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[PATCH][next] ASoC: fsl: fsl_easrc: Fix uninitialized variable st2_mem_alloc
by Colin King 04 Mar '21

04 Mar '21

From: Colin Ian King <colin.king(a)canonical.com> A previous cleanup commit removed the ininitialization of st2_mem_alloc. Fix this by restoring the original behaviour by initializing it to zero. Addresses-Coverity: ("Uninitialized scalar variable") Fixes: e80382fe721f ("ASoC: fsl: fsl_easrc: remove useless assignments") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/fsl/fsl_easrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 725a5d3aaa02..e823c9c13764 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -710,7 +710,7 @@ static int fsl_easrc_max_ch_for_slot(struct fsl_asrc_pair *ctx, struct fsl_easrc_slot *slot) { struct fsl_easrc_ctx_priv *ctx_priv = ctx->private; - int st1_mem_alloc = 0, st2_mem_alloc; + int st1_mem_alloc = 0, st2_mem_alloc = 0; int pf_mem_alloc = 0; int max_channels = 8 - slot->num_channel; int channels = 0; -- 2.30.0

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conf: Remove non-existent SPDIF output on Dell AE515
by GitHub pull_request - opened 03 Mar '21

03 Mar '21

alsa-project/alsa-lib pull request #124 was opened from drjhe: The Dell Professional Sound Bar AE515 shows up in GNOME Settings with a digital output that doesn't physically exist. Add it to the list to suppress this. Request URL : https://github.com/alsa-project/alsa-lib/pull/124 Patch URL : https://github.com/alsa-project/alsa-lib/pull/124.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ALSA: usb: Add Plantronics C320-M USB ctrl msg delay quirk
by John Ernberg 03 Mar '21

03 Mar '21

The microphone in the Plantronics C320-M headset will randomly fail to initialize properly, at least when using Microsoft Teams. Introducing a 20ms delay on the control messages appears to resolve the issue. Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1065 Tested-by: Andreas Kempe <kempe(a)lysator.liu.se> Signed-off-by: John Ernberg <john.ernberg(a)actia.se> --- Note: Microsoft Teams is the only application tests have been performed with and no other kinds of investigations have been performed, like USB packet logging. Originally reported as a possible PulseAudio issue, where it was suggested to try a delay quirk in the kernel. If other tests should be performed, please let me know which tests and I will try to do them. All testing was done on 5.10, patch was then applied on sound.git master for submission I did not register to the mailing lists, please keep me CC on any comments. Thanks! sound/usb/quirks.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d02dac5fcd40..a31b7db92b1e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1673,6 +1673,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) msleep(20); + /* + * Plantronics C320-M needs a delay to avoid random + * microhpone failures. + */ + if (chip->usb_id == USB_ID(0x047f, 0xc025) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + msleep(20); + /* Zoom R16/24, many Logitech(at least H650e/H570e/BCC950), * Jabra 550a, Kingston HyperX needs a tiny delay here, * otherwise requests like get/set frequency return -- 2.30.0

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[PATCH 0/4] ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec.
by Vitaly Rodionov 03 Mar '21

03 Mar '21

Dell's laptops Inspiron 3500, Inspiron 3501, Inspiron 3505 are using Cirrus Logic CS8409 HDA bridge with CS42L42 companion codec. The CS8409 is a multichannel HD audio routing controller. CS8409 includes support for four channels of digital microphone data and two bidirectional ASPs for up to 32 channels of TDM data or 4 channels of I2S data. The CS8409 is intended to be used with a remote companion codec that implements high performance analog functions in close physical proximity to the end-equipment audio port or speaker driver. The CS42L42 is a low-power audio codec with integrated MIPI SoundWire interface or I2C/I2S/TDM interfaces designed for portable applications. It provides a high-dynamic range, stereo DAC for audio playback and a mono high-dynamic-range ADC for audio capture Stefan Binding (1): ALSA: hda/cirrus: Add Headphone and Headset MIC Volume Control Vitaly Rodionov (3): ALSA: hda/cirrus: Increase AUTO_CFG_MAX_INS from 8 to 18 ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec. ALSA: hda/cirrus: Add jack detect interrupt support from CS42L42 companion codec. sound/pci/hda/hda_auto_parser.h | 2 +- sound/pci/hda/hda_local.h | 2 +- sound/pci/hda/patch_cirrus.c | 1033 +++++++++++++++++++++++++++++++ 3 files changed, 1035 insertions(+), 2 deletions(-) -- 2.25.1

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[PATCH] ASoC: SOF: Intel: broadwell: fix mutual exclusion with catpt driver
by Pierre-Louis Bossart 03 Mar '21

03 Mar '21

In v5.10, the "haswell" driver was replaced by the "catpt" driver, but the mutual exclusion with the SOF driver was not updated. This leads to errors with card names and UCM profiles not being loaded by PulseAudio. This fix should only be applied on v5.10-stable, the mutual exclusion was removed in 5.11. Reported-by: David Ward <david.ward(a)ll.mit.edu> Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=211985 Fixes: 6cbfa11d2694 ("ASoC: Intel: Select catpt and deprecate haswell") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: <stable(a)kernel.org> --- sound/soc/sof/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index de7ff2d097ab..6708a2c5a838 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -84,7 +84,7 @@ config SND_SOC_SOF_BAYTRAIL config SND_SOC_SOF_BROADWELL_SUPPORT bool "SOF support for Broadwell" - depends on SND_SOC_INTEL_HASWELL=n + depends on SND_SOC_INTEL_CATPT=n help This adds support for Sound Open Firmware for Intel(R) platforms using the Broadwell processors. -- 2.25.1

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[PATCH v2 0/5] soundwire: qcom: various improvements
by Srinivas Kandagatla 03 Mar '21

03 Mar '21

Thanks for reviewing v1 of this patchset! During testing SoundWire controller on SM8250 MTP, we found few issues like all the interrupts are not handled, all transport parameters are not read from device tree. Other major issue was register read/writes which was interrupt based was an overhead and puts lot of limitation on context it can be used from. With previous approach number of interrupts generated after enumeration are around 130: $ cat /proc/interrupts | grep soundwire 21: 130 0 0 0 0 0 0 0 GICv3 234 Edge soundwire after this patch they are just 3 interrupts $ cat /proc/interrupts | grep soundwire 21: 3 0 0 0 0 0 0 0 GICv3 234 Edge soundwire So this patchset add various improvements to the existing driver to address above issues. Tested it on SM8250 MTP with 2x WSA881x speakers, HeadPhones on WCD938x via lpass-rx-macro and Analog MICs via lpass-tx-macro. Also tested on DragonBoard DB845c with 2xWSA881x speakers. Changes since v1: - changed bgp_count to blk_group_count as suggested by Pierre - used version raw value as suggested by VKoul - updated read fifo logic and interrupt handling to do while() - renamed some of the variable to make reading easy! - removed patch that parses version Srinivas Kandagatla (5): soundwire: qcom: add support to missing transport params soundwire: qcom: set continue execution flag for ignored commands soundwire: qcom: start the clock during initialization soundwire: qcom: update register read/write routine soundwire: qcom: add support to new interrupts drivers/soundwire/qcom.c | 438 ++++++++++++++++++++++++++++++--------- 1 file changed, 335 insertions(+), 103 deletions(-) -- 2.21.0

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[PATCH] ALSA: hda/realtek: Apply dual codec quirks for MSI Godlike X570 board
by Takashi Iwai 03 Mar '21

03 Mar '21

There is another MSI board (1462:cc34) that has dual Realtek codecs, and we need to apply the existing quirk for fixing the conflicts of Master control. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211743 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7c24d13181e1..b47504fa8dfd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2532,6 +2532,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), + SND_PCI_QUIRK(0x1462, 0xcc34, "MSI Godlike X570", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), -- 2.26.2

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[PATCH] ALSA: hda - bind headset buttons to the headphone jack
by Hui Wang 03 Mar '21

03 Mar '21

With the HDA driver, if the headset buttons are supported, an audio Jack will be created for them. This audio Jack is a bit confusing to users since it can't report headphone/mic insertion events but it claims to support these events. And in addition, the driver already builds a headphone Jack and a mic Jack, and most of those buttons are used for headphone playback, so do some change to bind those buttons to the headphone Jack. After this change, the key events are generated from NID 0x55, and are reported to the input layer via headphone jack (NID 0x21). If there is no headphone Jack, then build an audio jack to support those buttons same as previously. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_jack.c | 43 ++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_jack.h | 5 ++++ sound/pci/hda/patch_realtek.c | 23 ++++++++++++++++--- 3 files changed, 67 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index b8b568046592..640b72b598e8 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -388,6 +388,39 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, } EXPORT_SYMBOL_GPL(snd_hda_jack_set_gating_jack); +/** + * snd_hda_jack_bind_keymap - bind keys generated from one NID to another jack. + * @codec: the HDA codec + * @key_nid: key event is generated by this pin NID + * @keymap: map of key type and key code + * @jack_nid: key reports to the jack of this pin NID + * + * This function is used in the case of key is generated from one NID while is + * reported to the jack of another NID. + */ +int snd_hda_jack_bind_keymap(struct hda_codec *codec, hda_nid_t key_nid, + const struct hda_jack_keymap *keymap, + hda_nid_t jack_nid) +{ + const struct hda_jack_keymap *map; + struct hda_jack_tbl *key_gen = snd_hda_jack_tbl_get(codec, key_nid); + struct hda_jack_tbl *report_to = snd_hda_jack_tbl_get(codec, jack_nid); + + WARN_ON(codec->dp_mst); + + if (!key_gen || !report_to || !report_to->jack) + return -EINVAL; + + key_gen->key_report_jack = jack_nid; + + if (keymap) + for (map = keymap; map->type; map++) + snd_jack_set_key(report_to->jack, map->type, map->key); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hda_jack_bind_keymap); + /** * snd_hda_jack_report_sync - sync the states of all jacks and report if changed * @codec: the HDA codec @@ -651,7 +684,15 @@ void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res) } if (!event) return; - event->jack_dirty = 1; + + if (event->key_report_jack) { + struct hda_jack_tbl *report_to = + snd_hda_jack_tbl_get_mst(codec, event->key_report_jack, + event->dev_id); + if (report_to) + report_to->jack_dirty = 1; + } else + event->jack_dirty = 1; call_jack_callback(codec, res, event); snd_hda_jack_report_sync(codec); diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 8ceaf0ef5df1..339746dd9578 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -40,6 +40,7 @@ struct hda_jack_tbl { unsigned int block_report:1; /* in a transitional state - do not report to userspace */ hda_nid_t gating_jack; /* valid when gating jack plugged */ hda_nid_t gated_jack; /* gated is dependent on this jack */ + hda_nid_t key_report_jack; /* key reports to this jack */ int type; int button_state; struct snd_jack *jack; @@ -99,6 +100,10 @@ snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid); +int snd_hda_jack_bind_keymap(struct hda_codec *codec, hda_nid_t key_nid, + const struct hda_jack_keymap *keymap, + hda_nid_t jack_nid); + u32 snd_hda_jack_pin_sense(struct hda_codec *codec, hda_nid_t nid, int dev_id); /* the jack state returned from snd_hda_jack_detect_state() */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52907506e16e..f381474f9403 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3102,6 +3102,16 @@ static void alc_headset_btn_callback(struct hda_codec *codec, if (jack->unsol_res & (7 << 10)) report |= SND_JACK_BTN_3; + if (jack->jack->key_report_jack) { + struct hda_jack_tbl *report_to = + snd_hda_jack_tbl_get(codec, jack->jack->key_report_jack); + + if (report_to) { + report_to->button_state = report; + return; + } + } + jack->jack->button_state = report; } @@ -3163,16 +3173,23 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: spec->has_hs_key = 1; snd_hda_jack_detect_enable_callback(codec, 0x55, alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); break; - case HDA_FIXUP_ACT_INIT: + case HDA_FIXUP_ACT_BUILD: + hp_pin = alc_get_hp_pin(spec); + if (!hp_pin || snd_hda_jack_bind_keymap(codec, 0x55, + alc_headset_btn_keymap, + hp_pin)) + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", + false, SND_JACK_HEADSET, + alc_headset_btn_keymap); + alc_enable_headset_jack_key(codec); break; } -- 2.25.1

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[PATCH] ASoC: soc-core: Prevent warning if no DMI table is present
by Jon Hunter 03 Mar '21

03 Mar '21

Many systems do not provide a DMI table and on these systems a warning, such as the following, is printed on boot ... WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name! If DMI support is enabled in the kernel, there is no simple way to detect if a DMI table is table or not. Note that the variable 'dmi_available' is not exported and so cannot be used by kernel modules. It could be possible to have every ASoC sound card driver set the long name to avoid the above message, but it might be intentional for the long name, that we fall back to using the sound card name. Therefore, make this a debug print by default to avoid the warning. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f6d4e99b590c..f1189e7c1fcc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1576,7 +1576,7 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) /* make up dmi long name as: vendor-product-version-board */ vendor = dmi_get_system_info(DMI_BOARD_VENDOR); if (!vendor || !is_dmi_valid(vendor)) { - dev_warn(card->dev, "ASoC: no DMI vendor name!\n"); + dev_dbg(card->dev, "ASoC: no DMI vendor name!\n"); return 0; } -- 2.25.1

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[PATCH v5 1/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Mar '21

03 Mar '21

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. CPU0 CPU1 dpcm_be_dai_trigger(): dpcm_be_disconnect(): for_each_dpcm_be(fe, stream, dpcm) { spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); struct snd_soc_pcm_runtime *be = dpcm->be; <-- Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_be_dai_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- include/sound/soc-dpcm.h | 5 ++++ sound/soc/soc-pcm.c | 59 +++++++++++++++++++++++++++++++++------- 2 files changed, 54 insertions(+), 10 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 0f6c50b17bba..599cd6054bc3 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -103,6 +103,11 @@ struct snd_soc_dpcm_runtime { int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */ }; +struct snd_soc_dpcm_rtd_list { + int num_rtds; + struct snd_soc_pcm_runtime *rtds[0]; +}; + #define for_each_dpcm_fe(be, stream, _dpcm) \ list_for_each_entry(_dpcm, &(be)->dpcm[stream].fe_clients, list_fe) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 14d85ca1e435..ccda0f77e827 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2028,15 +2028,51 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_be_list_create(struct snd_soc_pcm_runtime *fe, int stream, + struct snd_soc_dpcm_rtd_list **be_list) +{ + struct snd_soc_dpcm *dpcm; + unsigned long flags; + int size, i, ret = 0; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + + size = 0; + for_each_dpcm_be(fe, stream, dpcm) + size++; + + *be_list = kzalloc(struct_size(*be_list, rtds, size), GFP_ATOMIC); + if (*be_list) { + i = 0; + for_each_dpcm_be(fe, stream, dpcm) + (*be_list)->rtds[i++] = dpcm->be; + + if (i != size) + dev_err(fe->dev, "ASoC: abnormal change in be clients: %d -> %d\n", + size, i); + + (*be_list)->num_rtds = i; + } else { + ret = -ENOMEM; + } + + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} + int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { - struct snd_soc_dpcm *dpcm; - int ret = 0; + struct snd_soc_dpcm_rtd_list *be_list; + int i, ret = 0; - for_each_dpcm_be(fe, stream, dpcm) { + ret = dpcm_be_list_create(fe, stream, &be_list); + if (ret < 0) + return ret; - struct snd_soc_pcm_runtime *be = dpcm->be; + for (i = 0; i < be_list->num_rtds; i++) { + struct snd_soc_pcm_runtime *be = be_list->rtds[i]; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); @@ -2056,7 +2092,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2066,7 +2102,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2076,7 +2112,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2090,7 +2126,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2103,7 +2139,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2116,13 +2152,16 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = soc_pcm_trigger(be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret < 0) + break; } + kfree(be_list); return ret; } EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); -- 2.21.0

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