August 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Trying to create a developer account, but need a token
by Gene Cristan 15 Aug '18

15 Aug '18

To help protect against automated account creation, please answer the question that appears below (more info [1]): Token to prevent spam? Please, ask on mailing lists... Links: ------ [1] http://www.alsa-project.org/main/index.php/Special:Captcha/help

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[alsa-devel] [PATCH - hwmixvolume 4/9] hwmixvolume: use a with context to open files
by Emmanuel Gil Peyrot 15 Aug '18

15 Aug '18

Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 267228c..e82f32d 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -138,26 +138,20 @@ class Stream: subdevice = info.index filename = "/proc/asound/card%d/pcm%dp/sub%d/status" % (card, device, subdevice) try: - f = open(filename, "r") + with open(filename, "r") as f: + for line in f.readlines(): + if line[:9] == "owner_pid": + return int(line.split(':')[1].strip()) except IOError: return None - try: - for line in f.readlines(): - if line[:9] == "owner_pid": - return int(line.split(':')[1].strip()) - finally: - f.close() return None def get_pid_cmdline(self, pid): try: - f = open("/proc/%d/cmdline" % pid, "r") + with open("/proc/%d/cmdline" % pid, "r") as f: + cmdline = f.read() except IOError: return None - try: - cmdline = f.read() - finally: - f.close() return cmdline.replace('\x00', ' ').strip() class MixerWindow(Gtk.Window): -- 2.18.0

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[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 15 Aug '18

15 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 164c0f6454186d1b0e22c27dfe3d3d59fc80f979 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

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[alsa-devel] [PATCH] ASoC: dapm: Fix NULL pointer deference on CODEC to CODEC DAIs
by Charles Keepax 15 Aug '18

15 Aug '18

Commit a655de808cbde ("ASoC: core: Allow topology to override machine driver FE DAI link config.") caused soc_dai_hw_params to be come dependent on the substream private_data being set with a pointer to the snd_soc_pcm_runtime. Currently, CODEC to CODEC links don't set this, which causes a NULL pointer dereference: [<4069de54>] (soc_dai_hw_params) from [<40694b68>] (snd_soc_dai_link_event+0x1a0/0x380) Since the ASoC core in general assumes that the substream private_data will be set to a pointer to the snd_soc_pcm_runtime, update the CODEC to CODEC links to respect this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 4 ++++ 3 files changed, 7 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index af9ef16cc34d..fdaaafdc7a00 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -407,6 +407,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, const struct snd_soc_pcm_stream *params, unsigned int num_params, struct snd_soc_dapm_widget *source, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9cfe10d8040c..473eefe8658e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1447,7 +1447,7 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, sink = codec_dai->playback_widget; source = cpu_dai->capture_widget; if (sink && source) { - ret = snd_soc_dapm_new_pcm(card, dai_link->params, + ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params, dai_link->num_params, source, sink); if (ret != 0) { @@ -1460,7 +1460,7 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, sink = cpu_dai->playback_widget; source = codec_dai->capture_widget; if (sink && source) { - ret = snd_soc_dapm_new_pcm(card, dai_link->params, + ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params, dai_link->num_params, source, sink); if (ret != 0) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e96793050c9..461d951917c0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3652,6 +3652,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, { struct snd_soc_dapm_path *source_p, *sink_p; struct snd_soc_dai *source, *sink; + struct snd_soc_pcm_runtime *rtd = w->priv; const struct snd_soc_pcm_stream *config = w->params + w->params_select; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; @@ -3711,6 +3712,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, goto out; } substream.runtime = runtime; + substream.private_data = rtd; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3895,6 +3897,7 @@ snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card, } int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, const struct snd_soc_pcm_stream *params, unsigned int num_params, struct snd_soc_dapm_widget *source, @@ -3963,6 +3966,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, w->params = params; w->num_params = num_params; + w->priv = rtd; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) -- 2.11.0

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[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 15 Aug '18

15 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 164c0f6454186d1b0e22c27dfe3d3d59fc80f979 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

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Re: [alsa-devel] ASoC: pxa: possible regressions
by Daniel Mack 15 Aug '18

15 Aug '18

Hi Petr, On Wednesday, August 08, 2018 04:24 AM, Petr Cvek wrote: > I was trying to update my system (PXA Magician) to 4.18-next and i've > ran into few problems with sound. Ah, so you have a Magician platform, that's very good! > My setup has only a few changes from > vanilla (change to using devm_snd_soc_register_card+devm_gpio_request). Care to share these changes, or ideally rebase them on top of 4.18-next for mainline inclusion? Also, I'm curious what mainline version you used to be based on that worked for you. > I've started digging around: Thanks for that. Are you also willing to send patches to fix things once they work for you? > 1) unimportant goto > > Commit 7afd1b0b2ef9a812095 ("ASoC: pxa: move some functions to > pxa2xx-lib") is this construct necessary? > > if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { > ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, > SNDRV_PCM_STREAM_CAPTURE); > if (ret) > goto out; > } > out: > return ret; > > After calling pxa2xx_pcm_preallocate_dma_buffer() the code will jump to > out anyway. True. That's a cosmetic cleanup which can go in for 4.19. Would you want to send a patch after the merge window? > 2) wrong units (sample rate vs osc rate) > > Commit 05739375f1c0a1048 ("ASoC: pxa-ssp: remove .set_pll() and > .set_clkdiv() callbacks"). The function pxa_ssp_hw_params() get's called > with sample rate so: > > /* The values in the table are for 16 bits */ > if (width == 32) > acds--; > > ret = pxa_ssp_set_pll(priv, bclk); > if (ret < 0) > return ret; > > ssacd = pxa_ssp_read_reg(ssp, SSACD); > ssacd &= ~(SSACD_ACDS(7) | SSACD_SCDB_1X); > > pxa_ssp_set_pll() will try to set something like 8000, 44100, ... and > not the speed of an oscilator. The line should be probably: > > ret = pxa_ssp_set_pll(priv, m->pll); > > and there is a previous call to pxa_ssp_set_pll(): Ah, yes. Does changing that fix it for you? The changes in that machine driver where done blindly because I don't have Magician hardware, and I really don't like doing things that way. > 3) setup fails > > Recent changes in ASoC PXA DMA are causing a fail of a condition in > pxa_ssp_hw_params(): > > if ((sscr0 & SSCR0_MOD) && !ttsa) { What are the values of sscr0 and ttsa in your case? Can you cross-check that with a kernel that doesn't have my recent changes? > dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); > return -EINVAL; > } > > It seems the SSCR0_MOD flag is set in the fall through switch statement > in pxa_ssp_configure_dai_fmt(): > > case SND_SOC_DAIFMT_DSP_A: > sspsp |= SSPSP_FSRT; > /* fall through */ > case SND_SOC_DAIFMT_DSP_B: > sscr0 |= SSCR0_MOD | SSCR0_PSP; > sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; > break; > > the switch statement is an old code and it worked in the past versions. Yes, it was merely moved to some location that is called earlier. This was fixing a regression caused by a8bd0ee558714 ("ASoC: raumfeld: Use static DAI format setup"), so supposedly, all PXA sound was broken since v4.0. > 4) names are not unique > > The initialization of magician_dai structure wants to have (at least in > the past versions) for the .name item: > > static struct snd_soc_dai_link magician_dai[] = { > { > .name = "uda1380", > .stream_name = "UDA1380 Playback", > .cpu_dai_name = "pxa-ssp-dai.0", > ... > { > .name = "uda1380", > .stream_name = "UDA1380 Capture", > .cpu_dai_name = "pxa2xx-i2s", > > my personal branch has them named as "uda1380_playback" and > "uda1380_capture". Care to send a patch for that as well, please? > 5) Unbalanced references > > When trying to play a sound (with quick-and-dirty workarounds for > problems above): > > # cat /dev/urandom | aplay - > > the system will crash when terminating aplay (stopping playback). The > backtrace: So you did hear a sound? [...] > I've only managed to discover there are two calls of > pxa2xx_soc_pcm_new() (where DMA buffer is allocated) for both dai links: > > magician-audio magician-audio: uda1380-hifi-playback <-> pxa-ssp-dai.0 > mapping ok > ... > magician-audio magician-audio: uda1380-hifi-capture <-> pxa2xx-i2s > mapping ok > > but I don't know if this information is relevant. That seems fine. The question is why there are are unbalanced calls to snd_dmaengine_pcm_close_release_chan(). Could you dig a bit further maybe and add some printk()? Thanks for helping fix these regressions! Daniel

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[alsa-devel] [PATCH v2 00/10] ASoC: Add support to WCD9335 Audio Codec
by Srinivas Kandagatla 15 Aug '18

15 Aug '18

Thankyou for reviewing v1 patchset, here is v2 addressing comments from v1. Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support witout much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback and capture paths. This patchset also depends on the SLIMbus Stream apis which is already merged via char-misc tree for 4.19. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is my test branch incase someone want to try this out: https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=wcd9335 Thanks, Srini Changes since v1 (https://lkml.org/lkml/2018/7/23/889) - Fixed dt bindings as suggested by Mark B. - cleanup CLASS-H code, remove un-used states. - Various cleanup suggested by Vinod. Srinivas Kandagatla (10): ASoC: dt-bindings: add dt bindings for wcd9335 audio codec mfd: wcd9335: add support to wcd9335 core mfd: wcd9335: add wcd irq support ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: apq8096: add support to Analog audio via WCD9335 slim .../devicetree/bindings/sound/qcom,wcd9335.txt | 123 + drivers/mfd/Kconfig | 18 + drivers/mfd/Makefile | 4 + drivers/mfd/wcd9335-core.c | 277 ++ drivers/mfd/wcd9335-irq.c | 172 + include/dt-bindings/mfd/wcd9335.h | 43 + include/linux/mfd/wcd9335/registers.h | 580 +++ include/linux/mfd/wcd9335/wcd9335.h | 45 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh.c | 605 +++ sound/soc/codecs/wcd-clsh.h | 49 + sound/soc/codecs/wcd9335.c | 4963 ++++++++++++++++++++ sound/soc/qcom/apq8096.c | 68 + 14 files changed, 6954 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd9335.txt create mode 100644 drivers/mfd/wcd9335-core.c create mode 100644 drivers/mfd/wcd9335-irq.c create mode 100644 include/dt-bindings/mfd/wcd9335.h create mode 100644 include/linux/mfd/wcd9335/registers.h create mode 100644 include/linux/mfd/wcd9335/wcd9335.h create mode 100644 sound/soc/codecs/wcd-clsh.c create mode 100644 sound/soc/codecs/wcd-clsh.h create mode 100644 sound/soc/codecs/wcd9335.c -- 2.16.2

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[alsa-devel] Miscellaneous ioplug questions
by Rob Duncan 15 Aug '18

15 Aug '18

I have some questions about some ioplug implementation details: - Is the value returned from the pointer callback supposed to be a monotonically increasing value, or should it wrap at the buffer size? The documentation says "get the current DMA position" which isn't clear to me. - Must the pointer callback return values that are an integral multiple of the period size, or can it report partial periods? - Should the poll FD be made writeable as soon as data is available, or only when at least avail_min is available? Thanks, Rob.

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[alsa-devel] WARNING in __snd_rawmidi_transmit_ack (2)
by syzbot 14 Aug '18

14 Aug '18

Hello, syzbot found the following crash on: HEAD commit: 4e8b38549b50 Add linux-next specific files for 20180814 git tree: linux-next console output: https://syzkaller.appspot.com/x/log.txt?x=11f5e78c400000 kernel config: https://syzkaller.appspot.com/x/.config?x=18edf0289d1b5ab dashboard link: https://syzkaller.appspot.com/bug?extid=ec5f605c91812d200367 compiler: gcc (GCC) 8.0.1 20180413 (experimental) syzkaller repro:https://syzkaller.appspot.com/x/repro.syz?x=12f8c6d2400000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=16976b72400000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+ec5f605c91812d200367(a)syzkaller.appspotmail.com WARNING: CPU: 1 PID: 1691 at sound/core/rawmidi.c:1170 __snd_rawmidi_transmit_ack+0x263/0x2f0 sound/core/rawmidi.c:1170 Kernel panic - not syncing: panic_on_warn set ... CPU: 1 PID: 1691 Comm: kworker/1:1H Not tainted 4.18.0-next-20180814+ #38 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: events_highpri snd_vmidi_output_work Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1c9/0x2b4 lib/dump_stack.c:113 panic+0x238/0x4e7 kernel/panic.c:184 __warn.cold.8+0x163/0x1ba kernel/panic.c:536 report_bug+0x252/0x2d0 lib/bug.c:186 fixup_bug arch/x86/kernel/traps.c:178 [inline] do_error_trap+0x1fc/0x4d0 arch/x86/kernel/traps.c:296 do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:316 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:996 RIP: 0010:__snd_rawmidi_transmit_ack+0x263/0x2f0 sound/core/rawmidi.c:1170 Code: ba 01 00 00 00 be 03 00 00 00 e8 18 3a 66 fc e8 e3 44 86 fc 89 d8 48 83 c4 18 5b 41 5c 41 5d 41 5e 41 5f 5d c3 e8 cd 44 86 fc <0f> 0b e9 72 fe ff ff e8 21 5a c5 fc e9 1b ff ff ff bb ea ff ff ff RSP: 0018:ffff8801d2a6f5a8 EFLAGS: 00010093 RAX: ffff8801d2a52300 RBX: 0000000000000001 RCX: ffffffff84f64792 RDX: 0000000000000000 RSI: ffffffff84f64923 RDI: 0000000000000006 RBP: ffff8801d2a6f5e8 R08: ffff8801d2a52300 R09: ffffed003607acc1 R10: ffffed003607acc1 R11: ffff8801b03d660b R12: ffff8801ce2479c0 R13: 0000000000001001 R14: 0000000000000001 R15: ffff8801b03d65c0 snd_rawmidi_transmit_ack+0x51/0x80 sound/core/rawmidi.c:1201 snd_vmidi_output_work+0x3f4/0x4e0 sound/core/seq/seq_virmidi.c:153 process_one_work+0xc73/0x1aa0 kernel/workqueue.c:2153 worker_thread+0x189/0x13c0 kernel/workqueue.c:2296 kthread+0x35a/0x420 kernel/kthread.c:246 ret_from_fork+0x3a/0x50 arch/x86/entry/entry_64.S:415 Shutting down cpus with NMI Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#bug-status-tracking for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

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[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 14 Aug '18

14 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 164c0f6454186d1b0e22c27dfe3d3d59fc80f979 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

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