January 2021 - Alsa-devel - mailman.alsa-project.org

Wrong binds configuration for dmix plugin causes segmentation fault in alsa
by GitHub issues - opened 22 Jan '21

22 Jan '21

alsa-project/alsa-lib issue #117 was opened from elara-leitstellentechnik: On Debian buster I can get alsa to run into a segmentation fault with the following procedure: Configure dmix with two channels (binds) for a Mono device. (I am not sure that this is reproducible with any mono device, I tested with a Jabra LINK 230): ``` ~$ cat .asoundrc pcm."Jabra LINK 230" { type dmix ipc_key 1024 slave.pcm "hw:2,0" bindings { 0 0 # The next line is the problematic line that triggers the segfault 1 1 } } ``` Then when playing any sound file with e.g. mplayer and making it use the dmix device the following happens: ``` $ mplayer -ao alsa:device="Jabra LINK 230" /usr/share/sounds/alsa/Front_Left.wav MPlayer 1.3.0 (Debian), built with gcc-8 (C) 2000-2016 MPlayer Team do_connect: could not connect to socket connect: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing /usr/share/sounds/alsa/Front_Left.wav. libavformat version 58.20.100 (external) Mismatching header version 58.12.100 Audio only file format detected. Load subtitles in /usr/share/sounds/alsa/ ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder AUDIO: 48000 Hz, 1 ch, s16le, 768.0 kbit/100.00% (ratio: 96000->96000) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== [AO_ALSA] alsa-lib: pcm_hw.c:1711:(snd_pcm_hw_open) open '/dev/snd/pcmC2D0p' failed (-77): File descriptor in bad state AO: [alsa] 48000Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... MPlayer interrupted by signal 11 in module: play_audio - MPlayer crashed by bad usage of CPU/FPU/RAM. Recompile MPlayer with --enable-debug and make a 'gdb' backtrace and disassembly. Details in DOCS/HTML/en/bugreports_what.html#bugreports_crash. [ This binary of MPlayer in Debian is currently compiled with '--enable-debug'; the debugging symbols are in the package 'mplayer-dbgsym'.] ``` I appreciate that configuring two bindings for a single-channel audio device is an error, but I am not sure that this justifies a segfault. Here is some more information I have been able to extract using gdb: ``` ~$ gdb --args mplayer -ao alsa:device="Jabra LINK 230" /usr/share/sounds/alsa/Front_Left.wav GNU gdb (Debian 8.2.1-2+b3) 8.2.1 Copyright (C) 2018 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-linux-gnu". Type "show configuration" for configuration details. For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>. Find the GDB manual and other documentation resources online at: <http://www.gnu.org/software/gdb/documentation/>. For help, type "help". Type "apropos word" to search for commands related to "word"... Reading symbols from mplayer...Reading symbols from /usr/lib/debug/.build-id/3a/9816e9d363979a4f76e6cb809a33dffda67d05.debug...done. done. (gdb) run Starting program: /usr/bin/mplayer -ao alsa:device=Jabra\ LINK\ 230 /usr/share/sounds/alsa/Front_Left.wav [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". MPlayer 1.3.0 (Debian), built with gcc-8 (C) 2000-2016 MPlayer Team do_connect: could not connect to socket connect: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing /usr/share/sounds/alsa/Front_Left.wav. libavformat version 58.20.100 (external) Mismatching header version 58.12.100 Audio only file format detected. Load subtitles in /usr/share/sounds/alsa/ ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder AUDIO: 48000 Hz, 1 ch, s16le, 768.0 kbit/100.00% (ratio: 96000->96000) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== [AO_ALSA] alsa-lib: pcm_hw.c:1711:(snd_pcm_hw_open) open '/dev/snd/pcmC2D0p' failed (-77): File descriptor in bad state AO: [alsa] 48000Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... Program received signal SIGSEGV, Segmentation fault. 0x00007ffff7a09b76 in mix_areas_16_smp (size=4160473088, size@entry=17424, dst=0x555555aa9510, src=0xde020109, sum=0x21fe0f3c, dst_step=2, src_step=4, sum_step=8) at pcm_dmix_x86_64.h:48 48 pcm_dmix_x86_64.h: No such file or directory. (gdb) where #0 0x00007ffff7a09b76 in mix_areas_16_smp (size=4160473088, size@entry=17424, dst=0x555555aa9510, src=0xde020109, sum=0x21fe0f3c, dst_step=2, src_step=4, sum_step=8) at pcm_dmix_x86_64.h:48 #1 0x00007ffff7a0ba1c in mix_areas (size=17424, dst_ofs=576, src_ofs=0, dst_areas=0x555555a927f0, src_areas=0x7fffe88b3000, dmix=0x555555a929e0) at pcm_dmix.c:215 #2 snd_pcm_dmix_sync_area (pcm=pcm@entry=0x555555a94850) at pcm_dmix.c:382 #3 0x00007ffff7a0bdfa in snd_pcm_dmix_start (pcm=0x555555a94850) at pcm_dmix.c:619 #4 0x00007ffff79d8b4a in __snd_pcm_start (pcm=0x555555a94850, pcm=0x555555a94850) at pcm_local.h:434 #5 snd1_pcm_write_areas (pcm=pcm@entry=0x555555a94850, areas=areas@entry=0x7fffffffcfd0, offset=offset@entry=0, size=<optimized out>, size@entry=18000, func=func@entry=0x7ffff79e46c0 <snd_pcm_mmap_write_areas>) at pcm.c:7421 #6 0x00007ffff79e4b01 in snd_pcm_mmap_writei (pcm=0x555555a94850, buffer=<optimized out>, size=18000) at pcm_mmap.c:153 #7 0x00005555556242f8 in play (data=0x555555ac4f70, len=<optimized out>, flags=<optimized out>) at libao2/ao_alsa.c:812 #8 0x000055555560bbab in fill_audio_out_buffers () at mplayer.c:2212 #9 main (argc=<optimized out>, argv=<optimized out>) at mplayer.c:3794 (gdb) ``` To me this looks like the segfault is triggered in pcm_dmix_x86_64.h:48 so I believe it is an alsa problem rather than a problem with mplayer. I hope this makes sense. If I can support with further information please let me know. Issue URL : https://github.com/alsa-project/alsa-lib/issues/117 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 0/4] ASoC: mediatek: mt8192-mt6359: fix I2S corrupted for RT5682
by Tzung-Bi Shih 22 Jan '21

22 Jan '21

The series fixes the I2S corrupted when mt8192-mt6359 is playing and capturing via RT5682 in different formats. The 1st and 2nd patch refactor to use asoc_substream_to_rtd(). Just realized there is such a helper for the purpose. The 3rd patch simplies ops of Capture1 DAI link. The 4th patch adds format constriants to the I2S to make sure the settings are symmetric. Tzung-Bi Shih (4): ASoC: mediatek: mt8192-mt6359: use asoc_substream_to_rtd() ASoC: mediatek: mt8192: use asoc_substream_to_rtd() ASoC: mediatek: mt8192-mt6359: simply ops for Capture1 DAI link ASoC: mediatek: mt8192-mt6359: add format constraints for RT5682 sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 4 +- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 73 +++++++++++++++---- 2 files changed, 61 insertions(+), 16 deletions(-) -- 2.30.0.280.ga3ce27912f-goog

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[PATCH v2 00/12] MFD/extcon/ASoC: Rework arizona codec jack-detect support
by Hans de Goede 22 Jan '21

22 Jan '21

Hi All, This series reworks the arizona codec jack-detect support to use the snd_soc_jack helpers instead of direct extcon reporting. This is done by reworking the extcon driver into an arizona-jackdet library and then modifying the codec drivers to use that directly, replacing the old separate extcon child-devices and extcon-driver. This brings the arizona-codec jack-detect handling inline with how all other ASoC codec driver do this. This was developed and tested on a Lenovo Yoga Tablet 1051L with a WM5102 codec. The MFD, ASoC and extcon parts can be merged independent from each-other although that could lead to a case where both the extcon driver and the new arizona-jackdet library will try to do jack-detection. If we end up with a git tree in that state then one of the 2 will fail to load because the other will already have claimed the IRQs, so this is not a problem really. Or the entire series could be merged through the MFD tree if people prefer that. Note that this series also paves the way for some further cleanups, removing some jackdetect related variables like hp_ena and hp_clamp from the arizona data struct shared between all the MFD child devices. I've deliberately not done that cleanup as part of this patch-series, since IMHO the series is big enough as is. These cleanups can be done in a follow-up series once this series has landed. Regards, Hans

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[RFC][PATCH v5 0/2] audio jack software injection
by Hui Wang 22 Jan '21

22 Jan '21

the changes in the v5: - add a Kconfig to enable/disable the jack injection feature - replace all kzalloc with local char[] in the debugfs fops - replace the chars of !isalnum() to '_' for the jack folder's name - squash all .c files patches into one patch - add a document to explain jack injection, for easier review, put the document in a standalone patch. If needed, will squash this patch to the previous one. the changes in the v4: - change the sound-core to sound and change the sound_core_debugfs_root to sound_debugfs_root in the 0001-xxx.patch - change kzalloc/kfree to kvzalloc/kvfree in the debugfs fops for 0001-xxx.patch and 0003-xxx.patch - And if needed, I could squash 4 patches into 1 patch before merging. the changes in the v3 (for easy to review, divide change into 4 patches): - address the comment about the snd_jack_report() and _snd_jack_report(), the v2 design is hard to understand and is hard to review, in the v3, separate the jack_report to snd_jack_report() and snd_jack_inject_report(), hw jack events call snd_jack_report() as before, if a jack contains multi jack_kctl and the jack_kctl's sw_inject is enabled, the status and the related input-dev's events will not be reproted. The injection events call snd_jack_inject_report(). This change is squashed to 0001-xxx.patch, the rest part of 0001-xxx.patch is same as v2. - address the comment about folders'name in the 0002-xxx.patch, so far, drop the '/', ',', '=' and ' ' from the folders' name. - address the comment about adding more debugfs nodes in the 0003-xxx.patch, it adds kctl_id, mask_bits, status and type. - address the comment about save-n-restore jack's hw status in the 0004-xxx.patch, adding a hw_status_cache and save the last reported jack hw event, once the sw_inject is disabled, will restore all jack_kctl's state under the same snd_jack with hw_status_cache. [snip] the changes in the V2: - using debugfs instead of sysfs - using jack_ctrl to create a folder instead of snd_jack, since ASoC drivers could create multi jack_ctrls within a snd_jack - create a folder for each jack_ctrl instead for all jack_ctrls [ snip ] Hui Wang (2): alsa: jack: implement software jack injection via debugfs alsa: jack: add document for sound jack injection. Documentation/sound/designs/index.rst | 1 + .../sound/designs/jack-injection.rst | 124 ++++++++ include/sound/core.h | 2 + include/sound/jack.h | 1 + sound/core/Kconfig | 9 + sound/core/init.c | 10 + sound/core/jack.c | 298 +++++++++++++++++- sound/core/sound.c | 8 + 8 files changed, 449 insertions(+), 4 deletions(-) create mode 100644 Documentation/sound/designs/jack-injection.rst -- 2.25.1

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[PATCH] ASoC: qcom: Fix number of HDMI RDMA channels on sc7180
by Stephen Boyd 22 Jan '21

22 Jan '21

Suspending/resuming with an HDMI dongle attached leads to crashes from an audio regmap. Unable to handle kernel paging request at virtual address ffffffc018068000 Mem abort info: ESR = 0x96000047 EC = 0x25: DABT (current EL), IL = 32 bits SET = 0, FnV = 0 EA = 0, S1PTW = 0 Data abort info: ISV = 0, ISS = 0x00000047 CM = 0, WnR = 1 swapper pgtable: 4k pages, 39-bit VAs, pgdp=0000000081b12000 [ffffffc018068000] pgd=0000000275d14003, pud=0000000275d14003, pmd=000000026365d003, pte=0000000000000000 Internal error: Oops: 96000047 [#1] PREEMPT SMP Call trace: regmap_mmio_write32le+0x2c/0x40 regmap_mmio_write+0x48/0x6c _regmap_bus_reg_write+0x34/0x44 _regmap_write+0x100/0x150 regcache_default_sync+0xc0/0x138 regcache_sync+0x188/0x26c lpass_platform_pcmops_resume+0x48/0x54 [snd_soc_lpass_platform] snd_soc_component_resume+0x28/0x40 soc_resume_deferred+0x6c/0x178 process_one_work+0x208/0x3c8 worker_thread+0x23c/0x3e8 kthread+0x144/0x178 ret_from_fork+0x10/0x18 Code: d503201f d50332bf f94002a8 8b344108 (b9000113) I can reliably reproduce this problem by running 'tail' on the registers file in debugfs for the hdmi regmap. # tail /sys/kernel/debug/regmap/62d87000.lpass-lpass_hdmi/registers [ 84.658733] Unable to handle kernel paging request at virtual address ffffffd0128e800c This crash happens because we're trying to read registers from the regmap beyond the length of the mapping created by ioremap(). The number of hdmi_rdma_channels determines the size of the regmap via this code in sound/soc/qcom/lpass-cpu.c: lpass_hdmi_regmap_config.max_register = LPAIF_HDMI_RDMAPER_REG(variant, variant->hdmi_rdma_channels); According to debugfs the size of the regmap is 0x68010 but according to the DTS file posted in [1] the size is only 0x68000 (see the first reg property of the lpass_cpu node). Let's change the number of channels to be 3 instead of 4 so the math works out to have a max register of 0x67010, nicely fitting inside of the region size of 0x68000. Note: I tried to bump up the size of the register region to the next page to include the 0x68010 register but then the tail command caused SErrors with an async abort, implying that the register region doesn't exist or it isn't clocked because the bus is telling us that the register read failed. I reduce the number of channels and played audio through the HDMI channel and it kept working so I think this is correct. Fixes: 2ad63dc8df6b ("ASoC: qcom: sc7180: Add support for audio over DP") Link: https://lore.kernel.org/r/1601448168-18396-2-git-send-email-srivasam@codeau… [1] Cc: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Cc: Srinivasa Rao <srivasam(a)codeaurora.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Stephen Boyd <swboyd(a)chromium.org> --- sound/soc/qcom/lpass-sc7180.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 85db650c2169..3127b6022ea3 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -174,7 +174,7 @@ static struct lpass_variant sc7180_data = { .rdma_channels = 5, .hdmi_rdma_reg_base = 0x64000, .hdmi_rdma_reg_stride = 0x1000, - .hdmi_rdma_channels = 4, + .hdmi_rdma_channels = 3, .dmactl_audif_start = 1, .wrdma_reg_base = 0x18000, .wrdma_reg_stride = 0x1000, base-commit: 5c8fe583cce542aa0b84adc939ce85293de36e5e -- https://chromeos.dev

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[PATCH v1 0/2] isa: Make the remove callback for isa drivers return void
by Uwe Kleine-König 22 Jan '21

22 Jan '21

Hello, as described in the commit log of the 2nd patch returning an error code from a bus' remove callback doesn't make any difference as the driver core ignores it and still considers the device removed. So change the remove callback to return void to not give driver authors an incentive to believe they could return an error. There is only a single isa driver in the tree (assuming I didn't miss any) that has a remove callback that can return a non zero return code. This is "fixed" in the first patch, to make the second patch more obviously correct. Best regards Uwe Uwe Kleine-König (2): watchdog: pcwd: drop always-false if from remove callback isa: Make the remove callback for isa drivers return void drivers/base/isa.c | 2 +- drivers/i2c/busses/i2c-elektor.c | 4 +--- drivers/i2c/busses/i2c-pca-isa.c | 4 +--- drivers/input/touchscreen/htcpen.c | 4 +--- drivers/media/radio/radio-sf16fmr2.c | 4 +--- drivers/net/can/sja1000/tscan1.c | 4 +--- drivers/net/ethernet/3com/3c509.c | 3 +-- drivers/scsi/advansys.c | 3 +-- drivers/scsi/aha1542.c | 3 +-- drivers/scsi/fdomain_isa.c | 3 +-- drivers/scsi/g_NCR5380.c | 3 +-- drivers/watchdog/pcwd.c | 7 +------ include/linux/isa.h | 2 +- sound/isa/ad1848/ad1848.c | 3 +-- sound/isa/adlib.c | 3 +-- sound/isa/cmi8328.c | 3 +-- sound/isa/cmi8330.c | 3 +-- sound/isa/cs423x/cs4231.c | 3 +-- sound/isa/cs423x/cs4236.c | 3 +-- sound/isa/es1688/es1688.c | 3 +-- sound/isa/es18xx.c | 3 +-- sound/isa/galaxy/galaxy.c | 3 +-- sound/isa/gus/gusclassic.c | 3 +-- sound/isa/gus/gusextreme.c | 3 +-- sound/isa/gus/gusmax.c | 3 +-- sound/isa/gus/interwave.c | 3 +-- sound/isa/msnd/msnd_pinnacle.c | 3 +-- sound/isa/opl3sa2.c | 3 +-- sound/isa/opti9xx/miro.c | 3 +-- sound/isa/opti9xx/opti92x-ad1848.c | 3 +-- sound/isa/sb/jazz16.c | 3 +-- sound/isa/sb/sb16.c | 3 +-- sound/isa/sb/sb8.c | 3 +-- sound/isa/sc6000.c | 3 +-- sound/isa/sscape.c | 3 +-- sound/isa/wavefront/wavefront.c | 3 +-- 36 files changed, 36 insertions(+), 79 deletions(-) base-commit: 5a158981aafa7f29709034b17bd007b15cb29983 -- 2.29.2

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[PATCH] ALSA: hda/realtek: Enable headset of ASUS B1400CEPE with ALC256
by Jian-Hong Pan 22 Jan '21

22 Jan '21

ASUS B1400CEPE laptop's headset audio is not enabled until ALC256_FIXUP_ASUS_HPE quirk is applied. Here is the original pin node values: 0x12 0x40000000 0x13 0x411111f0 0x14 0x90170110 0x18 0x411111f0 0x19 0x411111f0 0x1a 0x411111f0 0x1b 0x411111f0 0x1d 0x40461b45 0x1e 0x411111f0 0x21 0x04211020 Signed-off-by: Jian-Hong Pan <jhp(a)endlessos.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ed5b6b894dc1..290645516313 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8006,6 +8006,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE), + SND_PCI_QUIRK(0x1043, 0x1982, "ASUS B1400CEPE", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 2.30.0

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[PATCH 1/2] ALSA: oxfw: fix info leak in hwdep_read()
by Dan Carpenter 22 Jan '21

22 Jan '21

If "count" is too large and "oxfw->dev_lock_changed" is false then this will copy beyond the end of the struct to user space. Fixes: 8985f4ac1c42 ("ALSA: oxfw: Add hwdep interface") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/firewire/oxfw/oxfw-hwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/oxfw/oxfw-hwdep.c b/sound/firewire/oxfw/oxfw-hwdep.c index 9e1b3e151bad..ca5477eeb663 100644 --- a/sound/firewire/oxfw/oxfw-hwdep.c +++ b/sound/firewire/oxfw/oxfw-hwdep.c @@ -35,12 +35,12 @@ static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, } memset(&event, 0, sizeof(event)); + count = min_t(long, count, sizeof(event.lock_status)); if (oxfw->dev_lock_changed) { event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; event.lock_status.status = (oxfw->dev_lock_count > 0); oxfw->dev_lock_changed = false; - count = min_t(long, count, sizeof(event.lock_status)); } spin_unlock_irq(&oxfw->lock); -- 2.29.2

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duplicate line at samsung speyside driver
by Kuninori Morimoto 22 Jan '21

22 Jan '21

Hi ALSA ML I noticed samsung speyside driver has duplicate line on audio_paths. I think it is typo or something... linux/sound/soc/samsung/speyside.c static struct snd_soc_dapm_route audio_paths[] = { => { "IN1RN", NULL, "MICB1" }, { "IN1RP", NULL, "MICB1" }, { "IN1RN", NULL, "MICB2" }, { "IN1RP", NULL, "MICB2" }, { "MICB1", NULL, "Headset Mic", speyside_get_micbias }, { "MICB2", NULL, "Headset Mic", speyside_get_micbias }, { "IN1LP", NULL, "MICB2" }, => { "IN1RN", NULL, "MICB1" }, { "MICB2", NULL, "Main AMIC" }, ... Thank you for your help !! Best regards --- Kuninori Morimoto

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[Regression] Game "Rise of the Tomb Raider" crashes at launch
by GitHub issues - edited 21 Jan '21

21 Jan '21

alsa-project/alsa-lib issue #116 was edited from Oschowa: The the native linux version of the game "Rise of the Tomb Raider" crashes at launch when it doesn't detect a working pulseaudio server and it falls back to alsa-lib directly. This is a regression, the last working alsa-lib version is v1.2.1.2 and bisection yields 5ee5ef31b5ff3f as the first bad commit. The crash happens when using "real" alsa hardware drivers as well as when running on the pulseaudio-alsa or pipewire-alsa software plug. Happy to provide more info if i can. Issue URL : https://github.com/alsa-project/alsa-lib/issues/116 Repository URL: https://github.com/alsa-project/alsa-lib

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