From abc3caac24501008465fdb55c5e89e16d58d5a3d Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 27 Mar 2020 16:47:24 -0400 Subject: ASoC: topology: Add missing memory checks MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit kstrdup is an allocation function and it can fail, so its return value should be checked and handled appropriately. In order to check all cases, we need to modify set_stream_info to return a value, so check that everything went correctly when doing kstrdup(). Later add proper checks and error handlers. Signed-off-by: Amadeusz Sławiński Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200327204729.397-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 62 ++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 49 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 87f75edba3dc..73fc304c9aca 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1766,10 +1766,13 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) return 0; } -static void set_stream_info(struct snd_soc_pcm_stream *stream, +static int set_stream_info(struct snd_soc_pcm_stream *stream, struct snd_soc_tplg_stream_caps *caps) { stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + if (!stream->stream_name) + return -ENOMEM; + stream->channels_min = le32_to_cpu(caps->channels_min); stream->channels_max = le32_to_cpu(caps->channels_max); stream->rates = le32_to_cpu(caps->rates); @@ -1777,6 +1780,8 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream, stream->rate_max = le32_to_cpu(caps->rate_max); stream->formats = le64_to_cpu(caps->formats); stream->sig_bits = le32_to_cpu(caps->sig_bits); + + return 0; } static void set_dai_flags(struct snd_soc_dai_driver *dai_drv, @@ -1812,20 +1817,29 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (dai_drv == NULL) return -ENOMEM; - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!dai_drv->name) { + ret = -ENOMEM; + goto err; + } + } dai_drv->id = le32_to_cpu(pcm->dai_id); if (pcm->playback) { stream = &dai_drv->playback; caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->capture) { stream = &dai_drv->capture; caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->compress) @@ -1835,11 +1849,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - kfree(dai_drv->playback.stream_name); - kfree(dai_drv->capture.stream_name); - kfree(dai_drv->name); - kfree(dai_drv); - return ret; + goto err; } dai_drv->dobj.index = tplg->index; @@ -1860,6 +1870,14 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, return ret; } + return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + kfree(dai_drv->name); + kfree(dai_drv); + return ret; } @@ -1916,11 +1934,20 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, if (strlen(pcm->pcm_name)) { link->name = kstrdup(pcm->pcm_name, GFP_KERNEL); link->stream_name = kstrdup(pcm->pcm_name, GFP_KERNEL); + if (!link->name || !link->stream_name) { + ret = -ENOMEM; + goto err; + } } link->id = le32_to_cpu(pcm->pcm_id); - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { link->cpus->dai_name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!link->cpus->dai_name) { + ret = -ENOMEM; + goto err; + } + } link->codecs->name = "snd-soc-dummy"; link->codecs->dai_name = "snd-soc-dummy-dai"; @@ -2436,13 +2463,17 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, if (d->playback) { stream = &dai_drv->playback; caps = &d->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->capture) { stream = &dai_drv->capture; caps = &d->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->flag_mask) @@ -2454,10 +2485,15 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - return ret; + goto err; } return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + return ret; } /* load physical DAI elements */ -- cgit v1.2.3 From 482db55ae87f3749db05810a38b1d618dfd4407c Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 27 Mar 2020 16:47:25 -0400 Subject: ASoC: topology: Check return value of soc_tplg_create_tlv MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Function soc_tplg_create_tlv can fail, so we should check if it succeded or not and proceed appropriately. Signed-off-by: Amadeusz Sławiński Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200327204729.397-3-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 73fc304c9aca..f37a72aebb5a 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -894,7 +894,13 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc, &mc->hdr); + err = soc_tplg_create_tlv(tplg, &kc, &mc->hdr); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to create TLV %s\n", + mc->hdr.name); + kfree(sm); + continue; + } /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc, @@ -1355,7 +1361,13 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr); + err = soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to create TLV %s\n", + mc->hdr.name); + kfree(sm); + continue; + } /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc[i], -- cgit v1.2.3 From 2ae548f30d7f6973388fc3769bb3c2f6fd13652b Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 27 Mar 2020 16:47:26 -0400 Subject: ASoC: topology: Check return value of soc_tplg_*_create MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Functions soc_tplg_denum_create, soc_tplg_dmixer_create, soc_tplg_dbytes_create can fail, so their return values should be checked and error should be propagated. Signed-off-by: Amadeusz Sławiński Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200327204729.397-4-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index f37a72aebb5a..3ada769cf823 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1124,6 +1124,7 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { struct snd_soc_tplg_ctl_hdr *control_hdr; + int ret; int i; if (tplg->pass != SOC_TPLG_PASS_MIXER) { @@ -1152,25 +1153,30 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, case SND_SOC_TPLG_CTL_RANGE: case SND_SOC_TPLG_DAPM_CTL_VOLSW: case SND_SOC_TPLG_DAPM_CTL_PIN: - soc_tplg_dmixer_create(tplg, 1, - le32_to_cpu(hdr->payload_size)); + ret = soc_tplg_dmixer_create(tplg, 1, + le32_to_cpu(hdr->payload_size)); break; case SND_SOC_TPLG_CTL_ENUM: case SND_SOC_TPLG_CTL_ENUM_VALUE: case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - soc_tplg_denum_create(tplg, 1, - le32_to_cpu(hdr->payload_size)); + ret = soc_tplg_denum_create(tplg, 1, + le32_to_cpu(hdr->payload_size)); break; case SND_SOC_TPLG_CTL_BYTES: - soc_tplg_dbytes_create(tplg, 1, - le32_to_cpu(hdr->payload_size)); + ret = soc_tplg_dbytes_create(tplg, 1, + le32_to_cpu(hdr->payload_size)); break; default: soc_bind_err(tplg, control_hdr, i); return -EINVAL; } + if (ret < 0) { + dev_err(tplg->dev, "ASoC: invalid control\n"); + return ret; + } + } return 0; -- cgit v1.2.3 From 6856e887eae3efc0fe56899cb3f969fe063171c5 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 27 Mar 2020 16:47:27 -0400 Subject: ASoC: topology: Check soc_tplg_add_route return value MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Function soc_tplg_add_route can propagate error code from callback, we should check its return value and handle fail in correct way. Signed-off-by: Amadeusz Sławiński Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200327204729.397-5-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 3ada769cf823..cb43994089de 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1284,7 +1284,9 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, routes[i]->dobj.index = tplg->index; list_add(&routes[i]->dobj.list, &tplg->comp->dobj_list); - soc_tplg_add_route(tplg, routes[i]); + ret = soc_tplg_add_route(tplg, routes[i]); + if (ret < 0) + break; /* add route, but keep going if some fail */ snd_soc_dapm_add_routes(dapm, routes[i], 1); -- cgit v1.2.3 From b3677fc3d68dd942c92de52f0bd9dd8b472a40e6 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 27 Mar 2020 16:47:28 -0400 Subject: ASoC: topology: Check return value of pcm_new_ver MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Function pcm_new_ver can fail, so we should check it's return value and handle possible error. Signed-off-by: Amadeusz Sławiński Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200327204729.397-6-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index cb43994089de..818657b06799 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2135,7 +2135,9 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, _pcm = pcm; } else { abi_match = false; - pcm_new_ver(tplg, pcm, &_pcm); + ret = pcm_new_ver(tplg, pcm, &_pcm); + if (ret < 0) + return ret; } /* create the FE DAIs and DAI links */ -- cgit v1.2.3 From dd8e871d4e560eeb8d22af82dde91457ad835a63 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 27 Mar 2020 16:47:29 -0400 Subject: ASoC: topology: Check return value of soc_tplg_dai_config MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Function soc_tplg_dai_config can fail, check for and handle possible failure. Signed-off-by: Amadeusz Sławiński Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200327204729.397-7-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 818657b06799..33e8d189ba2f 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2524,7 +2524,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, { struct snd_soc_tplg_dai *dai; int count; - int i; + int i, ret; count = le32_to_cpu(hdr->count); @@ -2539,7 +2539,12 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, return -EINVAL; } - soc_tplg_dai_config(tplg, dai); + ret = soc_tplg_dai_config(tplg, dai); + if (ret < 0) { + dev_err(tplg->dev, "ASoC: failed to configure DAI\n"); + return ret; + } + tplg->pos += (sizeof(*dai) + le32_to_cpu(dai->priv.size)); } -- cgit v1.2.3 From 4df933252827af69cb087e3df1294e4945a6f6c6 Mon Sep 17 00:00:00 2001 From: Xu Wang Date: Thu, 9 Apr 2020 19:20:52 +0800 Subject: ALSA: ctxfi: Remove unnecessary cast in kfree Remove unnecassary casts in the argument to kfree. Signed-off-by: Xu Wang Link: https://lore.kernel.org/r/20200409112052.13402-1-vulab@iscas.ac.cn Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cthw20k1.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 6e3177bcc709..015c0d676897 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -168,7 +168,7 @@ static int src_get_rsc_ctrl_blk(void **rblk) static int src_put_rsc_ctrl_blk(void *blk) { - kfree((struct src_rsc_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -494,7 +494,7 @@ static int src_mgr_get_ctrl_blk(void **rblk) static int src_mgr_put_ctrl_blk(void *blk) { - kfree((struct src_mgr_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -515,7 +515,7 @@ static int srcimp_mgr_get_ctrl_blk(void **rblk) static int srcimp_mgr_put_ctrl_blk(void *blk) { - kfree((struct srcimp_mgr_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -702,7 +702,7 @@ static int amixer_rsc_get_ctrl_blk(void **rblk) static int amixer_rsc_put_ctrl_blk(void *blk) { - kfree((struct amixer_rsc_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -909,7 +909,7 @@ static int dai_get_ctrl_blk(void **rblk) static int dai_put_ctrl_blk(void *blk) { - kfree((struct dai_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -958,7 +958,7 @@ static int dao_get_ctrl_blk(void **rblk) static int dao_put_ctrl_blk(void *blk) { - kfree((struct dao_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -1156,7 +1156,7 @@ static int daio_mgr_get_ctrl_blk(struct hw *hw, void **rblk) static int daio_mgr_put_ctrl_blk(void *blk) { - kfree((struct daio_mgr_ctrl_blk *)blk); + kfree(blk); return 0; } -- cgit v1.2.3 From 4963d66b8a26c489958063abb6900ea6ed8e4836 Mon Sep 17 00:00:00 2001 From: Adam Barber Date: Fri, 10 Apr 2020 17:00:32 +0800 Subject: ALSA: hda/realtek - Enable the headset mic on Asus FX505DT On Asus FX505DT with Realtek ALC233, the headset mic is connected to pin 0x19, with default 0x411111f0. Enable headset mic by reconfiguring the pin to an external mic associated with the headphone on 0x21. Mic jack detection was also found to be working. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207131 Signed-off-by: Adam Barber Cc: Link: https://lore.kernel.org/r/20200410090032.2759-1-barberadam995@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index de2826f90d34..dc5557d79c43 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7378,6 +7378,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), -- cgit v1.2.3 From 48cc42973509afac24e83d6edc23901d102872d1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 12 Apr 2020 10:13:28 +0200 Subject: ALSA: usb-audio: Filter error from connector kctl ops, too The ignore_ctl_error option should filter the error at kctl accesses, but there was an overlook: mixer_ctl_connector_get() returns an error from the request. This patch covers the forgotten code path and apply filter_error() properly. The locking error is still returned since this is a fatal error that has to be reported even with ignore_ctl_error option. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Cc: Link: https://lore.kernel.org/r/20200412081331.4742-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 721d12130d0c..d27e390dcd32 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1457,7 +1457,7 @@ error: usb_audio_err(chip, "cannot get connectors status: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", UAC_GET_CUR, validx, idx, cval->val_type); - return ret; + return filter_error(cval, ret); } ucontrol->value.integer.value[0] = val; -- cgit v1.2.3 From 3507245b82b4362dc9721cbc328644905a3efa22 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 12 Apr 2020 10:13:29 +0200 Subject: ALSA: usb-audio: Don't override ignore_ctl_error value from the map The mapping table may contain also ignore_ctl_error flag for devices that are known to behave wild. Since this flag always writes the card's own ignore_ctl_error flag, it overrides the value already set by the module option, so it doesn't follow user's expectation. Let's fix the code not to clear the flag that has been set by user. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Cc: Link: https://lore.kernel.org/r/20200412081331.4742-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index d27e390dcd32..83926b1be53b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3106,7 +3106,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (map->id == state.chip->usb_id) { state.map = map->map; state.selector_map = map->selector_map; - mixer->ignore_ctl_error = map->ignore_ctl_error; + mixer->ignore_ctl_error |= map->ignore_ctl_error; break; } } -- cgit v1.2.3 From 7dc3c5a0172e6c0449502103356c3628d05bc0e0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 12 Apr 2020 10:13:30 +0200 Subject: ALSA: usb-audio: Don't create jack controls for PCM terminals Some funky firmwares set the connector flag even on PCM terminals although it doesn't make sense (and even actually the firmware doesn't react properly!). Let's skip creation of jack controls in such a case. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Cc: Link: https://lore.kernel.org/r/20200412081331.4742-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 83926b1be53b..ab9c908a8771 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2109,7 +2109,8 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid, check_input_term(state, term_id, &iterm); /* Check for jack detection. */ - if (uac_v2v3_control_is_readable(bmctls, control)) + if ((iterm.type & 0xff00) != 0x0100 && + uac_v2v3_control_is_readable(bmctls, control)) build_connector_control(state->mixer, &iterm, true); return 0; @@ -3149,7 +3150,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (err < 0 && err != -EINVAL) return err; - if (uac_v2v3_control_is_readable(le16_to_cpu(desc->bmControls), + if ((state.oterm.type & 0xff00) != 0x0100 && + uac_v2v3_control_is_readable(le16_to_cpu(desc->bmControls), UAC2_TE_CONNECTOR)) { build_connector_control(state.mixer, &state.oterm, false); @@ -3174,7 +3176,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (err < 0 && err != -EINVAL) return err; - if (uac_v2v3_control_is_readable(le32_to_cpu(desc->bmControls), + if ((state.oterm.type & 0xff00) != 0x0100 && + uac_v2v3_control_is_readable(le32_to_cpu(desc->bmControls), UAC3_TE_INSERTION)) { build_connector_control(state.mixer, &state.oterm, false); -- cgit v1.2.3 From 934b96594ed66b07dbc7e576d28814466df3a494 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 12 Apr 2020 10:13:31 +0200 Subject: ALSA: usb-audio: Check mapping at creating connector controls, too Add the mapping check to build_connector_control() so that the device specific quirk can provide the node to skip for the badly behaving connector controls. As an example, ALC1220-VB-based codec implements the skip entry for the broken SPDIF connector detection. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Cc: Link: https://lore.kernel.org/r/20200412081331.4742-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 18 +++++++++++------- sound/usb/mixer_maps.c | 4 +++- 2 files changed, 14 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ab9c908a8771..e7b9040a54e6 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1771,11 +1771,15 @@ static void get_connector_control_name(struct usb_mixer_interface *mixer, /* Build a mixer control for a UAC connector control (jack-detect) */ static void build_connector_control(struct usb_mixer_interface *mixer, + const struct usbmix_name_map *imap, struct usb_audio_term *term, bool is_input) { struct snd_kcontrol *kctl; struct usb_mixer_elem_info *cval; + if (check_ignored_ctl(find_map(imap, term->id, 0))) + return; + cval = kzalloc(sizeof(*cval), GFP_KERNEL); if (!cval) return; @@ -2111,7 +2115,7 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid, /* Check for jack detection. */ if ((iterm.type & 0xff00) != 0x0100 && uac_v2v3_control_is_readable(bmctls, control)) - build_connector_control(state->mixer, &iterm, true); + build_connector_control(state->mixer, state->map, &iterm, true); return 0; } @@ -3072,13 +3076,13 @@ static int snd_usb_mixer_controls_badd(struct usb_mixer_interface *mixer, memset(&iterm, 0, sizeof(iterm)); iterm.id = UAC3_BADD_IT_ID4; iterm.type = UAC_BIDIR_TERMINAL_HEADSET; - build_connector_control(mixer, &iterm, true); + build_connector_control(mixer, map->map, &iterm, true); /* Output Term - Insertion control */ memset(&oterm, 0, sizeof(oterm)); oterm.id = UAC3_BADD_OT_ID3; oterm.type = UAC_BIDIR_TERMINAL_HEADSET; - build_connector_control(mixer, &oterm, false); + build_connector_control(mixer, map->map, &oterm, false); } return 0; @@ -3153,8 +3157,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if ((state.oterm.type & 0xff00) != 0x0100 && uac_v2v3_control_is_readable(le16_to_cpu(desc->bmControls), UAC2_TE_CONNECTOR)) { - build_connector_control(state.mixer, &state.oterm, - false); + build_connector_control(state.mixer, state.map, + &state.oterm, false); } } else { /* UAC_VERSION_3 */ struct uac3_output_terminal_descriptor *desc = p; @@ -3179,8 +3183,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if ((state.oterm.type & 0xff00) != 0x0100 && uac_v2v3_control_is_readable(le32_to_cpu(desc->bmControls), UAC3_TE_INSERTION)) { - build_connector_control(state.mixer, &state.oterm, - false); + build_connector_control(state.mixer, state.map, + &state.oterm, false); } } } diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 72b575c34860..b4e77000f441 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -360,9 +360,11 @@ static const struct usbmix_name_map corsair_virtuoso_map[] = { }; /* Some mobos shipped with a dummy HD-audio show the invalid GET_MIN/GET_MAX - * response for Input Gain Pad (id=19, control=12). Skip it. + * response for Input Gain Pad (id=19, control=12) and the connector status + * for SPDIF terminal (id=18). Skip them. */ static const struct usbmix_name_map asus_rog_map[] = { + { 18, NULL }, /* OT, connector control */ { 19, NULL, 12 }, /* FU, Input Gain Pad */ {} }; -- cgit v1.2.3 From 25faa4bd37c10f19e4b848b9032a17a3d44c6f09 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Apr 2020 10:20:29 +0200 Subject: ALSA: hda: Don't release card at firmware loading error At the error path of the firmware loading error, the driver tries to release the card object and set NULL to drvdata. This may be referred badly at the possible PM action, as the driver itself is still bound and the PM callbacks read the card object. Instead, we continue the probing as if it were no option set. This is often a better choice than the forced abort, too. Fixes: 5cb543dba986 ("ALSA: hda - Deferred probing with request_firmware_nowait()") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bd093593f8fb..a2e811375750 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2027,24 +2027,15 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) { struct snd_card *card = context; struct azx *chip = card->private_data; - struct pci_dev *pci = chip->pci; - - if (!fw) { - dev_err(card->dev, "Cannot load firmware, aborting\n"); - goto error; - } - chip->fw = fw; + if (fw) + chip->fw = fw; + else + dev_err(card->dev, "Cannot load firmware, continue without patching\n"); if (!chip->disabled) { /* continue probing */ - if (azx_probe_continue(chip)) - goto error; + azx_probe_continue(chip); } - return; /* OK */ - - error: - snd_card_free(card); - pci_set_drvdata(pci, NULL); } #endif -- cgit v1.2.3 From 10db5bccc390e8e4bd9fcd1fbd4f1b23f271a405 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Apr 2020 10:20:30 +0200 Subject: ALSA: hda: Honor PM disablement in PM freeze and thaw_noirq ops freeze_noirq and thaw_noirq need to check the PM availability like other PM ops. There are cases where the device got disabled due to the error, and the PM operation should be ignored for that. Fixes: 3e6db33aaf1d ("ALSA: hda - Set SKL+ hda controller power at freeze() and thaw()") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a2e811375750..f41d8b7864c1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1071,6 +1071,8 @@ static int azx_freeze_noirq(struct device *dev) struct azx *chip = card->private_data; struct pci_dev *pci = to_pci_dev(dev); + if (!azx_is_pm_ready(card)) + return 0; if (chip->driver_type == AZX_DRIVER_SKL) pci_set_power_state(pci, PCI_D3hot); @@ -1083,6 +1085,8 @@ static int azx_thaw_noirq(struct device *dev) struct azx *chip = card->private_data; struct pci_dev *pci = to_pci_dev(dev); + if (!azx_is_pm_ready(card)) + return 0; if (chip->driver_type == AZX_DRIVER_SKL) pci_set_power_state(pci, PCI_D0); -- cgit v1.2.3 From 2393e7555b531a534152ffe7bfd1862cacedaacb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Apr 2020 10:20:31 +0200 Subject: ALSA: hda: Release resources at error in delayed probe snd-hda-intel driver handles the most of its probe task in the delayed work (either via workqueue or via firmware loader). When an error happens in the later delayed probe, we can't deregister the device itself because the probe callback already returned success and the device was bound. So, for now, we set hda->init_failed flag and make the rest untouched until the device gets really unbound. However, this leaves the device up running, keeping the resources without any use that prevents other operations. In this patch, we release the resources at first when a probe error happens in the delayed probe stage, but keeps the top-level object, so that the PM and other ops can still refer to the object itself. Also for simplicity, snd_hda_intel object is allocated via devm, so that we can get rid of the explicit kfree calls. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 29 ++++++++++++++++------------- sound/pci/hda/hda_intel.h | 1 + 2 files changed, 17 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f41d8b7864c1..692857904d49 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1203,10 +1203,8 @@ static void azx_vs_set_state(struct pci_dev *pci, if (!disabled) { dev_info(chip->card->dev, "Start delayed initialization\n"); - if (azx_probe_continue(chip) < 0) { + if (azx_probe_continue(chip) < 0) dev_err(chip->card->dev, "initialization error\n"); - hda->init_failed = true; - } } } else { dev_info(chip->card->dev, "%s via vga_switcheroo\n", @@ -1339,12 +1337,15 @@ static int register_vga_switcheroo(struct azx *chip) /* * destructor */ -static int azx_free(struct azx *chip) +static void azx_free(struct azx *chip) { struct pci_dev *pci = chip->pci; struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); + if (hda->freed) + return; + if (azx_has_pm_runtime(chip) && chip->running) pm_runtime_get_noresume(&pci->dev); chip->running = 0; @@ -1388,9 +1389,8 @@ static int azx_free(struct azx *chip) if (chip->driver_caps & AZX_DCAPS_I915_COMPONENT) snd_hdac_i915_exit(bus); - kfree(hda); - return 0; + hda->freed = 1; } static int azx_dev_disconnect(struct snd_device *device) @@ -1406,7 +1406,8 @@ static int azx_dev_disconnect(struct snd_device *device) static int azx_dev_free(struct snd_device *device) { - return azx_free(device->device_data); + azx_free(device->device_data); + return 0; } #ifdef SUPPORT_VGA_SWITCHEROO @@ -1773,7 +1774,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, if (err < 0) return err; - hda = kzalloc(sizeof(*hda), GFP_KERNEL); + hda = devm_kzalloc(&pci->dev, sizeof(*hda), GFP_KERNEL); if (!hda) { pci_disable_device(pci); return -ENOMEM; @@ -1814,7 +1815,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, err = azx_bus_init(chip, model[dev]); if (err < 0) { - kfree(hda); pci_disable_device(pci); return err; } @@ -2340,13 +2340,16 @@ static int azx_probe_continue(struct azx *chip) pm_runtime_put_autosuspend(&pci->dev); out_free: - if (err < 0 || !hda->need_i915_power) + if (err < 0) { + azx_free(chip); + return err; + } + + if (!hda->need_i915_power) display_power(chip, false); - if (err < 0) - hda->init_failed = 1; complete_all(&hda->probe_wait); to_hda_bus(bus)->bus_probing = 0; - return err; + return 0; } static void azx_remove(struct pci_dev *pci) diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 2acfff3da1a0..3fb119f09040 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -27,6 +27,7 @@ struct hda_intel { unsigned int use_vga_switcheroo:1; unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ + unsigned int freed:1; /* resources already released */ bool need_i915_power:1; /* the hda controller needs i915 power */ }; -- cgit v1.2.3 From 9479e75fca370a5220784f7596bf598c4dad0b9b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Apr 2020 10:20:32 +0200 Subject: ALSA: hda: Keep the controller initialization even if no codecs found Currently, when the HD-audio controller driver doesn't detect any codecs, it tries to abort the probe. But this abort happens at the delayed probe, i.e. the primary probe call already returned success, hence the driver is never unbound until user does so explicitly. As a result, it may leave the HD-audio device in the running state without the runtime PM. More badly, if the device is a HD-audio bus that is tied with a GPU, GPU cannot reach to the full power down and consumes unnecessarily much power. This patch changes the logic after no-codec situation; it continues probing without the further codec initialization but keep the controller driver running normally. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Tested-by: Roy Spliet Link: https://lore.kernel.org/r/20200413082034.25166-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 692857904d49..aa0be85614b6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2009,7 +2009,7 @@ static int azx_first_init(struct azx *chip) /* codec detection */ if (!azx_bus(chip)->codec_mask) { dev_err(card->dev, "no codecs found!\n"); - return -ENODEV; + /* keep running the rest for the runtime PM */ } if (azx_acquire_irq(chip, 0) < 0) @@ -2303,9 +2303,11 @@ static int azx_probe_continue(struct azx *chip) #endif /* create codec instances */ - err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); - if (err < 0) - goto out_free; + if (bus->codec_mask) { + err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); + if (err < 0) + goto out_free; + } #ifdef CONFIG_SND_HDA_PATCH_LOADER if (chip->fw) { @@ -2319,7 +2321,7 @@ static int azx_probe_continue(struct azx *chip) #endif } #endif - if ((probe_only[dev] & 1) == 0) { + if (bus->codec_mask && !(probe_only[dev] & 1)) { err = azx_codec_configure(chip); if (err < 0) goto out_free; -- cgit v1.2.3 From c4c8dd6ef807663e42a5f04ea77cd62029eb99fa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Apr 2020 10:20:33 +0200 Subject: ALSA: hda: Skip controller resume if not needed The HD-audio controller does system-suspend and resume operations by directly calling its helpers __azx_runtime_suspend() and __azx_runtime_resume(). However, in general, we don't have to resume always the device fully at the system resume; typically, if a device has been runtime-suspended, we can leave it to runtime resume. Usually for achieving this, the driver would call pm_runtime_force_suspend() and pm_runtime_force_resume() pairs in the system suspend and resume ops. Unfortunately, this doesn't work for the resume path in our case. For handling the jack detection at the system resume, a child codec device may need the (literally) forcibly resume even if it's been runtime-suspended, and for that, the controller device must be also resumed even if it's been suspended. This patch is an attempt to improve the situation. It replaces the direct __azx_runtime_suspend()/_resume() calls with with pm_runtime_force_suspend() and pm_runtime_force_resume() with a slight trick as we've done for the codec side. More exactly: - azx_has_pm_runtime() check is dropped from azx_runtime_suspend() and azx_runtime_resume(), so that it can be properly executed from the system-suspend/resume path - The WAKEEN handling depends on the card's power state now; it's set and cleared only for the runtime-suspend - azx_resume() checks whether any codec may need the forcible resume beforehand. If the forcible resume is required, it does temporary PM refcount up/down for actually triggering the runtime resume. - A new helper function, hda_codec_need_resume(), is introduced for checking whether the codec needs a forcible runtime-resume, and the existing code is rewritten with that. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-6-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/hda_codec.h | 5 +++++ sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_intel.c | 38 +++++++++++++++++++++++++++----------- 3 files changed, 33 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 3ee8036f5436..225154a4f2ed 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -494,6 +494,11 @@ void snd_hda_update_power_acct(struct hda_codec *codec); static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {} #endif +static inline bool hda_codec_need_resume(struct hda_codec *codec) +{ + return !codec->relaxed_resume && codec->jacktbl.used; +} + #ifdef CONFIG_SND_HDA_PATCH_LOADER /* * patch firmware diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a34a2c9f4bcf..86a632bf4d50 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2951,7 +2951,7 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used; + bool forced_resume = hda_codec_need_resume(codec); int ret; /* The get/put pair below enforces the runtime resume even if the diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index aa0be85614b6..02c6308502b1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1027,7 +1027,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - __azx_runtime_suspend(chip); + pm_runtime_force_suspend(dev); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1044,7 +1044,9 @@ static int azx_suspend(struct device *dev) static int azx_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); + struct hda_codec *codec; struct azx *chip; + bool forced_resume = false; if (!azx_is_pm_ready(card)) return 0; @@ -1055,7 +1057,20 @@ static int azx_resume(struct device *dev) chip->msi = 0; if (azx_acquire_irq(chip, 1) < 0) return -EIO; - __azx_runtime_resume(chip, false); + + /* check for the forced resume */ + list_for_each_codec(codec, &chip->bus) { + if (hda_codec_need_resume(codec)) { + forced_resume = true; + break; + } + } + + if (forced_resume) + pm_runtime_get_noresume(dev); + pm_runtime_force_resume(dev); + if (forced_resume) + pm_runtime_put(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); @@ -1102,12 +1117,12 @@ static int azx_runtime_suspend(struct device *dev) if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - if (!azx_has_pm_runtime(chip)) - return 0; /* enable controller wake up event */ - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | - STATESTS_INT_MASK); + if (snd_power_get_state(card) == SNDRV_CTL_POWER_D0) { + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + } __azx_runtime_suspend(chip); trace_azx_runtime_suspend(chip); @@ -1118,17 +1133,18 @@ static int azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; + bool from_rt = snd_power_get_state(card) == SNDRV_CTL_POWER_D0; if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - if (!azx_has_pm_runtime(chip)) - return 0; - __azx_runtime_resume(chip, true); + __azx_runtime_resume(chip, from_rt); /* disable controller Wake Up event*/ - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & - ~STATESTS_INT_MASK); + if (from_rt) { + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & + ~STATESTS_INT_MASK); + } trace_azx_runtime_resume(chip); return 0; -- cgit v1.2.3 From 3ba21113bd33d49f3c300a23fc08cf114c434995 Mon Sep 17 00:00:00 2001 From: Roy Spliet Date: Mon, 13 Apr 2020 10:20:34 +0200 Subject: ALSA: hda: Explicitly permit using autosuspend if runtime PM is supported This fixes runtime PM not working after a suspend-to-RAM cycle at least for the codec-less HDA device found on NVIDIA GPUs. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Signed-off-by: Roy Spliet Link: https://lore.kernel.org/r/20200413082034.25166-7-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 02c6308502b1..8519051a426e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2354,8 +2354,10 @@ static int azx_probe_continue(struct azx *chip) set_default_power_save(chip); - if (azx_has_pm_runtime(chip)) + if (azx_has_pm_runtime(chip)) { + pm_runtime_use_autosuspend(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); + } out_free: if (err < 0) { -- cgit v1.2.3 From f8e4ae10de43fbb7ce85f79e04eca2988b6b2c40 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Apr 2020 22:19:19 +0200 Subject: ALSA: hda: Allow setting preallocation again for x86 The commit c31427d0d21e ("ALSA: hda: No preallocation on x86 platforms") changed CONFIG_SND_HDA_PREALLOC_SIZE setup and its default to zero for x86, as the preallocation should work almost all cases. However, this expectation was too naive; some applications try to allocate as the max buffer size as possible, and it leads to the memory exhaustion. More badly, the commit changed the kconfig no longer adjustable for x86, so you can't fix it statically (although it can be still adjusted via procfs). So, practically seen, it's more recommended to set a reasonable limit for x86, too. This patch follows to that experience, and changes the default to 2048 and allow the kconfig adjustable again. Fixes: c31427d0d21e ("ALSA: hda: No preallocation on x86 platforms") Cc: BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207223 Link: https://lore.kernel.org/r/20200413201919.24241-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/hda/Kconfig | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 4ca6b09056f3..3bc9224d5e4f 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -21,16 +21,17 @@ config SND_HDA_EXT_CORE select SND_HDA_CORE config SND_HDA_PREALLOC_SIZE - int "Pre-allocated buffer size for HD-audio driver" if !SND_DMA_SGBUF + int "Pre-allocated buffer size for HD-audio driver" range 0 32768 - default 0 if SND_DMA_SGBUF + default 2048 if SND_DMA_SGBUF default 64 if !SND_DMA_SGBUF help Specifies the default pre-allocated buffer-size in kB for the HD-audio driver. A larger buffer (e.g. 2048) is preferred for systems using PulseAudio. The default 64 is chosen just for compatibility reasons. - On x86 systems, the default is zero as we need no preallocation. + On x86 systems, the default is 2048 as a reasonable value for + most of modern systems. Note that the pre-allocation size can be changed dynamically via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. -- cgit v1.2.3 From ec21bdc6dd16d74b3674ef1fd12ae8e4e7418603 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 13 Apr 2020 14:45:48 +0200 Subject: ASoC: samsung: s3c24xx-i2s: Fix build after removal of DAI suspend/resume MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Commit 450312b640f9 ("ASoC: soc-core: remove DAI suspend/resume") removed the DAI side suspend/resume hooks and switched entirely to component suspend/resume. However the Samsung SoC s3c-i2s-v2 driver was not updated. Move the suspend/resume hooks from s3c-i2s-v2.c to s3c2412-i2s.c while changing dai to component which allows to keep the struct snd_soc_component_driver const. This fixes build errors: sound/soc/samsung/s3c-i2s-v2.c: In function ‘s3c_i2sv2_register_component’: sound/soc/samsung/s3c-i2s-v2.c:730:9: error: ‘struct snd_soc_dai_driver’ has no member named ‘suspend’ dai_drv->suspend = s3c2412_i2s_suspend; Reported-by: Arnd Bergmann Fixes: 450312b640f9 ("ASoC: soc-core: remove DAI suspend/resume") Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20200413124548.28197-1-krzk@kernel.org Signed-off-by: Mark Brown --- sound/soc/samsung/s3c-i2s-v2.c | 57 ----------------------------------------- sound/soc/samsung/s3c2412-i2s.c | 56 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 56 insertions(+), 57 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 358887848293..5e95c30fb2ba 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -656,60 +656,6 @@ void s3c_i2sv2_cleanup(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(s3c_i2sv2_cleanup); -#ifdef CONFIG_PM -static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - u32 iismod; - - if (dai->active) { - i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); - i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); - i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); - - /* some basic suspend checks */ - - iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) - pr_warn("%s: RXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) - pr_warn("%s: TXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_IIS_ACTIVE) - pr_warn("%s: IIS active\n", __func__); - } - - return 0; -} - -static int s3c2412_i2s_resume(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - - pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", - dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); - - if (dai->active) { - writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); - writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); - writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); - - writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - ndelay(250); - writel(0x0, i2s->regs + S3C2412_IISFIC); - } - - return 0; -} -#else -#define s3c2412_i2s_suspend NULL -#define s3c2412_i2s_resume NULL -#endif - int s3c_i2sv2_register_component(struct device *dev, int id, const struct snd_soc_component_driver *cmp_drv, struct snd_soc_dai_driver *dai_drv) @@ -727,9 +673,6 @@ int s3c_i2sv2_register_component(struct device *dev, int id, if (!ops->delay) ops->delay = s3c2412_i2s_delay; - dai_drv->suspend = s3c2412_i2s_suspend; - dai_drv->resume = s3c2412_i2s_resume; - return devm_snd_soc_register_component(dev, cmp_drv, dai_drv, 1); } EXPORT_SYMBOL_GPL(s3c_i2sv2_register_component); diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 787a3f6e9f24..b35d828c1cfe 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -117,6 +117,60 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } +#ifdef CONFIG_PM +static int s3c2412_i2s_suspend(struct snd_soc_component *component) +{ + struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component); + u32 iismod; + + if (component->active) { + i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); + i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); + i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); + + /* some basic suspend checks */ + + iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) + pr_warn("%s: RXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) + pr_warn("%s: TXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_IIS_ACTIVE) + pr_warn("%s: IIS active\n", __func__); + } + + return 0; +} + +static int s3c2412_i2s_resume(struct snd_soc_component *component) +{ + struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component); + + pr_info("component_active %d, IISMOD %08x, IISCON %08x\n", + component->active, i2s->suspend_iismod, i2s->suspend_iiscon); + + if (component->active) { + writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); + writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); + writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); + + writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + ndelay(250); + writel(0x0, i2s->regs + S3C2412_IISFIC); + } + + return 0; +} +#else +#define s3c2412_i2s_suspend NULL +#define s3c2412_i2s_resume NULL +#endif + #define S3C2412_I2S_RATES \ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ @@ -146,6 +200,8 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = { static const struct snd_soc_component_driver s3c2412_i2s_component = { .name = "s3c2412-i2s", + .suspend = s3c2412_i2s_suspend, + .resume = s3c2412_i2s_resume, }; static int s3c2412_iis_dev_probe(struct platform_device *pdev) -- cgit v1.2.3 From 595571cca4dec8ac48122a6d2733f790c9a2cade Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 9 Apr 2020 19:12:07 +0100 Subject: ASoC: dapm: Fix regression introducing multiple copies of DAI widgets Refactoring was done to factor out the linking of DAI widgets into a helper function, dapm_add_valid_dai_widget. However when this was done, a regression was introduced for CODEC to CODEC links. It was over looked that the playback and capture variables persisted across all CODEC DAIs being processed, which ensured that the special DAI widget that is added for CODEC to CODEC links was only created once. This bug causes kernel panics during DAPM shutdown. To stick with the spirit of the original refactoring whilst fixing the issue, variables to hold the DAI widgets are added to snd_soc_dai_link. Furthermore the dapm_add_valid_dai_widget function is renamed to dapm_connect_dai_pair, the function only adds DAI widgets in the CODEC to CODEC case and its primary job is to add routes connecting two DAI widgets, making the original name quite misleading. Fixes: 6c4b13b51aa3 ("ASoC: Add dapm_add_valid_dai_widget helper") Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20200409181209.30130-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/soc.h | 3 ++ sound/soc/soc-dapm.c | 91 +++++++++++++++++++++++++++------------------------- 2 files changed, 51 insertions(+), 43 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 13458e4fbb13..946f88a6c63d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -790,6 +790,9 @@ struct snd_soc_dai_link { const struct snd_soc_pcm_stream *params; unsigned int num_params; + struct snd_soc_dapm_widget *playback_widget; + struct snd_soc_dapm_widget *capture_widget; + unsigned int dai_fmt; /* format to set on init */ enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 679ed60d850e..fe907f0cc709 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4283,52 +4283,63 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -static void dapm_add_valid_dai_widget(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd, - struct snd_soc_dai *codec_dai, - struct snd_soc_dai *cpu_dai) +static void dapm_connect_dai_routes(struct snd_soc_dapm_context *dapm, + struct snd_soc_dai *src_dai, + struct snd_soc_dapm_widget *src, + struct snd_soc_dapm_widget *dai, + struct snd_soc_dai *sink_dai, + struct snd_soc_dapm_widget *sink) { - struct snd_soc_dapm_widget *playback = NULL, *capture = NULL; - struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu; + dev_dbg(dapm->dev, "connected DAI link %s:%s -> %s:%s\n", + src_dai->component->name, src->name, + sink_dai->component->name, sink->name); + + if (dai) { + snd_soc_dapm_add_path(dapm, src, dai, NULL, NULL); + src = dai; + } + + snd_soc_dapm_add_path(dapm, src, sink, NULL, NULL); +} + +static void dapm_connect_dai_pair(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_dai_link *dai_link = rtd->dai_link; + struct snd_soc_dapm_widget *dai, *codec, *playback_cpu, *capture_cpu; struct snd_pcm_substream *substream; struct snd_pcm_str *streams = rtd->pcm->streams; - if (rtd->dai_link->params) { + if (dai_link->params) { playback_cpu = cpu_dai->capture_widget; capture_cpu = cpu_dai->playback_widget; } else { - playback = cpu_dai->playback_widget; - capture = cpu_dai->capture_widget; - playback_cpu = playback; - capture_cpu = capture; + playback_cpu = cpu_dai->playback_widget; + capture_cpu = cpu_dai->capture_widget; } /* connect BE DAI playback if widgets are valid */ codec = codec_dai->playback_widget; if (playback_cpu && codec) { - if (!playback) { + if (dai_link->params && !dai_link->playback_widget) { substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - playback = snd_soc_dapm_new_dai(card, substream, - "playback"); - if (IS_ERR(playback)) { + dai = snd_soc_dapm_new_dai(card, substream, "playback"); + if (IS_ERR(dai)) { dev_err(rtd->dev, "ASoC: Failed to create DAI %s: %ld\n", codec_dai->name, - PTR_ERR(playback)); + PTR_ERR(dai)); goto capture; } - - snd_soc_dapm_add_path(&card->dapm, playback_cpu, - playback, NULL, NULL); + dai_link->playback_widget = dai; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - cpu_dai->component->name, playback_cpu->name, - codec_dai->component->name, codec->name); - - snd_soc_dapm_add_path(&card->dapm, playback, codec, - NULL, NULL); + dapm_connect_dai_routes(&card->dapm, cpu_dai, playback_cpu, + dai_link->playback_widget, + codec_dai, codec); } capture: @@ -4336,28 +4347,22 @@ capture: codec = codec_dai->capture_widget; if (codec && capture_cpu) { - if (!capture) { + if (dai_link->params && !dai_link->capture_widget) { substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; - capture = snd_soc_dapm_new_dai(card, substream, - "capture"); - if (IS_ERR(capture)) { + dai = snd_soc_dapm_new_dai(card, substream, "capture"); + if (IS_ERR(dai)) { dev_err(rtd->dev, "ASoC: Failed to create DAI %s: %ld\n", codec_dai->name, - PTR_ERR(capture)); + PTR_ERR(dai)); return; } - - snd_soc_dapm_add_path(&card->dapm, capture, - capture_cpu, NULL, NULL); + dai_link->capture_widget = dai; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - codec_dai->component->name, codec->name, - cpu_dai->component->name, capture_cpu->name); - - snd_soc_dapm_add_path(&card->dapm, codec, capture, - NULL, NULL); + dapm_connect_dai_routes(&card->dapm, codec_dai, codec, + dai_link->capture_widget, + cpu_dai, capture_cpu); } } @@ -4369,12 +4374,12 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, if (rtd->num_cpus == 1) { for_each_rtd_codec_dais(rtd, i, codec_dai) - dapm_add_valid_dai_widget(card, rtd, codec_dai, - rtd->cpu_dais[0]); + dapm_connect_dai_pair(card, rtd, codec_dai, + rtd->cpu_dais[0]); } else if (rtd->num_codecs == rtd->num_cpus) { for_each_rtd_codec_dais(rtd, i, codec_dai) - dapm_add_valid_dai_widget(card, rtd, codec_dai, - rtd->cpu_dais[i]); + dapm_connect_dai_pair(card, rtd, codec_dai, + rtd->cpu_dais[i]); } else { dev_err(card->dev, "N cpus to M codecs link is not supported yet\n"); -- cgit v1.2.3 From 9df8ba7c63073508e5aa677dade48fcab6a6773e Mon Sep 17 00:00:00 2001 From: Philipp Puschmann Date: Tue, 14 Apr 2020 13:27:54 +0200 Subject: ASoC: tas571x: disable regulators on failed probe If probe fails after enabling the regulators regulator_put is called for each supply without having them disabled before. This produces some warnings like WARNING: CPU: 0 PID: 90 at drivers/regulator/core.c:2044 _regulator_put.part.0+0x154/0x15c [] (unwind_backtrace) from [] (show_stack+0x10/0x14) [] (show_stack) from [] (__warn+0xd0/0xf4) [] (__warn) from [] (warn_slowpath_fmt+0x64/0xc4) [] (warn_slowpath_fmt) from [] (_regulator_put.part.0+0x154/0x15c) [] (_regulator_put.part.0) from [] (regulator_put+0x28/0x38) [] (regulator_put) from [] (regulator_bulk_free+0x28/0x38) [] (regulator_bulk_free) from [] (release_nodes+0x1d0/0x22c) [] (release_nodes) from [] (really_probe+0x108/0x34c) [] (really_probe) from [] (driver_probe_device+0xb8/0x16c) [] (driver_probe_device) from [] (device_driver_attach+0x58/0x60) [] (device_driver_attach) from [] (__driver_attach+0x58/0xcc) [] (__driver_attach) from [] (bus_for_each_dev+0x78/0xc0) [] (bus_for_each_dev) from [] (bus_add_driver+0x188/0x1e0) [] (bus_add_driver) from [] (driver_register+0x74/0x108) [] (driver_register) from [] (i2c_register_driver+0x3c/0x88) [] (i2c_register_driver) from [] (do_one_initcall+0x58/0x250) [] (do_one_initcall) from [] (do_init_module+0x60/0x244) [] (do_init_module) from [] (load_module+0x2180/0x2540) [] (load_module) from [] (sys_finit_module+0xd0/0xe8) [] (sys_finit_module) from [] (__sys_trace_return+0x0/0x20) Fixes: 3fd6e7d9a146 (ASoC: tas571x: New driver for TI TAS571x power amplifiers) Signed-off-by: Philipp Puschmann Link: https://lore.kernel.org/r/20200414112754.3365406-1-p.puschmann@pironex.de Signed-off-by: Mark Brown --- sound/soc/codecs/tas571x.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 1554631cb397..5b7f9fcf6cbf 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -820,8 +820,10 @@ static int tas571x_i2c_probe(struct i2c_client *client, priv->regmap = devm_regmap_init(dev, NULL, client, priv->chip->regmap_config); - if (IS_ERR(priv->regmap)) - return PTR_ERR(priv->regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + goto disable_regs; + } priv->pdn_gpio = devm_gpiod_get_optional(dev, "pdn", GPIOD_OUT_LOW); if (IS_ERR(priv->pdn_gpio)) { @@ -845,7 +847,7 @@ static int tas571x_i2c_probe(struct i2c_client *client, ret = regmap_write(priv->regmap, TAS571X_OSC_TRIM_REG, 0); if (ret) - return ret; + goto disable_regs; usleep_range(50000, 60000); @@ -861,12 +863,20 @@ static int tas571x_i2c_probe(struct i2c_client *client, */ ret = regmap_update_bits(priv->regmap, TAS571X_MVOL_REG, 1, 0); if (ret) - return ret; + goto disable_regs; } - return devm_snd_soc_register_component(&client->dev, + ret = devm_snd_soc_register_component(&client->dev, &priv->component_driver, &tas571x_dai, 1); + if (ret) + goto disable_regs; + + return ret; + +disable_regs: + regulator_bulk_disable(priv->chip->num_supply_names, priv->supplies); + return ret; } static int tas571x_i2c_remove(struct i2c_client *client) -- cgit v1.2.3 From 8dbcfcfc73d43df5a3dc306b6a4c1d996caf37e0 Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Tue, 14 Apr 2020 05:35:23 -0600 Subject: ASoC: amd: Fix button configuration RT5682 buttons were incorrectly mapped. Signed-off-by: Akshu Agrawal Link: https://lore.kernel.org/r/20200414113527.13532-1-akshu.agrawal@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 024a7ee54cd5..e499c00e0c66 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -89,9 +89,9 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) } snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); - snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); - snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); ret = snd_soc_component_set_jack(component, &pco_jack, NULL); if (ret) { -- cgit v1.2.3 From acda42b30fa6b67f07b4560577418df5ada77b52 Mon Sep 17 00:00:00 2001 From: Jason Yan Date: Fri, 10 Apr 2020 16:11:16 +0800 Subject: ASoC: intel: soc-acpi-intel-icl-match: remove useless 'rt1308_2_adr' MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix the following gcc warning: sound/soc/intel/common/soc-acpi-intel-icl-match.c:90:45: warning: ‘rt1308_2_adr’ defined but not used [-Wunused-const-variable=] static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { ^~~~~~~~~~~~ Reported-by: Hulk Robot Signed-off-by: Jason Yan Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200410081117.21319-1-yanaijie@huawei.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index ef8500349f2f..16ec9f382b0f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -87,14 +87,6 @@ static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = { } }; -static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { - { - .adr = 0x000210025D130800, - .num_endpoints = 1, - .endpoints = &single_endpoint, - } -}; - static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = { { .adr = 0x000110025D130800, -- cgit v1.2.3 From a306f04511148fade6bead59920dde864a54f017 Mon Sep 17 00:00:00 2001 From: Jason Yan Date: Fri, 10 Apr 2020 16:11:17 +0800 Subject: ASoC: Intel: soc-acpi-intel-cml-match: remove useless 'rt1308_2_adr' MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix the following gcc warning: sound/soc/intel/common/soc-acpi-intel-cml-match.c:116:45: warning: ‘rt1308_2_adr’ defined but not used [-Wunused-const-variable=] static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { ^~~~~~~~~~~~ Reported-by: Hulk Robot Signed-off-by: Jason Yan Link: https://lore.kernel.org/r/20200410081117.21319-2-yanaijie@huawei.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index bcedec6c6117..7d85bd5aff9f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -113,14 +113,6 @@ static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = { } }; -static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { - { - .adr = 0x000210025D130800, - .num_endpoints = 1, - .endpoints = &single_endpoint, - } -}; - static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = { { .adr = 0x000110025D130800, -- cgit v1.2.3 From 9a1bb600ecafcb07391c83f7c31057abdd3757b7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 9 Apr 2020 19:12:08 +0100 Subject: ASoC: dapm: Move error message to avoid some duplication Move the error message into snd_soc_dapm_new_dai from dapm_connect_dai_pair, since the two copies are almost identical and are the only callers. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20200409181209.30130-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fe907f0cc709..33168980619b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4165,6 +4165,8 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); if (IS_ERR(w)) { ret = PTR_ERR(w); + dev_err(rtd->dev, "ASoC: Failed to create %s widget: %d\n", + link_name, ret); goto outfree_kcontrol_news; } @@ -4327,13 +4329,8 @@ static void dapm_connect_dai_pair(struct snd_soc_card *card, if (dai_link->params && !dai_link->playback_widget) { substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; dai = snd_soc_dapm_new_dai(card, substream, "playback"); - if (IS_ERR(dai)) { - dev_err(rtd->dev, - "ASoC: Failed to create DAI %s: %ld\n", - codec_dai->name, - PTR_ERR(dai)); + if (IS_ERR(dai)) goto capture; - } dai_link->playback_widget = dai; } @@ -4350,13 +4347,8 @@ capture: if (dai_link->params && !dai_link->capture_widget) { substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; dai = snd_soc_dapm_new_dai(card, substream, "capture"); - if (IS_ERR(dai)) { - dev_err(rtd->dev, - "ASoC: Failed to create DAI %s: %ld\n", - codec_dai->name, - PTR_ERR(dai)); + if (IS_ERR(dai)) return; - } dai_link->capture_widget = dai; } -- cgit v1.2.3 From 0eaef95e3cef1f849e8873f929bef3039409c4fc Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 9 Apr 2020 19:12:09 +0100 Subject: ASoC: dapm: Remove dapm_connect_dai_link_widgets helper This helper is adding very little both it and is one caller are very small functions simply combine the two. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20200409181209.30130-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 36 ++++++++++++++---------------------- 1 file changed, 14 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 33168980619b..a4143ca190d0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4358,27 +4358,6 @@ capture: } } -static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *codec_dai; - int i; - - if (rtd->num_cpus == 1) { - for_each_rtd_codec_dais(rtd, i, codec_dai) - dapm_connect_dai_pair(card, rtd, codec_dai, - rtd->cpu_dais[0]); - } else if (rtd->num_codecs == rtd->num_cpus) { - for_each_rtd_codec_dais(rtd, i, codec_dai) - dapm_connect_dai_pair(card, rtd, codec_dai, - rtd->cpu_dais[i]); - } else { - dev_err(card->dev, - "N cpus to M codecs link is not supported yet\n"); - } - -} - static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, int event) { @@ -4419,6 +4398,8 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + int i; /* for each BE DAI link... */ for_each_card_rtds(card, rtd) { @@ -4429,7 +4410,18 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) if (rtd->dai_link->dynamic) continue; - dapm_connect_dai_link_widgets(card, rtd); + if (rtd->num_cpus == 1) { + for_each_rtd_codec_dais(rtd, i, codec_dai) + dapm_connect_dai_pair(card, rtd, codec_dai, + rtd->cpu_dais[0]); + } else if (rtd->num_codecs == rtd->num_cpus) { + for_each_rtd_codec_dais(rtd, i, codec_dai) + dapm_connect_dai_pair(card, rtd, codec_dai, + rtd->cpu_dais[i]); + } else { + dev_err(card->dev, + "N cpus to M codecs link is not supported yet\n"); + } } } -- cgit v1.2.3 From 9de300abb71f24b190362ff53907ab90505517bc Mon Sep 17 00:00:00 2001 From: Sebastian Fricke Date: Mon, 13 Apr 2020 06:29:52 +0200 Subject: soc/stm/stm32_sub_sai: Add missing '\n' in log messages Message logged by 'dev_xxx()' or 'pr_xxx()' should end with a '\n'. Fixes: 3e086ed("ASoC: stm32: add SAI drivers") Signed-off-by: Sebastian Fricke Link: https://lore.kernel.org/r/20200413042952.7675-1-sebastian.fricke.linux@gmail.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 0d0c9afd8791..34a7c3d6fb91 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -837,7 +837,7 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, cr1 = SAI_XCR1_DS_SET(SAI_DATASIZE_32); break; default: - dev_err(cpu_dai->dev, "Data format not supported"); + dev_err(cpu_dai->dev, "Data format not supported\n"); return -EINVAL; } -- cgit v1.2.3 From d0c56b307f37fd21e2424d3c210e5d85831dd132 Mon Sep 17 00:00:00 2001 From: Bjorn Andersson Date: Sun, 5 Apr 2020 17:32:29 -0700 Subject: ASoC: qcom: common: Silence duplicate parse error messages All error paths in qcom_snd_parse_of() prints more specific error messages, so silence the one in apq8096_platform_probe() and sdm845_snd_platform_probe() to avoid spamming the kernel log. Signed-off-by: Bjorn Andersson Link: https://lore.kernel.org/r/20200406003229.2354631-1-bjorn.andersson@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/apq8096.c | 4 +--- sound/soc/qcom/sdm845.c | 4 +--- 2 files changed, 2 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index d55e3ad96716..287ad2aa27f3 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -116,10 +116,8 @@ static int apq8096_platform_probe(struct platform_device *pdev) card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); - if (ret) { - dev_err(dev, "Error parsing OF data\n"); + if (ret) goto err; - } apq8096_add_be_ops(card); ret = snd_soc_register_card(card); diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index b2de65c7f95c..68e9388ff46f 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -559,10 +559,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); - if (ret) { - dev_err(dev, "Error parsing OF data\n"); + if (ret) goto parse_dt_fail; - } data->card = card; snd_soc_card_set_drvdata(card, data); -- cgit v1.2.3 From 0f2a3b02274c02eb97697c4d89c019d1d21ac225 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 14 Apr 2020 12:03:47 +0100 Subject: ASoC: wsa881x: mark read_only_wordlength flag WSA881x works in PDM mode so the wordlength is fixed, which also makes the only field "WordLength" in DPN_BlockCtrl1 register a read-only. Writing to this register will throw up errors with Qualcomm Controller. So use ro_blockctrl1_reg flag to mark this field as read-only so that core will not write to this register. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Link: https://lore.kernel.org/r/20200414110347.23829-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa881x.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index f2d6f2f81f14..d39d479e2378 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -394,6 +394,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* COMP */ .num = 2, @@ -401,6 +402,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* BOOST */ .num = 3, @@ -408,6 +410,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* VISENSE */ .num = 4, @@ -415,6 +418,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, } }; -- cgit v1.2.3 From bcbc13d28f7f7bcacb3594f72e68c8e57167a836 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 9 Apr 2020 19:13:11 +0100 Subject: ASoC: madera: Remove a couple of stray blank lines Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20200409181311.30247-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/madera.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index 40de9d7811d1..a448d2a2918a 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -1903,7 +1903,6 @@ const struct soc_enum madera_isrc_fsh[] = { MADERA_ISRC4_FSH_SHIFT, 0xf, MADERA_RATE_ENUM_SIZE, madera_rate_text, madera_rate_val), - }; EXPORT_SYMBOL_GPL(madera_isrc_fsh); @@ -1924,7 +1923,6 @@ const struct soc_enum madera_isrc_fsl[] = { MADERA_ISRC4_FSL_SHIFT, 0xf, MADERA_RATE_ENUM_SIZE, madera_rate_text, madera_rate_val), - }; EXPORT_SYMBOL_GPL(madera_isrc_fsl); @@ -1938,7 +1936,6 @@ const struct soc_enum madera_asrc1_rate[] = { MADERA_ASYNC_RATE_ENUM_SIZE, madera_rate_text + MADERA_SYNC_RATE_ENUM_SIZE, madera_rate_val + MADERA_SYNC_RATE_ENUM_SIZE), - }; EXPORT_SYMBOL_GPL(madera_asrc1_rate); @@ -1964,7 +1961,6 @@ const struct soc_enum madera_asrc2_rate[] = { MADERA_ASYNC_RATE_ENUM_SIZE, madera_rate_text + MADERA_SYNC_RATE_ENUM_SIZE, madera_rate_val + MADERA_SYNC_RATE_ENUM_SIZE), - }; EXPORT_SYMBOL_GPL(madera_asrc2_rate); -- cgit v1.2.3 From 9a6418487b566503c772cb6e7d3d44e652b019b0 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Tue, 14 Apr 2020 22:27:25 +0800 Subject: ALSA: hda: call runtime_allow() for all hda controllers Before the pci_driver->probe() is called, the pci subsystem calls runtime_forbid() and runtime_get_sync() on this pci dev, so only call runtime_put_autosuspend() is not enough to enable the runtime_pm on this device. For controllers with vgaswitcheroo feature, the pci/quirks.c will call runtime_allow() for this dev, then the controllers could enter rt_idle/suspend/resume, but for non-vgaswitcheroo controllers like Intel hda controllers, the runtime_pm is not enabled because the runtime_allow() is not called. Since it is no harm calling runtime_allow() twice, here let hda driver call runtime_allow() for all controllers. Then the runtime_pm is enabled on all controllers after the put_autosuspend() is called. Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20200414142725.6020-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8519051a426e..a5fab12defde 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2356,6 +2356,7 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); + pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- cgit v1.2.3 From aa7812737f2877e192d57626cbe8825cc7cf6de9 Mon Sep 17 00:00:00 2001 From: Sebastian Reichel Date: Tue, 14 Apr 2020 20:11:40 +0200 Subject: ASoC: sgtl5000: Fix VAG power-on handling As mentioned slightly out of patch context in the code, there is no reset routine for the chip. On boards where the chip is supplied by a fixed regulator, it might not even be resetted during (e.g. watchdog) reboot and can be in any state. If the device is probed with VAG enabled, the driver's probe routine will generate a loud pop sound when ANA_POWER is being programmed. Avoid this by properly disabling just the VAG bit and waiting the required power down time. Signed-off-by: Sebastian Reichel Reviewed-by: Fabio Estevam Link: https://lore.kernel.org/r/20200414181140.145825-1-sebastian.reichel@collabora.com Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 34 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/sgtl5000.h | 1 + 2 files changed, 35 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d5130193b4a2..e8a8bf7b4ffe 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1653,6 +1653,40 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, dev_err(&client->dev, "Error %d initializing CHIP_CLK_CTRL\n", ret); + /* Mute everything to avoid pop from the following power-up */ + ret = regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_CTRL, + SGTL5000_CHIP_ANA_CTRL_DEFAULT); + if (ret) { + dev_err(&client->dev, + "Error %d muting outputs via CHIP_ANA_CTRL\n", ret); + goto disable_clk; + } + + /* + * If VAG is powered-on (e.g. from previous boot), it would be disabled + * by the write to ANA_POWER in later steps of the probe code. This + * may create a loud pop even with all outputs muted. The proper way + * to circumvent this is disabling the bit first and waiting the proper + * cool-down time. + */ + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, &value); + if (ret) { + dev_err(&client->dev, "Failed to read ANA_POWER: %d\n", ret); + goto disable_clk; + } + if (value & SGTL5000_VAG_POWERUP) { + ret = regmap_update_bits(sgtl5000->regmap, + SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, + 0); + if (ret) { + dev_err(&client->dev, "Error %d disabling VAG\n", ret); + goto disable_clk; + } + + msleep(SGTL5000_VAG_POWERDOWN_DELAY); + } + /* Follow section 2.2.1.1 of AN3663 */ ana_pwr = SGTL5000_ANA_POWER_DEFAULT; if (sgtl5000->num_supplies <= VDDD) { diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index a4bf4bca95bf..56ec5863f250 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -233,6 +233,7 @@ /* * SGTL5000_CHIP_ANA_CTRL */ +#define SGTL5000_CHIP_ANA_CTRL_DEFAULT 0x0133 #define SGTL5000_LINE_OUT_MUTE 0x0100 #define SGTL5000_HP_SEL_MASK 0x0040 #define SGTL5000_HP_SEL_SHIFT 6 -- cgit v1.2.3 From 9b5db059366ae2087e07892b5fc108f81f4ec189 Mon Sep 17 00:00:00 2001 From: Stephan Gerhold Date: Wed, 15 Apr 2020 12:49:28 +0200 Subject: ASoC: soc-pcm: dpcm: Only allow playback/capture if supported At the moment, PCM devices for DPCM are only created based on the dpcm_playback/capture parameters of the DAI link, without considering if the CPU/FE DAI is actually capable of playback/capture. Normally the dpcm_playback/capture parameter should match the capabilities of the CPU DAI. However, there is no way to set that parameter from the device tree (e.g. with simple-audio-card or qcom sound cards). dpcm_playback/capture are always both set to 1. This causes problems when the CPU DAI does only support playback or capture. Attemting to open that PCM device with an unsupported stream type then results in a null pointer dereference: Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 Internal error: Oops: 96000044 [#1] PREEMPT SMP CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 pc : invalidate_paths_ep+0x30/0xe0 lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 Call trace: invalidate_paths_ep+0x30/0xe0 snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 dpcm_path_get+0x38/0xd0 dpcm_fe_dai_open+0x70/0x920 snd_pcm_open_substream+0x564/0x840 snd_pcm_open+0xfc/0x228 snd_pcm_capture_open+0x4c/0x78 snd_open+0xac/0x1a8 ... ... because the DAI playback/capture_widget is not set in that case. We could add checks there to fix the problem (maybe we should anyway), but much easier is to not expose the device as playback/capture in the first place. Attemting to use that device would always fail later anyway. Add checks for snd_soc_dai_stream_valid() to the DPCM case to avoid exposing playback/capture if it is not supported. Signed-off-by: Stephan Gerhold Link: https://lore.kernel.org/r/20200415104928.86091-1-stephan@gerhold.net Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 289aebc15529..1f302de44052 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2911,8 +2911,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) int i; if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { - playback = rtd->dai_link->dpcm_playback; - capture = rtd->dai_link->dpcm_capture; + cpu_dai = asoc_rtd_to_cpu(rtd, 0); + if (rtd->num_cpus > 1) { + dev_err(rtd->dev, + "DPCM doesn't support Multi CPU yet\n"); + return -EINVAL; + } + + playback = rtd->dai_link->dpcm_playback && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK); + capture = rtd->dai_link->dpcm_capture && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE); } else { /* Adapt stream for codec2codec links */ int cpu_capture = rtd->dai_link->params ? -- cgit v1.2.3 From 0c824ec094b5cda766c80d88c2036e28c24a4cb1 Mon Sep 17 00:00:00 2001 From: Stephan Gerhold Date: Wed, 15 Apr 2020 17:00:50 +0200 Subject: ASoC: q6dsp6: q6afe-dai: add missing channels to MI2S DAIs For some reason, the MI2S DAIs do not have channels_min/max defined. This means that snd_soc_dai_stream_valid() returns false, i.e. the DAIs have neither valid playback nor capture stream. It's quite surprising that this ever worked correctly, but in 5.7-rc1 this is now failing badly: :) Commit 0e9cf4c452ad ("ASoC: pcm: check if cpu-dai supports a given stream") introduced a check for snd_soc_dai_stream_valid() before calling hw_params(), which means that the q6i2s_hw_params() function was never called, eventually resulting in: qcom-q6afe aprsvc:q6afe:4:4: no line is assigned ... even though "qcom,sd-lines" is set in the device tree. Commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") now even avoids creating PCM devices if the stream is not supported, which means that it is failing even earlier with e.g.: Primary MI2S: ASoC: no backend playback stream Avoid all that trouble by adding channels_min/max for the MI2S DAIs. Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe dai driver") Signed-off-by: Stephan Gerhold Reviewed-by: Srinivas Kandagatla Cc: Srinivas Kandagatla Link: https://lore.kernel.org/r/20200415150050.616392-1-stephan@gerhold.net Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index c1a7624eaf17..2a5302f1db98 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -902,6 +902,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -917,6 +919,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -931,6 +935,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -946,6 +952,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -960,6 +968,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -975,6 +985,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -989,6 +1001,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -1004,6 +1018,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, -- cgit v1.2.3 From 26d87881590fd55ccdd8f829498d7b3033f81990 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Wed, 15 Apr 2020 12:24:35 -0400 Subject: ASoC: topology: Fix endianness issue MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit As done in already existing cases, we should use le32_to_cpu macro while accessing hdr->magic. Found with sparse. Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20200415162435.31859-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 33e8d189ba2f..6df3b0d12d87 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2652,7 +2652,7 @@ static int soc_valid_header(struct soc_tplg *tplg, } /* big endian firmware objects not supported atm */ - if (hdr->magic == SOC_TPLG_MAGIC_BIG_ENDIAN) { + if (le32_to_cpu(hdr->magic) == SOC_TPLG_MAGIC_BIG_ENDIAN) { dev_err(tplg->dev, "ASoC: pass %d big endian not supported header got %x at offset 0x%lx size 0x%zx.\n", tplg->pass, hdr->magic, -- cgit v1.2.3 From 5bd70440cb0a6f5c6a84019bb2aa93ab8310a5cd Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 14 Apr 2020 22:04:37 -0500 Subject: ASoC: soc-dai: revert all changes to DAI startup/shutdown sequence On Baytrail/Cherrytrail, the Atom/SST driver fails miserably: [ 9.741953] intel_sst_acpi 80860F28:00: FW Version 01.0c.00.01 [ 9.832992] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.833019] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.833028] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.833033] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.833037] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed [ 9.853942] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.853974] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.853984] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.853990] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.853994] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed Commit b56be800f1292 ("ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once") was the initial problematic commit. Commit 1ba616bd1a6d5e ("ASoC: soc-dai: fix DAI startup/shutdown sequence") was an attempt to fix things but it does not work on Baytrail, reverting all changes seems necessary for now. Fixes: 1ba616bd1a6d5e ("ASoC: soc-dai: fix DAI startup/shutdown sequence") Signed-off-by: Pierre-Louis Bossart Tested-by: Hans de Goede Link: https://lore.kernel.org/r/20200415030437.23803-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 1 - sound/soc/soc-dai.c | 11 ++--------- 2 files changed, 2 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index d4825b82c7a3..b33abe93b905 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -351,7 +351,6 @@ struct snd_soc_dai { /* bit field */ unsigned int probed:1; - unsigned int started[SNDRV_PCM_STREAM_LAST + 1]; }; static inline struct snd_soc_pcm_stream * diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 8f3cad8db89a..31c41559034b 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -295,24 +295,17 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, { int ret = 0; - if (!dai->started[substream->stream] && - dai->driver->ops->startup) + if (dai->driver->ops->startup) ret = dai->driver->ops->startup(substream, dai); - if (ret == 0) - dai->started[substream->stream] = 1; - return ret; } void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream) { - if (dai->started[substream->stream] && - dai->driver->ops->shutdown) + if (dai->driver->ops->shutdown) dai->driver->ops->shutdown(substream, dai); - - dai->started[substream->stream] = 0; } int snd_soc_dai_prepare(struct snd_soc_dai *dai, -- cgit v1.2.3 From a09fb3f28a60ba3e928a1fa94b0456780800299d Mon Sep 17 00:00:00 2001 From: Matthias Blankertz Date: Wed, 15 Apr 2020 16:10:16 +0200 Subject: ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode The parent SSI of a multi-SSI setup must be fully setup, started and stopped since it is also part of the playback/capture setup. So only skip the SSI (as per commit 203cdf51f288 ("ASoC: rsnd: SSI parent cares SWSP bit") and commit 597b046f0d99 ("ASoC: rsnd: control SSICR::EN correctly")) if the SSI is parent outside of a multi-SSI setup. Signed-off-by: Matthias Blankertz Acked-by: Kuninori Morimoto Link: https://lore.kernel.org/r/20200415141017.384017-2-matthias.blankertz@cetitec.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fc5d089868df..d51fb3a39448 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -407,7 +407,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, * We shouldn't exchange SWSP after running. * This means, parent needs to care it. */ - if (rsnd_ssi_is_parent(mod, io)) + if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io)) goto init_end; if (rsnd_io_is_play(io)) @@ -559,7 +559,7 @@ static int rsnd_ssi_start(struct rsnd_mod *mod, * EN is for data output. * SSI parent EN is not needed. */ - if (rsnd_ssi_is_parent(mod, io)) + if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io)) return 0; ssi->cr_en = EN; @@ -582,7 +582,7 @@ static int rsnd_ssi_stop(struct rsnd_mod *mod, if (!rsnd_ssi_is_run_mods(mod, io)) return 0; - if (rsnd_ssi_is_parent(mod, io)) + if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io)) return 0; cr = ssi->cr_own | @@ -620,7 +620,7 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod, if (rsnd_is_gen1(priv)) return 0; - if (rsnd_ssi_is_parent(mod, io)) + if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io)) return 0; if (!rsnd_ssi_is_run_mods(mod, io)) -- cgit v1.2.3 From b94e164759b82d0c1c80d4b1c8f12c9bee83f11d Mon Sep 17 00:00:00 2001 From: Matthias Blankertz Date: Wed, 15 Apr 2020 16:10:17 +0200 Subject: ASoC: rsnd: Fix HDMI channel mapping for multi-SSI mode The HDMI?_SEL register maps up to four stereo SSI data lanes onto the sdata[0..3] inputs of the HDMI output block. The upper half of the register contains four blocks of 4 bits, with the most significant controlling the sdata3 line and the least significant the sdata0 line. The shift calculation has an off-by-one error, causing the parent SSI to be mapped to sdata3, the first multi-SSI child to sdata0 and so forth. As the parent SSI transmits the stereo L/R channels, and the HDMI core expects it on the sdata0 line, this causes no audio to be output when playing stereo audio on a multichannel capable HDMI out, and multichannel audio has permutated channels. Fix the shift calculation to map the parent SSI to sdata0, the first child to sdata1 etc. Signed-off-by: Matthias Blankertz Acked-by: Kuninori Morimoto Link: https://lore.kernel.org/r/20200415141017.384017-3-matthias.blankertz@cetitec.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssiu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index f35d88211887..9c7c3e7539c9 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -221,7 +221,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, i; for_each_rsnd_mod_array(i, pos, io, rsnd_ssi_array) { - shift = (i * 4) + 16; + shift = (i * 4) + 20; val = (val & ~(0xF << shift)) | rsnd_mod_id(pos) << shift; } -- cgit v1.2.3 From 326b509238171d37402dbe308e154cc234ed1960 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Wed, 15 Apr 2020 12:28:49 -0400 Subject: ASoC: codecs: hdac_hdmi: Fix incorrect use of list_for_each_entry MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit If we don't find any pcm, pcm will point at address at an offset from the the list head and not a meaningful structure. Fix this by returning correct pcm if found and NULL if not. Found with coccinelle. Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20200415162849.308-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hdmi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index fba9b749839d..f26b77faed59 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -142,14 +142,14 @@ static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) { - struct hdac_hdmi_pcm *pcm = NULL; + struct hdac_hdmi_pcm *pcm; list_for_each_entry(pcm, &hdmi->pcm_list, head) { if (pcm->cvt == cvt) - break; + return pcm; } - return pcm; + return NULL; } static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, -- cgit v1.2.3 From b392350ec3f229ad9603d3816f753479e441d99a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Apr 2020 18:25:23 +0200 Subject: ALSA: hda/hdmi: Add module option to disable audio component binding As the recent regression showed, we want sometimes to turn off the audio component binding just for debugging. This patch adds the module option to control it easily without compilation. Fixes: ade49db337a9 ("ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207223 Cc: Link: https://lore.kernel.org/r/20200415162523.27499-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bb287a916dae..4eff16053bd5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -38,6 +38,10 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); +static bool enable_acomp = true; +module_param(enable_acomp, bool, 0444); +MODULE_PARM_DESC(enable_acomp, "Enable audio component binding (default=yes)"); + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -2505,6 +2509,11 @@ static void generic_acomp_init(struct hda_codec *codec, { struct hdmi_spec *spec = codec->spec; + if (!enable_acomp) { + codec_info(codec, "audio component disabled by module option\n"); + return; + } + spec->port2pin = port2pin; setup_drm_audio_ops(codec, ops); if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops, -- cgit v1.2.3 From e2bcb65782f91390952e849e21b82ed7cb05697f Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Fri, 17 Apr 2020 16:21:22 +0200 Subject: ASoC: stm32: sai: fix sai probe pcm config must be set before snd_dmaengine_pcm_register() call. Fixes: 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue") Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20200417142122.10212-1-olivier.moysan@st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 34a7c3d6fb91..41f01c3e639e 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1547,6 +1547,9 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) + conf = &stm32_sai_pcm_config_spdif; + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); if (ret) { if (ret != -EPROBE_DEFER) @@ -1556,15 +1559,10 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &stm32_component, &sai->cpu_dai_drv, 1); - if (ret) { + if (ret) snd_dmaengine_pcm_unregister(&pdev->dev); - return ret; - } - - if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) - conf = &stm32_sai_pcm_config_spdif; - return 0; + return ret; } static int stm32_sai_sub_remove(struct platform_device *pdev) -- cgit v1.2.3 From 8c05246c0b58cbe80580ea4be05f6d51228af8a9 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 17 Apr 2020 12:20:14 -0500 Subject: ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell Major regressions were detected by SOF CI on CherryTrail and Broadwell: [ 25.705750] SSP2-Codec: ASoC: no backend playback stream [ 27.923378] SSP2-Codec: ASoC: no users playback at close - state This is root-caused to the introduction of the DAI capability checks with snd_soc_dai_stream_valid(). Its use in soc-pcm.c makes it a requirement for all DAIs to report at least a non-zero min_channels field. For some reason the SSP structures used for SKL+ did provide this information but legacy platforms didn't. Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200417172014.11760-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/bdw.c | 16 ++++++++++++++++ sound/soc/sof/intel/byt.c | 48 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 64 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 6c23c5769330..a32a3ef78ec5 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -567,9 +567,25 @@ static void bdw_set_mach_params(const struct snd_soc_acpi_mach *mach, static struct snd_soc_dai_driver bdw_dai[] = { { .name = "ssp0-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp1-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index f84391294f12..29fd1d86156c 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -459,21 +459,69 @@ static void byt_set_mach_params(const struct snd_soc_acpi_mach *mach, static struct snd_soc_dai_driver byt_dai[] = { { .name = "ssp0-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp1-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp2-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + } }, { .name = "ssp3-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp4-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp5-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; -- cgit v1.2.3 From 1c826792586f526a5a5cd21d55aad388f5bb0b23 Mon Sep 17 00:00:00 2001 From: Alexander Tsoy Date: Sat, 18 Apr 2020 20:58:15 +0300 Subject: ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices Many Focusrite devices supports a limited set of sample rates per altsetting. These includes audio interfaces with ADAT ports: - Scarlett 18i6, 18i8 1st gen, 18i20 1st gen; - Scarlett 18i8 2nd gen, 18i20 2nd gen; - Scarlett 18i8 3rd gen, 18i20 3rd gen; - Clarett 2Pre USB, 4Pre USB, 8Pre USB. Maximum rate is exposed in the last 4 bytes of Format Type descriptor which has a non-standard bLength = 10. Tested-by: Alexey Skobkin Signed-off-by: Alexander Tsoy Cc: Link: https://lore.kernel.org/r/20200418175815.12211-1-alexander@tsoy.me Signed-off-by: Takashi Iwai --- sound/usb/format.c | 51 +++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) (limited to 'sound') diff --git a/sound/usb/format.c b/sound/usb/format.c index 50e1874c847c..5ffb457cc88c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -277,6 +277,52 @@ static bool s1810c_valid_sample_rate(struct audioformat *fp, return false; } +/* + * Many Focusrite devices supports a limited set of sampling rates per + * altsetting. Maximum rate is exposed in the last 4 bytes of Format Type + * descriptor which has a non-standard bLength = 10. + */ +static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip, + struct audioformat *fp, + unsigned int rate) +{ + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned char *fmt; + unsigned int max_rate; + + iface = usb_ifnum_to_if(chip->dev, fp->iface); + if (!iface) + return true; + + alts = &iface->altsetting[fp->altset_idx]; + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_FORMAT_TYPE); + if (!fmt) + return true; + + if (fmt[0] == 10) { /* bLength */ + max_rate = combine_quad(&fmt[6]); + + /* Validate max rate */ + if (max_rate != 48000 && + max_rate != 96000 && + max_rate != 192000 && + max_rate != 384000) { + + usb_audio_info(chip, + "%u:%d : unexpected max rate: %u\n", + fp->iface, fp->altsetting, max_rate); + + return true; + } + + return rate <= max_rate; + } + + return true; +} + /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -319,6 +365,11 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, !s1810c_valid_sample_rate(fp, rate)) goto skip_rate; + /* Filter out invalid rates on Focusrite devices */ + if (USB_ID_VENDOR(chip->usb_id) == 0x1235 && + !focusrite_valid_sample_rate(chip, fp, rate)) + goto skip_rate; + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) -- cgit v1.2.3 From 67791202c5e069cf2ba51db0718d56c634709e78 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 18 Apr 2020 21:06:39 +0200 Subject: ALSA: hda/realtek - Fix unexpected init_amp override The commit 1c76aa5fb48d ("ALSA: hda/realtek - Allow skipping spec->init_amp detection") changed the way to assign spec->init_amp field that specifies the way to initialize the amp. Along with the change, the commit also replaced a few fixups that set spec->init_amp in HDA_FIXUP_ACT_PROBE with HDA_FIXUP_ACT_PRE_PROBE. This was rather aligning to the other fixups, and not supposed to change the actual behavior. However, this change turned out to cause a regression on FSC S7020, which hit exactly the above. The reason was that there is still one place that overrides spec->init_amp after HDA_FIXUP_ACT_PRE_PROBE call, namely in alc_ssid_check(). This patch fixes the regression by adding the proper spec->init_amp override check, i.e. verifying whether it's still ALC_INIT_UNDEFINED. Fixes: 1c76aa5fb48d ("ALSA: hda/realtek - Allow skipping spec->init_amp detection") Cc: BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207329 Link: https://lore.kernel.org/r/20200418190639.10082-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dc5557d79c43..54e1c9c0a33f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -797,9 +797,11 @@ static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports) { if (!alc_subsystem_id(codec, ports)) { struct alc_spec *spec = codec->spec; - codec_dbg(codec, - "realtek: Enable default setup for auto mode as fallback\n"); - spec->init_amp = ALC_INIT_DEFAULT; + if (spec->init_amp == ALC_INIT_UNDEFINED) { + codec_dbg(codec, + "realtek: Enable default setup for auto mode as fallback\n"); + spec->init_amp = ALC_INIT_DEFAULT; + } } } -- cgit v1.2.3 From a8cf44f085ac12c0b5b8750ebb3b436c7f455419 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 19 Apr 2020 09:19:26 +0200 Subject: ALSA: hda: Remove ASUS ROG Zenith from the blacklist The commit 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") added a new blacklist for the devices that are known to have empty codecs, and one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f). However, it turned out that the very same PCI SSID is used for the previous model that does have the valid HD-audio codecs and the change broke the sound on it. This patch reverts the corresponding entry as a temporary solution. Although Zenith II and co will see get the empty HD-audio bus again, it'd be merely resource wastes and won't affect the functionality, so it's no end of the world. We'll need to address this later, e.g. by either switching to DMI string matching or using PCI ID & SSID pairs. Fixes: 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") Reported-by: Johnathan Smithinovic Cc: Link: https://lore.kernel.org/r/20200419071926.22683-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a5fab12defde..d7adae316c0d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2092,7 +2092,6 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, * should be ignored from the beginning. */ static const struct snd_pci_quirk driver_blacklist[] = { - SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0), SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), {} -- cgit v1.2.3 From a43c1c41bc5145971d06edc42a6b1e8faa0e2bc3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Apr 2020 08:20:36 +0200 Subject: ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobos TRX40 mobos from MSI and others with ALC1220-VB USB-audio device need yet more quirks for the proper control names. This patch provides the mapping table for those boards, correcting the FU names for volume and mute controls as well as the terminal names for jack controls. It also improves build_connector_control() not to add the directional suffix blindly if the string is given from the mapping table. With this patch applied, the new UCM profiles will be effective. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Link: https://lore.kernel.org/r/20200420062036.28567-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 12 +++++++++--- sound/usb/mixer_maps.c | 24 +++++++++++++++++++++--- sound/usb/quirks-table.h | 14 ++++++++++++++ 3 files changed, 44 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e7b9040a54e6..ecd5036a0b44 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1776,8 +1776,10 @@ static void build_connector_control(struct usb_mixer_interface *mixer, { struct snd_kcontrol *kctl; struct usb_mixer_elem_info *cval; + const struct usbmix_name_map *map; - if (check_ignored_ctl(find_map(imap, term->id, 0))) + map = find_map(imap, term->id, 0); + if (check_ignored_ctl(map)) return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -1809,8 +1811,12 @@ static void build_connector_control(struct usb_mixer_interface *mixer, usb_mixer_elem_info_free(cval); return; } - get_connector_control_name(mixer, term, is_input, kctl->id.name, - sizeof(kctl->id.name)); + + if (check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name))) + strlcat(kctl->id.name, " Jack", sizeof(kctl->id.name)); + else + get_connector_control_name(mixer, term, is_input, kctl->id.name, + sizeof(kctl->id.name)); kctl->private_free = snd_usb_mixer_elem_free; snd_usb_mixer_add_control(&cval->head, kctl); } diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index b4e77000f441..fb2c4a992951 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -369,6 +369,24 @@ static const struct usbmix_name_map asus_rog_map[] = { {} }; +/* TRX40 mobos with Realtek ALC1220-VB */ +static const struct usbmix_name_map trx40_mobo_map[] = { + { 18, NULL }, /* OT, IEC958 - broken response, disabled */ + { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */ + { 16, "Speaker" }, /* OT */ + { 22, "Speaker Playback" }, /* FU */ + { 7, "Line" }, /* IT */ + { 19, "Line Capture" }, /* FU */ + { 17, "Front Headphone" }, /* OT */ + { 23, "Front Headphone Playback" }, /* FU */ + { 8, "Mic" }, /* IT */ + { 20, "Mic Capture" }, /* FU */ + { 9, "Front Mic" }, /* IT */ + { 21, "Front Mic Capture" }, /* FU */ + { 24, "IEC958 Playback" }, /* FU */ + {} +}; + /* * Control map entries */ @@ -500,7 +518,7 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { }, { /* Gigabyte TRX40 Aorus Pro WiFi */ .id = USB_ID(0x0414, 0xa002), - .map = asus_rog_map, + .map = trx40_mobo_map, }, { /* ASUS ROG Zenith II */ .id = USB_ID(0x0b05, 0x1916), @@ -512,11 +530,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { }, { /* MSI TRX40 Creator */ .id = USB_ID(0x0db0, 0x0d64), - .map = asus_rog_map, + .map = trx40_mobo_map, }, { /* MSI TRX40 */ .id = USB_ID(0x0db0, 0x543d), - .map = asus_rog_map, + .map = trx40_mobo_map, }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e009d584e7d0..7598d6d6740c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3635,4 +3635,18 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +#define ALC1220_VB_DESKTOP(vend, prod) { \ + USB_DEVICE(vend, prod), \ + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \ + .vendor_name = "Realtek", \ + .product_name = "ALC1220-VB-DT", \ + .profile_name = "Realtek-ALC1220-VB-Desktop", \ + .ifnum = QUIRK_NO_INTERFACE \ + } \ +} +ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ +ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ +ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ +#undef ALC1220_VB_DESKTOP + #undef USB_DEVICE_VENDOR_SPEC -- cgit v1.2.3 From 1164284270779e1865cc2046a2a01b58a1e858a9 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Mon, 20 Apr 2020 13:45:10 +0200 Subject: ASoC: meson: axg-card: fix codec-to-codec link setup Since the addition of commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported"), meson-axg cards which have codec-to-codec links fail to init and Oops: Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 Internal error: Oops: 96000044 [#1] PREEMPT SMP CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 pc : invalidate_paths_ep+0x30/0xe0 lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 Call trace: invalidate_paths_ep+0x30/0xe0 snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 dpcm_path_get+0x38/0xd0 dpcm_fe_dai_open+0x70/0x920 snd_pcm_open_substream+0x564/0x840 snd_pcm_open+0xfc/0x228 snd_pcm_capture_open+0x4c/0x78 snd_open+0xac/0x1a8 ... While initiliazing the links, ASoC treats the codec-to-codec links of this card type as a DPCM backend. This error eventually leads to the Oops. Most of the card driver code is shared between DPCM backends and codec-to-codec links. The property "no_pcm" marking DCPM BE was left set on codec-to-codec links, leading to this problem. This commit fixes that. Fixes: 0a8f1117a680 ("ASoC: meson: axg-card: add basic codec-to-codec link support") Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200420114511.450560-2-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-card.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index af46845f4ef2..89f7f64747cd 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -338,8 +338,10 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node)) ret = axg_card_parse_tdm(card, np, index); - else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) + else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) { dai_link->params = &codec_params; + dai_link->no_pcm = 0; /* link is not a DPCM BE */ + } return ret; } -- cgit v1.2.3 From de911b4e683f9c28a063bb62991f2db206c38ba4 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Mon, 20 Apr 2020 13:45:11 +0200 Subject: ASoC: meson: gx-card: fix codec-to-codec link setup Since the addition of commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported"), meson-axg cards which have codec-to-codec links fail to init and Oops. Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 Internal error: Oops: 96000044 [#1] PREEMPT SMP CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 pc : invalidate_paths_ep+0x30/0xe0 lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 Call trace: invalidate_paths_ep+0x30/0xe0 snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 dpcm_path_get+0x38/0xd0 dpcm_fe_dai_open+0x70/0x920 snd_pcm_open_substream+0x564/0x840 snd_pcm_open+0xfc/0x228 snd_pcm_capture_open+0x4c/0x78 snd_open+0xac/0x1a8 ... While this error was initially reported the axg-card type, it also applies to the gx-card type. While initiliazing the links, ASoC treats the codec-to-codec links of this card type as a DPCM backend. This error eventually leads to the Oops. Most of the card driver code is shared between DPCM backends and codec-to-codec links. The property "no_pcm" marking DCPM BE was left set on codec-to-codec links, leading to this problem. This commit fixes that. Fixes: e37a0c313a0f ("ASoC: meson: gx: add sound card support") Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200420114511.450560-3-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/gx-card.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c index 7b01dcb73e5e..4abf7efb7eac 100644 --- a/sound/soc/meson/gx-card.c +++ b/sound/soc/meson/gx-card.c @@ -108,8 +108,10 @@ static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np, ret = gx_card_parse_i2s(card, np, index); /* Or apply codec to codec params if necessary */ - else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) + else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) { dai_link->params = &codec_params; + dai_link->no_pcm = 0; /* link is not a DPCM BE */ + } return ret; } -- cgit v1.2.3 From 0c258657ddfe81b4fc0183378d800c97ba0b7cdd Mon Sep 17 00:00:00 2001 From: Matthias Blankertz Date: Fri, 17 Apr 2020 17:30:16 +0200 Subject: ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent The master SSI of a multi-SSI setup was attached both to the RSND_MOD_SSI slot and the RSND_MOD_SSIP slot of the rsnd_dai_stream. This is not correct wrt. the meaning of being "parent" in the rest of the SSI code, where it seems to indicate an SSI that provides clock and word sync but is not transmitting/receiving audio data. Not treating the multi-SSI master as parent allows removal of various special cases to the rsnd_ssi_is_parent conditions introduced in commit a09fb3f28a60 ("ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode"). It also fixes the issue that operations performed via rsnd_dai_call() were performed twice for the master SSI. This caused some "status check failed" spam when stopping a multi-SSI stream as the driver attempted to stop the master SSI twice. Signed-off-by: Matthias Blankertz Acked-by: Kuninori Morimoto Link: https://lore.kernel.org/r/20200417153017.1744454-2-matthias.blankertz@cetitec.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index d51fb3a39448..9900a4f6f4e5 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -407,7 +407,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, * We shouldn't exchange SWSP after running. * This means, parent needs to care it. */ - if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io)) + if (rsnd_ssi_is_parent(mod, io)) goto init_end; if (rsnd_io_is_play(io)) @@ -559,7 +559,7 @@ static int rsnd_ssi_start(struct rsnd_mod *mod, * EN is for data output. * SSI parent EN is not needed. */ - if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io)) + if (rsnd_ssi_is_parent(mod, io)) return 0; ssi->cr_en = EN; @@ -582,7 +582,7 @@ static int rsnd_ssi_stop(struct rsnd_mod *mod, if (!rsnd_ssi_is_run_mods(mod, io)) return 0; - if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io)) + if (rsnd_ssi_is_parent(mod, io)) return 0; cr = ssi->cr_own | @@ -620,7 +620,7 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod, if (rsnd_is_gen1(priv)) return 0; - if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io)) + if (rsnd_ssi_is_parent(mod, io)) return 0; if (!rsnd_ssi_is_run_mods(mod, io)) @@ -737,6 +737,9 @@ static void rsnd_ssi_parent_attach(struct rsnd_mod *mod, if (!rsnd_rdai_is_clk_master(rdai)) return; + if (rsnd_ssi_is_multi_slave(mod, io)) + return; + switch (rsnd_mod_id(mod)) { case 1: case 2: -- cgit v1.2.3 From 54cb6221688660670a2e430892d7f4e6370263b8 Mon Sep 17 00:00:00 2001 From: Matthias Blankertz Date: Fri, 17 Apr 2020 17:30:17 +0200 Subject: ASoC: rsnd: Fix "status check failed" spam for multi-SSI Fix the rsnd_ssi_stop function to skip disabling the individual SSIs of a multi-SSI setup, as the actual stop is performed by rsnd_ssiu_stop_gen2 - the same logic as in rsnd_ssi_start. The attempt to disable these SSIs was harmless, but caused a "status check failed" message to be printed for every SSI in the multi-SSI setup. The disabling of interrupts is still performed, as they are enabled for all SSIs in rsnd_ssi_init, but care is taken to not accidentally set the EN bit for an SSI where it was not set by rsnd_ssi_start. Signed-off-by: Matthias Blankertz Acked-by: Kuninori Morimoto Link: https://lore.kernel.org/r/20200417153017.1744454-3-matthias.blankertz@cetitec.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 9900a4f6f4e5..4a7d3413917f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -594,10 +594,16 @@ static int rsnd_ssi_stop(struct rsnd_mod *mod, * Capture: It might not receave data. Do nothing */ if (rsnd_io_is_play(io)) { - rsnd_mod_write(mod, SSICR, cr | EN); + rsnd_mod_write(mod, SSICR, cr | ssi->cr_en); rsnd_ssi_status_check(mod, DIRQ); } + /* In multi-SSI mode, stop is performed by setting ssi0129 in + * SSI_CONTROL to 0 (in rsnd_ssio_stop_gen2). Do nothing here. + */ + if (rsnd_ssi_multi_slaves_runtime(io)) + return 0; + /* * disable SSI, * and, wait idle state -- cgit v1.2.3 From ebf1474745b4373fdde0fcf32d9d1f369b50b212 Mon Sep 17 00:00:00 2001 From: Gyeongtaek Lee Date: Sat, 18 Apr 2020 13:13:20 +0900 Subject: ASoC: dapm: fixup dapm kcontrol widget snd_soc_dapm_kcontrol widget which is created by autodisable control should contain correct on_val, mask and shift because it is set when the widget is powered and changed value is applied on registers by following code in dapm_seq_run_coalesced(). mask |= w->mask << w->shift; if (w->power) value |= w->on_val << w->shift; else value |= w->off_val << w->shift; Shift on the mask in dapm_kcontrol_data_alloc() is removed to prevent double shift. And, on_val in dapm_kcontrol_set_value() is modified to get correct value in the dapm_seq_run_coalesced(). Signed-off-by: Gyeongtaek Lee Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/000001d61537$b212f620$1638e260$@samsung.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a4143ca190d0..e2632841b321 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -423,7 +423,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, memset(&template, 0, sizeof(template)); template.reg = e->reg; - template.mask = e->mask << e->shift_l; + template.mask = e->mask; template.shift = e->shift_l; template.off_val = snd_soc_enum_item_to_val(e, 0); template.on_val = template.off_val; @@ -546,8 +546,22 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, if (data->value == value) return false; - if (data->widget) - data->widget->on_val = value; + if (data->widget) { + switch (dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + data->widget->on_val = value & data->widget->mask; + break; + case snd_soc_dapm_demux: + case snd_soc_dapm_mux: + data->widget->on_val = value >> data->widget->shift; + break; + default: + data->widget->on_val = value; + break; + } + } data->value = value; -- cgit v1.2.3 From 9bff3d3024e51122c0c09634056debcd6c7359ec Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Mon, 20 Apr 2020 20:53:43 +0800 Subject: ASoC: wm89xx: Add missing dependency sound/soc/codecs/wm8900.o: In function `wm8900_i2c_probe': wm8900.c:(.text+0xa36): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8900.o: In function `wm8900_modinit': wm8900.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8900.o: In function `wm8900_exit': wm8900.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' sound/soc/codecs/wm8988.o: In function `wm8988_i2c_probe': wm8988.c:(.text+0x857): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8988.o: In function `wm8988_modinit': wm8988.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8988.o: In function `wm8988_exit': wm8988.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' sound/soc/codecs/wm8995.o: In function `wm8995_i2c_probe': wm8995.c:(.text+0x1c4f): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8995.o: In function `wm8995_modinit': wm8995.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8995.o: In function `wm8995_exit': wm8995.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' Add SND_SOC_I2C_AND_SPI dependency to fix this. Fixes: ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS") Reported-by: Hulk Robot Signed-off-by: YueHaibing Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20200420125343.20920-1-yuehaibing@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e6a0c5d05fa5..e60e0b6a689c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1525,6 +1525,7 @@ config SND_SOC_WM8804_SPI config SND_SOC_WM8900 tristate + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8903 tristate "Wolfson Microelectronics WM8903 CODEC" @@ -1576,6 +1577,7 @@ config SND_SOC_WM8985 config SND_SOC_WM8988 tristate + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8990 tristate @@ -1594,6 +1596,7 @@ config SND_SOC_WM8994 config SND_SOC_WM8995 tristate + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8996 tristate -- cgit v1.2.3 From 6f4ea2074ddf689ac6f892afa58515032dabf2e4 Mon Sep 17 00:00:00 2001 From: Gregor Pintar Date: Mon, 20 Apr 2020 23:40:30 +0200 Subject: ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2 Force it to use asynchronous playback. Same quirk has already been added for Focusrite Scarlett Solo (2nd gen) with a commit 46f5710f0b88 ("ALSA: usb-audio: Add quirk for Focusrite Scarlett Solo"). This also seems to prevent regular clicks when playing at 44100Hz on Scarlett 2i2 (2nd gen). I did not notice any side effects. Moved both quirks to snd_usb_audioformat_attributes_quirk() as suggested. Signed-off-by: Gregor Pintar Reviewed-by: Alexander Tsoy Link: https://lore.kernel.org/r/20200420214030.2361-1-grpintar@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 84 ------------------------------------------------ sound/usb/quirks.c | 13 ++++++++ 2 files changed, 13 insertions(+), 84 deletions(-) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 7598d6d6740c..a1df4c5b4f8c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2756,90 +2756,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, -{ - /* - * Focusrite Scarlett Solo 2nd generation - * Reports that playback should use Synch: Synchronous - * while still providing a feedback endpoint. Synchronous causes - * snapping on some sample rates. - * Force it to use Synch: Asynchronous. - */ - USB_DEVICE(0x1235, 0x8205), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { - .formats = SNDRV_PCM_FMTBIT_S32_LE, - .channels = 2, - .iface = 1, - .altsetting = 1, - .altset_idx = 1, - .attributes = 0, - .endpoint = 0x01, - .ep_attr = USB_ENDPOINT_XFER_ISOC | - USB_ENDPOINT_SYNC_ASYNC, - .protocol = UAC_VERSION_2, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_176400 | - SNDRV_PCM_RATE_192000, - .rate_min = 44100, - .rate_max = 192000, - .nr_rates = 6, - .rate_table = (unsigned int[]) { - 44100, 48000, 88200, - 96000, 176400, 192000 - }, - .clock = 41 - } - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { - .formats = SNDRV_PCM_FMTBIT_S32_LE, - .channels = 2, - .iface = 2, - .altsetting = 1, - .altset_idx = 1, - .attributes = 0, - .endpoint = 0x82, - .ep_attr = USB_ENDPOINT_XFER_ISOC | - USB_ENDPOINT_SYNC_ASYNC | - USB_ENDPOINT_USAGE_IMPLICIT_FB, - .protocol = UAC_VERSION_2, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_176400 | - SNDRV_PCM_RATE_192000, - .rate_min = 44100, - .rate_max = 192000, - .nr_rates = 6, - .rate_table = (unsigned int[]) { - 44100, 48000, 88200, - 96000, 176400, 192000 - }, - .clock = 41 - } - }, - { - .ifnum = 3, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = -1 - } - } - } -}, /* Access Music devices */ { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a8ece1701068..6c2dfd3bfcbf 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1806,6 +1806,19 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, */ fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX; break; + case USB_ID(0x1235, 0x8202): /* Focusrite Scarlett 2i2 2nd gen */ + case USB_ID(0x1235, 0x8205): /* Focusrite Scarlett Solo 2nd gen */ + /* + * Reports that playback should use Synch: Synchronous + * while still providing a feedback endpoint. + * Synchronous causes snapping on some sample rates. + * Force it to use Synch: Asynchronous. + */ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; + fp->ep_attr |= USB_ENDPOINT_SYNC_ASYNC; + } + break; } } -- cgit v1.2.3 From 7686e3485253635c529cdd5f416fc640abaf076f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Apr 2020 09:55:29 +0200 Subject: ALSA: usx2y: Fix potential NULL dereference The error handling code in usX2Y_rate_set() may hit a potential NULL dereference when an error occurs before allocating all us->urb[]. Add a proper NULL check for fixing the corner case. Reported-by: Lin Yi Cc: Link: https://lore.kernel.org/r/20200420075529.27203-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/usx2y/usbusx2yaudio.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 37d290fe9d43..ecaf41265dcd 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -681,6 +681,8 @@ static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) us->submitted = 2*NOOF_SETRATE_URBS; for (i = 0; i < NOOF_SETRATE_URBS; ++i) { struct urb *urb = us->urb[i]; + if (!urb) + continue; if (urb->status) { if (!err) err = -ENODEV; -- cgit v1.2.3 From 1e060a453c8604311fb45ae2f84f67ed673329b4 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Tue, 21 Apr 2020 19:28:45 +0800 Subject: ASoC: wm8960: Fix wrong clock after suspend & resume After suspend & resume, wm8960_hw_params may be called when bias_level is not SND_SOC_BIAS_ON, then wm8960_configure_clocking is not called. But if sample rate is changed at that time, then the output clock rate will be not correct. So judgement of bias_level is SND_SOC_BIAS_ON in wm8960_hw_params is not necessary and it causes above issue. Fixes: 3176bf2d7ccd ("ASoC: wm8960: update pll and clock setting function") Signed-off-by: Shengjiu Wang Acked-by: Charles Keepax Link: https://lore.kernel.org/r/1587468525-27514-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 55112c1bba5e..6cf0f6612bda 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -860,8 +860,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, wm8960->is_stream_in_use[tx] = true; - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_ON && - !wm8960->is_stream_in_use[!tx]) + if (!wm8960->is_stream_in_use[!tx]) return wm8960_configure_clocking(component); return 0; -- cgit v1.2.3 From cf9fb7b8737694818e783fc61e8fe220b7eaaf60 Mon Sep 17 00:00:00 2001 From: Alexander Tsoy Date: Tue, 21 Apr 2020 22:09:08 +0300 Subject: ALSA: usb-audio: Apply async workaround for Scarlett 2i4 2nd gen Due to rounding error driver sometimes incorrectly calculate next packet size, which results in audible clicks on devices with synchronous playback endpoints. For example on a high speed bus and a sample rate 44.1 kHz it loses one sample every ~40.9 seconds. Fortunately playback interface on Scarlett 2i4 2nd gen has a working explicit feedback endpoint, so we can switch playback data endpoint to asynchronous mode as a workaround. Signed-off-by: Alexander Tsoy Link: https://lore.kernel.org/r/20200421190908.462860-1-alexander@tsoy.me Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6c2dfd3bfcbf..351ba214a9d3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1806,6 +1806,7 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, */ fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX; break; + case USB_ID(0x1235, 0x8200): /* Focusrite Scarlett 2i4 2nd gen */ case USB_ID(0x1235, 0x8202): /* Focusrite Scarlett 2i2 2nd gen */ case USB_ID(0x1235, 0x8205): /* Focusrite Scarlett Solo 2nd gen */ /* -- cgit v1.2.3 From fef66ae73a611e84c8b4b74ff6f805ec5f113477 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 22 Apr 2020 13:33:20 +0200 Subject: ALSA: usb-audio: Add connector notifier delegation It turned out that ALC1220-VB USB-audio device gives the interrupt event to some PCM terminals while those don't allow the connector state request but only the actual I/O terminals return the request. The recent commit 7dc3c5a0172e ("ALSA: usb-audio: Don't create jack controls for PCM terminals") excluded those phantom terminals, so those events are ignored, too. My first thought was that this could be easily deduced from the associated terminals, but some of them have even no associate terminal ID, hence it's not too trivial to figure out. Since the number of such terminals are small and limited, this patch implements another quirk table for the simple mapping of the connectors. It's not really scalable, but let's hope that there will be not many such funky devices in future. Fixes: 7dc3c5a0172e ("ALSA: usb-audio: Don't create jack controls for PCM terminals") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Link: https://lore.kernel.org/r/20200422113320.26664-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 25 +++++++++++++++++++++++++ sound/usb/mixer.h | 10 ++++++++++ sound/usb/mixer_maps.c | 13 +++++++++++++ 3 files changed, 48 insertions(+) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ecd5036a0b44..a88d7854513b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3117,6 +3117,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (map->id == state.chip->usb_id) { state.map = map->map; state.selector_map = map->selector_map; + mixer->connector_map = map->connector_map; mixer->ignore_ctl_error |= map->ignore_ctl_error; break; } @@ -3198,10 +3199,32 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) return 0; } +static int delegate_notify(struct usb_mixer_interface *mixer, int unitid, + u8 *control, u8 *channel) +{ + const struct usbmix_connector_map *map = mixer->connector_map; + + if (!map) + return unitid; + + for (; map->id; map++) { + if (map->id == unitid) { + if (control && map->control) + *control = map->control; + if (channel && map->channel) + *channel = map->channel; + return map->delegated_id; + } + } + return unitid; +} + void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) { struct usb_mixer_elem_list *list; + unitid = delegate_notify(mixer, unitid, NULL, NULL); + for_each_mixer_elem(list, mixer, unitid) { struct usb_mixer_elem_info *info = mixer_elem_list_to_info(list); @@ -3271,6 +3294,8 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, return; } + unitid = delegate_notify(mixer, unitid, &control, &channel); + for_each_mixer_elem(list, mixer, unitid) count++; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 65d6d08c96f5..41ec9dc4139b 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -6,6 +6,13 @@ struct media_mixer_ctl; +struct usbmix_connector_map { + u8 id; + u8 delegated_id; + u8 control; + u8 channel; +}; + struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -18,6 +25,9 @@ struct usb_mixer_interface { /* the usb audio specification version this interface complies to */ int protocol; + /* optional connector delegation map */ + const struct usbmix_connector_map *connector_map; + /* Sound Blaster remote control stuff */ const struct rc_config *rc_cfg; u32 rc_code; diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index fb2c4a992951..0260c750e156 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -27,6 +27,7 @@ struct usbmix_ctl_map { u32 id; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; + const struct usbmix_connector_map *connector_map; int ignore_ctl_error; }; @@ -387,6 +388,15 @@ static const struct usbmix_name_map trx40_mobo_map[] = { {} }; +static const struct usbmix_connector_map trx40_mobo_connector_map[] = { + { 10, 16 }, /* (Back) Speaker */ + { 11, 17 }, /* Front Headphone */ + { 13, 7 }, /* Line */ + { 14, 8 }, /* Mic */ + { 15, 9 }, /* Front Mic */ + {} +}; + /* * Control map entries */ @@ -519,6 +529,7 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { { /* Gigabyte TRX40 Aorus Pro WiFi */ .id = USB_ID(0x0414, 0xa002), .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, }, { /* ASUS ROG Zenith II */ .id = USB_ID(0x0b05, 0x1916), @@ -531,10 +542,12 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { { /* MSI TRX40 Creator */ .id = USB_ID(0x0db0, 0x0d64), .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, }, { /* MSI TRX40 */ .id = USB_ID(0x0db0, 0x543d), .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, }, { 0 } /* terminator */ }; -- cgit v1.2.3 From 59e1947ca09ebd1cae147c08c7c41f3141233c84 Mon Sep 17 00:00:00 2001 From: Xiyu Yang Date: Thu, 23 Apr 2020 12:54:19 +0800 Subject: ALSA: usb-audio: Fix usb audio refcnt leak when getting spdif snd_microii_spdif_default_get() invokes snd_usb_lock_shutdown(), which increases the refcount of the snd_usb_audio object "chip". When snd_microii_spdif_default_get() returns, local variable "chip" becomes invalid, so the refcount should be decreased to keep refcount balanced. The reference counting issue happens in several exception handling paths of snd_microii_spdif_default_get(). When those error scenarios occur such as usb_ifnum_to_if() returns NULL, the function forgets to decrease the refcnt increased by snd_usb_lock_shutdown(), causing a refcnt leak. Fix this issue by jumping to "end" label when those error scenarios occur. Fixes: 447d6275f0c2 ("ALSA: usb-audio: Add sanity checks for endpoint accesses") Signed-off-by: Xiyu Yang Signed-off-by: Xin Tan Cc: Link: https://lore.kernel.org/r/1587617711-13200-1-git-send-email-xiyuyang19@fudan.edu.cn Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 02b036b2aefb..a5f65a9a0254 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1509,11 +1509,15 @@ static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol, /* use known values for that card: interface#1 altsetting#1 */ iface = usb_ifnum_to_if(chip->dev, 1); - if (!iface || iface->num_altsetting < 2) - return -EINVAL; + if (!iface || iface->num_altsetting < 2) { + err = -EINVAL; + goto end; + } alts = &iface->altsetting[1]; - if (get_iface_desc(alts)->bNumEndpoints < 1) - return -EINVAL; + if (get_iface_desc(alts)->bNumEndpoints < 1) { + err = -EINVAL; + goto end; + } ep = get_endpoint(alts, 0)->bEndpointAddress; err = snd_usb_ctl_msg(chip->dev, -- cgit v1.2.3 From 7fbdcd8301a84c09cebfa64f1317a6dafeec9188 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 23 Apr 2020 14:18:31 +0800 Subject: ALSA: hda/realtek - Add new codec supported for ALC245 Enable new codec supported for ALC245. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/8c0804738b2c42439f59c39c8437817f@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 54e1c9c0a33f..c1a85c8f7b69 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -377,6 +377,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0233: case 0x10ec0235: case 0x10ec0236: + case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: case 0x10ec0257: @@ -8198,6 +8199,7 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0215: + case 0x10ec0245: case 0x10ec0285: case 0x10ec0289: spec->codec_variant = ALC269_TYPE_ALC215; @@ -9459,6 +9461,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269), HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269), HDA_CODEC_ENTRY(0x10ec0236, "ALC236", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0245, "ALC245", patch_alc269), HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269), HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269), HDA_CODEC_ENTRY(0x10ec0257, "ALC257", patch_alc269), -- cgit v1.2.3 From 8d6762af302d69f76fa788a277a56a9d9cd275d5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 22 Apr 2020 22:37:44 +0200 Subject: ALSA: hda: Always use jackpoll helper for jack update after resume HD-audio codec driver applies a tricky procedure to forcibly perform the runtime resume by mimicking the usage count even if the device has been runtime-suspended beforehand. This was needed to assure to trigger the jack detection update after the system resume. And recently we also applied the similar logic to the HD-audio controller side. However this seems leading to some inconsistency, and eventually PCI controller gets screwed up. This patch is an attempt to fix and clean up those behavior: instead of the tricky runtime resume procedure, the existing jackpoll work is scheduled when such a forced codec resume is required. The jackpoll work will power up the codec, and this alone should suffice for the jack status update in usual cases. If the extra polling is requested (by checking codec->jackpoll_interval), the manual update is invoked after that, and the codec is powered down again. Also, we filter the spurious wake up of the codec from the controller runtime resume by checking codec->relaxed_resume flag. If this flag is set, basically we don't need to wake up explicitly, but it's supposed to be done via the audio component notifier. Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed") Link: https://lore.kernel.org/r/20200422203744.26299-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 28 +++++++++++++++++----------- sound/pci/hda/hda_intel.c | 17 ++--------------- 2 files changed, 19 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 86a632bf4d50..7e3ae4534df9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -641,8 +641,18 @@ static void hda_jackpoll_work(struct work_struct *work) struct hda_codec *codec = container_of(work, struct hda_codec, jackpoll_work.work); - snd_hda_jack_set_dirty_all(codec); - snd_hda_jack_poll_all(codec); + /* for non-polling trigger: we need nothing if already powered on */ + if (!codec->jackpoll_interval && snd_hdac_is_power_on(&codec->core)) + return; + + /* the power-up/down sequence triggers the runtime resume */ + snd_hda_power_up_pm(codec); + /* update jacks manually if polling is required, too */ + if (codec->jackpoll_interval) { + snd_hda_jack_set_dirty_all(codec); + snd_hda_jack_poll_all(codec); + } + snd_hda_power_down_pm(codec); if (!codec->jackpoll_interval) return; @@ -2951,18 +2961,14 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = hda_codec_need_resume(codec); int ret; - /* The get/put pair below enforces the runtime resume even if the - * device hasn't been used at suspend time. This trick is needed to - * update the jack state change during the sleep. - */ - if (forced_resume) - pm_runtime_get_noresume(dev); ret = pm_runtime_force_resume(dev); - if (forced_resume) - pm_runtime_put(dev); + /* schedule jackpoll work for jack detection update */ + if (codec->jackpoll_interval || + (pm_runtime_suspended(dev) && hda_codec_need_resume(codec))) + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); return ret; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d7adae316c0d..457a2c065485 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1004,7 +1004,8 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt) if (status && from_rt) { list_for_each_codec(codec, &chip->bus) - if (status & (1 << codec->addr)) + if (!codec->relaxed_resume && + (status & (1 << codec->addr))) schedule_delayed_work(&codec->jackpoll_work, codec->jackpoll_interval); } @@ -1044,9 +1045,7 @@ static int azx_suspend(struct device *dev) static int azx_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); - struct hda_codec *codec; struct azx *chip; - bool forced_resume = false; if (!azx_is_pm_ready(card)) return 0; @@ -1058,19 +1057,7 @@ static int azx_resume(struct device *dev) if (azx_acquire_irq(chip, 1) < 0) return -EIO; - /* check for the forced resume */ - list_for_each_codec(codec, &chip->bus) { - if (hda_codec_need_resume(codec)) { - forced_resume = true; - break; - } - } - - if (forced_resume) - pm_runtime_get_noresume(dev); pm_runtime_force_resume(dev); - if (forced_resume) - pm_runtime_put(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); -- cgit v1.2.3 From 977dfef40c8996b69afe23a9094d184049efb7bb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Apr 2020 08:12:22 +0200 Subject: ALSA: hda: Match both PCI ID and SSID for driver blacklist The commit 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") added a new blacklist for the devices that are known to have empty codecs, and one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f). However, it turned out that the very same PCI SSID is used for the previous model that does have the valid HD-audio codecs and the change broke the sound on it. Since the empty codec problem appear on the certain AMD platform (PCI ID 1022:1487), this patch changes the blacklist matching to both PCI ID and SSID using pci_match_id(). Also, the entry that was removed by the previous fix for ASUS ROG Zenigh II is re-added. Link: https://lore.kernel.org/r/20200424061222.19792-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 457a2c065485..0310193ea1bd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2078,9 +2078,10 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, * some HD-audio PCI entries are exposed without any codecs, and such devices * should be ignored from the beginning. */ -static const struct snd_pci_quirk driver_blacklist[] = { - SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), - SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), +static const struct pci_device_id driver_blacklist[] = { + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1043, 0x874f) }, /* ASUS ROG Zenith II / Strix */ + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb59) }, /* MSI TRX40 Creator */ + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb60) }, /* MSI TRX40 */ {} }; @@ -2100,7 +2101,7 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; - if (snd_pci_quirk_lookup(pci, driver_blacklist)) { + if (pci_match_id(driver_blacklist, pci)) { dev_info(&pci->dev, "Skipping the blacklisted device\n"); return -ENODEV; } -- cgit v1.2.3 From ac957e8c54115c1ed32e41e0072af3a63576cda6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Apr 2020 21:38:43 +0200 Subject: ALSA: pcm: oss: Place the plugin buffer overflow checks correctly (for 5.7) [ This is again a forward-port of the fix applied for 5.6-base code (commit 4285de0725b1) to 5.7-base, hence neither Fixes nor Cc-to-stable tags are included here -- tiwai ] The checks of the plugin buffer overflow in the previous fix by commit f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") are put in the wrong places mistakenly, which leads to the expected (repeated) sound when the rate plugin is involved. Fix in the right places. Also, at those right places, the zero check is needed for the termination node, so added there as well, and let's get it done, finally. Link: https://lore.kernel.org/r/20200424193843.20397-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_plugin.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 59d62f05658f..1545f8fdb4db 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -205,13 +205,14 @@ static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_first(plug); while (plugin && frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; if (plugin->dst_frames) { frames = plugin->dst_frames(plugin, frames); if (frames < 0) return frames; } - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin = plugin_next; } return frames; @@ -225,14 +226,15 @@ static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_last(plug); while (plugin && frames > 0) { - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); if (frames < 0) return frames; } + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_prev; } return frames; -- cgit v1.2.3 From cc18b2f4f3f1d7ed3125ac1840794f9feab0325c Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Sat, 25 Apr 2020 13:11:15 -0700 Subject: ALSA: line6: Fix POD HD500 audio playback Apparently interface 1 is control interface akin to HD500X, setting LINE6_CAP_CONTROL and choosing it as ctrl_if fixes audio playback on POD HD500. Signed-off-by: Vasily Khoruzhick Cc: Link: https://lore.kernel.org/r/20200425201115.3430-1-anarsoul@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/line6/podhd.c | 22 +++++----------------- 1 file changed, 5 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index d37db32ecd3b..e39dc85c355a 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -21,8 +21,7 @@ enum { LINE6_PODHD300, LINE6_PODHD400, - LINE6_PODHD500_0, - LINE6_PODHD500_1, + LINE6_PODHD500, LINE6_PODX3, LINE6_PODX3LIVE, LINE6_PODHD500X, @@ -318,8 +317,7 @@ static const struct usb_device_id podhd_id_table[] = { /* TODO: no need to alloc data interfaces when only audio is used */ { LINE6_DEVICE(0x5057), .driver_info = LINE6_PODHD300 }, { LINE6_DEVICE(0x5058), .driver_info = LINE6_PODHD400 }, - { LINE6_IF_NUM(0x414D, 0), .driver_info = LINE6_PODHD500_0 }, - { LINE6_IF_NUM(0x414D, 1), .driver_info = LINE6_PODHD500_1 }, + { LINE6_IF_NUM(0x414D, 0), .driver_info = LINE6_PODHD500 }, { LINE6_IF_NUM(0x414A, 0), .driver_info = LINE6_PODX3 }, { LINE6_IF_NUM(0x414B, 0), .driver_info = LINE6_PODX3LIVE }, { LINE6_IF_NUM(0x4159, 0), .driver_info = LINE6_PODHD500X }, @@ -352,23 +350,13 @@ static const struct line6_properties podhd_properties_table[] = { .ep_audio_r = 0x82, .ep_audio_w = 0x01, }, - [LINE6_PODHD500_0] = { + [LINE6_PODHD500] = { .id = "PODHD500", .name = "POD HD500", - .capabilities = LINE6_CAP_PCM + .capabilities = LINE6_CAP_PCM | LINE6_CAP_CONTROL | LINE6_CAP_HWMON, .altsetting = 1, - .ep_ctrl_r = 0x81, - .ep_ctrl_w = 0x01, - .ep_audio_r = 0x86, - .ep_audio_w = 0x02, - }, - [LINE6_PODHD500_1] = { - .id = "PODHD500", - .name = "POD HD500", - .capabilities = LINE6_CAP_PCM - | LINE6_CAP_HWMON, - .altsetting = 0, + .ctrl_if = 1, .ep_ctrl_r = 0x81, .ep_ctrl_w = 0x01, .ep_audio_r = 0x86, -- cgit v1.2.3 From ef0b3203c758b6b8abdb5dca651880347eae6b8c Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Mon, 27 Apr 2020 11:00:39 +0800 Subject: ALSA: hda/realtek - Two front mics on a Lenovo ThinkCenter This new Lenovo ThinkCenter has two front mics which can't be handled by PA so far, so apply the fixup ALC283_FIXUP_HEADSET_MIC to change the location for one of the mics. Cc: Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20200427030039.10121-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c1a85c8f7b69..c16f63957c5a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7420,6 +7420,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), + SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), -- cgit v1.2.3 From ca76282b6faffc83601c25bd2a95f635c03503ef Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Tue, 28 Apr 2020 15:38:36 +0300 Subject: ALSA: hda/hdmi: fix race in monitor detection during probe A race exists between build_pcms() and build_controls() phases of codec setup. Build_pcms() sets up notifier for jack events. If a monitor event is received before build_controls() is run, the initial jack state is lost and never reported via mixer controls. The problem can be hit at least with SOF as the controller driver. SOF calls snd_hda_codec_build_controls() in its workqueue-based probe and this can be delayed enough to hit the race condition. Fix the issue by invalidating the per-pin ELD information when build_controls() is called. The existing call to hdmi_present_sense() will update the ELD contents. This ensures initial monitor state is correctly reflected via mixer controls. BugLink: https://github.com/thesofproject/linux/issues/1687 Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200428123836.24512-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4eff16053bd5..2e2c382fe4b5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2198,7 +2198,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + struct hdmi_eld *pin_eld = &per_pin->sink_eld; + pin_eld->eld_valid = false; hdmi_present_sense(per_pin, 0); } -- cgit v1.2.3 From a2f647240998aa49632fb09b01388fdf2b87acfc Mon Sep 17 00:00:00 2001 From: Wu Bo Date: Sun, 26 Apr 2020 21:17:22 +0800 Subject: ALSA: hda/hdmi: fix without unlocked before return Fix the following coccicheck warning: sound/pci/hda/patch_hdmi.c:1852:2-8: preceding lock on line 1846 After add sanity check to pass klockwork check, The spdif_mutex should be unlock before return true in check_non_pcm_per_cvt(). Fixes: 960a581e22d9 ("ALSA: hda: fix some klockwork scan warnings") Signed-off-by: Wu Bo Cc: Link: https://lore.kernel.org/r/1587907042-694161-1-git-send-email-wubo40@huawei.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2e2c382fe4b5..93760a3564cf 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1848,8 +1848,10 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) /* Add sanity check to pass klockwork check. * This should never happen. */ - if (WARN_ON(spdif == NULL)) + if (WARN_ON(spdif == NULL)) { + mutex_unlock(&codec->spdif_mutex); return true; + } non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); mutex_unlock(&codec->spdif_mutex); return non_pcm; -- cgit v1.2.3 From 5ce00760a84848d008554c693ceb6286f4d9c509 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 29 Apr 2020 21:02:03 +0200 Subject: ALSA: opti9xx: shut up gcc-10 range warning gcc-10 points out a few instances of suspicious integer arithmetic leading to value truncation: sound/isa/opti9xx/opti92x-ad1848.c: In function 'snd_opti9xx_configure': sound/isa/opti9xx/opti92x-ad1848.c:322:43: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_opti9xx_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow] 322 | (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~ sound/isa/opti9xx/opti92x-ad1848.c:351:3: note: in expansion of macro 'snd_opti9xx_write_mask' 351 | snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); | ^~~~~~~~~~~~~~~~~~~~~~ sound/isa/opti9xx/miro.c: In function 'snd_miro_configure': sound/isa/opti9xx/miro.c:873:40: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_miro_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow] 873 | (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask))) | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~ sound/isa/opti9xx/miro.c:1010:3: note: in expansion of macro 'snd_miro_write_mask' 1010 | snd_miro_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); | ^~~~~~~~~~~~~~~~~~~ These are all harmless here as only the low 8 bit are passed down anyway. Change the macros to inline functions to make the code more readable and also avoid the warning. Strictly speaking those functions also need locking to make the read/write pair atomic, but it seems unlikely that anyone would still run into that issue. Fixes: 1841f613fd2e ("[ALSA] Add snd-miro driver") Signed-off-by: Arnd Bergmann Link: https://lore.kernel.org/r/20200429190216.85919-1-arnd@arndb.de Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 9 ++++++--- sound/isa/opti9xx/opti92x-ad1848.c | 9 ++++++--- 2 files changed, 12 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index e764816a8f7a..b039429e6871 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -867,10 +867,13 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg, spin_unlock_irqrestore(&chip->lock, flags); } +static inline void snd_miro_write_mask(struct snd_miro *chip, + unsigned char reg, unsigned char value, unsigned char mask) +{ + unsigned char oldval = snd_miro_read(chip, reg); -#define snd_miro_write_mask(chip, reg, value, mask) \ - snd_miro_write(chip, reg, \ - (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask))) + snd_miro_write(chip, reg, (oldval & ~mask) | (value & mask)); +} /* * Proc Interface diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index d06b29693c85..0e6d20e49158 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -317,10 +317,13 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, } -#define snd_opti9xx_write_mask(chip, reg, value, mask) \ - snd_opti9xx_write(chip, reg, \ - (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) +static inline void snd_opti9xx_write_mask(struct snd_opti9xx *chip, + unsigned char reg, unsigned char value, unsigned char mask) +{ + unsigned char oldval = snd_opti9xx_read(chip, reg); + snd_opti9xx_write(chip, reg, (oldval & ~mask) | (value & mask)); +} static int snd_opti9xx_configure(struct snd_opti9xx *chip, long port, -- cgit v1.2.3 From 547d2c9cf4f1f72adfecacbd5b093681fb0e8b3e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Apr 2020 14:47:55 +0200 Subject: ALSA: usb-audio: Correct a typo of NuPrime DAC-10 USB ID The USB vendor ID of NuPrime DAC-10 is not 16b0 but 16d0. Fixes: f656891c6619 ("ALSA: usb-audio: add more quirks for DSD interfaces") Cc: Link: https://lore.kernel.org/r/20200430124755.15940-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 351ba214a9d3..848a4cc25bed 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1687,7 +1687,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ - case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ + case USB_ID(0x16d0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ -- cgit v1.2.3