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

[PATCH v4 00/10] topology: decode: Various fixes
by Piotr Maziarz 03 Sep '20

03 Sep '20

This series fixes various problems with topology decoding mechanism. Some of the problems were critical like improper memory management or infinite loops that were causing undefined behaviour or program crashes, while other resulted in losing some data during conversion. Bugs found while testing with Intel SST topologies. Signed-off-by: Piotr Maziarz <piotrx.maziarz(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Changelog: v2: -Divide into more patches, critical fixes are in separate patches now -More specific descriptions -Fix a typo UML to UCM -Add error reporting in topology: decode: fix channel map memory allocation -Remove goto again in topology: Make buffer for saving dynamic size for better readability v3: -No functional changes -Changed UCM to more descriptive standard ALSA configuration files -removed Gerrit's Change-Id -Added missing signed-offs v4: -Resend, added Reviewed-by tags Piotr Maziarz (10): topology: decode: Fix channel map memory allocation topology: decode: Fix infinite loop in decoding enum control topology: decode: Remove decoding values for enum control topology: decode: Add enum control texts as separate element topology: decode: Fix printing texts section topology: decode: Change declaration of enum decoding function topology: decode: Fix decoding PCM formats and rates topology: decode: Print sig_bits field in PCM capabilities section topology: decode: Add DAI name printing topology: Make buffer for saving dynamic size src/topology/ctl.c | 51 ++++++++++++++++++++++------------------------- src/topology/dapm.c | 3 +-- src/topology/pcm.c | 11 +++++++--- src/topology/save.c | 34 ++++++++++++++++++++++++++----- src/topology/text.c | 2 +- src/topology/tplg_local.h | 2 +- 6 files changed, 64 insertions(+), 39 deletions(-) -- 2.7.4

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[PATCH] ALSA: hda/realtek - Add control fixup for Lenovo Thinkpad X1 Carbon 7th
by Benjamin Poirier 03 Sep '20

03 Sep '20

As a result of commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen"), the sound output level on my machine, an X1 Carbon 7th gen, was reduced to ~65% of its previous level when playing certain sounds. [1] Internally, this laptop model has three outputs (PCM-OUT1, connection 0x02; PCM-OUT2, connection 0x03; SP-OUT PCM, connection 0x06) which can be routed to two sets of stereo speakers: Front (tweeters, node 0x14) and Bass (woofers, node 0x17, aka Rear in some contexts) and one headphone output (node 0x21). The tweeters are noticeably less powerful than the woofers. [2] Before commit d2cd795c4ece, the bass speakers were connected to SP-OUT PCM. SP-OUT PCM is meant for s/pdif output and does not have volume control. This connection made volume control commonly ineffective (using the Master slider in alsa or pulseaudio apparently had little effect or alternated between mute or max with nothing in between). commit d2cd795c4ece added quirk ALC285_FIXUP_SPEAKER2_TO_DAC1 which resulted in assigning both sets of speakers to PCM-OUT1, bringing the two sets of speakers under one effective volume control but also lowering the output volume noticeably. Fix this by connecting PCM-OUT1 to Front speakers and PCM-OUT2 to Rear speakers. The max output volume is restored to what it was before commit d2cd795c4ece. This is done by setting the connection of node 0x17 to 0x03. However, when we do this, the HDA auto config automatically changes the connection of node 0x21 to 0x02. This output, meant for the front speakers, has some "secret" equalizer which changes the output volume according to the level of what's being played, after some delay[3]. This is undesirable with headphones. Therefore, this patch manually limits the connection of node 0x21 to 0x03. Even though there are three speaker/line outputs (nodes 0x14, 0x17, 0x21), there are only two amps for volume control (nodes 0x02, 0x03). When setting node 0x17 to connection 0x03 and node 0x21 to connection 0x03, amp 0x03 is shared by the Woofer and Headphone outputs. Therefore, they cannot have independent volume control. This patch uses the Speaker control element to regulate nodes 0x02 (acts on tweeters) and 0x03 (acts on woofers and headphones). A virtual control element is created for the Headphone mixer which mirrors the Speaker mixer. These changes are only for volume controls; there are 3 hardware mute controls. The current fixup chain applied to this machine is ALC285_FIXUP_THINKPAD_HEADSET_JACK ALC285_FIXUP_SPEAKER2_TO_DAC1 ALC269_FIXUP_THINKPAD_ACPI ALC269_FIXUP_SKU_IGNORE The fixup added by this patch is an alternative to ALC285_FIXUP_SPEAKER2_TO_DAC1 which must be selected explicitly using the module parameter snd_sof_intel_hda_common.hda_model=alc285-tpx1-dual-speakers when using the new SOF driver architecture, or snd_hda_intel.model=alc285-tpx1-dual-speakers when using the snd_hda_intel driver in legacy mode (see the snd_intel_dspcfg.dsp_driver module parameter). In order to be able to apply ALC285_FIXUP_THINKPAD_HEADSET_JACK without changing current behavior, a new variant of that fixup is introduced. When using the model added by this patch, the fixup chain is: ALC285_FIXUP_TPX1_DUAL_SPEAKERS ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI ALC269_FIXUP_THINKPAD_ACPI ALC269_FIXUP_SKU_IGNORE [1] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… [2] https://bugzilla.kernel.org/show_bug.cgi?id=207407#c10 [3] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Link: https://lore.kernel.org/alsa-devel/20200210025249.GA2700@f3/ Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Kailang Yang <kailang(a)realtek.com> Tested-by: Even Brenden <evenbrenden(a)gmail.com> Tested-by: Vincent Bernat <vincent(a)bernat.ch> Signed-off-by: Benjamin Poirier <benjamin.poirier(a)gmail.com> --- This is an updated version of the patchset (now a single patch) to fix the low output volume on the Lenovo ThinkPad X1 Carbon 7th gen. With this patch, the volume output with the Linux driver matches the one with the Windows driver. Volume control is effective and headphone output is unaffected. Multiple users of this machine have reported a low output volume [1] when using a kernel with commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen"). Many have tried these patches or a similar workaround using `hda-verb` and prefer this configuration[2]. [1] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… ... [2] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… ... Changes: v1 == https://lore.kernel.org/alsa-devel/20200211055651.4405-1-benjamin.poirier@g… v2 == https://lore.kernel.org/alsa-devel/20200703080005.8942-1-benjamin.poirier@g… * patch 1 Shortened log * patch 2 Change log to replace experimental observations by information from diagram posted on kernel.org bugzilla. Select exactly the desired connection for node 0x17 (bass speakers) instead of reusing alc295_fixup_disable_dac3(). Control node 0x21 (headphones) connection to avoid the volume "wobble" effect described in the log. Include SET_CONNECT_SEL command in fixup since it is not done by the hda core. Rename one control to reflect its new function and to use as a condition in the ucm config. v3 == * Remove patch 1 which was merged separately as commit 9774dc218bb6 ("ALSA: hda/realtek - Fix Lenovo Thinkpad X1 Carbon 7th quirk subdevice id") * Make fixup activation manual using "alc285-tpx1-dual-speakers" model instead of automatic based on codec id. Suggested by Jaroslav. * Control both tweeter and woofer volume from "Speaker Playback Volume" control. This way a separate ucm change is not needed, unlike in v2 of the patchset. Suggested by Jaroslav. sound/pci/hda/patch_realtek.c | 355 +++++++++++++++++++++++++++++++++- 1 file changed, 351 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1fa983d2a94..8012c4aaf282 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5821,6 +5821,338 @@ static void alc285_fixup_speaker2_to_dac1(struct hda_codec *codec, } } +static struct snd_kcontrol *find_ctl_verbose(struct hda_codec *codec, + const char *name) +{ + struct snd_kcontrol *kctl; + + kctl = snd_hda_find_mixer_ctl(codec, name); + if (!kctl) + codec_warn(codec, "Did not find control \"%s\"\n", name); + return kctl; +} + +struct tpx1_dual_speaker { + struct hda_codec *codec; + struct snd_kcontrol underlying, hp_vol; +}; + +static int tpx1_dual_speaker_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + return speaker_priv->underlying.info(&speaker_priv->underlying, uinfo); +} + +static int tpx1_dual_speaker_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + return speaker_priv->underlying.get(&speaker_priv->underlying, + ucontrol); +} + +static int tpx1_dual_speaker_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + int err; + + /* Control tweeter volume */ + err = speaker_priv->underlying.put(&speaker_priv->underlying, + ucontrol); + if (err < 0) + return err; + + /* Control woofer volume (shared with headphone) */ + err = speaker_priv->hp_vol.put(&speaker_priv->hp_vol, ucontrol); + if (err < 0) + return err; + + snd_ctl_notify(speaker_priv->codec->card, SNDRV_CTL_EVENT_MASK_VALUE, + &speaker_priv->hp_vol.id); + return err; +} + +static int tpx1_dual_speaker_vol_tlv(struct snd_kcontrol *kcontrol, + int op_flag, unsigned int size, + unsigned int __user *tlv) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + return speaker_priv->underlying.tlv.c(&speaker_priv->underlying, + op_flag, size, tlv); +} + +static void tpx1_dual_speaker_vol_free(struct snd_kcontrol *kcontrol) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + if (speaker_priv->underlying.private_free) + speaker_priv->underlying.private_free( + &speaker_priv->underlying); + kfree(speaker_priv); +} + +static int tpx1_dual_override_speaker_vol(struct hda_codec *codec, + struct snd_kcontrol *speaker_vol, + struct snd_kcontrol *hp_vol) +{ + struct tpx1_dual_speaker *speaker_priv; + + speaker_priv = kmalloc(sizeof(struct tpx1_dual_speaker), GFP_KERNEL); + if (!speaker_priv) + return -ENOMEM; + speaker_priv->codec = codec; + memcpy(&speaker_priv->underlying, speaker_vol, + sizeof(struct snd_kcontrol)); + memcpy(&speaker_priv->hp_vol, hp_vol, sizeof(struct snd_kcontrol)); + + speaker_vol->info = &tpx1_dual_speaker_vol_info; + speaker_vol->get = &tpx1_dual_speaker_vol_get; + speaker_vol->put = &tpx1_dual_speaker_vol_put; + speaker_vol->tlv.c = &tpx1_dual_speaker_vol_tlv; + speaker_vol->private_data = speaker_priv; + speaker_vol->private_free = &tpx1_dual_speaker_vol_free; + + return 0; +} + +struct tpx1_dual_hp { + struct hda_codec *codec; + struct snd_kcontrol underlying, *speaker_vol; +}; + +static int tpx1_dual_hp_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + return hp_priv->underlying.info(&hp_priv->underlying, uinfo); +} + +static int tpx1_dual_hp_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + return hp_priv->speaker_vol->get(hp_priv->speaker_vol, ucontrol); +} + +static int tpx1_dual_hp_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + int err; + + err = hp_priv->speaker_vol->put(hp_priv->speaker_vol, ucontrol); + if (err < 0) + return err; + snd_ctl_notify(hp_priv->codec->card, SNDRV_CTL_EVENT_MASK_VALUE, + &hp_priv->speaker_vol->id); + + return 0; +} + +static int tpx1_dual_hp_vol_tlv(struct snd_kcontrol *kcontrol, + int op_flag, unsigned int size, + unsigned int __user *tlv) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + return hp_priv->underlying.tlv.c(&hp_priv->underlying, op_flag, size, + tlv); +} + +static void tpx1_dual_hp_vol_free(struct snd_kcontrol *kcontrol) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + if (hp_priv->underlying.private_free) + hp_priv->underlying.private_free(&hp_priv->underlying); + kfree(hp_priv); +} + +static int tpx1_dual_override_hp_vol(struct hda_codec *codec, + struct snd_kcontrol *speaker_vol, + struct snd_kcontrol *hp_vol) +{ + struct tpx1_dual_hp *hp_priv; + + hp_priv = kmalloc(sizeof(struct tpx1_dual_hp), GFP_KERNEL); + if (!hp_priv) + return -ENOMEM; + hp_priv->codec = codec; + memcpy(&hp_priv->underlying, hp_vol, sizeof(struct snd_kcontrol)); + hp_priv->speaker_vol = speaker_vol; + + hp_vol->info = &tpx1_dual_hp_vol_info; + hp_vol->get = &tpx1_dual_hp_vol_get; + hp_vol->put = &tpx1_dual_hp_vol_put; + hp_vol->tlv.c = &tpx1_dual_hp_vol_tlv; + hp_vol->private_data = hp_priv; + hp_vol->private_free = &tpx1_dual_hp_vol_free; + + return 0; +} + +/* We cannot use snd_hda_add_vmaster() because it expects the controls to be + * HDA (see init_follower_0dB()). This is not the case here. + */ +static int tpx1_dual_create_main_volume_ctl(struct hda_codec *codec) +{ + struct snd_kcontrol *main_vol, *speaker_vol; + struct snd_ctl_elem_value *ucontrol = NULL; + struct hda_gen_spec *spec = codec->spec; + struct snd_ctl_elem_info *uinfo; + int ch, err; + + speaker_vol = find_ctl_verbose(codec, "Speaker Playback Volume"); + if (!speaker_vol) + return -ENOENT; + + main_vol = snd_ctl_make_virtual_master("Master Playback Volume", + spec->vmaster_tlv); + if (!main_vol) + return -ENOMEM; + err = snd_hda_ctl_add(codec, 0, main_vol); + if (err < 0) + return err; + err = snd_ctl_add_follower(main_vol, speaker_vol); + if (err < 0) + return err; + + /* Initialize values */ + uinfo = kmalloc(sizeof(struct snd_ctl_elem_info), GFP_KERNEL); + if (!uinfo) + return -ENOMEM; + + uinfo->id = speaker_vol->id; + err = speaker_vol->info(speaker_vol, uinfo); + if (err) + goto out; + + ucontrol = kmalloc(sizeof(struct snd_ctl_elem_value), GFP_KERNEL); + if (!ucontrol) { + err = -ENOMEM; + goto out; + } + + ucontrol->id = speaker_vol->id; + for (ch = 0; ch < uinfo->count; ch++) { + ucontrol->value.integer.value[ch] = + uinfo->value.integer.max; + } + err = speaker_vol->put(speaker_vol, ucontrol); + if (err < 0) + goto out; + else + err = 0; + +out: + kfree(ucontrol); + kfree(uinfo); + return err; +} + +static void alc285_fixup_tpx1_dual_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + static const hda_nid_t conn[] = { 0x03 }; + + /* For NID 0x17 (bass speakers), the connection list is {0x02, + * 0x03, 0x06}. Disable SP-OUT PCM (0x06) selection since it + * has no volume control, disable PCM1 (0x02) selection since + * it is for front speakers. This leaves PCM2 (0x03). + */ + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), + conn); + /* For NID 0x21 (headphone out), the connection list is {0x02, + * 0x03}. Disable LOUT1 (0x02) selection since its volume + * fluctuates according to input level. This leaves LOUT2 + * (0x03). + */ + snd_hda_override_conn_list(codec, 0x21, ARRAY_SIZE(conn), + conn); + + /* "Master" controls will be created manually after + * subordinate controls have been changed in the + * HDA_FIXUP_ACT_BUILD phase. + */ + spec->gen.suppress_vmaster = 1; + } else if (action == HDA_FIXUP_ACT_INIT) { + /* Because the overridden connection lists contain only a + * single node, __parse_nid_path() does not label the output + * as "multi". This leads snd_hda_activate_path() to skip the + * AC_VERB_SET_CONNECT_SEL even though it might be needed. Do + * it here instead. + * Note that when doing AC_VERB_SET_CONNECT_SEL, the + * connection is specified by index instead of nid. + */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + 0x1); + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_CONNECT_SEL, + 0x1); + } else if (action == HDA_FIXUP_ACT_BUILD) { + struct snd_kcontrol *speaker_vol, *hp_vol; + int err; + + speaker_vol = find_ctl_verbose(codec, "Speaker Playback Volume"); + if (!speaker_vol) + return; + + hp_vol = find_ctl_verbose(codec, "Headphone Playback Volume"); + if (!hp_vol) + return; + + err = tpx1_dual_override_speaker_vol(codec, speaker_vol, hp_vol); + if (err) { + codec_warn(codec, "Failed to override speaker volume control, err %d\n", + err); + return; + } + + err = tpx1_dual_override_hp_vol(codec, speaker_vol, hp_vol); + if (err) { + codec_warn(codec, "Failed to override headphone volume control, err %d\n", + err); + return; + } + + err = tpx1_dual_create_main_volume_ctl(codec); + if (err) { + codec_warn(codec, "Failed to create main volume control, err %d\n", + err); + return; + } + + err = __snd_hda_add_vmaster( + codec, "Master Playback Switch", NULL, + ((const char * const[]){ + "Speaker", "Bass Speaker", "Headphone", NULL + }), "Playback Switch", true, + &spec->gen.vmaster_mute.sw_kctl); + if (err) { + codec_warn(codec, "Failed to create main mute control, err %d\n", + err); + return; + } + + if (spec->gen.vmaster_mute.hook) { + snd_hda_add_vmaster_hook(codec, + &spec->gen.vmaster_mute, + spec->gen.vmaster_mute_enum); + snd_hda_sync_vmaster_hook(&spec->gen.vmaster_mute); + } + } +} + /* Hook to update amp GPIO4 for automute */ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) @@ -6083,6 +6415,7 @@ enum { ALC225_FIXUP_DISABLE_MIC_VREF, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, + ALC285_FIXUP_TPX1_DUAL_SPEAKERS, ALC285_FIXUP_SPEAKER2_TO_DAC1, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, @@ -6135,7 +6468,8 @@ enum { ALC289_FIXUP_DUAL_SPK, ALC294_FIXUP_SPK2_TO_DAC1, ALC294_FIXUP_ASUS_DUAL_SPK, - ALC285_FIXUP_THINKPAD_HEADSET_JACK, + ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK, + ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, ALC285_FIXUP_HP_GPIO_LED, @@ -6905,6 +7239,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_disable_dac3, }, + [ALC285_FIXUP_TPX1_DUAL_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_tpx1_dual_speakers, + .chained = true, + .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI + }, [ALC285_FIXUP_SPEAKER2_TO_DAC1] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_speaker2_to_dac1, @@ -7280,12 +7620,18 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, - [ALC285_FIXUP_THINKPAD_HEADSET_JACK] = { + [ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_jack, .chained = true, .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1 }, + [ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_jack, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, [ALC294_FIXUP_ASUS_HPE] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -7741,8 +8087,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), - SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK), + SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), @@ -7934,6 +8280,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc225-dell1"}, {.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"}, + {.id = ALC285_FIXUP_TPX1_DUAL_SPEAKERS, .name = "alc285-tpx1-dual-speakers"}, {.id = ALC285_FIXUP_SPEAKER2_TO_DAC1, .name = "alc285-speaker2-to-dac1"}, {.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"}, {.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"}, -- 2.28.0

3 6

[PATCH] ALSA: hda: Refactor calculating SDnFMT according to specification
by Pawel Harlozinski 02 Sep '20

02 Sep '20

Fix setting SDnFMT based on High Definition Audio Specification Rev. 1.0a page 48. Bits per Sample (BITS): 000 = 8 bits. The data will be packed in memory in 8-bit containers on 16-bit boundaries. 001 = 16 bits. The data will be packed in memory in 16-bit containers on 16-bit boundaries. 010 = 20 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 011 = 24 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 100 = 32 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 101-111 = Reserved Set SDnFMT depending on which format was given, as maxbps only describes container size. Henceforth split cases for formats 20, 24, 32 bits. For format SNDRV_PCM_FORMAT_FLOAT_LE width is equal 32 so as previously it will end up with mask for 32 bits. Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/hda/hdac_device.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 333220f0f8af..2ccf79866f99 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -762,14 +762,13 @@ unsigned int snd_hdac_calc_stream_format(unsigned int rate, val |= AC_FMT_BITS_16; break; case 20: + val |= AC_FMT_BITS_20; + break; case 24: + val |= AC_FMT_BITS_24; + break; case 32: - if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) - val |= AC_FMT_BITS_32; - else if (maxbps >= 24) - val |= AC_FMT_BITS_24; - else - val |= AC_FMT_BITS_20; + val |= AC_FMT_BITS_32; break; default: return 0; -- 2.17.1

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[bug report] 'ASoC: Intel: haswell: Power transition refactor' and PulseAudio
by Christian Bundy 01 Sep '20

01 Sep '20

After upgrading to Linux 5.8 I discovered an audio issue on my device that was introduced in 8ec7d6043263ecf250b9b7c0dd8ade899487538a [0]. I used 'git bisect' to identify the commit that introduced the bug and have confirmed that reverting the commit resolves the problem Reproduction: 1. Play any audio via PulseAudio. 2. Observe that the audio output is fuzzy and choppy. I can use programs like mpv to play audio without PulseAudio, and the audio is fine, but as soon as I open a process that uses PulseAudio it will ruin the audio output for all processes (including mpv) until I reboot. I'm using a 2015 Chromebook Pixel ("Samus") and have confirmed this problem with a friend who has the same device. Is there anything I can do to help debug this instead of sending a patch to revert the commit? Relevant lspci output: 00:03.0 Audio device: Intel Corporation Broadwell-U Audio Controller (rev 09) Subsystem: Intel Corporation Broadwell-U Audio Controller Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 50 Region 0: Memory at e1218000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee00378 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE- FLReset+ DevCtl: CorrErr- NonFatalErr- FatalErr- UnsupReq- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- FLReset- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- NonFatalErr- FatalErr- UnsupReq- AuxPwr- TransPend- Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel [0]: https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git/commit/?h=…

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[PATCH] soundwire: fix error handling
by trix＠redhat.com 01 Sep '20

01 Sep '20

From: Tom Rix <trix(a)redhat.com> clang static analysis flags this problem stream.c:844:9: warning: Use of memory after it is freed kfree(bus->defer_msg.msg->buf); ^~~~~~~~~~~~~~~~~~~~~~~ This happens in an error handler cleaning up memory allocated for elements in a list. list_for_each_entry(m_rt, &stream->master_list, stream_node) { bus = m_rt->bus; kfree(bus->defer_msg.msg->buf); kfree(bus->defer_msg.msg); } And is triggered when the call to sdw_bank_switch() fails. There are a two problems. First, when sdw_bank_switch() fails, though it frees memory it does not clear bus's reference 'defer_msg.msg' to that memory. The second problem is the freeing msg->buf. In some cases msg will be NULL so this will dereference a null pointer. Need to check before freeing. Fixes: 99b8a5d608a6 ("soundwire: Add bank switch routine") Signed-off-by: Tom Rix <trix(a)redhat.com> --- drivers/soundwire/stream.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 37290a799023..6e36deb505b1 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -717,6 +717,7 @@ static int sdw_bank_switch(struct sdw_bus *bus, int m_rt_count) kfree(wbuf); error_1: kfree(wr_msg); + bus->defer_msg.msg = NULL; return ret; } @@ -840,9 +841,10 @@ static int do_bank_switch(struct sdw_stream_runtime *stream) error: list_for_each_entry(m_rt, &stream->master_list, stream_node) { bus = m_rt->bus; - - kfree(bus->defer_msg.msg->buf); - kfree(bus->defer_msg.msg); + if (bus->defer_msg.msg) { + kfree(bus->defer_msg.msg->buf); + kfree(bus->defer_msg.msg); + } } msg_unlock: -- 2.18.1

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[PATCH] ASoC: rt5682: Prefer async probe
by Douglas Anderson 01 Sep '20

01 Sep '20

The probe of rt5682 is pretty slow. A quick measurement shows that it takes ~650 ms on at least one board. There's no reason to block all other drivers waiting for this probe to finish. Set the flag to allow other drivers to probe while we're probing. Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- NOTE: I haven't done any analysis of the driver to see _why_ it's so slow, only that I have measured it to be slow. Someone could certainly take the time to profile / optimize it, but in any case it still won't hurt to be async. This is a very safe flag to turn on since: 1. It's not like our probe order was defined by anything anyway. When we probe is at the whim of when our i2c controller probes and that can be any time. 2. If some other driver needs us then they have to handle the fact that we might not have probed yet anyway. 3. There may be other drivers probing at the same time as us anyway because _they_ used async probe. While I won't say that it's impossible to tickle a bug by turning on async probe, I would assert that in almost all cases the bug was already there and needed to be fixed anyway. sound/soc/codecs/rt5682-i2c.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 85aba311bdc8..6b4e0eb30c89 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -294,6 +294,7 @@ static struct i2c_driver rt5682_i2c_driver = { .name = "rt5682", .of_match_table = rt5682_of_match, .acpi_match_table = rt5682_acpi_match, + .probe_type = PROBE_PREFER_ASYNCHRONOUS, }, .probe = rt5682_i2c_probe, .shutdown = rt5682_i2c_shutdown, -- 2.28.0.402.g5ffc5be6b7-goog

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[PATCH 1/2] ASoC: wm8994: Skip setting of the WM8994_MICBIAS register for WM1811
by Sylwester Nawrocki 01 Sep '20

01 Sep '20

The WM8994_MICBIAS register is not available in the WM1811 CODEC so skip initialization of that register for that device. This suppresses an error during boot: "wm8994-codec: ASoC: error at snd_soc_component_update_bits on wm8994-codec" Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/codecs/wm8994.c | 2 ++ sound/soc/codecs/wm_hubs.c | 3 +++ sound/soc/codecs/wm_hubs.h | 1 + 3 files changed, 6 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 038be66..b3ba053 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4193,11 +4193,13 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994->hubs.dcs_readback_mode = 2; break; } + wm8994->hubs.micd_scthr = true; break; case WM8958: wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; + wm8994->hubs.micd_scthr = true; switch (control->revision) { case 0: diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 891effe..0c88184 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1223,6 +1223,9 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_component *component, snd_soc_component_update_bits(component, WM8993_ADDITIONAL_CONTROL, WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB); + if (!hubs->micd_scthr) + return 0; + snd_soc_component_update_bits(component, WM8993_MICBIAS, WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK | WM8993_MICB1_LVL | WM8993_MICB2_LVL, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 4b8e5f0..988b29e 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -27,6 +27,7 @@ struct wm_hubs_data { int hp_startup_mode; int series_startup; int no_series_update; + bool micd_scthr; bool no_cache_dac_hp_direct; struct list_head dcs_cache; -- 2.7.4

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[PATCH 00/11] soundwire: intel: add multi-link support
by Bard Liao 01 Sep '20

01 Sep '20

This series enables multi-link support for Intel platforms. Bard Liao (1): soundwire: intel: Only call sdw stream APIs for the first cpu_dai Pierre-Louis Bossart (10): soundwire: intel: disable shim wake on suspend soundwire: intel: ignore software command retries soundwire: intel: add multi-link support soundwire: intel: add missing support for all clock stop modes soundwire: bus: update multi-link definition with hw sync details soundwire: intel: add multi-link hw_synchronization information soundwire: stream: enable hw_sync as needed by hardware soundwire: intel: add dynamic debug trace for clock-stop invalid configs soundwire: intel: pass link_mask information to each master soundwire: intel: don't manage link power individually drivers/soundwire/intel.c | 309 ++++++++++++++++++++++++++++----- drivers/soundwire/intel.h | 2 + drivers/soundwire/intel_init.c | 1 + drivers/soundwire/stream.c | 15 +- include/linux/soundwire/sdw.h | 6 + 5 files changed, 282 insertions(+), 51 deletions(-) -- 2.17.1

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[PATCH 00/10] sound: convert tasklets to use new tasklet_setup()
by Allen Pais 01 Sep '20

01 Sep '20

From: Allen Pais <allen.lkml(a)gmail.com> Commit 12cc923f1ccc ("tasklet: Introduce new initialization API")' introduced a new tasklet initialization API. This series converts all the sound drivers to use the new tasklet_setup() API Allen Pais (10): sound: core: convert tasklets to use new tasklet_setup() API sound: firewire: convert tasklets to use new tasklet_setup() API sound: asihpi: convert tasklets to use new tasklet_setup() API sound: riptide: convert tasklets to use new tasklet_setup() API sound: rm9652: convert tasklets to use new tasklet_setup() API sound/soc: fsl_esai: convert tasklets to use new tasklet_setup() API sound/soc: sh: convert tasklets to use new tasklet_setup() API sound/soc: txx9: convert tasklets to use new tasklet_setup() API sound: midi: convert tasklets to use new tasklet_setup() API sound: ua101: convert tasklets to use new tasklet_setup() API sound/core/timer.c | 7 +++---- sound/firewire/amdtp-stream.c | 8 ++++---- sound/pci/asihpi/asihpi.c | 9 ++++----- sound/pci/riptide/riptide.c | 6 +++--- sound/pci/rme9652/hdsp.c | 6 +++--- sound/pci/rme9652/hdspm.c | 7 +++---- sound/soc/fsl/fsl_esai.c | 7 +++---- sound/soc/sh/siu_pcm.c | 10 ++++------ sound/soc/txx9/txx9aclc.c | 7 +++---- sound/usb/midi.c | 7 +++---- sound/usb/misc/ua101.c | 7 +++---- 11 files changed, 36 insertions(+), 45 deletions(-) -- 2.17.1

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[PATCH v3 0/3] ASoC: soundwire: fix port_ready[] dynamic allocation
by Bard Liao 01 Sep '20

01 Sep '20

The existing code allocates memory for the total number of ports. This only works if the ports are contiguous, but will break if e.g. a Devices uses port0, 1, and 14. The port_ready[] array would contain 3 elements, which would lead to an out-of-bounds access. Conversely in other cases, the wrong port index would be used leading to timeouts on prepare. This can be fixed by allocating for the worst-case of 15 ports (DP0..DP14). In addition since the number is now fixed, we can use an array instead of a dynamic allocation. Changes in v3: - Add ASoC tag in the cover letter title. - Edit the title and commit message of the third patch for better understanding. Changes in v2: - Split patches into sdw and asoc patches. Please note that "soundwire: fix port_ready[] dynamic allocation in mipi_disco" and "ASoC: codecs: fix port_ready[] dynamic allocation in ASoC codecs" should be merged at the same time. Pierre-Louis Bossart (3): soundwire: add definition for maximum number of ports soundwire: fix port_ready[] dynamic allocation in mipi_disco ASoC: codecs: soundwire: remove port_ready[] usage from codecs. drivers/soundwire/mipi_disco.c | 18 +----------------- drivers/soundwire/slave.c | 4 ++++ include/linux/soundwire/sdw.h | 5 +++-- sound/soc/codecs/max98373-sdw.c | 15 +-------------- sound/soc/codecs/rt1308-sdw.c | 14 +------------- sound/soc/codecs/rt5682-sdw.c | 15 +-------------- sound/soc/codecs/rt700-sdw.c | 15 +-------------- sound/soc/codecs/rt711-sdw.c | 15 +-------------- sound/soc/codecs/rt715-sdw.c | 33 +-------------------------------- 9 files changed, 14 insertions(+), 120 deletions(-) -- 2.17.1

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