- 23 Jan, 2020 1 commit
-
-
Colin Ian King authored
There is a spelling mistake in a dev_err message. Fix it. Signed-off-by:
Colin Ian King <colin.king@canonical.com> Link: https://lore.kernel.org/r/20200123000050.2831088-1-colin.king@canonical.comSigned-off-by:
Takashi Iwai <tiwai@suse.de>
-
- 21 Jan, 2020 3 commits
-
-
Kai Vehmanen authored
The initial snd_hda_get_sub_node() can fail on certain devices (e.g. some Chromebook models using Intel GLK). The failure rate is very low, but as this is is part of the probe process, end-user impact is high. In observed cases, related hardware status registers have expected values, but the node query still fails. Retrying the node query does seem to help, so fix the problem by adding retry logic to the query. This does not impact non-Intel platforms. BugLink: https://github.com/thesofproject/linux/issues/1642Signed-off-by:
Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by:
-
Takashi Iwai authored
Like many other drivers, HD-audio drivers also do PCM buffer preallocation to assure the buffer pages allocated at the early boot stage. This step is useful for platforms that may fail to allocate the PCM hardware buffers -- which is mostly for either large continuous pages or with the specific DMA mask (like emu10k1). OTOH, when a buffer is allocated as SG-buffer and the DMA mask is either 32 or 64 bits, the allocation almost never fails unless it hits the real OOM situation. In such a case, we don't need the preallocation inevitably unlike the cases above. That said, we may drop the preallocation for HD-audio that does allocate via SG-buffers, and the patch achieves it. However, there is one caveat: the buffer allocation behavior depends on CONFIG_SND_DMA_SGBUF, and it falls back to the continuous pages when it's not set. And, currently this SG buffer allocation is enabled only on x86 platforms. So, covering those fall-outs, the patch adjusts CONFIG_SND_HDA_PREALLOC_SIZE depending on the condition, and keeps the old behavior as-is for non-x86 platforms. On x86, the kconfig item is no longer adjustable but always set to zero for disabling the preallocation. You can still enable the preallocation via procfs interface at any time later, too. Link: https://lore.kernel.org/r/20200120124423.11862-2-tiwai@suse.deSigned-off-by:
-
Takashi Iwai authored
Currently, the available buffer allocation size for a PCM stream depends on the preallocated size; when a buffer has been preallocated, the max buffer size is set to that size, so that application won't re-allocate too much memory. OTOH, when no preallocation is done, each substream may allocate arbitrary size of buffers as long as snd_pcm_hardware.buffer_bytes_max allows -- which can be quite high, HD-audio sets 1GB there. It means that the system may consume a high amount of pages for PCM buffers, and they are pinned and never swapped out. This can lead to OOM easily. For avoiding such a situation, this patch adds the upper limit per card. Each snd_pcm_lib_malloc_pages() and _free_pages() calls are tracked and it will return an error if the total amount of buffers goes over the defined upper limit. The default value is set to 32MB, which should be really large enough for usual operations. If larger buffers are needed for any specific usage, it can be adjusted (also dynamically) via snd_pcm.max_alloc_per_card option. Setting zero there means no chceck is performed, and again, unlimited amount of buffers are allowed. Link: https://lore.kernel.org/r/20200120124423.11862-1-tiwai@suse.deSigned-off-by:
-
- 20 Jan, 2020 2 commits
-
-
Takashi Iwai authored
Resolved the merge conflict in HD-audio Tegra driver. Signed-off-by: -
Takashi Iwai authored
It turned out that the recent simplification of HD-audio bus access helpers caused a regression on the virtual HD-audio device on QEMU with ARM platforms. The driver got a CORB/RIRB timeout and couldn't probe any codecs. The essential difference that caused a problem was the enforced aligned MMIO accesses by simplification. Since snd-hda-tegra driver is enabled on ARM, it enables CONFIG_SND_HDA_ALIGNED_MMIO, which makes the all HD-audio drivers using the aligned MMIO accesses. While this is mandatory for snd-hda-tegra, it seems that snd-hda-intel on ARM gets broken by this access pattern. For addressing the regression, this patch introduces a new flag, aligned_mmio, to hdac_bus object, and applies the aligned MMIO only when this flag is set. This change affects only platforms with CONFIG_SND_HDA_ALIGNED_MMIO set, i.e. mostly only for ARM platforms. Unfortunately the patch became a big bigger than it should be, just because the former calls didn't take hdac_bus object in the argument, hence we had to extend the call patterns. Fixes: 19abfefd ("ALSA: hda: Direct MMIO accesses") BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1161152 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200120104127.28985-1-tiwai@suse.deSigned-off-by:
-
- 17 Jan, 2020 1 commit
-
-
Kailang Yang authored
HP ALC671 need to support Headset Mic. Signed-off-by:
Kailang Yang <kailang@realtek.com> Link: https://lore.kernel.org/r/06a9d2b176e14706976d6584cbe2d92a@realtek.comSigned-off-by:
-
- 16 Jan, 2020 4 commits
-
-
Takashi Iwai authored
Both snd_pcm_hw_constraints_init() and _complete() functions are called only from pcm_native.c, hence they can be static for further optimization. Link: https://lore.kernel.org/r/20200116162825.24792-1-tiwai@suse.deSigned-off-by:
-
Takashi Iwai authored
AD HD-audio codec driver has a few code lines invoking snd_get_num_conns() and using its return value as the array index without checking. This is basically safe in all those places; at the second and later calls snd_get_num_conns() returns the value cached from the first invocation, hence the value is always consistent. However, it looks a bit confusing as if a lack of the proper check. This patch introduces a new field num_smux_conns in ad198x_spec for simplifying the code. Now we store and refer to the value more locally without invoking the extra function at each time. Reported-by:
-
Takashi Iwai authored
Merge tag 'asoc-fix-v5.5-rc6' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.5 This is mostly driver specific fixes, plus an error handling fix in the core. There is a rather large diffstat for the stm32 SAI driver, this is a very large but mostly mechanical update which wraps every register access in the driver to allow a fix to the locking which avoids circular locks, the active change is much smaller and more reasonably sized.
-
Alexander Tsoy authored
This fixes crackling sound during playback. Further note: MOTU is known for reusing Product IDs for different devices or different generations of the device (e.g. MicroBook I/II/IIc shares a single Product ID). This patch was only tested with M4 audio interface, but the same Product ID is also used by M2. Hope it will work for M2 as well. Signed-off-by:
Alexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20200115151358.56672-1-alexander@tsoy.meSigned-off-by:
-
- 15 Jan, 2020 2 commits
-
-
Takashi Iwai authored
snd_seq_info_timer_read() reads the information of the timer assigned for each queue, but it's done in a racy way which may lead to UAF as spotted by syzkaller. This patch applies the missing q->timer_mutex lock while accessing the timer object as well as a slight code change to adapt the standard coding style. Reported-by: syzbot+2b2ef983f973e5c40943@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200115203733.26530-1-tiwai@suse.deSigned-off-by:
-
Dan Carpenter authored
We need to unlock before we returning on this error path. Fixes: 73ac9f5e ("ALSA: usb-audio: Add boot quirk for MOTU M Series") Signed-off-by:
Dan Carpenter <dan.carpenter@oracle.com> Link: https://lore.kernel.org/r/20200115174604.rhanfgy4j3uc65cx@kili.mountainSigned-off-by:
-
- 14 Jan, 2020 4 commits
-
-
Johan Hovold authored
The altsetting sanity check in set_sync_ep_implicit_fb_quirk() was checking for there to be at least one altsetting but then went on to access the second one, which may not exist. This could lead to random slab data being used to initialise the sync endpoint in snd_usb_add_endpoint(). Fixes: c75a8a7a ("ALSA: snd-usb: add support for implicit feedback") Fixes: ca10a7eb ("ALSA: usb-audio: FT C400 sync playback EP to capture EP") Fixes: 5e35dc03 ("ALSA: usb-audio: add implicit fb quirk for Behringer UFX1204") Fixes: 17f08b0d ("ALSA: usb-audio: add implicit fb quirk for Axe-Fx II") Fixes: 103e9625 ("ALSA: usb-audio: simplify set_sync_ep_implicit_fb_quirk") Cc: stable <stable@vger.kernel.org> # 3.5 Signed-off-by:
Johan Hovold <johan@kernel.org> Link: https://lore.kernel.org/r/20200114083953.1106-1-johan@kernel.orgSigned-off-by:
-
Pierre-Louis Bossart authored
make W=1 reports the following warnings, fix as suggested sound/pci/hda/patch_hdmi.c: In function ‘hdmi_non_intrinsic_event’: sound/pci/hda/patch_hdmi.c:824:3: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] 824 | ; | ^ sound/pci/hda/patch_hdmi.c:826:3: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] 826 | ; | ^ Signed-off-by:Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200113211405.28070-3-pierre-louis.bossart@linux.intel.comSigned-off-by:
-
Pierre-Louis Bossart authored
make W=1 throws warnings, provide missing documentation Signed-off-by:
-
Keyon Jie authored
Make W=1 throws a lot of warnings, with multiple misalignments between function params and their descriptions. Signed-off-by:
Keyon Jie <yang.jie@linux.intel.com> Signed-off-by:
-
- 13 Jan, 2020 17 commits
-
-
Stephan Gerhold authored
For some reason, attempting to route audio through QDSP6 on MSM8916 causes the RX interpolation path to get "stuck" after playing audio a few times. In this situation, the analog codec part is still working, but the RX path in the digital codec stops working, so you only hear the analog parts powering up. After a reboot everything works again. So far I was not able to reproduce the problem when using lpass-cpu. The downstream kernel driver avoids this by resetting the RX interpolation path after use. In mainline we do something similar for the TX decimator (LPASS_CDC_CLK_TX_RESET_B1_CTL), but the interpolator reset (LPASS_CDC_CLK_RX_RESET_CTL) got lost when the msm8916-wcd driver was split into analog and digital. Fix this problem by adding the reset to msm8916_wcd_digital_enable_interpolator(). Fixes: 150db8c5 ("ASoC: codecs: Add msm8916-wcd digital codec") Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by:
Stephan Gerhold <stephan@gerhold.net> Link: https://lore.kernel.org/r/20200105102753.83108-1-stephan@gerhold.netSigned-off-by:
Mark Brown <broonie@kernel.org>
-
Stephan Gerhold authored
MIC BIAS Internal1 is broken at the moment because we always enable the internal rbias resistor to the TX2 line (connected to the headset microphone), rather than enabling the resistor connected to TX1. Move the RBIAS code to pm8916_wcd_analog_enable_micbias_int1/2() to fix this. Fixes: 585e881e ("ASoC: codecs: Add msm8916-wcd analog codec") Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by:
-
Yu-Hsuan Hsu authored
Add ACPI entry for cros_ec_codec. Signed-off-by:
Yu-Hsuan Hsu <yuhsuan@chromium.org> Link: https://lore.kernel.org/r/20200112054900.236576-1-yuhsuan@chromium.orgSigned-off-by:
-
Arnaud Pouliquen authored
Change mutex and spinlock management to avoid sleep in atomic issue. Signed-off-by:
Arnaud Pouliquen <arnaud.pouliquen@st.com> Link: https://lore.kernel.org/r/20200113100400.30472-1-arnaud.pouliquen@st.comSigned-off-by:
-
Stephan Gerhold authored
MIC BIAS External1 sets pm8916_wcd_analog_enable_micbias_ext1() as event handler, which ends up in pm8916_wcd_analog_enable_micbias_ext(). But pm8916_wcd_analog_enable_micbias_ext() only handles the POST_PMU event, which is not specified in the event flags for MIC BIAS External1. This means that the code in the event handler is never actually run. Set SND_SOC_DAPM_POST_PMU as the only event for the handler to fix this. Fixes: 585e881e ("ASoC: codecs: Add msm8916-wcd analog codec") Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by:
-
Kai Vehmanen authored
In case system has multiple HDA codecs, and codec probe fails for at least one but not all codecs, driver will end up cancelling a non-initialized timer context upon driver removal. Call trace of typical case: [ 60.593646] WARNING: CPU: 1 PID: 1147 at kernel/workqueue.c:3032 __flush_work+0x18b/0x1a0 [...] [ 60.593670] __cancel_work_timer+0x11f/0x1a0 [ 60.593673] hdac_hda_dev_remove+0x25/0x30 [snd_soc_hdac_hda] [ 60.593674] device_release_driver_internal+0xe0/0x1c0 [ 60.593675] bus_remove_device+0xd6/0x140 [ 60.593677] device_del+0x175/0x3e0 [ 60.593679] ? widget_tree_free.isra.7+0x90/0xb0 [snd_hda_core] [ 60.593680] snd_hdac_device_unregister+0x34/0x50 [snd_hda_core] [ 60.593682] snd_hdac_ext_bus_device_remove+0x2a/0x60 [snd_hda_ext_core] [ 60.593684] hda_dsp_remove+0x26/0x100 [snd_sof_intel_hda_common] [ 60.593686] snd_sof_device_remove+0x84/0xa0 [snd_sof] [ 60.593687] sof_pci_remove+0x10/0x30 [snd_sof_pci] [ 60.593689] pci_device_remove+0x36/0xb0 Signed-off-by:
-
Kai Vehmanen authored
In case system has multiple HDA controllers, it can happen that same HDA codec driver is used for codecs of multiple controllers. In this case, SOF may fail to probe the HDA driver and SOF initialization fails. SOF HDA code currently relies that a call to request_module() will also run device matching logic to attach driver to the codec instance. However if driver for another HDA controller was already loaded and it already loaded the HDA codec driver, this breaks current logic in SOF. In this case the request_module() SOF does becomes a no-op and HDA Codec driver is not attached to the codec instance sitting on the HDA bus SOF is controlling. Typical scenario would be a system with both external and internal GPUs, with driver of the external GPU loaded first. Fix this by adding similar logic as is used in legacy HDA driver where an explicit device_attach() call is done after request_module(). Also add logic to propagate errors reported by device_attach() back to caller. This also works in the case where drivers are not built as modules. Signed-off-by:
-
Bard liao authored
We will reinit DSP in a loop when it fails to initialize the first time, as recommended. So, it is not an error before we finally give up. And reorder the trace to make it more readable. Signed-off-by:
Bard liao <yung-chuan.liao@linux.intel.com> Signed-off-by:
-
Takashi Iwai authored
In the commit 8e85def5 ("ALSA: hda: enable regmap internal locking"), we re-enabled the regmap lock due to the reported regression that showed the possible concurrent accesses. It was a temporary workaround, and there are still a few opened races even after the revert. In this patch, we cover those still opened windows with a proper mutex lock and disable the regmap internal lock again. First off, the patch introduces a new snd_hdac_device.regmap_lock mutex that is applied for each snd_hdac_regmap_*() call, including read, write and update helpers. The mutex is applied carefully so that it won't block the self-power-up procedure in the helper function. Also, this assures the protection for the accesses without regmap, too. The snd_hdac_regmap_update_raw() is refactored to use the standard regmap_update_bits_check() function instead of the open-code. The non-regmap case is still open-coded but it's an easy part. The all read and write operations are in the single mutex protection, so it's now race-free. In addition, a couple of new helper functions are added: snd_hdac_regmap_update_raw_once() and snd_hdac_regmap_sync(). Both are called from HD-audio legacy driver. The former is to initialize the given verb bits but only once when it's not initialized yet. Due to this condition, the function invokes regcache_cache_only(), and it's now performed inside the regmap_lock (formerly it was racy) too. The latter function is for simply invoking regcache_sync() inside the regmap_lock, which is called from the codec resume call path. Along with that, the HD-audio codec driver code is slightly modified / simplified to adapt those new functions. And finally, snd_hdac_regmap_read_raw(), *_write_raw(), etc are rewritten with the helper macro. It's just for simplification because the code logic is identical among all those functions. Tested-by:
-
Alexander Tsoy authored
Add delay to make sure that audio urbs are not sent too early. Otherwise the device hangs. Windows driver makes ~2s delay, so use about the same time delay value. snd_usb_apply_boot_quirk() is called 3 times for my MOTU M4, which is an overkill. Thus a quirk that is called only once is implemented. Also send two vendor-specific control messages before and after the delay. This behaviour is blindly copied from the Windows driver. Signed-off-by:
-
Takashi Sakamoto authored
Alesis MasterControl was shipped 2009 and already discontinued. This model consists of: * TSB41AB2 for physical layer of IEEE 1394 * WaveFront Dice II STD for link layer and protocol implementation * FreeScale DSPB56374AE Although the firmware of this model can respond against read transaction to address space for TCAT extension protocol, the content is not valid for protocol extension. This results in sound card without any PCM/MIDI interfaces. $ ./firewire-request /dev/fw1 read 0xffffe0200000 0x48 result: 000: 00 00 00 20 00 00 04 94 00 00 04 b4 00 00 00 b4 result: 010: 00 00 05 68 00 00 00 24 00 00 05 8c 00 00 00 48 result: 020: 00 00 00 20 00 00 00 08 00 00 00 20 00 00 00 20 result: 030: 00 00 00 10 00 00 00 08 00 00 00 08 00 00 00 04 result: 040: 00 00 00 00 00 00 00 00 This commit adds support the model by adding hard-coded stream formats. $ python3 ~/git/linux-firewire-utils/src/crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04041ad7 bus_info_length 4, crc_length 4, crc 6871 404 31333934 bus_name "1394" 408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255, max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400) 40c 00059504 company_id 000595 | Alesis Corporation 410 008003f5 device_id 04008003f5 | EUI-64 00059504008003f5 root directory ----------------------------------------------------------------- 414 0006a620 directory_length 6, crc 42528 418 03000595 vendor: Alesis Corporation 41c 8100000a --> descriptor leaf at 444 420 17000002 model 424 8100000d --> descriptor leaf at 458 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ----------------------------------------------------------------- 430 00041b9f directory_length 4, crc 7071 434 12000595 specifier id: Alesis Corporation 438 13000001 version: audio 43c 17000002 model 440 8100000d --> descriptor leaf at 474 descriptor leaf at 444 ----------------------------------------------------------------- 444 000494c2 leaf_length 4, crc 38082 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 416c6573 "Ales" 454 69730000 "is" descriptor leaf at 458 ----------------------------------------------------------------- 458 0006c2ec leaf_length 6, crc 49900 45c 00000000 textual descriptor 460 00000000 minimal ASCII 464 4d617374 "Mast" 468 6572436f "erCo" 46c 6e74726f "ntro" 470 6c000000 "l" descriptor leaf at 474 ----------------------------------------------------------------- 474 0006c2ec leaf_length 6, crc 49900 478 00000000 textual descriptor 47c 00000000 minimal ASCII 480 4d617374 "Mast" 484 6572436f "erCo" 488 6e74726f "ntro" 48c 6c000000 "l" Signed-off-by:Takashi Sakamoto <o-takashi@sakamocchi.jp> Link: https://lore.kernel.org/r/20200113084630.14305-4-o-takashi@sakamocchi.jpSigned-off-by:
-
Takashi Sakamoto authored
ALSA dice driver expects devices to multiplex MIDI messages into first port of isochronous communication. Actually devices perform for it. However, check of stream format is invalid for second port of isochronous communication. As a result, when the device supports two ports for isochronous communication and the stream format is hard-coded, ALSA dice driver fails to start packet streaming. This commit loosens stream format check for MIDI conformant data channel. Signed-off-by:
-
Takashi Sakamoto authored
At failure of attempt to detect protocol extension, ALSA dice driver should be fallback to limited functionality. However it's not. This commit fixes it. Cc: <stable@vger.kernel.org> # v4.18+ Fixes: 58579c05 ("ALSA: dice: use extended protocol to detect available stream formats") Signed-off-by:
-
Takashi Sakamoto authored
Stanton SCS.1d uses Oxford Semiconductor FW 971 ASIC (FW971) for communication. Although the unit is bound to ALSA oxfw driver, the instance of sound card can not be added due to its quirk of plug information. This bug was added when snd-scs1x is merged into snd-oxfw at commit 9e2004f9 ("ALSA: oxfw: obsolete scs1x module"). This commit fixes the driver for the quirk. In cases that the unit returns NOT IMPLEMENTED for some AV/C commands, the sound card is added without any PCM/MIDI interfaces for packet streaming. For SCS.1d, model dependent operation adds MIDI interface and applications can use it to operate according to HSS1394 protocol from reverse-engineering work by Sean M. Pappalardo. Plug Control Register (PCR) has information that the unit has a pair of plugs for isochronous communication: (oMPR) $ ./firewire-request /dev/fw1 read 0xfffff0000900 result: 80ff0001 (iMPR) $ ./firewire-request /dev/fw1 read 0xfffff0000980 result: 80ff0001 AV/C PLUG INFO also returns information that the unit has a pair of plugs for isochronous communication. (AV/C PLUG INFO command) $ ./firewire-request /dev/fw1 fcp 0x01ff0200ffffffff response: 000: 0c ff 02 00 01 01 02 02 However, AV/C PLUG SIGNAL INFO command is rejected for both plugs. (AV/C OUTPUT PLUG SIGNAL INFO command) $ ./firewire-request /dev/fw1 fcp 0x01ff1800ffffffff response: 000: 0a ff 18 00 ff ff ff ff (AV/C INPUT PLUG SIGNAL INFO command) $ ./firewire-request /dev/fw1 fcp 0x01ff1900ffffffff response: 000: 0a ff 19 00 ff ff ff ff Furthermore, AV/C EXTENDED STREAM FORMAT INFO is not implemented. (AV/C EXTENDED STREAM FORMAT INFO list subfunction for input plug) $ ./firewire-request /dev/fw1 fcp 0x01ffbfc000000000ffff00ff response: 000: 08 ff bf c0 00 00 00 00 ff ff 00 ff (AV/C EXTENDED STREAM FORMAT INFO list subfunction for output plug) $ ./firewire-request /dev/fw1 fcp 0x01ffbfc001000000ffff00ff response: 000: 08 ff bf c0 01 00 00 00 ff ff 00 ff (AV/C EXTENDED STREAM FORMAT INFO single subfunction for input plug) $ ./firewire-request /dev/fw1 fcp 0x01ffbfc100000000ffffffff response: 000: 08 ff bf c1 00 00 00 00 ff ff ff ff (AV/C EXTENDED STREAM FORMAT INFO single subfunction for output plug) $ ./firewire-request /dev/fw1 fcp 0x01ffbfc101000000ffffffff response: 000: 08 ff bf c1 01 00 00 00 ff ff ff ff Reference: https://mailman.alsa-project.org/pipermail/alsa-devel/2012-May/052264.htmlSigned-off-by:
-
Takashi Sakamoto authored
Stanton SCS.1d doesn't support packet streaming even if it has plugs for isochronous communication. This commit is a preparation for this case. The 'has_input' member is added to specific structure, and MIDI/PCM interfaces are not added when the member is false. Signed-off-by:
-
Takashi Sakamoto authored
When AV/C command returns 'NOT IMPLEMENTED' status in its response, ALSA oxfw driver uses ENOSYS as error code. However, it's expected just to be used for missing system call number. This commit replaces it with ENXIO. Signed-off-by:
-
Takashi Sakamoto authored
ALSA firewire-tascam driver can bring corruption due to spin lock without restoration of IRQ flag in SoftIRQ context. This commit fixes the bug. Cc: Scott Bahling <sbahling@suse.com> Cc: <stable@vger.kernel.org> # v4.21 Fixes: d7167422 ("ALSA: firewire-tascam: queue events for change of control surface") Signed-off-by:
-
- 12 Jan, 2020 3 commits
-
-
Pierre-Louis Bossart authored
GCC reports the following warning with W=1 sound/usb/mixer_quirks.c: In function ‘snd_microii_controls_create’: sound/usb/mixer_quirks.c:1694:2: warning: ‘static’ is not at beginning of declaration [-Wold-style-declaration] 1694 | const static usb_mixer_elem_resume_func_t resume_funcs[] = { | ^~~~~ Move static to the beginning of declaration Signed-off-by: -
Pierre-Louis Bossart authored
GCC reports the following warning with W=1 sound/pci/hda/patch_realtek.c: In function ‘alc269_suspend’: sound/pci/hda/patch_realtek.c:3616:29: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] 3616 | alc5505_dsp_suspend(codec); | ^ sound/pci/hda/patch_realtek.c: In function ‘alc269_resume’: sound/pci/hda/patch_realtek.c:3651:28: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] 3651 | alc5505_dsp_resume(codec); | ^ This is a classic macro problem and can indeed lead to bad program flows. Fix by using the usual "do { } while (0)" pattern Signed-off-by: -
Pierre-Louis Bossart authored
GCC reports a warning with W=1: sound/core/timer.c: In function ‘snd_timer_user_read’: sound/core/timer.c:2219:19: warning: initialized field overwritten [-Woverride-init] 2219 | .tstamp_sec = tread->tstamp_nsec, | ^~~~~ sound/core/timer.c:2219:19: note: (near initialization for ‘(anonymous).tstamp_sec’) Assigning nsec values to sec fields is problematic in general, even more so when the initial goal was to survive the 2030 timer armageddon. Fix by using the proper field in the initialization Cc: Baolin Wang <baolin.wang@linaro.org> Cc: Arnd Bergmann <arnd@arndb.de> Fixes: 07094ae6 ("ALSA: Avoid using timespec for struct snd_timer_tread") Signed-off-by:Arnd Bergmann <arnd@arndb.de> Link: https://lore.kernel.org/r/20200111203325.20498-1-pierre-louis.bossart@linux.intel.comSigned-off-by:
-
- 11 Jan, 2020 2 commits
-
-
Takashi Iwai authored
Sync 5.5-devel branch once again for applying the HD-audio fixes. Signed-off-by: -
Takashi Iwai authored
We've got quite a few bug reports showing the SOF driver being loaded unintentionally recently, and the reason seems to be that users didn't know the module option change: with the recent kernel, a new option dsp_driver=1 has to be passed to a new module snd-intel-dspcfg instead of snd_hda_intel.dmic_detect=0 option. That is, actually there are two tricky things here: - We changed the whole detection in another module and another option semantics. - The existing option for skipping the DSP probe was also renamed. For avoiding the confusion and giving user more hint, this patch reverts the renamed option dsp_driver back to dmic_detect for snd-hda-intel module, and show the warning about the module option change when the non-default value is passed. Fixes: 82d9d54a ("ALSA: hda: add Intel DSP configuration / probe code") Link: https://lore.kernel.org/r/20200109082000.26729-1-tiwai@suse.deSigned-off-by:
-
- 10 Jan, 2020 1 commit
-
-
Olivier Moysan authored
In stm32_afsdm_pcm_cb function, the transfer size is provided in bytes. However, samples are copied as 16 bits words from iio buffer. Divide by two the transfer size, to copy the right number of samples. Fixes: 1e7f6e1c ("ASoC: stm32: dfsdm: add 16 bits audio record support") Signed-off-by:
Olivier Moysan <olivier.moysan@st.com> Link: https://lore.kernel.org/r/20200110131131.3191-1-olivier.moysan@st.comSigned-off-by:
-
