From 9fe9efd6924c9a62ebb759025bb8927e398f51f7 Mon Sep 17 00:00:00 2001 From: Tomasz Figa Date: Wed, 14 Oct 2020 14:16:24 +0000 Subject: ASoC: Intel: kbl_rt5663_max98927: Fix kabylake_ssp_fixup function MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This is a copy of commit 5c5f1baee85a ("ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function") applied to the kbl_rt5663_max98927 board file. Original explanation of the change: kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The HW parameters are changed based on the codec DAI of the stream. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). 2 codecs on the SSP are dmic (capture) and speakers (playback). Based on the stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime. Fixes a boot crash on a HP Chromebook x2: [ 16.582225] BUG: kernel NULL pointer dereference, address: 0000000000000050 [ 16.582231] #PF: supervisor read access in kernel mode [ 16.582233] #PF: error_code(0x0000) - not-present page [ 16.582234] PGD 0 P4D 0 [ 16.582238] Oops: 0000 [#1] PREEMPT SMP PTI [ 16.582241] CPU: 0 PID: 1980 Comm: cras Tainted: G C 5.4.58 #1 [ 16.582243] Hardware name: HP Soraka/Soraka, BIOS Google_Soraka.10431.75.0 08/30/2018 [ 16.582247] RIP: 0010:kabylake_ssp_fixup+0x19/0xbb [snd_soc_kbl_rt5663_max98927] [ 16.582250] Code: c6 6f c5 80 c0 44 89 f2 31 c0 e8 3e c9 4c d6 eb de 0f 1f 44 00 00 55 48 89 e5 41 57 41 56 53 48 89 f3 48 8b 46 c8 48 8b 4e d0 <48> 8b 49 10 4c 8b 78 10 4c 8b 31 4c 89 f7 48 c7 c6 4b c2 80 c0 e8 [ 16.582252] RSP: 0000:ffffaf7e81e0b958 EFLAGS: 00010282 [ 16.582254] RAX: ffffffff96f13e0d RBX: ffffaf7e81e0ba00 RCX: 0000000000000040 [ 16.582256] RDX: ffffaf7e81e0ba00 RSI: ffffaf7e81e0ba00 RDI: ffffa3b208558028 [ 16.582258] RBP: ffffaf7e81e0b970 R08: ffffa3b203b54160 R09: ffffaf7e81e0ba00 [ 16.582259] R10: 0000000000000000 R11: ffffffffc080b345 R12: ffffa3b209fb6e00 [ 16.582261] R13: ffffa3b1b1a47838 R14: ffffa3b1e6197f28 R15: ffffaf7e81e0ba00 [ 16.582263] FS: 00007eb3f25aaf80(0000) GS:ffffa3b236a00000(0000) knlGS:0000000000000000 [ 16.582265] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 16.582267] CR2: 0000000000000050 CR3: 0000000246bc8006 CR4: 00000000003606f0 [ 16.582269] Call Trace: [ 16.582275] snd_soc_link_be_hw_params_fixup+0x21/0x68 [ 16.582278] snd_soc_dai_hw_params+0x25/0x94 [ 16.582282] soc_pcm_hw_params+0x2d8/0x583 [ 16.582288] dpcm_be_dai_hw_params+0x172/0x29e [ 16.582291] dpcm_fe_dai_hw_params+0x9f/0x12f [ 16.582295] snd_pcm_hw_params+0x137/0x41c [ 16.582298] snd_pcm_hw_params_user+0x3c/0x71 [ 16.582301] snd_pcm_common_ioctl+0x2c6/0x565 [ 16.582304] snd_pcm_ioctl+0x32/0x36 [ 16.582307] do_vfs_ioctl+0x506/0x783 [ 16.582311] ksys_ioctl+0x58/0x83 [ 16.582313] __x64_sys_ioctl+0x1a/0x1e [ 16.582316] do_syscall_64+0x54/0x7e [ 16.582319] entry_SYSCALL_64_after_hwframe+0x44/0xa9 [ 16.582322] RIP: 0033:0x7eb3f1886157 [ 16.582324] Code: 8a 66 90 48 8b 05 11 dd 2b 00 64 c7 00 26 00 00 00 48 c7 c0 ff ff ff ff c3 66 2e 0f 1f 84 00 00 00 00 00 b8 10 00 00 00 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 8b 0d e1 dc 2b 00 f7 d8 64 89 01 48 [ 16.582326] RSP: 002b:00007ffff7559818 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 [ 16.582329] RAX: ffffffffffffffda RBX: 00005acc9188b140 RCX: 00007eb3f1886157 [ 16.582330] RDX: 00007ffff7559940 RSI: 00000000c2604111 RDI: 000000000000001e [ 16.582332] RBP: 00007ffff7559840 R08: 0000000000000004 R09: 0000000000000000 [ 16.582333] R10: 0000000000000000 R11: 0000000000000246 R12: 000000000000bb80 [ 16.582335] R13: 00005acc91702e80 R14: 00007ffff7559940 R15: 00005acc91702e80 [ 16.582337] Modules linked in: rfcomm cmac algif_hash algif_skcipher af_alg uinput hid_google_hammer snd_soc_kbl_rt5663_max98927 snd_soc_hdac_hdmi snd_soc_dmic snd_soc_skl_ssp_clk snd_soc_skl snd_soc_sst_ipc snd_soc_sst_dsp snd_soc_hdac_hda snd_soc_acpi_intel_match snd_soc_acpi snd_hda_ext_core snd_intel_dspcfg snd_hda_codec snd_hwdep snd_hda_core ipu3_cio2 ipu3_imgu(C) videobuf2_v4l2 videobuf2_common videobuf2_dma_sg videobuf2_memops snd_soc_rt5663 snd_soc_max98927 snd_soc_rl6231 ov5670 ov13858 acpi_als v4l2_fwnode dw9714 fuse xt_MASQUERADE iio_trig_sysfs cros_ec_light_prox cros_ec_sensors cros_ec_sensors_core cros_ec_sensors_ring industrialio_triggered_buffer kfifo_buf industrialio cros_ec_sensorhub cdc_ether usbnet btusb btrtl btintel btbcm bluetooth ecdh_generic ecc lzo_rle lzo_compress iwlmvm zram iwl7000_mac80211 r8152 mii iwlwifi cfg80211 joydev [ 16.584243] gsmi: Log Shutdown Reason 0x03 [ 16.584246] CR2: 0000000000000050 [ 16.584248] ---[ end trace c8511d090c11edff ]--- Suggested-by: Łukasz Majczak Fixes: 2e5894d73789e ("ASoC: pcm: Add support for DAI multicodec") Signed-off-by: Tomasz Figa Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20201014141624.4143453-1-tfiga@chromium.org Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 39 ++++++++++++++++++++++------ 1 file changed, 31 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 3ea4602dfb3e..9a4b3d0973f6 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -401,17 +401,40 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; + + /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); @@ -421,7 +444,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; -- cgit v1.2.3 From 1849a3872f035494639201fdefb394425233647b Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 12 Oct 2020 11:50:05 +0200 Subject: ASoC: Intel: atom: Remove duplicate kconfigs SND_SST_IPC and its _PCI and _ACPI variants all target sound/soc/intel/atom solution alone. SND_SST_IPC is the core component, required for PCI and ACPI based atom platforms both. _PCI and _ACPI target Merrifield/Edison and Baytrial/Cherrytrail platforms respectively. On top of that, there is an equivalent set of configs targeting the same solution: - SND_SST_ATOM_HIFI2_PLATFORM (core) - SND_SST_ATOM_HIFI2_PLATFORM_PCI - SND_SST_ATOM_HIFI2_PLATFORM_ACPI As both sets do the same job - allow for granular platform selection - remove the duplicate set and rely on SND_SST_ATOM_HIFI2_PLATOFRM_XXX configs alone. Signed-off-by: Cezary Rojewski Acked-by: Hans de Goede Link: https://lore.kernel.org/r/20201012095005.29859-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 18 ------------------ sound/soc/intel/atom/Makefile | 2 +- sound/soc/intel/atom/sst/Makefile | 6 +++--- 3 files changed, 4 insertions(+), 22 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index d5bae5d1ab6f..a5b446d5af19 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -15,22 +15,6 @@ config SND_SOC_INTEL_SST_TOPLEVEL if SND_SOC_INTEL_SST_TOPLEVEL -config SND_SST_IPC - tristate - # This option controls the IPC core for HiFi2 platforms - -config SND_SST_IPC_PCI - tristate - select SND_SST_IPC - # This option controls the PCI-based IPC for HiFi2 platforms - # (Medfield, Merrifield). - -config SND_SST_IPC_ACPI - tristate - select SND_SST_IPC - # This option controls the ACPI-based IPC for HiFi2 platforms - # (Baytrail, Cherrytrail) - config SND_SOC_INTEL_SST tristate @@ -57,7 +41,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM config SND_SST_ATOM_HIFI2_PLATFORM_PCI tristate "PCI HiFi2 (Merrifield) Platforms" depends on X86 && PCI - select SND_SST_IPC_PCI select SND_SST_ATOM_HIFI2_PLATFORM help If you have a Intel Merrifield/Edison platform, then @@ -70,7 +53,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI depends on X86 && ACPI && PCI - select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH select IOSF_MBI diff --git a/sound/soc/intel/atom/Makefile b/sound/soc/intel/atom/Makefile index a9326d5ec44c..c66f03f5d8d6 100644 --- a/sound/soc/intel/atom/Makefile +++ b/sound/soc/intel/atom/Makefile @@ -6,4 +6,4 @@ snd-soc-sst-atom-hifi2-platform-objs := sst-mfld-platform-pcm.o \ obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-soc-sst-atom-hifi2-platform.o # DSP driver -obj-$(CONFIG_SND_SST_IPC) += sst/ +obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += sst/ diff --git a/sound/soc/intel/atom/sst/Makefile b/sound/soc/intel/atom/sst/Makefile index f17c905df3e2..5761d30a5f9d 100644 --- a/sound/soc/intel/atom/sst/Makefile +++ b/sound/soc/intel/atom/sst/Makefile @@ -3,6 +3,6 @@ snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_ snd-intel-sst-pci-objs += sst_pci.o snd-intel-sst-acpi-objs += sst_acpi.o -obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o -obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o -obj-$(CONFIG_SND_SST_IPC_ACPI) += snd-intel-sst-acpi.o +obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-intel-sst-core.o +obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_PCI) += snd-intel-sst-pci.o +obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) += snd-intel-sst-acpi.o -- cgit v1.2.3 From 1d159edf19542793851a04202e5b0dd548a9415c Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 12 Oct 2020 12:32:20 +0200 Subject: ASoC: Intel: catpt: Wake up device before configuring SSP port catpt_dai_pcm_new() invoked during new PCM runtime creation configures SSP by sending IPC to DSP firmware. For that to succeed device needs to be up and running. While components default probing behavior - snd_soc_catpt causing machine board module to load just after it - needs no changes, machine board's module may be unloaded and re-loaded at a different time e.g.: when catpt is already asleep. Wake device explicitly in catpt_dai_pcm_new() to ensure communication is established before sending any IPCs, enabling those advanced scenarios in the process. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20201012103221.30759-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/catpt/pcm.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index f78018c857b8..ba653ebea7d1 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -667,7 +667,17 @@ static int catpt_dai_pcm_new(struct snd_soc_pcm_runtime *rtm, break; } + /* see if this is a new configuration */ + if (!memcmp(&cdev->devfmt[devfmt.iface], &devfmt, sizeof(devfmt))) + return 0; + + pm_runtime_get_sync(cdev->dev); + ret = catpt_ipc_set_device_format(cdev, &devfmt); + + pm_runtime_mark_last_busy(cdev->dev); + pm_runtime_put_autosuspend(cdev->dev); + if (ret) return CATPT_IPC_ERROR(ret); -- cgit v1.2.3 From 3d53c6df4299134525ad9e197f480e89bc8b06af Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 12 Oct 2020 12:32:21 +0200 Subject: ASoC: Intel: catpt: Relax clock selection conditions Stress tests show that DSP may occasionally be late with signaling WAIT state when all pins are made use of simultaneously plus start/stop (pause) gets involved. While this isn't tied to standard audio scenarios where only System Pin (playback and capture) is involved, ensure user is not hindered when playing with more advanced scenarios. >From DSP perspective, clock acts as a resource: low clock equals less resources, high clock more resources. Relax clock selection procedure so only low -> high switch is allowed when awaiting WAIT signal times out. Once active stream count decreases, DSP will have more time internally to adjust thus low clock selection becomes possible again. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20201012103221.30759-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/catpt/dsp.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/catpt/dsp.c b/sound/soc/intel/catpt/dsp.c index 7d2968571951..9e807b941732 100644 --- a/sound/soc/intel/catpt/dsp.c +++ b/sound/soc/intel/catpt/dsp.c @@ -267,9 +267,12 @@ static int catpt_dsp_select_lpclock(struct catpt_dev *cdev, bool lp, bool waiti) reg, (reg & CATPT_ISD_DCPWM), 500, 10000); if (ret) { - dev_err(cdev->dev, "await WAITI timeout\n"); - mutex_unlock(&cdev->clk_mutex); - return ret; + dev_warn(cdev->dev, "await WAITI timeout\n"); + /* no signal - only high clock selection allowed */ + if (lp) { + mutex_unlock(&cdev->clk_mutex); + return 0; + } } } -- cgit v1.2.3 From eb5a558705c7f63d06b4ddd072898b1ca894e053 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Mon, 19 Oct 2020 12:47:24 +0800 Subject: ASoC: mediatek: mt8183-da7219: fix DAPM paths for rt1015 RT1015's output widget name is "SPO" instead of "Speaker". Fixes it to use the correct names. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20201019044724.1601476-1-tzungbi@google.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 31 +++++++++++++++++----- 1 file changed, 25 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 4d69ea31bfe4..9630637b8ab9 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -630,15 +630,34 @@ static struct snd_soc_codec_conf mt8183_da7219_rt1015_codec_conf[] = { }, }; +static const struct snd_kcontrol_new mt8183_da7219_rt1015_snd_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const +struct snd_soc_dapm_widget mt8183_da7219_rt1015_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_PINCTRL("TDM_OUT_PINCTRL", + "aud_tdm_out_on", "aud_tdm_out_off"), +}; + +static const struct snd_soc_dapm_route mt8183_da7219_rt1015_dapm_routes[] = { + {"Left Spk", NULL, "Left SPO"}, + {"Right Spk", NULL, "Right SPO"}, + {"I2S Playback", NULL, "TDM_OUT_PINCTRL"}, +}; + static struct snd_soc_card mt8183_da7219_rt1015_card = { .name = "mt8183_da7219_rt1015", .owner = THIS_MODULE, - .controls = mt8183_da7219_max98357_snd_controls, - .num_controls = ARRAY_SIZE(mt8183_da7219_max98357_snd_controls), - .dapm_widgets = mt8183_da7219_max98357_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_max98357_dapm_widgets), - .dapm_routes = mt8183_da7219_max98357_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_max98357_dapm_routes), + .controls = mt8183_da7219_rt1015_snd_controls, + .num_controls = ARRAY_SIZE(mt8183_da7219_rt1015_snd_controls), + .dapm_widgets = mt8183_da7219_rt1015_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_widgets), + .dapm_routes = mt8183_da7219_rt1015_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_routes), .dai_link = mt8183_da7219_dai_links, .num_links = ARRAY_SIZE(mt8183_da7219_dai_links), .aux_dev = &mt8183_da7219_max98357_headset_dev, -- cgit v1.2.3 From f47d0742515748162d3fc35f04331c5b81c0ed47 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 22 Oct 2020 14:05:18 +0100 Subject: ASoC: codecs: wsa881x: add missing stream rates and format Add missing supported rates and formats for the stream, without which attempt to do playback will fail to find any matching rates/format. Fixes: a0aab9e1404a ("ASoC: codecs: add wsa881x amplifier support") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20201022130518.31723-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa881x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index d39d479e2378..5456124457a7 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1026,6 +1026,8 @@ static struct snd_soc_dai_driver wsa881x_dais[] = { .id = 0, .playback = { .stream_name = "SPKR Playback", + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, .rate_max = 48000, .rate_min = 48000, .channels_min = 1, -- cgit v1.2.3 From 3f48b6eba15ea342ef4cb420b580f5ed6605669f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 23 Oct 2020 10:58:49 +0100 Subject: ASoC: qcom: sdm845: set driver name correctly With the current state of code, we would endup with something like below in /proc/asound/cards for 2 machines based on this driver. Machine 1: 0 [DB845c ]: DB845c - DB845c DB845c Machine 2: 0 [LenovoYOGAC6301]: Lenovo-YOGA-C63 - Lenovo-YOGA-C630-13Q50 LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216 This is not very UCM friendly both w.r.t to common up configs and card identification, and UCM2 became totally not usefull with just one ucm sdm845.conf for two machines which have different setups w.r.t HDMI and other dais. Reasons for such thing is partly because Qualcomm machine drivers never cared to set driver_name. This patch sets up driver name for the this driver to sort out the UCM integration issues! after this patch contents of /proc/asound/cards: Machine 1: 0 [DB845c ]: sdm845 - DB845c DB845c Machine 2: 0 [LenovoYOGAC6301]: sdm845 - Lenovo-YOGA-C630-13Q50 LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216 with this its possible to align with what UCM2 expects and we can have sdm845/DB845.conf sdm845/LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216.conf ... for board variants. This should scale much better! Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20201023095849.22894-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/sdm845.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index ab1bf23c21a6..6c2760e27ea6 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -17,6 +17,7 @@ #include "qdsp6/q6afe.h" #include "../codecs/rt5663.h" +#define DRIVER_NAME "sdm845" #define DEFAULT_SAMPLE_RATE_48K 48000 #define DEFAULT_MCLK_RATE 24576000 #define TDM_BCLK_RATE 6144000 @@ -552,6 +553,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) if (!data) return -ENOMEM; + card->driver_name = DRIVER_NAME; card->dapm_widgets = sdm845_snd_widgets; card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; -- cgit v1.2.3 From 20afe581c9b980848ad097c4d54dde9bec7593ef Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Tue, 20 Oct 2020 17:01:09 +0200 Subject: ASoC: cs42l51: manage mclk shutdown delay A delay must be introduced before the shutdown down of the mclk, as stated in CS42L51 datasheet. Otherwise the codec may produce some noise after the end of DAPM power down sequence. The delay between DAC and CLOCK_SUPPLY widgets is too short. Add a delay in mclk shutdown request to manage the shutdown delay explicitly. From experiments, at least 10ms delay is necessary. Set delay to 20ms as recommended in Documentation/timers/timers-howto.rst when using msleep(). Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20201020150109.482-1-olivier.moysan@st.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 097c4e8d9950..c61b17dc2af8 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -254,8 +254,28 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = { &cs42l51_adcr_mux_controls), }; +static int mclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + struct cs42l51_private *cs42l51 = snd_soc_component_get_drvdata(comp); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return clk_prepare_enable(cs42l51->mclk_handle); + case SND_SOC_DAPM_POST_PMD: + /* Delay mclk shutdown to fulfill power-down sequence requirements */ + msleep(20); + clk_disable_unprepare(cs42l51->mclk_handle); + break; + } + + return 0; +} + static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = { - SND_SOC_DAPM_CLOCK_SUPPLY("MCLK") + SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0, mclk_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route cs42l51_routes[] = { -- cgit v1.2.3 From 6e5329c6e6032cd997400b43b8299f607a61883e Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 21 Oct 2020 21:24:19 +0300 Subject: ASoC: SOF: loader: handle all SOF_IPC_EXT types Do not emit a warning for extended firmware header fields that are not used by kernel. This creates unnecessary noise to kernel logs like: sof-audio-pci 0000:00:1f.3: warning: unknown ext header type 3 size 0x1c sof-audio-pci 0000:00:1f.3: warning: unknown ext header type 4 size 0x10 Signed-off-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Guennadi Liakhovetski Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20201021182419.1160391-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/loader.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 68ed454f7ddf..ba9ed66f98bc 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -118,6 +118,11 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) case SOF_IPC_EXT_CC_INFO: ret = get_cc_info(sdev, ext_hdr); break; + case SOF_IPC_EXT_UNUSED: + case SOF_IPC_EXT_PROBE_INFO: + case SOF_IPC_EXT_USER_ABI_INFO: + /* They are supported but we don't do anything here */ + break; default: dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n", ext_hdr->type, ext_hdr->hdr.size); -- cgit v1.2.3 From f7b6603c666798a1f8379e692d11d500885f32d8 Mon Sep 17 00:00:00 2001 From: Mauro Carvalho Chehab Date: Fri, 23 Oct 2020 18:33:35 +0200 Subject: ALSA: fix kernel-doc markups Kernel-doc markups should use this format: identifier - description There is a common comment marked, instead, with kernel-doc notation. Some identifiers have different names between their prototypes and the kernel-doc markup. Signed-off-by: Mauro Carvalho Chehab Acked-by: Mark Brown Link: https://lore.kernel.org/r/535182d6f55d7a7de293dda9676df68f5f60afc6.1603469755.git.mchehab+huawei@kernel.org Signed-off-by: Takashi Iwai --- include/sound/core.h | 3 ++- include/sound/pcm.h | 4 ++-- include/uapi/sound/compress_offload.h | 2 +- sound/core/control.c | 4 ++-- sound/core/pcm_dmaengine.c | 3 ++- sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 4 ++-- sound/soc/soc-core.c | 2 +- sound/soc/soc-dapm.c | 2 +- 9 files changed, 14 insertions(+), 12 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 381a010a1bd4..0462c577d7a3 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -332,7 +332,8 @@ void __snd_printk(unsigned int level, const char *file, int line, #define snd_BUG() WARN(1, "BUG?\n") /** - * Suppress high rates of output when CONFIG_SND_DEBUG is enabled. + * snd_printd_ratelimit - Suppress high rates of output when + * CONFIG_SND_DEBUG is enabled. */ #define snd_printd_ratelimit() printk_ratelimit() diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2ba5df2c9e23..2336bf9243e1 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1284,8 +1284,8 @@ snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs) } /** - * snd_pcm_sgbuf_chunk_size - Compute the max size that fits within the contig. - * page from the given size + * snd_pcm_sgbuf_get_chunk_size - Compute the max size that fits within the + * contig. page from the given size * @substream: PCM substream * @ofs: byte offset * @size: byte size to examine diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 7184265c0b0d..9555f31c8425 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -144,7 +144,7 @@ struct snd_compr_metadata { __u32 value[8]; } __attribute__((packed, aligned(4))); -/** +/* * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec diff --git a/sound/core/control.c b/sound/core/control.c index 421ddc76f264..4373de42a5a0 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1925,8 +1925,8 @@ EXPORT_SYMBOL(snd_ctl_unregister_ioctl); #ifdef CONFIG_COMPAT /** - * snd_ctl_unregister_ioctl - de-register the device-specific compat 32bit - * control-ioctls + * snd_ctl_unregister_ioctl_compat - de-register the device-specific compat + * 32bit control-ioctls * @fcn: ioctl callback function to unregister */ int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 4d059ff2b2e4..4d0e8fe535a1 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -356,7 +356,8 @@ int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close); /** - * snd_dmaengine_pcm_release_chan_close - Close a dmaengine based PCM substream and release channel + * snd_dmaengine_pcm_close_release_chan - Close a dmaengine based PCM + * substream and release channel * @substream: PCM substream * * Releases the DMA channel associated with the PCM substream. diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index d531e1bc2b81..bda3514c7b2d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -490,7 +490,7 @@ void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, EXPORT_SYMBOL(snd_pcm_set_ops); /** - * snd_pcm_sync - set the PCM sync id + * snd_pcm_set_sync - set the PCM sync id * @substream: the pcm substream * * Sets the PCM sync identifier for the card. diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 9e0b2d73faf6..47b155a49226 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -112,7 +112,7 @@ void snd_pcm_stream_lock(struct snd_pcm_substream *substream) EXPORT_SYMBOL_GPL(snd_pcm_stream_lock); /** - * snd_pcm_stream_lock - Unlock the PCM stream + * snd_pcm_stream_unlock - Unlock the PCM stream * @substream: PCM substream * * This unlocks the PCM stream that has been locked via snd_pcm_stream_lock(). @@ -595,7 +595,7 @@ static void snd_pcm_sync_stop(struct snd_pcm_substream *substream) } /** - * snd_pcm_hw_param_choose - choose a configuration defined by @params + * snd_pcm_hw_params_choose - choose a configuration defined by @params * @pcm: PCM instance * @params: the hw_params instance * diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ea3986a46c12..05a085f6dc7c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2341,7 +2341,7 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, } /** - * snd_soc_unregister_dai - Unregister DAIs from the ASoC core + * snd_soc_unregister_dais - Unregister DAIs from the ASoC core * * @component: The component for which the DAIs should be unregistered */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 980f2c330b87..7f87b449f950 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1276,7 +1276,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, } /** - * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * snd_soc_dapm_dai_get_connected_widgets - query audio path and it's widgets. * @dai: the soc DAI. * @stream: stream direction. * @list: list of active widgets for this stream. -- cgit v1.2.3 From 2a6eca16f376f6b83aaf73c57f0b6547907a5ed3 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 26 Oct 2020 17:52:18 +0100 Subject: ALSA: make snd_kcontrol_new name a normal string When building with W=2, there are lots of warnings about the snd_kcontrol_new name field being an array of 'unsigned char' but initialized to a string: include/sound/soc.h:93:48: warning: pointer targets in initialization of 'const unsigned char *' from 'char *' differ in signedness [-Wpointer-sign] Make it a regular 'char *' to avoid flooding the build log with this. Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Signed-off-by: Arnd Bergmann Link: https://lore.kernel.org/r/20201026165715.3723704-1-arnd@kernel.org Signed-off-by: Takashi Iwai --- include/sound/control.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/control.h b/include/sound/control.h index e128cff10dfa..77d9fa10812d 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -42,7 +42,7 @@ struct snd_kcontrol_new { snd_ctl_elem_iface_t iface; /* interface identifier */ unsigned int device; /* device/client number */ unsigned int subdevice; /* subdevice (substream) number */ - const unsigned char *name; /* ASCII name of item */ + const char *name; /* ASCII name of item */ unsigned int index; /* index of item */ unsigned int access; /* access rights */ unsigned int count; /* count of same elements */ -- cgit v1.2.3 From f15cfca818d756dd1c9492530091dfd583359db3 Mon Sep 17 00:00:00 2001 From: Keith Winstein Date: Sun, 25 Oct 2020 22:05:47 -0700 Subject: ALSA: usb-audio: Add implicit feedback quirk for Zoom UAC-2 The Zoom UAC-2 USB audio interface provides an async playback endpoint ("1 OUT (ASYNC)") and capture endpoint ("2 IN (ASYNC)"), both with 2-channel S32_LE in 44.1, 48, 88.2, 96, 176.4, or 192 kilosamples/s. The device provides explicit feedback to adjust the host's playback rate, but the feedback appears unstable and biased relative to the device's capture rate. "alsaloop -t 1000" experiences playback underruns and tries to resample the captured audio to match the varying playback rate. Forcing the kernel to use implicit feedback appears to produce more stable results. This causes the host to transmit one playback sample for each capture sample received. (Zoom North America has been notified of this change.) Signed-off-by: Keith Winstein Tested-by: Keith Winstein Cc: BugLink: https://lore.kernel.org/r/20201027071841.GA164525@trolley.csail.mit.edu Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b401ee894e1b..31b1edf53c41 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -352,6 +352,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; + case USB_ID(0x1686, 0xf029): /* Zoom UAC-2 */ + ep = 0x82; + ifnum = 2; + goto add_sync_ep_from_ifnum; case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */ case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */ ep = 0x81; -- cgit v1.2.3 From 8a8de09cb2adc119104f35044d1a840dd47aa9d8 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 27 Oct 2020 16:46:38 +0800 Subject: ALSA: hda/realtek - Fixed HP headset Mic can't be detected System boot with plugged headset. It will not detect headset Mic. It will happen on cold boot restart resume state. Quirk by SSID change to quirk by pin verb. Fixes: 13468bfa8c58 ("ALSA: hda/realtek - set mic to auto detect on a HP AIO machine") Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/f42ae1ede1cf47029ae2bef1a42caf03@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 54 ++++++++++++++++++++++++++++++++++--------- 1 file changed, 43 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f2398721ac1e..4f0ea8013bf6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6008,6 +6008,27 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } +static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) +{ + switch (codec->core.vendor_id) { + case 0x10ec0274: + case 0x10ec0294: + case 0x10ec0225: + case 0x10ec0295: + case 0x10ec0299: + alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */ + alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15); + break; + case 0x10ec0235: + case 0x10ec0236: + case 0x10ec0255: + case 0x10ec0256: + alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ + alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); + break; + } +} + static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6018,16 +6039,7 @@ static void alc295_fixup_chromebook(struct hda_codec *codec, spec->ultra_low_power = true; break; case HDA_FIXUP_ACT_INIT: - switch (codec->core.vendor_id) { - case 0x10ec0295: - alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */ - alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15); - break; - case 0x10ec0236: - alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ - alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); - break; - } + alc_combo_jack_hp_jd_restart(codec); break; } } @@ -6083,6 +6095,16 @@ static void alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec, alc_write_coef_idx(codec, 0x65, 0x0); } +static void alc274_fixup_hp_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + switch (action) { + case HDA_FIXUP_ACT_INIT: + alc_combo_jack_hp_jd_restart(codec); + break; + } +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -6277,6 +6299,7 @@ enum { ALC256_FIXUP_INTEL_NUC8_RUGGED, ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, + ALC274_FIXUP_HP_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -7664,6 +7687,12 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC274_FIXUP_HP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc274_fixup_hp_headset_mic, + .chained = true, + .chain_id = ALC274_FIXUP_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7815,7 +7844,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT), - SND_PCI_QUIRK(0x103c, 0x874e, "HP", ALC274_FIXUP_HP_MIC), SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED), @@ -8339,6 +8367,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1a, 0x90a70130}, {0x1b, 0x90170110}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0274, 0x103c, "HP", ALC274_FIXUP_HP_HEADSET_MIC, + {0x17, 0x90170110}, + {0x19, 0x03a11030}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x14, 0x90170110}, -- cgit v1.2.3 From 215a22ed31a1332075866eca07744d442367c04b Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Tue, 27 Oct 2020 21:00:36 +0800 Subject: ALSA: hda: Refactor codec PM to use direct-complete optimization Upon system resume, hda_codec_pm_resume() uses hda_codec_force_resume() to resume the codec. However, pm_runtime_force_resume() won't really resume the codec because of pm_runtime_need_not_resume() check. Hence, hda_codec_force_resume() schedules a jackpoll work, which is to really power up the codec. Instead of doing that, we can use direct-complete to make the PM flow more straightforward, and keep codec always suspended through system PM flow if conditions are met. On system suspend, PM core will decide what to do based on hda_codec_pm_prepare(): - If codec is not runtime-suspended, PM core will suspend and resume the device as normal. - If codec is runtime-suspended, PM core will try to keep it suspended. If it's still suspended after system resume, we use hda_codec_pm_complete() to resume codec if it's needed. Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20201027130038.16463-2-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 45 +++++++++++++++++++++++++++++---------------- 1 file changed, 29 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a356c21edb90..4bb58e8b08a8 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2934,7 +2934,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) snd_hdac_leave_pm(&codec->core); } -static int hda_codec_runtime_suspend(struct device *dev) +static int hda_codec_suspend(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); unsigned int state; @@ -2953,7 +2953,7 @@ static int hda_codec_runtime_suspend(struct device *dev) return 0; } -static int hda_codec_runtime_resume(struct device *dev) +static int hda_codec_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); @@ -2967,57 +2967,70 @@ static int hda_codec_runtime_resume(struct device *dev) pm_runtime_mark_last_busy(dev); return 0; } + +static int hda_codec_runtime_suspend(struct device *dev) +{ + return hda_codec_suspend(dev); +} + +static int hda_codec_runtime_resume(struct device *dev) +{ + return hda_codec_resume(dev); +} + #endif /* CONFIG_PM */ #ifdef CONFIG_PM_SLEEP -static int hda_codec_force_resume(struct device *dev) +static int hda_codec_pm_prepare(struct device *dev) +{ + return pm_runtime_suspended(dev); +} + +static void hda_codec_pm_complete(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - int ret; - ret = pm_runtime_force_resume(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; + if (pm_runtime_suspended(dev) && (codec->jackpoll_interval || + hda_codec_need_resume(codec) || codec->forced_resume)) + pm_request_resume(dev); } static int hda_codec_pm_suspend(struct device *dev) { dev->power.power_state = PMSG_SUSPEND; - return pm_runtime_force_suspend(dev); + return hda_codec_suspend(dev); } static int hda_codec_pm_resume(struct device *dev) { dev->power.power_state = PMSG_RESUME; - return hda_codec_force_resume(dev); + return hda_codec_resume(dev); } static int hda_codec_pm_freeze(struct device *dev) { dev->power.power_state = PMSG_FREEZE; - return pm_runtime_force_suspend(dev); + return hda_codec_suspend(dev); } static int hda_codec_pm_thaw(struct device *dev) { dev->power.power_state = PMSG_THAW; - return hda_codec_force_resume(dev); + return hda_codec_resume(dev); } static int hda_codec_pm_restore(struct device *dev) { dev->power.power_state = PMSG_RESTORE; - return hda_codec_force_resume(dev); + return hda_codec_resume(dev); } #endif /* CONFIG_PM_SLEEP */ /* referred in hda_bind.c */ const struct dev_pm_ops hda_codec_driver_pm = { #ifdef CONFIG_PM_SLEEP + .prepare = hda_codec_pm_prepare, + .complete = hda_codec_pm_complete, .suspend = hda_codec_pm_suspend, .resume = hda_codec_pm_resume, .freeze = hda_codec_pm_freeze, -- cgit v1.2.3 From f5dac54d9d93826a776dffc848df76746f7135bb Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Tue, 27 Oct 2020 21:00:37 +0800 Subject: ALSA: hda: Separate runtime and system suspend Both pm_runtime_force_suspend() and pm_runtime_force_resume() have some implicit checks, so it can make code flow more straightforward if we separate runtime and system suspend callbacks. High Definition Audio Specification, 4.5.9.3 Codec Wake From System S3 states that codec can wake the system up from S3 if WAKEEN is toggled. Since HDA controller has different wakeup settings for runtime and system susend, we also need to explicitly disable direct-complete which can be enabled automatically by PCI core. In addition to that, avoid waking up codec if runtime resume is for system suspend, to not break direct-complete for codecs. While at it, also remove AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP, as the original bug commit a6630529aecb ("ALSA: hda: Workaround for spurious wakeups on some Intel platforms") solves doesn't happen with this patch. Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20201027130038.16463-3-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.h | 3 +- sound/pci/hda/hda_intel.c | 62 +++++++++++++++++++++++------------------- 2 files changed, 36 insertions(+), 29 deletions(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index be63ead8161f..68f9668788ea 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -41,7 +41,7 @@ /* 24 unused */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -#define AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP (1 << 27) /* Workaround for spurious wakeups after suspend */ +/* 27 unused */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ @@ -143,6 +143,7 @@ struct azx { unsigned int align_buffer_size:1; unsigned int region_requested:1; unsigned int disabled:1; /* disabled by vga_switcheroo */ + unsigned int pm_prepared:1; /* GTS present */ unsigned int gts_present:1; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 749b88090970..be83f1747c5f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -297,8 +297,7 @@ enum { /* PCH for HSW/BDW; with runtime PM */ /* no i915 binding for this as HSW/BDW has another controller for HDMI */ #define AZX_DCAPS_INTEL_PCH \ - (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ - AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME) /* HSW HDMI */ #define AZX_DCAPS_INTEL_HASWELL \ @@ -985,7 +984,7 @@ static void __azx_runtime_suspend(struct azx *chip) display_power(chip, false); } -static void __azx_runtime_resume(struct azx *chip, bool from_rt) +static void __azx_runtime_resume(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); @@ -1002,7 +1001,8 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt) azx_init_pci(chip); hda_intel_init_chip(chip, true); - if (from_rt) { + /* Avoid codec resume if runtime resume is for system suspend */ + if (!chip->pm_prepared) { list_for_each_codec(codec, &chip->bus) { if (codec->relaxed_resume) continue; @@ -1018,6 +1018,29 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt) } #ifdef CONFIG_PM_SLEEP +static int azx_prepare(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip; + + chip = card->private_data; + chip->pm_prepared = 1; + + /* HDA controller always requires different WAKEEN for runtime suspend + * and system suspend, so don't use direct-complete here. + */ + return 0; +} + +static void azx_complete(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip; + + chip = card->private_data; + chip->pm_prepared = 0; +} + static int azx_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -1029,15 +1052,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - /* An ugly workaround: direct call of __azx_runtime_suspend() and - * __azx_runtime_resume() for old Intel platforms that suffer from - * spurious wakeups after S3 suspend - */ - if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) - __azx_runtime_suspend(chip); - else - pm_runtime_force_suspend(dev); + __azx_runtime_suspend(chip); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1066,11 +1081,7 @@ static int azx_resume(struct device *dev) if (azx_acquire_irq(chip, 1) < 0) return -EIO; - if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) - __azx_runtime_resume(chip, false); - else - pm_runtime_force_resume(dev); - snd_power_change_state(card, SNDRV_CTL_POWER_D0); + __azx_runtime_resume(chip); trace_azx_resume(chip); return 0; @@ -1118,10 +1129,7 @@ static int azx_runtime_suspend(struct device *dev) chip = card->private_data; /* enable controller wake up event */ - if (snd_power_get_state(card) == SNDRV_CTL_POWER_D0) { - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | - STATESTS_INT_MASK); - } + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | STATESTS_INT_MASK); __azx_runtime_suspend(chip); trace_azx_runtime_suspend(chip); @@ -1132,18 +1140,14 @@ 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; - __azx_runtime_resume(chip, from_rt); + __azx_runtime_resume(chip); /* disable controller Wake Up event*/ - if (from_rt) { - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & - ~STATESTS_INT_MASK); - } + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & ~STATESTS_INT_MASK); trace_azx_runtime_resume(chip); return 0; @@ -1177,6 +1181,8 @@ static int azx_runtime_idle(struct device *dev) static const struct dev_pm_ops azx_pm = { SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) #ifdef CONFIG_PM_SLEEP + .prepare = azx_prepare, + .complete = azx_complete, .freeze_noirq = azx_freeze_noirq, .thaw_noirq = azx_thaw_noirq, #endif -- cgit v1.2.3 From 9fc149c3bce7bdbb94948a8e6bd025e3b3538603 Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Tue, 27 Oct 2020 21:00:38 +0800 Subject: ALSA: hda: Reinstate runtime_allow() for all hda controllers The broken jack detection should be fixed by commit a6e7d0a4bdb0 ("ALSA: hda: fix jack detection with Realtek codecs when in D3"), let's try enabling runtime PM by default again. Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20201027130038.16463-4-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index be83f1747c5f..d539f52009a1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2362,6 +2362,7 @@ 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 fc0522bbe02fa4beb95c0514ace66b585616f111 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 28 Oct 2020 15:43:39 +0000 Subject: ASoC: codecs: wcd934x: Set digital gain range correctly digital gain range is -84dB min to 40dB max, however this was not correctly specified in the range. Fix this by with correct range! Fixes: 1cde8b822332 ("ASoC: wcd934x: add basic controls") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20201028154340.17090-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd934x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 35697b072367..40f682f5dab8 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -551,7 +551,7 @@ struct wcd_iir_filter_ctl { struct soc_bytes_ext bytes_ext; }; -static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0); +static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400); static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1); static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1); static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0); -- cgit v1.2.3 From 6d6bc54ab4f2404d46078abc04bf4dee4db01def Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 28 Oct 2020 15:43:40 +0000 Subject: ASoC: codecs: wcd9335: Set digital gain range correctly digital gain range is -84dB min to 40dB max, however this was not correctly specified in the range. Fix this by with correct range! Fixes: 8c4f021d806a ("ASoC: wcd9335: add basic controls") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20201028154340.17090-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd9335.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index f2d9d52ee171..4d2b1ec7c03b 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -618,7 +618,7 @@ static const char * const sb_tx8_mux_text[] = { "ZERO", "RX_MIX_TX8", "DEC8", "DEC8_192" }; -static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0); +static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400); static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1); static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1); static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0); -- cgit v1.2.3 From cf9d21984da2c8e852320d12c03ddb7d11760a32 Mon Sep 17 00:00:00 2001 From: V Sujith Kumar Reddy Date: Tue, 27 Oct 2020 17:04:34 +0530 Subject: ASoC: qcom: lpass-sc7180: Fix MI2S bitwidth field bit positions Update SC7180 lpass_variant structure with proper I2S bitwidth field bit positions, as bitwidth denotes 0 to 1 bits, but previously used only 0 bit. Signed-off-by: V Sujith Kumar Reddy Signed-off-by: Srinivasa Rao Mandadapu Link: https://lore.kernel.org/r/1603798474-4897-1-git-send-email-srivasam@codeaurora.org Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-sc7180.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index c6292f9e613f..bc998d501600 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -188,7 +188,7 @@ static struct lpass_variant sc7180_data = { .micmode = REG_FIELD_ID(0x1000, 4, 8, 3, 0x1000), .micmono = REG_FIELD_ID(0x1000, 3, 3, 3, 0x1000), .wssrc = REG_FIELD_ID(0x1000, 2, 2, 3, 0x1000), - .bitwidth = REG_FIELD_ID(0x1000, 0, 0, 3, 0x1000), + .bitwidth = REG_FIELD_ID(0x1000, 0, 1, 3, 0x1000), .rdma_dyncclk = REG_FIELD_ID(0xC000, 21, 21, 5, 0x1000), .rdma_bursten = REG_FIELD_ID(0xC000, 20, 20, 5, 0x1000), -- cgit v1.2.3 From 6ec6c3693a389841d8ca952072aea8020da54ef4 Mon Sep 17 00:00:00 2001 From: V Sujith Kumar Reddy Date: Mon, 19 Oct 2020 14:36:03 +0530 Subject: ASoC: qcom: lpass-cpu: Fix clock disable failure Disable MI2S bit clock from PAUSE/STOP/SUSPEND usecase instead of shutdown time. Acheive this by invoking clk_disable API from cpu daiops trigger instead of cpu daiops shutdown. Change non-atomic API "clk_prepare_enable" to atomic API "clk_enable" in trigger, as trigger is being called from atomic context. Fixes: 7e6799d8f87d ("ASoC: qcom: lpass-cpu: Enable MI2S BCLK and LRCLK together") Signed-off-by: V Sujith Kumar Reddy Signed-off-by: Srinivasa Rao Mandadapu Link: https://lore.kernel.org/r/1603098363-9251-1-git-send-email-srivasam@codeaurora.org Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index ba2aca301a9b..9d17c87445a9 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -80,6 +80,12 @@ static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream, dev_err(dai->dev, "error in enabling mi2s osr clk: %d\n", ret); return ret; } + ret = clk_prepare(drvdata->mi2s_bit_clk[dai->driver->id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + return ret; + } return 0; } @@ -88,9 +94,8 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); - clk_disable_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); - clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, @@ -303,10 +308,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); - ret = clk_prepare_enable(drvdata->mi2s_bit_clk[id]); + ret = clk_enable(drvdata->mi2s_bit_clk[id]); if (ret) { dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); - clk_disable_unprepare(drvdata->mi2s_osr_clk[id]); + clk_disable(drvdata->mi2s_osr_clk[id]); return ret; } @@ -324,6 +329,7 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, if (ret) dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); break; } -- cgit v1.2.3 From 158e1886b6262c1d1c96a18c85fac5219b8bf804 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 3 Nov 2020 13:18:07 +0300 Subject: ALSA: hda: prevent undefined shift in snd_hdac_ext_bus_get_link() This is harmless, but the "addr" comes from the user and it could lead to a negative shift or to shift wrapping if it's too high. Fixes: 0b00a5615dc4 ("ALSA: hdac_ext: add hdac extended controller") Signed-off-by: Dan Carpenter Link: https://lore.kernel.org/r/20201103101807.GC1127762@mwanda Signed-off-by: Takashi Iwai --- sound/hda/ext/hdac_ext_controller.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 4d060d5b1db6..b0c0ef824d7d 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -148,6 +148,8 @@ struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus, return NULL; if (bus->idx != bus_idx) return NULL; + if (addr < 0 || addr > 31) + return NULL; list_for_each_entry(hlink, &bus->hlink_list, list) { for (i = 0; i < HDA_MAX_CODECS; i++) { -- cgit v1.2.3 From ef9ce66fab959c66d270bbee7ca79b92ee957893 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 3 Nov 2020 15:40:35 +0800 Subject: ALSA: hda/realtek - Enable headphone for ASUS TM420 ASUS TM420 had depop circuit for headphone. It need to turn on by COEF bit. [ fixed the missing enum definition by tiwai ] Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/3d6177d7023b4783bf2793861c577ada@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4f0ea8013bf6..6899089d132e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6300,6 +6300,7 @@ enum { ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, ALC274_FIXUP_HP_HEADSET_MIC, + ALC256_FIXUP_ASUS_HPE, }; static const struct hda_fixup alc269_fixups[] = { @@ -7693,6 +7694,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC274_FIXUP_HP_MIC }, + [ALC256_FIXUP_ASUS_HPE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Set EAPD high */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x0f }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x7778 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7876,6 +7888,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), -- cgit v1.2.3 From 07815a2b3501adeaae6384a25b9c4a9c81dae59f Mon Sep 17 00:00:00 2001 From: Artem Lapkin Date: Tue, 3 Nov 2020 18:08:09 +0800 Subject: ALSA: usb-audio: add usb vendor id as DSD-capable for Khadas devices Khadas audio devices ( USB_ID_VENDOR 0x3353 ) have DSD-capable implementations from XMOS need add new usb vendor id for recognition Signed-off-by: Artem Lapkin Cc: Link: https://lore.kernel.org/r/20201103103311.5435-1-art@khadas.com Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b4fa80ef730d..c989ad8052ae 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1800,6 +1800,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x278b: /* Rotel? */ case 0x292b: /* Gustard/Ess based devices */ case 0x2ab6: /* T+A devices */ + case 0x3353: /* Khadas devices */ case 0x3842: /* EVGA */ case 0xc502: /* HiBy devices */ if (fp->dsd_raw) -- cgit v1.2.3 From 26201ddc1373c99b2a67c5774da2f0eecd749b93 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 4 Nov 2020 22:37:05 +1030 Subject: ALSA: usb-audio: Add implicit feedback quirk for MODX This patch fixes audio distortion on playback for the Yamaha MODX. Signed-off-by: Geoffrey D. Bennett Tested-by: Frank Slotta Cc: Link: https://lore.kernel.org/r/20201104120705.GA19126@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 31b1edf53c41..7227ccb7e57e 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -345,6 +345,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 2; goto add_sync_ep_from_ifnum; case USB_ID(0x2466, 0x8003): /* Fractal Audio Axe-Fx II */ + case USB_ID(0x0499, 0x172a): /* Yamaha MODX */ ep = 0x86; ifnum = 2; goto add_sync_ep_from_ifnum; -- cgit v1.2.3 From f9d7c6eb23f7e55e7a0ca5451da06909bdfdd0e4 Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu Date: Wed, 4 Nov 2020 17:57:38 +0200 Subject: ASoC: mchp-spdiftx: Do not set Validity bit(s) The Validity bits (bit 28) must not be set in order to have the samples valid. Some controllers look for this bit and ignore the samples if it is set. Fixes: 06ca24e98e6b ("ASoC: mchp-spdiftx: add driver for S/PDIF TX Controller") Signed-off-by: Codrin Ciubotariu Link: https://lore.kernel.org/r/20201104155738.68403-1-codrin.ciubotariu@microchip.com Signed-off-by: Mark Brown --- sound/soc/atmel/mchp-spdiftx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index 82c1eecd2528..3bd350afb743 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -487,7 +487,6 @@ static int mchp_spdiftx_hw_params(struct snd_pcm_substream *substream, } mchp_spdiftx_channel_status_write(dev); spin_unlock_irqrestore(&ctrl->lock, flags); - mr |= SPDIFTX_MR_VALID1 | SPDIFTX_MR_VALID2; if (dev->gclk_enabled) { clk_disable_unprepare(dev->gclk); -- cgit v1.2.3 From 0938ecae432e7ac8b01080c35dd81d50a1e43033 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 4 Nov 2020 22:27:17 +1030 Subject: ALSA: usb-audio: Add implicit feedback quirk for Qu-16 This patch fixes audio distortion on playback for the Allen&Heath Qu-16. Signed-off-by: Geoffrey D. Bennett Cc: Link: https://lore.kernel.org/r/20201104115717.GA19046@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 1 + 1 file c