June 2020 - Alsa-devel - mailman.alsa-project.org

Re: KASAN: use-after-free Read in line6_submit_audio_in_all_urbs
by Alan Stern 10 Jul '20

10 Jul '20

On Fri, Jun 26, 2020 at 10:18:12AM -0700, syzbot wrote: > Hello, > > syzbot found the following crash on: > > HEAD commit: fb574682 usbip: tools: fix module name in man page > git tree: https://git.kernel.org/pub/scm/linux/kernel/git/gregkh/usb.git usb-testing > console output: https://syzkaller.appspot.com/x/log.txt?x=156560b1100000 > kernel config: https://syzkaller.appspot.com/x/.config?x=63b40b2ae167bad6 > dashboard link: https://syzkaller.appspot.com/bug?extid=145012a46658ac00fc9e > compiler: gcc (GCC) 10.1.0-syz 20200507 > > Unfortunately, I don't have any reproducer for this crash yet. > > IMPORTANT: if you fix the bug, please add the following tag to the commit: > Reported-by: syzbot+145012a46658ac00fc9e(a)syzkaller.appspotmail.com > > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > ================================================================== > BUG: KASAN: use-after-free in line6_submit_audio_in_all_urbs+0x10b/0x120 sound/usb/line6/capture.c:72 > Read of size 8 at addr ffff8881cffb1800 by task kworker/1:5/3257 > > CPU: 1 PID: 3257 Comm: kworker/1:5 Not tainted 5.8.0-rc1-syzkaller #0 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 > Workqueue: events line6_startup_work > Call Trace: > __dump_stack lib/dump_stack.c:77 [inline] > dump_stack+0xf6/0x16e lib/dump_stack.c:118 > print_address_description.constprop.0+0x1a/0x210 mm/kasan/report.c:383 > __kasan_report mm/kasan/report.c:513 [inline] > kasan_report.cold+0x37/0x7c mm/kasan/report.c:530 > line6_submit_audio_in_all_urbs+0x10b/0x120 sound/usb/line6/capture.c:72 > line6_stream_start+0x207/0x230 sound/usb/line6/pcm.c:197 > line6_pcm_acquire+0x161/0x210 sound/usb/line6/pcm.c:318 > line6_startup_work+0x42/0x50 sound/usb/line6/driver.c:734 > process_one_work+0x94c/0x15f0 kernel/workqueue.c:2269 > worker_thread+0x64c/0x1120 kernel/workqueue.c:2415 > kthread+0x392/0x470 kernel/kthread.c:291 > ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:293 > > Allocated by task 76: > save_stack+0x1b/0x40 mm/kasan/common.c:48 > set_track mm/kasan/common.c:56 [inline] > __kasan_kmalloc.constprop.0+0xc2/0xd0 mm/kasan/common.c:494 > kmalloc include/linux/slab.h:555 [inline] > kzalloc include/linux/slab.h:669 [inline] > line6_init_pcm+0x2a7/0x9e0 sound/usb/line6/pcm.c:533 > toneport_init+0xdd/0x6b0 sound/usb/line6/toneport.c:419 > line6_probe+0xaa0/0x1330 sound/usb/line6/driver.c:809 It look like the cancel_delayed_work() in line6_disconnect() needs to be cancel_delayed_work_sync(). Unfortunately we can't test this until syzbot is able to reproduce the bug. Alan Stern

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echoaudio: Fix some long standing bugs
by Mark Hills 09 Jul '20

09 Jul '20

Here follows some patches for the echoaudio PCM handling. Tested on the Layla3G on x86-64. I do have a Darla somewhere, but it's not practical for me to test with that right now. -- Mark

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Fixing sound on Asus UX534F / some UX533 models
by Hans de Goede 09 Jul '20

09 Jul '20

Hi Kai-Heng Feng, Takashi, I see that you are on the notification list for this bug: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1850439 So you may have already seen this. Some owners of affected laptops (which have non working speakers / headphone output atm), have done some heroic debugging work and come up with a set of 2 hda-verb commands which fix this. I'm not all that familiar with writing hda quirks, so I was hoping that one of you 2 can come up with a patch to fix this at the kernel level. This would also resolve these 2 bugs, which I believe are the same bug really: https://bugzilla.kernel.org/show_bug.cgi?id=206289 https://bugzilla.redhat.com/show_bug.cgi?id=1834751 Regards, Hans

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Ignored USB-audio implicit feedback in kernel 5.8rc3
by Pavel Hofman 07 Jul '20

07 Jul '20

Hi, Audio analyzer RTX6001 (XMOS-based) is using implicit feedback, yet the feedback is not used by the latest kernel 5.8 rc3 (i.e. already with the latest implicit-feedback patches). Two analyzers on two different PCs are getting clicks in duplex-mode loobpack, one every 10 secs, the other one every 50 secs. While running aplay/arecord loopback I captured USB packets with tcpdump/tshark. The dump contained 931484 IN packets with 48bytes payload, and 684 IN packets with 40 bytes payload. Packet interval 125us (bInterval = 1), 32bits, 2ch => precise samplerate 47,994.1298Hz. Clearly the incoming stream works OK. Yet stream0 file states for Capture: Momentary freq = 48000 Hz (0x6.0000) But ONLY 48bytes packets were sent by the host in OUT packets, i.e. exactly 48kHz output rate. 6 frames OUT/IN difference per second, the incoming FIFO in the device overflows after several seconds. Please may I ask for help with making the driver respect the implicit feedback stream? Best regards, Pavel. 5.8.0-050800-generic #202006282330 SMP Sun Jun 28 23:35:41 UTC 2020 x86_64 GNU/Linux cat /proc/asound/card4/stream0 RTX A/S RTX6001 USB Audio 2.0 at usb-0000:00:13.2-4, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 72 Momentary freq = 48000 Hz (0x6.0000) Interface 1 Altset 1 Format: S32_LE Channels: 2 Endpoint: 1 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 32 Channel map: FL FR Interface 1 Altset 2 Format: S32_LE Channels: 2 Endpoint: 1 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 24 Channel map: FL FR Interface 1 Altset 3 Format: S16_LE Channels: 2 Endpoint: 1 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 16 Channel map: FL FR Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 2 Endpoint: 1 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 32 Channel map: FL FR Interface 2 Altset 2 Format: S32_LE Channels: 2 Endpoint: 1 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 24 Channel map: FL FR Interface 2 Altset 3 Format: S16_LE Channels: 2 Endpoint: 1 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 16 Channel map: FL FR lsusb: Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 007 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 006 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 002 Device 002: ID 058f:6362 Alcor Micro Corp. Flash Card Reader/Writer Bus 002 Device 003: ID 0d9a:00df RTX AS RTX6001 USB Audio 2.0 <---- Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 005 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 008: ID 2047:0301 Texas Instruments Arduino Micro Bus 004 Device 007: ID 18f8:0f97 [Maxxter] Elo TouchSystems 2216 AccuTouch® USB Touchmonitor Interface Bus 004 Device 006: ID 413c:2001 Dell Computer Corp. Keyboard HID Support Bus 004 Device 005: ID 413c:1001 Dell Computer Corp. Keyboard Hub Bus 004 Device 004: ID 0bd9:0001 Liberty Instruments, Inc. Liberty Praxis AudPod 1.1 Bus 004 Device 003: ID 2341:8037 Arduino SA Arduino Micro Bus 004 Device 002: ID 04e7:0050 Elo TouchSystems 2216 AccuTouch® Touchmonitor Interface Bus 004 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 009 Device 002: ID 18a5:0422 Verbatim, Ltd Portable SSD Bus 009 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 008 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub lsusb -v see the attachment

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[PATCHv3 0/5] ASoC: da7213: support for usage with simple-card
by Sebastian Reichel 07 Jul '20

07 Jul '20

Hi, This extends the da7213 driver to be used with simple-audio-card in combination with a fixed clock. Here is a snippet of the downstream board's DT, that is supposed to be supported by this patchset. --------------------------------------------------------------------- / { sound { compatible = "simple-audio-card"; simple-audio-card,name = "audio-card"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&dailink_master>; simple-audio-card,frame-master = <&dailink_master>; simple-audio-card,widgets = "Speaker", "Ext Spk"; simple-audio-card,audio-routing = "Ext Spk", "LINE"; simple-audio-card,cpu { sound-dai = <&ssi1>; }; dailink_master: simple-audio-card,codec { sound-dai = <&codec>; }; }; clk_ext_audio_codec: clock-codec { compatible = "fixed-clock"; #clock-cells = <0>; clock-frequency = <12288000>; }; }; &i2c1 { codec: audio-codec@1a { compatible = "dlg,da7212"; reg = <0x1a>; #sound-dai-cells = <0>; VDDA-supply = <&reg_2v5_audio>; VDDSP-supply = <&reg_5v0_audio>; VDDMIC-supply = <&reg_3v3_audio>; VDDIO-supply = <&reg_3v3_audio>; clocks = <&clk_ext_audio_codec>; clock-names = "mclk"; }; }; --------------------------------------------------------------------- This is mostly a resend of PATCHv2. There was quite a bit of discussion for the last patch. As far as I can see no better solution was found and the patch is fine as is. Sorry for the delay in sending another revision. Changes since PATCHv2: * https://lore.kernel.org/alsa-devel/20191120152406.2744-1-sebastian.reichel@… * dropped patch converting DA7213 into selectable Kconfig option (merged) * fix compatible string in patch 1 (DT binding) as pointed out by Adam Thomson * collected Reviewed-by from Adam Thomson for patch 2-4 Changes since PATCHv1: * https://lore.kernel.org/alsa-devel/20191108174843.11227-1-sebastian.reichel… * add patch adding da7212 compatible to DT bindings * update regulator patch, so that VDDA is enabled together with VDDIO while the device is enabled to avoid device reset * update clock patch, so that automatic PLL handling is not enabled when PLL is configured manually * update clock patch, so that automatic PLL is disabled when the device is suspended * update clock patch, so that automatic PLL is configured into bypass mode when possible -- Sebastian Sebastian Reichel (5): ASoC: da7213: Add da7212 DT compatible ASoC: da7213: Add regulator support ASoC: da7213: move set_sysclk to codec level ASoC: da7213: move set_pll to codec level ASoC: da7213: add default clock handling .../devicetree/bindings/sound/da7213.txt | 8 +- sound/soc/codecs/da7213.c | 172 ++++++++++++++++-- sound/soc/codecs/da7213.h | 11 ++ 3 files changed, 175 insertions(+), 16 deletions(-) -- 2.26.2

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[PATCH v3 0/2] Introduce multi PM domains helpers
by Daniel Baluta 07 Jul '20

07 Jul '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> i.MX8QXP/i.MX8QM has IPs that need multiple power domains to be up in order to work. In order to help drivers, we introduce multi PM domains helpers that are able to activate/deactivate multi PM domains. First patch introduces the helpers and second patch demonstrates how a driver can use them instead of hardcoding the PM domains handling. Changes since v2: » - fix kernel test robot reported issues (missing static inline for newly introduced functions in headers and arguments swapped for devm_kzalloc). Changes since v1: (addressed Ranjani's comments) » - enhanced description for dev_multi_pm_attach return value » - renamed exit_unroll_pm label to exit_detach_pm Ideally would be to have patch 1/2 merged via power tree and then I will submit again patch 2/2 on sound tree. Daniel Baluta (2): PM / domains: Introduce multi PM domains helpers ASoC: SOF: Use multi PM domains helpers drivers/base/power/common.c | 93 +++++++++++++++++++++++++++++++++++++ include/linux/pm_domain.h | 19 ++++++++ sound/soc/sof/imx/imx8.c | 60 ++++-------------------- 3 files changed, 121 insertions(+), 51 deletions(-) -- 2.17.1

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[PATCH alsa-lib v2 0/2] Make dmix lockless operation optional
by Takashi Iwai 07 Jul '20

07 Jul '20

Hi, this is a revised patch set for the PCM dmix, to make the lockless operation optional and fix the semaphore usage in the lockless mode. Unfortunately I got a server problem in the last week and lost many posts during the weekend, so I couldn't follow up to the thread (I just read from the lore archive). In this patch set, to make my intention clearer, I swapped the patches: the first to disable the lockless operation as default and the second to fix the semaphore usage in the lockless mode. In the first patch, the magic number is changed as Jaroslav suggested, and the description has been updated. The second patch is equivalent with the previous one. thanks, Takashi === Takashi Iwai (2): pcm: dmix: make lockless operation optional pcm: dmix: Fix semaphore usage with lockless operation configure.ac | 13 +++++++++++++ src/pcm/pcm_direct.c | 16 +++++++++++++--- src/pcm/pcm_direct.h | 1 + src/pcm/pcm_dmix.c | 18 +++++++++++------- src/pcm/pcm_dmix_generic.c | 2 +- src/pcm/pcm_dmix_i386.c | 1 + src/pcm/pcm_dmix_x86_64.c | 1 + 7 files changed, 41 insertions(+), 11 deletions(-) -- 2.16.4

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[PATCH v5 0/3] ALSA: compress: Document stream states and fix gapless SM
by Vinod Koul 07 Jul '20

07 Jul '20

Srini found issue with gapless implementation which prompted to look deeper into SM for compressed stream. So documenting SM was first step, so first two patches add that. Last patch fixes the issue by keeping track on partial_drain and then moving state to 'running' in snd_compr_drain_notify() for partial_drain case on success. While at it, noticed snd_compr_drain_notify() is lockless state change, so fixed that as well. I have tested this on Dragon board RB3, compressed audio works out of the box on that platform and Srini will send driver and fcplay patches for gapless soon. Changes in v5: - Update SM diagram as suggested by Amadeusz Changes in v4: - Add review tag by Charles - fix the lock deadlock pointed by Charles Changes in v3: - Add pause->stop->free transition - Add setup->free transition - remove running->free as that goes thru stop Changes in v2: - Added tested tag by Srini - Update compress SM with Free state and compr_stop() transitions Vinod Koul (3): ALSA: compress: document the compress audio state machine ALSA: compress: document the compress gapless audio state machine ALSA: compress: fix partial_drain completion state .../sound/designs/compress-offload.rst | 83 +++++++++++++++++++ include/sound/compress_driver.h | 10 ++- sound/core/compress_offload.c | 4 + 3 files changed, 96 insertions(+), 1 deletion(-) -- 2.26.2

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[PATCH v2] ASoC: fsl_asrc: Add an option to select internal ratio mode
by Shengjiu Wang 04 Jul '20

04 Jul '20

The ASRC not only supports ideal ratio mode, but also supports internal ratio mode. For internal rato mode, the rate of clock source should be divided with no remainder by sample rate, otherwise there is sound distortion. Add function fsl_asrc_select_clk() to find proper clock source for internal ratio mode, if the clock source is available then internal ratio mode will be selected. With change, the ideal ratio mode is not the only option for user. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - update according to Nicolin's comments sound/soc/fsl/fsl_asrc.c | 54 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 52 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 95f6a9617b0b..4105ef2c4f99 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -582,11 +582,55 @@ static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, &fsl_asrc_rate_constraints); } +/** + * Select proper clock source for internal ratio mode + */ +static int fsl_asrc_select_clk(struct fsl_asrc_priv *asrc_priv, + struct fsl_asrc_pair *pair, + int in_rate, + int out_rate) +{ + struct fsl_asrc_pair_priv *pair_priv = pair->private; + struct asrc_config *config = pair_priv->config; + int rate[2], select_clk[2]; /* Array size 2 means IN and OUT */ + int clk_rate, clk_index; + int i = 0, j = 0; + + rate[0] = in_rate; + rate[1] = out_rate; + + /* Select proper clock source for internal ratio mode */ + for (j = 0; j < 2; j++) { + for (i = 0; i < ASRC_CLK_MAP_LEN; i++) { + clk_index = asrc_priv->clk_map[j][i]; + clk_rate = clk_get_rate(asrc_priv->asrck_clk[clk_index]); + /* Only match a perfect clock source with no remainder */ + if (clk_rate != 0 && (clk_rate / rate[j]) <= 1024 && + (clk_rate % rate[j]) == 0) + break; + } + + select_clk[j] = i; + } + + /* Switch to ideal ratio mode if there is no proper clock source */ + if (select_clk[IN] == ASRC_CLK_MAP_LEN || select_clk[OUT] == ASRC_CLK_MAP_LEN) { + select_clk[IN] = INCLK_NONE; + select_clk[OUT] = OUTCLK_ASRCK1_CLK; + } + + config->inclk = select_clk[IN]; + config->outclk = select_clk[OUT]; + + return 0; +} + static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct fsl_asrc *asrc = snd_soc_dai_get_drvdata(dai); + struct fsl_asrc_priv *asrc_priv = asrc->private; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; struct fsl_asrc_pair_priv *pair_priv = pair->private; @@ -605,8 +649,6 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.pair = pair->index; config.channel_num = channels; - config.inclk = INCLK_NONE; - config.outclk = OUTCLK_ASRCK1_CLK; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config.input_format = params_format(params); @@ -620,6 +662,14 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } + ret = fsl_asrc_select_clk(asrc_priv, pair, + config.input_sample_rate, + config.output_sample_rate); + if (ret) { + dev_err(dai->dev, "fail to select clock\n"); + return ret; + } + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); -- 2.21.0

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[HDAudio][ALC295] speakers not working on Acer Travelmate P614
by Stanislav Kazmin 03 Jul '20

03 Jul '20

Hello, I have the following problem on my Acer TravelMate P614-51T-G2: I have all the audio sinks correctly identified (like HDMI, DMic, Headphones) but the internal speakers do not produce any sound. What I have tried so far: - hard/soft shutdown on Windows and reboot - disable/re-enable speakers and microphone in BIOS - uninstall pulseaudio and test alsa alone (so it is defeitely **not** a pulseaudio issue) - switch from sof-hda-dsp to snd-hda-intel driver (without Dmic support) - retask pins 0x14, 0x16, 0x1b to "Internal Speakers" through hdajackretask (only basic, without advanced features) - removed/reconnected the headphones `alsamixer` shows all needed sinks and nothing is muted. I already discussed the issue at sof github https://github.com/thesofproject/ sof/issues/3058 but since the same issue occurs on `snd-hda-intel` legacy river, I was advised to communicate with alsa-devel team. The alsa-info.sh result are linked at http://alsa-project.org/db/? f=252f92c7a1df3c755d16ee69353b26d2535a4d81 I have tested the kernel 5.8-rc1 but it does not make any difference. Let me know if I can do anything else to test the issue.

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