- 11 Apr, 2020 1 commit
-
-
Adam Barber authored
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=207131Signed-off-by: Adam Barber <barberadam995@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200410090032.2759-1-barberadam995@gmail.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
- 09 Apr, 2020 1 commit
-
-
Xu Wang authored
Remove unnecassary casts in the argument to kfree. Signed-off-by: Xu Wang <vulab@iscas.ac.cn> Link: https://lore.kernel.org/r/20200409112052.13402-1-vulab@iscas.ac.cnSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
- 08 Apr, 2020 4 commits
-
-
Takashi Iwai authored
Merge tag 'asoc-fix-v5.7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.7 A collection of fixes that have been accumilated since the merge window, mainly relating to x86 platform support.
-
Takashi Iwai authored
The recent AMD platform exposes an HD-audio bus but without any actual codecs, which is internally tied with a USB-audio device, supposedly. It results in "no codecs" error of HD-audio bus driver, and it's nothing but a waste of resources. This patch introduces a static blacklist table for skipping such a known bogus PCI SSID entry. As of writing this patch, the known SSIDs are: * 1043:874f - ASUS ROG Zenith II / Strix * 1462:cb59 - MSI TRX40 Creator * 1462:cb60 - MSI TRX40 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200408140449.22319-2-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
Takashi Iwai authored
Some recent boards (supposedly with a new AMD platform) contain the USB audio class 2 device that is often tied with HD-audio. The device exposes an Input Gain Pad control (id=19, control=12) but this node doesn't behave correctly, returning an error for each inquiry of GET_MIN and GET_MAX that should have been mandatory. As a workaround, simply ignore this node by adding a usbmix_name_map table entry. The currently known devices are: * 0414:a002 - Gigabyte TRX40 Aorus Pro WiFi * 0b05:1916 - ASUS ROG Zenith II * 0b05:1917 - ASUS ROG Strix * 0db0:0d64 - MSI TRX40 Creator * 0db0:543d - MSI TRX40 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200408140449.22319-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
Takashi Iwai authored
MSI GL63 laptop requires the similar quirk like other MSI models, ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID for the device, hence we need to take the codec SSID (1462:1275) instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
- 07 Apr, 2020 5 commits
-
-
Takashi Iwai authored
The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e55 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139Reviewed-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
Takashi Iwai authored
The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e854 ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139Reviewed-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
Mike Willard authored
Pull the RST line low then high when initializing the driver, in order to force a reset of the chip. Previously, the line was not pulled low, which could result in the chip registers not resetting to their default values on boot. Signed-off-by: Mike Willard <mwillard@izotope.com> Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20200401205454.79792-1-mwillard@izotope.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
Kailang Yang authored
HP new platform has new mute led feature. COEF index 0x34 bit 5 to control playback mute led. COEF index 0x35 bit 2 and bit 3 to control Mic mute led. [ corrected typos by tiwai ] Signed-off-by: Kailang Yang <kailang@realtek.com> Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
Kailang Yang authored
HP Note Book supported new mute Led. Hardware PIN was not enough to meet old LED rule. JD2 to control playback mute led. GPO3 to control capture mute led. (ALC285 didn't control GPO3 via verb command) This two PIN just could control by COEF registers. [ corrected typos by tiwai ] Signed-off-by: Kailang Yang <kailang@realtek.com> Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
- 06 Apr, 2020 3 commits
-
-
Hans de Goede authored
The Medion E1239T uses the default jack-detect mode 3, but instead of using an analog microphone it is using a DMIC on dmic-data-pin 1, like other models following Intel's Brasswell's reference design. This commit adds a DMI quirk pointing to the intel_braswell_platform_data for this model. Signed-off-by: Hans de Goede <hdegoede@redhat.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200402185257.3355-1-hdegoede@redhat.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
Hans de Goede authored
The MPMAN MPWIN895CL tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200405133726.24154-1-hdegoede@redhat.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
Julia Lawall authored
The commit 0d6defc7 ("ASoC: stm32: sai: manage rebind issue") converts some function calls to their non-devm equivalents. The appropriate cleanup code was added to the remove function, but not to the probe function. Add a call to snd_dmaengine_pcm_unregister to compensate for the call to snd_dmaengine_pcm_register in case of subsequent failure. Fixes: commit 0d6defc7 ("ASoC: stm32: sai: manage rebind issue") Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr> Acked-by: Olivier Moysan <olivier.moysan@st.com> Link: https://lore.kernel.org/r/1586099028-5104-1-git-send-email-Julia.Lawall@inria.frSigned-off-by: Mark Brown <broonie@kernel.org>
-
- 04 Apr, 2020 1 commit
-
-
Emmanuel Pescosta authored
Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud Alpha S (0951:16d8) uses two interfaces, but only the second interface contains the capture stream. This patch delays the registration until the second interface appears. Signed-off-by: Emmanuel Pescosta <emmanuelpescosta099@gmail.com> Link: https://lore.kernel.org/r/20200404153843.9288-1-emmanuelpescosta099@gmail.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
- 03 Apr, 2020 8 commits
-
-
Hans de Goede authored
GCC 10 gives a "variable might be used uninitialized" warning for the block variable in sst_prepare_and_post_msg(). This is a false-positive warning, but lets fix it anyways. Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-3-hdegoede@redhat.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
Hans de Goede authored
sst_fill_and_send_cmd_unlocked must be called with the drv->lock mutex locked already. In the past there have been cases where this was not the case, add a WARN_ON to check for drv->lock being locked. Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-2-hdegoede@redhat.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
Hans de Goede authored
sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d141 ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
Hans de Goede authored
Using a Canon Lake machine with the SOF driver causes dmesg to fill up with a ton of these messages: [ 275.902194] sof-audio-pci 0000:00:1f.3: firmware boot complete [ 351.529358] sof-audio-pci 0000:00:1f.3: firmware boot complete [ 560.049047] sof-audio-pci 0000:00:1f.3: firmware boot complete etc. Since the DSP is powered down when not in used this happens everytime e.g. a notification plays, polluting dmesg. Turn this messages into a debug message, matching what the code already does for the ""booting DSP firmware" message. Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200402184948.3014-2-hdegoede@redhat.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
František Kučera authored
Pioneer DJ DJM-250MK2 is a mixer that acts like a USB sound card. The MIDI controller part is standard but the PCM part is "vendor specific". Output is enabled by this quirk: 8 channels, 48 000 Hz, S24_3LE. Input is not working. Signed-off-by: František Kučera <franta-linux@frantovo.cz> Link: https://lore.kernel.org/r/20200401095907.3387-1-konference@frantovo.czSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
Takashi Iwai authored
An empty merge for the original fix for PCM OSS regression where the same fix is already applied in a different form. Signed-off-by: Takashi Iwai <tiwai@suse.de>
-
Takashi Iwai authored
[ This is essentially the same fix as commit ae769d35, but it's adapted to the latest code for 5.7; hence it contains no Fixes or other tags for avoid backport confusion -- tiwai ] The recent fix for the OOB access in PCM OSS plugins (commit f2ecf903: "ALSA: pcm: oss: Avoid plugin buffer overflow") caused a regression on OSS applications. The patch introduced the size check in client and slave size calculations to limit to each plugin's buffer size, but I overlooked that some code paths call those without allocating the buffer but just for estimation. This patch fixes the bug by skipping the size check for those code paths while keeping checking in the actual transfer calls. Link: https://lore.kernel.org/r/20200403073818.27943-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
Takashi Iwai authored
The recent fix for the OOB access in PCM OSS plugins (commit f2ecf903: "ALSA: pcm: oss: Avoid plugin buffer overflow") caused a regression on OSS applications. The patch introduced the size check in client and slave size calculations to limit to each plugin's buffer size, but I overlooked that some code paths call those without allocating the buffer but just for estimation. This patch fixes the bug by skipping the size check for those code paths while keeping checking in the actual transfer calls. Fixes: f2ecf903 ("ALSA: pcm: oss: Avoid plugin buffer overflow") Tested-and-reported-by: Jari Ruusu <jari.ruusu@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200403072515.25539-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
- 02 Apr, 2020 2 commits
-
-
Takashi Iwai authored
Since snprintf() returns the would-be-output size instead of the actual output size, the succeeding calls may go beyond the given buffer limit. Fix it by replacing with scnprintf(). Link: https://lore.kernel.org/r/20200320084429.1803-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
Hans de Goede authored
The audio setup on the Lenovo Carbon X1 8th gen is the same as that on the Lenovo Carbon X1 7th gen, as such it needs the same ALC285_FIXUP_THINKPAD_HEADSET_JACK quirk. This fixes volume control of the speaker not working among other things. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1820196 Cc: stable@vger.kernel.org Suggested-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Hans de Goede <hdegoede@redhat.com> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Link: https://lore.kernel.org/r/20200402174311.238614-1-hdegoede@redhat.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
- 01 Apr, 2020 3 commits
-
-
이경택 authored
Current topology doesn't add prefix of component to new kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com> Link: https://lore.kernel.org/r/009b01d60804$ae25c2d0$0a714870$@samsung.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
YueHaibing authored
If I2C is n but SoundWire is m, building fails: sound/soc/codecs/rt5682.c:3716:1: warning: data definition has no type or storage class module_i2c_driver(rt5682_i2c_driver); ^~~~~~~~~~~~~~~~~ sound/soc/codecs/rt5682.c:3716:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] sound/soc/codecs/rt5682.c:3716:1: warning: parameter names (without types) in function declaration Guard this use #ifdef CONFIG_I2C. Fixes: 5549ea64 ("ASoC: rt5682: fix unmet dependencies") Signed-off-by: YueHaibing <yuehaibing@huawei.com> Link: https://lore.kernel.org/r/20200401091055.34112-1-yuehaibing@huawei.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
이경택 authored
soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com> Reviewed-by: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
- 31 Mar, 2020 7 commits
-
-
이경택 authored
Since a virtual mixer has no backing registers to decide which path to connect, it will try to match with initial state. This is to ensure that the default mixer choice will be correctly powered up during initialization. Invert flag is used to select initial state of the virtual switch. Since actual hardware can't be disconnected by virtual switch, connected is better choice as initial state in many cases. Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com> Link: https://lore.kernel.org/r/01a301d60731$b724ea10$256ebe30$@samsung.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
Stephan Gerhold authored
At the moment, playing audio with PulseAudio with the qdsp6 driver results in distorted sound. It seems like its timer-based scheduling does not work properly with qdsp6 since setting tsched=0 in the PulseAudio configuration avoids the issue. Apparently this happens when the pointer() callback is not accurate enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop PulseAudio from using timer-based scheduling by default. According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html: The flag is being used in the sense explained in the previous audio meeting -- the data transfer granularity isn't fine enough but aligned to the period size (or less). q6asm-dai reports the position as multiple of prtd->pcm_count = snd_pcm_lib_period_bytes(substream) so it indeed just a multiple of the period size. Therefore adding the flag here seems appropriate and makes audio work out of the box. Fixes: 2a9e92d3 ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Signed-off-by: Stephan Gerhold <stephan@gerhold.net> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20200330175210.47518-1-stephan@gerhold.netSigned-off-by: Mark Brown <broonie@kernel.org>
-
Andreas Steinmetz authored
The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra endpoint descriptor. The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00 As the code in snd_usbmidi_get_ms_info() looks only at the first extra descriptor to find USB_DT_CS_ENDPOINT the device as such is recognized but there is neither input nor output configured. The patch iterates through the extra descriptors to find the proper one. With this patch the device is correctly configured. Signed-off-by: Andreas Steinmetz <ast@domdv.de> Link: https://lore.kernel.org/r/1c3b431a86f69e1d60745b6110cdb93c299f120b.camel@domdv.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
Takashi Iwai authored
This reverts commit 645c08f1 which was reported to break the build a program using this header. The original issue was addressed in the alsa-lib side recently, so we can make the header more self-contained again. Reported-by: Dmitry V. Levin <ldv@altlinux.org> Fixes: 645c08f1 ("ALSA: uapi: Drop asound.h inclusion from asoc.h") Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200331090023.8112-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
Thomas Hebb authored
patch_realtek.c has historically failed to properly configure the PC Beep Hidden Register for the ALC256 codec (among others). Depending on your kernel version, symptoms of this misconfiguration can range from chassis noise, picked up by a poorly-shielded PCBEEP trace, getting amplified and played on your internal speaker and/or headphones to loud feedback, which responds to the "Headphone Mic Boost" ALSA control, getting played through your headphones. For details of the problem, see the patch in this series titled "ALSA: hda/realtek - Set principled PC Beep configuration for ALC256", which fixes the configuration. These symptoms have been most noticed on the Dell XPS 13 9350 and 9360, popular laptops that use the ALC256. As a result, several model-specific fixups have been introduced to try and fix the problem, the most egregious of which locks the "Headphone Mic Boost" control as a hack to minimize noise from a feedback loop that shouldn't have been there in the first place. Now that the underlying issue has been fixed, remove all these fixups. Remaining fixups needed by the XPS 13 are all picked up by existing pin quirks. This change should, for the XPS 13 9350/9360 - Significantly increase volume and audio quality on headphones - Eliminate headphone popping on suspend/resume - Allow "Headphone Mic Boost" to be set again, making the headphone jack fully usable as a microphone jack too. Fixes: 8c69729b ("ALSA: hda - Fix headphone noise after Dell XPS 13 resume back from S3") Fixes: 423cd785 ("ALSA: hda - Fix headphone noise on Dell XPS 13 9360") Fixes: e4c9fd10 ("ALSA: hda - Apply headphone noise quirk for another Dell XPS 13 variant") Fixes: 1099f484 ("ALSA: hda/realtek: Reduce the Headphone static noise on XPS 9350/9360") Cc: stable@vger.kernel.org Signed-off-by: Thomas Hebb <tommyhebb@gmail.com> Link: https://lore.kernel.org/r/b649a00edfde150cf6eebbb4390e15e0c2deb39a.1585584498.git.tommyhebb@gmail.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
Thomas Hebb authored
The Realtek PC Beep Hidden Register[1] is currently set by patch_realtek.c in two different places: In alc_fill_eapd_coef(), it's set to the value 0x5757, corresponding to non-beep input on 1Ah and no 1Ah loopback to either headphones or speakers. (Although, curiously, the loopback amp is still enabled.) This write was added fairly recently by commit e3743f431143 ("ALSA: hda/realtek - Dell headphone has noise on unmute for ALC236") and is a safe default. However, it happens in the wrong place: alc_fill_eapd_coef() runs on module load and cold boot but not on S3 resume, meaning the register loses its value after suspend. Conversely, in alc256_init(), the register is updated to unset bit 13 (disable speaker loopback) and set bit 5 (set non-beep input on 1Ah). Although this write does run on S3 resume, it's not quite enough to fix up the register's default value of 0x3717. What's missing is a set of bit 14 to disable headphone loopback. Without that, we end up with a feedback loop where the headphone jack is being driven by amplified samples of itself[2]. This change eliminates the update in alc256_init() and replaces it with the 0x5757 write from alc_fill_eapd_coef(). Kailang says that 0x5757 is supposed to be the codec's default value, so using it will make debugging easier for Realtek. Affects the ALC255, ALC256, ALC257, ALC235, and ALC236 codecs. [1] Newly documented in Documentation/sound/hd-audio/realtek-pc-beep.rst [2] Setting the "Headphone Mic Boost" control from userspace changes this feedback loop and has been a widely-shared workaround for headphone noise on laptops like the Dell XPS 13 9350. This commit eliminates the feedback loop and makes the workaround unnecessary. Fixes: e1e8c1fd ("ALSA: hda/realtek - Dell headphone has noise on unmute for ALC236") Cc: stable@vger.kernel.org Signed-off-by: Thomas Hebb <tommyhebb@gmail.com> Link: https://lore.kernel.org/r/bf22b417d1f2474b12011c2a39ed6cf8b06d3bf5.1585584498.git.tommyhebb@gmail.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
Thomas Hebb authored
This codec (among others) has a hidden set of audio routes, apparently designed to allow PC Beep output without a mixer widget on the output path, which are controlled by an undocumented Realtek vendor register. The default configuration of these routes means that certain inputs aren't accessible, necessitating driver control of the register. However, Realtek has provided no documentation of the register, instead opting to fix issues by providing magic numbers, most of which have been at least somewhat erroneous. These magic numbers then get copied by others into model-specific fixups, leading to a fragmented and buggy set of configurations. To get out of this situation, I've reverse engineered the register by flipping bits and observing how the codec's behavior changes. This commit documents my findings. It does not change any code. Cc: stable@vger.kernel.org Signed-off-by: Thomas Hebb <tommyhebb@gmail.com> Link: https://lore.kernel.org/r/bd69dfdeaf40ff31c4b7b797c829bb320031739c.1585584498.git.tommyhebb@gmail.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
-
- 30 Mar, 2020 5 commits
-
-
Mark Brown authored
Merge series "ASoC: Intel: boards: Remove ignore_suspend flag from SSP0 dai link" from Cezary Rojewski <cezary.rojewski@intel.com>: As of commit: ASoC: soc-core: care .ignore_suspend for Component suspend function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend' flag for dai links. While BE dai link for System Pin is supposed to follow standard suspend-resume flow, appended 'ignore_suspend' flag disturbs that flow and causes audio to break right after resume. Remove the flag to address this. Link to first message in conversation: https://lkml.org/lkml/2020/3/18/54 Cezary Rojewski (4): ASoC: Intel: broadwell: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: haswell: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: bdw-rt5677: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: bdw-rt5650: Remove ignore_suspend flag from SSP0 dai link sound/soc/intel/boards/bdw-rt5650.c | 1 - sound/soc/intel/boards/bdw-rt5677.c | 1 - sound/soc/intel/boards/broadwell.c | 1 - sound/soc/intel/boards/haswell.c | 1 - 4 files changed, 4 deletions(-) -- 2.17.1
-
Pierre-Louis Bossart authored
The addition of a single flag to track the DAI status prevents the DAI startup sequence from being called on capture if the DAI is already used for playback. Fix by extending the existing code with one flag per direction. Fixes: b56be800 ("ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once") Reported-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Tested-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Link: https://lore.kernel.org/r/20200330160602.10180-1-pierre-louis.bossart@linux.intel.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
이경택 authored
If regwshift is 32 and the selected architecture compiles '<<' operator for signed int literal into rotating shift, '1<<regwshift' became 1 and it makes regwmask to 0x0. The literal is set to unsigned long to get intended regwmask. Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com> Link: https://lore.kernel.org/r/001001d60665$db7af3e0$9270dba0$@samsung.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
Cezary Rojewski authored
As of commit: ASoC: soc-core: care .ignore_suspend for Component suspend function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend' flag for dai links. While BE dai link for System Pin is supposed to follow standard suspend-resume flow, appended 'ignore_suspend' flag disturbs that flow and causes audio to break right after resume. Remove the flag to address this. Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Dominik Brodowski <linux@dominikbrodowski.net> Cc: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20200319204947.18963-5-cezary.rojewski@intel.comSigned-off-by: Mark Brown <broonie@kernel.org>
-
Cezary Rojewski authored
As of commit: ASoC: soc-core: care .ignore_suspend for Component suspend function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend' flag for dai links. While BE dai link for System Pin is supposed to follow standard suspend-resume flow, appended 'ignore_suspend' flag disturbs that flow and causes audio to break right after resume. Remove the flag to address this. Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Dominik Brodowski <linux@dominikbrodowski.net> Cc: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20200319204947.18963-4-cezary.rojewski@intel.comSigned-off-by: Mark Brown <broonie@kernel.org>
-