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| * | | ASoC: Intel: avs: Access path components under lockAmadeusz Sławiński2023-05-221-1/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Path and its components should be accessed under lock to prevent problems with one thread modifying them while other tries to read. Fixes: c8c960c10971 ("ASoC: Intel: avs: APL-based platforms support") Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Link: https://lore.kernel.org/r/20230519201711.4073845-3-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| * | | ASoC: Intel: avs: Fix module lookupAmadeusz Sławiński2023-05-221-7/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When changing value of kcontrol, FW module to which data should be send needs to be found. Currently it is done in improper way, fix it. Change function name to indicate that it looks only for volume module. This allows to change volume during runtime, instead of only changing init value. Fixes: be2b81b519d7 ("ASoC: Intel: avs: Parse control tuples") Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Link: https://lore.kernel.org/r/20230519201711.4073845-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| * | | ASoC: soc-pcm: test if a BE can be preparedRanjani Sridharan2023-05-191-0/+20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the BE hw_params configuration, the existing code checks if any of the existing FEs are prepared, running, paused or suspended - and skips the configuration in those cases. This allows multiple calls of hw_params which the ALSA state machine supports. This check is not handled for the prepare stage, which can lead to the same BE being prepared multiple times. This patch adds a check similar to that of the hw_params, with the main difference being that the suspended state is allowed: the ALSA state machine allows a transition from suspended to prepared with hw_params skipped. This problem was detected on Intel IPC4/SoundWire devices, where the BE dailink .prepare stage is used to configure the SoundWire stream with a bank switch. Multiple .prepare calls lead to conflicts with the .trigger operation with IPC4 configurations. This problem was not detected earlier on Intel devices, HDaudio BE dailinks detect that the link is already prepared and skip the configuration, and for IPC3 devices there is no BE trigger. Link: https://github.com/thesofproject/sof/issues/7596 Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Link: https://lore.kernel.org/r/20230517185731.487124-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| * | | ASoC: rt5682: Disable jack detection interrupt during suspendMatthias Kaehlcke2023-05-173-1/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The rt5682 driver switches its regmap to cache-only when the device suspends and back to regular mode on resume. When the jack detect interrupt fires rt5682_irq() schedules the jack detect work. This can result in invalid reads from the regmap in cache-only mode if the work runs before the device has resumed: [ 56.245502] rt5682 9-001a: ASoC: error at soc_component_read_no_lock on rt5682.9-001a for register: [0x000000f0] -16 Disable the jack detection interrupt during suspend and re-enable it on resume. The driver already schedules the jack detection work on resume, so any state change during suspend is still handled. This is essentially the same as commit f7d00a9be147 ("SoC: rt5682s: Disable jack detection interrupt during suspend") for the rt5682s. Cc: stable@kernel.org Signed-off-by: Matthias Kaehlcke <mka@chromium.org Reviewed-by: Douglas Anderson <dianders@chromium.org Reviewed-by: Stephen Boyd <swboyd@chromium.org Link: https://lore.kernel.org/r/20230516164629.1.Ibf79e94b3442eecc0054d2b478779cc512d967fc@changeid Signed-off-by: Mark Brown <broonie@kernel.org
| * | | ASoC: lpass: Fix for KASAN use_after_free out of boundsRavulapati Vishnu Vardhan Rao2023-05-171-0/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When we run syzkaller we get below Out of Bounds error. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: BUG: KASAN: slab-out-of-bounds in regcache_flat_read+0x10c/0x110 Read of size 4 at addr ffffff8088fbf714 by task syz-executor.4/14144 CPU: 6 PID: 14144 Comm: syz-executor.4 Tainted: G W Hardware name: Qualcomm Technologies, Inc. sc7280 CRD platform (rev5+) (DT) Call trace: dump_backtrace+0x0/0x4ec show_stack+0x34/0x50 dump_stack_lvl+0xdc/0x11c print_address_description+0x30/0x2d8 kasan_report+0x178/0x1e4 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf8/0x5a0 _regmap_read+0x45c/0x86c _regmap_update_bits+0x128/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_put_dec_enum+0x1d0/0x268 snd_ctl_elem_write+0x288/0x474 By Error checking and checking valid values issue gets rectifies. Signed-off-by: Ravulapati Vishnu Vardhan Rao <quic_visr@quicinc.com Link: https://lore.kernel.org/r/20230511112532.16106-1-quic_visr@quicinc.com Signed-off-by: Mark Brown <broonie@kernel.org
* | | | ALSA: hda: Fix unhandled register update during auto-suspend periodTakashi Iwai2023-05-221-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | It's reported that the recording started right after the driver probe doesn't work properly, and it turned out that this is related with the codec auto-suspend. Namely, after the probe phase, the usage count goes zero, and the auto-suspend is programmed, but the codec is kept still active until the auto-suspend expiration. When an application (e.g. alsactl) updates the mixer values at this moment, the values are cached but not actually written. Then, starting arecord thereafter also results in the silence because of the missing unmute. The root cause is the handling of "lazy update" mode; when a mixer value is updated *after* the suspend, it should update only the cache and exits. At the resume, the cached value is written to the device, in turn. The problem is that the current code misinterprets the state of auto-suspend as if it were already suspended. Although we can add the check of the actual device state after pm_runtime_get_if_in_use() for catching the missing state, this won't suffice; the second call of regmap_update_bits_check() will skip writing the register because the cache has been already updated by the first call. So we'd need fixes in two different places. OTOH, a simpler fix is to replace pm_runtime_get_if_in_use() with pm_runtime_get_if_active() (with ign_usage_count=true). This change implies that the driver takes the pm refcount if the device is still in ACTIVE state and continues the processing. A small caveat is that this will leave the auto-suspend timer. But, since the timer callback itself checks the device state and aborts gracefully when it's active, this won't be any substantial problem. Long story short: we address the missing register-write problem just by replacing the pm_runtime_*() call in snd_hda_keep_power_up(). Fixes: fc4f000bf8c0 ("ALSA: hda - Fix unexpected resume through regmap code path") Reported-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Closes: https://lore.kernel.org/r/a7478636-af11-92ab-731c-9b13c582a70d@linux.intel.com Suggested-by: Cezary Rojewski <cezary.rojewski@intel.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230518113520.15213-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: hda/ca0132: add quirk for EVGA X299 DARKAdam Stylinski2023-05-221-0/+1
| |_|/ |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This quirk is necessary for surround and other DSP effects to work with the onboard ca0132 based audio chipset for the EVGA X299 dark mainboard. Signed-off-by: Adam Stylinski <kungfujesus06@gmail.com> Cc: <stable@vger.kernel.org> Link: https://bugzilla.kernel.org/show_bug.cgi?id=67071 Link: https://lore.kernel.org/r/ZGopOe19T1QOwizS@eggsbenedict.adamsnet Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda: Add NVIDIA codec IDs a3 through a7 to patch tableNikhil Mahale2023-05-171-0/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | These IDs are for AD102, AD103, AD104, AD106, and AD107 gpus with audio functions that are largely similar to the existing ones. Tested audio using gnome-settings, over HDMI, DP-SST and DP-MST connections on AD106 gpu. Signed-off-by: Nikhil Mahale <nmahale@nvidia.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230517090736.15088-1-nmahale@nvidia.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: oss: avoid missing-prototype warningsArnd Bergmann2023-05-171-8/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Two functions are defined and used in pcm_oss.c but also optionally used from io.c, with an optional prototype. If CONFIG_SND_PCM_OSS_PLUGINS is disabled, this causes a warning as the functions are not static and have no prototype: sound/core/oss/pcm_oss.c:1235:19: error: no previous prototype for 'snd_pcm_oss_write3' [-Werror=missing-prototypes] sound/core/oss/pcm_oss.c:1266:19: error: no previous prototype for 'snd_pcm_oss_read3' [-Werror=missing-prototypes] Avoid this by making the prototypes unconditional. Signed-off-by: Arnd Bergmann <arnd@arndb.de> Link: https://lore.kernel.org/r/20230516195046.550584-2-arnd@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: cs46xx: mark snd_cs46xx_download_image as staticArnd Bergmann2023-05-171-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_cs46xx_download_image() was originally called from dsp_spos.c, but is now local to cs46xx_lib.c. Mark it as 'static' to avoid a warning about it lacking a declaration, and '__maybe_unused' to avoid a warning about it being unused when CONFIG_SND_CS46XX_NEW_DSP is disabled: sound/pci/cs46xx/cs46xx_lib.c:534:5: error: no previous prototype for 'snd_cs46xx_download_image' Fixes: 89f157d9e6bf ("[ALSA] cs46xx - Fix PM resume") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Link: https://lore.kernel.org/r/20230516195046.550584-1-arnd@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda: Fix Oops by 9.1 surround channel namesTakashi Iwai2023-05-161-3/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | get_line_out_pfx() may trigger an Oops by overflowing the static array with more than 8 channels. This was reported for MacBookPro 12,1 with Cirrus codec. As a workaround, extend for the 9.1 channels and also fix the potential Oops by unifying the code paths accessing the same array with the proper size check. Reported-by: Olliver Schinagl <oliver@schinagl.nl> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/64d95eb0-dbdb-cff8-a8b1-988dc22b24cd@schinagl.nl Link: https://lore.kernel.org/r/20230516184412.24078-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | Merge tag 'asoc-fix-v6.4-rc2' of ↵Takashi Iwai2023-05-1620-71/+247
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v6.4 More fixes that came in since the merge window, the bulk of which are for the SOF code, I suspect as a result of the wide usage, active development and large code size rather than huge quality problems. There's also a couple of MAINTAINERS updates and some new device quirks.
| * | ASoC: SOF: Intel: hda-mlink: fixes and extensionsMark Brown2023-05-161-9/+87
| |\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>: With additional testing with multiple links and multiple DAI types, we found a couple of mistakes with refcounts, base address, missing initialization. A new helper was also added due to a change in the SoundWire programming sequences, with the host driver in charge of setting up the DMA channel mapping instead of the firmware.
| | * | ASoC: SOF: Intel: hda-mlink: add helper to program SoundWire PCMSyCM registersPierre-Louis Bossart2023-05-151-0/+50
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | These registers enable the HDaudio DMA hardware to split/merge data from different PDIs, possibly on different links. This capability exists for all types of HDaudio extended links, but for now is only required for SoundWire. In the SSP/DMIC case, the IP is programmed by the DSP firmware. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Rander Wang <rander.wang@intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| | * | ASoC: SOF: Intel: hda-mlink: initialize instance_offset memberPierre-Louis Bossart2023-05-151-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We defined the values but never initialized it for SoundWire/SSP, fix this miss. A Fixes: tag is not provided as instance_offset was not used so far, so nothing was really broken. This patch is only required for the SoundWire support in the following patch. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Rander Wang <rander.wang@intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| | * | ASoC: SOF: Intel: hda-mlink: use 'ml_addr' parameter consistentlyPierre-Louis Bossart2023-05-151-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We mix the use of hlink->ml_addr and the 'ml_addr' parameter. It's the same thing, let's align on using the parameter. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Rander Wang <rander.wang@intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| | * | ASoC: SOF: Intel: hda-mlink: fix base_ptr computationPierre-Louis Bossart2023-05-151-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The base_ptr value needs to be derived from the remap_addr pointer, not the ml_addr. This base_ptr was used only in debug logs that were so far not contributed upstream so the issue was not detected. It needs to be fixed for SoundWire support on LunarLake. Fixes: 17c9b6ec35c0 ("ASoC: SOF: Intel: hda-mlink: add structures to parse ALT links") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Rander Wang <rander.wang@intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| | * | ASoC: SOF: Intel: hda-mlink: add helper to get SoundWire hlinkPierre-Louis Bossart2023-05-151-0/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Same functionality as for DMIC/SSP with different ID. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Rander Wang <rander.wang@intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| | * | ASoC: SOF: Intel: hda-mlink: fix sublink refcountingPierre-Louis Bossart2023-05-151-5/+19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In hindsight it was a very bad idea to use the same refcount for Extended and 'legacy' HDaudio multi-links. The existing solution only powers-up the first sublink, which causes SoundWire and SSP tests to fail when more than one DAI is used concurrently. Solving this problem requires per-sublink refcounting, as suggested in this patch. The existing refcounting remains for 'legacy' HdAudio links, mainly to avoid changing the obscure programming sequence in snd_hdac_ext_bus_link_put(). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| * | | ASoC: SOF: topology: Fix tuples array allocationRanjani Sridharan2023-05-151-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The memory allocated for the tuples array assumes that there's 1 instance of all tokens already. So for those tokens that have multiple instances in topology, we need to exclude the initial instance that has already been accounted for. Fixes: 4fdef47a44d6 ("ASoC: SOF: ipc4-topology: Add new tokens for input/output pin format count") Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230515085200.17094-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| * | | ASoC: SOF: Separate the tokens for input and output pin indexRanjani Sridharan2023-05-151-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Using the same token ID for both input and output format pin index results in collisions and incorrect pin index getting parsed from topology. Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com Reviewed-by: Paul Olaru <paul.olaru@oss.nxp.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230515104403.32207-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| * | | ASoC: SOF: Various runtime pm fixes, improvementsMark Brown2023-05-153-13/+16
| |\ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>: Three patch to correct error path PM runtime handling in few places. Regards, Peter --- Pierre-Louis Bossart (3): ASoC: SOF: debug: conditionally bump runtime_pm counter on exceptions ASoC: SOF: pcm: fix pm_runtime imbalance in error handling ASoC: SOF: sof-client-probes: fix pm_runtime imbalance in error handling sound/soc/sof/debug.c | 4 ++-- sound/soc/sof/pcm.c | 11 ++++++----- sound/soc/sof/sof-client-probes.c | 14 ++++++++------ 3 files changed, 16 insertions(+), 13 deletions(-) -- 2.40.1
| | * | | ASoC: SOF: sof-client-probes: fix pm_runtime imbalance in error handlingPierre-Louis Bossart2023-05-151-6/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When an error occurs, we need to make sure the device can pm_runtime suspend instead of keeping it active. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| | * | | ASoC: SOF: pcm: fix pm_runtime imbalance in error handlingPierre-Louis Bossart2023-05-151-5/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When an error occurs, we need to make sure the device can pm_runtime suspend instead of keeping it active. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| | * | | ASoC: SOF: debug: conditionally bump runtime_pm counter on exceptionsPierre-Louis Bossart2023-05-151-2/+2
| | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When a firmware IPC error happens during a pm_runtime suspend, we ignore the error and suspend anyways. However, the code unconditionally increases the runtime_pm counter. This results in a confusing configuration where the code will suspend, resume but never suspend again due to the use of pm_runtime_get_noresume(). The intent of the counter increase was to prevent entry in D3, but if that transition to D3 is already started it cannot be stopped. In addition, there's no point in that case in trying to prevent anything, the firmware error is handled and the next resume will re-initialize the firmware completely. This patch changes the logic to prevent suspend when the device is pm_runtime active and has a use_count > 0. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| * | | ASoC: cs35l56: Prevent unbalanced pm_runtime in dsp_work() on SoundWireSimon Trimmer2023-05-151-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Flush the SoundWire interrupt handler work instead of cancelling it. When a SoundWire interrupt is triggered the pm_runtime is held until the work has completed. It's therefore unsafe to cancel the work, it must be flushed. Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com Link: https://lore.kernel.org/r/20230512144237.739000-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org
| * | | ASoC: SOF: topology: Fix logic for copying tuplesRanjani Sridharan2023-05-151-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Topology could have more instances of the tokens being searched for than the number of sets that need to be copied. Stop copying token after the limit of number of token instances has been reached. This worked before only by chance as we had allocated more size for the tuples array than the number of actual tokens being parsed. Fixes: 7006d20e5e9d ("ASoC: SOF: Introduce IPC3 ops") Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230512114630.24439-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| * | | ASoC: SOF: pm: save io region state in case of errors in resumeKai Vehmanen2023-05-151-1/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If there are failures in DSP runtime resume, the device state will not reach active and this makes it impossible e.g. to retrieve a possible DSP panic dump via "exception" debugfs node. If CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE=y is set, the data in cache is stale. If debugfs cache is not used, the region simply cannot be read. To allow debugging these scenarios, update the debugfs cache contents in resume error handler. User-space can then later retrieve DSP panic and other state via debugfs (requires SOF debugfs cache to be enabled in build). Reported-by: Curtis Malainey <cujomalainey@chromium.org Link: https://github.com/thesofproject/linux/issues/4274 Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Curtis Malainey <cujomalainey@chromium.org Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230512104638.21376-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| * | | ASoC: mediatek: mt8186: Fix use-after-free in driver remove pathDouglas Anderson2023-05-155-34/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When devm runs function in the "remove" path for a device it runs them in the reverse order. That means that if you have parts of your driver that aren't using devm or are using "roll your own" devm w/ devm_add_action_or_reset() you need to keep that in mind. The mt8186 audio driver didn't quite get this right. Specifically, in mt8186_init_clock() it called mt8186_audsys_clk_register() and then went on to call a bunch of other devm function. The caller of mt8186_init_clock() used devm_add_action_or_reset() to call mt8186_deinit_clock() but, because of the intervening devm functions, the order was wrong. Specifically at probe time, the order was: 1. mt8186_audsys_clk_register() 2. afe_priv->clk = devm_kcalloc(...) 3. afe_priv->clk[i] = devm_clk_get(...) At remove time, the order (which should have been 3, 2, 1) was: 1. mt8186_audsys_clk_unregister() 3. Free all of afe_priv->clk[i] 2. Free afe_priv->clk The above seemed to be causing a use-after-free. Luckily, it's easy to fix this by simply using devm more correctly. Let's move the devm_add_action_or_reset() to the right place. In addition to fixing the use-after-free, code inspection shows that this fixes a leak (missing call to mt8186_audsys_clk_unregister()) that would have happened if any of the syscon_regmap_lookup_by_phandle() calls in mt8186_init_clock() had failed. Fixes: 55b423d5623c ("ASoC: mediatek: mt8186: support audio clock control in platform driver") Signed-off-by: Douglas Anderson <dianders@chromium.org Link: https://lore.kernel.org/r/20230511092437.1.I31cceffc8c45bb1af16eb613e197b3df92cdc19e@changeid Signed-off-by: Mark Brown <broonie@kernel.org
| * | | ASoC: SOF: ipc3-topology: Make sure that only one cmd is sent in dai_configPeter Ujfalusi2023-05-151-2/+5
| |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The commands in sof_ipc_dai_config.flags are encoded as bits: 1 (bit0) - hw_params 2 (bit1) - hw_free 4 (bit2) - pause These are commands, they cannot be combined as one would assume, for example 3 (bit0 | bit1) is invalid. This can happen right at the second start of a stream as at the end of the first stream we set the hw_free command (bit1) and on the second start we would OR on top of it the hw_params (bit0). Fixes: b66bfc3a9810 ("ASoC: SOF: sof-audio: Fix broken early bclk feature for SSP") Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Link: https://lore.kernel.org/r/20230512110317.5180-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
| * | ASoC: ssm2602: Add workaround for playback distortionsPaweł Anikiel2023-05-121-0/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Apply a workaround for what appears to be a hardware quirk. The problem seems to happen when enabling "whole chip power" (bit D7 register R6) for the very first time after the chip receives power. If either "output" (D4) or "DAC" (D3) aren't powered on at that time, playback becomes very distorted later on. This happens on the Google Chameleon v3, as well as on a ZYBO Z7-10: https://ez.analog.com/audio/f/q-a/543726/solved-ssm2603-right-output-offset-issue/480229 I suspect this happens only when using an external MCLK signal (which is the case for both of these boards). Here are some experiments run on a Google Chameleon v3. These were run in userspace using a wrapper around the i2cset utility: ssmset() { i2cset -y 0 0x1a $(($1*2)) $2 } For each of the following sequences, we apply power to the ssm2603 chip, set the configuration registers R0-R5 and R7-R8, run the selected sequence, and check for distortions on playback. ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac OK ssmset 0x09 0x01 # core ssmset 0x06 0x87 # out, dac ssmset 0x06 0x07 # chip OK (disable MCLK) ssmset 0x09 0x01 # core ssmset 0x06 0x1f # chip ssmset 0x06 0x07 # out, dac (enable MCLK) OK ssmset 0x09 0x01 # core ssmset 0x06 0x1f # chip ssmset 0x06 0x07 # out, dac NOT OK ssmset 0x06 0x1f # chip ssmset 0x09 0x01 # core ssmset 0x06 0x07 # out, dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x0f # chip, out ssmset 0x06 0x07 # dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x17 # chip, dac ssmset 0x06 0x07 # out NOT OK For each of the following sequences, we apply power to the ssm2603 chip, run the selected sequence, issue a reset with R15, configure R0-R5 and R7-R8, run one of the NOT OK sequences from above, and check for distortions. ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac OK (disable MCLK) ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac (enable MCLK after reset) NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x17 # chip, dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x0f # chip, out NOT OK ssmset 0x06 0x07 # chip, out, dac NOT OK Signed-off-by: Paweł Anikiel <pan@semihalf.com Link: https://lore.kernel.org/r/20230508113037.137627-8-pan@semihalf.com Signed-off-by: Mark Brown <broonie@kernel.org
| * | ASoC: jz4740-i2s: Make I2S divider calculations more robustAidan MacDonald2023-05-111-4/+50
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When the CPU supplies bit/frame clocks, the system clock (clk_i2s) is divided to produce the bit clock. This is a simple 1/N divider with a fairly limited range, so for a given system clock frequency only a few sample rates can be produced. Usually a wider range of sample rates is supported by varying the system clock frequency. The old calculation method was not very robust and could easily produce the wrong clock rate, especially with non-standard rates. For example, if the system clock is 1.99x the target bit clock rate, the divider would be calculated as 1 instead of the more accurate 2. Instead, use a more accurate method that considers two adjacent divider settings and selects the one that produces the least error versus the requested rate. If the error is 5% or higher then the rate setting is rejected to prevent garbled audio. Skip divider calculation when the codec is supplying both the bit and frame clock; in that case, the divider outputs are unused and we don't want to constrain the sample rate. Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0@gmail.com Link: https://lore.kernel.org/r/20230509125134.208129-1-aidanmacdonald.0x0@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org
| * | ASoC: SOF: amd: Fix NULL pointer crash in acp_sof_ipc_msg_data functionV sujith kumar Reddy2023-05-091-1/+6
| | | | | | | | | | | | | | | | | | | | | | | | Check substream and runtime variables before assigning. Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy@amd.com Link: https://lore.kernel.org/r/20230508070510.6100-1-Vsujithkumar.Reddy@amd.com Signed-off-by: Mark Brown <broonie@kernel.org
| * | ASoC: fsl_micfil: Fix error handler with pm_runtime_enableShengjiu Wang2023-05-081-1/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There is error message when defer probe happens: fsl-micfil-dai 30ca0000.micfil: Unbalanced pm_runtime_enable! Fix the error handler with pm_runtime_enable and add fsl_micfil_remove() for pm_runtime_disable. Fixes: 47a70e6fc9a8 ("ASoC: Add MICFIL SoC Digital Audio Interface driver.") Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com Link: https://lore.kernel.org/r/1683540996-6136-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown <broonie@kernel.org
| * | ASoC: dwc: limit the number of overrun messagesMaxim Kochetkov2023-05-081-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | On slow CPU (FPGA/QEMU emulated) printing overrun messages from interrupt handler to uart console may leads to more overrun errors. So use dev_err_ratelimited to limit the number of error messages. Signed-off-by: Maxim Kochetkov <fido_max@inbox.ru Link: https://lore.kernel.org/r/20230505062820.21840-1-fido_max@inbox.ru Signed-off-by: Mark Brown <broonie@kernel.org
| * | ASoC: amd: yc: Add DMI entry to support System76 Pangolin 12Jeremy Soller2023-05-081-0/+7
| |/ | | | | | | | | | | | | | | | | Add pang12 quirk to enable the internal microphone. Signed-off-by: Jeremy Soller <jeremy@system76.com Signed-off-by: Tim Crawford <tcrawford@system76.com Link: https://lore.kernel.org/r/20230505161458.19676-1-tcrawford@system76.com Signed-off-by: Mark Brown <broonie@kernel.org
* | ALSA: hda/realtek: Fix mute and micmute LEDs for yet another HP laptopKai-Heng Feng2023-05-121-0/+1
| | | | | | | | | | | | | | | | | | | | There's yet another laptop that needs the fixup to enable mute and micmute LEDs. So do it accordingly. Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230512083417.157127-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek: Apply HP B&O top speaker profile to Pavilion 15Ryan C. Underwood2023-05-121-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | The Pavilion 15 line has B&O top speakers similar to the x360 and applying the same profile produces good sound. Without this, the sound would be tinny and underpowered without either applying model=alc295-hp-x360 or booting another OS first. Signed-off-by: Ryan Underwood <nemesis@icequake.net> Fixes: 563785edfcef ("ALSA: hda/realtek - Add quirk entry for HP Pavilion 15") Link: https://lore.kernel.org/r/ZF0mpcMz3ezP9KQw@icequake.net Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Add a sample rate workaround for Line6 Pod GoTakashi Iwai2023-05-121-0/+1
| | | | | | | | | | | | | | | | | | | | | | Line6 Pod Go (0e41:424b) requires the similar workaround for the fixed 48k sample rate like other Line6 models. This patch adds the corresponding entry to line6_parse_audio_format_rate_quirk(). Reported-by: John Humlick <john@humlick.org> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230512075858.22813-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: firewire-digi00x: prevent potential use after freeDan Carpenter2023-05-121-1/+3
| | | | | | | | | | | | | | | | | | | | | | This code was supposed to return an error code if init_stream() failed, but it instead freed dg00x->rx_stream and returned success. This potentially leads to a use after free. Fixes: 9a08067ec318 ("ALSA: firewire-digi00x: support AMDTP domain") Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org> Link: https://lore.kernel.org/r/c224cbd5-d9e2-4cd4-9bcf-2138eb1d35c6@kili.mountain Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek: Add quirks for ASUS GU604V and GU603VAlexandru Sorodoc2023-05-111-0/+20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | These models use 2 CS35L41 amplifiers using SPI for down-facing speakers. alc285_fixup_speaker2_to_dac1 is needed to fix volume control of the down-facing speakers. Pin configs are needed to enable headset mic detection. Note that these models lack the ACPI _DSD properties needed to initialize the amplifiers. They can be added during boot to get working sound out of the speakers: https://gist.github.com/lamperez/862763881c0e1c812392b5574727f6ff Signed-off-by: Alexandru Sorodoc <ealex95@gmail.com> Link: https://lore.kernel.org/r/20230511161510.315170-1-ealex95@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek: Add quirk for HP EliteBook G10 laptopsVitaly Rodionov2023-05-111-1/+7
| | | | | | | | | | | | | | | | | | | | Add support for HP EliteBook 835/845/845W/865 G10 laptops with CS35L41 amplifiers on I2C/SPI bus connected to Realtek codec. Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230510142227.32945-1-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek: Add a quirk for HP EliteDesk 805Ai Chao2023-05-081-0/+1
| | | | | | | | | | | | | | | | | | Add a quirk for HP EliteDesk 805 to fixup ALC3867 headset MIC no sound. Signed-off-by: Ai Chao <aichao@kylinos.cn> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230506022653.2074343-1-aichao@kylinos.cn Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek: Add quirk for 2nd ASUS GU603Luke D. Jones2023-05-081-0/+1
| | | | | | | | | | | | | | | | | | Add quirk for GU603 with 0x1c62 variant of codec. Signed-off-by: Luke D. Jones <luke@ljones.dev> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230505235824.49607-2-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek: Add quirk for Clevo L140AUJeremy Soller2023-05-081-0/+1
|/ | | | | | | | | | Fixes headset detection on Clevo L140AU. Signed-off-by: Jeremy Soller <jeremy@system76.com> Signed-off-by: Tim Crawford <tcrawford@system76.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230505163651.21257-1-tcrawford@system76.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge tag 'sound-fix-6.4-rc1' of ↵Linus Torvalds2023-05-0622-1115/+1325
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "A collection of small fixes for rc1. The only (LOC-wise) dominant change was ASoC Qualcomm fix, but most of it was merely a code shuffling. Another significant change here is for ALSA PCM core; it received a revert and a series of fixes for PCM auto-silencing where it caused a regression in the previous PR for rc1. Others are all small: ASoC Intel fixes, various quirks for ASoC AMD, HD-audio and USB-audio, the continued legacy emu10k1 code cleanup, and some documentation updates" * tag 'sound-fix-6.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (23 commits) ALSA: pcm: use exit controlled loop in snd_pcm_playback_silence() ALSA: pcm: simplify top-up mode init in snd_pcm_playback_silence() ALSA: pcm: playback silence - move silence variable updates to separate function ALSA: pcm: playback silence - remove extra code ALSA: pcm: fix playback silence - correct incremental silencing ALSA: pcm: fix playback silence - use the actual new_hw_ptr for the threshold mode ALSA: pcm: Revert "ALSA: pcm: rewrite snd_pcm_playback_silence()" ALSA: hda/realtek: Fix mute and micmute LEDs for an HP laptop ALSA: caiaq: input: Add error handling for unsupported input methods in `snd_usb_caiaq_input_init` ALSA: usb-audio: Add quirk for Pioneer DDJ-800 ALSA: hda/realtek: support HP Pavilion Aero 13-be0xxx Mute LED ASoC: Intel: soc-acpi-cht: Add quirk for Nextbook Ares 8A tablet ASoC: amd: yc: Add Asus VivoBook Pro 14 OLED M6400RC to the quirks list for acp6x ASoC: codecs: wcd938x: fix accessing regmap on unattached devices ALSA: docs: Fix code block indentation in ALSA driver example ALSA: docs: Extend module parameters description ALSA: hda/realtek: Add quirk for ASUS UM3402YAR using CS35L41 ALSA: emu10k1: use more existing defines instead of open-coded numbers ASoC: amd: yc: Add ASUS M3402RA into DMI table ALSA: hda/realtek: Add quirk for ThinkPad P1 Gen 6 ...
| * ALSA: pcm: use exit controlled loop in snd_pcm_playback_silence()Oswald Buddenhagen2023-05-051-2/+2
| | | | | | | | | | | | | | | | | | | | We already know that `frames` is greater than zero, because we just checked it. So we don't need to check the loop condition on the first iteration. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de> Link: https://lore.kernel.org/r/20230505155244.2312199-7-oswald.buddenhagen@gmx.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: pcm: simplify top-up mode init in snd_pcm_playback_silence()Oswald Buddenhagen2023-05-051-7/+24
| | | | | | | | | | | | | | | | | | | | Inline the remaining call of snd_pcm_playback_hw_avail(). This makes the top-up branch more congruent with the thresholded one, and allows simplifying the handling of the corner cases. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de> Link: https://lore.kernel.org/r/20230505155244.2312199-6-oswald.buddenhagen@gmx.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: pcm: playback silence - move silence variable updates to separate functionJaroslav Kysela2023-05-051-20/+22
| | | | | | | | | | | | | | | | | | | | | | | | The code tracking the added samples in thresholded mode and the code tracking the just played samples in top-up mode are semantically identical, so factor it out to a common function to enhance readability. Co-developed-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de> Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz> Link: https://lore.kernel.org/r/20230505155244.2312199-5-oswald.buddenhagen@gmx.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: pcm: playback silence - remove extra codeJaroslav Kysela2023-05-051-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | The removed condition handles de facto only one situation where runtime->silence_filled variable is equal to runtime->buffer_size, because this variable cannot go over the buffer size. This case is implicitly caught by the required comparison of the noise distance with the threshold. Suggested-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de> Link: https://lore.kernel.org/r/20230505155244.2312199-4-oswald.buddenhagen@gmx.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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