0) - return err; + hwc_debug("applying %s\n", __func__); + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + return apply_hw_params_minmax(it, ep->cur_rate, ep->cur_rate); +} - return 1; /* notify the finding */ +static int hw_rule_period_size_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_interval *it; + + ep = get_sync_ep_from_substream(subs); + if (!ep) + return 0; + + hwc_debug("applying %s\n", __func__); + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + return apply_hw_params_minmax(it, ep->cur_period_frames, + ep->cur_period_frames); +} + +static int hw_rule_periods_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_interval *it; + + ep = get_sync_ep_from_substream(subs); + if (!ep) + return 0; + + hwc_debug("applying %s\n", __func__); + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIODS); + return apply_hw_params_minmax(it, ep->cur_buffer_periods, + ep->cur_buffer_periods); } /* @@ -899,20 +932,11 @@ static int apply_hw_constraint_from_sync(struct snd_pcm_runtime *runtime, static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) { - struct snd_usb_audio *chip = subs->stream->chip; const struct audioformat *fp; unsigned int pt, ptmin; int param_period_time_if_needed = -1; int err; - mutex_lock(&chip->mutex); - err = apply_hw_constraint_from_sync(runtime, subs); - mutex_unlock(&chip->mutex); - if (err < 0) - return err; - if (err > 0) /* found the matching? */ - goto add_extra_rules; - runtime->hw.formats = subs->formats; runtime->hw.rate_min = 0x7fffffff; @@ -964,7 +988,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (err < 0) return err; -add_extra_rules: err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, subs, SNDRV_PCM_HW_PARAM_FORMAT, @@ -993,6 +1016,28 @@ add_extra_rules: return err; } + /* additional hw constraints for implicit fb */ + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_rate_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + hw_rule_period_size_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, + hw_rule_periods_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_PERIODS, -1); + if (err < 0) + return err; + return 0; } From 495dc7637cb5ca8e39c46db818328410bb6e73a1 Mon Sep 17 00:00:00 2001 From: Chris Chiu Date: Thu, 14 Jan 2021 16:27:28 +0800 Subject: [PATCH 09/19] ALSA: hda/realtek - Limit int mic boost on Acer Aspire E5-575T The Acer Apire E5-575T laptop with codec ALC255 has a terrible background noise comes from internal mic capture. And the jack sensing dose not work for headset like some other Acer laptops. This patch limits the internal mic boost on top of the existing ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk for Acer Aspire E5-575T. Signed-off-by: Chris Chiu Cc: Link: https://lore.kernel.org/r/20210114082728.74729-1-chiu@endlessos.org Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dd82ff2bd5d6..ed5b6b894dc1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6371,6 +6371,7 @@ enum { ALC256_FIXUP_HP_HEADSET_MIC, ALC236_FIXUP_DELL_AIO_HEADSET_MIC, ALC282_FIXUP_ACER_DISABLE_LINEOUT, + ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -7808,6 +7809,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC255_FIXUP_ACER_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7826,6 +7833,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x1025, 0x1094, "Acer Aspire E5-575T", ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1166, "Acer Veriton N4640G", ALC269_FIXUP_LIFEBOOK), From 67ea698c3950d10925be33c21ca49ffb64e21842 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jan 2021 08:24:53 +0100 Subject: [PATCH 10/19] ALSA: hda/via: Add minimum mute flag It turned out that VIA codecs also mute the sound in the lowest mixer level. Turn on the dac_min_mute flag to indicate the mute-as-minimum in TLV like already done in Conexant and IDT codecs. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210559 Cc: Link: https://lore.kernel.org/r/20210114072453.11379-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 0ab40a8a68fb..834367dd54e1 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -113,6 +113,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->codec_type = VT1708S; spec->gen.indep_hp = 1; spec->gen.keep_eapd_on = 1; + spec->gen.dac_min_mute = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; codec->power_save_node = 1; From 217bfbb8b0bfa24619b11ab75c135fec99b99b20 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 Jan 2021 10:34:28 +0100 Subject: [PATCH 11/19] ALSA: seq: oss: Fix missing error check in snd_seq_oss_synth_make_info() snd_seq_oss_synth_make_info() didn't check the error code from snd_seq_oss_midi_make_info(), and this leads to the call of strlcpy() with the uninitialized string as the source, which may lead to the access over the limit. Add the proper error check for avoiding the failure. Reported-by: syzbot+e42504ff21cff05a595f@syzkaller.appspotmail.com Cc: Link: https://lore.kernel.org/r/20210115093428.15882-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_synth.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 11554d0412f0..1b8409ec2c97 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -611,7 +611,8 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in if (info->is_midi) { struct midi_info minf; - snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf); + if (snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf)) + return -ENXIO; inf->synth_type = SYNTH_TYPE_MIDI; inf->synth_subtype = 0; inf->nr_voices = 16; From f84d3a1ec375e46a55cc3ba85c04272b24bd3921 Mon Sep 17 00:00:00 2001 From: Kai-Chuan Hsieh Date: Fri, 15 Jan 2021 11:15:15 +0800 Subject: [PATCH 12/19] ALSA: hda: Add Cometlake-R PCI ID Add HD Audio Device PCI ID for the Intel Cometlake-R platform Reviewed-by: Kai Vehmanen Signed-off-by: Kai-Chuan Hsieh Link: https://lore.kernel.org/r/20210115031515.13100-1-kaichuan.hsieh@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8d568277088a..5a50d3a46445 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2484,6 +2484,9 @@ static const struct pci_device_id azx_ids[] = { /* CometLake-S */ { PCI_DEVICE(0x8086, 0xa3f0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* CometLake-R */ + { PCI_DEVICE(0x8086, 0xf0c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, From 9b268be3adbb410cc1a857477b638a71258891a8 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 15 Jan 2021 16:55:19 +0000 Subject: [PATCH 13/19] MAINTAINERS: update maintainers of qcom audio Add myself as maintainer of qcom audio drivers, as Patrick has very little time to look at the patches. Signed-off-by: Srinivas Kandagatla Reviewed-by: Banajit Goswami Acked-by: Patrick Lai Link: https://lore.kernel.org/r/20210115165520.6023-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index bff306838d57..3fc059ec786a 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -14510,7 +14510,7 @@ S: Supported F: drivers/crypto/qat/ QCOM AUDIO (ASoC) DRIVERS -M: Patrick Lai +M: Srinivas Kandagatla M: Banajit Goswami L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Supported From 7505c06dabb5e814bda610c8d83338544f15db45 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 15 Jan 2021 16:55:20 +0000 Subject: [PATCH 14/19] MAINTAINERS: update qcom ASoC drivers list Add full list of ASoC drivers that are maintained! Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20210115165520.6023-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- MAINTAINERS | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 3fc059ec786a..f32dcf49c27c 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -14514,6 +14514,14 @@ M: Srinivas Kandagatla M: Banajit Goswami L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Supported +F: sound/soc/codecs/lpass-va-macro.c +F: sound/soc/codecs/lpass-wsa-macro.* +F: sound/soc/codecs/msm8916-wcd-analog.c +F: sound/soc/codecs/msm8916-wcd-digital.c +F: sound/soc/codecs/wcd9335.* +F: sound/soc/codecs/wcd934x.c +F: sound/soc/codecs/wcd-clsh-v2.* +F: sound/soc/codecs/wsa881x.c F: sound/soc/qcom/ QCOM IPA DRIVER From 87cb9af9f8a2b242cea7f828206d619e8cbb6a1a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2021 08:58:14 +0100 Subject: [PATCH 15/19] ALSA: usb-audio: Fix UAC1 rate setup for secondary endpoints The current sample rate setup function for UAC1 assumes only the first endpoint retrieved from the interface:altset pair, but the rate set up may be needed also for the secondary endpoint. Also, retrieving the endpoint number from the interface descriptor is redundant; we have already the target endpoint in the given audioformat object. This patch simplifies the code and corrects the target endpoint as described in the above. It simply refers to fmt->endpoint directly. Also, this patch drops the pioneer_djm_set_format_quirk() that is caleld from snd_usb_set_format_quirk(); this function does the sample rate setup but for the capture endpoint (0x82), and that's exactly what the change above fixes. Link: https://lore.kernel.org/r/20210118075816.25068-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 21 ++++++--------------- sound/usb/quirks.c | 28 ---------------------------- 2 files changed, 6 insertions(+), 43 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 31051f2be46d..dc68ed65e478 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -485,18 +485,9 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, const struct audioformat *fmt, int rate) { struct usb_device *dev = chip->dev; - struct usb_host_interface *alts; - unsigned int ep; unsigned char data[3]; int err, crate; - alts = snd_usb_get_host_interface(chip, fmt->iface, fmt->altsetting); - if (!alts) - return -EINVAL; - if (get_iface_desc(alts)->bNumEndpoints < 1) - return -EINVAL; - ep = get_endpoint(alts, 0)->bEndpointAddress; - /* if endpoint doesn't have sampling rate control, bail out */ if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) return 0; @@ -506,11 +497,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, data[2] = rate >> 16; err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data)); + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, + fmt->endpoint, data, sizeof(data)); if (err < 0) { dev_err(&dev->dev, "%d:%d: cannot set freq %d to ep %#x\n", - fmt->iface, fmt->altsetting, rate, ep); + fmt->iface, fmt->altsetting, rate, fmt->endpoint); return err; } @@ -524,11 +515,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data)); + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, + fmt->endpoint, data, sizeof(data)); if (err < 0) { dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", - fmt->iface, fmt->altsetting, ep); + fmt->iface, fmt->altsetting, fmt->endpoint); chip->sample_rate_read_error++; return 0; /* some devices don't support reading */ } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 89e172642d98..e196e364cef1 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1470,30 +1470,6 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs, subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; } - -/* - * Pioneer DJ DJM-900NXS2 - * Device needs to know the sample rate each time substream is started - */ -static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs) -{ - unsigned int cur_rate = subs->data_endpoint->cur_rate; - /* Convert sample rate value to little endian */ - u8 sr[3]; - - sr[0] = cur_rate & 0xff; - sr[1] = (cur_rate >> 8) & 0xff; - sr[2] = (cur_rate >> 16) & 0xff; - - /* Configure device */ - usb_set_interface(subs->dev, 0, 1); - snd_usb_ctl_msg(subs->stream->chip->dev, - usb_rcvctrlpipe(subs->stream->chip->dev, 0), - 0x01, 0x22, 0x0100, 0x0082, &sr, 0x0003); - - return 0; -} - void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1504,10 +1480,6 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; - case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ - case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ - pioneer_djm_set_format_quirk(subs); - break; case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; From 3784d449d795ba11a92681bd22d183329f976421 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2021 08:58:15 +0100 Subject: [PATCH 16/19] ALSA: usb-audio: Set sample rate for all sharing EPs on UAC1 The UAC2/3 sample rate setup is based on the clock node, which is usually shared in the interface, and can't be re-setup without deselecting the interface once, and that's how the current code behaves. OTOH, the sample rate setup of UAC1 is per endpoint, hence we basically need to call for each endpoint usage even if those share the same interface. This patch fixes the behavior of UAC1 to call always snd_usb_init_sample_rate() in snd_usb_endpoint_configure(). Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Link: https://lore.kernel.org/r/20210118075816.25068-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index fe73fe3ff2bc..8e568823c992 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1252,6 +1252,15 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, /* If the interface has been already set up, just set EP parameters */ if (!ep->iface_ref->need_setup) { + /* sample rate setup of UAC1 is per endpoint, and we need + * to update at each EP configuration + */ + if (ep->cur_audiofmt->protocol == UAC_VERSION_1) { + err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, + ep->cur_rate); + if (err < 0) + goto unlock; + } err = snd_usb_endpoint_set_params(chip, ep); if (err < 0) goto unlock; From 532a208ad61018b586cebfca8431291fe9c10ce7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2021 08:58:16 +0100 Subject: [PATCH 17/19] ALSA: usb-audio: Avoid implicit feedback on Pioneer devices For addressing the regression on Pioneer devices, we recently corrected the quirk code to enable the implicit feedback mode on those devices properly. However, the devices still showed problems with the full duplex operations with JACK, and after debug sessions, we figured out that the older kernels that had worked with JACK also didn't use the implicit feedback mode at all although they had the quirk code to enable it; instead, the old code worked just to skip the normal sync endpoint setup that would have been detected without it. IOW, what broke without the implicit-fb quirk in the past was the application of the normal sync endpoint that is actually the capture data endpoint on these devices. This patch covers the overseen piece: it modifies the quirk code again not to enable the implicit feedback mode but just to make the driver skipping the sync endpoint detection. This made the driver working with JACK full-duplex mode again. Still it's not quite clear why the implicit feedback doesn't work on those devices yet; maybe it's about some issues in the URB setup. But at least, with this patch, the driver should work in the level of the older kernels again. Fixes: 167c9dc84ec3 ("ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices") Link: https://lore.kernel.org/r/20210118075816.25068-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/implicit.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 1ac2cc6c33fb..521cc846d9d9 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -175,11 +175,13 @@ static int add_roland_implicit_fb(struct snd_usb_audio *chip, ifnum, alts); } -/* Pioneer devices: playback and capture streams sharing the same iface/altset +/* Playback and capture EPs on Pioneer devices share the same iface/altset, + * but they don't seem working with the implicit fb mode well, hence we + * just return as if the sync were already set up. */ -static int add_pioneer_implicit_fb(struct snd_usb_audio *chip, - struct audioformat *fmt, - struct usb_host_interface *alts) +static int skip_pioneer_sync_ep(struct snd_usb_audio *chip, + struct audioformat *fmt, + struct usb_host_interface *alts) { struct usb_endpoint_descriptor *epd; @@ -194,8 +196,7 @@ static int add_pioneer_implicit_fb(struct snd_usb_audio *chip, (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != USB_ENDPOINT_USAGE_IMPLICIT_FB)) return 0; - return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 1, - alts->desc.bInterfaceNumber, alts); + return 1; /* don't handle with the implicit fb, just skip sync EP */ } static int __add_generic_implicit_fb(struct snd_usb_audio *chip, @@ -298,11 +299,11 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, return 1; } - /* Pioneer devices implicit feedback with vendor spec class */ + /* Pioneer devices with vendor spec class */ if (attr == USB_ENDPOINT_SYNC_ASYNC && alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC && USB_ID_VENDOR(chip->usb_id) == 0x2b73 /* Pioneer */) { - if (add_pioneer_implicit_fb(chip, fmt, alts)) + if (skip_pioneer_sync_ep(chip, fmt, alts)) return 1; } From 2b73649cee65b8e33c75c66348cb1bfe0ff9d766 Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Tue, 19 Jan 2021 23:21:43 +0800 Subject: [PATCH 18/19] ALSA: hda: Balance runtime/system PM if direct-complete is disabled After hibernation, HDA controller can't be runtime-suspended after commit 215a22ed31a1 ("ALSA: hda: Refactor codjc PM to use direct-complete optimization"), which enables direct-complete for HDA codec. The HDA codec driver didn't expect direct-complete will be disabled after it returns a positive value from prepare() callback. However, there are some places that PM core can disable direct-complete. For instance, system hibernation or when codec has subordinates like LEDs. So if the codec is prepared for direct-complete but PM core still calls codec's suspend or freeze callback, partially revert the commit and take the original approach, which uses pm_runtime_force_*() helpers to ensure PM refcount are balanced. Meanwhile, still keep prepare() and complete() callbacks to enable direct-complete and request a resume for jack detection, respectively. Reported-by: Kenneth R. Crudup Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization") Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20210119152145.346558-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 24 +++++++----------------- 1 file changed, 7 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 687216e74526..eec1775dfffe 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2934,7 +2934,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) snd_hdac_leave_pm(&codec->core); } -static int hda_codec_suspend(struct device *dev) +static int hda_codec_runtime_suspend(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); unsigned int state; @@ -2953,7 +2953,7 @@ static int hda_codec_suspend(struct device *dev) return 0; } -static int hda_codec_resume(struct device *dev) +static int hda_codec_runtime_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); @@ -2968,16 +2968,6 @@ static int hda_codec_resume(struct device *dev) return 0; } -static int hda_codec_runtime_suspend(struct device *dev) -{ - return hda_codec_suspend(dev); -} - -static int hda_codec_runtime_resume(struct device *dev) -{ - return hda_codec_resume(dev); -} - #endif /* CONFIG_PM */ #ifdef CONFIG_PM_SLEEP @@ -2998,31 +2988,31 @@ static void hda_codec_pm_complete(struct device *dev) static int hda_codec_pm_suspend(struct device *dev) { dev->power.power_state = PMSG_SUSPEND; - return hda_codec_suspend(dev); + return pm_runtime_force_suspend(dev); } static int hda_codec_pm_resume(struct device *dev) { dev->power.power_state = PMSG_RESUME; - return hda_codec_resume(dev); + return pm_runtime_force_resume(dev); } static int hda_codec_pm_freeze(struct device *dev) { dev->power.power_state = PMSG_FREEZE; - return hda_codec_suspend(dev); + return pm_runtime_force_suspend(dev); } static int hda_codec_pm_thaw(struct device *dev) { dev->power.power_state = PMSG_THAW; - return hda_codec_resume(dev); + return pm_runtime_force_resume(dev); } static int hda_codec_pm_restore(struct device *dev) { dev->power.power_state = PMSG_RESTORE; - return hda_codec_resume(dev); + return pm_runtime_force_resume(dev); } #endif /* CONFIG_PM_SLEEP */ From 506c203cc3de6e26666b8476d287dee81595d6dc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jan 2021 21:45:54 +0100 Subject: [PATCH 19/19] ALSA: usb-audio: Fix hw constraints dependencies Since the recent refactoring, it's been reported that some USB-audio devices (typically webcams) are no longer detected properly by PulseAudio. The debug session revealed that it's failing at probing by PA to try the sample rate 44.1kHz while the device has discrete sample rates other than 44.1kHz. But the puzzle was that arecord works as is, and some other devices with the discrete rates work, either. After all, this turned out to be the lack of the dependencies in a few hw constraint rules: snd_pcm_hw_rule_add() has the (variable) arguments specifying the dependent parameters, and some functions didn't set the target parameter itself as the dependencies. This resulted in an invalid parameter that could be generated only in a certain call pattern. This bug itself has been present in the code, but it didn't trigger errors just because the rules were casually avoiding such a corner case. After the recent refactoring and cleanup, however, the hw constraints work "as expected", and the problem surfaced now. For fixing the problem above, this patch adds the missing dependent parameters to each snd_pcm_hw_rule() call. Fixes: bc4e94aa8e72 ("ALSA: usb-audio: Handle discrete rates properly in hw constraints") BugLink: http://bugzilla.opensuse.org/show_bug.cgi?id=1181014 Link: https://lore.kernel.org/r/20210120204554.30177-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f71965bf815f..078bb4c94033 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -981,6 +981,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_rate, subs, + SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, param_period_time_if_needed, @@ -990,6 +991,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, subs, + SNDRV_PCM_HW_PARAM_CHANNELS, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_RATE, param_period_time_if_needed, @@ -998,6 +1000,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre return err; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_format, subs, + SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_CHANNELS, param_period_time_if_needed,