March 2021 - Alsa-devel - mailman.alsa-project.org

[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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[PATCH 0/3] ALSA: digigram: fix kernel-doc warnings
by Pierre-Louis Bossart 03 Mar '21

03 Mar '21

make W=1 reports these warnings with v5.12-rc1. Pierre-Louis Bossart (3): ALSA: vx: fix kernel-doc warning ALSA: pci: mixart: fix kernel-doc warning ALSA: pci: vx222: fix kernel-doc warning sound/drivers/vx/vx_core.c | 2 +- sound/pci/mixart/mixart_hwdep.c | 3 ++- sound/pci/vx222/vx222_ops.c | 10 +++++----- 3 files changed, 8 insertions(+), 7 deletions(-) -- 2.25.1

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[PATCH] ALSA: hda/realtek: Add quirk for Intel NUC 10
by Werner Sembach 03 Mar '21

03 Mar '21

ALSA: hda/realtek: Add quirk for Intel NUC 10 This adds a new SND_PCI_QUIRK(...) and applies it to the Intel NUC 10 devices. This fixes the issue of the devices not having audio input and output on the headset jack because the kernel does not recognize when something is plugged in. The new quirk was inspired by the quirk for the Intel NUC 8 devices, but it turned out that the NUC 10 uses another pin. This information was acquired by black box testing likely pins. Co-developed-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- Resend of this patch with git send-email, because last patch got tabs replaced with spaces. >From d281364b8ca6c054a0e5ce20caa599bf7d08160d Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:54:30 +0100 Subject: [PATCH] Fix Intel NUC10 no output and input on headset jack --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..c14d624dbaf1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6362,6 +6362,7 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC256_FIXUP_INTEL_NUC10, ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, ALC274_FIXUP_HP_HEADSET_MIC, @@ -7744,6 +7745,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC256_FIXUP_INTEL_NUC10] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, [ALC255_FIXUP_XIAOMI_HEADSET_MIC] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -8183,6 +8193,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), + SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), #if 0 /* Below is a quirk table taken from the old code. -- 2.25.1

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[PATCH v5 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Mar '21

03 Mar '21

On Wed, 2021-02-17 16:29, Takashi Iwai wrote: >On Wed, 17 Feb 2021 05:31:49 +0100, >Gyeongtaek Lee wrote: >> >> 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(). > >I don't think we can apply spin lock there blindly. There is >non-atomic PCM that must not take a spin lock there, too. Hi, Takashi I fixed the patch to hold dpcm_lock during accessing dpcm related data only, not on the trigger callback of driver. Thank you. Gyeongtaek Lee > > >thanks, > >Takashi > Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() include/sound/soc-dpcm.h | 5 ++++ sound/soc/soc-pcm.c | 59 +++++++++++++++++++++++++++++++++------- 2 files changed, 54 insertions(+), 10 deletions(-) base-commit: fe07bfda2fb9cdef8a4d4008a409bb02f35f1bd8 -- 2.21.0

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Re: [PATCH] ASoC: wm8960: Remove bitclk relax condition
by S.j. Wang 03 Mar '21

03 Mar '21

> > > > Using a higher bitclk then expected doesn't always work. > > Here is an example: > > > > aplay -Dhw:0,0 -d 5 -r 48000 -f S24_LE -c 2 audio48k24b2c.wav > > > > In this case, the required bitclk is 48000 * 24 * 2 = 2304000 but the > > closest bitclk that can be derived is 3072000. Since the clock is > > faster than expected, it will start to send bytes from the next > > channel so the sound will be corrupted. > > > > Fixes: 82bab88910ee ("ASoC: codec: wm8960: Relax bit clock computation > > when using PLL") > > Fixes: 3c01b9ee2ab9 ("ASoC: codec: wm8960: Relax bit clock > > computation") > > Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > I think this is probably going to need a much more involved fix. > The problem is that there are systems that depend on this behaviour, so you > can't just flat out revert it. And to be fair the I2S specification says that bit > clock can run at a higher rate than required for the data, so the behaviour is > correct as well. > > Probably the best solution here is to add additional contraints from the > machine driver on which rates/bit depths/channels are supported. > Just double check the issue, the real reason is in below: The call sequence in wm8960_configure_clocking is ret = wm8960_configure_sysclk(); if (ret >= 0) goto configure_clock; .... ret = wm8960_configure_pll(); configure_clock: ... wm8960_configure_sysclk is called before wm8960_configure_pll, as there is bitclk relax on both functions, so wm8960_configure_sysclk always return success, then wm8960_configure_pll() never be called. So bitclk relax condition should be removed in wm8960_configure_sysclk, That wm8960_configure_pll can be called, and there is also bitclk relax function in wm8960_configure_pll. I will update and resend this patch. Best regards Wang shengjiu

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