- 06 Aug, 2020 4 commits
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Mirko Dietrich authored
Adds an entry for Creative USB X-Fi to the rc_config array in mixer_quirks.c to allow use of volume knob on the device. Adds support for newer X-Fi Pro card, known as "Model No. SB1095" with USB ID "041e:3263" Signed-off-by:
Mirko Dietrich <buzz@l4m1.de> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200806124850.20334-1-buzz@l4m1.deSigned-off-by:
Takashi Iwai <tiwai@suse.de>
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Colin Ian King authored
There is a spelling mistake in a usb_audio_dbg debug message. Also replace "param" with "parameter". Fix these. Signed-off-by:
Colin Ian King <colin.king@canonical.com> Link: https://lore.kernel.org/r/20200806105134.46447-1-colin.king@canonical.comSigned-off-by:
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Randy Dunlap authored
Drop duplicated words in sound/pci/. {and, the, at} Signed-off-by:Randy Dunlap <rdunlap@infradead.org> Link: https://lore.kernel.org/r/20200806021926.32418-1-rdunlap@infradead.orgSigned-off-by:
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Randy Dunlap authored
Drop duplicated words in sound/isa/. {be, bit} Signed-off-by:
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- 05 Aug, 2020 4 commits
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Mohan Kumar authored
Tegra HDA has audio data buffer for upto tens of frames, this buffer can help to avoid underflow. HW will keep issuing new data fetch request when buffers are not full and current BDL is not done. When SW disable DMA RUN bit for a stream, HW can't cancel the already issued data fetch request and hence it can't stop DMA. HW has to wait for all issued data fetch request get data returned before it stops DMA. This HW behavior is not in sync with HDA spec which says DMA RUN bit should be cleared within 1 audio frame. For Tegra, DMA RUN bit was active for more than one audio frame, due to this the timeout in snd_hdac_stream_sync function is not helping. When Stream reset set and clear happens during DMA RUN bit active state it results in Memory Decode error. Unfortunately, there is no way to detect when these data accesses have completed, but testing has shown that a 100us delay between Stream reset set and clear operation for Tegra avoids the memory decode error. Therefore, adding a 100us dma stop delay. Signed-off-by:
Mohan Kumar <mkumard@nvidia.com> Link: https://lore.kernel.org/r/20200805095221.5476-4-mkumard@nvidia.comSigned-off-by:
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Mohan Kumar authored
A variable dma_stop_delay is added as a new member in hdac_bus structure to avoid memory decode error incase DMA RUN bit is not disabled in the given timeout from snd_hdac_stream_sync function and followed by stream reset which results in memory decode error between reset set and clear operation. Signed-off-by:
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Mohan Kumar authored
Set chip->align_buffer_size to 1 for Tegra platforms to make the buffer alignment to be multiple of 128 bytes. This fix is applied as gstreamer alsasink gets stuck with the default buffer-time and latency-time parameters with 4 byte buffer alignment. Signed-off-by:
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Takashi Iwai authored
Some ioctls via OSS sequencer API may race and lead to UAF when the port create and delete are performed concurrently, as spotted by a couple of syzkaller cases. This patch is an attempt to address it by serializing the ioctls with the existing register_mutex. Basically OSS sequencer API is an obsoleted interface and was designed without much consideration of the concurrency. There are very few applications with it, and the concurrent performance isn't asked, hence this "big hammer" approach should be good enough. Reported-by: syzbot+1a54a94bd32716796edd@syzkaller.appspotmail.com Reported-by: syzbot+9d2abfef257f3e2d4713@syzkaller.appspotmail.com Suggested-by:
Hillf Danton <hdanton@sina.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200804185815.2453-1-tiwai@suse.deSigned-off-by:
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- 04 Aug, 2020 2 commits
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Kai-Heng Feng authored
HDMI on some platforms doesn't enable audio support because its Port Connectivity [31:30] is set to AC_JACK_PORT_NONE: Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02 0x03* 0x04 For now, use a quirk to force connectivity based on SSID. If there are more platforms affected by the same issue, we can eye for a more generic solution. Signed-off-by:Kai-Heng Feng <kai.heng.feng@canonical.com> Link: https://lore.kernel.org/r/20200804155836.16252-1-kai.heng.feng@canonical.comSigned-off-by:
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Curtis Malainey authored
The dock sold from startech (PID: ICUSBAUDIO7D) has no friendly name and shows up currently as "USB Sound Device" in ALSA. Signed-off-by:
Curtis Malainey <cujomalainey@chromium.org> Link: https://lore.kernel.org/r/20200804010616.3399256-1-cujomalainey@chromium.orgSigned-off-by:
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- 03 Aug, 2020 7 commits
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Kai-Heng Feng authored
Lenovo ThinkStation P620 is like other TRX40 boards, is equipped with two USB audio cards. USB device (17aa:104d) provides functionality for Internal Speaker and Front Headset. It's UAC v2, so it supports insertion control (jack detection). However, when trying to get the connector status of the speaker, an error occurs: [ 5.787405] usb 3-1: cannot get connectors status: req = 0x81, wValue = 0x200, wIndex = 0x1000, type = 0 Since the insertion control works perfectly for the headset, the error for speaker is probably casued by connecting internally. So let's relax the error for a bit if it's a speaker, and always reports it's connected. USB device (17aa:1046) is for rear Line-in, Line-out and Microphone. The insertion control works for all three jacks. However, there's an Function Unit that doesn't work: [ 5.905415] usb 3-6: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 5.905418] usb 3-6: 19:0: cannot get min/max values for control 12 (id 19) So turn off the FU to avoid the error. Also, add specific card name for both devices, so userspace can easily indentify both cards. Signed-off-by:
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Takashi Iwai authored
Merge tag 'asoc-v5.9' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Updates for v5.9 The biggest changes here one again come from Mormioto-san who has continued his dilligent work cleaning up long standing issues in the APIs, it's particularly nice to see the transition from digital_mute() to mute_stream() finally completed. There's also been a lot of work on the x86 code again, this time a big focus has been on cleaning up some issues identified by various static tests, and on the Freescale systems. Otherwise the biggest thing has been a lot of driver additions: - Convert users of digital_mute() to mute_stream(). - Simplify I/O helper functions. - Add a helper for getting the RTD from a substream. - Many, many fixes and cleanups to the x86 code. - New drivers for Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim MAX98360A and MAX98373 Soundwire, several Mediatek boards, nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards (some of the first phones I worked on!) and TI J721e EVM.
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Hui Wang authored
This reverts commit 9a641848 ("ALSA: hda: call runtime_allow() for all hda controllers"). The reverted patch already introduced some regressions on some machines: - on gemini-lake machines, the error of "azx_get_response timeout" happens in the hda driver. - on the machines with alc662 codec, the audio jack detection doesn't work anymore. Fixes: 9a641848 ("ALSA: hda: call runtime_allow() for all hda controllers") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208511 Cc: <stable@vger.kernel.org> Signed-off-by:
Hui Wang <hui.wang@canonical.com> Link: https://lore.kernel.org/r/20200803064638.6139-1-hui.wang@canonical.comSigned-off-by:
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Connor McAdams authored
The ca0113 command had the wrong group_id, 0x48 when it should've been 0x30. The front microphone selection should now work. Signed-off-by:
Connor McAdams <conmanx360@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200803002928.8638-3-conmanx360@gmail.comSigned-off-by:
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Connor McAdams authored
Add a new quirk ID for the Recon3D, as tested by me. Signed-off-by:
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Connor McAdams authored
When the ZxR headphone gain control was added, the ca0132_switch_get function was not updated, which meant that the changes to the control state were not saved when entering/exiting alsamixer. Signed-off-by:
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Takashi Iwai authored
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- 02 Aug, 2020 1 commit
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Huacai Chen authored
There are several Loongson-3 based laptops produced by CZC or Lemote, they use alc269/alc662 codecs and need specific pin-tables, this patch add their pin-tables. Signed-off-by:
Huacai Chen <chenhc@lemote.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/1596360400-32425-1-git-send-email-chenhc@lemote.comSigned-off-by:
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- 01 Aug, 2020 2 commits
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Julia Lawall authored
GFP_KRENEL -> GFP_KERNEL Signed-off-by:
Julia Lawall <Julia.Lawall@inria.fr> Link: https://lore.kernel.org/r/1596224129-7699-1-git-send-email-Julia.Lawall@inria.frSigned-off-by:
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Julia Lawall authored
CONFIG_PCM_XRUN_DEBUG should be CONFIG_SND_PCM_XRUN_DEBUG Signed-off-by:
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- 31 Jul, 2020 10 commits
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Mark Brown authored
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Mark Brown authored
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Mark Brown authored
Merge series "ASoC: core: Two step component registration" from Cezary Rojewski <cezary.rojewski@intel.com>: Provide a mechanism for true two-step component registration. This mimics device registration flow where initialization is the first step while addition goes as second in line. Drivers may choose to modify component's fields before registering component to ASoC subsystem via snd_soc_add_component. Patchset achieves status quo - behavior of snd_soc_register_component remains unchanged. Cezary Rojewski (3): ASoC: core: Relocate and expose snd_soc_component_initialize ASoC: core: Simplify snd_soc_component_initialize declaration ASoC: core: Two step component registration include/sound/soc-component.h | 3 -- include/sound/soc.h | 11 +++--- sound/soc/soc-component.c | 16 --------- sound/soc/soc-core.c | 52 +++++++++++++++++---------- sound/soc/soc-generic-dmaengine-pcm.c | 14 +++++--- sound/soc/stm/stm32_adfsdm.c | 9 +++-- 6 files changed, 55 insertions(+), 50 deletions(-) -- 2.17.1
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Cezary Rojewski authored
Modify snd_soc_add_component so it calls snd_soc_component_initialize no longer and thus providing true two-step registration. Drivers may choose to change component's fields before actually adding it to ASoC subsystem. Signed-off-by:
Cezary Rojewski <cezary.rojewski@intel.com> Link: https://lore.kernel.org/r/20200731144146.6678-4-cezary.rojewski@intel.comSigned-off-by:
Mark Brown <broonie@kernel.org>
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Cezary Rojewski authored
Move 'name' field initialization responsibility back to snd_soc_component_initialize to prepare snd_soc_add_component function for being called separatelly as a second registration step. Signed-off-by:
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Cezary Rojewski authored
To allow for two-step component registration, expose snd_soc_component_initialize function and move it back to soc-core.c. Signed-off-by:
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Laurent Pinchart authored
Enabling a whole subsystem from a single driver 'select' is frowned upon and won't be accepted in new drivers, that need to use 'depends on' instead. Existing selection of DMADEVICES will then cause circular dependencies. Replace them with a dependency. Signed-off-by:
Laurent Pinchart <laurent.pinchart@ideasonboard.com> Acked-by:
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Shengjiu Wang authored
The fifo_depth is 64 on i.MX8QM/i.MX8QXP, 128 on i.MX8MQ, 16 on i.MX7ULP. Original FSL_SAI_CR1_RFW_MASK value 0x1F is not suitable for these platform, the FIFO watermark mask should be updated according to the fifo_depth. Fixes: a860fac4 ("ASoC: fsl_sai: Add support for imx7ulp/imx8mq") Signed-off-by:
Shengjiu Wang <shengjiu.wang@nxp.com> Reviewed-by:
Fabio Estevam <festevam@gmail.com> Link: https://lore.kernel.org/r/1596176895-28724-1-git-send-email-shengjiu.wang@nxp.comSigned-off-by:
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Jerome Brunet authored
Commit b73287f0 ("ASoC: soc-pcm: dpcm: fix playback/capture checks") changed the meaning of dpcm_playback/dpcm_capture and now requires the CPU DAI BE to aligned with those flags. This broke all Amlogic cards with uni-directional backends (All gx and most axg cards). While I'm still confused as to how this change is an improvement, those cards can't remain broken forever. Hopefully, next time an API change is done like that, all the users will be updated as part of the change, and not left to fend for themselves. Fixes: b73287f0 ("ASoC: soc-pcm: dpcm: fix playback/capture checks") Signed-off-by:
Jerome Brunet <jbrunet@baylibre.com> Link: https://lore.kernel.org/r/20200731120603.2243261-1-jbrunet@baylibre.comSigned-off-by:
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Pierre-Louis Bossart authored
Previous updates to set dailink capabilities and check dailink capabilities were based on a flawed assumption that all dais support the same capabilities as the dailink. This is true for TDM configurations but existing configurations use an amplifier and a capture device on the same dailink, and the tests would prevent the card from probing. This patch modifies the snd_soc_dai_link_set_capabilities() helper so that the dpcm_playback (resp. dpcm_capture) dailink capabilities are set if at least one dai supports playback (resp. capture). Likewise the checks are modified so that an error is reported only when dpcm_playback (resp. dpcm_capture) is set but none of the CPU DAIs support playback (resp. capture). Fixes: 25612477 ('ASoC: soc-dai: set dai_link dpcm_ flags with a helper') Fixes: b73287f0 ('ASoC: soc-pcm: dpcm: fix playback/capture checks') Suggested-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200723180533.220312-1-pierre-louis.bossart@linux.intel.comSigned-off-by:
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- 30 Jul, 2020 10 commits
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Mark Brown authored
The various list iterators are able to handle an empty list. The only effect of avoiding the loop is not initializing some index variables. Drop list_empty tests in cases where these variables are not used. The semantic patch that makes these changes is as follows: (http://coccinelle.lip6.fr/) <smpl> @@ expression x,e; iterator name list_for_each_entry; statement S; identifier i; @@ -if (!(list_empty(x))) { list_for_each_entry(i,x,...) S - } ... when != i ? i = e @@ expression x,e; iterator name list_for_each_entry_safe; statement S; identifier i,j; @@ -if (!(list_empty(x))) { list_for_each_entry_safe(i,j,x,...) S - } ... when != i when != j ( i = e; | ? j = e; ) @@ expression x,e; iterator name list_for_each; statement S; identifier i; @@ -if (!(list_empty(x))) { list_for_each(i,x) S - } ... when != i ? i = e @@ expression x,e; iterator name list_for_each_safe; statement S; identifier i,j; @@ -if (!(list_empty(x))) { list_for_each_safe(i,j,x) S - } ... when != i when != j ( i = e; | ? j = e; ) // ------------------- @@ expression x,e; statement S; identifier i; @@ -if (!(list_empty(x))) list_for_each_entry(i,x,...) S ... when != i ? i = e @@ expression x,e; statement S; identifier i,j; @@ -if (!(list_empty(x))) list_for_each_entry_safe(i,j,x,...) S ... when != i when != j ( i = e; | ? j = e; ) @@ expression x,e; statement S; identifier i; @@ -if (!(list_empty(x))) list_for_each(i,x) S ... when != i ? i = e @@ expression x,e; statement S; identifier i,j; @@ -if (!(list_empty(x))) list_for_each_safe(i,j,x) S ... when != i when != j ( i = e; | ? j = e; ) </smpl> --- drivers/media/pci/saa7134/saa7134-core.c | 14 ++--- drivers/media/usb/cx231xx/cx231xx-core.c | 16 ++---- drivers/media/usb/tm6000/tm6000-core.c | 24 +++------- drivers/net/ethernet/mellanox/mlx5/core/steering/dr_matcher.c | 13 ++--- drivers/net/ethernet/mellanox/mlx5/core/steering/dr_rule.c | 5 -- drivers/net/ethernet/sfc/ptp.c | 20 +++----- drivers/net/wireless/ath/dfs_pattern_detector.c | 15 ++---- sound/soc/intel/atom/sst/sst_loader.c | 10 +--- sound/soc/intel/skylake/skl-pcm.c | 8 +-- sound/soc/intel/skylake/skl-topology.c | 5 -- 10 files changed, 53 insertions(+), 77 deletions(-)
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Alper Nebi Yasak authored
PulseAudio (and perhaps other userspace utilities) can not detect any jack for rk3399_gru_sound as the driver doesn't expose related Jack kcontrols. This patch adds two DAPM pins to the headset jack, where the snd_soc_card_jack_new() call automatically creates "Headphones Jack" and "Headset Mic Jack" kcontrols from them. With an appropriate ALSA UCM config specifying JackControl fields for the "Headphones" and "Headset" (mic) devices, PulseAudio can detect plug/unplug events for both of them after this patch. Signed-off-by:
Alper Nebi Yasak <alpernebiyasak@gmail.com> Link: https://lore.kernel.org/r/20200721182709.6895-1-alpernebiyasak@gmail.comSigned-off-by:
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Fabio Estevam authored
According to the WM8962 datasheet, there is no register at address 0x200. WM8962_GPIO_BASE is just a base address for the GPIO registers and not a real register, so remove it from wm8962_readable_register(). Also, Register 515 (WM8962_GPIO_BASE + 3) does not exist, so skip its access. This fixes the following errors: wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 Signed-off-by:
Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20200717135959.19212-1-festevam@gmail.comSigned-off-by:
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Julia Lawall authored
Use resource_size rather than a verbose computation on the end and start fields. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) <smpl> @@ struct resource ptr; @@ - (ptr.end - ptr.start + 1) + resource_size(&ptr) </smpl> Signed-off-by:
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Julia Lawall authored
list_for_each_entry_safe is able to handle an empty list. The only effect of avoiding the loop is not initializing the index variable. Drop list_empty tests in cases where these variables are not used. Note that list_for_each_entry_safe is defined in terms of list_first_entry, which indicates that it should not be used on an empty list. But in list_for_each_entry_safe, the element obtained by list_first_entry is not really accessed, only the address of its list_head field is compared to the address of the list head, so the list_first_entry is safe. The semantic patch that makes this change is as follows (with another variant for the no brace case): (http://coccinelle.lip6.fr/) <smpl> @@ expression x,e; iterator name list_for_each_entry_safe; statement S; identifier i,j; @@ -if (!(list_empty(x))) { list_for_each_entry_safe(i,j,x,...) S - } ... when != i when != j ( i = e; | ? j = e; ) </smpl> Signed-off-by:
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Mark Brown authored
Merge series "ASoC: Intel: KMB: TDM Enablement patches" from Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>: This patch series is to enable multiple features on the Keembay Platform Michael Sit Wei Hong (4): ASoC: Intel: KMB: Add 8kHz audio support ASoC: Intel: KMB: Rework disable channel function ASoC: Intel: KMB: Enable TDM audio capture dt-bindings: sound: intel,keembay-i2s: Add channel-max property .../bindings/sound/intel,keembay-i2s.yaml | 8 + sound/soc/intel/keembay/kmb_platform.c | 137 +++++++++++++----- sound/soc/intel/keembay/kmb_platform.h | 1 + 3 files changed, 112 insertions(+), 34 deletions(-) -- 2.17.1
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Mark Brown authored
This patcheset is collection of fixes for the TDM input and output the axg audio architecture. Its fixes: - slave mode format setting - g12 and sm1 skew offset - tdm clock inversion - standard daifmt props names which don't require a specific prefix Jerome Brunet (4): ASoC: meson: axg-tdm-interface: fix link fmt setup ASoC: meson: axg-tdmin: fix g12a skew ASoC: meson: axg-tdm-formatters: fix sclk inversion ASoC: meson: cards: remove DT_PREFIX for standard daifmt properties sound/soc/meson/axg-tdm-formatter.c | 11 ++++++----- sound/soc/meson/axg-tdm-formatter.h | 1 - sound/soc/meson/axg-tdm-interface.c | 26 +++++++++++++++++--------- sound/soc/meson/axg-tdmin.c | 16 +++++++++++++++- sound/soc/meson/axg-tdmout.c | 3 --- sound/soc/meson/meson-card-utils.c | 2 +- 6 files changed, 39 insertions(+), 20 deletions(-) -- 2.25.4
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Kuninori Morimoto authored
This patch switches from .txt base to .yaml base Document. Signed-off-by:
Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by:
Rob Herring <robh@kernel.org> Link: https://lore.kernel.org/r/87mu4cxlo2.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87o8pf3923.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/873659bpbk.wl-kuninori.morimoto.gx@renesas.comSigned-off-by:
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Ravulapati Vishnu vardhan rao authored
Adding rt1015 hw_params which set Bit-clock ratio, PLL and appropriate sys clk specific with RTK1015. Signed-off-by:
Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com> Link: https://lore.kernel.org/r/20200728160255.31020-6-Vishnuvardhanrao.Ravulapati@amd.comSigned-off-by:
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Ravulapati Vishnu vardhan rao authored
DAI link support for RTK 1015 and providing the codec details depending on the snd_soc_card selected by ACPI ID. Signed-off-by:
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