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[alsa-devel] [PATCH V2] ASoC: soc-pcm: BE dai needs prepare when pause release after resume
by libin.yang＠intel.com 13 May '19

13 May '19

From: Libin Yang <libin.yang(a)intel.com> If playback/capture is paused and system enters S3, after system returns from suspend, BE dai needs to call prepare() callback when playback/capture is released from pause if RESUME_INFO flag is not set. Currently, the dpcm_be_dai_prepare() function will block calling prepare() if the pcm is in SND_SOC_DPCM_STATE_PAUSED state. This will cause the following test case fail if the pcm uses BE: playback -> pause -> S3 suspend -> S3 resume -> pause release The playback may exit abnormally when pause is released because the BE dai prepare() is not called. This patch allows dpcm_be_dai_prepare() to call dai prepare() callback in SND_SOC_DPCM_STATE_PAUSED state. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/soc-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d2aa560..0888995 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2471,7 +2471,8 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; dev_dbg(be->dev, "ASoC: prepare BE %s\n", -- 2.7.4

5 13

[alsa-devel] [PATCH v2] ASoC: max98357a: request GPIO when device get probed
by Tzung-Bi Shih 13 May '19

13 May '19

devm_gpiod_get_optional() returns EBUSY after component rebound. Request GPIO in max98357a_platform_probe() to support component rebinding. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- v1: ASoC: max98357a: manage GPIO for component rebinding => don't move from devm to non-devm (https://patchwork.kernel.org/patch/10922979/) v2: ASoC: max98357a: release GPIO when component removing => only put things that really need the card in component's probe() => don't use devm_gpiod_put() if possible (https://patchwork.kernel.org/patch/10932279/) v3: ASoC: max98357a: request GPIO when device get probed => remove empty function max98357a_component_probe() and max98357a_platform_remove() (https://patchwork.kernel.org/patch/10936239/) sound/soc/codecs/max98357a.c | 29 +++++++++-------------------- 1 file changed, 9 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index d037a3e4d323..80080a6415b3 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -59,21 +59,7 @@ static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { {"Speaker", NULL, "HiFi Playback"}, }; -static int max98357a_component_probe(struct snd_soc_component *component) -{ - struct gpio_desc *sdmode; - - sdmode = devm_gpiod_get_optional(component->dev, "sdmode", GPIOD_OUT_LOW); - if (IS_ERR(sdmode)) - return PTR_ERR(sdmode); - - snd_soc_component_set_drvdata(component, sdmode); - - return 0; -} - static const struct snd_soc_component_driver max98357a_component_driver = { - .probe = max98357a_component_probe, .dapm_widgets = max98357a_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98357a_dapm_widgets), .dapm_routes = max98357a_dapm_routes, @@ -112,16 +98,20 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { static int max98357a_platform_probe(struct platform_device *pdev) { + struct gpio_desc *sdmode; + + sdmode = devm_gpiod_get_optional(&pdev->dev, + "sdmode", GPIOD_OUT_LOW); + if (IS_ERR(sdmode)) + return PTR_ERR(sdmode); + + dev_set_drvdata(&pdev->dev, sdmode); + return devm_snd_soc_register_component(&pdev->dev, &max98357a_component_driver, &max98357a_dai_driver, 1); } -static int max98357a_platform_remove(struct platform_device *pdev) -{ - return 0; -} - #ifdef CONFIG_OF static const struct of_device_id max98357a_device_id[] = { { .compatible = "maxim,max98357a" }, @@ -145,7 +135,6 @@ static struct platform_driver max98357a_platform_driver = { .acpi_match_table = ACPI_PTR(max98357a_acpi_match), }, .probe = max98357a_platform_probe, - .remove = max98357a_platform_remove, }; module_platform_driver(max98357a_platform_driver); -- 2.21.0.1020.gf2820cf01a-goog

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[alsa-devel] [PATCH 0/6] ASoC: SOF: small update to align with Mark's tree
by Pierre-Louis Bossart 13 May '19

13 May '19

This small patchset removes unneeded include files, fixes a race condition seen in QEMU tests, removes unnecessary PM calls, fixes typos and updates ICL configurations. Jaska Uimonen (1): ASoC: SOF: uapi: remove unused sof header files Kai Vehmanen (1): ASoC: SOF: fix race in FW boot timeout handling Ranjani Sridharan (2): ASoC: SOF: pcm: remove runtime PM calls during pcm open/close ASoC: SOF: fix error in verbose ipc command parsing Zhu Yingjiang (2): ASoC: SOF: Intel: ICL: add Icelake SSP count ASoC: SOF: Intel: ICL add Icelake chip info struct include/uapi/sound/sof/eq.h | 172 --------------------------- include/uapi/sound/sof/manifest.h | 188 ------------------------------ include/uapi/sound/sof/tone.h | 21 ---- include/uapi/sound/sof/trace.h | 66 ----------- sound/soc/sof/intel/cnl.c | 19 +++ sound/soc/sof/intel/hda.h | 2 + sound/soc/sof/ipc.c | 2 +- sound/soc/sof/loader.c | 2 + sound/soc/sof/pcm.c | 29 +---- sound/soc/sof/sof-pci-dev.c | 2 +- 10 files changed, 27 insertions(+), 476 deletions(-) delete mode 100644 include/uapi/sound/sof/eq.h delete mode 100644 include/uapi/sound/sof/manifest.h delete mode 100644 include/uapi/sound/sof/tone.h delete mode 100644 include/uapi/sound/sof/trace.h -- 2.17.1

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[alsa-devel] [PATCH] ASoC: hdmi-codec: re-introduce mutex locking
by Jerome Brunet 13 May '19

13 May '19

Replace the bit atomic operations by a mutex to ensure only one dai at a time is active on the hdmi codec. This is a follow up on change: 3fcf94ef4d41 ("ASoC: hdmi-codec: remove reference to the current substream") Suggested-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/codecs/hdmi-codec.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 90a892766625..6a0cc8d7e141 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -281,7 +281,7 @@ struct hdmi_codec_priv { uint8_t eld[MAX_ELD_BYTES]; struct snd_pcm_chmap *chmap_info; unsigned int chmap_idx; - unsigned long busy; + struct mutex lock; }; static const struct snd_soc_dapm_widget hdmi_widgets[] = { @@ -395,8 +395,8 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); int ret = 0; - ret = test_and_set_bit(0, &hcp->busy); - if (ret) { + ret = mutex_trylock(&hcp->lock); + if (!ret) { dev_err(dai->dev, "Only one simultaneous stream supported!\n"); return -EINVAL; } @@ -424,7 +424,7 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, err: /* Release the exclusive lock on error */ - clear_bit(0, &hcp->busy); + mutex_unlock(&hcp->lock); return ret; } @@ -436,7 +436,7 @@ static void hdmi_codec_shutdown(struct snd_pcm_substream *substream, hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; hcp->hcd.ops->audio_shutdown(dai->dev->parent, hcp->hcd.data); - clear_bit(0, &hcp->busy); + mutex_unlock(&hcp->lock); } static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, @@ -773,6 +773,8 @@ static int hdmi_codec_probe(struct platform_device *pdev) return -ENOMEM; hcp->hcd = *hcd; + mutex_init(&hcp->lock); + daidrv = devm_kcalloc(dev, dai_count, sizeof(*daidrv), GFP_KERNEL); if (!daidrv) return -ENOMEM; -- 2.20.1

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Re: [alsa-devel] Boot procudure on HDA driver
by Takashi Iwai 13 May '19

13 May '19

On Mon, 13 May 2019 13:52:32 +0200, Kailang wrote: > > > > >Well, it's not clear to me whether you'd like to reduce the depop > >procedure above or to add it somewhere else. Could you clarify? > > This depop procedure in alc225_shutup. it doesn't need to run in system boot up. > > If system boot up, it will run alc225_init() already. alc225_init() include power up depop procedure. > So, if system boot up, it only need to run depop procedure in alc225_init(). This is the runtime suspend and resume. That is: > The current boot up procedure was below. > 1. alc225_init() ... which is called at the driver probe time, from snd_hda_codec_build_controls(). Then the codec driver initialization finishes at this point, hence it goes to runtime suspend. > 2. alc225_shutup() ==> this include power off and suspend depop procedure. some codec run this field will have pop noise at boot up. ... which is a part of the codec runtime suspend. Then, the codec gets woken up for some access, and it performs the runtime resume. > 3. alc225_init() ... which is a part of the codec runtime resume. That is, the step 2 and 3 are already out of the initialization phase, and they are normal runtime PM procedure. If the runtime depop code is superfluous for the runtime PM, you can skip it by checking the power_state, for example. But, again, it has nothing to do with the boot or not. It's just the normal runtime suspend and resume. thanks, Takashi > > > ________________________________________ > 從: Takashi Iwai [tiwai(a)suse.de] > 寄件日期: 2019年5月13日下午 07:30 > 至: Kailang > 副本: (alsa-devel(a)alsa-project.org) > 主旨: Re: Boot procudure on HDA driver > > On Mon, 13 May 2019 12:36:36 +0200, > Kailang wrote: > > > > > > Maybe I confuse and I also confuse you. Sorry!! > > > > When system suspend, it will run alc_suspend(). alc_suspend() was include alc_shutup(). > > Right. > > > When system shutdown(power off), will it run alc_shutup()? > > Yes, it's called via alc_reboot_notify(). > If spec->reboot_notify is defined, this supersedes the call, so you > may define the callback if you'd need to avoid alc_shutup() call from > there. > > > > in our patch_realtek.c. > > spec->shutup = alc225_shutup; > > Some codec didn't run the depop procudure during boot up. > > > > Maybe disable EAPD and shutup PIN will cause pop noise. > > But this procudure need to run in SUSPEND and HIBERNATE and Power off. > > It also need to run it at runtime suspend. > > > > static void alc225_shutup(struct hda_codec *codec) > > { > > struct alc_spec *spec = codec->spec; > > hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; > > bool hp1_pin_sense, hp2_pin_sense; > > > > *********** depop procudure ****************************************** > > if (!hp_pin) > > hp_pin = 0x21; > > > > /* 3k pull low control for Headset jack. */ > > alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); > > > > hp1_pin_sense = snd_hda_jack_detect(codec, hp_pin); > > hp2_pin_sense = snd_hda_jack_detect(codec, 0x16); > > > > if (hp1_pin_sense || hp2_pin_sense) > > msleep(2); > > > > if (hp1_pin_sense) > > snd_hda_codec_write(codec, hp_pin, 0, > > AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); > > if (hp2_pin_sense) > > snd_hda_codec_write(codec, 0x16, 0, > > AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); > > > > if (hp1_pin_sense || hp2_pin_sense) > > msleep(85); > > > > if (hp1_pin_sense) > > snd_hda_codec_write(codec, hp_pin, 0, > > AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); > > if (hp2_pin_sense) > > snd_hda_codec_write(codec, 0x16, 0, > > AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); > > > > if (hp1_pin_sense || hp2_pin_sense) > > msleep(100); > > > > alc_auto_setup_eapd(codec, false); > > snd_hda_shutup_pins(codec); > > ************** depop procudure ************************************* > > } > > Well, it's not clear to me whether you'd like to reduce the depop > procedure above or to add it somewhere else. Could you clarify? > > > thanks, > > Takashi > > > > > > > > > > > > > > ________________________________________ > > 從: Takashi Iwai [tiwai(a)suse.de] > > 寄件日期: 2019年5月13日下午 05:42 > > 至: Kailang > > 副本: (alsa-devel(a)alsa-project.org) > > 主旨: Re: Boot procudure on HDA driver > > > > On Mon, 13 May 2019 11:30:56 +0200, > > Kailang wrote: > > > > > > > > > > > > > -----Original Message----- > > > > From: Takashi Iwai <tiwai(a)suse.de> > > > > Sent: Monday, May 13, 2019 5:11 PM > > > > To: Kailang <kailang(a)realtek.com> > > > > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > > > > Subject: Re: Boot procudure on HDA driver > > > > > > > > On Mon, 13 May 2019 11:00:18 +0200, > > > > Kailang wrote: > > > > > > > > > > Hi Takashi, > > > > > > > > > > When System Boot up. > > > > > The Hda Driver running step was as below. > > > > > > > > > > alc_init(); > > > > > alc_shutup(); > > > > > alc_init(); > > > > > > > > > > The depop procedure was put in spec->init_hook and spec->shutup. > > > > > > > > > > But I find more codec which run spec->shutup at boot up. It will occur pop > > > > noise. > > > > > If it doesn't run spec->shutup, it will not occur pop noise or reduce pop noise. > > > > > > > > > > How could the spec->shutup not run at boot up? > > > > > > > > > > I ever test to put the check in spec->shutup() for PM_EVENT_SUSPEND and > > > > PM_EVENT_HIBERNATE and shutdown. > > > > > But if power_save=1, it was have issue for this. > > > > > Codec was idle already in power_save=1 state. If system go suspend, it will > > > > not run spec->shutup() again. > > > > > > > > I guess it's a runtime PM, hence it's neither PM_EVENT_SUSPEND nor > > > > PM_EVENT_HIBERNATE. > > > > > > > runtime PM and suspend and hibernate and shutdown need to run > > > spec->shutup(). > > > > Is the call really mandatory? > > > > > It's no problem. > > > But spec->shutup() doesn't need to run in boot up. > > > > The call of spec->shutup() at boot up *is* the runtime PM. > > Or any other call path I overlooked? > > > > > And it will set power_save=1 on all dell machine. > > > > > > If (codec->auto_runtime_pm || codec->bus->shutdown || > > > codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND || > > > codec->core.dev.power.power_state.event == PM_EVENT_HIBERNATE) > > > > > > So, I need to put upper check code in spec->shutup(). Right? > > > Thanks. > > > > No, the auto_runtime_pm is for a completely different purpose. > > > > At the boot up, the runtime PM can be kicked in at any time. So if > > you disable the shutup during the runtime PM, it means you'd need to > > call the shutup at runtime PM completely. > > > > > > Takashi > > > > > > > > > Actually, if the shutup procedure makes the problem on a certain platform, just > > > > skip it. It's an optional behavior and would be fine without it (of course only if > > > > it's confirmed to work :) > > > > > > > > > > > > Takashi > > > > > > > > ------Please consider the environment before printing this e-mail. > > >

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Re: [alsa-devel] Boot procudure on HDA driver
by Takashi Iwai 13 May '19

13 May '19

On Mon, 13 May 2019 12:36:36 +0200, Kailang wrote: > > > Maybe I confuse and I also confuse you. Sorry!! > > When system suspend, it will run alc_suspend(). alc_suspend() was include alc_shutup(). Right. > When system shutdown(power off), will it run alc_shutup()? Yes, it's called via alc_reboot_notify(). If spec->reboot_notify is defined, this supersedes the call, so you may define the callback if you'd need to avoid alc_shutup() call from there. > in our patch_realtek.c. > spec->shutup = alc225_shutup; > Some codec didn't run the depop procudure during boot up. > > Maybe disable EAPD and shutup PIN will cause pop noise. > But this procudure need to run in SUSPEND and HIBERNATE and Power off. > It also need to run it at runtime suspend. > > static void alc225_shutup(struct hda_codec *codec) > { > struct alc_spec *spec = codec->spec; > hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; > bool hp1_pin_sense, hp2_pin_sense; > > *********** depop procudure ****************************************** > if (!hp_pin) > hp_pin = 0x21; > > /* 3k pull low control for Headset jack. */ > alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); > > hp1_pin_sense = snd_hda_jack_detect(codec, hp_pin); > hp2_pin_sense = snd_hda_jack_detect(codec, 0x16); > > if (hp1_pin_sense || hp2_pin_sense) > msleep(2); > > if (hp1_pin_sense) > snd_hda_codec_write(codec, hp_pin, 0, > AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); > if (hp2_pin_sense) > snd_hda_codec_write(codec, 0x16, 0, > AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); > > if (hp1_pin_sense || hp2_pin_sense) > msleep(85); > > if (hp1_pin_sense) > snd_hda_codec_write(codec, hp_pin, 0, > AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); > if (hp2_pin_sense) > snd_hda_codec_write(codec, 0x16, 0, > AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); > > if (hp1_pin_sense || hp2_pin_sense) > msleep(100); > > alc_auto_setup_eapd(codec, false); > snd_hda_shutup_pins(codec); > ************** depop procudure ************************************* > } Well, it's not clear to me whether you'd like to reduce the depop procedure above or to add it somewhere else. Could you clarify? thanks, Takashi > > > > > > ________________________________________ > 從: Takashi Iwai [tiwai(a)suse.de] > 寄件日期: 2019年5月13日下午 05:42 > 至: Kailang > 副本: (alsa-devel(a)alsa-project.org) > 主旨: Re: Boot procudure on HDA driver > > On Mon, 13 May 2019 11:30:56 +0200, > Kailang wrote: > > > > > > > > > -----Original Message----- > > > From: Takashi Iwai <tiwai(a)suse.de> > > > Sent: Monday, May 13, 2019 5:11 PM > > > To: Kailang <kailang(a)realtek.com> > > > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > > > Subject: Re: Boot procudure on HDA driver > > > > > > On Mon, 13 May 2019 11:00:18 +0200, > > > Kailang wrote: > > > > > > > > Hi Takashi, > > > > > > > > When System Boot up. > > > > The Hda Driver running step was as below. > > > > > > > > alc_init(); > > > > alc_shutup(); > > > > alc_init(); > > > > > > > > The depop procedure was put in spec->init_hook and spec->shutup. > > > > > > > > But I find more codec which run spec->shutup at boot up. It will occur pop > > > noise. > > > > If it doesn't run spec->shutup, it will not occur pop noise or reduce pop noise. > > > > > > > > How could the spec->shutup not run at boot up? > > > > > > > > I ever test to put the check in spec->shutup() for PM_EVENT_SUSPEND and > > > PM_EVENT_HIBERNATE and shutdown. > > > > But if power_save=1, it was have issue for this. > > > > Codec was idle already in power_save=1 state. If system go suspend, it will > > > not run spec->shutup() again. > > > > > > I guess it's a runtime PM, hence it's neither PM_EVENT_SUSPEND nor > > > PM_EVENT_HIBERNATE. > > > > > runtime PM and suspend and hibernate and shutdown need to run > > spec->shutup(). > > Is the call really mandatory? > > > It's no problem. > > But spec->shutup() doesn't need to run in boot up. > > The call of spec->shutup() at boot up *is* the runtime PM. > Or any other call path I overlooked? > > > And it will set power_save=1 on all dell machine. > > > > If (codec->auto_runtime_pm || codec->bus->shutdown || > > codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND || > > codec->core.dev.power.power_state.event == PM_EVENT_HIBERNATE) > > > > So, I need to put upper check code in spec->shutup(). Right? > > Thanks. > > No, the auto_runtime_pm is for a completely different purpose. > > At the boot up, the runtime PM can be kicked in at any time. So if > you disable the shutup during the runtime PM, it means you'd need to > call the shutup at runtime PM completely. > > > Takashi > > > > > > Actually, if the shutup procedure makes the problem on a certain platform, just > > > skip it. It's an optional behavior and would be fine without it (of course only if > > > it's confirmed to work :) > > > > > > > > > Takashi > > > > > > ------Please consider the environment before printing this e-mail. > >

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[alsa-devel] Boot procudure on HDA driver
by Kailang 13 May '19

13 May '19

Hi Takashi, When System Boot up. The Hda Driver running step was as below. alc_init(); alc_shutup(); alc_init(); The depop procedure was put in spec->init_hook and spec->shutup. But I find more codec which run spec->shutup at boot up. It will occur pop noise. If it doesn't run spec->shutup, it will not occur pop noise or reduce pop noise. How could the spec->shutup not run at boot up? I ever test to put the check in spec->shutup() for PM_EVENT_SUSPEND and PM_EVENT_HIBERNATE and shutdown. But if power_save=1, it was have issue for this. Codec was idle already in power_save=1 state. If system go suspend, it will not run spec->shutup() again. BR, Kailang

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[alsa-devel] alsa-lib-1.1.6.tar.bz2: some files are missing in the tarball
by Lars Wendler 13 May '19

13 May '19

Hello list, please have a look at this Gentoo bug report [1] where a user reported that he cannot compile alsa-lib-1.1.6 with --enable-alisp configure option. The error message is: make[3]: *** No rule to make target 'SI7018/sndoc-mixer.alisp', needed by 'all-am'. Stop. This happens because the directory src/conf/cards/SI7018 is missing as well as the following files: src/conf/cards/aliases.alisp src/conf/smixer.conf src/conf/sndo-mixer.alisp First I thought those files were removed intentionally but checking the alsa-lib git tree I can see all files still being available. So at Gentoo we fixed this by re-adding the files through a patch [2]. Any chance this was an accidental oversight? Kind regards Lars [1] https://bugs.gentoo.org/652422 [2] https://gitweb.gentoo.org/repo/gentoo.git/tree/media-libs/alsa-lib/files/al… P.S.: Please CC me on replies as I am not subscribed to the list. -- Lars Wendler Gentoo package maintainer GPG: 21CC CF02 4586 0A07 ED93 9F68 498F E765 960E 9B39

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Re: [alsa-devel] [PATCH 03/16] lib, treewide: add new match_string() helper/macro
by Andy Shevchenko 13 May '19

13 May '19

On Wed, May 08, 2019 at 02:28:29PM +0300, Alexandru Ardelean wrote: > This change re-introduces `match_string()` as a macro that uses > ARRAY_SIZE() to compute the size of the array. > The macro is added in all the places that do > `match_string(_a, ARRAY_SIZE(_a), s)`, since the change is pretty > straightforward. Can you split include/linux/ change from the rest? > > Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> > --- > drivers/clk/bcm/clk-bcm2835.c | 4 +--- > drivers/gpio/gpiolib-of.c | 2 +- > drivers/gpu/drm/i915/intel_pipe_crc.c | 2 +- > drivers/mfd/omap-usb-host.c | 2 +- > drivers/net/wireless/intel/iwlwifi/mvm/debugfs.c | 2 +- > drivers/pci/pcie/aer.c | 2 +- > drivers/usb/common/common.c | 4 ++-- > drivers/usb/typec/class.c | 8 +++----- > drivers/usb/typec/tps6598x.c | 2 +- > drivers/vfio/vfio.c | 4 +--- > include/linux/string.h | 9 +++++++++ > sound/firewire/oxfw/oxfw.c | 2 +- > sound/soc/codecs/max98088.c | 2 +- > sound/soc/codecs/max98095.c | 2 +- > 14 files changed, 25 insertions(+), 22 deletions(-) > > diff --git a/drivers/clk/bcm/clk-bcm2835.c b/drivers/clk/bcm/clk-bcm2835.c > index a775f6a1f717..1ab388590ead 100644 > --- a/drivers/clk/bcm/clk-bcm2835.c > +++ b/drivers/clk/bcm/clk-bcm2835.c > @@ -1390,9 +1390,7 @@ static struct clk_hw *bcm2835_register_clock(struct bcm2835_cprman *cprman, > for (i = 0; i < data->num_mux_parents; i++) { > parents[i] = data->parents[i]; > > - ret = __match_string(cprman_parent_names, > - ARRAY_SIZE(cprman_parent_names), > - parents[i]); > + ret = match_string(cprman_parent_names, parents[i]); > if (ret >= 0) > parents[i] = cprman->real_parent_names[ret]; > } > diff --git a/drivers/gpio/gpiolib-of.c b/drivers/gpio/gpiolib-of.c > index 27d6f04ab58e..71e886869d78 100644 > --- a/drivers/gpio/gpiolib-of.c > +++ b/drivers/gpio/gpiolib-of.c > @@ -279,7 +279,7 @@ static struct gpio_desc *of_find_regulator_gpio(struct device *dev, const char * > if (!con_id) > return ERR_PTR(-ENOENT); > > - i = __match_string(whitelist, ARRAY_SIZE(whitelist), con_id); > + i = match_string(whitelist, con_id); > if (i < 0) > return ERR_PTR(-ENOENT); > > diff --git a/drivers/gpu/drm/i915/intel_pipe_crc.c b/drivers/gpu/drm/i915/intel_pipe_crc.c > index 286fad1f0e08..6fc4f3d3d1f6 100644 > --- a/drivers/gpu/drm/i915/intel_pipe_crc.c > +++ b/drivers/gpu/drm/i915/intel_pipe_crc.c > @@ -449,7 +449,7 @@ display_crc_ctl_parse_source(const char *buf, enum intel_pipe_crc_source *s) > return 0; > } > > - i = __match_string(pipe_crc_sources, ARRAY_SIZE(pipe_crc_sources), buf); > + i = match_string(pipe_crc_sources, buf); > if (i < 0) > return i; > > diff --git a/drivers/mfd/omap-usb-host.c b/drivers/mfd/omap-usb-host.c > index 9aaacb5bdb26..53dff34c0afc 100644 > --- a/drivers/mfd/omap-usb-host.c > +++ b/drivers/mfd/omap-usb-host.c > @@ -509,7 +509,7 @@ static int usbhs_omap_get_dt_pdata(struct device *dev, > continue; > > /* get 'enum usbhs_omap_port_mode' from port mode string */ > - ret = __match_string(port_modes, ARRAY_SIZE(port_modes), mode); > + ret = match_string(port_modes, mode); > if (ret < 0) { > dev_warn(dev, "Invalid port%d-mode \"%s\" in device tree\n", > i, mode); > diff --git a/drivers/net/wireless/intel/iwlwifi/mvm/debugfs.c b/drivers/net/wireless/intel/iwlwifi/mvm/debugfs.c > index 59ce3ff35553..778b4dfd8b75 100644 > --- a/drivers/net/wireless/intel/iwlwifi/mvm/debugfs.c > +++ b/drivers/net/wireless/intel/iwlwifi/mvm/debugfs.c > @@ -667,7 +667,7 @@ iwl_dbgfs_bt_force_ant_write(struct iwl_mvm *mvm, char *buf, > }; > int ret, bt_force_ant_mode; > > - ret = __match_string(modes_str, ARRAY_SIZE(modes_str), buf); > + ret = match_string(modes_str, buf); > if (ret < 0) > return ret; > > diff --git a/drivers/pci/pcie/aer.c b/drivers/pci/pcie/aer.c > index 41a0773a1cbc..2278caba109c 100644 > --- a/drivers/pci/pcie/aer.c > +++ b/drivers/pci/pcie/aer.c > @@ -203,7 +203,7 @@ void pcie_ecrc_get_policy(char *str) > { > int i; > > - i = __match_string(ecrc_policy_str, ARRAY_SIZE(ecrc_policy_str), str); > + i = match_string(ecrc_policy_str, str); > if (i < 0) > return; > > diff --git a/drivers/usb/common/common.c b/drivers/usb/common/common.c > index bca0c404c6ca..5a651d311d38 100644 > --- a/drivers/usb/common/common.c > +++ b/drivers/usb/common/common.c > @@ -68,7 +68,7 @@ enum usb_device_speed usb_get_maximum_speed(struct device *dev) > if (ret < 0) > return USB_SPEED_UNKNOWN; > > - ret = __match_string(speed_names, ARRAY_SIZE(speed_names), maximum_speed); > + ret = match_string(speed_names, maximum_speed); > > return (ret < 0) ? USB_SPEED_UNKNOWN : ret; > } > @@ -106,7 +106,7 @@ static enum usb_dr_mode usb_get_dr_mode_from_string(const char *str) > { > int ret; > > - ret = __match_string(usb_dr_modes, ARRAY_SIZE(usb_dr_modes), str); > + ret = match_string(usb_dr_modes, str); > return (ret < 0) ? USB_DR_MODE_UNKNOWN : ret; > } > > diff --git a/drivers/usb/typec/class.c b/drivers/usb/typec/class.c > index 4abc5a76ec51..38ac776cba8a 100644 > --- a/drivers/usb/typec/class.c > +++ b/drivers/usb/typec/class.c > @@ -1409,8 +1409,7 @@ EXPORT_SYMBOL_GPL(typec_set_pwr_opmode); > */ > int typec_find_port_power_role(const char *name) > { > - return __match_string(typec_port_power_roles, > - ARRAY_SIZE(typec_port_power_roles), name); > + return match_string(typec_port_power_roles, name); > } > EXPORT_SYMBOL_GPL(typec_find_port_power_role); > > @@ -1424,7 +1423,7 @@ EXPORT_SYMBOL_GPL(typec_find_port_power_role); > */ > int typec_find_power_role(const char *name) > { > - return __match_string(typec_roles, ARRAY_SIZE(typec_roles), name); > + return match_string(typec_roles, name); > } > EXPORT_SYMBOL_GPL(typec_find_power_role); > > @@ -1438,8 +1437,7 @@ EXPORT_SYMBOL_GPL(typec_find_power_role); > */ > int typec_find_port_data_role(const char *name) > { > - return __match_string(typec_port_data_roles, > - ARRAY_SIZE(typec_port_data_roles), name); > + return match_string(typec_port_data_roles, name); > } > EXPORT_SYMBOL_GPL(typec_find_port_data_role); > > diff --git a/drivers/usb/typec/tps6598x.c b/drivers/usb/typec/tps6598x.c > index 0389e4391faf..0c4e47868590 100644 > --- a/drivers/usb/typec/tps6598x.c > +++ b/drivers/usb/typec/tps6598x.c > @@ -423,7 +423,7 @@ static int tps6598x_check_mode(struct tps6598x *tps) > if (ret) > return ret; > > - switch (__match_string(modes, ARRAY_SIZE(modes), mode)) { > + switch (match_string(modes, mode)) { > case TPS_MODE_APP: > return 0; > case TPS_MODE_BOOT: > diff --git a/drivers/vfio/vfio.c b/drivers/vfio/vfio.c > index b31585ecf48f..fe8283d3781b 100644 > --- a/drivers/vfio/vfio.c > +++ b/drivers/vfio/vfio.c > @@ -637,9 +637,7 @@ static bool vfio_dev_whitelisted(struct device *dev, struct device_driver *drv) > return true; > } > > - return __match_string(vfio_driver_whitelist, > - ARRAY_SIZE(vfio_driver_whitelist), > - drv->name) >= 0; > + return match_string(vfio_driver_whitelist, drv->name) >= 0; > } > > /* > diff --git a/include/linux/string.h b/include/linux/string.h > index 531d04308ff9..07e9f89088df 100644 > --- a/include/linux/string.h > +++ b/include/linux/string.h > @@ -194,6 +194,15 @@ static inline int strtobool(const char *s, bool *res) > int __match_string(const char * const *array, size_t n, const char *string); > int __sysfs_match_string(const char * const *array, size_t n, const char *s); > > +/** > + * match_string - matches given string in an array > + * @_a: array of strings > + * @_s: string to match with > + * > + * Helper for __match_string(). Calculates the size of @a automatically. > + */ > +#define match_string(_a, _s) __match_string(_a, ARRAY_SIZE(_a), _s) > + > /** > * sysfs_match_string - matches given string in an array > * @_a: array of strings > diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c > index 9ec5316f3bb5..433fc84c4f90 100644 > --- a/sound/firewire/oxfw/oxfw.c > +++ b/sound/firewire/oxfw/oxfw.c > @@ -57,7 +57,7 @@ static bool detect_loud_models(struct fw_unit *unit) > if (err < 0) > return false; > > - return __match_string(models, ARRAY_SIZE(models), model) >= 0; > + return match_string(models, model) >= 0; > } > > static int name_card(struct snd_oxfw *oxfw) > diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c > index 3ef743075bda..911ffe84c37e 100644 > --- a/sound/soc/codecs/max98088.c > +++ b/sound/soc/codecs/max98088.c > @@ -1405,7 +1405,7 @@ static int max98088_get_channel(struct snd_soc_component *component, const char > { > int ret; > > - ret = __match_string(eq_mode_name, ARRAY_SIZE(eq_mode_name), name); > + ret = match_string(eq_mode_name, name); > if (ret < 0) > dev_err(component->dev, "Bad EQ channel name '%s'\n", name); > return ret; > diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c > index cd69916d5dcb..d182d45d0c83 100644 > --- a/sound/soc/codecs/max98095.c > +++ b/sound/soc/codecs/max98095.c > @@ -1636,7 +1636,7 @@ static int max98095_get_bq_channel(struct snd_soc_component *component, > { > int ret; > > - ret = __match_string(bq_mode_name, ARRAY_SIZE(bq_mode_name), name); > + ret = match_string(bq_mode_name, name); > if (ret < 0) > dev_err(component->dev, "Bad biquad channel name '%s'\n", name); > return ret; > -- > 2.17.1 > -- With Best Regards, Andy Shevchenko

4 5

[alsa-devel] [PATCH v2 0/4] ASoC: hdmi-codec: fixes and improvements
by Jerome Brunet 12 May '19

12 May '19

This patchset is a collection of fixes and improvement on the hdmi codec driver. It should be completely transparent for the current users of the codec. The most important change is removal the current_substream pointer which allows the codec to be used on codec-to-codec links. I plan to use this for the HDMI sound support of the Amlogic g12a SoC family. Changes since v1: * Drop already applied patches * Rebase patchset on sound/for-next Jerome Brunet (4): ASoC: hdmi-codec: remove function name debug traces ASoC: hdmi-codec: remove reference to the current substream ASoC: hdmi-codec: remove reference to the dai drivers in the private data ASoC: hdmi-codec: remove ops dependency on the dai id sound/soc/codecs/hdmi-codec.c | 188 ++++++++++++++++------------------ 1 file changed, 91 insertions(+), 97 deletions(-) -- 2.20.1

2 15

[alsa-devel] 1.1.9: test suite issue: axfer/test -> container-test loops forever
by GitHub issues - opened 11 May '19

11 May '19

alsa-project/alsa-utils issue #19 was opened from kloczek: Looks like it is some issue with axfer/test -> container-test. This test cannot finishe and looks like it is looping and cannot finish. Just in case: nothing is written on stdout/stderr or container-test.log. axfer/test/hoge file is constantly written and deleted. <pre>[tkloczko@domek test]$ strace -f /bin/sh ../../test-driver --test-name container-test --log-file container-test.log --trs-file container-test.trs --color-tests yes --enable-hard-errors yes --expect-failure no -- ./container-test execve("/bin/sh", ["/bin/sh", "../../test-driver", "--test-name", "container-test", "--log-file", "container-test.log", "--trs-file", "container-test.trs", "--color-tests", "yes", "--enable-hard-errors", "yes", "--expect-failure", "no", "--", "./container-test"], 0x7fff8aab0350 /* 67 vars */) = 0 brk(NULL) = 0x559a60017000 arch_prctl(0x3001 /* ARCH_??? */, 0x7ffce069bb50) = -1 EINVAL (Invalid argument) access("/etc/ld.so.preload", R_OK) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/etc/ld.so.cache", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=188179, ...}) = 0 mmap(NULL, 188179, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7fd3e422c000 close(3) = 0 openat(AT_FDCWD, "/lib64/libtinfo.so.6", O_RDONLY|O_CLOEXEC) = 3 read(3, "\177ELF\2\1\1\0\0\0\0\0\0\0\0\0\3\0>\0\1\0\0\0\320\363\0\0\0\0\0\0"..., 832) = 832 lseek(3, 169216, SEEK_SET) = 169216 read(3, "\4\0\0\0\20\0\0\0\5\0\0\0GNU\0\2\0\0\300\4\0\0\0\3\0\0\0\0\0\0\0", 32) = 32 fstat(3, {st_mode=S_IFREG|0755, st_size=285768, ...}) = 0 mmap(NULL, 8192, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x7fd3e422a000 lseek(3, 169216, SEEK_SET) = 169216 read(3, "\4\0\0\0\20\0\0\0\5\0\0\0GNU\0\2\0\0\300\4\0\0\0\3\0\0\0\0\0\0\0", 32) = 32 mmap(NULL, 190848, PROT_READ, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x7fd3e41fb000 mmap(0x7fd3e4209000, 61440, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xe000) = 0x7fd3e4209000 mmap(0x7fd3e4218000, 53248, PROT_READ, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1d000) = 0x7fd3e4218000 mmap(0x7fd3e4225000, 20480, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x29000) = 0x7fd3e4225000 close(3) = 0 openat(AT_FDCWD, "/lib64/libdl.so.2", O_RDONLY|O_CLOEXEC) = 3 read(3, "\177ELF\2\1\1\0\0\0\0\0\0\0\0\0\3\0>\0\1\0\0\0p\"\0\0\0\0\0\0"..., 832) = 832 fstat(3, {st_mode=S_IFREG|0755, st_size=56640, ...}) = 0 mmap(NULL, 24688, PROT_READ, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x7fd3e41f4000 mmap(0x7fd3e41f6000, 8192, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2000) = 0x7fd3e41f6000 mmap(0x7fd3e41f8000, 4096, PROT_READ, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x4000) = 0x7fd3e41f8000 mmap(0x7fd3e41f9000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x4000) = 0x7fd3e41f9000 mmap(0x7fd3e41fa000, 112, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x7fd3e41fa000 close(3) = 0 openat(AT_FDCWD, "/lib64/libc.so.6", O_RDONLY|O_CLOEXEC) = 3 read(3, "\177ELF\2\1\1\3\0\0\0\0\0\0\0\0\3\0>\0\1\0\0\0\200p\2\0\0\0\0\0"..., 832) = 832 lseek(3, 64, SEEK_SET) = 64 read(3, "\6\0\0\0\4\0\0\0@\0\0\0\0\0\0\0@\0\0\0\0\0\0\0@\0\0\0\0\0\0\0"..., 784) = 784 lseek(3, 848, SEEK_SET) = 848 read(3, "\4\0\0\0\20\0\0\0\5\0\0\0GNU\0\2\0\0\300\4\0\0\0\3\0\0\0\0\0\0\0", 32) = 32 lseek(3, 880, SEEK_SET) = 880 read(3, "\4\0\0\0\24\0\0\0\3\0\0\0GNU\0\0052`U\224\217\3<\204\4\231e\235\313\324 "..., 68) = 68 fstat(3, {st_mode=S_IFREG|0755, st_size=5577040, ...}) = 0 lseek(3, 64, SEEK_SET) = 64 read(3, "\6\0\0\0\4\0\0\0@\0\0\0\0\0\0\0@\0\0\0\0\0\0\0@\0\0\0\0\0\0\0"..., 784) = 784 lseek(3, 848, SEEK_SET) = 848 read(3, "\4\0\0\0\20\0\0\0\5\0\0\0GNU\0\2\0\0\300\4\0\0\0\3\0\0\0\0\0\0\0", 32) = 32 lseek(3, 880, SEEK_SET) = 880 read(3, "\4\0\0\0\24\0\0\0\3\0\0\0GNU\0\0052`U\224\217\3<\204\4\231e\235\313\324 "..., 68) = 68 mmap(NULL, 1852600, PROT_READ, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x7fd3e402f000 mprotect(0x7fd3e4054000, 1662976, PROT_NONE) = 0 mmap(0x7fd3e4054000, 1359872, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x25000) = 0x7fd3e4054000 mmap(0x7fd3e41a0000, 299008, PROT_READ, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x171000) = 0x7fd3e41a0000 mmap(0x7fd3e41ea000, 24576, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1ba000) = 0x7fd3e41ea000 mmap(0x7fd3e41f0000, 13496, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x7fd3e41f0000 close(3) = 0 mmap(NULL, 12288, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x7fd3e402c000 arch_prctl(ARCH_SET_FS, 0x7fd3e402c740) = 0 mprotect(0x7fd3e41ea000, 12288, PROT_READ) = 0 mprotect(0x7fd3e41f9000, 4096, PROT_READ) = 0 mprotect(0x7fd3e4225000, 16384, PROT_READ) = 0 mprotect(0x559a5eca6000, 16384, PROT_READ) = 0 mprotect(0x7fd3e4284000, 4096, PROT_READ) = 0 munmap(0x7fd3e422c000, 188179) = 0 openat(AT_FDCWD, "/dev/tty", O_RDWR|O_NONBLOCK) = 3 close(3) = 0 brk(NULL) = 0x559a60017000 brk(0x559a60038000) = 0x559a60038000 brk(NULL) = 0x559a60038000 openat(AT_FDCWD, "/usr/lib/locale/locale-archive", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/usr/share/locale/locale.alias", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=2997, ...}) = 0 read(3, "# Locale name alias data base.\n#"..., 4096) = 2997 read(3, "", 4096) = 0 close(3) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.UTF-8/LC_IDENTIFICATION", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/usr/lib/locale/en_GB.utf8/LC_IDENTIFICATION", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=373, ...}) = 0 mmap(NULL, 373, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7fd3e4283000 close(3) = 0 openat(AT_FDCWD, "/usr/lib64/gconv/gconv-modules.cache", O_RDONLY) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=26998, ...}) = 0 mmap(NULL, 26998, PROT_READ, MAP_SHARED, 3, 0) = 0x7fd3e4253000 close(3) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.utf8/LC_MEASUREMENT", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=23, ...}) = 0 mmap(NULL, 23, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7fd3e4252000 close(3) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.UTF-8/LC_TELEPHONE", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/usr/lib/locale/en_GB.utf8/LC_TELEPHONE", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=56, ...}) = 0 mmap(NULL, 56, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7fd3e4251000 close(3) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.UTF-8/LC_ADDRESS", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/usr/lib/locale/en_GB.utf8/LC_ADDRESS", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=158, ...}) = 0 mmap(NULL, 158, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7fd3e4250000 close(3) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.UTF-8/LC_NAME", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/usr/lib/locale/en_GB.utf8/LC_NAME", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=77, ...}) = 0 mmap(NULL, 77, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7fd3e424f000 close(3) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.utf8/LC_PAPER", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=34, ...}) = 0 mmap(NULL, 34, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7fd3e424e000 close(3) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.UTF-8/LC_MESSAGES", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/usr/lib/locale/en_GB.utf8/LC_MESSAGES", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFDIR|0755, st_size=30, ...}) = 0 close(3) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.utf8/LC_MESSAGES/SYS_LC_MESSAGES", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=57, ...}) = 0 mmap(NULL, 57, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7fd3e424d000 close(3) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.utf8/LC_MONETARY", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=290, ...}) = 0 mmap(NULL, 290, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7fd3e424c000 close(3) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.UTF-8/LC_COLLATE", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/usr/lib/locale/en_GB.utf8/LC_COLLATE", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=2586930, ...}) = 0 mmap(NULL, 2586930, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7fd3e3db4000 close(3) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.utf8/LC_TIME", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=3332, ...}) = 0 mmap(NULL, 3332, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7fd3e424b000 close(3) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.utf8/LC_NUMERIC", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=54, ...}) = 0 mmap(NULL, 54, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7fd3e424a000 close(3) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.UTF-8/LC_CTYPE", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/usr/lib/locale/en_GB.utf8/LC_CTYPE", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=340008, ...}) = 0 mmap(NULL, 340008, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7fd3e3d60000 close(3) = 0 getuid() = 1000 getgid() = 1000 geteuid() = 1000 getegid() = 1000 rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0 ioctl(-1, TIOCGPGRP, 0x7ffce069b9b4) = -1 EBADF (Bad file descriptor) sysinfo({uptime=447376, loads=[64256, 115168, 149856], totalram=8272330752, freeram=130670592, sharedram=986529792, bufferram=106496, totalswap=0, freeswap=0, procs=1208, totalhigh=0, freehigh=0, mem_unit=1}) = 0 rt_sigaction(SIGCHLD, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER|SA_RESTART, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=0}, 8) = 0 rt_sigaction(SIGCHLD, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER|SA_RESTART, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER|SA_RESTART, sa_restorer=0x7fd3e4069f40}, 8) = 0 rt_sigaction(SIGINT, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=0}, 8) = 0 rt_sigaction(SIGINT, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 rt_sigaction(SIGQUIT, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=0}, 8) = 0 rt_sigaction(SIGQUIT, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 rt_sigaction(SIGTSTP, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=0}, 8) = 0 rt_sigaction(SIGTSTP, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 rt_sigaction(SIGTTIN, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=0}, 8) = 0 rt_sigaction(SIGTTIN, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 rt_sigaction(SIGTTOU, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=0}, 8) = 0 rt_sigaction(SIGTTOU, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0 rt_sigaction(SIGQUIT, {sa_handler=SIG_IGN, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 uname({sysname="Linux", nodename="domek", ...}) = 0 rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0 rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0 stat("/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.9/axfer/test", {st_mode=S_IFDIR|0755, st_size=532, ...}) = 0 stat(".", {st_mode=S_IFDIR|0755, st_size=532, ...}) = 0 stat("/home", {st_mode=S_IFDIR|0755, st_size=16, ...}) = 0 stat("/home/tkloczko", {st_mode=S_IFDIR|0755, st_size=2148, ...}) = 0 stat("/home/tkloczko/rpmbuild", {st_mode=S_IFDIR|0755, st_size=1464, ...}) = 0 stat("/home/tkloczko/rpmbuild/BUILD", {st_mode=S_IFDIR|0755, st_size=2344, ...}) = 0 stat("/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.9", {st_mode=S_IFDIR|0755, st_size=966, ...}) = 0 stat("/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.9/axfer", {st_mode=S_IFDIR|0755, st_size=1634, ...}) = 0 stat("/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.9/axfer/test", {st_mode=S_IFDIR|0755, st_size=532, ...}) = 0 stat("/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.9/axfer", {st_mode=S_IFDIR|0755, st_size=1634, ...}) = 0 getpid() = 24893 getppid() = 24890 getpid() = 24893 getpgrp() = 24890 ioctl(2, TIOCGPGRP, [24890]) = 0 rt_sigaction(SIGCHLD, {sa_handler=0x559a5ebecd60, sa_mask=[], sa_flags=SA_RESTORER|SA_RESTART, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER|SA_RESTART, sa_restorer=0x7fd3e4069f40}, 8) = 0 prlimit64(0, RLIMIT_NPROC, NULL, {rlim_cur=31438, rlim_max=31438}) = 0 openat(AT_FDCWD, "/usr/lib/locale/en_GB.UTF-8/LC_MEASUREMENT", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/usr/lib/locale/en_GB.UTF-8/LC_PAPER", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/usr/lib/locale/en_GB.UTF-8/LC_MONETARY", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/usr/lib/locale/en_GB.UTF-8/LC_TIME", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/usr/lib/locale/en_GB.UTF-8/LC_NUMERIC", O_RDONLY|O_CLOEXEC) = -1 ENOENT (No such file or directory) rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0 openat(AT_FDCWD, "../../test-driver", O_RDONLY) = 3 stat("../../test-driver", {st_mode=S_IFREG|0755, st_size=4645, ...}) = 0 ioctl(3, TCGETS, 0x7ffce069b940) = -1 ENOTTY (Inappropriate ioctl for device) lseek(3, 0, SEEK_CUR) = 0 read(3, "#!/usr/bin/sh\n# test-driver - ba"..., 80) = 80 lseek(3, 0, SEEK_SET) = 0 prlimit64(0, RLIMIT_NOFILE, NULL, {rlim_cur=1024, rlim_max=512*1024}) = 0 fcntl(255, F_GETFD) = -1 EBADF (Bad file descriptor) dup2(3, 255) = 255 close(3) = 0 fcntl(255, F_SETFD, FD_CLOEXEC) = 0 fcntl(255, F_GETFL) = 0x8000 (flags O_RDONLY|O_LARGEFILE) fstat(255, {st_mode=S_IFREG|0755, st_size=4645, ...}) = 0 lseek(255, 0, SEEK_CUR) = 0 read(255, "#!/usr/bin/sh\n# test-driver - ba"..., 4645) = 4645 rt_sigaction(SIGHUP, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=0}, 8) = 0 rt_sigaction(SIGHUP, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 rt_sigprocmask(SIG_BLOCK, [HUP], [], 8) = 0 rt_sigaction(SIGHUP, {sa_handler=0x559a5ec06c40, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 rt_sigprocmask(SIG_SETMASK, [], NULL, 8) = 0 rt_sigprocmask(SIG_BLOCK, [INT], [], 8) = 0 rt_sigaction(SIGINT, {sa_handler=0x559a5ec06c40, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 rt_sigprocmask(SIG_SETMASK, [], NULL, 8) = 0 rt_sigaction(SIGPIPE, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=0}, 8) = 0 rt_sigaction(SIGPIPE, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 rt_sigprocmask(SIG_BLOCK, [PIPE], [], 8) = 0 rt_sigaction(SIGPIPE, {sa_handler=0x559a5ec06c40, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 rt_sigprocmask(SIG_SETMASK, [], NULL, 8) = 0 rt_sigaction(SIGTERM, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=0}, 8) = 0 rt_sigaction(SIGTERM, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 rt_sigprocmask(SIG_BLOCK, [TERM], [], 8) = 0 rt_sigaction(SIGTERM, {sa_handler=0x559a5ec06c40, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 rt_sigprocmask(SIG_SETMASK, [], NULL, 8) = 0 rt_sigprocmask(SIG_BLOCK, [INT CHLD], [], 8) = 0 lseek(255, -1344, SEEK_CUR) = 3301 clone(strace: Process 24894 attached child_stack=NULL, flags=CLONE_CHILD_CLEARTID|CLONE_CHILD_SETTID|SIGCHLD, child_tidptr=0x7fd3e402ca10) = 24894 [pid 24894] getpid( <unfinished ...> [pid 24893] rt_sigprocmask(SIG_SETMASK, [], <unfinished ...> [pid 24894] <... getpid resumed>) = 24894 [pid 24893] <... rt_sigprocmask resumed>NULL, 8) = 0 [pid 24894] close(255 <unfinished ...> [pid 24893] rt_sigprocmask(SIG_BLOCK, [CHLD], <unfinished ...> [pid 24894] <... close resumed>) = 0 [pid 24893] <... rt_sigprocmask resumed>[], 8) = 0 [pid 24894] rt_sigprocmask(SIG_SETMASK, [], <unfinished ...> [pid 24893] rt_sigprocmask(SIG_SETMASK, [], <unfinished ...> [pid 24894] <... rt_sigprocmask resumed>NULL, 8) = 0 [pid 24893] <... rt_sigprocmask resumed>NULL, 8) = 0 [pid 24894] rt_sigaction(SIGTSTP, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, <unfinished ...> [pid 24893] rt_sigprocmask(SIG_BLOCK, [CHLD], <unfinished ...> [pid 24894] <... rt_sigaction resumed>{sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 [pid 24893] <... rt_sigprocmask resumed>[], 8) = 0 [pid 24894] rt_sigaction(SIGTTIN, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, <unfinished ...> [pid 24893] rt_sigaction(SIGINT, {sa_handler=0x559a5ebe9810, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, <unfinished ...> [pid 24894] <... rt_sigaction resumed>{sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 [pid 24893] <... rt_sigaction resumed>{sa_handler=0x559a5ec06c40, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 [pid 24894] rt_sigaction(SIGTTOU, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, <unfinished ...> [pid 24893] wait4(-1, <unfinished ...> [pid 24894] <... rt_sigaction resumed>{sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 [pid 24894] rt_sigaction(SIGHUP, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=0x559a5ec06c40, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 [pid 24894] rt_sigaction(SIGINT, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=0x559a5ec06c40, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 [pid 24894] rt_sigaction(SIGQUIT, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=SIG_IGN, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 [pid 24894] rt_sigaction(SIGPIPE, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=0x559a5ec06c40, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 [pid 24894] rt_sigaction(SIGTERM, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, {sa_handler=0x559a5ec06c40, sa_mask=[], sa_flags=SA_RESTORER, sa_restorer=0x7fd3e4069f40}, 8) = 0 [pid 24894] rt_sigaction(SIGCHLD, {sa_handler=SIG_DFL, sa_mask=[], sa_flags=SA_RESTORER|SA_RESTART, sa_restorer=0x7fd3e4069f40}, {sa_handler=0x559a5ebecd60, sa_mask=[], sa_flags=SA_RESTORER|SA_RESTART, sa_restorer=0x7fd3e4069f40}, 8) = 0 [pid 24894] openat(AT_FDCWD, "container-test.log", O_WRONLY|O_CREAT|O_TRUNC, 0666) = 3 [pid 24894] dup2(3, 1) = 1 [pid 24894] close(3) = 0 [pid 24894] dup2(1, 2) = 2 [pid 24894] fcntl(1, F_GETFD) = 0 [pid 24894] execve("./container-test", ["./container-test"], 0x559a60029510 /* 67 vars */) = 0 [pid 24894] brk(NULL) = 0x55711c860000 [pid 24894] arch_prctl(0x3001 /* ARCH_??? */, 0x7ffff5c380a0) = -1 EINVAL (Invalid argument) [pid 24894] access("/etc/ld.so.preload", R_OK) = -1 ENOENT (No such file or directory) [pid 24894] openat(AT_FDCWD, "/etc/ld.so.cache", O_RDONLY|O_CLOEXEC) = 3 [pid 24894] fstat(3, {st_mode=S_IFREG|0644, st_size=188179, ...}) = 0 [pid 24894] mmap(NULL, 188179, PROT_READ, MAP_PRIVATE, 3, 0) = 0x7f302eae3000 [pid 24894] close(3) = 0 [pid 24894] openat(AT_FDCWD, "/lib64/libc.so.6", O_RDONLY|O_CLOEXEC) = 3 [pid 24894] read(3, "\177ELF\2\1\1\3\0\0\0\0\0\0\0\0\3\0>\0\1\0\0\0\200p\2\0\0\0\0\0"..., 832) = 832 [pid 24894] lseek(3, 64, SEEK_SET) = 64 [pid 24894] read(3, "\6\0\0\0\4\0\0\0@\0\0\0\0\0\0\0@\0\0\0\0\0\0\0@\0\0\0\0\0\0\0"..., 784) = 784 [pid 24894] lseek(3, 848, SEEK_SET) = 848 [pid 24894] read(3, "\4\0\0\0\20\0\0\0\5\0\0\0GNU\0\2\0\0\300\4\0\0\0\3\0\0\0\0\0\0\0", 32) = 32 [pid 24894] lseek(3, 880, SEEK_SET) = 880 [pid 24894] read(3, "\4\0\0\0\24\0\0\0\3\0\0\0GNU\0\0052`U\224\217\3<\204\4\231e\235\313\324 "..., 68) = 68 [pid 24894] fstat(3, {st_mode=S_IFREG|0755, st_size=5577040, ...}) = 0 [pid 24894] mmap(NULL, 8192, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x7f302eae1000 [pid 24894] lseek(3, 64, SEEK_SET) = 64 [pid 24894] read(3, "\6\0\0\0\4\0\0\0@\0\0\0\0\0\0\0@\0\0\0\0\0\0\0@\0\0\0\0\0\0\0"..., 784) = 784 [pid 24894] lseek(3, 848, SEEK_SET) = 848 [pid 24894] read(3, "\4\0\0\0\20\0\0\0\5\0\0\0GNU\0\2\0\0\300\4\0\0\0\3\0\0\0\0\0\0\0", 32) = 32 [pid 24894] lseek(3, 880, SEEK_SET) = 880 [pid 24894] read(3, "\4\0\0\0\24\0\0\0\3\0\0\0GNU\0\0052`U\224\217\3<\204\4\231e\235\313\324 "..., 68) = 68 [pid 24894] mmap(NULL, 1852600, PROT_READ, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x7f302e91c000 [pid 24894] mprotect(0x7f302e941000, 1662976, PROT_NONE) = 0 [pid 24894] mmap(0x7f302e941000, 1359872, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x25000) = 0x7f302e941000 [pid 24894] mmap(0x7f302ea8d000, 299008, PROT_READ, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x171000) = 0x7f302ea8d000 [pid 24894] mmap(0x7f302ead7000, 24576, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1ba000) = 0x7f302ead7000 [pid 24894] mmap(0x7f302eadd000, 13496, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x7f302eadd000 [pid 24894] close(3) = 0 [pid 24894] openat(AT_FDCWD, "/lib64/libasound.so.2", O_RDONLY|O_CLOEXEC) = 3 [pid 24894] read(3, "\177ELF\2\1\1\0\0\0\0\0\0\0\0\0\3\0>\0\1\0\0\0\260'\3\0\0\0\0\0"..., 832) = 832 [pid 24894] fstat(3, {st_mode=S_IFREG|0755, st_size=1157768, ...}) = 0 [pid 24894] mmap(NULL, 955152, PROT_READ, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x7f302e832000 [pid 24894] mmap(0x7f302e85f000, 552960, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2d000) = 0x7f302e85f000 [pid 24894] mmap(0x7f302e8e6000, 184320, PROT_READ, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xb4000) = 0x7f302e8e6000 [pid 24894] mmap(0x7f302e913000, 36864, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xe0000) = 0x7f302e913000 [pid 24894] close(3) = 0 [pid 24894] openat(AT_FDCWD, "/lib64/libm.so.6", O_RDONLY|O_CLOEXEC) = 3 [pid 24894] read(3, "\177ELF\2\1\1\3\0\0\0\0\0\0\0\0\3\0>\0\1\0\0\0\220\363\0\0\0\0\0\0"..., 832) = 832 [pid 24894] fstat(3, {st_mode=S_IFREG|0755, st_size=3315312, ...}) = 0 [pid 24894] mmap(NULL, 1331216, PROT_READ, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x7f302e6ec000 [pid 24894] mmap(0x7f302e6fb000, 638976, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xf000) = 0x7f302e6fb000 [pid 24894] mmap(0x7f302e797000, 626688, PROT_READ, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xab000) = 0x7f302e797000 [pid 24894] mmap(0x7f302e830000, 8192, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x143000) = 0x7f302e830000 [pid 24894] close(3) = 0 [pid 24894] openat(AT_FDCWD, "/lib64/libdl.so.2", O_RDONLY|O_CLOEXEC) = 3 [pid 24894] read(3, "\177ELF\2\1\1\0\0\0\0\0\0\0\0\0\3\0>\0\1\0\0\0p\"\0\0\0\0\0\0"..., 832) = 832 [pid 24894] fstat(3, {st_mode=S_IFREG|0755, st_size=56640, ...}) = 0 [pid 24894] mmap(NULL, 24688, PROT_READ, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x7f302e6e5000 [pid 24894] mmap(0x7f302e6e7000, 8192, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2000) = 0x7f302e6e7000 [pid 24894] mmap(0x7f302e6e9000, 4096, PROT_READ, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x4000) = 0x7f302e6e9000 [pid 24894] mmap(0x7f302e6ea000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x4000) = 0x7f302e6ea000 [pid 24894] mmap(0x7f302e6eb000, 112, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x7f302e6eb000 [pid 24894] close(3) = 0 [pid 24894] openat(AT_FDCWD, "/lib64/libpthread.so.0", O_RDONLY|O_CLOEXEC) = 3 [pid 24894] read(3, "\177ELF\2\1\1\3\0\0\0\0\0\0\0\0\3\0>\0\1\0\0\0\0\202\0\0\0\0\0\0"..., 832) = 832 [pid 24894] lseek(3, 824, SEEK_SET) = 824 [pid 24894] read(3, "\4\0\0\0\24\0\0\0\3\0\0\0GNU\0=K\6\363\272\306<\320\35\37vOsr\212\341"..., 68) = 68 [pid 24894] fstat(3, {st_mode=S_IFREG|0755, st_size=695360, ...}) = 0 [pid 24894] lseek(3, 824, SEEK_SET) = 824 [pid 24894] read(3, "\4\0\0\0\24\0\0\0\3\0\0\0GNU\0=K\6\363\272\306<\320\35\37vOsr\212\341"..., 68) = 68 [pid 24894] mmap(NULL, 131552, PROT_READ, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x7f302e6c4000 [pid 24894] mmap(0x7f302e6cb000, 61440, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x7000) = 0x7f302e6cb000 [pid 24894] mmap(0x7f302e6da000, 20480, PROT_READ, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x16000) = 0x7f302e6da000 [pid 24894] mmap(0x7f302e6df000, 8192, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1a000) = 0x7f302e6df000 [pid 24894] mmap(0x7f302e6e1000, 12768, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x7f302e6e1000 [pid 24894] close(3) = 0 [pid 24894] mmap(NULL, 8192, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x7f302e6c2000 [pid 24894] arch_prctl(ARCH_SET_FS, 0x7f302e6c32c0) = 0 [pid 24894] mprotect(0x7f302ead7000, 12288, PROT_READ) = 0 [pid 24894] mprotect(0x7f302e6df000, 4096, PROT_READ) = 0 [pid 24894] mprotect(0x7f302e6ea000, 4096, PROT_READ) = 0 [pid 24894] mprotect(0x7f302e830000, 4096, PROT_READ) = 0 [pid 24894] mprotect(0x7f302e913000, 32768, PROT_READ) = 0 [pid 24894] mprotect(0x55711c760000, 4096, PROT_READ) = 0 [pid 24894] mprotect(0x7f302eb3b000, 4096, PROT_READ) = 0 [pid 24894] munmap(0x7f302eae3000, 188179) = 0 [pid 24894] set_tid_address(0x7f302e6c3590) = 24894 [pid 24894] set_robust_list(0x7f302e6c35a0, 24) = 0 [pid 24894] rt_sigaction(SIGRTMIN, {sa_handler=0x7f302e6cbc50, sa_mask=[], sa_flags=SA_RESTORER|SA_SIGINFO, sa_restorer=0x7f302e6d7d40}, NULL, 8) = 0 [pid 24894] rt_sigaction(SIGRT_1, {sa_handler=0x7f302e6cbcf0, sa_mask=[], sa_flags=SA_RESTORER|SA_RESTART|SA_SIGINFO, sa_restorer=0x7f302e6d7d40}, NULL, 8) = 0 [pid 24894] rt_sigprocmask(SIG_UNBLOCK, [RTMIN RT_1], NULL, 8) = 0 [pid 24894] prlimit64(0, RLIMIT_STACK, NULL, {rlim_cur=8192*1024, rlim_max=RLIM64_INFINITY}) = 0 [pid 24894] openat(AT_FDCWD, "/dev/urandom", O_RDONLY) = 3 [pid 24894] brk(NULL) = 0x55711c860000 [pid 24894] brk(0x55711c881000) = 0x55711c881000 [pid 24894] brk(NULL) = 0x55711c881000 [pid 24894] read(3, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25\246\334W\337\222RK\262\200"..., 4500) = 4500 [pid 24894] unlink("hoge") = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDWR|O_CREAT|O_TRUNC|O_NONBLOCK, 0644) = 4 [pid 24894] write(4, "RIFF\377\377\377\377", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0D\254\0\0D\254\0\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\333\377\377\377", 8) = 8 [pid 24894] write(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25", 23) = 23 [pid 24894] lseek(4, 0, SEEK_SET) = 0 [pid 24894] write(4, "RIFF_\0\0\0", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0D\254\0\0D\254\0\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\27\0\0\0", 8) = 8 [pid 24894] fsync(4) = 0 [pid 24894] close(4) = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDONLY|O_NONBLOCK) = 4 [pid 24894] read(4, "RIFF", 4) = 4 [pid 24894] read(4, "_\0\0\0", 4) = 4 [pid 24894] read(4, "WAVE", 4) = 4 [pid 24894] read(4, "fmt \20\0\0\0", 8) = 8 [pid 24894] read(4, "\1\0\1\0D\254\0\0D\254\0\0\1\0\10\0", 16) = 16 [pid 24894] read(4, "data\27\0\0\0", 8) = 8 [pid 24894] read(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25", 23) = 23 [pid 24894] close(4) = 0 [pid 24894] unlink("hoge") = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDWR|O_CREAT|O_TRUNC|O_NONBLOCK, 0644) = 4 [pid 24894] write(4, "RIFF\377\377\377\377", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\200\273\0\0\200\273\0\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\333\377\377\377", 8) = 8 [pid 24894] write(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25", 23) = 23 [pid 24894] lseek(4, 0, SEEK_SET) = 0 [pid 24894] write(4, "RIFF_\0\0\0", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\200\273\0\0\200\273\0\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\27\0\0\0", 8) = 8 [pid 24894] fsync(4) = 0 [pid 24894] close(4) = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDONLY|O_NONBLOCK) = 4 [pid 24894] read(4, "RIFF", 4) = 4 [pid 24894] read(4, "_\0\0\0", 4) = 4 [pid 24894] read(4, "WAVE", 4) = 4 [pid 24894] read(4, "fmt \20\0\0\0", 8) = 8 [pid 24894] read(4, "\1\0\1\0\200\273\0\0\200\273\0\0\1\0\10\0", 16) = 16 [pid 24894] read(4, "data\27\0\0\0", 8) = 8 [pid 24894] read(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25", 23) = 23 [pid 24894] close(4) = 0 [pid 24894] unlink("hoge") = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDWR|O_CREAT|O_TRUNC|O_NONBLOCK, 0644) = 4 [pid 24894] write(4, "RIFF\377\377\377\377", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\210X\1\0\210X\1\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\333\377\377\377", 8) = 8 [pid 24894] write(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25", 23) = 23 [pid 24894] lseek(4, 0, SEEK_SET) = 0 [pid 24894] write(4, "RIFF_\0\0\0", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\210X\1\0\210X\1\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\27\0\0\0", 8) = 8 [pid 24894] fsync(4) = 0 [pid 24894] close(4) = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDONLY|O_NONBLOCK) = 4 [pid 24894] read(4, "RIFF", 4) = 4 [pid 24894] read(4, "_\0\0\0", 4) = 4 [pid 24894] read(4, "WAVE", 4) = 4 [pid 24894] read(4, "fmt \20\0\0\0", 8) = 8 [pid 24894] read(4, "\1\0\1\0\210X\1\0\210X\1\0\1\0\10\0", 16) = 16 [pid 24894] read(4, "data\27\0\0\0", 8) = 8 [pid 24894] read(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25", 23) = 23 [pid 24894] close(4) = 0 [pid 24894] unlink("hoge") = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDWR|O_CREAT|O_TRUNC|O_NONBLOCK, 0644) = 4 [pid 24894] write(4, "RIFF\377\377\377\377", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\0w\1\0\0w\1\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\333\377\377\377", 8) = 8 [pid 24894] write(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25", 23) = 23 [pid 24894] lseek(4, 0, SEEK_SET) = 0 [pid 24894] write(4, "RIFF_\0\0\0", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\0w\1\0\0w\1\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\27\0\0\0", 8) = 8 [pid 24894] fsync(4) = 0 [pid 24894] close(4) = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDONLY|O_NONBLOCK) = 4 [pid 24894] read(4, "RIFF", 4) = 4 [pid 24894] read(4, "_\0\0\0", 4) = 4 [pid 24894] read(4, "WAVE", 4) = 4 [pid 24894] read(4, "fmt \20\0\0\0", 8) = 8 [pid 24894] read(4, "\1\0\1\0\0w\1\0\0w\1\0\1\0\10\0", 16) = 16 [pid 24894] read(4, "data\27\0\0\0", 8) = 8 [pid 24894] read(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25", 23) = 23 [pid 24894] close(4) = 0 [pid 24894] unlink("hoge") = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDWR|O_CREAT|O_TRUNC|O_NONBLOCK, 0644) = 4 [pid 24894] write(4, "RIFF\377\377\377\377", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\20\261\2\0\20\261\2\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\333\377\377\377", 8) = 8 [pid 24894] write(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25", 23) = 23 [pid 24894] lseek(4, 0, SEEK_SET) = 0 [pid 24894] write(4, "RIFF_\0\0\0", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\20\261\2\0\20\261\2\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\27\0\0\0", 8) = 8 [pid 24894] fsync(4) = 0 [pid 24894] close(4) = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDONLY|O_NONBLOCK) = 4 [pid 24894] read(4, "RIFF", 4) = 4 [pid 24894] read(4, "_\0\0\0", 4) = 4 [pid 24894] read(4, "WAVE", 4) = 4 [pid 24894] read(4, "fmt \20\0\0\0", 8) = 8 [pid 24894] read(4, "\1\0\1\0\20\261\2\0\20\261\2\0\1\0\10\0", 16) = 16 [pid 24894] read(4, "data\27\0\0\0", 8) = 8 [pid 24894] read(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25", 23) = 23 [pid 24894] close(4) = 0 [pid 24894] unlink("hoge") = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDWR|O_CREAT|O_TRUNC|O_NONBLOCK, 0644) = 4 [pid 24894] write(4, "RIFF\377\377\377\377", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\0\356\2\0\0\356\2\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\333\377\377\377", 8) = 8 [pid 24894] write(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25", 23) = 23 [pid 24894] lseek(4, 0, SEEK_SET) = 0 [pid 24894] write(4, "RIFF_\0\0\0", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\0\356\2\0\0\356\2\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\27\0\0\0", 8) = 8 [pid 24894] fsync(4) = 0 [pid 24894] close(4) = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDONLY|O_NONBLOCK) = 4 [pid 24894] read(4, "RIFF", 4) = 4 [pid 24894] read(4, "_\0\0\0", 4) = 4 [pid 24894] read(4, "WAVE", 4) = 4 [pid 24894] read(4, "fmt \20\0\0\0", 8) = 8 [pid 24894] read(4, "\1\0\1\0\0\356\2\0\0\356\2\0\1\0\10\0", 16) = 16 [pid 24894] read(4, "data\27\0\0\0", 8) = 8 [pid 24894] read(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25", 23) = 23 [pid 24894] close(4) = 0 [pid 24894] unlink("hoge") = 0 [pid 24894] unlink("hoge") = -1 ENOENT (No such file or directory) [pid 24894] openat(AT_FDCWD, "hoge", O_RDWR|O_CREAT|O_TRUNC|O_NONBLOCK, 0644) = 4 [pid 24894] write(4, "RIFF\377\377\377\377", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0D\254\0\0D\254\0\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\333\377\377\377", 8) = 8 [pid 24894] write(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25\246\334W\337\222RK\262\200"..., 1047) = 1047 [pid 24894] lseek(4, 0, SEEK_SET) = 0 [pid 24894] write(4, "RIFF_\4\0\0", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0D\254\0\0D\254\0\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\27\4\0\0", 8) = 8 [pid 24894] fsync(4) = 0 [pid 24894] close(4) = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDONLY|O_NONBLOCK) = 4 [pid 24894] read(4, "RIFF", 4) = 4 [pid 24894] read(4, "_\4\0\0", 4) = 4 [pid 24894] read(4, "WAVE", 4) = 4 [pid 24894] read(4, "fmt \20\0\0\0", 8) = 8 [pid 24894] read(4, "\1\0\1\0D\254\0\0D\254\0\0\1\0\10\0", 16) = 16 [pid 24894] read(4, "data\27\4\0\0", 8) = 8 [pid 24894] read(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25\246\334W\337\222RK\262\200"..., 1047) = 1047 [pid 24894] close(4) = 0 [pid 24894] unlink("hoge") = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDWR|O_CREAT|O_TRUNC|O_NONBLOCK, 0644) = 4 [pid 24894] write(4, "RIFF\377\377\377\377", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\200\273\0\0\200\273\0\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\333\377\377\377", 8) = 8 [pid 24894] write(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25\246\334W\337\222RK\262\200"..., 1047) = 1047 [pid 24894] lseek(4, 0, SEEK_SET) = 0 [pid 24894] write(4, "RIFF_\4\0\0", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\200\273\0\0\200\273\0\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\27\4\0\0", 8) = 8 [pid 24894] fsync(4) = 0 [pid 24894] close(4) = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDONLY|O_NONBLOCK) = 4 [pid 24894] read(4, "RIFF", 4) = 4 [pid 24894] read(4, "_\4\0\0", 4) = 4 [pid 24894] read(4, "WAVE", 4) = 4 [pid 24894] read(4, "fmt \20\0\0\0", 8) = 8 [pid 24894] read(4, "\1\0\1\0\200\273\0\0\200\273\0\0\1\0\10\0", 16) = 16 [pid 24894] read(4, "data\27\4\0\0", 8) = 8 [pid 24894] read(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25\246\334W\337\222RK\262\200"..., 1047) = 1047 [pid 24894] close(4) = 0 [pid 24894] unlink("hoge") = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDWR|O_CREAT|O_TRUNC|O_NONBLOCK, 0644) = 4 [pid 24894] write(4, "RIFF\377\377\377\377", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\210X\1\0\210X\1\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\333\377\377\377", 8) = 8 [pid 24894] write(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25\246\334W\337\222RK\262\200"..., 1047) = 1047 [pid 24894] lseek(4, 0, SEEK_SET) = 0 [pid 24894] write(4, "RIFF_\4\0\0", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\210X\1\0\210X\1\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\27\4\0\0", 8) = 8 [pid 24894] fsync(4) = 0 [pid 24894] close(4) = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDONLY|O_NONBLOCK) = 4 [pid 24894] read(4, "RIFF", 4) = 4 [pid 24894] read(4, "_\4\0\0", 4) = 4 [pid 24894] read(4, "WAVE", 4) = 4 [pid 24894] read(4, "fmt \20\0\0\0", 8) = 8 [pid 24894] read(4, "\1\0\1\0\210X\1\0\210X\1\0\1\0\10\0", 16) = 16 [pid 24894] read(4, "data\27\4\0\0", 8) = 8 [pid 24894] read(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25\246\334W\337\222RK\262\200"..., 1047) = 1047 [pid 24894] close(4) = 0 [pid 24894] unlink("hoge") = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDWR|O_CREAT|O_TRUNC|O_NONBLOCK, 0644) = 4 [pid 24894] write(4, "RIFF\377\377\377\377", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\0w\1\0\0w\1\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\333\377\377\377", 8) = 8 [pid 24894] write(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25\246\334W\337\222RK\262\200"..., 1047) = 1047 [pid 24894] lseek(4, 0, SEEK_SET) = 0 [pid 24894] write(4, "RIFF_\4\0\0", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\0w\1\0\0w\1\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\27\4\0\0", 8) = 8 [pid 24894] fsync(4) = 0 [pid 24894] close(4) = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDONLY|O_NONBLOCK) = 4 [pid 24894] read(4, "RIFF", 4) = 4 [pid 24894] read(4, "_\4\0\0", 4) = 4 [pid 24894] read(4, "WAVE", 4) = 4 [pid 24894] read(4, "fmt \20\0\0\0", 8) = 8 [pid 24894] read(4, "\1\0\1\0\0w\1\0\0w\1\0\1\0\10\0", 16) = 16 [pid 24894] read(4, "data\27\4\0\0", 8) = 8 [pid 24894] read(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25\246\334W\337\222RK\262\200"..., 1047) = 1047 [pid 24894] close(4) = 0 [pid 24894] unlink("hoge") = 0 [pid 24894] openat(AT_FDCWD, "hoge", O_RDWR|O_CREAT|O_TRUNC|O_NONBLOCK, 0644) = 4 [pid 24894] write(4, "RIFF\377\377\377\377", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\20\261\2\0\20\261\2\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\333\377\377\377", 8) = 8 [pid 24894] write(4, "\346\317xj\341.\354\2\213C9\370\240\320\216\3340=\366\27#\257\25\246\334W\337\222RK\262\200"..., 1047) = 1047 [pid 24894] lseek(4, 0, SEEK_SET) = 0 [pid 24894] write(4, "RIFF_\4\0\0", 8) = 8 [pid 24894] write(4, "WAVE", 4) = 4 [pid 24894] write(4, "fmt \20\0\0\0\1\0\1\0\20\261\2\0\20\261\2\0\1\0\10\0", 24) = 24 [pid 24894] write(4, "data\27\4\0\0", 8) = 8 [pid 24894] fsync(4) = 0 [pid 24894] close(4) = 0 </pre> Issue URL : https://github.com/alsa-project/alsa-utils/issues/19 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] Fix for Lenovo B50-70 inverted internal microphone bug
by Michał Wadowski 11 May '19

11 May '19

I prepared patch for this bug: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1524215 I tested this patch on Linux 5.1.0 and it fixes problem on my Lenovo B50-70 laptop.

1 1

[alsa-devel] [PATCH] ASoC: SOF: Fix build error with CONFIG_SND_SOC_SOF_NOCODEC=m
by YueHaibing 10 May '19

10 May '19

Fix gcc build error while CONFIG_SND_SOC_SOF_NOCODEC=m sound/soc/sof/core.o: In function `snd_sof_device_probe': core.c:(.text+0x4af): undefined reference to `sof_nocodec_setup' Change SND_SOC_SOF_NOCODEC to bool to fix this. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: c16211d6226d ("ASoC: SOF: Add Sound Open Firmware driver core") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index b204c65..9c280c9 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -44,7 +44,7 @@ config SND_SOC_SOF_OPTIONS if SND_SOC_SOF_OPTIONS config SND_SOC_SOF_NOCODEC - tristate "SOF nocodec mode Support" + bool "SOF nocodec mode Support" help This adds support for a dummy/nocodec machine driver fallback option if no known codec is detected. This is typically only -- 2.7.4

3 15

[alsa-devel] [PATCH] Headset fixup for System76 Gazelle (gaze14)
by Jeremy Soller 10 May '19

10 May '19

A mistake was made in the identification of the four variants of the System76 Gazelle (gaze14). This patch corrects the PCI ID of the 17-inch, GTX 1660 Ti variant from 0x8560 to 0x8551. This patch also adds the correct fixups for the 15-inch and 17-inch GTX 1650 variants with PCI IDs 0x8560 and 0x8561. Tests were done on all four variants ensuring full audio capability. Signed-off-by: Jeremy Soller <jeremy(a)system76.com> --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b3712d87e98b..3e790b49c1c0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6901,7 +6901,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8550, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8551, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),

2 2

[alsa-devel] [PATCH] ALSA: hda: Fix race between creating and refreshing sysfs entries
by Amadeusz Sławiński 10 May '19

10 May '19

hda_widget_sysfs_reinit() can free underlying codec->widgets structure on which widget_tree_create() operates. Add locking to prevent such issues from happening. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=110382 Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- include/sound/hdaudio.h | 1 + sound/hda/hdac_sysfs.c | 22 +++++++++++++++++++--- sound/pci/hda/hda_codec.c | 2 ++ 3 files changed, 22 insertions(+), 3 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 45f944d57982..85835d0c33cc 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -81,6 +81,7 @@ struct hdac_device { atomic_t in_pm; /* suspend/resume being performed */ /* sysfs */ + struct mutex widget_lock; struct hdac_widget_tree *widgets; /* regmap */ diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index fb2aa344981e..5352e5db814c 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -399,21 +399,28 @@ int hda_widget_sysfs_init(struct hdac_device *codec) { int err; - if (codec->widgets) + mutex_lock(&codec->widget_lock); + if (codec->widgets) { + mutex_unlock(&codec->widget_lock); return 0; /* already created */ + } err = widget_tree_create(codec); if (err < 0) { widget_tree_free(codec); + mutex_unlock(&codec->widget_lock); return err; } + mutex_unlock(&codec->widget_lock); return 0; } void hda_widget_sysfs_exit(struct hdac_device *codec) { + mutex_lock(&codec->widget_lock); widget_tree_free(codec); + mutex_unlock(&codec->widget_lock); } int hda_widget_sysfs_reinit(struct hdac_device *codec, @@ -424,16 +431,23 @@ int hda_widget_sysfs_reinit(struct hdac_device *codec, hda_nid_t nid; int i; - if (!codec->widgets) + mutex_lock(&codec->widget_lock); + + if (!codec->widgets) { + mutex_unlock(&codec->widget_lock); return hda_widget_sysfs_init(codec); + } tree = kmemdup(codec->widgets, sizeof(*tree), GFP_KERNEL); - if (!tree) + if (!tree) { + mutex_unlock(&codec->widget_lock); return -ENOMEM; + } tree->nodes = kcalloc(num_nodes + 1, sizeof(*tree->nodes), GFP_KERNEL); if (!tree->nodes) { kfree(tree); + mutex_unlock(&codec->widget_lock); return -ENOMEM; } @@ -460,5 +474,7 @@ int hda_widget_sysfs_reinit(struct hdac_device *codec, codec->widgets = tree; kobject_uevent(tree->root, KOBJ_CHANGE); + + mutex_unlock(&codec->widget_lock); return 0; } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 701a69d856f5..a5746df5c94b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -870,6 +870,8 @@ static int snd_hda_codec_device_init(struct hda_bus *bus, struct snd_card *card, if (!codec) return -ENOMEM; + mutex_init(&codec->widget_lock); + sprintf(name, "hdaudioC%dD%d", card->number, codec_addr); err = snd_hdac_device_init(&codec->core, &bus->core, name, codec_addr); if (err < 0) { -- 2.17.1

2 1

Re: [alsa-devel] [PATCH 09/16] mmc: sdhci-xenon: use new match_string() helper/macro
by Ardelean, Alexandru 10 May '19

10 May '19

On Wed, 2019-05-08 at 16:26 +0300, Alexandru Ardelean wrote: > On Wed, 2019-05-08 at 15:20 +0300, Dan Carpenter wrote: > > > > > > On Wed, May 08, 2019 at 02:28:35PM +0300, Alexandru Ardelean wrote: > > > -static const char * const phy_types[] = { > > > - "emmc 5.0 phy", > > > - "emmc 5.1 phy" > > > -}; > > > - > > > enum xenon_phy_type_enum { > > > EMMC_5_0_PHY, > > > EMMC_5_1_PHY, > > > NR_PHY_TYPES > > > > There is no need for NR_PHY_TYPES now so you could remove that as well. > > > > I thought the same. > The only reason to keep NR_PHY_TYPES, is for potential future patches, > where it would be just 1 addition > > enum xenon_phy_type_enum { > EMMC_5_0_PHY, > EMMC_5_1_PHY, > + EMMC_5_2_PHY, > NR_PHY_TYPES > } > > Depending on style/preference of how to do enums (allow comma on last > enum > or not allow comma on last enum value), adding new enum values woudl be 2 > additions + 1 deletion lines. > > enum xenon_phy_type_enum { > EMMC_5_0_PHY, > - EMMC_5_1_PHY > + EMM > C_5_1_PHY, > + EMMC_5_2_PHY > } > > Either way (leave NR_PHY_TYPES or remove NR_PHY_TYPES) is fine from my > side. > Preference on this ? If no objection [nobody insists] I would keep. I don't feel strongly about it [dropping NR_PHY_TYPES or not]. Thanks Alex > Thanks > Alex > > > regards, > > dan carpenter > >

2 2

[alsa-devel] [PATCH v2 1/2] dt-bindings: sound: Convert Allwinner SPDIF binding to YAML
by Maxime Ripard 10 May '19

10 May '19

The Allwinner SoCs feature an SPDIF controller across multiple SoC generations. However, earlier generations were a bit simpler than the subsequent ones, and for example would always have RX and TX capabilities, and no reset lines. In order to express this, let's create two YAML schemas instead of the free form text we had before. Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- Changes from v1: - Merged the two schemas together and used the draft-7 conditionals --- Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml | 98 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++- Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt | 42 +------------------------------- 2 files changed, 98 insertions(+), 42 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml new file mode 100644 index 000000000000..5698e5de5e31 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 S/PDIF Controller Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens(a)csie.org> + - Liam Girdwood <lgirdwood(a)gmail.com> + - Mark Brown <broonie(a)kernel.org> + - Maxime Ripard <maxime.ripard(a)bootlin.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + oneOf: + - const: allwinner,sun4i-a10-spdif + - const: allwinner,sun6i-a31-spdif + - const: allwinner,sun8i-h3-spdif + - items: + - const: allwinner,sun8i-a83t-spdif + - const: allwinner,sun8i-h3-spdif + - items: + - const: allwinner,sun50i-a64-spdif + - const: allwinner,sun8i-h3-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: spdif + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + +allOf: + - if: + properties: + compatible: + contains: + enum: + - allwinner,sun6i-a31-spdif + - allwinner,sun8i-h3-spdif + + then: + required: + - resets + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + spdif: spdif@1c21000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun4i-a10-spdif"; + reg = <0x01c21000 0x40>; + interrupts = <13>; + clocks = <&apb0_gates 1>, <&spdif_clk>; + clock-names = "apb", "spdif"; + dmas = <&dma 0 2>, <&dma 0 2>; + dma-names = "rx", "tx"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt b/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt deleted file mode 100644 index 0c64a209c2e9..000000000000 --- a/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt +++ /dev/null @@ -1,42 +0,0 @@ -Allwinner Sony/Philips Digital Interface Format (S/PDIF) Controller - -The Allwinner S/PDIF audio block is a transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. -For now only playback is supported. - -Required properties: - - - compatible : should be one of the following: - - "allwinner,sun4i-a10-spdif": for the Allwinner A10 SoC - - "allwinner,sun6i-a31-spdif": for the Allwinner A31 SoC - - "allwinner,sun8i-h3-spdif": for the Allwinner H3 SoC - - - reg : Offset and length of the register set for the device. - - - interrupts : Contains the spdif interrupt. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - clocks : Contains an entry for each entry in clock-names. - - - clock-names : Includes the following entries: - "apb" clock for the spdif bus. - "spdif" clock for spdif controller. - - - resets : reset specifier for the ahb reset (A31 and newer only) - -Example: - -spdif: spdif@1c21000 { - compatible = "allwinner,sun4i-a10-spdif"; - reg = <0x01c21000 0x40>; - interrupts = <13>; - clocks = <&apb0_gates 1>, <&spdif_clk>; - clock-names = "apb", "spdif"; - dmas = <&dma 0 2>, <&dma 0 2>; - dma-names = "rx", "tx"; -}; base-commit: 00c3bc00f0cadbb48f2414370733892676511e79 -- git-series 0.9.1

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[alsa-devel] FW: move eapd coef function before ACT_PRE_PROBE state
by Kailang 10 May '19

10 May '19

Hi Takashi, This one. -----Original Message----- From: Kailang Sent: Monday, May 6, 2019 2:46 PM To: Takashi Iwai (tiwai(a)suse.de) <tiwai(a)suse.de> Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> Subject: RE: move eapd coef function before ACT_PRE_PROBE state Hi Takashi, Are you available for apply this ? Thanks. BR, Kailang > -----Original Message----- > From: Kailang > Sent: Tuesday, April 30, 2019 3:41 PM > To: Takashi Iwai (tiwai(a)suse.de) <tiwai(a)suse.de> > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > Subject: move eapd coef function before ACT_PRE_PROBE state > > Hi Takashi, > > alc_fill_eapd_coef(),this function was change EAPD control to default. > Default was set EAPD by verb control. > This function was run in ACT_INIT state. > Move it to ACT_PRE_PROBE above. It will have a chance to change EAPD > control on ACT_PRE_PROBE state. It could change control by 0x20 coef bit. > > BR, > Kailang

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[alsa-devel] ALC298 headphone noise
by Kailang 10 May '19

10 May '19

Hi Takashi, Dell platform with ALC298. system enter to runtime suspend. Headphone had noise. Let Headset Mic not shutup will solve this issue. BR, Kailang

2 1

[alsa-devel] Alsa-loopback
by Mats Lindestam 10 May '19

10 May '19

Hi, I am new to Alsa and I have been told that there is some kind of buffer/queue, in Alsa-loopback, that regulates when 'sound-buffers' is dropped. What I would like to achieve is make the buffer/queue larger (500 ms). >From what I have heard is that it should be possible to configure the buffer/queue, in Alsa-loopback, via the /etc/asound.conf configuration file. It seems that the parameters period_time, period_size, buffer_time, buffer_size & rate are important to achieve what I want. I haven't found any information on what these parameters do and how or if they are used in conjunction with each other. Here is my asound.conf file: # cat /etc/asound.conf pcm_slave.aloopcommon { pcm "hw:Loopback,0,0" ... rate 16000 period_time 64000 buffer_time 512000 } pcm.!audiosource0 { type dsnoop ... slave aloopcommon ... } Is there somewhere where these parameters are explained and how they interact? BR /Mats ?

1 0

[alsa-devel] [RFC PATCH 0/7] soundwire: add sysfs and debugfs support
by Pierre-Louis Bossart 09 May '19

09 May '19

This patchset is not fully-tested and is not a candidate for any merge. Since I am not very comfortable with sysfs and debugfs support, and I am not the initial author for this code, I could use feedback from smarter people than me. This series is intented to be applied on top of the 'soundwire: corrections to ACPI and DisCo properties' series Parts of this code was initially written by my Intel colleagues Vinod Koul, Sanyog Kale, Shreyas Nc and Hardik Shah, who are either no longer with Intel or no longer involved in SoundWire development. When relevant, I explictly added a note in commit messages to give them credit for their hard work, but I removed their signed-off-by tags to avoid email bounces and avoid spamming them forever with SoundWire patches. The sysfs parts essentially expose the values of MIPI DisCo properties. My contribution here was mainly to align with the specification, a number of properties from the Intel internal code were missing. I also split the code to make sure the same attribute names can be used at different levels, as described in the spec. One of the main questions I have is whether there is really a need to add new devices, or if the attributes can be added to the *existing* ones. For example, the sysfs hierarchy for the SoundWire 0 master shows as: /sys/bus/acpi/devices/PRP00001:00/int-sdw.0# ls sdw* 'sdw:0:25d:700:0:0': bank_delay_support hda_reg paging_support source_ports ch_prep_timeout high_PHY_capable power src-dp2 clk_stop_mode1 index_reg reset_behave src-dp4 clk_stop_timeout master_count simple_clk_stop_capable subsystem dp0 mipi_revision sink-dp1 test_mode_capable driver modalias sink-dp3 uevent firmware_node p15_behave sink_ports wake_capable 'sdw:0:25d:701:0:0': bank_delay_support high_PHY_capable paging_support source_ports ch_prep_timeout master_count power subsystem clk_stop_mode1 mipi_revision reset_behave test_mode_capable clk_stop_timeout modalias simple_clk_stop_capable uevent firmware_node p15_behave sink_ports wake_capable 'sdw-master:0': clk_stop_modes default_col dynamic_frame power uevent clock_frequencies default_frame_rate err_threshold revision clock_gears default_row max_clk_freq subsystem For the first two Slaves, this results in pretend-devices being added below each master, the actual Slave devices are children of the PRP00001 devices, so here we add a bit of complexity. Likewise, the 'sdw-master:0' is a pretend-device whose purpose is only to expose property values. Is this the recommended direction? Or should all the sysfs properties be added to the devices exposed by ACPI? The debugfs part is mainly to dump the Master and Slave registers, as well as the Intel-specific parts. One of the main changes from the previous code was to harden the code with scnprintf Feedback welcome. ~Pierre Pierre-Louis Bossart (7): soundwire: Add sysfs support for master(s) soundwire: add Slave sysfs support ABI: testing: Add description of soundwire master sysfs files ABI: testing: Add description of soundwire slave sysfs files soundwire: add debugfs support soundwire: cadence_master: add debugfs register dump soundwire: intel: add debugfs register dump .../ABI/testing/sysfs-bus-soundwire-master | 21 + .../ABI/testing/sysfs-bus-soundwire-slave | 85 ++++ drivers/soundwire/Makefile | 4 +- drivers/soundwire/bus.c | 13 + drivers/soundwire/bus.h | 30 ++ drivers/soundwire/bus_type.c | 8 + drivers/soundwire/cadence_master.c | 98 +++++ drivers/soundwire/cadence_master.h | 5 + drivers/soundwire/debugfs.c | 214 ++++++++++ drivers/soundwire/intel.c | 115 ++++++ drivers/soundwire/slave.c | 2 + drivers/soundwire/sysfs.c | 376 ++++++++++++++++++ drivers/soundwire/sysfs_local.h | 42 ++ drivers/soundwire/sysfs_slave_dp0.c | 112 ++++++ drivers/soundwire/sysfs_slave_dpn.c | 168 ++++++++ include/linux/soundwire/sdw.h | 24 ++ 16 files changed, 1316 insertions(+), 1 deletion(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-soundwire-master create mode 100644 Documentation/ABI/testing/sysfs-bus-soundwire-slave create mode 100644 drivers/soundwire/debugfs.c create mode 100644 drivers/soundwire/sysfs.c create mode 100644 drivers/soundwire/sysfs_local.h create mode 100644 drivers/soundwire/sysfs_slave_dp0.c create mode 100644 drivers/soundwire/sysfs_slave_dpn.c -- 2.17.1

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[alsa-devel] [PATCH v2 00/30] Update cros_ec_commands.h
by Gwendal Grignou 09 May '19

09 May '19

The interface between CrosEC embedded controller and the host, described by cros_ec_commands.h, as diverged from what the embedded controller really support. The source of thruth is at https://chromium.googlesource.com/chromiumos/platform/ec/+/master/include/e… That include file is converted to remove ACPI and Embedded only code. >From now on, cros_ec_commands.h will be automatically generated from the file above, do not modify directly. Fell free to squash the commits below. Changes in v2: - Move I2S changes at the end of the patchset, squashed with change in sound/soc/codecs/cros_ec_codec.c to match new interface. - Add Acked-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> Gwendal Grignou (30): mfd: cros_ec: Update license term mfd: cros_ec: Zero BUILD_ macro mfd: cros_ec: set comments properly mfd: cros_ec: add ec_align macros mfd: cros_ec: Define commands as 4-digit UPPER CASE hex values mfd: cros_ec: use BIT macro mfd: cros_ec: Update ACPI interface definition mfd: cros_ec: move HDMI CEC API definition mfd: cros_ec: Remove zero-size structs mfd: cros_ec: Add Flash V2 commands API mfd: cros_ec: Add PWM_SET_DUTY API mfd: cros_ec: Add lightbar v2 API mfd: cros_ec: Expand hash API mfd: cros_ec: Add EC transport protocol v4 mfd: cros_ec: Complete MEMS sensor API mfd: cros_ec: Fix event processing API mfd: cros_ec: Add fingerprint API mfd: cros_ec: Fix temperature API mfd: cros_ec: Complete Power and USB PD API mfd: cros_ec: Add API for keyboard testing mfd: cros_ec: Add Hibernate API mfd: cros_ec: Add Smart Battery Firmware update API mfd: cros_ec: Add I2C passthru protection API mfd: cros_ec: Add API for EC-EC communication mfd: cros_ec: Add API for Touchpad support mfd: cros_ec: Add API for Fingerprint support mfd: cros_ec: Add API for rwsig mfd: cros_ec: Add SKU ID and Secure storage API mfd: cros_ec: Add Management API entry points mfd: cros_ec: Update I2S API include/linux/mfd/cros_ec_commands.h | 3735 +++++++++++++++++++++----- sound/soc/codecs/cros_ec_codec.c | 8 +- 2 files changed, 2997 insertions(+), 746 deletions(-) -- 2.21.0.1020.gf2820cf01a-goog

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[alsa-devel] [GIT PULL] ASoC fix for v5.2
by Mark Brown 09 May '19

09 May '19

The following changes since commit ce38a75089f70f6380fc63a5478a7659b4eb3f47: ASoC: SOF: core: fix undefined nocodec reference (2019-05-08 15:42:00 +0900) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.2-4 for you to fetch changes up to 2a79e3d342e4b8883ff7fe2cf1b3e9192c26f61a: ASoC: SOF: Fix unused variable warnings (2019-05-08 18:33:34 +0900) ---------------------------------------------------------------- ASoC: ...and one more This didn't get applied as quickly as expected due to a missing list CC and I didn't notice my scripts delaying it. ---------------------------------------------------------------- Emil Renner Berthing (1): spi: rockchip: turn down tx dma bursts Libin Yang (1): ASoC: codec: hdac_hdmi: no checking monitor in hw_params Olivier Moysan (1): ASoC: stm32: i2s: update pcm hardware constraints Takashi Iwai (1): ASoC: SOF: Fix unused variable warnings Tzung-Bi Shih (3): ASoC: mediatek: mt6358: initialize setting when ramping volume ASoC: mediatek: mt6358: save output volume for mixer controls ASoC: mediatek: mt6358: save PGA for mixer control drivers/spi/spi-rockchip.c | 4 +- sound/soc/codecs/hdac_hdmi.c | 13 ----- sound/soc/codecs/mt6358.c | 131 +++++++++++++++++++++++++++---------------- sound/soc/sof/core.c | 2 + sound/soc/stm/stm32_i2s.c | 3 +- 5 files changed, 89 insertions(+), 64 deletions(-)

2 7

[alsa-devel] [PATCH] ASoC: max98357a: request GPIO when device get probed
by Tzung-Bi Shih 09 May '19

09 May '19

devm_gpiod_get_optional() returns EBUSY after component rebound. Request GPIO in max98357a_platform_probe() to support component rebinding. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- v1: ASoC: max98357a: manage GPIO for component rebinding => don't move from devm to non-devm v2: ASoC: max98357a: release GPIO when component removing => only put things that really need the card in component's probe() => don't use devm_gpiod_put() if possible sound/soc/codecs/max98357a.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index d037a3e4d323..0a14f9dacbee 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -61,14 +61,6 @@ static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { static int max98357a_component_probe(struct snd_soc_component *component) { - struct gpio_desc *sdmode; - - sdmode = devm_gpiod_get_optional(component->dev, "sdmode", GPIOD_OUT_LOW); - if (IS_ERR(sdmode)) - return PTR_ERR(sdmode); - - snd_soc_component_set_drvdata(component, sdmode); - return 0; } @@ -112,6 +104,15 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { static int max98357a_platform_probe(struct platform_device *pdev) { + struct gpio_desc *sdmode; + + sdmode = devm_gpiod_get_optional(&pdev->dev, + "sdmode", GPIOD_OUT_LOW); + if (IS_ERR(sdmode)) + return PTR_ERR(sdmode); + + dev_set_drvdata(&pdev->dev, sdmode); + return devm_snd_soc_register_component(&pdev->dev, &max98357a_component_driver, &max98357a_dai_driver, 1); -- 2.21.0.1020.gf2820cf01a-goog

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: Intel: Add Cometlake PCI IDs
by Evan Green 09 May '19

09 May '19

This small series adds PCI IDs for Cometlake platforms, for a dazzling audio experience. Changes in v2: - Add CML-H ID 0x06c8 (Pierre-Louis) - Add 0x06c8 for CML-H (Pierre-Louis) Evan Green (2): ASoC: SOF: Add Comet Lake PCI IDs ASoC: Intel: Skylake: Add Cometlake PCI IDs sound/soc/intel/Kconfig | 18 +++++++++++++++ sound/soc/intel/skylake/skl-messages.c | 16 +++++++++++++ sound/soc/intel/skylake/skl.c | 10 ++++++++ sound/soc/sof/intel/Kconfig | 32 ++++++++++++++++++++++++++ sound/soc/sof/sof-pci-dev.c | 8 +++++++ 5 files changed, 84 insertions(+) -- 2.20.1

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[alsa-devel] [PATCH 0/8] soundwire: corrections to ACPI and DisCo properties
by Pierre-Louis Bossart 08 May '19

08 May '19

Now that we are done with cleanups, we can start fixing the code with actual semantic or functional changes. This patchset applies on top of everything Vinod and I contributed this week. The fist patch correct a simplifying assumption made earlier. With the IceLake BIOS the existing code is fooled by the presence of a second child device and the namespace walk needs to be filtered. This was not needed on previous generations. The second patch fixes a long-standing misalignment between code and DisCo specification, preventing MIPI DisCo properties from being parsed successfully. The third and fourth patch remove restrictions preventing codec drivers from reading DisCo properties. The fifth patch adds definitions from the SoundWire spec that were missed somehow, but will be very much needed for dynamic bandwidth allocation. The last 3 patches realign the code with the MIPI specification. The existing code exposes properties that don't exist, or exposes them with ambiguous wording. Sticking to the specification helps avoid interpretation issues for integrators, and will make sure the follow-up sysfs support is self-explanatory. Parts of this code was initially written by my Intel colleagues Vinod Koul, Sanyog Kale, Shreyas Nc and Hardik Shah, who are either no longer with Intel or no longer involved in SoundWire development. When relevant, I explictly added a note in commit messages to give them credit for their hard work, but I removed their signed-off-by tags to avoid email bounces and avoid spamming them forever with SoundWire patches. Pierre-Louis Bossart (8): soundwire: intel: filter SoundWire controller device search soundwire: mipi_disco: fix master/link error soundwire: mipi_disco: expose sdw_slave_read_dpn as symbol soundwire: mipi_disco: expose sdw_slave_read_dp0 as symbol soundwire: add port-related definitions soundwire: remove master data port properties soundwire: fix master properties soundwire: rename/clarify MIPI DisCo properties drivers/soundwire/bus.c | 6 +-- drivers/soundwire/intel.c | 11 ++-- drivers/soundwire/intel_init.c | 19 ++++++- drivers/soundwire/mipi_disco.c | 49 +++++++++--------- drivers/soundwire/stream.c | 8 +-- include/linux/soundwire/sdw.h | 94 +++++++++++++++++++++++++++------- 6 files changed, 132 insertions(+), 55 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH 0/2] ASoC: SOF: fix undefined references
by Pierre-Louis Bossart 08 May '19

08 May '19

kbuild reported a couple of valid issues with undefined references. It's a bit embarrassing that they were not found earlier since the SOF tree is tracked for each PR... Pierre-Louis Bossart (2): ASoC: SOF: xtensa: fix undefined references ASoC: SOF: core: fix undefined nocodec reference sound/soc/sof/core.c | 10 +++++----- sound/soc/sof/intel/Kconfig | 2 +- 2 files changed, 6 insertions(+), 6 deletions(-) -- 2.17.1

3 7

[alsa-devel] [PATCH 0/4] ALSA: line6: Cleanup startup sequences
by Takashi Iwai 08 May '19

08 May '19

Hi, while debugging line6 toneport driver bug, I noticed that other line6 drivers have too complex and fragile startup sequences. This is the result of the rewrites with a single delayed work. Maybe for 5.3, as these are no critical fixes. Takashi === Takashi Iwai (4): ALSA: line6: pod: Rewrite complex timer & work combo with a delayed work ALSA: line6: podhd: Rewrite complex timer & work combo with a delayed work ALSA: line6: variax: Rewrite complex timer & work combo with a delayed work ALSA: line6: Drop superfluous timer helper function sound/usb/line6/driver.c | 11 ----- sound/usb/line6/driver.h | 9 ---- sound/usb/line6/pod.c | 85 ++++++++++++-------------------- sound/usb/line6/podhd.c | 73 +++++---------------------- sound/usb/line6/variax.c | 125 ++++++++++++++--------------------------------- 5 files changed, 78 insertions(+), 225 deletions(-) -- 2.16.4

1 4

[alsa-devel] [PATCH] ALSA: line6: toneport: Fix broken usage of timer for delayed execution
by Takashi Iwai 08 May '19

08 May '19

The line6 toneport driver has code for some delayed initialization, and this hits the kernel Oops because mutex and other sleepable functions are used in the timer callback. Fix the abuse by a delayed work instead so that everything works gracefully. Reported-by: syzbot+a07d0142e74fdd595cfb(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/line6/toneport.c | 16 +++++++++------- 1 file changed, 9 insertions(+), 7 deletions(-) diff --git a/sound/usb/line6/toneport.c b/sound/usb/line6/toneport.c index 19bee725de00..325b07b98b3c 100644 --- a/sound/usb/line6/toneport.c +++ b/sound/usb/line6/toneport.c @@ -54,8 +54,8 @@ struct usb_line6_toneport { /* Firmware version (x 100) */ u8 firmware_version; - /* Timer for delayed PCM startup */ - struct timer_list timer; + /* Work for delayed PCM startup */ + struct delayed_work pcm_work; /* Device type */ enum line6_device_type type; @@ -241,9 +241,10 @@ static int snd_toneport_source_put(struct snd_kcontrol *kcontrol, return 1; } -static void toneport_start_pcm(struct timer_list *t) +static void toneport_start_pcm(struct work_struct *work) { - struct usb_line6_toneport *toneport = from_timer(toneport, t, timer); + struct usb_line6_toneport *toneport = + container_of(work, struct usb_line6_toneport, pcm_work.work); struct usb_line6 *line6 = &toneport->line6; line6_pcm_acquire(line6->line6pcm, LINE6_STREAM_MONITOR, true); @@ -393,7 +394,8 @@ static int toneport_setup(struct usb_line6_toneport *toneport) if (toneport_has_led(toneport)) toneport_update_led(toneport); - mod_timer(&toneport->timer, jiffies + TONEPORT_PCM_DELAY * HZ); + schedule_delayed_work(&toneport->pcm_work, + msecs_to_jiffies(TONEPORT_PCM_DELAY * 1000)); return 0; } @@ -405,7 +407,7 @@ static void line6_toneport_disconnect(struct usb_line6 *line6) struct usb_line6_toneport *toneport = (struct usb_line6_toneport *)line6; - del_timer_sync(&toneport->timer); + cancel_delayed_work_sync(&toneport->pcm_work); if (toneport_has_led(toneport)) toneport_remove_leds(toneport); @@ -422,7 +424,7 @@ static int toneport_init(struct usb_line6 *line6, struct usb_line6_toneport *toneport = (struct usb_line6_toneport *) line6; toneport->type = id->driver_info; - timer_setup(&toneport->timer, toneport_start_pcm, 0); + INIT_DELAYED_WORK(&toneport->pcm_work, toneport_start_pcm); line6->disconnect = line6_toneport_disconnect; -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: aica: Fix a long-time build breakage
by Takashi Iwai 08 May '19

08 May '19

The build of aica sound driver has been broken since the timer API conversion and some code rewrite. This patch fixes the breakage by using the common substream field, as well as a bit cleaning up wrt the timer handling in the code. Fixes: d522bb6a105f ("ALSA: sh: aica: Convert timers to use timer_setup()") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/sh/aica.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index e7fef3fce44a..a24e486d9d83 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -303,7 +303,7 @@ static void aica_period_elapsed(struct timer_list *t) { struct snd_card_aica *dreamcastcard = from_timer(dreamcastcard, t, timer); - struct snd_pcm_substream *substream = dreamcastcard->timer_substream; + struct snd_pcm_substream *substream = dreamcastcard->substream; /*timer function - so cannot sleep */ int play_period; struct snd_pcm_runtime *runtime; @@ -335,13 +335,6 @@ static void spu_begin_dma(struct snd_pcm_substream *substream) dreamcastcard = substream->pcm->private_data; /*get the queue to do the work */ schedule_work(&(dreamcastcard->spu_dma_work)); - /* Timer may already be running */ - if (unlikely(dreamcastcard->timer_substream)) { - mod_timer(&dreamcastcard->timer, jiffies + 4); - return; - } - timer_setup(&dreamcastcard->timer, aica_period_elapsed, 0); - dreamcastcard->timer_substream = substream; mod_timer(&dreamcastcard->timer, jiffies + 4); } @@ -379,8 +372,8 @@ static int snd_aicapcm_pcm_close(struct snd_pcm_substream { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; flush_work(&(dreamcastcard->spu_dma_work)); - if (dreamcastcard->timer_substream) - del_timer(&dreamcastcard->timer); + del_timer(&dreamcastcard->timer); + dreamcastcard->substream = NULL; kfree(dreamcastcard->channel); spu_disable(); return 0; @@ -613,6 +606,7 @@ static int snd_aica_probe(struct platform_device *devptr) "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); /* Prepare to use the queue */ INIT_WORK(&(dreamcastcard->spu_dma_work), run_spu_dma); + timer_setup(&dreamcastcard->timer, aica_period_elapsed, 0); /* Load the PCM 'chip' */ err = snd_aicapcmchip(dreamcastcard, 0); if (unlikely(err < 0)) -- 2.16.4

1 0

Re: [alsa-devel] [PATCH 01/16] lib: fix match_string() helper on -1 array size
by Ardelean, Alexandru 08 May '19

08 May '19

On Wed, 2019-05-08 at 14:28 +0300, Alexandru Ardelean wrote: > The documentation the `_match_string()` helper mentions that `n` > should be: > * @n: number of strings in the array or -1 for NULL terminated arrays > > The behavior of the function is different, in the sense that it exits on > the first NULL element in the array, regardless of whether `n` is -1 or a > positive number. > > This patch changes the behavior, to exit the loop when a NULL element is > found and n == -1. Essentially, this aligns the behavior with the > doc-string. > > There are currently many users of `match_string()`, and so, in order to > go > through them, the next patches in the series will focus on doing some > cosmetic changes, which are aimed at grouping the users of > `match_string()`. > This is the duplicate & should be dropped. Sorry for this. > Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> > --- > lib/string.c | 5 ++++- > 1 file changed, 4 insertions(+), 1 deletion(-) > > diff --git a/lib/string.c b/lib/string.c > index 3ab861c1a857..76edb7bf76cb 100644 > --- a/lib/string.c > +++ b/lib/string.c > @@ -648,8 +648,11 @@ int match_string(const char * const *array, size_t > n, const char *string) > > for (index = 0; index < n; index++) { > item = array[index]; > - if (!item) > + if (!item) { > + if (n != (size_t)-1) > + continue; > break; > + } > if (!strcmp(item, string)) > return index; > }

1 0

[alsa-devel] Fwd: [PATCH] ASoC: SOF: Fix unused variable warnings
by Takashi Iwai 08 May '19

08 May '19

Oops, forgot to add to Cc to alsa-devel ML. Forwarding now. Takashi

2 4

Re: [alsa-devel] [PATCH 1/2] ASoC: Add support for Conexant CX2072X CODEC
by Takashi Iwai 08 May '19

08 May '19

On Mon, 06 May 2019 06:26:25 +0200, Mark Brown wrote: > > On Fri, May 03, 2019 at 09:18:26AM +0200, Takashi Iwai wrote: > > On Fri, 03 May 2019 08:47:29 +0200, > > Mark Brown wrote: > > > On Thu, May 02, 2019 at 09:04:06AM +0200, Takashi Iwai wrote: > > > > Mark Brown wrote: > > > > > > This looks *very* much like board configuration rather than a patch - > > > > > there's no kind of test bit and the comments talk specifically about > > > > > OK, will replace with the straight regmap_multi_reg_write(). > > > > That's probably not addressing the issue, a lot of that stuff just > > > doesn't seem like it should be in some fixed configuration table at all. > > > So what's your alternative suggestion? > > Make it either platform data or runtime controls depending on the > specific thing being configured. Though... > > > > Oh, joy. What's the story here? Do you have a datasheet for the part? > > > Not at all. I just refreshed the already submitted patches since I > > have a laptop with the codec. I tried to contact Conexant, but in > > vain, so I decided to submit the renewed one. > > ...this is not going to help. At the very least it needs to be > abundantly clear to anyone reading the code tht any magic register write > sequences like this are being done the way they are due to a badly > documented part from an unsupportive vendor rather than because it's a > good idea in case someone decides to use the driver as an example. Some > comments or something. For safety the bits that look board specific > like the pinmuxing should ideally be behind DMI checks, though if nobody > else ever bought the CODEC that's probably not so important. Fair enough. I'm going to put more comments on that part. Since this is the only code for this codec on the net and all existing platforms (Intel devices, RPi extensions and else) seem working with it -- i.e. all these depend on the same init sequence more or less, so splitting to platform data won't help much in practice. > > > > The jack detection in ASoC is anyway a bit funky, especially when > > > > involved with PM... > > > > What do you mean here? I'm not aware of any issues and the systems I've > > > worked with seemed robust... > > > There are tons of different ways of implementation for jack controls, > > with different API usages. IOW, no consistency. > > The board registration interfaces do vary as the hardware isn't very > standard so there's a bunch of things you can do for configuration > (though there is a generic interface for the 90% case, the main > limitation these days on using it is Intel's lack of any sensible > firmware interface). I'm not aware of any issues with power management > though? Yeah, I understand the various use cases and coverage for a wide range of hardware implementations, but the lack of the documentation is... not a thing we can be proud of. Regarding the PM, my statement wasn't straight, sorry. The fact is that ASoC manages so many different levels of (a sort of) "PM". We have the Linux device-object system and runtime PMs, and we have BIAS level control. The machine needed the jack handling via ACPI GPIO IRQ, that is outside the codec, which looks rather like a rare case, and I had hard time to look for the right choice of the API usage. > > > > > > +EXPORT_SYMBOL_GPL(snd_soc_cx2072x_get_jack_state); > > > > > > Why is this symbol exported? > > > > > It's called from the machine driver. > > > > snd_soc_jack_add_gpios() is called in the machine driver side, and it > > > > needs the jack_status_check callback that calls this function. > > > > That code shouldn't be in the machine driver, the CODEC driver should > > > request any interrupts it needs itself. > > > The similar things are done on many other Intel SST board drivers. > > The current patch just follows the pattern. > > The only example I can find is the max98095 which it just plain buggy in > this regard, that example shouldn't be followed - it looks like it's > just plain an oversight in review. Frankly the export was done by me > and looks like I was just rushing to fix up a build breakage. This is rather a question how generic the codec driver should be written. I changed the code in v5 patchset to embed the jack_gpio stuff inside the codec driver side rather than the machine driver, so the whole exported functions can be reduced now. But, of course, it slightly gives more implicit assumption about the hardware implementations, too. Though, the existing code seems to have already fixed gpio / pin setups, so the other setups wouldn't have worked, in anyway. > > > No, that's not the question. The question is why there is any > > > initialization at all? > > > Again, no idea. These are likely no default values of the hardware > > registers, and we need to set up some. I *guess* ditto for the > > initial register table in the above, too. > > Or it's just a taste setting someone had for their particular system. > With the earlier register default table an awful lot of it is fairly > obvious board specific configuration, doing things like pinmuxing which > will vary with the board. > > It's unlikely there are *no* default values - you'd kind of have to try > to have a specific register range like this with genuinely floating > values. Given that the code for configuring the EQ was broken to IIRC > never take effect I'd not be 100% surprisd if someone couldn't figure > out why their settings weren't taking effect and just bodged something > directly in the driver. Actually I'm fine to drop the whole EQ stuff that brings lots of black magic. Certainly it'll benefit us for code simplification. Let's see. thanks, Takashi

2 7

[alsa-devel] [PATCH] ASoC: max98357a: release GPIO when component removing
by Tzung-Bi Shih 08 May '19

08 May '19

Component's probe() gets GPIO for "sdmode-gpios" via devm_gpiod_get_optional(). The GPIO binds to the device. When the component get removed but the device does not, the GPIO still binds to the device. Then, when the component be probed again, devm_gpiod_get_optional() returns EBUSY. Release the GPIO proactively when component is removing. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/codecs/max98357a.c | 42 ++++++++++++++++++++++++++++-------- 1 file changed, 33 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index d037a3e4d323..b11c6cb81f55 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -27,24 +27,28 @@ #include <sound/soc-dai.h> #include <sound/soc-dapm.h> +struct max98357a_priv { + struct gpio_desc *sdmode; +}; + static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai); + struct max98357a_priv *max98357a = snd_soc_dai_get_drvdata(dai); - if (!sdmode) + if (!max98357a->sdmode) return 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - gpiod_set_value(sdmode, 1); + gpiod_set_value(max98357a->sdmode, 1); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - gpiod_set_value(sdmode, 0); + gpiod_set_value(max98357a->sdmode, 0); break; } @@ -61,19 +65,31 @@ static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { static int max98357a_component_probe(struct snd_soc_component *component) { - struct gpio_desc *sdmode; + struct max98357a_priv *max98357a = + snd_soc_component_get_drvdata(component); - sdmode = devm_gpiod_get_optional(component->dev, "sdmode", GPIOD_OUT_LOW); - if (IS_ERR(sdmode)) - return PTR_ERR(sdmode); + max98357a->sdmode = devm_gpiod_get_optional(component->dev, + "sdmode", GPIOD_OUT_LOW); + if (IS_ERR(max98357a->sdmode)) + return PTR_ERR(max98357a->sdmode); - snd_soc_component_set_drvdata(component, sdmode); + snd_soc_component_set_drvdata(component, max98357a); return 0; } +static void max98357a_component_remove(struct snd_soc_component *component) +{ + struct max98357a_priv *max98357a = + snd_soc_component_get_drvdata(component); + + if (max98357a->sdmode) + devm_gpiod_put(component->dev, max98357a->sdmode); +} + static const struct snd_soc_component_driver max98357a_component_driver = { .probe = max98357a_component_probe, + .remove = max98357a_component_remove, .dapm_widgets = max98357a_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98357a_dapm_widgets), .dapm_routes = max98357a_dapm_routes, @@ -112,6 +128,14 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { static int max98357a_platform_probe(struct platform_device *pdev) { + struct max98357a_priv *max98357a; + + max98357a = devm_kzalloc(&pdev->dev, sizeof(*max98357a), GFP_KERNEL); + if (!max98357a) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, max98357a); + return devm_snd_soc_register_component(&pdev->dev, &max98357a_component_driver, &max98357a_dai_driver, 1); -- 2.21.0.1020.gf2820cf01a-goog

2 3

[alsa-devel] Applied "ASoC: SOF: Fix unused variable warnings" to the asoc tree
by Mark Brown 08 May '19

08 May '19

The patch ASoC: SOF: Fix unused variable warnings 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 2a79e3d342e4b8883ff7fe2cf1b3e9192c26f61a Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Wed, 8 May 2019 10:50:37 +0200 Subject: [PATCH] ASoC: SOF: Fix unused variable warnings MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The recent fix for the build fix caused a couple of unused variable compiler warnings when CONFIG_SND_SOC_SOF_NOCODEC isn't set: sound/soc/sof/core.c:263:6: warning: unused variable ‘ret’ [-Wunused-variable] sound/soc/sof/core.c:262:28: warning: unused variable ‘machine’ [-Wunused-variable] Fix them by adding another ifdef. Fixes: ce38a75089f7 ("ASoC: SOF: core: fix undefined nocodec reference") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 5ddbfa8f1a28..32105e0fabe8 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -259,8 +259,10 @@ int snd_sof_create_page_table(struct snd_sof_dev *sdev, static int sof_machine_check(struct snd_sof_dev *sdev) { struct snd_sof_pdata *plat_data = sdev->pdata; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC) struct snd_soc_acpi_mach *machine; int ret; +#endif if (plat_data->machine) return 0; -- 2.20.1

1 0

[alsa-devel] [GIT PULL] ASoC fixes for v5.2
by Mark Brown 08 May '19

08 May '19

The following changes since commit ce38a75089f70f6380fc63a5478a7659b4eb3f47: ASoC: SOF: core: fix undefined nocodec reference (2019-05-08 15:42:00 +0900) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.2-3 for you to fetch changes up to ecc425f06c19a296622c827cde54735ebc5320b8: spi: rockchip: turn down tx dma bursts (2019-05-08 18:04:32 +0900) ---------------------------------------------------------------- ASoC: Even more fixes Yet more stuff that came in through the day. ---------------------------------------------------------------- Emil Renner Berthing (1): spi: rockchip: turn down tx dma bursts Libin Yang (1): ASoC: codec: hdac_hdmi: no checking monitor in hw_params Olivier Moysan (1): ASoC: stm32: i2s: update pcm hardware constraints Tzung-Bi Shih (3): ASoC: mediatek: mt6358: initialize setting when ramping volume ASoC: mediatek: mt6358: save output volume for mixer controls ASoC: mediatek: mt6358: save PGA for mixer control drivers/spi/spi-rockchip.c | 4 +- sound/soc/codecs/hdac_hdmi.c | 13 ----- sound/soc/codecs/mt6358.c | 131 +++++++++++++++++++++++++++---------------- sound/soc/stm/stm32_i2s.c | 3 +- 4 files changed, 87 insertions(+), 64 deletions(-)

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[alsa-devel] [PATCH 0/3] ASoC: mediatek: mt6358: fix mixer control issues
by Tzung-Bi Shih 08 May '19

08 May '19

This series fixes the following mixer control related issues: - "Headphone Volume" and "PGA Volume" have been reset after previous stream closed. - Set "Headphone Volume" to the smallest value (-10dB) still get mute, because ramp from -10dB to -10dB is a NOP in original implementation. "Lineout Volume" and "Handset Volume" are un-tested. Fixing them by applying similar logic with "Headphone Volume". Tzung-Bi Shih (3): ASoC: mediatek: mt6358: initialize setting when ramping volume ASoC: mediatek: mt6358: save output volume for mixer controls ASoC: mediatek: mt6358: save PGA for mixer control sound/soc/codecs/mt6358.c | 131 ++++++++++++++++++++++++-------------- 1 file changed, 83 insertions(+), 48 deletions(-) -- 2.21.0.1020.gf2820cf01a-goog

2 6

[alsa-devel] [PATCH 0/2] ASoC: stm32: i2s: add some features
by Olivier Moysan 08 May '19

08 May '19

Update and add some features in STM32 I2S driver. - update pcm hardware constraints - add management of identification registers Olivier Moysan (2): ASoC: stm32: i2s: update pcm hardware constraints ASoC: stm32: i2s: manage identification registers sound/soc/stm/stm32_i2s.c | 63 ++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 59 insertions(+), 4 deletions(-) -- 2.7.4

2 4

[alsa-devel] [PATCH] ASoC: codec: hdac_hdmi: no checking monitor in hw_params
by libin.yang＠intel.com 08 May '19

08 May '19

From: Libin Yang <libin.yang(a)intel.com> Some userspace apps, like pulseaudio, may call open, hw_params, prepare to judge whether the pcm is ready or not. Current hdac_hdmi will return -ENODEV if monitor is not connected, which will cause the apps believe the pcm is not ready. Actually PCM for hdmi is ready, even the monitor is not connected. This patch removes the check of monitor presence in hw_params, just like what the legacy HD-Audio driver does. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4de1fbf..660e058 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -455,24 +455,11 @@ static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hparams, struct snd_soc_dai *dai) { struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); - struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_dai_port_map *dai_map; - struct hdac_hdmi_port *port; struct hdac_hdmi_pcm *pcm; int format; dai_map = &hdmi->dai_map[dai->id]; - port = dai_map->port; - - if (!port) - return -ENODEV; - - if ((!port->eld.monitor_present) || (!port->eld.eld_valid)) { - dev_err(&hdev->dev, - "device is not configured for this pin:port%d:%d\n", - port->pin->nid, port->id); - return -ENODEV; - } format = snd_hdac_calc_stream_format(params_rate(hparams), params_channels(hparams), params_format(hparams), -- 2.7.4

4 3

[alsa-devel] [RESEND PATCH v3] ASoC: fsl_sai: Move clock operation to PM runtime
by Daniel Baluta 08 May '19

08 May '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> Turn off/on clocks when device enters suspend/resume. This can help saving power. As a further optimization, we turn off/on mclk only when SAI is in master mode because otherwise mclk is externally provided. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Viorel Suman <viorel.suman(a)nxp.com> --- - initially part of this 3 patch series https://patchwork.kernel.org/cover/10911519/ - first 2 patches were merged sound/soc/fsl/fsl_sai.c | 54 +++++++++++++++++++++++++++++++++-------- 1 file changed, 44 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 8593269156bd..d58cc3ae90d8 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -596,15 +596,8 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - struct device *dev = &sai->pdev->dev; int ret; - ret = clk_prepare_enable(sai->bus_clk); - if (ret) { - dev_err(dev, "failed to enable bus clock: %d\n", ret); - return ret; - } - regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE, FSL_SAI_CR3_TRCE); @@ -621,8 +614,6 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE, 0); - - clk_disable_unprepare(sai->bus_clk); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -935,6 +926,14 @@ static int fsl_sai_runtime_suspend(struct device *dev) { struct fsl_sai *sai = dev_get_drvdata(dev); + if (sai->mclk_streams & BIT(SNDRV_PCM_STREAM_CAPTURE)) + clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[0]]); + + if (sai->mclk_streams & BIT(SNDRV_PCM_STREAM_PLAYBACK)) + clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[1]]); + + clk_disable_unprepare(sai->bus_clk); + regcache_cache_only(sai->regmap, true); regcache_mark_dirty(sai->regmap); @@ -944,6 +943,25 @@ static int fsl_sai_runtime_suspend(struct device *dev) static int fsl_sai_runtime_resume(struct device *dev) { struct fsl_sai *sai = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(sai->bus_clk); + if (ret) { + dev_err(dev, "failed to enable bus clock: %d\n", ret); + return ret; + } + + if (sai->mclk_streams & BIT(SNDRV_PCM_STREAM_PLAYBACK)) { + ret = clk_prepare_enable(sai->mclk_clk[sai->mclk_id[1]]); + if (ret) + goto disable_bus_clk; + } + + if (sai->mclk_streams & BIT(SNDRV_PCM_STREAM_CAPTURE)) { + ret = clk_prepare_enable(sai->mclk_clk[sai->mclk_id[0]]); + if (ret) + goto disable_tx_clk; + } regcache_cache_only(sai->regmap, false); regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); @@ -951,7 +969,23 @@ static int fsl_sai_runtime_resume(struct device *dev) usleep_range(1000, 2000); regmap_write(sai->regmap, FSL_SAI_TCSR, 0); regmap_write(sai->regmap, FSL_SAI_RCSR, 0); - return regcache_sync(sai->regmap); + + ret = regcache_sync(sai->regmap); + if (ret) + goto disable_rx_clk; + + return 0; + +disable_rx_clk: + if (sai->mclk_streams & BIT(SNDRV_PCM_STREAM_CAPTURE)) + clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[0]]); +disable_tx_clk: + if (sai->mclk_streams & BIT(SNDRV_PCM_STREAM_PLAYBACK)) + clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[1]]); +disable_bus_clk: + clk_disable_unprepare(sai->bus_clk); + + return ret; } #endif /* CONFIG_PM */ -- 2.17.1

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[alsa-devel] Applied "sound: soc-acpi: fix implicit header use of module.h/export.h" to the asoc tree
by Mark Brown 08 May '19

08 May '19

The patch sound: soc-acpi: fix implicit header use of module.h/export.h has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.3 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 473849e7f208b2d7de004c6c6c0eabe037d125ac Mon Sep 17 00:00:00 2001 From: Paul Gortmaker <paul.gortmaker(a)windriver.com> Date: Sat, 13 Apr 2019 11:15:18 -0400 Subject: [PATCH] sound: soc-acpi: fix implicit header use of module.h/export.h This file is implicitly relying on an instance of including module.h from <linux/acpi.h>. Ideally, header files under include/linux shouldn't be adding includes of other headers, in anticipation of their consumers, but just the headers needed for the header itself to pass parsing with CPP. The module.h is particularly bad in this sense, as it itself does include a whole bunch of other headers, due to the complexity of module support. Here, we make the include explicit, in order to allow the future removal of module.h from linux/acpi.h without causing build breakage. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Paul Gortmaker <paul.gortmaker(a)windriver.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-acpi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index 4fb29f0e561e..444ce0602f76 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -4,6 +4,8 @@ // // Copyright (c) 2013-15, Intel Corporation. +#include <linux/export.h> +#include <linux/module.h> #include <sound/soc-acpi.h> struct snd_soc_acpi_mach * -- 2.20.1

1 0

[alsa-devel] Support low power consumption for ALC3204
by Kailang 08 May '19

08 May '19

Hi Takashi, This patch will support low consumption for Chrome ALC3204. Thanks. BR, Kailang

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v5.2
by Mark Brown 08 May '19

08 May '19

The following changes since commit 1587a061ef562de0d97c82a95863e191bcd69d63: ASoC: sprd: Add reserved DMA memory support (2019-05-06 22:48:03 +0900) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.2-2 for you to fetch changes up to ce38a75089f70f6380fc63a5478a7659b4eb3f47: ASoC: SOF: core: fix undefined nocodec reference (2019-05-08 15:42:00 +0900) ---------------------------------------------------------------- ASoC: Fixes for v5.2 A bunch of driver specific fixes that came in since the initial pull request for v5.2, mainly warning fixes for the newly added Sound Open Firmware code which people appeared to only start looking at after I'd sent the pull request. ---------------------------------------------------------------- Curtis Malainey (1): ASoC: RT5677-SPI: Disable 16Bit SPI Transfers Jon Hunter (1): ASoC: max98090: Fix restore of DAPM Muxes Olivier Moysan (2): ASoC: stm32: spdifrx: update pcm hardware constraints ASoC: stm32: spdifrx: change trace level on iec control Pierre-Louis Bossart (2): ASoC: SOF: xtensa: fix undefined references ASoC: SOF: core: fix undefined nocodec reference Randy Dunlap (1): ASoC: sound/soc/sof/: fix kconfig dependency warning Takashi Iwai (4): ASoC: da7219: Fix a compile warning at CONFIG_COMMON_CLK=n ASoC: SOF: Fix a compile warning with CONFIG_PCI=n ASoC: rockchip: Fix an uninitialized variable compile warning ASoC: SOF: Propagate sof_get_ctrl_copy_params() error properly sound/soc/codecs/da7219.c | 2 ++ sound/soc/codecs/max98090.c | 12 ++++++------ sound/soc/codecs/rt5677-spi.c | 35 ++++++++++++++++------------------- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/sof/Kconfig | 2 +- sound/soc/sof/core.c | 10 +++++----- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/ipc.c | 10 +++++++--- sound/soc/sof/ops.c | 2 +- sound/soc/stm/stm32_spdifrx.c | 5 +++-- 10 files changed, 43 insertions(+), 39 deletions(-)

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[alsa-devel] [PATCH] ASoC: cs42xx8: Add reset gpio handling
by S.j. Wang 08 May '19

08 May '19

Handle the reset GPIO and reset the device in pm_runtime_resume Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/cs42xx8.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index ebb9e0cf8364..fc28e6d26c6d 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -14,6 +14,7 @@ #include <linux/delay.h> #include <linux/module.h> #include <linux/of_device.h> +#include <linux/of_gpio.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> #include <sound/pcm_params.h> @@ -45,6 +46,7 @@ struct cs42xx8_priv { bool slave_mode; unsigned long sysclk; u32 tx_channels; + int gpio_reset; }; /* -127.5dB to 0dB with step of 0.5dB */ @@ -467,6 +469,17 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) return -EINVAL; } + cs42xx8->gpio_reset = of_get_named_gpio(dev->of_node, "gpio-reset", 0); + if (gpio_is_valid(cs42xx8->gpio_reset)) { + ret = devm_gpio_request_one(dev, cs42xx8->gpio_reset, + GPIOF_OUT_INIT_LOW, "cs42xx8 reset"); + if (ret) { + dev_err(dev, "unable to get reset gpio\n"); + return ret; + } + gpio_set_value_cansleep(cs42xx8->gpio_reset, 1); + } + cs42xx8->clk = devm_clk_get(dev, "mclk"); if (IS_ERR(cs42xx8->clk)) { dev_err(dev, "failed to get the clock: %ld\n", @@ -547,6 +560,11 @@ static int cs42xx8_runtime_resume(struct device *dev) return ret; } + if (gpio_is_valid(cs42xx8->gpio_reset)) { + gpio_set_value_cansleep(cs42xx8->gpio_reset, 0); + gpio_set_value_cansleep(cs42xx8->gpio_reset, 1); + } + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); if (ret) { @@ -559,6 +577,7 @@ static int cs42xx8_runtime_resume(struct device *dev) regcache_cache_only(cs42xx8->regmap, false); + regcache_mark_dirty(cs42xx8->regmap); ret = regcache_sync(cs42xx8->regmap); if (ret) { dev_err(dev, "failed to sync regmap: %d\n", ret); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: SOF: Propagate sof_get_ctrl_copy_params() error properly
by Takashi Iwai 08 May '19

08 May '19

This fixes a compile warning below by properly handling the error code from sof_get_ctrl_copy_params(): include/linux/kernel.h:843:43: warning: 'sparams.pl_size' may be used uninitialized in this function [-Wmaybe-uninitialized] sound/soc/sof/ipc.c:639:34: note: 'sparams.pl_size' was declared here The function returns an error before setting sparams.pl_size, so it'd assign an uninitialized value at a later point. Fixes: 53e0c72d98ba ("ASoC: SOF: Add support for IPC IO between DSP and Host") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/sof/ipc.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index ba1bb17a8d1e..f0b9d3c53f6f 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -567,7 +567,7 @@ static int sof_set_get_large_ctrl_data(struct snd_sof_dev *sdev, size_t offset = 0; size_t msg_bytes; size_t pl_size; - int err = 0; + int err; int i; /* allocate max ipc size because we have at least one */ @@ -576,9 +576,13 @@ static int sof_set_get_large_ctrl_data(struct snd_sof_dev *sdev, return -ENOMEM; if (send) - sof_get_ctrl_copy_params(cdata->type, cdata, partdata, sparams); + err = sof_get_ctrl_copy_params(cdata->type, cdata, partdata, + sparams); else - sof_get_ctrl_copy_params(cdata->type, partdata, cdata, sparams); + err = sof_get_ctrl_copy_params(cdata->type, partdata, cdata, + sparams); + if (err < 0) + return err; msg_bytes = sparams->msg_bytes; pl_size = sparams->pl_size; -- 2.16.4

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[alsa-devel] [PATCH] ASoC: rt5677: Add component driver name
by Curtis Malainey 08 May '19

08 May '19

Add name to component driver so it is possible to lookup the component when needed. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677.c | 7 ++++--- sound/soc/codecs/rt5677.h | 2 ++ 2 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 6fc70e441458..44071c7a4670 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4650,7 +4650,7 @@ static int rt5677_to_irq(struct gpio_chip *chip, unsigned offset) } static const struct gpio_chip rt5677_template_chip = { - .label = "rt5677", + .label = RT5677_DRV_NAME, .owner = THIS_MODULE, .direction_output = rt5677_gpio_direction_out, .set = rt5677_gpio_set, @@ -4958,6 +4958,7 @@ static struct snd_soc_dai_driver rt5677_dai[] = { }; static const struct snd_soc_component_driver soc_component_dev_rt5677 = { + .name = RT5677_DRV_NAME, .probe = rt5677_probe, .remove = rt5677_remove, .suspend = rt5677_suspend, @@ -5079,7 +5080,7 @@ static struct regmap_irq rt5677_irqs[] = { }; static struct regmap_irq_chip rt5677_irq_chip = { - .name = "rt5677", + .name = RT5677_DRV_NAME, .irqs = rt5677_irqs, .num_irqs = ARRAY_SIZE(rt5677_irqs), @@ -5267,7 +5268,7 @@ static int rt5677_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt5677_i2c_driver = { .driver = { - .name = "rt5677", + .name = RT5677_DRV_NAME, .of_match_table = rt5677_of_match, .acpi_match_table = ACPI_PTR(rt5677_acpi_match), }, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 183d92b03045..11a2ffceec3b 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1674,6 +1674,8 @@ #define RT5677_FIRMWARE1 "rt5677_dsp_fw1.bin" #define RT5677_FIRMWARE2 "rt5677_dsp_fw2.bin" +#define RT5677_DRV_NAME "rt5677" + /* System Clock Source */ enum { RT5677_SCLK_S_MCLK, -- 2.21.0.1020.gf2820cf01a-goog

2 2

[alsa-devel] [PATCH V5] ASoC: fsl_esai: Fix missing break in switch statement
by S.j. Wang 08 May '19

08 May '19

case ESAI_HCKT_EXTAL and case ESAI_HCKR_EXTAL should be independent of each other, so replace fall-through with break. Fixes: 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: <stable(a)vger.kernel.org> --- Changes in v5 - remove new line after Fixes Changes in v4 - Add acked-by Changes in v3 - Update subject line and cc stable Changes in v2 - Fix "Fixes" tag sound/soc/fsl/fsl_esai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 3623aa9a6f2e..15202a637197 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -251,7 +251,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, break; case ESAI_HCKT_EXTAL: ecr |= ESAI_ECR_ETI; - /* fall through */ + break; case ESAI_HCKR_EXTAL: ecr |= ESAI_ECR_ERI; break; -- 1.9.1

3 2

[alsa-devel] [PATCH v4 0/3] Fix jack detection for Chromebook Pixel
by Fletcher Woodruff 08 May '19

08 May '19

Headphone/mic jack detection doesn't work on the Chromebook Pixel 2015. This patch changes the irq implementation to support polarity flipping and fixes the configuration code so that correct GPIO pins are read from ACPI. With this series, plugging and unplugging the headphone jack switches between headphones and speakers automatically, and headset microphones are also detected. v4: - Fix incorrect void* cast in rt5677_irq() v3: - Update commit message for patch 1/3 to clarify why we implement our own irq_chip. v2: - Split IRQ change into two patches: adding and fixing potential race - Change config reading code to try both DT and ACPI style names Ben Zhang (1): ASoC: rt5677: allow multiple interrupt sources Fletcher Woodruff (2): ASoC: rt5677: handle concurrent interrupts ASoC: rt5677: fall back to DT prop names on error sound/soc/codecs/rt5677.c | 346 ++++++++++++++++++++++++++------------ sound/soc/codecs/rt5677.h | 14 +- 2 files changed, 256 insertions(+), 104 deletions(-) -- 2.21.0.1020.gf2820cf01a-goog

3 6

[alsa-devel] cs4265 : alsactl store and "C data buffer" fails
by Matt Flax 08 May '19

08 May '19

Hi, When alsactl store is run with a cs4265 chip, the following error occurs (from https://github.com/Audio-Injector/Ultra/issues/25) : $ sudo alsactl store alsactl: get_control:256: Cannot read control '2,0,0,C Data Buffer,0': Input/output error Upon commenting out the C data buffer control it works : https://github.com/torvalds/linux/blob/master/sound/soc/codecs/cs4265.c#L163 Can you suggest a suitable method for fixing this bug ? thanks Matt

2 2

[alsa-devel] [PATCH] Headset fixup for System76 Gazelle (gaze14)
by Jeremy Soller 08 May '19

08 May '19

On the System76 Gazelle (gaze14), there is a headset microphone input attached to 0x1a that does not have a jack detect. In order to get it working, the pin configuration needs to be set correctly, and the ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC fixup needs to be applied. This is identical to the patch already applied for the System76 Darter Pro (darp5). Signed-off-by: Jeremy Soller <jeremy(a)system76.com> --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f5b510f119ed..4ea3affad617 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6931,6 +6931,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8550, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),

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[alsa-devel] Question: Selecting one codec from multiple codecs using rk3288_hdmi_analog driver
by Cheng-yi Chiang 08 May '19

08 May '19

Hi all, I am wondering whether it is possible to selectively play to one codec out of multiple codecs. There is a multiple codec usage in driver rk3288_hdmi_analog: https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rk3288_hdm… https://patchwork.kernel.org/patch/9554267/ I am not sure whether the driver supports playing to analog output or HDMI selectively one at a time ? I don't have a Firefly-RK3288 board to test that driver, but I have RK3288 veyron board. I followed the similar way like that driver does to support max98090 and HDMI on RK3288 veyron board. However, when I play to the device, it goes to both max98090 and HDMI codec. >From rk3288_hdmi_analog implementation, I failed to find out how to turn off HDMI codec when HDMI path is unwanted. The DAPM pin switch "HDMI" does not help since that even if DAPM path I2S Playback -> TX -> HDMI is turned off, the startup ops of HDMI codec is still called from soc_pcm_open path. Could someone please explain whether this is supported in the framework ? There seems to be two ways to achieve this One is to use TDM and select different channel for two codecs. The second is to add mixer controls and turn codec on/off to disable codec completely. But those requires changes on codec driver and the capability of TDM or mixer control on each codec. Maybe there should be a way to bypass startup ops of one of multiple codecs if it is unwanted ? Any help or suggestion is appreciated. Thanks!

1 0

[alsa-devel] [PATCH v1 0/2] ASoC: Intel: Add Cometlake PCI IDs
by Evan Green 08 May '19

08 May '19

This small series adds PCI IDs for Cometlake platforms, for a dazzling audio experience. Evan Green (2): ASoC: SOF: Add Comet Lake PCI ID ASoC: Intel: Skylake: Add Cometlake PCI IDs sound/soc/intel/Kconfig | 9 +++++++++ sound/soc/intel/skylake/skl-messages.c | 8 ++++++++ sound/soc/intel/skylake/skl.c | 5 +++++ sound/soc/sof/intel/Kconfig | 16 ++++++++++++++++ sound/soc/sof/sof-pci-dev.c | 4 ++++ 5 files changed, 42 insertions(+) -- 2.20.1

3 9

[alsa-devel] [PATCH] ALSA: hda/intel: add CometLake PCI IDs
by Pierre-Louis Bossart 07 May '19

07 May '19

Add PCI IDs for LP and H skews. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 789308f54785..0741eae23f10 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2378,6 +2378,12 @@ static const struct pci_device_id azx_ids[] = { /* Cannonlake */ { PCI_DEVICE(0x8086, 0x9dc8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* CometLake-LP */ + { PCI_DEVICE(0x8086, 0x02C8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* CometLake-H */ + { PCI_DEVICE(0x8086, 0x06C8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- 2.17.1

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[alsa-devel] [asoc:for-5.2 248/316] sound/soc/sof/core.c:279: undefined reference to `sof_nocodec_setup'
by kbuild test robot 07 May '19

07 May '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: f153bf49ddf6169544b24ef5ea26fa3eb4996b95 commit: 7e978fa37df7f771a5d02934ff828a0ee36bcf11 [248/316] ASoC: SOF: Add Build support for SOF core and Intel drivers config: i386-randconfig-l1-05071509 (attached as .config) compiler: gcc-5 (Debian 5.5.0-3) 5.4.1 20171010 reproduce: git checkout 7e978fa37df7f771a5d02934ff828a0ee36bcf11 # save the attached .config to linux build tree make ARCH=i386 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): ld: sound/soc/sof/core.o: in function `sof_machine_check': >> sound/soc/sof/core.c:279: undefined reference to `sof_nocodec_setup' vim +279 sound/soc/sof/core.c c16211d6 Liam Girdwood 2019-04-12 255 c16211d6 Liam Girdwood 2019-04-12 256 /* c16211d6 Liam Girdwood 2019-04-12 257 * SOF Driver enumeration. c16211d6 Liam Girdwood 2019-04-12 258 */ c16211d6 Liam Girdwood 2019-04-12 259 static int sof_machine_check(struct snd_sof_dev *sdev) c16211d6 Liam Girdwood 2019-04-12 260 { c16211d6 Liam Girdwood 2019-04-12 261 struct snd_sof_pdata *plat_data = sdev->pdata; c16211d6 Liam Girdwood 2019-04-12 262 struct snd_soc_acpi_mach *machine; c16211d6 Liam Girdwood 2019-04-12 263 int ret; c16211d6 Liam Girdwood 2019-04-12 264 c16211d6 Liam Girdwood 2019-04-12 265 if (plat_data->machine) c16211d6 Liam Girdwood 2019-04-12 266 return 0; c16211d6 Liam Girdwood 2019-04-12 267 c16211d6 Liam Girdwood 2019-04-12 268 if (!IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC)) { c16211d6 Liam Girdwood 2019-04-12 269 dev_err(sdev->dev, "error: no matching ASoC machine driver found - aborting probe\n"); c16211d6 Liam Girdwood 2019-04-12 270 return -ENODEV; c16211d6 Liam Girdwood 2019-04-12 271 } c16211d6 Liam Girdwood 2019-04-12 272 c16211d6 Liam Girdwood 2019-04-12 273 /* fallback to nocodec mode */ c16211d6 Liam Girdwood 2019-04-12 274 dev_warn(sdev->dev, "No ASoC machine driver found - using nocodec\n"); c16211d6 Liam Girdwood 2019-04-12 275 machine = devm_kzalloc(sdev->dev, sizeof(*machine), GFP_KERNEL); c16211d6 Liam Girdwood 2019-04-12 276 if (!machine) c16211d6 Liam Girdwood 2019-04-12 277 return -ENOMEM; c16211d6 Liam Girdwood 2019-04-12 278 c16211d6 Liam Girdwood 2019-04-12 @279 ret = sof_nocodec_setup(sdev->dev, plat_data, machine, c16211d6 Liam Girdwood 2019-04-12 280 plat_data->desc, plat_data->desc->ops); c16211d6 Liam Girdwood 2019-04-12 281 if (ret < 0) c16211d6 Liam Girdwood 2019-04-12 282 return ret; c16211d6 Liam Girdwood 2019-04-12 283 c16211d6 Liam Girdwood 2019-04-12 284 plat_data->machine = machine; c16211d6 Liam Girdwood 2019-04-12 285 c16211d6 Liam Girdwood 2019-04-12 286 return 0; c16211d6 Liam Girdwood 2019-04-12 287 } c16211d6 Liam Girdwood 2019-04-12 288 :::::: The code at line 279 was first introduced by commit :::::: c16211d6226dbde8819b84da07508083a1138a06 ASoC: SOF: Add Sound Open Firmware driver core :::::: TO: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] Right internal speaker and internal microphone doesn't seem to work
by Drago Praznik 07 May '19

07 May '19

My Mint 19 (kernel: 4.15.0-48-generic) on Lenovo V110 never succeeded to use an internal microphone and the right internal speaker also doesn't seem to work. Previously I tried to install Windows 10 and all the hardware works normally whilst using Win10. Here's some additional info of my setup: http://alsa-project.org/db/?f=9e9d2f773b9682f2dbd2a22cd1f4fd0613eb0a2d Originally I submitted a post on forum: https://forums.linuxmint.com/viewtopic.php?f=49&t=292732 ... since it didn't help, I ended up here. Cheers, Drago

3 4

[alsa-devel] [asoc:for-5.2 248/316] ld: sound/soc/sof/sof-acpi-dev.o:undefined reference to `sof_xtensa_arch_ops'
by kbuild test robot 07 May '19

07 May '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: f153bf49ddf6169544b24ef5ea26fa3eb4996b95 commit: 7e978fa37df7f771a5d02934ff828a0ee36bcf11 [248/316] ASoC: SOF: Add Build support for SOF core and Intel drivers config: i386-randconfig-l0-05071509 (attached as .config) compiler: gcc-5 (Debian 5.5.0-3) 5.4.1 20171010 reproduce: git checkout 7e978fa37df7f771a5d02934ff828a0ee36bcf11 # save the attached .config to linux build tree make ARCH=i386 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): >> ld: sound/soc/sof/sof-acpi-dev.o:(.rodata+0x120): undefined reference to `sof_xtensa_arch_ops' ld: sound/soc/sof/sof-acpi-dev.o:(.rodata+0x180): undefined reference to `sof_xtensa_arch_ops' ld: sound/soc/sof/sof-acpi-dev.o:(.rodata+0x1e0): undefined reference to `sof_xtensa_arch_ops' --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] KASAN: use-after-free Write in check_and_subscribe_port
by syzbot 07 May '19

07 May '19

Hello, syzbot found the following crash on: HEAD commit: bf3bd966 Merge tag 'usb-5.1-rc8' of git://git.kernel.org/p.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=1431a0e0a00000 kernel config: https://syzkaller.appspot.com/x/.config?x=ef1b87b455c397cf dashboard link: https://syzkaller.appspot.com/bug?extid=20ab495fadf081e8a2b0 compiler: gcc (GCC) 9.0.0 20181231 (experimental) 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+20ab495fadf081e8a2b0(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in __list_add include/linux/list.h:64 [inline] BUG: KASAN: use-after-free in list_add_tail include/linux/list.h:93 [inline] BUG: KASAN: use-after-free in check_and_subscribe_port+0x7bd/0x860 sound/core/seq/seq_ports.c:524 Write of size 8 at addr ffff88809587bba0 by task syz-executor.2/19168 CPU: 0 PID: 19168 Comm: syz-executor.2 Not tainted 5.1.0-rc7+ #95 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x172/0x1f0 lib/dump_stack.c:113 print_address_description.cold+0x7c/0x20d mm/kasan/report.c:187 kasan_report.cold+0x1b/0x40 mm/kasan/report.c:317 __asan_report_store8_noabort+0x17/0x20 mm/kasan/generic_report.c:137 __list_add include/linux/list.h:64 [inline] list_add_tail include/linux/list.h:93 [inline] check_and_subscribe_port+0x7bd/0x860 sound/core/seq/seq_ports.c:524 snd_seq_port_connect+0x388/0x510 sound/core/seq/seq_ports.c:587 snd_seq_ioctl_subscribe_port+0x1e5/0x310 sound/core/seq/seq_clientmgr.c:1458 snd_seq_ioctl+0x224/0x3e0 sound/core/seq/seq_clientmgr.c:2138 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:509 [inline] do_vfs_ioctl+0xd6e/0x1390 fs/ioctl.c:696 ksys_ioctl+0xab/0xd0 fs/ioctl.c:713 __do_sys_ioctl fs/ioctl.c:720 [inline] __se_sys_ioctl fs/ioctl.c:718 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:718 do_syscall_64+0x103/0x610 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x458da9 Code: ad b8 fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 7b b8 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007f6a5c605c78 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000003 RCX: 0000000000458da9 RDX: 0000000020000000 RSI: 0000000040505330 RDI: 0000000000000003 RBP: 000000000073bf00 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f6a5c6066d4 R13: 00000000004cc598 R14: 00000000004d5ee8 R15: 00000000ffffffff Allocated by task 19168: save_stack+0x45/0xd0 mm/kasan/common.c:75 set_track mm/kasan/common.c:87 [inline] __kasan_kmalloc mm/kasan/common.c:497 [inline] __kasan_kmalloc.constprop.0+0xcf/0xe0 mm/kasan/common.c:470 kasan_kmalloc+0x9/0x10 mm/kasan/common.c:511 kmem_cache_alloc_trace+0x151/0x760 mm/slab.c:3622 kmalloc include/linux/slab.h:547 [inline] kzalloc include/linux/slab.h:742 [inline] snd_seq_port_connect+0x60/0x510 sound/core/seq/seq_ports.c:571 snd_seq_ioctl_subscribe_port+0x1e5/0x310 sound/core/seq/seq_clientmgr.c:1458 snd_seq_ioctl+0x224/0x3e0 sound/core/seq/seq_clientmgr.c:2138 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:509 [inline] do_vfs_ioctl+0xd6e/0x1390 fs/ioctl.c:696 ksys_ioctl+0xab/0xd0 fs/ioctl.c:713 __do_sys_ioctl fs/ioctl.c:720 [inline] __se_sys_ioctl fs/ioctl.c:718 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:718 do_syscall_64+0x103/0x610 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe Freed by task 19167: save_stack+0x45/0xd0 mm/kasan/common.c:75 set_track mm/kasan/common.c:87 [inline] __kasan_slab_free+0x102/0x150 mm/kasan/common.c:459 kasan_slab_free+0xe/0x10 mm/kasan/common.c:467 __cache_free mm/slab.c:3499 [inline] kfree+0xcf/0x230 mm/slab.c:3822 snd_seq_port_disconnect+0x209/0x280 sound/core/seq/seq_ports.c:632 snd_seq_ioctl_unsubscribe_port+0x1e5/0x310 sound/core/seq/seq_clientmgr.c:1499 snd_seq_ioctl+0x224/0x3e0 sound/core/seq/seq_clientmgr.c:2138 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:509 [inline] do_vfs_ioctl+0xd6e/0x1390 fs/ioctl.c:696 ksys_ioctl+0xab/0xd0 fs/ioctl.c:713 __do_sys_ioctl fs/ioctl.c:720 [inline] __se_sys_ioctl fs/ioctl.c:718 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:718 do_syscall_64+0x103/0x610 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe The buggy address belongs to the object at ffff88809587bb40 which belongs to the cache kmalloc-128 of size 128 The buggy address is located 96 bytes inside of 128-byte region [ffff88809587bb40, ffff88809587bbc0) The buggy address belongs to the page: page:ffffea0002561ec0 count:1 mapcount:0 mapping:ffff8880aa400640 index:0xffff88809587bd80 flags: 0x1fffc0000000200(slab) raw: 01fffc0000000200 ffffea0002949788 ffffea0002906888 ffff8880aa400640 raw: ffff88809587bd80 ffff88809587b000 000000010000000f 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff88809587ba80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff88809587bb00: fc fc fc fc fc fc fc fc fb fb fb fb fb fb fb fb > ffff88809587bb80: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ^ ffff88809587bc00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff88809587bc80: fc fc fc fc fc fc fc fc 00 00 00 00 00 00 00 00 ================================================================== --- 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#status for how to communicate with syzbot.

1 0

[alsa-devel] Support low power consumption for ALC3254
by Kailang 07 May '19

07 May '19

Hi Takashi, Chrome OS want to support low power consumption. This patch will support low consumption. Thanks. BR, Kailang

2 6

[alsa-devel] [PATCH] ASoC: soc-pcm: BE dai needs prepare when pause release after resume
by libin.yang＠intel.com 07 May '19

07 May '19

From: Libin Yang <libin.yang(a)intel.com> If playback/capture is paused and system enters S3, after system returns from suspend, BE dai needs to call prepare() callback when playback/capture is released from pause. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/soc-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d2aa560..0888995 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2471,7 +2471,8 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; dev_dbg(be->dev, "ASoC: prepare BE %s\n", -- 2.7.4

3 3

Re: [alsa-devel] [PATCH 00/19] ASoC: SOF: improvements for ABI checks and Intel code
by Pierre-Louis Bossart 07 May '19

07 May '19

On 5/5/19 10:51 PM, Mark Brown wrote: > On Fri, May 03, 2019 at 09:47:39AM -0500, Pierre-Louis Bossart wrote: >> On 5/3/19 12:40 AM, Mark Brown wrote: > >>> reviewer fatigue setting in with this stuff, one thing that'd really >>> help here is if there were some help from Intel people with review for >>> the DPCM code. > >> I can certainly understand reviewer fatigue, i've had to put a time limit on >> daily reviews to keep my sanity, but I don't get your last point. These >> patches were submitted and reviewed by Intel people on GitHub, what you see >> here is the result of multiple iterations precisely to make sure the patches >> are acceptable for upstream. we've set the goal of having two Intel aprovers >> for each patch. Can you elaborate on how we can make your life simpler? > > The last point there is that Intel only write patches, nobody from Intel > is reviewing other people's patches especially in areas of the code like > DPCM which are complex, fragile and where Intel is by far the most > active user. It's a valid point, and for now we indeed only check for non-regressions and provide point solutions without looking at the bigger picture. We have a couple of people ramping up (Ranjani, Libin, Guennadi, Jaska) and hopefully at some point we'll be able to review and improve.

2 1

[alsa-devel] [PATCH RESEND v4] ASoC: Intel: boards: kbl_da7219_max98927: add dai_trigger function
by mac.chiang＠intel.com 07 May '19

07 May '19

From: Mac Chiang <mac.chiang(a)intel.com> amplifier feedback is not modeled as being dependent on any active output. Even when there is no playback happening, parts of the graph, specifically the IV sense->speaker protection->output remains active and this prevents the DSP from entering low-power states. This patch suggest a machine driver level approach where the speaker pins are enabled/disabled dynamically depending on stream start/stop events. DPAM graph representations show the feedback loop is indeed disabled and low-power states can be reached. Signed-off-by: Jenny TC <jenny.tc(a)intel.com> Signed-off-by: Harshapriya.n <harshapriya.n(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- Changelog: v4: - Changed device name in in kabylake_ssp0_trigger to address a conflict with recently merged patch MAXIM_DEV0_NAME -> MAX98927_DEV0_NAME ... v3: - fixed the return logic at the end of kabylake_card_late_probe() v2: - described clearly in commit message - added snd_soc_dapm_disable_pin in kabylake_card_late_probe() --- --- sound/soc/intel/boards/kbl_da7219_max98927.c | 73 +++++++++++++++++++++++++++- 1 file changed, 71 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index f72a7bf..1efe7fd 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -219,8 +219,60 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return 0; } +static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int j, ret; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + const char *name = codec_dai->component->name; + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + char pin_name[20]; + + if (strcmp(name, MAX98927_DEV0_NAME) && + strcmp(name, MAX98927_DEV1_NAME) && + strcmp(name, MAX98373_DEV0_NAME) && + strcmp(name, MAX98373_DEV1_NAME)) + continue; + + snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk", + codec_dai->component->name_prefix); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = snd_soc_dapm_enable_pin(dapm, pin_name); + if (ret) { + dev_err(rtd->dev, "failed to enable %s: %d\n", + pin_name, ret); + return ret; + } + snd_soc_dapm_sync(dapm); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = snd_soc_dapm_disable_pin(dapm, pin_name); + if (ret) { + dev_err(rtd->dev, "failed to disable %s: %d\n", + pin_name, ret); + return ret; + } + snd_soc_dapm_sync(dapm); + break; + } + } + + return 0; +} + static struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, + .trigger = kabylake_ssp0_trigger, }; static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, @@ -950,6 +1002,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); struct kbl_hdmi_pcm *pcm; + struct snd_soc_dapm_context *dapm = &card->dapm; struct snd_soc_component *component = NULL; int err, i = 0; char jack_name[NAME_SIZE]; @@ -976,9 +1029,25 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) if (!component) return -EINVAL; - return hdac_hdmi_jack_port_init(component, &card->dapm); - return 0; + err = hdac_hdmi_jack_port_init(component, &card->dapm); + + if (err < 0) + return err; + + err = snd_soc_dapm_disable_pin(dapm, "Left Spk"); + if (err) { + dev_err(card->dev, "failed to disable Left Spk: %d\n", err); + return err; + } + + err = snd_soc_dapm_disable_pin(dapm, "Right Spk"); + if (err) { + dev_err(card->dev, "failed to disable Right Spk: %d\n", err); + return err; + } + + return snd_soc_dapm_sync(dapm); } /* kabylake audio machine driver for SPT + DA7219 */ -- 2.7.4

3 5

[alsa-devel] [PATCH] ASoC: rt5677-spi: Rename driver to differentiate from main codec
by Curtis Malainey 07 May '19

07 May '19

Currently the SPI driver and the main codec share the same name. This will become confusing when looking up components when using both drivers. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677-spi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 84b6bd8b50e1..871ccb37318d 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -29,6 +29,8 @@ #include "rt5677-spi.h" +#define DRV_NAME "rt5677spi" + #define RT5677_SPI_BURST_LEN 240 #define RT5677_SPI_HEADER 5 #define RT5677_SPI_FREQ 6000000 @@ -232,7 +234,7 @@ MODULE_DEVICE_TABLE(acpi, rt5677_spi_acpi_id); static struct spi_driver rt5677_spi_driver = { .driver = { - .name = "rt5677", + .name = DRV_NAME, .acpi_match_table = ACPI_PTR(rt5677_spi_acpi_id), }, .probe = rt5677_spi_probe, -- 2.21.0.1020.gf2820cf01a-goog

2 4

[alsa-devel] [PATCH v5] ASoC: fsl_esai: Add pm runtime function
by Nicolin Chen 07 May '19

07 May '19

From: "S.j. Wang" <shengjiu.wang(a)nxp.com> Add pm runtime support and move clock handling there. Close the clocks at suspend to reduce the power consumption. fsl_esai_suspend is replaced by pm_runtime_force_suspend. fsl_esai_resume is replaced by pm_runtime_force_resume. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changes in v5 -Replaced my Acked-by with Signed-off-by as a resend. Changes in v4 -resend base on for-5.2 Changes in v3 -refine the commit comments. -add acked-by Changes in v2 -refine the commit comments. -move regcache_mark_dirty to runtime suspend. sound/soc/fsl/fsl_esai.c | 141 +++++++++++++++++++++------------------ 1 file changed, 77 insertions(+), 64 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c7410bbfd2af..022368c8a074 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -9,6 +9,7 @@ #include <linux/module.h> #include <linux/of_irq.h> #include <linux/of_platform.h> +#include <linux/pm_runtime.h> #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> @@ -466,30 +467,6 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); - int ret; - - /* - * Some platforms might use the same bit to gate all three or two of - * clocks, so keep all clocks open/close at the same time for safety - */ - ret = clk_prepare_enable(esai_priv->coreclk); - if (ret) - return ret; - if (!IS_ERR(esai_priv->spbaclk)) { - ret = clk_prepare_enable(esai_priv->spbaclk); - if (ret) - goto err_spbaclk; - } - if (!IS_ERR(esai_priv->extalclk)) { - ret = clk_prepare_enable(esai_priv->extalclk); - if (ret) - goto err_extalck; - } - if (!IS_ERR(esai_priv->fsysclk)) { - ret = clk_prepare_enable(esai_priv->fsysclk); - if (ret) - goto err_fsysclk; - } if (!dai->active) { /* Set synchronous mode */ @@ -506,16 +483,6 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, return 0; -err_fsysclk: - if (!IS_ERR(esai_priv->extalclk)) - clk_disable_unprepare(esai_priv->extalclk); -err_extalck: - if (!IS_ERR(esai_priv->spbaclk)) - clk_disable_unprepare(esai_priv->spbaclk); -err_spbaclk: - clk_disable_unprepare(esai_priv->coreclk); - - return ret; } static int fsl_esai_hw_params(struct snd_pcm_substream *substream, @@ -576,20 +543,6 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, return 0; } -static void fsl_esai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); - - if (!IS_ERR(esai_priv->fsysclk)) - clk_disable_unprepare(esai_priv->fsysclk); - if (!IS_ERR(esai_priv->extalclk)) - clk_disable_unprepare(esai_priv->extalclk); - if (!IS_ERR(esai_priv->spbaclk)) - clk_disable_unprepare(esai_priv->spbaclk); - clk_disable_unprepare(esai_priv->coreclk); -} - static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -658,7 +611,6 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, static const struct snd_soc_dai_ops fsl_esai_dai_ops = { .startup = fsl_esai_startup, - .shutdown = fsl_esai_shutdown, .trigger = fsl_esai_trigger, .hw_params = fsl_esai_hw_params, .set_sysclk = fsl_esai_set_dai_sysclk, @@ -947,6 +899,10 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + pm_runtime_enable(&pdev->dev); + + regcache_cache_only(esai_priv->regmap, true); + ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); if (ret) dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); @@ -954,6 +910,13 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } +static int fsl_esai_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + + return 0; +} + static const struct of_device_id fsl_esai_dt_ids[] = { { .compatible = "fsl,imx35-esai", }, { .compatible = "fsl,vf610-esai", }, @@ -961,22 +924,35 @@ static const struct of_device_id fsl_esai_dt_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); -#ifdef CONFIG_PM_SLEEP -static int fsl_esai_suspend(struct device *dev) -{ - struct fsl_esai *esai = dev_get_drvdata(dev); - - regcache_cache_only(esai->regmap, true); - regcache_mark_dirty(esai->regmap); - - return 0; -} - -static int fsl_esai_resume(struct device *dev) +#ifdef CONFIG_PM +static int fsl_esai_runtime_resume(struct device *dev) { struct fsl_esai *esai = dev_get_drvdata(dev); int ret; + /* + * Some platforms might use the same bit to gate all three or two of + * clocks, so keep all clocks open/close at the same time for safety + */ + ret = clk_prepare_enable(esai->coreclk); + if (ret) + return ret; + if (!IS_ERR(esai->spbaclk)) { + ret = clk_prepare_enable(esai->spbaclk); + if (ret) + goto err_spbaclk; + } + if (!IS_ERR(esai->extalclk)) { + ret = clk_prepare_enable(esai->extalclk); + if (ret) + goto err_extalclk; + } + if (!IS_ERR(esai->fsysclk)) { + ret = clk_prepare_enable(esai->fsysclk); + if (ret) + goto err_fsysclk; + } + regcache_cache_only(esai->regmap, false); /* FIFO reset for safety */ @@ -987,22 +963,59 @@ static int fsl_esai_resume(struct device *dev) ret = regcache_sync(esai->regmap); if (ret) - return ret; + goto err_regcache_sync; /* FIFO reset done */ regmap_update_bits(esai->regmap, REG_ESAI_TFCR, ESAI_xFCR_xFR, 0); regmap_update_bits(esai->regmap, REG_ESAI_RFCR, ESAI_xFCR_xFR, 0); + return 0; + +err_regcache_sync: + if (!IS_ERR(esai->fsysclk)) + clk_disable_unprepare(esai->fsysclk); +err_fsysclk: + if (!IS_ERR(esai->extalclk)) + clk_disable_unprepare(esai->extalclk); +err_extalclk: + if (!IS_ERR(esai->spbaclk)) + clk_disable_unprepare(esai->spbaclk); +err_spbaclk: + clk_disable_unprepare(esai->coreclk); + + return ret; +} + +static int fsl_esai_runtime_suspend(struct device *dev) +{ + struct fsl_esai *esai = dev_get_drvdata(dev); + + regcache_cache_only(esai->regmap, true); + regcache_mark_dirty(esai->regmap); + + if (!IS_ERR(esai->fsysclk)) + clk_disable_unprepare(esai->fsysclk); + if (!IS_ERR(esai->extalclk)) + clk_disable_unprepare(esai->extalclk); + if (!IS_ERR(esai->spbaclk)) + clk_disable_unprepare(esai->spbaclk); + clk_disable_unprepare(esai->coreclk); + return 0; } -#endif /* CONFIG_PM_SLEEP */ +#endif /* CONFIG_PM */ static const struct dev_pm_ops fsl_esai_pm_ops = { - SET_SYSTEM_SLEEP_PM_OPS(fsl_esai_suspend, fsl_esai_resume) + SET_RUNTIME_PM_OPS(fsl_esai_runtime_suspend, + fsl_esai_runtime_resume, + NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) }; static struct platform_driver fsl_esai_driver = { .probe = fsl_esai_probe, + .remove = fsl_esai_remove, .driver = { .name = "fsl-esai-dai", .pm = &fsl_esai_pm_ops, -- 2.17.1

2 4

[alsa-devel] [PATCH 0/3] ASoC: stm32: spdifrx: add some features
by Olivier Moysan 07 May '19

07 May '19

Update and add some features in STM32 SPDIFRX driver. - pdate pcm hardware constraints - change trace level on iec control - add management of identification registers Olivier Moysan (3): ASoC: stm32: spdifrx: update pcm hardware constraints ASoC: stm32: spdifrx: change trace level on iec control ASoC: stm32: spdifrx: manage identification registers sound/soc/stm/stm32_spdifrx.c | 42 ++++++++++++++++++++++++++++++++++++++---- 1 file changed, 38 insertions(+), 4 deletions(-) -- 2.7.4

2 9

[alsa-devel] [PATCH] ASoC: rt5677: use more of the volume range from DACs
by Curtis Malainey 07 May '19

07 May '19

From: Dylan Reid <dgreid(a)chromium.org> The DACs volume can go over 0, both according to the data sheet and real world testing. The control can go up to +30dB. This was tested by playing audio at full volume on a samus chromebook. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> Reviewed-by: Hsinyu Chao <hychao(a)chromium.org> Signed-off-by: Ross Zwisler <zwisler(a)google.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 6fc70e441458..907c957b614d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -832,13 +832,13 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5677_DAC1_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 127, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC2 Playback Volume", RT5677_DAC2_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 127, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC3 Playback Volume", RT5677_DAC3_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 127, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC4 Playback Volume", RT5677_DAC4_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 127, 0, dac_vol_tlv), /* IN1/IN2 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5677_IN1, RT5677_BST_SFT1, 8, 0, bst_tlv), -- 2.21.0.1020.gf2820cf01a-goog

2 3

[alsa-devel] [PATCH] ASoC: sound/soc/sof/: fix kconfig dependency warning
by Randy Dunlap 07 May '19

07 May '19

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kconfig warning for unmet dependency for IOSF_MBI when PCI is not set/enabled. Fixes this warning: WARNING: unmet direct dependencies detected for IOSF_MBI Depends on [n]: PCI [=n] Selected by [y]: - SND_SOC_SOF_ACPI [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_SOF_TOPLEVEL [=y] && (ACPI [=y] || COMPILE_TEST [=n]) && X86 [=y] Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/sof/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20190506.orig/sound/soc/sof/Kconfig +++ linux-next-20190506/sound/soc/sof/Kconfig @@ -28,7 +28,7 @@ config SND_SOC_SOF_ACPI select SND_SOC_ACPI if ACPI select SND_SOC_SOF_OPTIONS select SND_SOC_SOF_INTEL_ACPI if SND_SOC_SOF_INTEL_TOPLEVEL - select IOSF_MBI if X86 + select IOSF_MBI if X86 && PCI help This adds support for ACPI enumeration. This option is required to enable Intel Haswell/Broadwell/Baytrail/Cherrytrail devices

3 3

[alsa-devel] [PATCH 0/3] ASoC: trivial compile warning fixes
by Takashi Iwai 07 May '19

07 May '19

Hi, here are the three fixes for trivial compile warnings I caught during build test. Takashi === Takashi Iwai (3): ASoC: da7219: Fix a compile warning at CONFIG_COMMON_CLK=n ASoC: SOF: Fix a compile warning with CONFIG_PCI=n ASoC: rockchip: Fix an uninitialized variable compile warning sound/soc/codecs/da7219.c | 2 ++ sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/sof/ops.c | 2 +- 3 files changed, 4 insertions(+), 2 deletions(-) -- 2.16.4

4 11

[alsa-devel] [PATCH] ASoC: sprd: Add reserved DMA memory support
by Baolin Wang 07 May '19

07 May '19

For Spreadtrum audio platform driver, it need allocate a larger DMA buffer dynamically to copy audio data between userspace and kernel space, but that will increase the risk of memory allocation failure especially the system is under heavy load situation. To make sure the audio can work in this scenario, we usually reserve one region of memory to be used as a shared pool of DMA buffers for the platform component. So add of_reserved_mem_device_init_by_idx() function to initialize the shared pool of DMA buffers to be used by the platform component. Signed-off-by: Baolin Wang <baolin.wang(a)linaro.org> --- sound/soc/sprd/sprd-pcm-dma.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sprd/sprd-pcm-dma.c b/sound/soc/sprd/sprd-pcm-dma.c index 9be6d4b..d38ebbb 100644 --- a/sound/soc/sprd/sprd-pcm-dma.c +++ b/sound/soc/sprd/sprd-pcm-dma.c @@ -6,6 +6,7 @@ #include <linux/dma/sprd-dma.h> #include <linux/kernel.h> #include <linux/module.h> +#include <linux/of_reserved_mem.h> #include <linux/platform_device.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -530,8 +531,14 @@ static void sprd_pcm_free(struct snd_pcm *pcm) static int sprd_soc_platform_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; int ret; + ret = of_reserved_mem_device_init_by_idx(&pdev->dev, np, 0); + if (ret) + dev_warn(&pdev->dev, + "no reserved DMA memory for audio platform device\n"); + ret = devm_snd_soc_register_component(&pdev->dev, &sprd_soc_component, NULL, 0); if (ret) -- 1.7.9.5

2 5

[alsa-devel] [RFC PATCH] ASoC: codec: hdac_hdmi: no checking monitor in hw_params
by libin.yang＠intel.com 07 May '19

07 May '19

From: Libin Yang <libin.yang(a)intel.com> This patch move the check of monitor from hw_params to trigger callback. The original code will check the monitor presence in hw_params. If the monitor doesn't exist, hw_params will return -ENODEV. Mostly this is OK. However, pulseaudio will check the pcm devices when kernel is booting up. It will try to open, set hw_params, prepare such pcm devices. We can't guarantee that the monitor will be connected when kernel is booting up. Especially, hdac_hdmi will export 3 pcms at most. It's hard to say users will connect 3 monitors to the HDMI/DP ports. This will cause pulseaudio fail in parsing the pcm devices because the driver will return -ENODEV in hw_params. This patch tries to move the check of monitor presence into trigger callback. This can "trick" the pulseaudio the pcm is ready. This bug is found when we try to enable HDMI detection in gnome-sound-setting for ASoC hdac_hdmi. After we enable the hdmi in UCM, pulseaudio will try to parse the hdmi pcm devices. It will cause failure if there are no monitors connected. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 44 +++++++++++++++++++++++++++++++------------- 1 file changed, 31 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4de1fbf..f482e09 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -455,24 +455,11 @@ static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hparams, struct snd_soc_dai *dai) { struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); - struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_dai_port_map *dai_map; - struct hdac_hdmi_port *port; struct hdac_hdmi_pcm *pcm; int format; dai_map = &hdmi->dai_map[dai->id]; - port = dai_map->port; - - if (!port) - return -ENODEV; - - if ((!port->eld.monitor_present) || (!port->eld.eld_valid)) { - dev_err(&hdev->dev, - "device is not configured for this pin:port%d:%d\n", - port->pin->nid, port->id); - return -ENODEV; - } format = snd_hdac_calc_stream_format(params_rate(hparams), params_channels(hparams), params_format(hparams), @@ -630,6 +617,36 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream, dai_map->port = NULL; } +static int hdac_hdmi_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct hdac_hdmi_port *port; + struct hdac_hdmi_dai_port_map *dai_map; + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); + struct hdac_device *hdev = hdmi->hdev; + + /* + * When start, if there is no monitor, + * It should not start audio. + */ + if (cmd == SNDRV_PCM_TRIGGER_START) { + dai_map = &hdmi->dai_map[dai->id]; + port = dai_map->port; + + if (!port) + return -ENODEV; + + if ((!port->eld.monitor_present) || (!port->eld.eld_valid)) { + dev_err(&hdev->dev, + "device is not configured for this pin:port%d:%d\n", + port->pin->nid, port->id); + return -ENODEV; + } + } + + return 0; +} + static int hdac_hdmi_query_cvt_params(struct hdac_device *hdev, struct hdac_hdmi_cvt *cvt) { @@ -1389,6 +1406,7 @@ static const struct snd_soc_dai_ops hdmi_dai_ops = { .startup = hdac_hdmi_pcm_open, .shutdown = hdac_hdmi_pcm_close, .hw_params = hdac_hdmi_set_hw_params, + .trigger = hdac_hdmi_pcm_trigger, .set_tdm_slot = hdac_hdmi_set_tdm_slot, }; -- 2.7.4

6 17

[alsa-devel] ALSA -> pulseaudio channel mapping, what is it (and why?)
by frederik＠ofb.net 07 May '19

07 May '19

Dear ALSA developers, I ran into a problem which seems to be caused by some lines in "alsa-plugins-1.1.8/pulse/pcm_pulse.c". The problem is that if I create a 6-channel PulseAudio device, and then I configure an ALSA device in ~/.asoundrc to output to the PulseAudio device, the channels of the ALSA device are mixed together in various ways and don't correspond 1-1 with the channels of the PulseAudio device. I find this to be counter-intuitive. If I wanted channels to be mixed and remapped in various ways, I would use the PulseAudio module module-remap-sink on the PulseAudio side, since AFAICT ALSA does not support named channels. I would not expect remixing to happen in the ALSA->Pulse bridge, since the relevant PCM plugin contains no channel map parameter. I don't even know what channel map I should give the PulseAudio device in order to prevent remixing from occurring. The file pcm_pulse.c contains the lines: for (c = pcm->ss.channels; c > 0; c--) if (pa_channel_map_init_auto(&map, c, PA_CHANNEL_MAP_ALSA)) break; This refers to a macro in /usr/include/pulse/channelmap.h: /** \cond fulldocs */ PA_CHANNEL_MAP_ALSA, /**< The default mapping used by ALSA. This mapping is probably * not too useful since ALSA's default channel mapping depends on * the device string used. */ /** \endcond */ The comment says that the ALSA mapping is "not too useful" - so why are we using it? Also, how do I find out what it is, if it "depends on the device string used"? My preference would simply be to have a 1-1 channel map in pcm_pulse, so that a PulseAudio device with 6 channels will turn into an ALSA device with THE SAME 6 CHANNELS. No mixing. Mixing is confusing in this context, and seems undocumented outside of the code. I tried using aux0 thru aux5 as the PulseAudio channel map, to prevent geometry-based remixing. However, this results in silence - apparently none of the 6 ALSA channels end up going anywhere. That was also unexpected. Since I am playing 6-channel audio to a 6-channel device, which emulates another 6-channel device, I would have thought that the channel mapping wouldn't be complicated. Instead, it appears that the PulseAudio sink needs to have channels with specific names in order to receive data from ALSA. Can we fix this, or improve the user experience? Thanks, Frederick

2 2

[alsa-devel] [PATCH v2 1/2] ALSA: hda/hdmi - Read the pin sense from register when repolling
by Hui Wang 07 May '19

07 May '19

The driver will check the monitor presence when resuming from suspend, starting poll or interrupt triggers. In these 3 situations, the jack_dirty will be set to 1 first, then the hda_jack.c reads the pin_sense from register, after reading the register, the jack_dirty will be set to 0. But hdmi_repoll_work() is enabled in these 3 situations, It will read the pin_sense a couple of times subsequently, since the jack_dirty is 0 now, It does not read the register anymore, instead it uses the shadow pin_sense which is read at the first time. It is meaningless to check the shadow pin_sense a couple of times, we need to read the register to check the real plugging state, so we set the jack_dirty to 1 in the hdmi_repoll_work(). Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8b3ac690efa3..410d94301941 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1663,6 +1663,11 @@ static void hdmi_repoll_eld(struct work_struct *work) container_of(to_delayed_work(work), struct hdmi_spec_per_pin, work); struct hda_codec *codec = per_pin->codec; struct hdmi_spec *spec = codec->spec; + struct hda_jack_tbl *jack; + + jack = snd_hda_jack_tbl_get(codec, per_pin->pin_nid); + if (jack) + jack = jack->jack_dirty = 1; if (per_pin->repoll_count++ > 6) per_pin->repoll_count = 0; -- 2.17.1

2 4

[alsa-devel] alsa-utils: axfer/test: fix compiler warnings due to uninitialized return value
by GitHub pull_request - opened 07 May '19

07 May '19

alsa-project/alsa-utils pull request #18 was opened from takaswie: Hi, One of test programs for axfer internals has bugs to let compiler generate warnings due to uninitialized return value. This patchset fixes them. ``` Takashi Sakamoto (2): axfer/test: fix uninitialized warning axfer/test: fix uninitialized warning axfer/test/mapper-test.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/18 Patch URL : https://github.com/alsa-project/alsa-utils/pull/18.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH] ASoC: Intel: bytcr_5640.c:Refactored if statement and removed buffer
by Nariman 06 May '19

06 May '19

From: Nariman Etemadi <narimantos(a)gmail.com> in function snd_byt_rt5640_mc_probe and removed buffer yt_rt5640_codec_aif_name & byt_rt5640_cpu_dai_name Signed-off-by: Nariman Etemadi <narimantos(a)gmail.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 26 ++++---------------------- 1 file changed, 4 insertions(+), 22 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 940eb27158da..0d91642ca287 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1075,8 +1075,6 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { /* SoC card */ static char byt_rt5640_codec_name[SND_ACPI_I2C_ID_LEN]; -static char byt_rt5640_codec_aif_name[12]; /* = "rt5640-aif[1|2]" */ -static char byt_rt5640_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ static char byt_rt5640_long_name[40]; /* = "bytcr-rt5640-*-spk-*-mic" */ static int byt_rt5640_suspend(struct snd_soc_card *card) @@ -1268,28 +1266,12 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) log_quirks(&pdev->dev); if ((byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) || - (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { - - /* fixup codec aif name */ - snprintf(byt_rt5640_codec_aif_name, - sizeof(byt_rt5640_codec_aif_name), - "%s", "rt5640-aif2"); - - byt_rt5640_dais[dai_index].codec_dai_name = - byt_rt5640_codec_aif_name; - } + (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) + byt_rt5640_dais[dai_index].codec_dai_name = "rt5640-aif2"; if ((byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) || - (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { - - /* fixup cpu dai name name */ - snprintf(byt_rt5640_cpu_dai_name, - sizeof(byt_rt5640_cpu_dai_name), - "%s", "ssp0-port"); - - byt_rt5640_dais[dai_index].cpu_dai_name = - byt_rt5640_cpu_dai_name; - } + (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) + byt_rt5640_dais[dai_index].cpu_dai_name = "ssp0-port"; if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) { priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); -- 2.20.1

4 12

Re: [alsa-devel] [PATCH v2 2/2] regmap: soundwire: fix Kconfig select/depend issue
by Pierre-Louis Bossart 06 May '19

06 May '19

On 5/5/19 11:40 PM, Mark Brown wrote: > On Fri, May 03, 2019 at 09:32:53AM -0500, Pierre-Louis Bossart wrote: > >> As I mentioned it'll compile the bus even if there is no user for it, but >> it's your call: alignment or optimization. > > You can have both. Alignment is a requirement. If you want to optimize > this then it'd be better to optimize all the bus types rather than just > having the one weird bus type that does something different for no > documented reason. Fine, I'll align if this is the requirement. Thanks for the feedback.

1 0

[alsa-devel] [PATCH] ASoC: max98090: Fix restore of DAPM Muxes
by Jon Hunter 06 May '19

06 May '19

The max98090 driver defines 3 DAPM muxes; one for the right line output (LINMOD Mux), one for the left headphone mixer source (MIXHPLSEL Mux) and one for the right headphone mixer source (MIXHPRSEL Mux). The same bit is used for the mux as well as the DAPM enable, and although the mux can be correctly configured, after playback has completed, the mux will be reset during the disable phase. This is preventing the state of these muxes from being saved and restored correctly on system reboot. Fix this by marking these muxes as SND_SOC_NOPM. Note this has been verified this on the Tegra124 Nyan Big which features the MAX98090 codec. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- If this is the correct fix for this issue, can we tag this for stable? Or I can re-send with the stable tag. However, wanted to send out for review first. sound/soc/codecs/max98090.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 30c242c38d99..7619ea31ab50 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1194,14 +1194,14 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { &max98090_right_rcv_mixer_controls[0], ARRAY_SIZE(max98090_right_rcv_mixer_controls)), - SND_SOC_DAPM_MUX("LINMOD Mux", M98090_REG_LOUTR_MIXER, - M98090_LINMOD_SHIFT, 0, &max98090_linmod_mux), + SND_SOC_DAPM_MUX("LINMOD Mux", SND_SOC_NOPM, 0, 0, + &max98090_linmod_mux), - SND_SOC_DAPM_MUX("MIXHPLSEL Mux", M98090_REG_HP_CONTROL, - M98090_MIXHPLSEL_SHIFT, 0, &max98090_mixhplsel_mux), + SND_SOC_DAPM_MUX("MIXHPLSEL Mux", SND_SOC_NOPM, 0, 0, + &max98090_mixhplsel_mux), - SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL, - M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), + SND_SOC_DAPM_MUX("MIXHPRSEL Mux", SND_SOC_NOPM, 0, 0, + &max98090_mixhprsel_mux), SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE, M98090_HPLEN_SHIFT, 0, NULL, 0), -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: RT5677-SPI: Disable 16Bit SPI Transfers
by Curtis Malainey 06 May '19

06 May '19

The current algorithm allows 3 types of transfers, 16bit, 32bit and burst. According to Realtek, 16bit transfers have a special restriction in that it is restricted to the memory region of 0x18020000 ~ 0x18021000. This region is the memory location of the I2C registers. The current algorithm does not uphold this restriction and therefore fails to complete writes. Since this has been broken for some time it likely no one is using it. Better to simply disable the 16 bit writes. This will allow users to properly load firmware over SPI without data corruption. CC: Ben Zhang <benzh(a)chromium.org> CC: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677-spi.c | 35 ++++++++++++++++------------------- 1 file changed, 16 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 167a02773a0b..84b6bd8b50e1 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -58,13 +58,15 @@ static DEFINE_MUTEX(spi_mutex); * RT5677_SPI_READ/WRITE_32: Transfer 4 bytes * RT5677_SPI_READ/WRITE_BURST: Transfer any multiples of 8 bytes * - * For example, reading 260 bytes at 0x60030002 uses the following commands: - * 0x60030002 RT5677_SPI_READ_16 2 bytes + * Note: + * 16 Bit writes and reads are restricted to the address range + * 0x18020000 ~ 0x18021000 + * + * For example, reading 256 bytes at 0x60030004 uses the following commands: * 0x60030004 RT5677_SPI_READ_32 4 bytes * 0x60030008 RT5677_SPI_READ_BURST 240 bytes * 0x600300F8 RT5677_SPI_READ_BURST 8 bytes * 0x60030100 RT5677_SPI_READ_32 4 bytes - * 0x60030104 RT5677_SPI_READ_16 2 bytes * * Input: * @read: true for read commands; false for write commands @@ -79,15 +81,13 @@ static u8 rt5677_spi_select_cmd(bool read, u32 align, u32 remain, u32 *len) { u8 cmd; - if (align == 2 || align == 6 || remain == 2) { - cmd = RT5677_SPI_READ_16; - *len = 2; - } else if (align == 4 || remain <= 6) { + if (align == 4 || remain <= 4) { cmd = RT5677_SPI_READ_32; *len = 4; } else { cmd = RT5677_SPI_READ_BURST; - *len = min_t(u32, remain & ~7, RT5677_SPI_BURST_LEN); + *len = (((remain - 1) >> 3) + 1) << 3; + *len = min_t(u32, *len, RT5677_SPI_BURST_LEN); } return read ? cmd : cmd + 1; } @@ -108,7 +108,7 @@ static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen) } } -/* Read DSP address space using SPI. addr and len have to be 2-byte aligned. */ +/* Read DSP address space using SPI. addr and len have to be 4-byte aligned. */ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) { u32 offset; @@ -124,7 +124,7 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) if (!g_spi) return -ENODEV; - if ((addr & 1) || (len & 1)) { + if ((addr & 3) || (len & 3)) { dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len); return -EACCES; } @@ -159,13 +159,13 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) } EXPORT_SYMBOL_GPL(rt5677_spi_read); -/* Write DSP address space using SPI. addr has to be 2-byte aligned. - * If len is not 2-byte aligned, an extra byte of zero is written at the end +/* Write DSP address space using SPI. addr has to be 4-byte aligned. + * If len is not 4-byte aligned, then extra zeros are written at the end * as padding. */ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) { - u32 offset, len_with_pad = len; + u32 offset; int status = 0; struct spi_transfer t; struct spi_message m; @@ -178,22 +178,19 @@ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) if (!g_spi) return -ENODEV; - if (addr & 1) { + if (addr & 3) { dev_err(&g_spi->dev, "Bad write align 0x%x(%zu)\n", addr, len); return -EACCES; } - if (len & 1) - len_with_pad = len + 1; - memset(&t, 0, sizeof(t)); t.tx_buf = buf; t.speed_hz = RT5677_SPI_FREQ; spi_message_init_with_transfers(&m, &t, 1); - for (offset = 0; offset < len_with_pad;) { + for (offset = 0; offset < len;) { spi_cmd = rt5677_spi_select_cmd(false, (addr + offset) & 7, - len_with_pad - offset, &t.len); + len - offset, &t.len); /* Construct SPI message header */ buf[0] = spi_cmd; -- 2.21.0.1020.gf2820cf01a-goog

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[alsa-devel] [GIT PULL] ASoC updates for v5.2
by Mark Brown 06 May '19

06 May '19

The following changes since commit e93c9c99a629c61837d5a7fc2120cd2b6c70dbdd: Linux 5.1 (2019-05-05 17:42:58 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.2 for you to fetch changes up to 378d590c494551a68a824b939c711bb9a280e9ef: Merge branch 'asoc-5.2' into asoc-next (2019-05-06 22:51:54 +0900) ---------------------------------------------------------------- ASoC: Updates for v5.2 This is a pretty huge set of changes, it's been a pretty active release all round but the big thing with this release is the Sound Open Firmware changes from Intel, providing another DSP framework for use with the DSPs in their SoCs. This one works with the firmware of the same name which is free software (unlike the previous DSP firmwares and framework) and there has been some interest in adoption by other systems already so hopefully we will see adoption by other vendors in the future. Other highlights incldue: - Support for MCLK/sample rate ratio setting in the generic cards. - Support for pin switches in the generic cards. - A big set of improvements to the TLV320AIC32x4 drivers from Annaliese McDermond. - New drivers for Freescale audio mixers, several Intel machines, several Mediatek machines, Meson G12A, Sound Open Firmware and Spreadtrum compressed audio and DMA devices. ---------------------------------------------------------------- Adam Thomson (4): ASoC: da7219: Update DAI clock binding info to cover WCLK/BCLK ASoC: da7219: Expose BCLK and WCLK control through CCF ASoC: AMD: Update DA7219 DAI clock name to align with codec updates ASoC: da7219: Use clk_round_rate to handle enabled bclk/wclk case Aditya Pakki (1): ASoC: sirf-audio: Remove redundant of_match_node call Agrawal, Akshu (1): ASoC: AMD: Configure wclk and bclk of master codec Anders Roxell (1): ASoC: Mediatek: MT8183: TS3A227: fix build error Andi Kleen (1): ASoC: AMD: Fix incorrect extern Andra Danciu (21): ASoC: imx-pcm: Switch to SPDX identifier ASoC: eukrea-tlv320: Switch to SPDX identifier ASoC: fsl-dma: Switch to SPDX identifier ASoC: imx-audmux: Switch to SPDX identifier ASoC: imx-es8328: Switch to SPDX identifier ASoC: imx-mc13783: Switch to SPDX identifier ASoC: imx-pcm-fiq: Switch to SPDX identifier ASoC: imx-spdif: Switch to SPDX identifier ASoC: imx-ssi: Switch to SPDX identifier ASoC: mpc5200_dma: Switch to SPDX identifier ASoC: mpc5200_psc_ac97: Switch to SPDX identifier ASoC: mpc5200_psc_i2s: Switch to SPDX identifier ASoC: mpc8610_hpcd: Switch to SPDX identifier ASoC: mx27vis-aic32x4: Switch to SPDX identifier ASoC: p1022_ds: Switch to SPDX identifier ASoC: p1022_rdk: Switch to SPDX identifier ASoC: pcm030-audio-fabric: Switch to SPDX identifier ASoC: phycore-ac97: Switch to SPDX identifier ASoC: wm1133-ev1: Switch to SPDX identifier ASoC: mpc5200_psc_i2s: Fix invalid license ID ASoC: mpc5200_dma: Fix invalid license ID Andrew F. Davis (2): ASoC: tlv320aic31xx: Add headphone/headset detection ASoC: tlv320aic31xx: Add button press detection Andy Shevchenko (10): ACPI / utils: Introduce acpi_dev_get_first_match_dev() helper extcon: axp288: Convert to use acpi_dev_get_first_match_dev() gpio: merrifield: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: bytcht_da7213: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: bytcht_es8316: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: bytcr_rt5640: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: bytcr_rt5651: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: cht_bsw_rt5645: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: cht_bsw_rt5672: Convert to use acpi_dev_get_first_match_dev() ACPI / utils: Remove deprecated function since no user left Annaliese McDermond (17): ASoC: tlv320aic32x4: Break out clock setting into separate function ASoC: tlv320aic32x4: Properly Set Processing Blocks ASoC: tlv320aic32x4: Model PLL in CCF ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF ASoC: tlv320aic32x4: Model DAC/ADC dividers in CCF ASoC: tlv320aic32x4: Model BDIV divider in CCF ASoC: tlv320aic32x4: Control clock gating with CCF ASoC: tlv320aic32x4: Move aosr and dosr setting to separate functions ASoC: tlv320aic32x4: Dynamically Determine Clocking ASoC: tlv320aic32x4: Restructure set_dai_sysclk ASoC: tlv320aic32x4: Remove mclk references ASoC: tlv320aic32x4: Allow 192000 Sample Rate ASoC: tlv320aic32x4: Update copyright and use SPDX identifier ASoC: tlv320aic32x4: Add Switch for Setting Common Mode Voltage ASoC: tlv320aic32x4: Add Playback PowerTune Controls ASoC: tlv320aic32x4: Fix spacing ASoC: tlv320aic32x4: Fix potential uninitialized variable Arnaud Pouliquen (1): ASoC: stm32: sai: simplify dai driver initialisation Arnd Bergmann (6): ASoC: davinci-mcasp: Fix clang warning without CONFIG_PM ASoC: qcom: add i2c dependency for SND_SOC_SDM845 ASoC: ti: remove compat dma probing ASoC: ams-delta: remove duplicate 'const' ASoC: ti: fix davinci_mcasp_probe dependencies ASoC: imx: fix fiq dependencies Baolin Wang (5): ASoC: sprd: Add Spreadtrum audio compress offload support ASoC: sprd: Fix the smatch warning dt-bindings: ASoC: Add Spreadtrum multi-channel data transfer support ASoC: sprd: Add Spreadtrum multi-channel data transfer support ASoC: sprd: Add reserved DMA memory support Bard liao (4): ASoC: topology: create tlv before soc_tplg_init_kcontrol ASoC: topology: free stream_name of dai_drv ASoC: topology: free link string in error ASoC: Intel: add sof-rt5682 machine driver Cao Van Dong (1): ASoC: rsnd: Document r8a77470 bindings Charles Keepax (12): ASoC: wm_adsp: Don't export wm_adsp2_lock ASoC: wm_adsp: Refactor firmware status reading ASoC: wm_adsp: Factor out parsing of firmware ID header ASoC: wm_adsp: Factor out DSP specific operations ASoC: wm_adsp: Correct some minor issues ASoC: lochnagar: Add device tree binding document ASoC: wm_adsp: Fix typo in comment ASoC: wm_adsp: Remove redundant NULL check in wm_adsp_buffer_free ASoC: wm_adsp: Make use of local variables ASoC: wm_adsp: Avoid calling snd_compr_stop_error from WDT expiry ASoC: wm_adsp: Use DSP ops pointers to stop watchdog in error handlers ASoC: wm_adsp: Correct region base typo in wm_halo_setup_algs Codrin Ciubotariu (2): ASoC: mchp-i2s-mcc: dt-bindings: add DT bindings for I2S Multi-Channel Controller ASoC: mchp-i2s-mcc: add driver for I2SC Multi-Channel Controller Colin Ian King (5): ASoC: fsl: fix spelling mistake: "missign" -> "missing" ASoC: sprd: Fix spelling mistake "faied" -> "failed" ASoC: hdmi-codec: fix spelling mistake "plalform" -> "platform" ASoC: SOF: Intel: fix spelling mistake "incompatble" -> "incompatible" ASoC: SOF: remove redundant null checks of dai Daniel Baluta (2): ASoC: fsl_sai: Update is_slave_mode with correct value ASoC: fsl_sai: Add support for runtime pm Fabien Dessenne (1): ASoC: stm32: i2s: return the get_irq error Guennadi Liakhovetski (1): ASoC: SOF: Intel: Add legacy IPC support Hans de Goede (2): ASoC: rt5651: Add support for active-high jack detect ASoC: Intel: bytcr_rt5651: Add BYT_RT5651_JD_NOT_INV quirk Hui Wang (1): ASoC: rt5645: Headphone Jack sense inverts on the LattePanda board Jenny TC (1): ASoC: Intel: kbl_da7219_max98927: remove headset Jerome Brunet (11): ASoC: max98357a: add missing supported rates ASoC: meson: add g12a compatibles ASoC: meson: axg-fifo: add g12a support ASoC: meson: axg-toddr: add g12a support ASoC: meson: axg-frddr: add g12a support ASoC: meson: axg-tdm-formatter: rework quirks settings ASoC: meson: axg-tdmout: add g12a support ASoC: hdmi-codec: unlock the device on startup errors ASoC: fix valid stream condition ASoC: skip hw_free on codec dai for which the stream is invalid ASoC: hdmi-codec: stream is already locked in hw_params Jiada Wang (1): ASoC: rsnd: src: fix compiler warnings John Hsu (4): ASoC: nau8810: use 64-bit arithmetic instead of 32-bit ASoC: nau8810: fix the typo of function name ASoC: nau8810: automatic selecting BCLK in I2S master mode ASoC: nau8810: fix the issue of 64 bits division KaiChieh Chuang (2): ASoC: dpcm: prevent snd_soc_dpcm use after free ASoC: mediatek: btcvsd fix irq mask Kangjie Lu (2): ASoC: rt5645: fix a NULL pointer dereference ASoC: cs43130: fix a NULL pointer dereference Keyon Jie (9): ASoC: SOF: Intel: Add hda-bus support and initialization ASoC: SOF: Intel: add SKL+ platform DAIs ASoC: SOF: Intel: cnl: add pointer ops to use DPIB position ASoC: SOF: PCM: add period_elapsed work to fix race condition in interrupt context ASoC: SOF: Intel: use snd_sof_pcm_period_elapsed ASoC: SOF: ipc: use snd_sof_pcm_period_elapsed ASoC: SOF: Intel: hda-ipc: simplify handling of IPC IRQ ASoC: SOF: Intel: hda-stream: store stream capabilities ASoC: SOF: Intel: hda-stream: handle real stream interrupts only Kuninori Morimoto (18): ASoC: rsnd: add missing reg/reg-name at Document ASoC: rsnd: explain extended Audio DMAC peri peri address ASoC: simple-card: rename simple_get_conversion() to simple_parse_convert() ASoC: simple-card: add simple_parse_mclk_fs() ASoC: audio-graph: rename graph_get_conversion() to graph_parse_convert() ASoC: audio-graph: add graph_parse_mclk_fs() ASoC: simple-card-utils: add asoc_simple_debug_info() ASoC: simple_card_utils: share common priv for simple-card/audio-graph ASoC: simple-card-utils: share asoc_simple_startup() ASoC: simple-card-utils: share asoc_simple_shutdown() ASoC: simple-card-utils: share asoc_simple_hw_param() ASoC: simple-card-utils: share asoc_simple_dai_init() ASoC: simple-card-utils: share asoc_simple_be_hw_params_fixup() ASoC: simple-card-utils: share asoc_simple_card_init_priv() ASoC: simple-card-utils: separate asoc_simple_card_parse_dai() ASoC: simple-card-utils: rename asoc_simple_card_xxx() to asoc_simple_() ASoC: simple-card-utils: test memory allocation ASoC: rsnd: fixup 6ch settings to 8ch Liam Girdwood (28): ASoC: topology: Align tplg pointer increment across all kcontrols ASoC: SOF: Add Sound Open Firmware driver core ASoC: SOF: Add Sound Open Firmware KControl support ASoC: SOF: Add driver debug support. ASoC: SOF: Add support for IPC IO between DSP and Host ASoC: SOF: Add PCM operations support ASoC: SOF: Add support for loading topologies ASoC: SOF: Add DSP firmware logger support ASoC: SOF: Add DSP HW abstraction operations ASoC: SOF: Add firmware loader support ASoC: SOF: Add userspace ABI support ASoC: SOF: Add PM support ASoC: SOF: Add Nocodec machine driver support ASoC: SOF: Intel: Add BYT, CHT and BSW DSP HW support. ASoC: SOF: Intel: Add BDW HW DSP support ASoC: SOF: Intel: Add APL/CNL HW DSP support ASoC: SOF: Intel: Add HDA controller for Intel DSP ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations ASoC: SOF: Intel: Add Intel specific HDA IPC mechanisms. ASoC: SOF: Add ACPI device support ASoC: SOF: Add PCI device support ASoC: SOF: Intel: Add Intel specific HDA firmware loader ASoC: SOF: Intel: Add Intel specific HDA PCM operations ASoC: SOF: Intel: Add Intel specific HDA stream operations ASoC: SOF: Intel: Add Intel specific HDA trace operations ASoC: SOF: Intel: Add platform differentiation for APL and CNL ASoC: SOF: Add Build support for SOF core and Intel drivers ASoC: Intel: Make sure BDW based machine drivers build for SOF Libin Yang (2): ASoC: pcm: save fixed-up hw_params of BE ASoC: codec: hdac_hdmi add device_link to card device Logesh (1): ASoC: da7213: fix DAI_CLK_EN register bit overwrite Luca Ceresoli (2): ASoC: axi-i2s: let both capture and playback be optional ASoC: axi-i2s: make both "rx" and "tx" optional Mac Chiang (1): ASoC: Intel: kbl_da7219_max98357a: Map BTN_0 to KEY_PLAYPAUSE Mark Brown (11): Merge tag 'v5.1-rc1' into asoc-5.2 Merge branch 'for-5.1' of https://git.kernel.org/.../broonie/sound into asoc-5.2 Merge branch 'asoc-5.1' into asoc-5.2 for wmadsp ASoC: tlv320aic32x4: Only enable with common clock Merge branch 'acpi-utils' of git://git.kernel.org/.../rafael/linux-pm into asoc-5.2 ASoC: sprd: Fix modular build Merge branch 'asoc-5.1' into asoc-5.2 Merge branch 'asoc-5.1' into asoc-5.2 ASoC: Intel: Make boards more available for compile test Merge branch 'asoc-5.1' into asoc-linus Merge branch 'asoc-5.2' into asoc-next Mylène Josserand (1): ASoC: simple-amplifier: Make gpio property optional Nicolin Chen (1): ASoC: fsl_sai: Add missing return 0 in remove() Oder Chiou (1): ASoC: rt5677-spi: Add ACPI ID Olivier Moysan (14): ASoC: stm32: i2s: change trigger traces ASoC: stm32: i2s: improve channel capabilities handling ASoC: stm32: i2s: use default dai name ASoC: stm32: spdifrx: add power management ASoC: stm32: dfsdm: add mmap support ASoC: stm32: sai: add power management ASoC: cs42l51: add multi endpoint support ASoC: dt-bindings: update cs42l51 bindings ASoC: cs42l51: add regulator management ASoC: cs42l51: add reset management ASoC: cs42l51: add support of master mode ASoC: cs42l51: add power management ASoC: cs42l51: add adc volume control ASoC: cs42l51: change mic bias DAPM Pan Xiuli (3): ASoC: SOF: IPC: add ipc dump function ASoC: SOF: Intel: APL: add ipc dump function ASoC: SOF: Intel: CNL: add ipc dump function Pankaj Bharadiya (1): ASoC: dapm: Initialize private_value in snd_soc_dapm_new_dai Paul Cercueil (7): ASoC: jz4740: Depend on MIPS instead of MACH_JZ4740 ASoC: es8316: Add support for inverted jack detect ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect ASoC: Intel: bytcht_es8316: Add quirk for the Teclast X98+ II ASoC: doc: simple-card: Add pin-switches property ASoC: simple-card-utils: add asoc_simple_parse_pin_switches() ASoC: simple-card: Read pin switches conf from devicetree Peter Ujfalusi (1): ASoC: pcm3168a: Enable TDM support for DSP_A/B modes Pi-Hsun Shih (2): ASoC: mediatek: Fix UBSAN warning. ASoC: mediatek: Reduce repititive code on mtk_regmap_update_bits. Pierre-Louis Bossart (15): ASoC: core: support driver alias names for FE topology overrides ASoC: topology: fix endianness issues ASoC: topology: fix big-endian check ASoC: Intel: bytcht_es8316: fix compilation warning ASoC: Intel: bytcr_rt5640: align quirk override handling ASoC: Intel: bytcr_rt5651: align quirk override handling ASoC: SOF: Add xtensa support ASoC: SOF: Add utils ASoC: SOF: Intel: Add support for HDAudio codecs ASoC: Intel: Kconfig: expose common option between SST and SOF drivers ASoC: Intel: select relevant machine drivers for SOF ASoC: SOF: add Kconfig option for strict ABI checks ASOC: SOF: ipc: add support for stricter ABI checks ASoC: SOF: topology: add support for stricter ABI checks ASoC: SOF: Intel: hda-pcm: remove useless dependency on hdac_ext Piotr Stankiewicz (1): ASoC: lochnagar: Add driver to support Lochnagar 2 sound card Rander Wang (2): ASoC: Intel: skl_hda_dsp_generic: add DMIC support ASoC: Intel: skl_hda_dsp_generic:refine code style Randy Dunlap (1): ASoC: fix SND_SOC_LOCHNAGAR_SC kconfig warning Ranjani Sridharan (2): ASoC: core: remove link components before cleaning up card resources ASoC: SOF: intel: hda: add hw_params_upon_resume flag for hda stream Richard Fitzgerald (2): ASoC: wm_adsp: Move wm_adsp2_set_dspclk to CODEC drivers ASoC: wm_adsp: Add HALO MPU fault logging Ross Zwisler (1): ASoC: Intel: avoid Oops if DMA setup fails S.j. Wang (2): ASoC: fsl_esai: Support synchronous mode ASoC: fsl_esai: Fix missing break in switch statement Seppo Ingalsuo (1): ASoC: dapm: fix kcontrols for effect widgets Shunli Wang (5): ASoC: Mediatek: MT8183: Add machine driver with TS3A227 ASoC: Mediatek: Add document for mt8183-mt6358-ts3a227-max98357 ASoC: Mediatek: MT8183: Add machine driver with DA7219 ASoC: Mediatek: Add document for mt8183-da7219-max98357 ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3 Stuart Henderson (1): ASoC: wm_adsp: Add WDT expiry IRQ handling support for Halo core Sugar Zhang (10): ASoC: rockchip: pdm: using left justified store mode ASoC: rockchip: pdm: add default regs ASoC: rockchip: pdm: optimize clear logic ASoC: rockchip: pdm: fixup pdm fractional div dt-bindings: sound: rockchip: add compatible for rk3308/px30 ASoC: rockchip: pdm: add compatible for rk1808 dt-bindings: sound: add compatible for rk1808 ASoC: rockchip: pdm: adjust waterlevel in frame unit ASoC: rockchip: pdm: Correct PDM_CTRL0 reg value ASoC: rockchip: pdm: Mark RXFIFO_DATA as volatile and precious Sylwester Nawrocki (28): ASoC: samsung: i2s: Don't use register read to determine slave/master mode ASoC: samsung: arndale: Convert to SPDX License Identifier ASoC: samsung: bells: Convert to SPDX License Identifier ASoC: samsung: dmaengine: Convert to SPDX License Indentifier ASoC: samsung: h1940_uda1380: Convert to SPDX License Identifier ASoC: samsung: i2s: Convert to SPDX License Identifier ASoC: samsung: idma: Convert to SPDX License Identifier ASoC: samsung: jive_wm8750: Convert to SPDX License Identifier ASoC: samsung: littlemill: Convert to SPDX License Identifier ASoC: samsung: lowland: Convert to SPDX License Identifier ASoC: samsung: neo1973_wm8753: Convert to SPDX License Identifier ASoC: samsung: odroid: Convert to SPDX License Identifier ASoC: samsung: pcm: Convert to SPDX License Identifier ASoC: samsung: s3c-i2s-v2: Convert to SPDX License Identifier ASoC: samsung: s3c24xx-i2s: Convert to SPDX License Identifier ASoC: samsung: rx1950_uda1380: Convert to SPDX License Identifier ASoC: samsung: s3c2412-i2s: Convert to SPDX License Identifier ASoC: samsung: s3c24xx_simtec*: Convert to SPDX License Identifier ASoC: samsung: s3c24xx_uda134x: Convert to SPDX License Identifier ASoC: samsung: smartq_wm8987: Convert to SPDX License Identifier ASoC: samsung: smdk_spdif: Convert to SPDX License Identifier ASoC: samsung: smdk_wm8580: Convert to SPDX License Identifier ASoC: samsung: smdk_wm8994*: Convert to SPDX License Identifier ASoC: samsung: snow: Convert to SPDX License Identifier ASoC: samsung: spdif: Convert to SPDX License Identifier ASoC: samsung: speyside: Convert to SPDX License Identifier ASoC: samsung: tm2_wm5110: Convert to SPDX License Identifier ASoC: samsung: tobermory: Convert to SPDX License Identifier Tzung-Bi Shih (1): ASoC: Mediatek: MT8183: change supported formats of DL2 and UL1 Viorel Suman (8): ASoC: fsl: Add Audio Mixer CPU DAI driver ASoC: add fsl_audmix DT binding documentation ASoC: fsl: Add Audio Mixer machine driver ASoC: fsl_audmix: Fix kbuild failure ASoC: imx-audmix: fix object reference leaks in probe ASoC: fsl_audmix: remove "model" attribute from DT document ASoC: fsl_audmix: remove "model" attribute ASoC: fsl_audmix: cache pdev->dev pointer Wei Yongjun (2): ASoC: sprd: Fix return value check in sprd_mcdt_probe() ASoC: sprd: Fix to use list_for_each_entry_safe() when delete items Wen Shi (1): ASoC: wm_adsp: Add support for new Halo core DSPs Wen Yang (3): ASoC: wcd9335: fix a leaked reference by adding missing of_node_put ASoC: fsl_utils: fix a leaked reference by adding missing of_node_put ASoC: eukrea-tlv320: fix a leaked reference by adding missing of_node_put Xun Zhang (1): ASoC: codecs: rt5682: initialize mutex before using Yong Zhi (1): ASoC: Intel: common: add Geminilake Realtek+Maxim machine driver entry Yu-Hsuan Hsu (1): ASoC: da7219: Update the support rate list YueHaibing (10): ASoC: simple-card-utils: remove set but not used variable 'dai_name' ASoC: Mediatek: MT8183: Fix build error in mt8183_da7219_max98357_dev_probe ASoC: Mediatek: MT8183: Fix platform_no_drv_owner.cocci warnings ASoC: Mediatek: MT8183: Fix platform_no_drv_owner.cocci warnings ASoC: Intel: Haswell: Remove set but not used variable 'stage_type' ASoC: Mediatek: MT8183: Fix build err while CONFIG_I2C set to module ASoC: fsl_micfil: Remove set but not used variable 'osr' ASoC: tlv320aic32x4: Remove set but not used variable 'mclk_rate' ASoC: amd: acp3x: Make acp3x_dai_i2s_ops static ASoC: atmel: tse850: Make some functions static Zhu Yingjiang (4): ASoC: SOF: Intel: hda: add the SSP Host Device memory space ASoC: SOF: Intel: hda: add SSP info to the chip info struct ASoC: SOF: Intel: hda: set I2S slave before enabling DSP ASoC: SOF: Intel: hda: set bus->idx as 0 kbuild test robot (1): ASoC: tlv320aic31xx: aic31xx_set_jack() can be static .../devicetree/bindings/sound/adi,axi-i2s.txt | 7 +- .../devicetree/bindings/sound/amlogic,axg-fifo.txt | 4 +- .../devicetree/bindings/sound/amlogic,axg-pdm.txt | 3 +- .../bindings/sound/amlogic,axg-spdifin.txt | 3 +- .../bindings/sound/amlogic,axg-spdifout.txt | 3 +- .../bindings/sound/amlogic,axg-tdm-formatters.txt | 4 +- .../devicetree/bindings/sound/cirrus,lochnagar.txt | 39 + .../devicetree/bindings/sound/cs42l51.txt | 16 + Documentation/devicetree/bindings/sound/da7219.txt | 8 +- .../devicetree/bindings/sound/fsl,audmix.txt | 50 + .../devicetree/bindings/sound/mchp-i2s-mcc.txt | 43 + .../bindings/sound/mt8183-da7219-max98357.txt | 15 + .../sound/mt8183-mt6358-ts3a227-max98357.txt | 15 + .../devicetree/bindings/sound/renesas,rsnd.txt | 8 +- .../devicetree/bindings/sound/rockchip,pdm.txt | 5 + Documentation/devicetree/bindings/sound/rt5651.txt | 5 + .../devicetree/bindings/sound/simple-amplifier.txt | 2 +- .../devicetree/bindings/sound/simple-card.txt | 2 + .../devicetree/bindings/sound/sprd-mcdt.txt | 19 + MAINTAINERS | 2 + drivers/acpi/utils.c | 16 +- drivers/extcon/extcon-axp288.c | 9 +- drivers/gpio/gpio-merrifield.c | 18 +- include/acpi/acpi_bus.h | 4 +- include/linux/acpi.h | 4 +- include/sound/da7219.h | 8 +- include/sound/simple_card_utils.h | 238 +- include/sound/sof.h | 100 + include/sound/sof/control.h | 158 + include/sound/sof/dai-intel.h | 178 ++ include/sound/sof/dai.h | 75 + include/sound/sof/header.h | 158 + include/sound/sof/info.h | 118 + include/sound/sof/pm.h | 48 + include/sound/sof/stream.h | 148 + include/sound/sof/topology.h | 256 ++ include/sound/sof/trace.h | 67 + include/sound/sof/xtensa.h | 44 + include/uapi/sound/sof/abi.h | 62 + include/uapi/sound/sof/eq.h | 172 ++ include/uapi/sound/sof/fw.h | 78 + include/uapi/sound/sof/header.h | 27 + include/uapi/sound/sof/manifest.h | 188 ++ include/uapi/sound/sof/tokens.h | 107 + include/uapi/sound/sof/tone.h | 21 + include/uapi/sound/sof/trace.h | 66 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/adi/axi-i2s.c | 68 +- sound/soc/amd/acp-da7219-max98357a.c | 68 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- sound/soc/atmel/Kconfig | 14 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/mchp-i2s-mcc.c | 974 ++++++ sound/soc/atmel/tse850-pcm5142.c | 16 +- sound/soc/codecs/Kconfig | 15 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/cs42l51-i2c.c | 13 +- sound/soc/codecs/cs42l51.c | 225 +- sound/soc/codecs/cs42l51.h | 3 + sound/soc/codecs/cs43130.c | 2 + sound/soc/codecs/cs47l24.c | 4 +- sound/soc/codecs/da7213.c | 5 +- sound/soc/codecs/da7213.h | 2 + sound/soc/codecs/da7219.c | 540 +++- sound/soc/codecs/da7219.h | 6 +- sound/soc/codecs/es8316.c | 14 + sound/soc/codecs/hdac_hdmi.c | 11 + sound/soc/codecs/hdmi-codec.c | 12 +- sound/soc/codecs/lochnagar-sc.c | 266 ++ sound/soc/codecs/max98357a.c | 3 + sound/soc/codecs/nau8810.c | 25 +- sound/soc/codecs/pcm3168a.c | 7 +- sound/soc/codecs/rt5645.c | 17 + sound/soc/codecs/rt5651.c | 47 +- sound/soc/codecs/rt5651.h | 1 + sound/soc/codecs/rt5677-spi.c | 8 + sound/soc/codecs/rt5682.c | 2 +- sound/soc/codecs/simple-amplifier.c | 3 +- sound/soc/codecs/sirf-audio-codec.c | 3 - sound/soc/codecs/tlv320aic31xx.c | 61 +- sound/soc/codecs/tlv320aic31xx.h | 12 + sound/soc/codecs/tlv320aic32x4-clk.c | 483 +++ sound/soc/codecs/tlv320aic32x4-i2c.c | 14 +- sound/soc/codecs/tlv320aic32x4-spi.c | 14 +- sound/soc/codecs/tlv320aic32x4.c | 441 +-- sound/soc/codecs/tlv320aic32x4.h | 13 + sound/soc/codecs/wcd9335.c | 1 + sound/soc/codecs/wm5102.c | 4 +- sound/soc/codecs/wm5110.c | 4 +- sound/soc/codecs/wm_adsp.c | 1032 +++++-- sound/soc/codecs/wm_adsp.h | 50 +- sound/soc/codecs/wmfw.h | 30 + sound/soc/fsl/Kconfig | 25 +- sound/soc/fsl/Makefile | 5 + sound/soc/fsl/eukrea-tlv320.c | 30 +- sound/soc/fsl/fsl_audmix.c | 578 ++++ sound/soc/fsl/fsl_audmix.h | 102 + sound/soc/fsl/fsl_dma.c | 26 +- sound/soc/fsl/fsl_dma.h | 5 +- sound/soc/fsl/fsl_esai.c | 23 +- sound/soc/fsl/fsl_micfil.c | 3 - sound/soc/fsl/fsl_sai.c | 26 +- sound/soc/fsl/fsl_utils.c | 1 + sound/soc/fsl/imx-audmix.c | 331 ++ sound/soc/fsl/imx-audmux.c | 26 +- sound/soc/fsl/imx-es8328.c | 15 +- sound/soc/fsl/imx-mc13783.c | 22 +- sound/soc/fsl/imx-pcm-fiq.c | 21 +- sound/soc/fsl/imx-pcm.h | 6 +- sound/soc/fsl/imx-spdif.c | 13 +- sound/soc/fsl/imx-ssi.c | 57 +- sound/soc/fsl/imx-ssi.h | 6 +- sound/soc/fsl/mpc5200_dma.c | 14 +- sound/soc/fsl/mpc5200_psc_ac97.c | 16 +- sound/soc/fsl/mpc5200_psc_i2s.c | 14 +- sound/soc/fsl/mpc8610_hpcd.c | 18 +- sound/soc/fsl/mx27vis-aic32x4.c | 29 +- sound/soc/fsl/p1022_ds.c | 18 +- sound/soc/fsl/p1022_rdk.c | 32 +- sound/soc/fsl/pcm030-audio-fabric.c | 18 +- sound/soc/fsl/phycore-ac97.c | 16 +- sound/soc/fsl/wm1133-ev1.c | 21 +- sound/soc/generic/audio-graph-card.c | 429 ++- sound/soc/generic/simple-card-utils.c | 440 +-- sound/soc/generic/simple-card.c | 383 +-- sound/soc/intel/Kconfig | 7 +- sound/soc/intel/boards/Kconfig | 117 +- sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5677.c | 4 + sound/soc/intel/boards/broadwell.c | 4 + sound/soc/intel/boards/bytcht_da7213.c | 9 +- sound/soc/intel/boards/bytcht_es8316.c | 51 +- sound/soc/intel/boards/bytcr_rt5640.c | 16 +- sound/soc/intel/boards/bytcr_rt5651.c | 42 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 9 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 9 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 52 +- sound/soc/intel/boards/skl_hda_dsp_common.c | 22 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 19 +- sound/soc/intel/boards/sof_rt5682.c | 563 ++++ sound/soc/intel/common/soc-acpi-intel-byt-match.c | 6 + sound/soc/intel/common/soc-acpi-intel-cht-match.c | 6 + sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 19 + sound/soc/intel/common/soc-acpi-intel-glk-match.c | 9 + sound/soc/intel/common/soc-acpi-intel-icl-match.c | 6 + sound/soc/intel/common/sst-firmware.c | 8 +- sound/soc/intel/haswell/sst-haswell-ipc.c | 8 +- sound/soc/jz4740/Kconfig | 2 +- sound/soc/mediatek/Kconfig | 27 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 51 +- sound/soc/mediatek/common/mtk-btcvsd.c | 4 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 13 - sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 16 - sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 16 - sound/soc/mediatek/mt8183/Makefile | 2 + sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 18 + sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 471 +++ .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 423 +++ sound/soc/meson/axg-fifo.c | 34 +- sound/soc/meson/axg-fifo.h | 2 + sound/soc/meson/axg-frddr.c | 143 +- sound/soc/meson/axg-tdm-formatter.c | 6 +- sound/soc/meson/axg-tdm-formatter.h | 11 +- sound/soc/meson/axg-tdmin.c | 16 +- sound/soc/meson/axg-tdmout.c | 29 +- sound/soc/meson/axg-toddr.c | 53 + sound/soc/qcom/Kconfig | 2 +- sound/soc/rockchip/rockchip_pdm.c | 205 +- sound/soc/rockchip/rockchip_pdm.h | 12 + sound/soc/samsung/arndale_rt5631.c | 17 +- sound/soc/samsung/bells.c | 15 +- sound/soc/samsung/dma.h | 8 +- sound/soc/samsung/dmaengine.c | 22 +- sound/soc/samsung/h1940_uda1380.c | 22 +- sound/soc/samsung/i2s-regs.h | 8 +- sound/soc/samsung/i2s.c | 27 +- sound/soc/samsung/i2s.h | 8 +- sound/soc/samsung/idma.c | 20 +- sound/soc/samsung/idma.h | 9 +- sound/soc/samsung/jive_wm8750.c | 19 +- sound/soc/samsung/littlemill.c | 15 +- sound/soc/samsung/lowland.c | 15 +- sound/soc/samsung/neo1973_wm8753.c | 25 +- sound/soc/samsung/odroid.c | 10 +- sound/soc/samsung/pcm.c | 19 +- sound/soc/samsung/pcm.h | 8 +- sound/soc/samsung/regs-i2s-v2.h | 10 +- sound/soc/samsung/regs-iis.h | 7 +- sound/soc/samsung/rx1950_uda1380.c | 30 +- sound/soc/samsung/s3c-i2s-v2.c | 26 +- sound/soc/samsung/s3c-i2s-v2.h | 11 +- sound/soc/samsung/s3c2412-i2s.c | 28 +- sound/soc/samsung/s3c2412-i2s.h | 11 +- sound/soc/samsung/s3c24xx-i2s.c | 25 +- sound/soc/samsung/s3c24xx-i2s.h | 6 +- sound/soc/samsung/s3c24xx_simtec.c | 11 +- sound/soc/samsung/s3c24xx_simtec.h | 10 +- sound/soc/samsung/s3c24xx_simtec_hermes.c | 11 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 11 +- sound/soc/samsung/s3c24xx_uda134x.c | 20 +- sound/soc/samsung/smartq_wm8987.c | 21 +- sound/soc/samsung/smdk_spdif.c | 16 +- sound/soc/samsung/smdk_wm8580.c | 15 +- sound/soc/samsung/smdk_wm8994.c | 9 +- sound/soc/samsung/smdk_wm8994pcm.c | 16 +- sound/soc/samsung/snow.c | 15 +- sound/soc/samsung/spdif.c | 17 +- sound/soc/samsung/spdif.h | 8 +- sound/soc/samsung/speyside.c | 15 +- sound/soc/samsung/tm2_wm5110.c | 17 +- sound/soc/samsung/tobermory.c | 15 +- sound/soc/sh/rcar/core.c | 18 +- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/ssi.c | 2 + sound/soc/soc-core.c | 18 +- sound/soc/soc-dapm.c | 5 +- sound/soc/soc-pcm.c | 14 +- sound/soc/soc-topology.c | 330 +- sound/soc/sof/Kconfig | 156 + sound/soc/sof/Makefile | 18 + sound/soc/sof/control.c | 552 ++++ sound/soc/sof/core.c | 506 ++++ sound/soc/sof/debug.c | 232 ++ sound/soc/sof/intel/Kconfig | 230 ++ sound/soc/sof/intel/Makefile | 19 + sound/soc/sof/intel/apl.c | 113 + sound/soc/sof/intel/bdw.c | 713 +++++ sound/soc/sof/intel/byt.c | 874 ++++++ sound/soc/sof/intel/cnl.c | 268 ++ sound/soc/sof/intel/hda-bus.c | 111 + sound/soc/sof/intel/hda-codec.c | 171 ++ sound/soc/sof/intel/hda-ctrl.c | 181 ++ sound/soc/sof/intel/hda-dai.c | 356 +++ sound/soc/sof/intel/hda-dsp.c | 471 +++ sound/soc/sof/intel/hda-ipc.c | 455 +++ sound/soc/sof/intel/hda-loader.c | 382 +++ sound/soc/sof/intel/hda-pcm.c | 239 ++ sound/soc/sof/intel/hda-stream.c | 701 +++++ sound/soc/sof/intel/hda-trace.c | 94 + sound/soc/sof/intel/hda.c | 689 +++++ sound/soc/sof/intel/hda.h | 583 ++++ sound/soc/sof/intel/intel-ipc.c | 92 + sound/soc/sof/intel/shim.h | 185 ++ sound/soc/sof/ipc.c | 842 ++++++ sound/soc/sof/loader.c | 400 +++ sound/soc/sof/nocodec.c | 109 + sound/soc/sof/ops.c | 163 + sound/soc/sof/ops.h | 411 +++ sound/soc/sof/pcm.c | 767 +++++ sound/soc/sof/pm.c | 388 +++ sound/soc/sof/sof-acpi-dev.c | 312 ++ sound/soc/sof/sof-pci-dev.c | 373 +++ sound/soc/sof/sof-priv.h | 635 ++++ sound/soc/sof/topology.c | 3179 ++++++++++++++++++++ sound/soc/sof/trace.c | 297 ++ sound/soc/sof/utils.c | 112 + sound/soc/sof/xtensa/Kconfig | 2 + sound/soc/sof/xtensa/Makefile | 5 + sound/soc/sof/xtensa/core.c | 138 + sound/soc/sprd/Kconfig | 9 + sound/soc/sprd/Makefile | 6 +- sound/soc/sprd/sprd-mcdt.c | 1011 +++++++ sound/soc/sprd/sprd-mcdt.h | 107 + sound/soc/sprd/sprd-pcm-compress.c | 674 +++++ sound/soc/sprd/sprd-pcm-dma.c | 9 +- sound/soc/sprd/sprd-pcm-dma.h | 43 + sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/stm/stm32_i2s.c | 30 +- sound/soc/stm/stm32_sai.c | 80 +- sound/soc/stm/stm32_sai.h | 2 + sound/soc/stm/stm32_sai_sub.c | 88 +- sound/soc/stm/stm32_spdifrx.c | 42 +- sound/soc/ti/Kconfig | 4 +- sound/soc/ti/ams-delta.c | 2 +- sound/soc/ti/davinci-mcasp.c | 2 + sound/soc/ti/edma-pcm.c | 5 +- sound/soc/ti/sdma-pcm.c | 9 +- 280 files changed, 29423 insertions(+), 2743 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,lochnagar.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt create mode 100644 Documentation/devicetree/bindings/sound/mchp-i2s-mcc.txt create mode 100644 Documentation/devicetree/bindings/sound/mt8183-da7219-max98357.txt create mode 100644 Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt create mode 100644 Documentation/devicetree/bindings/sound/sprd-mcdt.txt create mode 100644 include/sound/sof.h create mode 100644 include/sound/sof/control.h create mode 100644 include/sound/sof/dai-intel.h create mode 100644 include/sound/sof/dai.h create mode 100644 include/sound/sof/header.h create mode 100644 include/sound/sof/info.h create mode 100644 include/sound/sof/pm.h create mode 100644 include/sound/sof/stream.h create mode 100644 include/sound/sof/topology.h create mode 100644 include/sound/sof/trace.h create mode 100644 include/sound/sof/xtensa.h create mode 100644 include/uapi/sound/sof/abi.h create mode 100644 include/uapi/sound/sof/eq.h create mode 100644 include/uapi/sound/sof/fw.h create mode 100644 include/uapi/sound/sof/header.h create mode 100644 include/uapi/sound/sof/manifest.h create mode 100644 include/uapi/sound/sof/tokens.h create mode 100644 include/uapi/sound/sof/tone.h create mode 100644 include/uapi/sound/sof/trace.h create mode 100644 sound/soc/atmel/mchp-i2s-mcc.c create mode 100644 sound/soc/codecs/lochnagar-sc.c create mode 100644 sound/soc/codecs/tlv320aic32x4-clk.c create mode 100644 sound/soc/fsl/fsl_audmix.c create mode 100644 sound/soc/fsl/fsl_audmix.h create mode 100644 sound/soc/fsl/imx-audmix.c create mode 100644 sound/soc/intel/boards/sof_rt5682.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c create mode 100644 sound/soc/sof/Kconfig create mode 100644 sound/soc/sof/Makefile create mode 100644 sound/soc/sof/control.c create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/debug.c create mode 100644 sound/soc/sof/intel/Kconfig create mode 100644 sound/soc/sof/intel/Makefile create mode 100644 sound/soc/sof/intel/apl.c create mode 100644 sound/soc/sof/intel/bdw.c create mode 100644 sound/soc/sof/intel/byt.c create mode 100644 sound/soc/sof/intel/cnl.c create mode 100644 sound/soc/sof/intel/hda-bus.c create mode 100644 sound/soc/sof/intel/hda-codec.c create mode 100644 sound/soc/sof/intel/hda-ctrl.c create mode 100644 sound/soc/sof/intel/hda-dai.c create mode 100644 sound/soc/sof/intel/hda-dsp.c create mode 100644 sound/soc/sof/intel/hda-ipc.c create mode 100644 sound/soc/sof/intel/hda-loader.c create mode 100644 sound/soc/sof/intel/hda-pcm.c create mode 100644 sound/soc/sof/intel/hda-stream.c create mode 100644 sound/soc/sof/intel/hda-trace.c create mode 100644 sound/soc/sof/intel/hda.c create mode 100644 sound/soc/sof/intel/hda.h create mode 100644 sound/soc/sof/intel/intel-ipc.c create mode 100644 sound/soc/sof/intel/shim.h create mode 100644 sound/soc/sof/ipc.c create mode 100644 sound/soc/sof/loader.c create mode 100644 sound/soc/sof/nocodec.c create mode 100644 sound/soc/sof/ops.c create mode 100644 sound/soc/sof/ops.h create mode 100644 sound/soc/sof/pcm.c create mode 100644 sound/soc/sof/pm.c create mode 100644 sound/soc/sof/sof-acpi-dev.c create mode 100644 sound/soc/sof/sof-pci-dev.c create mode 100644 sound/soc/sof/sof-priv.h create mode 100644 sound/soc/sof/topology.c create mode 100644 sound/soc/sof/trace.c create mode 100644 sound/soc/sof/utils.c create mode 100644 sound/soc/sof/xtensa/Kconfig create mode 100644 sound/soc/sof/xtensa/Makefile create mode 100644 sound/soc/sof/xtensa/core.c create mode 100644 sound/soc/sprd/sprd-mcdt.c create mode 100644 sound/soc/sprd/sprd-mcdt.h create mode 100644 sound/soc/sprd/sprd-pcm-compress.c

2 6

[alsa-devel] [PATCH] ALSA: synth: emux: soundfont.c: divide by zero in calc_gus_envelope_time()
by Dan Carpenter 06 May '19

06 May '19

This function is called from load_guspatch() and the rate is specified by the user. If they accidentally selected zero then it would crash the kernel. I've just changed the zero to a one. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/synth/emux/soundfont.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 31a4ea94830e..9b5d70104489 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -856,6 +856,8 @@ calc_gus_envelope_time(int rate, int start, int end) int r, p, t; r = (3 - ((rate >> 6) & 3)) * 3; p = rate & 0x3f; + if (!p) + p = 1; t = end - start; if (t < 0) t = -t; if (13 > r) -- 2.18.0

2 1

[alsa-devel] 1.1.8: compile warnings
by GitHub issues - opened 06 May '19

06 May '19

alsa-project/alsa-utils issue #17 was opened from kloczek: It would be good to clean this :) <pre>alsactl.c: In function 'main': alsactl.c:345:3: warning: ignoring return value of 'daemon', declared with attribute warn_unused_result [-Wunused-result] 345 | daemon(0, 0); | ^~~~~~~~~~~~ utils.c: In function 'initfailed': utils.c:95:2: warning: ignoring return value of 'write', declared with attribute warn_unused_result [-Wunused-result] 95 | write(fp, str, strlen(str)); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~ utils.c:96:2: warning: ignoring return value of 'write', declared with attribute warn_unused_result [-Wunused-result] 96 | write(fp, ":", 1); | ^~~~~~~~~~~~~~~~~ utils.c:97:2: warning: ignoring return value of 'write', declared with attribute warn_unused_result [-Wunused-result] 97 | write(fp, reason, strlen(reason)); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ utils.c:98:2: warning: ignoring return value of 'write', declared with attribute warn_unused_result [-Wunused-result] 98 | write(fp, ":", 1); | ^~~~~~~~~~~~~~~~~ utils.c:99:2: warning: ignoring return value of 'write', declared with attribute warn_unused_result [-Wunused-result] 99 | write(fp, sexitcode, strlen(sexitcode)); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ utils.c:100:2: warning: ignoring return value of 'write', declared with attribute warn_unused_result [-Wunused-result] 100 | write(fp, "\n", 1); | ^~~~~~~~~~~~~~~~~~ Note: Writing alsactl_init.7 mem.c: In function 'casprintf': mem.c:63:6: warning: implicit declaration of function 'vasprintf'; did you mean 'casprintf'? [-Wimplicit-function-declaration] 63 | if (vasprintf(&str, fmt, ap) < 0) | ^~~~~~~~~ | casprintf utils.c:19: warning: "_XOPEN_SOURCE" redefined 19 | #define _XOPEN_SOURCE | <command-line>: note: this is the location of the previous definition volume_mapping.c: In function 'get_normalized_volume': volume_mapping.c:116:15: warning: implicit declaration of function 'exp10' [-Wimplicit-function-declaration] 116 | normalized = exp10((value - max) / 6000.0); | ^~~~~ volume_mapping.c:116:15: warning: incompatible implicit declaration of built-in function 'exp10' volume_mapping.c: In function 'set_normalized_volume': volume_mapping.c:151:14: warning: incompatible implicit declaration of built-in function 'exp10' 151 | min_norm = exp10((min - max) / 6000.0); | ^~~~~ ../alsamixer/volume_mapping.c: In function 'get_normalized_volume': ../alsamixer/volume_mapping.c:116:15: warning: implicit declaration of function 'exp10' [-Wimplicit-function-declaration] 116 | normalized = exp10((value - max) / 6000.0); | ^~~~~ ../alsamixer/volume_mapping.c:116:15: warning: incompatible implicit declaration of built-in function 'exp10' ../alsamixer/volume_mapping.c: In function 'set_normalized_volume': ../alsamixer/volume_mapping.c:151:14: warning: incompatible implicit declaration of built-in function 'exp10' 151 | min_norm = exp10((min - max) / 6000.0); | ^~~~~ amidi.c: In function 'main': amidi.c:692:5: warning: ignoring return value of 'write', declared with attribute warn_unused_result [-Wunused-result] 692 | write(receive_file, buf, length); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ alsaloop.c: In function 'parse_config': alsaloop.c:398:6: warning: variable 'arg_effect' set but not used [-Wunused-but-set-variable] 398 | int arg_effect = 0; | ^~~~~~~~~~ aseqnet.c: In function 'flush_writebuf': aseqnet.c:495:5: warning: ignoring return value of 'write', declared with attribute warn_unused_result [-Wunused-result] 495 | write(netfd[i], writebuf, cur_wrlen); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ </pre> Issue URL : https://github.com/alsa-project/alsa-utils/issues/17 Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] 1.1.8: test suite is failing
by GitHub issues - opened 06 May '19

06 May '19

alsa-project/alsa-utils issue #16 was opened from kloczek: <pre>+ /usr/bin/make -O -j4 check Making check in include Making check in alsactl Making check in init make[2]: Nothing to be done for 'check'. make[2]: Nothing to be done for 'check-am'. Making check in utils make[1]: Nothing to be done for 'check'. Making check in m4 make[1]: Nothing to be done for 'check'. Making check in po make[1]: Nothing to be done for 'check'. Making check in alsa-info make[1]: Nothing to be done for 'check'. Making check in alsamixer make[1]: Nothing to be done for 'check'. Making check in amixer make[1]: Nothing to be done for 'check'. Making check in amidi make[1]: Nothing to be done for 'check'. Making check in aplay make[1]: Nothing to be done for 'check'. Making check in iecset make[1]: Nothing to be done for 'check'. Making check in speaker-test Making check in samples make[2]: Nothing to be done for 'check'. make[2]: Nothing to be done for 'check-am'. Making check in axfer Making check in test /usr/bin/make container-test mapper-test make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.8/axfer/test' gcc -DHAVE_CONFIG_H -I. -I../../include -O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection -flto -Os -c -o container-au.o `test -f '../container-au.c' || echo './'`../container-au.c make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.8/axfer/test' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.8/axfer/test' gcc -DHAVE_CONFIG_H -I. -I../../include -O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection -flto -Os -c -o container-riff-wave.o `test -f '../container-riff-wave.c' || echo './'`../container-riff-wave.c make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.8/axfer/test' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.8/axfer/test' gcc -DHAVE_CONFIG_H -I. -I../../include -O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection -flto -Os -c -o generator.o generator.c generator.c:9:10: fatal error: generator.h: No such file or directory 9 | #include "generator.h" | ^~~~~~~~~~~~~ compilation terminated. make[3]: *** [Makefile:640: generator.o] Error 1 make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.8/axfer/test' make[3]: *** Waiting for unfinished jobs.... make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.8/axfer/test' gcc -DHAVE_CONFIG_H -I. -I../../include -O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection -flto -Os -c -o container.o `test -f '../container.c' || echo './'`../container.c make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.8/axfer/test' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.8/axfer/test' gcc -DHAVE_CONFIG_H -I. -I../../include -O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection -flto -Os -c -o container-voc.o `test -f '../container-voc.c' || echo './'`../container-voc.c make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.8/axfer/test' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.8/axfer/test' gcc -DHAVE_CONFIG_H -I. -I../../include -O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection -flto -Os -c -o container-raw.o `test -f '../container-raw.c' || echo './'`../container-raw.c make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-utils-1.1.8/axfer/test' make[2]: *** [Makefile:1017: check-am] Error 2 make[1]: *** [Makefile:650: check-recursive] Error 1 make: *** [Makefile:428: check-recursive] Error 1 error: Bad exit status from /var/tmp/rpm-tmp.VI5lrR (%check) </pre> Issue URL : https://github.com/alsa-project/alsa-utils/issues/16 Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] [PATCH v3 00/26] compat_ioctl: cleanups
by Arnd Bergmann 06 May '19

06 May '19

Hi Al, It took me way longer than I had hoped to revisit this series, see https://lore.kernel.org/lkml/20180912150142.157913-1-arnd@arndb.de/ for the previously posted version. I've come to the point where all conversion handlers and most COMPATIBLE_IOCTL() entries are gone from this file, but for now, this series only has the parts that have either been reviewed previously, or that are simple enough to include. The main missing piece is the SG_IO/SG_GET_REQUEST_TABLE conversion. I'll post the patches I made for that later, as they need more testing and review from the scsi maintainers. I hope you can still take these for the coming merge window, unless new problems come up. Arnd Arnd Bergmann (26): compat_ioctl: pppoe: fix PPPOEIOCSFWD handling compat_ioctl: move simple ppp command handling into driver compat_ioctl: avoid unused function warning for do_ioctl compat_ioctl: move PPPIOCSCOMPRESS32 to ppp-generic.c compat_ioctl: move PPPIOCSPASS32/PPPIOCSACTIVE32 to ppp_generic.c compat_ioctl: handle PPPIOCGIDLE for 64-bit time_t compat_ioctl: move rtc handling into rtc-dev.c compat_ioctl: add compat_ptr_ioctl() compat_ioctl: move drivers to compat_ptr_ioctl compat_ioctl: use correct compat_ptr() translation in drivers ceph: fix compat_ioctl for ceph_dir_operations compat_ioctl: move more drivers to compat_ptr_ioctl compat_ioctl: move tape handling into drivers compat_ioctl: move ATYFB_CLK handling to atyfb driver compat_ioctl: move isdn/capi ioctl translation into driver compat_ioctl: move rfcomm handlers into driver compat_ioctl: move hci_sock handlers into driver compat_ioctl: remove HCIUART handling compat_ioctl: remove HIDIO translation compat_ioctl: remove translation for sound ioctls compat_ioctl: remove IGNORE_IOCTL() compat_ioctl: remove /dev/random commands compat_ioctl: remove joystick ioctl translation compat_ioctl: remove PCI ioctl translation compat_ioctl: remove /dev/raw ioctl translation compat_ioctl: remove last RAID handling code Documentation/networking/ppp_generic.txt | 2 + arch/um/drivers/hostaudio_kern.c | 1 + drivers/android/binder.c | 2 +- drivers/char/ppdev.c | 12 +- drivers/char/random.c | 1 + drivers/char/tpm/tpm_vtpm_proxy.c | 12 +- drivers/crypto/qat/qat_common/adf_ctl_drv.c | 2 +- drivers/dma-buf/dma-buf.c | 4 +- drivers/dma-buf/sw_sync.c | 2 +- drivers/dma-buf/sync_file.c | 2 +- drivers/firewire/core-cdev.c | 12 +- drivers/gpu/drm/amd/amdkfd/kfd_chardev.c | 2 +- drivers/hid/hidraw.c | 4 +- drivers/hid/usbhid/hiddev.c | 11 +- drivers/hwtracing/stm/core.c | 12 +- drivers/ide/ide-tape.c | 31 +- drivers/iio/industrialio-core.c | 2 +- drivers/infiniband/core/uverbs_main.c | 4 +- drivers/isdn/capi/capi.c | 31 + drivers/isdn/i4l/isdn_ppp.c | 14 +- drivers/media/rc/lirc_dev.c | 4 +- drivers/mfd/cros_ec_dev.c | 4 +- drivers/misc/cxl/flash.c | 8 +- drivers/misc/genwqe/card_dev.c | 23 +- drivers/misc/mei/main.c | 22 +- drivers/misc/vmw_vmci/vmci_host.c | 2 +- drivers/mtd/ubi/cdev.c | 36 +- drivers/net/ppp/ppp_generic.c | 99 +++- drivers/net/ppp/pppoe.c | 7 + drivers/net/ppp/pptp.c | 3 + drivers/net/tap.c | 12 +- drivers/nvdimm/bus.c | 4 +- drivers/nvme/host/core.c | 2 +- drivers/pci/switch/switchtec.c | 2 +- drivers/platform/x86/wmi.c | 2 +- drivers/rpmsg/rpmsg_char.c | 4 +- drivers/rtc/dev.c | 13 +- drivers/rtc/rtc-vr41xx.c | 10 + drivers/s390/char/tape_char.c | 41 +- drivers/sbus/char/display7seg.c | 2 +- drivers/sbus/char/envctrl.c | 4 +- drivers/scsi/3w-xxxx.c | 4 +- drivers/scsi/cxlflash/main.c | 2 +- drivers/scsi/esas2r/esas2r_main.c | 2 +- drivers/scsi/megaraid/megaraid_mm.c | 28 +- drivers/scsi/osst.c | 34 +- drivers/scsi/pmcraid.c | 4 +- drivers/scsi/st.c | 35 +- drivers/staging/android/ion/ion.c | 4 +- drivers/staging/pi433/pi433_if.c | 12 +- drivers/staging/vme/devices/vme_user.c | 2 +- drivers/tee/tee_core.c | 2 +- drivers/usb/class/cdc-wdm.c | 2 +- drivers/usb/class/usbtmc.c | 4 +- drivers/usb/core/devio.c | 16 +- drivers/usb/gadget/function/f_fs.c | 12 +- drivers/vfio/vfio.c | 39 +- drivers/vhost/net.c | 12 +- drivers/vhost/scsi.c | 12 +- drivers/vhost/test.c | 12 +- drivers/vhost/vsock.c | 12 +- drivers/video/fbdev/aty/atyfb_base.c | 12 +- drivers/virt/fsl_hypervisor.c | 2 +- fs/btrfs/super.c | 2 +- fs/ceph/dir.c | 1 + fs/ceph/file.c | 2 +- fs/compat_ioctl.c | 602 +------------------- fs/fat/file.c | 13 +- fs/fuse/dev.c | 2 +- fs/notify/fanotify/fanotify_user.c | 2 +- fs/userfaultfd.c | 2 +- include/linux/fs.h | 7 + include/linux/if_pppox.h | 2 + include/linux/mtio.h | 58 ++ include/uapi/linux/ppp-ioctl.h | 2 + include/uapi/linux/ppp_defs.h | 14 + net/bluetooth/hci_sock.c | 21 +- net/bluetooth/rfcomm/sock.c | 14 +- net/l2tp/l2tp_ppp.c | 3 + net/rfkill/core.c | 2 +- sound/core/oss/pcm_oss.c | 4 + sound/oss/dmasound/dmasound_core.c | 2 + 82 files changed, 452 insertions(+), 1034 deletions(-) create mode 100644 include/linux/mtio.h -- 2.20.0 Cc: "David S. Miller" <davem(a)davemloft.net> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Karsten Keil <isdn(a)linux-pingi.de> Cc: "James E.J. Bottomley" <jejb(a)linux.ibm.com> Cc: "Martin K. Petersen" <martin.petersen(a)oracle.com> Cc: Marcel Holtmann <marcel(a)holtmann.org> Cc: netdev(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: devel(a)driverdev.osuosl.org Cc: linux-integrity(a)vger.kernel.org Cc: qat-linux(a)intel.com Cc: linux-crypto(a)vger.kernel.org Cc: linux-media(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux1394-devel(a)lists.sourceforge.net Cc: amd-gfx(a)lists.freedesktop.org Cc: linux-input(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-ide(a)vger.kernel.org Cc: linux-iio(a)vger.kernel.org Cc: linux-rdma(a)vger.kernel.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-mtd(a)lists.infradead.org Cc: linux-ppp(a)vger.kernel.org Cc: linux-nvme(a)lists.infradead.org Cc: platform-driver-x86(a)vger.kernel.org Cc: linux-remoteproc(a)vger.kernel.org Cc: linux-rtc(a)vger.kernel.org Cc: linux-s390(a)vger.kernel.org Cc: sparclinux(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: linux-fbdev(a)vger.kernel.org Cc: linux-btrfs(a)vger.kernel.org Cc: ceph-devel(a)vger.kernel.org Cc: linux-fsdevel(a)vger.kernel.org Cc: linux-bluetooth(a)vger.kernel.org Cc: linux-wireless(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: y2038(a)lists.linaro.org

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Re: [alsa-devel] move eapd coef function before ACT_PRE_PROBE state
by Kailang 06 May '19

06 May '19

Hi Takashi, Are you available for apply this ? Thanks. BR, Kailang > -----Original Message----- > From: Kailang > Sent: Tuesday, April 30, 2019 3:41 PM > To: Takashi Iwai (tiwai(a)suse.de) <tiwai(a)suse.de> > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > Subject: move eapd coef function before ACT_PRE_PROBE state > > Hi Takashi, > > alc_fill_eapd_coef(),this function was change EAPD control to default. > Default was set EAPD by verb control. > This function was run in ACT_INIT state. > Move it to ACT_PRE_PROBE above. It will have a chance to change EAPD > control on ACT_PRE_PROBE state. It could change control by 0x20 coef bit. > > BR, > Kailang

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[alsa-devel] [PATCH v5 0/2] ASoC: CX2072X codec support (revised)
by Takashi Iwai 05 May '19

05 May '19

Hi, this is a v5 revision of ASoC CX2072X codec support patchset for Intel platforms. I've pushed this out in a bit rush because many things have been addressed since the last post. Takashi === v4->v5: Move clk management to runtime PM Sparse warning fixes Some more code simplification Drop tricky regcache fiddling Apply mutex locks around possible racy sequences Move exported jack detection stuff into set_jack callback v3->v4: Fix the wrong endianess conversion in reg write Minor code cleanups v2->v3: CX2072X codec driver: * Move register tables to appropriate place * Remove some confusing codes * Set snd_ctl_boolean_* helpers directly * Fix EQ put callback * Rename to "DAC1 Switch" from "DAC1 Mute Switch" * Drop superfluous regmap calls at shutdown * Avoid regmap_register_patch() * Add missing register definitions * Fix register access on big-endian machine * Remove regcache messes CX2072X machine driver: * Add Pierre's ack * Drop superfluous ssp0 routes v1->v2: Uncomment SOF entries in ACPI binding Move snd_soc_dai_set_bclk_ratio() call into init callback The original cover letter is below: === It's been long time ago the first version of CX2072X codec patch was submitted, and I hoped that it'd be resubmitted, but never happened, as the original author seems to have left, unfortunately. Since I have such a laptop (ASUS E200HA), I took some time to cook and brush up the patches. The patches are for 5.2, should be applicable to ASoC tree for-next branch. Will post the corresponding UCM profile patch once when this gets accepted. thanks, Takashi === Simon Ho (1): ASoC: Add support for Conexant CX2072X CODEC Takashi Iwai (1): ASoC: Intel: Add machine driver for CX2072X on BYT/CHT platforms sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2072x.c | 2139 +++++++++++++++++++++ sound/soc/codecs/cx2072x.h | 309 +++ sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcht_cx2072x.c | 262 +++ sound/soc/intel/common/soc-acpi-intel-byt-match.c | 8 + sound/soc/intel/common/soc-acpi-intel-cht-match.c | 8 + 9 files changed, 2748 insertions(+) create mode 100644 sound/soc/codecs/cx2072x.c create mode 100644 sound/soc/codecs/cx2072x.h create mode 100644 sound/soc/intel/boards/bytcht_cx2072x.c -- 2.16.4

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[alsa-devel] [PATCH] Aseqnet, no nagle and dual stack
by Rob van der Putten 05 May '19

05 May '19

Hi there Not an ALSA source patch, but a patch for an ALSA related util. Aseqnet sends ALSA sound_seq MIDI over TCP/IP. The patch below disables nagle, enables quickack and makes aseqnet dual-stack. --- aseqnet.c.bak 2012-01-25 10:43:38.000000000 +0100 +++ aseqnet.c 2017-08-26 14:17:58.261868853 +0200 @@ -3,6 +3,8 @@ * ver.0.1 * * Copyright (C) 1999-2000 Takashi Iwai + * Modified by Rob van der Putten, Leiden, Holland, + * rob at sput dot nl. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -15,18 +17,21 @@ * */ +#include <alsa/asoundlib.h> +#include <arpa/inet.h> +#include <assert.h> +#include <ctype.h> +#include <getopt.h> +#include <locale.h> +#include <netdb.h> +#include <netinet/in.h> +#include <netinet/tcp.h> +#include <signal.h> #include <stdio.h> #include <stdlib.h> -#include <ctype.h> #include <string.h> -#include <netinet/in.h> #include <sys/socket.h> -#include <netdb.h> -#include <locale.h> -#include <alsa/asoundlib.h> -#include <getopt.h> -#include <signal.h> -#include <assert.h> +#include <sys/types.h> #include "aconfig.h" #include "gettext.h" @@ -327,17 +332,24 @@ */ static void init_server(int port) { + /* + * RvdP, changed to support IPv6 + * Dual stack only! + */ int i; int curstate = 1; - struct sockaddr_in addr; + int ipv6only = 0; + int nodelay = 1; + int quickack = 1; + struct sockaddr_in6 addr; memset(&addr, 0, sizeof(addr)); - addr.sin_family = AF_INET; - addr.sin_addr.s_addr = INADDR_ANY; - addr.sin_port = htons(port); + addr.sin6_family = AF_INET6; + inet_pton(AF_INET6, "::", &(addr.sin6_addr)); + addr.sin6_port = htons(port); - sockfd = socket(AF_INET, SOCK_STREAM, 0); + sockfd = socket(PF_INET6, SOCK_STREAM, 0); if (sockfd < 0) { perror("create socket"); exit(1); @@ -345,7 +357,19 @@ setsockopt(sockfd, SOL_SOCKET, SO_REUSEADDR, &curstate, sizeof(curstate)); /* the return value is ignored.. */ - if (bind(sockfd, (struct sockaddr *)&addr, sizeof(addr)) < 0) { + /* Force dual stack */ + setsockopt(sockfd, IPPROTO_IPV6, IPV6_V6ONLY, &ipv6only, sizeof(ipv6only)); + /* the return value is ignored.. */ + + /* Nagle and quickack */ + if ((setsockopt(sockfd, IPPROTO_TCP, TCP_NODELAY, &nodelay, sizeof(nodelay))) < 0) { + perror("Error setsockopt tcp_nodelay"); + } + if ((setsockopt(sockfd, IPPROTO_TCP, TCP_QUICKACK, &quickack, sizeof(quickack))) < 0) { + perror("Error setsockopt tcp_quickack"); + } + + if (bind(sockfd, (struct sockaddr *) &addr, sizeof(addr)) < 0) { perror("can't bind"); exit(1); } @@ -394,30 +418,58 @@ */ static void init_client(char *server, int port) { - struct sockaddr_in addr; - struct hostent *host; - int curstate = 1; - int fd; + /* + * RvdP, changed to support IPv6 + */ - if ((fd = socket(AF_INET, SOCK_STREAM, IPPROTO_TCP)) < 0){ - perror("create socket"); + struct addrinfo hints; + struct addrinfo *result, *rp; + int curstate = 1; + int nodelay = 1; + int quickack = 1; + int fd, s; + char portstr[8]; + + memset(&hints, 0, sizeof(struct addrinfo)); + hints.ai_family = AF_UNSPEC; + hints.ai_socktype = SOCK_STREAM; + /* hints.ai_protocol = IPPROTO_TCP; */ + hints.ai_flags = 0; + + memset(portstr, 0, 8); + snprintf(portstr, 6, "%d", port); + + s = getaddrinfo(server, portstr, &hints, &result); + if (s != 0) { + fprintf(stderr, "getaddrinfo: %s\n", gai_strerror(s)); + exit(1); + } + for (rp = result; rp != NULL; rp = rp->ai_next) { + fd = socket(rp->ai_family, rp->ai_socktype, rp->ai_protocol); + if (fd == -1) + continue; + if (connect(fd, rp->ai_addr, rp->ai_addrlen) != -1) + break; + close(fd); + } + if (rp == NULL) { + fprintf(stderr, "Could not connect\n"); exit(1); } + if (setsockopt(fd, SOL_SOCKET, SO_REUSEADDR, &curstate, sizeof(curstate)) < 0) { perror("setsockopt"); exit(1); } - if ((host = gethostbyname(server)) == NULL){ - fprintf(stderr, _("can't get address %s\n"), server); - exit(1); + + /* RvdP, nagle and quickack */ + if ((setsockopt(fd, IPPROTO_TCP, TCP_NODELAY, &nodelay, sizeof(nodelay))) < 0) { + perror("Error setsockopt tcp_nodelay"); } - addr.sin_port = htons(port); - addr.sin_family = AF_INET; - memcpy(&addr.sin_addr, host->h_addr, host->h_length); - if (connect(fd, (struct sockaddr *)&addr, sizeof(addr)) < 0) { - perror("connect"); - exit(1); + if ((setsockopt(fd, IPPROTO_TCP, TCP_QUICKACK, &quickack, sizeof(quickack))) < 0) { + perror("Error setsockopt tcp_quickack"); } + if (verbose) fprintf(stderr, _("ok.. connected\n")); netfd[0] = fd; Regards, Rob

2 3

[alsa-devel] [PATCH - maemo fixes 1/1] maemo plugin has two crashes I was able to see in a valgrind log from another user:
by stian.skjelstad＠gmail.com 05 May '19

05 May '19

From: Stian Skjelstad <stian.skjelstad(a)gmail.com> * maximum write size was calculated in words (16bit), but checked against byte-size length. This causes memcpy later to overflow the buffer (normally by up to 12KB). * remove a double free (by marking free'd data with NULL) * mmap returns MMAP_FAILED on error, not NULL I suspect that this plugin/driver might have other issues aswell, since I am unable to find any logic for checking DSP buffer status, and no implementation for odelay reporting. Author: Stian Skjelstad <stian.skjelstad(a)gmail.com> Signed-off-by: Stian Skjelstad <stian.skjelstad(a)gmail.com> diff --git a/maemo/alsa-dsp.c b/maemo/alsa-dsp.c index 7e04f6a..3307384 100644 --- a/maemo/alsa-dsp.c +++ b/maemo/alsa-dsp.c @@ -135,18 +135,18 @@ static snd_pcm_sframes_t alsa_dsp_transfer(snd_pcm_ioplug_t * io, snd_pcm_alsa_dsp_t *alsa_dsp = io->private_data; DENTER(); char *buf; - int words; + int bytes, words; ssize_t result; - words = size * alsa_dsp->bytes_per_frame; - words /= 2; - DPRINT("***** Info: words %d size %lu bpf: %d\n", words, size, - alsa_dsp->bytes_per_frame); - if (words > alsa_dsp->dsp_protocol->mmap_buffer_size) { - DERROR("Requested too much data transfer (playing only %d)\n", - alsa_dsp->dsp_protocol->mmap_buffer_size); - words = alsa_dsp->dsp_protocol->mmap_buffer_size; + bytes = size * alsa_dsp->bytes_per_frame; + DPRINT("***** Info: samples %lu * bpf %d => bytes %d\n", + size, alsa_dsp->bytes_per_frame, bytes); + if (bytes > alsa_dsp->dsp_protocol->mmap_buffer_size) { + DERROR("Requested too much data transfer (requested %d, playing only %d)\n", + bytes, alsa_dsp->dsp_protocol->mmap_buffer_size); + bytes = alsa_dsp->dsp_protocol->mmap_buffer_size; } + words = bytes / 2; if (alsa_dsp->dsp_protocol->state != STATE_PLAYING) { DPRINT("I did nothing - No start sent\n"); alsa_dsp_start(io); diff --git a/maemo/dsp-ctl.c b/maemo/dsp-ctl.c index 5bcda36..ac05942 100644 --- a/maemo/dsp-ctl.c +++ b/maemo/dsp-ctl.c @@ -93,6 +93,7 @@ static void dsp_ctl_close(snd_ctl_ext_t * ext) snd_ctl_dsp_t *dsp_ctl = ext->private_data; DENTER(); free(dsp_ctl->controls); + dsp_ctl->controls = NULL; free_control_list(&(dsp_ctl->playback_devices)); free_control_list(&(dsp_ctl->recording_devices)); // free(dsp_ctl); diff --git a/maemo/dsp-protocol.c b/maemo/dsp-protocol.c index af67251..32193d3 100644 --- a/maemo/dsp-protocol.c +++ b/maemo/dsp-protocol.c @@ -194,7 +194,7 @@ int dsp_protocol_open_node(dsp_protocol_t * dsp_protocol, const char *device) mmap((void *)0, dsp_protocol->mmap_buffer_size, PROT_READ | PROT_WRITE, MAP_SHARED, dsp_protocol->fd, 0); - if (dsp_protocol->mmap_buffer == NULL) { + if (dsp_protocol->mmap_buffer == MAP_FAILED) { report_dsp_protocol("Cannot mmap data buffer", dsp_protocol); ret = -ENOMEM; goto unlock; -- 2.17.1

2 1

Re: [alsa-devel] [PATCH V4] ASoC: fsl_esai: Add pm runtime function
by S.j. Wang 05 May '19

05 May '19

Hi > Hi Mark, > On Fri, May 03, 2019 at 01:27:31PM +0900, Mark Brown wrote: > > On Thu, May 02, 2019 at 09:13:58AM +0000, S.j. Wang wrote: > > > > > I am checking, but I don't know why this patch failed in your side. > > > I Tried to apply this patch on for-5.1, for 5.2, for-linus and > > > for-next, all are Successful. The git is > git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git. > > > > > I can't reproduce your problem. Is there any my operation wrong? > > > > The error message I got was: > > > > Applying: ASoC: fsl_esai: Add pm runtime function > > error: patch failed: sound/soc/fsl/fsl_esai.c:9 > > error: sound/soc/fsl/fsl_esai.c: patch does not apply Patch failed at > > 0001 ASoC: fsl_esai: Add pm runtime function > > > > which is the header addition. I can't spot any obvious issues > > visually looking at the patch, only thing I can think is some kind of > > whitespace damage somewhere. > > I downloaded this v4 from patchwork and resubmitted a v5 for a test. > Would you please try to apply that one? > > If my v5 works vs. having merge conflict at v4, maybe something wrong with > Git version of Shengjiu's? I compared my v5 and his > v4 using vimdiff, there is no much difference of whitespace. > > Thanks > Nicolin We find that maybe it is caused by the Transfer-Encoding format. We sent the patch by the --transfer-encoding=8bit, but in the receiver side it shows: Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: base64 It may be caused by our company's mail server. We are checking... Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH] ALSA: hda/hdmi - Don't report Jack event if no need to do that
by Hui Wang 04 May '19

04 May '19

On the machines with AMD GPU or Nvidia GPU, we often meet this issues: after s3, there are 4 HDMI/DP audio devices in the gnome-sound-setting even there is no any monitors plugged. When this problem happens, we check the /proc/asound/cardX/eld#N.M, we will find the monitor_present=1, eld_valid=0. The root cause is somehow the pin_sense reports the monitor is present and eld is valid when there is no monitor plugged. The current driver will read the eld data if the pin_sense reports the eld is valid, because of no monitor is plugged, there is no valid eld data, then the eld->valid is set to 0. If we don't let driver report Jack event when monitor_present=1 while eld_valid=0, there will be no this issue. After this change, the driver only reports Jack event with one of the below 2 conditons: eld->monitor_present=1 and eld->eld_valid=1 (a valid monitor detect) eld->monitor_present=0 (a monitor is unplugged) Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8b3ac690efa3..e5a34fa6f358 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1548,7 +1548,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, else update_eld(codec, per_pin, eld); - ret = !repoll || !eld->monitor_present || eld->eld_valid; + ret = !eld->monitor_present || eld->eld_valid; jack = snd_hda_jack_tbl_get(codec, pin_nid); if (jack) -- 2.17.1

2 10

[alsa-devel] [PATCH V4] ASoC: fsl_esai: Add pm runtime function
by S.j. Wang 03 May '19

03 May '19

Add pm runtime support and move clock handling there. Close the clocks at suspend to reduce the power consumption. fsl_esai_suspend is replaced by pm_runtime_force_suspend. fsl_esai_resume is replaced by pm_runtime_force_resume. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changes in v4 -resend base on for-5.2 Changes in v3 -refine the commit comments. -add acked-by Changes in v2 -refine the commit comments. -move regcache_mark_dirty to runtime suspend. sound/soc/fsl/fsl_esai.c | 141 ++++++++++++++++++++++++++--------------------- 1 file changed, 77 insertions(+), 64 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index bad0dfed6b68..10d2210c91ef 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -9,6 +9,7 @@ #include <linux/module.h> #include <linux/of_irq.h> #include <linux/of_platform.h> +#include <linux/pm_runtime.h> #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> @@ -466,30 +467,6 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); - int ret; - - /* - * Some platforms might use the same bit to gate all three or two of - * clocks, so keep all clocks open/close at the same time for safety - */ - ret = clk_prepare_enable(esai_priv->coreclk); - if (ret) - return ret; - if (!IS_ERR(esai_priv->spbaclk)) { - ret = clk_prepare_enable(esai_priv->spbaclk); - if (ret) - goto err_spbaclk; - } - if (!IS_ERR(esai_priv->extalclk)) { - ret = clk_prepare_enable(esai_priv->extalclk); - if (ret) - goto err_extalck; - } - if (!IS_ERR(esai_priv->fsysclk)) { - ret = clk_prepare_enable(esai_priv->fsysclk); - if (ret) - goto err_fsysclk; - } if (!dai->active) { /* Set synchronous mode */ @@ -506,16 +483,6 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, return 0; -err_fsysclk: - if (!IS_ERR(esai_priv->extalclk)) - clk_disable_unprepare(esai_priv->extalclk); -err_extalck: - if (!IS_ERR(esai_priv->spbaclk)) - clk_disable_unprepare(esai_priv->spbaclk); -err_spbaclk: - clk_disable_unprepare(esai_priv->coreclk); - - return ret; } static int fsl_esai_hw_params(struct snd_pcm_substream *substream, @@ -576,20 +543,6 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, return 0; } -static void fsl_esai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); - - if (!IS_ERR(esai_priv->fsysclk)) - clk_disable_unprepare(esai_priv->fsysclk); - if (!IS_ERR(esai_priv->extalclk)) - clk_disable_unprepare(esai_priv->extalclk); - if (!IS_ERR(esai_priv->spbaclk)) - clk_disable_unprepare(esai_priv->spbaclk); - clk_disable_unprepare(esai_priv->coreclk); -} - static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -658,7 +611,6 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, static const struct snd_soc_dai_ops fsl_esai_dai_ops = { .startup = fsl_esai_startup, - .shutdown = fsl_esai_shutdown, .trigger = fsl_esai_trigger, .hw_params = fsl_esai_hw_params, .set_sysclk = fsl_esai_set_dai_sysclk, @@ -947,6 +899,10 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + pm_runtime_enable(&pdev->dev); + + regcache_cache_only(esai_priv->regmap, true); + ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); if (ret) dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); @@ -954,6 +910,13 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } +static int fsl_esai_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + + return 0; +} + static const struct of_device_id fsl_esai_dt_ids[] = { { .compatible = "fsl,imx35-esai", }, { .compatible = "fsl,vf610-esai", }, @@ -961,22 +924,35 @@ static int fsl_esai_probe(struct platform_device *pdev) }; MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); -#ifdef CONFIG_PM_SLEEP -static int fsl_esai_suspend(struct device *dev) -{ - struct fsl_esai *esai = dev_get_drvdata(dev); - - regcache_cache_only(esai->regmap, true); - regcache_mark_dirty(esai->regmap); - - return 0; -} - -static int fsl_esai_resume(struct device *dev) +#ifdef CONFIG_PM +static int fsl_esai_runtime_resume(struct device *dev) { struct fsl_esai *esai = dev_get_drvdata(dev); int ret; + /* + * Some platforms might use the same bit to gate all three or two of + * clocks, so keep all clocks open/close at the same time for safety + */ + ret = clk_prepare_enable(esai->coreclk); + if (ret) + return ret; + if (!IS_ERR(esai->spbaclk)) { + ret = clk_prepare_enable(esai->spbaclk); + if (ret) + goto err_spbaclk; + } + if (!IS_ERR(esai->extalclk)) { + ret = clk_prepare_enable(esai->extalclk); + if (ret) + goto err_extalclk; + } + if (!IS_ERR(esai->fsysclk)) { + ret = clk_prepare_enable(esai->fsysclk); + if (ret) + goto err_fsysclk; + } + regcache_cache_only(esai->regmap, false); /* FIFO reset for safety */ @@ -987,22 +963,59 @@ static int fsl_esai_resume(struct device *dev) ret = regcache_sync(esai->regmap); if (ret) - return ret; + goto err_regcache_sync; /* FIFO reset done */ regmap_update_bits(esai->regmap, REG_ESAI_TFCR, ESAI_xFCR_xFR, 0); regmap_update_bits(esai->regmap, REG_ESAI_RFCR, ESAI_xFCR_xFR, 0); return 0; + +err_regcache_sync: + if (!IS_ERR(esai->fsysclk)) + clk_disable_unprepare(esai->fsysclk); +err_fsysclk: + if (!IS_ERR(esai->extalclk)) + clk_disable_unprepare(esai->extalclk); +err_extalclk: + if (!IS_ERR(esai->spbaclk)) + clk_disable_unprepare(esai->spbaclk); +err_spbaclk: + clk_disable_unprepare(esai->coreclk); + + return ret; } -#endif /* CONFIG_PM_SLEEP */ + +static int fsl_esai_runtime_suspend(struct device *dev) +{ + struct fsl_esai *esai = dev_get_drvdata(dev); + + regcache_cache_only(esai->regmap, true); + regcache_mark_dirty(esai->regmap); + + if (!IS_ERR(esai->fsysclk)) + clk_disable_unprepare(esai->fsysclk); + if (!IS_ERR(esai->extalclk)) + clk_disable_unprepare(esai->extalclk); + if (!IS_ERR(esai->spbaclk)) + clk_disable_unprepare(esai->spbaclk); + clk_disable_unprepare(esai->coreclk); + + return 0; +} +#endif /* CONFIG_PM */ static const struct dev_pm_ops fsl_esai_pm_ops = { - SET_SYSTEM_SLEEP_PM_OPS(fsl_esai_suspend, fsl_esai_resume) + SET_RUNTIME_PM_OPS(fsl_esai_runtime_suspend, + fsl_esai_runtime_resume, + NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) }; static struct platform_driver fsl_esai_driver = { .probe = fsl_esai_probe, + .remove = fsl_esai_remove, .driver = { .name = "fsl-esai-dai", .pm = &fsl_esai_pm_ops, -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: core: Allow topology to override machine driver FE DAI link config.
by Liam Girdwood 03 May '19

03 May '19

Machine drivers statically define a number of DAI links that currently cannot be changed or removed by topology. This means PCMs and platform components cannot be changed by topology at runtime AND machine drivers are tightly coupled to topology. This patch allows topology to override the machine driver DAI link config in order to reuse machine drivers with different topologies and platform components. The patch supports :- 1) create new FE PCMs with a topology defined PCM ID. 2) destroy existing static FE PCMs 3) change the platform component driver. 4) assign any new HW params fixups. 5) assign a new card name prefix to differentiate this topology to userspace. The patch requires no changes to the machine drivers, but does add some platform component flags that the platform component driver can assign before loading topologies. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1. o Now iterate over component_list to fix crash with DT enumerated cards. o Make sure name prefix is only added once with deferred probiing. include/sound/soc.h | 13 ++++++ sound/soc/soc-core.c | 101 +++++++++++++++++++++++++++++++++++++++++-- sound/soc/soc-pcm.c | 12 +++++ 3 files changed, 123 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f7579f82cc00..b1d65fcb8756 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -806,6 +806,14 @@ struct snd_soc_component_driver { unsigned int use_pmdown_time:1; /* care pmdown_time at stop */ unsigned int endianness:1; unsigned int non_legacy_dai_naming:1; + + /* this component uses topology and ignore machine driver FEs */ + const char *ignore_machine; + const char *topology_name_prefix; + int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); + bool use_dai_pcm_id; /* use the DAI link PCM ID as PCM device number */ + int be_pcm_base; /* base device ID for all BE PCMs */ }; struct snd_soc_component { @@ -963,6 +971,9 @@ struct snd_soc_dai_link { /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; + /* Do not create a PCM for this DAI link (Backend link) */ + unsigned int ignore:1; + struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; @@ -1002,6 +1013,7 @@ struct snd_soc_card { const char *long_name; const char *driver_name; char dmi_longname[80]; + char topology_shortname[32]; struct device *dev; struct snd_card *snd_card; @@ -1011,6 +1023,7 @@ struct snd_soc_card { struct mutex dapm_mutex; bool instantiated; + bool topology_shortname_created; int (*probe)(struct snd_soc_card *card); int (*late_probe)(struct snd_soc_card *card); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4663de3cf495..38bf7d01894b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -852,6 +852,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, const char *platform_name; int i; + if (dai_link->ignore) + return 0; + dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); if (soc_is_dai_link_bound(card, dai_link)) { @@ -1461,7 +1464,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + int i, ret, num; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, rtd->num, order); @@ -1507,9 +1512,28 @@ static int soc_probe_link_dais(struct snd_soc_card *card, soc_dpcm_debugfs_add(rtd); #endif + num = rtd->num; + + /* + * most drivers will register their PCMs using DAI link ordering but + * topology based drivers can use the DAI link id field to set PCM + * device number and then use rtd + a base offset of the BEs. + */ + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (!component->driver->use_dai_pcm_id) + continue; + + if (rtd->dai_link->no_pcm) + num += component->driver->be_pcm_base; + else + num = rtd->dai_link->id; + } + if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = cpu_dai->driver->compress_new(rtd, rtd->num); + ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); @@ -1519,7 +1543,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, if (!dai_link->params) { /* create the pcm */ - ret = soc_new_pcm(rtd, rtd->num); + ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); @@ -1846,6 +1870,74 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name); #endif /* CONFIG_DMI */ +static void soc_check_tplg_fes(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + const struct snd_soc_component_driver *comp_drv; + struct snd_soc_dai_link *dai_link; + int i; + + list_for_each_entry(component, &component_list, list) { + + /* does this component override FEs ? */ + if (!component->driver->ignore_machine) + continue; + + /* for this machine ? */ + if (strcmp(component->driver->ignore_machine, + card->dev->driver->name)) + continue; + + /* machine matches, so override the rtd data */ + for (i = 0; i < card->num_links; i++) { + + dai_link = &card->dai_link[i]; + + /* ignore this FE */ + if (dai_link->dynamic) { + dai_link->ignore = true; + continue; + } + + dev_info(card->dev, "info: override FE DAI link %s\n", + card->dai_link[i].name); + + /* override platform component */ + dai_link->platform_name = component->name; + + /* convert non BE into BE */ + dai_link->no_pcm = 1; + + /* override any BE fixups */ + dai_link->be_hw_params_fixup = + component->driver->be_hw_params_fixup; + + /* most BE links don't set stream name, so set it to + * dai link name if it's NULL to help bind widgets. + */ + if (!dai_link->stream_name) + dai_link->stream_name = dai_link->name; + } + + /* Inform userspace we are using alternate topology */ + if (component->driver->topology_name_prefix) { + + /* topology shortname created ? */ + if (!card->topology_shortname_created) { + comp_drv = component->driver; + + snprintf(card->topology_shortname, 32, "%s-%s", + comp_drv->topology_name_prefix, + card->name); + card->topology_shortname_created = true; + } + + /* use topology shortname */ + card->name = card->topology_shortname; + } + } +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -1855,6 +1947,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + /* check whether any platform is ignore machine FE and using topology */ + soc_check_tplg_fes(card); + /* bind DAIs */ for (i = 0; i < card->num_links; i++) { ret = soc_bind_dai_link(card, &card->dai_link[i]); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 153218aa4909..db51849ae9bd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -865,8 +865,20 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret; + /* perform any topology hw_params fixups before DAI */ + if (rtd->dai_link->be_hw_params_fixup) { + ret = rtd->dai_link->be_hw_params_fixup(rtd, params); + if (ret < 0) { + dev_err(rtd->dev, + "ASoC: hw_params topology fixup failed %d\n", + ret); + return ret; + } + } + if (dai->driver->ops->hw_params) { ret = dai->driver->ops->hw_params(substream, params, dai); if (ret < 0) { -- 2.17.1

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[alsa-devel] [PATCH] Headset, microphone and internal speaker support for Tuxedo XC 1509
by Richard Sailer 03 May '19

03 May '19

This adds a SND_PCI_QUIRK(...) line for the Tuxedo XC 1509. The Tuxedo XC 1509 and the System76 oryp5 are the same barebone notebooks manufactured by Clevo. To name the fixups both use after the actual underlying hardware, this patch also changes System76_orpy5 to clevo_pb51ed in 2 enum symbols and one function name, matching the other pci_quirk entries which are also named after the device ODM. Signed-off-by: Richard Sailer <rs(a)tuxedocomputers.com> --- sound/pci/hda/patch_realtek.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 715ab2342151..9df868ea96ee 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1864,8 +1864,8 @@ enum { ALC887_FIXUP_BASS_CHMAP, ALC1220_FIXUP_GB_DUAL_CODECS, ALC1220_FIXUP_CLEVO_P950, - ALC1220_FIXUP_SYSTEM76_ORYP5, - ALC1220_FIXUP_SYSTEM76_ORYP5_PINS, + ALC1220_FIXUP_CLEVO_PB51ED, + ALC1220_FIXUP_CLEVO_PB51ED_PINS, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2070,7 +2070,7 @@ static void alc1220_fixup_clevo_p950(struct hda_codec *codec, static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, const struct hda_fixup *fix, int action); -static void alc1220_fixup_system76_oryp5(struct hda_codec *codec, +static void alc1220_fixup_clevo_pb51ed(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2322,18 +2322,18 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc1220_fixup_clevo_p950, }, - [ALC1220_FIXUP_SYSTEM76_ORYP5] = { + [ALC1220_FIXUP_CLEVO_PB51ED] = { .type = HDA_FIXUP_FUNC, - .v.func = alc1220_fixup_system76_oryp5, + .v.func = alc1220_fixup_clevo_pb51ed, }, - [ALC1220_FIXUP_SYSTEM76_ORYP5_PINS] = { + [ALC1220_FIXUP_CLEVO_PB51ED_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ {} }, .chained = true, - .chain_id = ALC1220_FIXUP_SYSTEM76_ORYP5, + .chain_id = ALC1220_FIXUP_CLEVO_PB51ED, }, }; @@ -2411,8 +2411,9 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x9501, "Clevo P950HR", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x95e1, "Clevo P95xER", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x95e2, "Clevo P950ER", ALC1220_FIXUP_CLEVO_P950), - SND_PCI_QUIRK(0x1558, 0x96e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_SYSTEM76_ORYP5_PINS), - SND_PCI_QUIRK(0x1558, 0x97e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_SYSTEM76_ORYP5_PINS), + SND_PCI_QUIRK(0x1558, 0x96e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x97e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x65d1, "Tuxedo Book XC1509", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), -- 2.17.1

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[alsa-devel] [PATCH v2 0/5] Allwinner H6 SPDIF support
by Clément Péron 03 May '19

03 May '19

*H6 DMA support IS REQUIRED* Allwinner H6 SoC has a SPDIF controller called One Wire Audio (OWA) which is different from the previous H3 generation and not compatible. Difference are an increase of fifo sizes, some memory mapping are different and there is now the possibility to output the master clock on a pin. Also this series require the DMA working on H6, a first version has been submitted by Jernej Škrabec but is not yet accepted (as this moment): https://patchwork.kernel.org/project/linux-arm-kernel/list/?series=89011 This series has been tested on Beelink GS1 on top of sunxi/for-next with Jernej DMA support patch. Clément Péron (5): dt-bindings: sound: sun4i-spdif: Add Allwinner H6 compatible ASoC: sun4i-spdif: Add support for H6 SoC arm64: dts: allwinner: Add SPDIF node for Allwinner H6 arm64: dts: allwinner: h6: Enable SPDIF for Beelink GS1 arm64: defconfig: Enable Sun4i SPDIF module .../bindings/sound/sunxi,sun4i-spdif.txt | 3 +- .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 4 ++ arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 38 +++++++++++++++++ arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-spdif.c | 42 ++++++++++++++++--- 5 files changed, 81 insertions(+), 7 deletions(-) -- 2.17.1

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Re: [alsa-devel] [PATCH 00/19] ASoC: SOF: improvements for ABI checks and Intel code
by Pierre-Louis Bossart 03 May '19

03 May '19

On 5/3/19 12:40 AM, Mark Brown wrote: > On Tue, Apr 30, 2019 at 06:09:15PM -0500, Pierre-Louis Bossart wrote: >> This series is a set of relatively small SOF changes after the big >> batch merged last week (Thanks!). Since we are very close to the merge >> window and in May where most of the world takes time off, it'd be >> perfectly understandable if those patches were queued for 5.3, after >> feedback and corrections as needed. > > The headline thing I see when this pops up in my inbox is yet another 20 > patch series for these DSPs, I'm not going to notice the size of the > patches on a first pass. As I think you've noticed there's a lot of > reviewer fatigue setting in with this stuff, one thing that'd really > help here is if there were some help from Intel people with review for > the DPCM code. I can certainly understand reviewer fatigue, i've had to put a time limit on daily reviews to keep my sanity, but I don't get your last point. These patches were submitted and reviewed by Intel people on GitHub, what you see here is the result of multiple iterations precisely to make sure the patches are acceptable for upstream. we've set the goal of having two Intel aprovers for each patch. Can you elaborate on how we can make your life simpler? FYI we are going to have a number of fixes coming in the next 2 months for HDaudio support, changes to IPC/power flows to support D0ix states and SoundWire, 3rd party firmware module support, etc, so I really want to make sure you are comfortable with SOF-related patches, their level of maturity and the submission rate. Thanks! -Pierre

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[alsa-devel] [PATCH v2 0/2] soundwire: fix Kconfig select/depend issues
by Pierre-Louis Bossart 03 May '19

03 May '19

0-day/Kbuild starts complaining about missed module dependencies and compilation issues. Since codecs and soc drivers need to be compilable independently, let's fix this using the following model: SOF_INTEL_XYZ --- select ----> SOUNDWIRE_INTEL ---> select ---- | v SOUNDWIRE_BUS ^ | CODEC_ABC ------- select -----> REGMAP_SOUNDWIRE ---> select --- The SOUNDWIRE option remains a top-level boolean selector without impact on compilation, as recommended by Linus [1] and done for all Intel ASoC stuff. [1] https://lkml.org/lkml/2017/11/15/554 Changes since v1: Removed useless depend in REGMAP_SOUNDWIRE (select will ignore dependencies) Reordered patches to avoid circular dependencies with git bisect Pierre-Louis Bossart (2): soundwire: fix SOUNDWIRE_BUS option regmap: soundwire: fix Kconfig select/depend issue drivers/base/regmap/Kconfig | 2 +- drivers/soundwire/Kconfig | 1 - 2 files changed, 1 insertion(+), 2 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH] MAINTAINERS: update git tree for sound entries
by Ross Zwisler 03 May '19

03 May '19

Several sound related entries in MAINTAINERS refer to the old git tree at "git://git.alsa-project.org/alsa-kernel.git". This is no longer used for development, and Takashi Iwai's kernel.org tree is used instead. Signed-off-by: Ross Zwisler <zwisler(a)google.com> --- MAINTAINERS | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index e17ebf70b5480..d373d976a9317 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3351,7 +3351,7 @@ F: include/uapi/linux/bsg.h BT87X AUDIO DRIVER M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -T: git git://git.alsa-project.org/alsa-kernel.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: Documentation/sound/cards/bt87x.rst F: sound/pci/bt87x.c @@ -3404,7 +3404,7 @@ F: drivers/scsi/FlashPoint.* C-MEDIA CMI8788 DRIVER M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -T: git git://git.alsa-project.org/alsa-kernel.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: sound/pci/oxygen/ @@ -5696,7 +5696,7 @@ F: drivers/edac/qcom_edac.c EDIROL UA-101/UA-1000 DRIVER M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -T: git git://git.alsa-project.org/alsa-kernel.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: sound/usb/misc/ua101.c @@ -6036,7 +6036,7 @@ F: include/linux/f75375s.h FIREWIRE AUDIO DRIVERS M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -T: git git://git.alsa-project.org/alsa-kernel.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: sound/firewire/ @@ -11593,7 +11593,7 @@ F: Documentation/devicetree/bindings/opp/ OPL4 DRIVER M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -T: git git://git.alsa-project.org/alsa-kernel.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: sound/drivers/opl4/ @@ -14490,7 +14490,6 @@ M: Takashi Iwai <tiwai(a)suse.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://www.alsa-project.org/ T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git -T: git git://git.alsa-project.org/alsa-kernel.git Q: http://patchwork.kernel.org/project/alsa-devel/list/ S: Maintained F: Documentation/sound/ @@ -16100,7 +16099,7 @@ F: drivers/usb/storage/ USB MIDI DRIVER M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -T: git git://git.alsa-project.org/alsa-kernel.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: sound/usb/midi.* -- 2.21.0.593.g511ec345e18-goog

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Re: [alsa-devel] [PATCH 1/2] ASoC: Add support for Conexant CX2072X CODEC
by Takashi Iwai 03 May '19

03 May '19

On Fri, 03 May 2019 08:47:29 +0200, Mark Brown wrote: > > > > > +#define cx2072x_plbk_eq_en_info snd_ctl_boolean_mono_info > > > > Why not just use the function directly rather than hiding it? > > > Just a standard idiom. Can be replaced if preferred. > > Please. BTW, a good reason for the style above is that it makes code more undrestandable. For defining a ctl element, typically we define three callback functions, info, get and put, in that order: static int foo_info() { .... } static int foo_get() { .... } static int foo_put() { ... }; and often the actual snd_kcontrol_new table containing these callbacks appears much later, where you'd need to scroll down a few screens to read that point. In the above, especially defining the info callback at the beginning is important. By reading foo_init() at first, readers can know which type (int, boolean, enum, etc) and the number of elements to be used in get/put callbacks beforehand. It's a commonly seen mistake, for example, that a wrong type (e.g. integer) is passed to info callback while enum type is used in the get/put. The #define above keeps this foo_info() definition while optimizing with the standard helper. If we drop this and set snd_ctl_boolean_mono_info directly in the snd_kcontrol_new entry, you'll have to go and back the screen just to take a look at the info callback. That's why I called it a standard idiom. It's not strictly necessary, but often help reading / reviewing the code. Though, it's just another bikeshed theme, so I'm not sticking with this style. thanks, Takashi

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Re: [alsa-devel] [PATCH 1/2] ASoC: Add support for Conexant CX2072X CODEC
by Takashi Iwai 03 May '19

03 May '19

On Fri, 03 May 2019 08:47:29 +0200, Mark Brown wrote: > > On Thu, May 02, 2019 at 09:04:06AM +0200, Takashi Iwai wrote: > > Mark Brown wrote: > > > On Tue, Apr 23, 2019 at 04:13:35PM +0200, Takashi Iwai wrote: > > > > This looks *very* much like board configuration rather than a patch - > > > there's no kind of test bit and the comments talk specifically about > > > things like gain settings and pad configuration which look very board > > > specific. Register patches are supposed to be for things like early > > > revisions of the chip which have different register defaults or magic > > > sequences that vendors tell you to run on startup, usually to tune test > > > registers. > > > OK, will replace with the straight regmap_multi_reg_write(). > > That's probably not addressing the issue, a lot of that stuff just > doesn't seem like it should be in some fixed configuration table at all. So what's your alternative suggestion? > > > > +#define cx2072x_plbk_eq_en_info snd_ctl_boolean_mono_info > > > > Why not just use the function directly rather than hiding it? > > > Just a standard idiom. Can be replaced if preferred. > > Please. > > > > > +int snd_soc_cx2072x_enable_jack_detect(struct snd_soc_component *codec) > > > > +{ > > > > + struct cx2072x_priv *cx2072x = snd_soc_component_get_drvdata(codec); > > > > + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(codec); > > > > + > > > > + /* No-sticky input type */ > > > > + regmap_write(cx2072x->regmap, CX2072X_GPIO_STICKY_MASK, 0x1f); > > > > + > > > > + /* Use GPOI0 as interrupt pin */ > > > > + regmap_write(cx2072x->regmap, CX2072X_UM_INTERRUPT_CRTL_E, 0x12 << 24); > > > > This isn't board specific is it? > > > I have no idea. It's been so from the original code, and there > > doesn't seem any other hardware implementations. > > Oh, joy. What's the story here? Do you have a datasheet for the part? Not at all. I just refreshed the already submitted patches since I have a laptop with the codec. I tried to contact Conexant, but in vain, so I decided to submit the renewed one. > > The jack detection in ASoC is anyway a bit funky, especially when > > involved with PM... > > What do you mean here? I'm not aware of any issues and the systems I've > worked with seemed robust... There are tons of different ways of implementation for jack controls, with different API usages. IOW, no consistency. > > > > + dev_dbg(codec->dev, "CX2072X_HSDETECT type=0x%X,Jack state = %x\n", > > > > + type, state); > > > > + return state; > > > > +} > > > > +EXPORT_SYMBOL_GPL(snd_soc_cx2072x_get_jack_state); > > > > Why is this symbol exported? > > > It's called from the machine driver. > > snd_soc_jack_add_gpios() is called in the machine driver side, and it > > needs the jack_status_check callback that calls this function. > > That code shouldn't be in the machine driver, the CODEC driver should > request any interrupts it needs itself. The similar things are done on many other Intel SST board drivers. The current patch just follows the pattern. > > > > + /* use flat eq by default */ > > > > + for (ch = 0 ; ch < 2 ; ch++) { > > > > + for (band = 0; band < CX2072X_PLBK_EQ_BAND_NUM; band++) { > > > > + cx2072x->plbk_eq[ch][band][1] = 64; > > > > + cx2072x->plbk_eq[ch][band][10] = 3; > > > > + } > > > > + } > > > > Why not use the register defaults? > > > Because it'll become too messy for put flatten array values? > > The initialization using loop makes more sense in such a case, IMO. > > No, that's not the question. The question is why there is any > initialization at all? Again, no idea. These are likely no default values of the hardware registers, and we need to set up some. I *guess* ditto for the initial register table in the above, too. thanks, Takashi

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[alsa-devel] [PATCH 0/2] ASoC: CX2072X codec support (revised)
by Takashi Iwai 03 May '19

03 May '19

Hi, it's been long time ago the first version of CX2072X codec patch was submitted, and I hoped that it'd be resubmitted, but never happened, as the original author seems to have left, unfortunately. Since I have such a laptop (ASUS E200HA), I took some time to cook and brush up the patches. The patches are for 5.2, should be applicable to ASoC tree for-next branch. Will post the corresponding UCM profile patch once when this gets accepted. thanks, Takashi === Simon Ho (1): ASoC: Add support for Conexant CX2072X CODEC Takashi Iwai (1): ASoC: Intel: Add machine driver for CX2072X on BYT/CHT platforms sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2072x.c | 2197 +++++++++++++++++++++ sound/soc/codecs/cx2072x.h | 317 +++ sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcht_cx2072x.c | 285 +++ sound/soc/intel/common/soc-acpi-intel-byt-match.c | 8 + sound/soc/intel/common/soc-acpi-intel-cht-match.c | 8 + 9 files changed, 2837 insertions(+) create mode 100644 sound/soc/codecs/cx2072x.c create mode 100644 sound/soc/codecs/cx2072x.h create mode 100644 sound/soc/intel/boards/bytcht_cx2072x.c -- 2.16.4

3 9

[alsa-devel] [PATCH v2] ASoC: mediatek: mt8516: register ADDA DAI
by Fabien Parent 03 May '19

03 May '19

Register the ADDA DAI driver into the MT8516 PCM driver. Signed-off-by: Fabien Parent <fparent(a)baylibre.com> --- This patch depends on patch serie: [PATCH 0/5] ASoC: mediatek: Add basic PCM driver for MT8516 v2: * Register ADDA before memif to fix ordering issue. --- sound/soc/mediatek/mt8516/mt8516-afe-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/mediatek/mt8516/mt8516-afe-pcm.c b/sound/soc/mediatek/mt8516/mt8516-afe-pcm.c index 84fbb5dbbd14..dea9221c67aa 100644 --- a/sound/soc/mediatek/mt8516/mt8516-afe-pcm.c +++ b/sound/soc/mediatek/mt8516/mt8516-afe-pcm.c @@ -10,6 +10,7 @@ #include <linux/module.h> #include <linux/of.h> +#include "mt8516-afe-common.h" #include "mt8516-afe-regs.h" #include "../common/mtk-afe-platform-driver.h" @@ -669,6 +670,7 @@ static int mt8516_dai_memif_register(struct mtk_base_afe *afe) typedef int (*dai_register_cb)(struct mtk_base_afe *); static const dai_register_cb dai_register_cbs[] = { + mt8516_dai_adda_register, mt8516_dai_memif_register, }; -- 2.20.1

2 1

[alsa-devel] [RFC PATCH] ASoC: max98357a: manage GPIO for component rebinding
by Tzung-Bi Shih 03 May '19

03 May '19

Component's probe() gets GPIO for "sdmode-gpios" via devm_gpiod_get_optional(). The GPIO binds to the device. When the component get removed but the device does not, the GPIO still binds to the device. Then, when the component be probed again, devm_gpiod_get_optional() returns EBUSY. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Hi, We discovered the issue when rebinding sound card. We found that the rebinding failed due to the GPIO was hold by the first time component's probe() in max98357. The patch changes devm to non-devm API, because we don't have equivalent API for ASoC components (e.g. compm_). Question: we are not very sure to move from devm to non-devm, because devm should be most encouraged. Question: does it suggest not to use devm for GPIO in component's probe()? It seems there are still some such usages, for example sound/soc/codecs/dmic.c. sound/soc/codecs/max98357a.c | 42 ++++++++++++++++++++++++++++-------- 1 file changed, 33 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index d037a3e4d323..4e90876f6ef6 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -27,24 +27,28 @@ #include <sound/soc-dai.h> #include <sound/soc-dapm.h> +struct max98357a_priv { + struct gpio_desc *sdmode; +}; + static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai); + struct max98357a_priv *max98357a = snd_soc_dai_get_drvdata(dai); - if (!sdmode) + if (!max98357a->sdmode) return 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - gpiod_set_value(sdmode, 1); + gpiod_set_value(max98357a->sdmode, 1); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - gpiod_set_value(sdmode, 0); + gpiod_set_value(max98357a->sdmode, 0); break; } @@ -61,19 +65,31 @@ static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { static int max98357a_component_probe(struct snd_soc_component *component) { - struct gpio_desc *sdmode; + struct max98357a_priv *max98357a = + snd_soc_component_get_drvdata(component); - sdmode = devm_gpiod_get_optional(component->dev, "sdmode", GPIOD_OUT_LOW); - if (IS_ERR(sdmode)) - return PTR_ERR(sdmode); + max98357a->sdmode = gpiod_get_optional(component->dev, + "sdmode", GPIOD_OUT_LOW); + if (IS_ERR(max98357a->sdmode)) + return PTR_ERR(max98357a->sdmode); - snd_soc_component_set_drvdata(component, sdmode); + snd_soc_component_set_drvdata(component, max98357a); return 0; } +static void max98357a_component_remove(struct snd_soc_component *component) +{ + struct max98357a_priv *max98357a = + snd_soc_component_get_drvdata(component); + + if (max98357a->sdmode) + gpiod_put(max98357a->sdmode); +} + static const struct snd_soc_component_driver max98357a_component_driver = { .probe = max98357a_component_probe, + .remove = max98357a_component_remove, .dapm_widgets = max98357a_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98357a_dapm_widgets), .dapm_routes = max98357a_dapm_routes, @@ -112,6 +128,14 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { static int max98357a_platform_probe(struct platform_device *pdev) { + struct max98357a_priv *max98357a; + + max98357a = devm_kzalloc(&pdev->dev, sizeof(*max98357a), GFP_KERNEL); + if (!max98357a) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, max98357a); + return devm_snd_soc_register_component(&pdev->dev, &max98357a_component_driver, &max98357a_dai_driver, 1); -- 2.21.0.593.g511ec345e18-goog

2 1

[alsa-devel] [PATCH] ASoC: Intel: boards: kbl_da7219_max98927: add dai_trigger function
by mac.chiang＠intel.com 03 May '19

03 May '19

From: Mac Chiang <mac.chiang(a)intel.com> amplifier feedback is not modeled as being dependent on any active output. Even when there is no playback happening, parts of the graph, specifically the IV sense->speaker protection->output remains active and this prevents the DSP from entering low-power states. This patch suggest a machine driver level approach where the speaker pins are enabled/disabled dynamically depending on stream start/stop events. DPAM graph representations show the feedback loop is indeed disabled and low-power states can be reached. Signed-off-by: Jenny TC <jenny.tc(a)intel.com> Signed-off-by: Harshapriya.n <harshapriya.n(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 73 +++++++++++++++++++++++++++- 1 file changed, 71 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index f72a7bf..1efe7fd 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -219,8 +219,60 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return 0; } +static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int j, ret; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + const char *name = codec_dai->component->name; + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + char pin_name[20]; + + if (strcmp(name, MAX98927_DEV0_NAME) && + strcmp(name, MAX98927_DEV1_NAME) && + strcmp(name, MAX98373_DEV0_NAME) && + strcmp(name, MAX98373_DEV1_NAME)) + continue; + + snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk", + codec_dai->component->name_prefix); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = snd_soc_dapm_enable_pin(dapm, pin_name); + if (ret) { + dev_err(rtd->dev, "failed to enable %s: %d\n", + pin_name, ret); + return ret; + } + snd_soc_dapm_sync(dapm); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = snd_soc_dapm_disable_pin(dapm, pin_name); + if (ret) { + dev_err(rtd->dev, "failed to disable %s: %d\n", + pin_name, ret); + return ret; + } + snd_soc_dapm_sync(dapm); + break; + } + } + + return 0; +} + static struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, + .trigger = kabylake_ssp0_trigger, }; static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, @@ -950,6 +1002,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); struct kbl_hdmi_pcm *pcm; + struct snd_soc_dapm_context *dapm = &card->dapm; struct snd_soc_component *component = NULL; int err, i = 0; char jack_name[NAME_SIZE]; @@ -976,9 +1029,25 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) if (!component) return -EINVAL; - return hdac_hdmi_jack_port_init(component, &card->dapm); - return 0; + err = hdac_hdmi_jack_port_init(component, &card->dapm); + + if (err < 0) + return err; + + err = snd_soc_dapm_disable_pin(dapm, "Left Spk"); + if (err) { + dev_err(card->dev, "failed to disable Left Spk: %d\n", err); + return err; + } + + err = snd_soc_dapm_disable_pin(dapm, "Right Spk"); + if (err) { + dev_err(card->dev, "failed to disable Right Spk: %d\n", err); + return err; + } + + return snd_soc_dapm_sync(dapm); } /* kabylake audio machine driver for SPT + DA7219 */ -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: Intel: boards: kbl_da7219_max98927: add dai_trigger function
by mac.chiang＠intel.com 03 May '19

03 May '19

From: Mac Chiang <mac.chiang(a)intel.com> amplifier feedback is not modeled as being dependent on any active output. Even when there is no playback happening, parts of the graph, specifically the IV sense->speaker protection->output remains active and this prevents the DSP from entering low-power states. This patch suggest a machine driver level approach where the speaker pins are enabled/disabled dynamically depending on stream start/stop events. DPAM graph representations show the feedback loop is indeed disabled and low-power states can be reached. Signed-off-by: Jenny TC <jenny.tc(a)intel.com> Signed-off-by: Harshapriya.n <harshapriya.n(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- Changelog: v4: - Changed device name in in kabylake_ssp0_trigger to address a conflict with recently merged patch MAXIM_DEV0_NAME -> MAX98927_DEV0_NAME ... v3: - fixed the return logic at the end of kabylake_card_late_probe() v2: - described clearly in commit message - added snd_soc_dapm_disable_pin in kabylake_card_late_probe() --- --- sound/soc/intel/boards/kbl_da7219_max98927.c | 73 +++++++++++++++++++++++++++- 1 file changed, 71 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index f72a7bf..1efe7fd 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -219,8 +219,60 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return 0; } +static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int j, ret; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + const char *name = codec_dai->component->name; + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + char pin_name[20]; + + if (strcmp(name, MAX98927_DEV0_NAME) && + strcmp(name, MAX98927_DEV1_NAME) && + strcmp(name, MAX98373_DEV0_NAME) && + strcmp(name, MAX98373_DEV1_NAME)) + continue; + + snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk", + codec_dai->component->name_prefix); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = snd_soc_dapm_enable_pin(dapm, pin_name); + if (ret) { + dev_err(rtd->dev, "failed to enable %s: %d\n", + pin_name, ret); + return ret; + } + snd_soc_dapm_sync(dapm); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = snd_soc_dapm_disable_pin(dapm, pin_name); + if (ret) { + dev_err(rtd->dev, "failed to disable %s: %d\n", + pin_name, ret); + return ret; + } + snd_soc_dapm_sync(dapm); + break; + } + } + + return 0; +} + static struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, + .trigger = kabylake_ssp0_trigger, }; static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, @@ -950,6 +1002,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); struct kbl_hdmi_pcm *pcm; + struct snd_soc_dapm_context *dapm = &card->dapm; struct snd_soc_component *component = NULL; int err, i = 0; char jack_name[NAME_SIZE]; @@ -976,9 +1029,25 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) if (!component) return -EINVAL; - return hdac_hdmi_jack_port_init(component, &card->dapm); - return 0; + err = hdac_hdmi_jack_port_init(component, &card->dapm); + + if (err < 0) + return err; + + err = snd_soc_dapm_disable_pin(dapm, "Left Spk"); + if (err) { + dev_err(card->dev, "failed to disable Left Spk: %d\n", err); + return err; + } + + err = snd_soc_dapm_disable_pin(dapm, "Right Spk"); + if (err) { + dev_err(card->dev, "failed to disable Right Spk: %d\n", err); + return err; + } + + return snd_soc_dapm_sync(dapm); } /* kabylake audio machine driver for SPT + DA7219 */ -- 2.7.4

1 0

[alsa-devel] [sound:topic/soc-cx2072x-5.2 1/2] sound/soc/codecs/cx2072x.c:597:18: sparse: sparse: cast to restricted __le32
by kbuild test robot 03 May '19

03 May '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/soc-cx2072x-5.2 head: 19e7a6a1d903117b3f4520690e7f399a53852c44 commit: b3d0d3217c3c84b473918b74c33f2d626c49d267 [1/2] ASoC: Add support for Conexant CX2071X CODEC reproduce: # apt-get install sparse git checkout b3d0d3217c3c84b473918b74c33f2d626c49d267 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/cx2072x.c:597:18: sparse: sparse: cast to restricted __le32 vim +597 sound/soc/codecs/cx2072x.c 564 565 static int cx2072x_reg_read(void *context, unsigned int reg, 566 unsigned int *value) 567 { 568 struct i2c_client *client = context; 569 struct device *dev = &client->dev; 570 unsigned int recv_buf = 0; 571 struct i2c_msg msgs[2]; 572 unsigned int size; 573 u8 send_buf[2]; 574 int ret; 575 576 size = cx2072x_register_size(reg); 577 578 send_buf[0] = reg >> 8; 579 send_buf[1] = reg & 0xff; 580 581 msgs[0].addr = client->addr; 582 msgs[0].len = sizeof(send_buf); 583 msgs[0].buf = send_buf; 584 msgs[0].flags = 0; 585 586 msgs[1].addr = client->addr; 587 msgs[1].len = size; 588 msgs[1].buf = (u8 *)&recv_buf; 589 msgs[1].flags = I2C_M_RD; 590 591 ret = i2c_transfer(client->adapter, msgs, ARRAY_SIZE(msgs)); 592 if (ret != ARRAY_SIZE(msgs)) { 593 dev_err(dev, "Failed to read register, ret = %d\n", ret); 594 return ret < 0 ? ret : -EIO; 595 } 596 > 597 *value = le32_to_cpu(recv_buf); 598 return 0; 599 } 600 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: RT5677-SPI: Disable 16Bit SPI Transfers
by Curtis Malainey 03 May '19

03 May '19

The current algorithm allows 3 types of transfers, 16bit, 32bit and burst. According to Realtek, 16bit transfers have a special restriction in that it is restricted to the memory region of 0x18020000 ~ 0x18021000. This region is the memory location of the I2C registers. The current algorithm does not uphold this restriction and therefore fails to complete writes. Since this has been broken for some time it likely no one is using it. Better to simply disable the 16 bit writes. This will allow users to properly load firmware over SPI without data corruption. CC: Ben Zhang <benzh(a)chromium.org> CC: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677-spi.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 167a02773a0b..b296e62fdbb4 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -58,13 +58,15 @@ static DEFINE_MUTEX(spi_mutex); * RT5677_SPI_READ/WRITE_32: Transfer 4 bytes * RT5677_SPI_READ/WRITE_BURST: Transfer any multiples of 8 bytes * - * For example, reading 260 bytes at 0x60030002 uses the following commands: - * 0x60030002 RT5677_SPI_READ_16 2 bytes + * Note: + * 16 Bit writes and reads are restricted to the address range + * 0x18020000 ~ 0x18021000 + * + * For example, reading 256 bytes at 0x60030004 uses the following commands: * 0x60030004 RT5677_SPI_READ_32 4 bytes * 0x60030008 RT5677_SPI_READ_BURST 240 bytes * 0x600300F8 RT5677_SPI_READ_BURST 8 bytes * 0x60030100 RT5677_SPI_READ_32 4 bytes - * 0x60030104 RT5677_SPI_READ_16 2 bytes * * Input: * @read: true for read commands; false for write commands @@ -79,15 +81,13 @@ static u8 rt5677_spi_select_cmd(bool read, u32 align, u32 remain, u32 *len) { u8 cmd; - if (align == 2 || align == 6 || remain == 2) { - cmd = RT5677_SPI_READ_16; - *len = 2; - } else if (align == 4 || remain <= 6) { + if (align == 4 || remain <= 4) { cmd = RT5677_SPI_READ_32; *len = 4; } else { cmd = RT5677_SPI_READ_BURST; - *len = min_t(u32, remain & ~7, RT5677_SPI_BURST_LEN); + *len = (((remain - 1) >> 3) + 1) << 3; + *len = min_t(u32, *len, RT5677_SPI_BURST_LEN); } return read ? cmd : cmd + 1; } @@ -108,7 +108,7 @@ static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen) } } -/* Read DSP address space using SPI. addr and len have to be 2-byte aligned. */ +/* Read DSP address space using SPI. addr and len have to be 4-byte aligned. */ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) { u32 offset; @@ -124,7 +124,7 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) if (!g_spi) return -ENODEV; - if ((addr & 1) || (len & 1)) { + if ((addr & 3) || (len & 3)) { dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len); return -EACCES; } @@ -159,8 +159,8 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) } EXPORT_SYMBOL_GPL(rt5677_spi_read); -/* Write DSP address space using SPI. addr has to be 2-byte aligned. - * If len is not 2-byte aligned, an extra byte of zero is written at the end +/* Write DSP address space using SPI. addr has to be 4-byte aligned. + * If len is not 4-byte aligned, then extra zeros are written at the end * as padding. */ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) @@ -178,7 +178,7 @@ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) if (!g_spi) return -ENODEV; - if (addr & 1) { + if (addr & 3) { dev_err(&g_spi->dev, "Bad write align 0x%x(%zu)\n", addr, len); return -EACCES; } -- 2.21.0.593.g511ec345e18-goog

3 2

[alsa-devel] Kernel patch needed for Asus Q325UAR
by ecmk 03 May '19

03 May '19

No sound combined with quickly fading sounds for a few seconds. Randomly working but with low quality distorted sound. The model needs to be added to the kernel patch listed bellow. Asus Q325UAR suffers from the same issues as discussed here: [alsa-devel] [PATCH] ALSA: hda/realtek: Enable audio jacks of ASUS UX391UA with, ALC294 https://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143282.… A detailed issue of the problem plus the solution is described here https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1784485?comments=all Thanks

2 6

[alsa-devel] [PATCH v3] ASoC: tlv320aic3x: Add support for high power analog output
by Saravanan Sekar 03 May '19

03 May '19

Add support to output level control for the analog high power output drivers HPOUT and HPCOM. Signed-off-by: Saravanan Sekar <sravanhome(a)gmail.com> --- Notes: Changes in V3: -Fixed compilation error Changes in V2: - Removed power control as it is handled by DAPM - Added level control for left channel sound/soc/codecs/tlv320aic3x.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 516d17cb2182..489a6d89d63d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -324,6 +324,9 @@ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 50, 0); */ static DECLARE_TLV_DB_SCALE(output_stage_tlv, -5900, 50, 1); +/* HP/HPCOM volumes. From 0 to 9 dB in 1 dB steps */ +static DECLARE_TLV_DB_SCALE(hp_tlv, 0, 100, 0); + static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Output */ SOC_DOUBLE_R_TLV("PCM Playback Volume", @@ -419,6 +422,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Pop reduction */ SOC_ENUM("Output Driver Power-On time", aic3x_poweron_time_enum), SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum), + + /* Analog HPOUT, HPCOM output level controls */ + SOC_DOUBLE_R_TLV("HP Playback Volume", HPLOUT_CTRL, HPROUT_CTRL, + 4, 9, 0, hp_tlv), + SOC_DOUBLE_R_TLV("HPCOM Playback Volume", HPLCOM_CTRL, HPRCOM_CTRL, + 4, 9, 0, hp_tlv), }; /* For other than tlv320aic3104 */ -- 2.17.1

2 1

[alsa-devel] [PATCH v4 0/2] ASoC: CX2072X codec support (revised)
by Takashi Iwai 02 May '19

02 May '19

Hi, this is a revised patchset for ASoC CX2072X codec support on Intel platforms. Takashi === v3->v4: Fix the wrong endianess conversion in reg write Minor code cleanups v2->v3: CX2072X codec driver: * Move register tables to appropriate place * Remove some confusing codes * Set snd_ctl_boolean_* helpers directly * Fix EQ put callback * Rename to "DAC1 Switch" from "DAC1 Mute Switch" * Drop superfluous regmap calls at shutdown * Avoid regmap_register_patch() * Add missing register definitions * Fix register access on big-endian machine * Remove regcache messes CX2072X machine driver: * Add Pierre's ack * Drop superfluous ssp0 routes v1->v2: Uncomment SOF entries in ACPI binding Move snd_soc_dai_set_bclk_ratio() call into init callback === Simon Ho (1): ASoC: Add support for Conexant CX2071X CODEC Takashi Iwai (1): ASoC: Intel: Add machine driver for CX2072X on BYT/CHT platforms sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2072x.c | 2151 +++++++++++++++++++++ sound/soc/codecs/cx2072x.h | 319 +++ sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcht_cx2072x.c | 284 +++ sound/soc/intel/common/soc-acpi-intel-byt-match.c | 8 + sound/soc/intel/common/soc-acpi-intel-cht-match.c | 8 + 9 files changed, 2792 insertions(+) create mode 100644 sound/soc/codecs/cx2072x.c create mode 100644 sound/soc/codecs/cx2072x.h create mode 100644 sound/soc/intel/boards/bytcht_cx2072x.c -- 2.16.4

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[alsa-devel] [PATCH v3 0/2] ASoC: CX2072X codec support (revised)
by Takashi Iwai 02 May '19

02 May '19

Hi, this is a revised patchset for ASoC CX2072X codec support on Intel platforms. v1->v2: Uncomment SOF entries in ACPI binding Move snd_soc_dai_set_bclk_ratio() call into init callback v2->v3: CX2072X codec driver: * Move register tables to appropriate place * Remove some confusing codes * Set snd_ctl_boolean_* helpers directly * Fix EQ put callback * Rename to "DAC1 Switch" from "DAC1 Mute Switch" * Drop superfluous regmap calls at shutdown * Avoid regmap_register_patch() * Add missing register definitions * Fix register access on big-endian machine * Remove regcache messes CX2072X machine driver: * Add Pierre's ack * Drop superfluous ssp0 routes Simon Ho (1): ASoC: Add support for Conexant CX2071X CODEC Takashi Iwai (1): ASoC: Intel: Add machine driver for CX2072X on BYT/CHT platforms sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2072x.c | 2158 +++++++++++++++++++++ sound/soc/codecs/cx2072x.h | 319 +++ sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcht_cx2072x.c | 284 +++ sound/soc/intel/common/soc-acpi-intel-byt-match.c | 8 + sound/soc/intel/common/soc-acpi-intel-cht-match.c | 8 + 9 files changed, 2799 insertions(+) create mode 100644 sound/soc/codecs/cx2072x.c create mode 100644 sound/soc/codecs/cx2072x.h create mode 100644 sound/soc/intel/boards/bytcht_cx2072x.c -- 2.16.4

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[alsa-devel] [GIT PULL]: Second soundwire updates
by Vinod Koul 02 May '19

02 May '19

Hi Greg, As discussed here is the second soundwire PULL request for v5.2-rc1. The following changes since commit 4abbd783d126cabfc20d1b8d50b1d5026b5cba09: soundwire: intel: fix implicit header use of module.h/export.h (2019-04-14 15:52:50 +0530) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-5.2-rc1_2 for you to fetch changes up to f3ed52bc798d7a5bc229d62374bf24173498092b: soundwire: remove multiple blank lines (2019-05-02 22:12:26 +0530) ---------------------------------------------------------------- 2nd soundwire updates for v5.2-rc1 This includes - the code cleanup series from Pierre - subsequent cleanups from me ---------------------------------------------------------------- Pierre-Louis Bossart (27): soundwire: Kconfig: fix help format soundwire: bus: fix alignment issues soundwire: bus: fix typos in comments soundwire: bus: fix boolean comparisons soundwire: bus: remove spurious newline soundwire: bus_type: fix alignment issues soundwire: mipi_disco: fix alignment issues soundwire: mipi_disco: fix boolean comparisons soundwire: stream: fix alignment issues soundwire: slave: fix alignment issues soundwire: intel_init: fix alignment issues soundwire: intel: fix alignment issues soundwire: intel: protect macro parameters soundwire: intel: fix boolean comparison soundwire: cadence_master: fix alignment issues soundwire: cadence_master: balance parentheses soundwire: cadence_master: fix boolean comparisons soundwire: cadence_master: remove spurious newline soundwire: fix SPDX license for header files soundwire: cadence: fix SPDX license for header files soundwire: intel: fix SPDX license for header files soundwire: fix alignment issues in header files soundwire: cadence: fix alignment issues in header files soundwire: bus: remove useless parentheses soundwire: add missing newlines in dynamic debug logs soundwire: cadence: add missing newlines in dynamic debug logs soundwire: intel: add missing newlines in dynamic debug logs Vinod Koul (14): soundwire: fix kconfig help format soundwire: fix SPDX license for header files soundwire: intel: fix SPDX license for header file soundwire: remove empty line before/after braces soundwire: cadence: remove empty line after braces soundwire: intel: remove empty line after braces soundwire: add argument to function definition soundwire: more alignment fixes soundwire: intel: more alignment fixes soundwire: avoid multiple assignments soundwire: fix more typos soundwire: wrap macro argument in parenthesis soundwire: add a blank line between functions soundwire: remove multiple blank lines drivers/soundwire/Kconfig | 4 +- drivers/soundwire/bus.c | 150 ++++++++--------- drivers/soundwire/bus.h | 16 +- drivers/soundwire/bus_type.c | 4 +- drivers/soundwire/cadence_master.c | 100 +++++------ drivers/soundwire/cadence_master.h | 22 +-- drivers/soundwire/intel.c | 135 ++++++++------- drivers/soundwire/intel.h | 4 +- drivers/soundwire/intel_init.c | 13 +- drivers/soundwire/mipi_disco.c | 122 +++++++------- drivers/soundwire/slave.c | 10 +- drivers/soundwire/stream.c | 283 ++++++++++++++++---------------- include/linux/soundwire/sdw.h | 16 +- include/linux/soundwire/sdw_intel.h | 6 +- include/linux/soundwire/sdw_registers.h | 5 +- include/linux/soundwire/sdw_type.h | 6 +- 16 files changed, 449 insertions(+), 447 deletions(-) Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH 00/14] soundwire: more code cleanups
by Vinod Koul 02 May '19

02 May '19

After applying cleanup from Pierre, I realized few more things can be cleaned up as well, so fixing these instance in the subsystem. Looks like bool is no longer encouraged, but I have skipped those and few alignment as they caused code to look worse. Greg, if you are okay I would like to add this as well (i know it is late, but they are cosmetic changes and no logic ones, let me know and I can send a updated PR) Vinod Koul (14): soundwire: fix kconfig help format soundwire: fix SPDX license for header files soundwire: intel: fix SPDX license for header file soundwire: remove empty line before/after braces soundwire: cadence: remove empty line after braces soundwire: intel: remove empty line after braces soundwire: add argument to function definition soundwire: more alignment fixes soundwire: intel: more alignment fixes soundwire: avoid multiple assignments soundwire: fix more typos soundwire: wrap macro argument in parenthesis soundwire: add a blank line between functions soundwire: remove multiple blank lines drivers/soundwire/Kconfig | 2 +- drivers/soundwire/bus.c | 17 +++++------- drivers/soundwire/cadence_master.c | 1 - drivers/soundwire/intel.c | 36 ++++++++++++------------- drivers/soundwire/intel_init.c | 1 - drivers/soundwire/mipi_disco.c | 6 ----- drivers/soundwire/stream.c | 28 +++++++++---------- include/linux/soundwire/sdw.h | 16 +++++------ include/linux/soundwire/sdw_intel.h | 6 ++--- include/linux/soundwire/sdw_registers.h | 5 ++-- include/linux/soundwire/sdw_type.h | 6 ++--- 11 files changed, 54 insertions(+), 70 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ASoC: Intel: kbl_da7219_max98357a: Map BTN_0 to KEY_PLAYPAUSE
by "mac.chiang＠intel.com--cc=pierre-louis.bossart"＠linux.intel.com 02 May '19

02 May '19

From: Mac Chiang <mac.chiang(a)intel.com> On kbl_rt5663_max98927, commit 38a5882e4292 ("ASoC: Intel: kbl_rt5663_max98927: Map BTN_0 to KEY_PLAYPAUSE") This key pair mapping to play/pause when playing Youtube The Android 3.5mm Headset jack specification mentions that BTN_0 should be mapped to KEY_MEDIA, but this is less logical than KEY_PLAYPAUSE, which has much broader userspace support. For example, the Chrome OS userspace now supports KEY_PLAYPAUSE to toggle play/pause of videos and audio, but does not handle KEY_MEDIA. Furthermore, Android itself now supports KEY_PLAYPAUSE equivalently, as the new USB headset spec requires KEY_PLAYPAUSE for BTN_0. https://source.android.com/devices/accessories/headset/usb-headset-spec The same fix is required on Chrome kbl_da7219_max98357a. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Reviewed-by: Benson Leung <bleung(a)chromium.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 38f6ab7..07491a0 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -188,7 +188,7 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); -- 2.7.4

2 1

[alsa-devel] [PATCH 1/5] pcm: direct: Add generic hw_ptr_alignment function for dmix, dshare and dsnoop
by vanitha.channaiah＠in.bosch.com 02 May '19

02 May '19

From: Vanitha Channaiah <vanitha.channaiah(a)in.bosch.com> This patch is same as the fix in below commit commit 6b058fda9dce ("pcm: dmix: Add option to allow alignment of slave pointers") Signed-off-by: Vanitha Channaiah <vanitha.channaiah(a)in.bosch.com> --- src/pcm/pcm_direct.c | 19 +++++++++++++++++++ src/pcm/pcm_direct.h | 8 ++++++++ src/pcm/pcm_dmix.c | 25 ++----------------------- 3 files changed, 29 insertions(+), 23 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 666a8ce..411a035 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -2040,3 +2040,22 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, return 0; } + +void snd_pcm_direct_reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) +{ + dmix->slave_appl_ptr = dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; + + if (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_ROUNDUP || + (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_AUTO && + pcm->buffer_size <= pcm->period_size * 2)) + dmix->slave_appl_ptr = + ((dmix->slave_appl_ptr + dmix->slave_period_size - 1) / + dmix->slave_period_size) * dmix->slave_period_size; + else if (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_ROUNDDOWN || + (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_AUTO && + (dmix->slave_period_size * SEC_TO_MS) / + pcm->rate < LOW_LATENCY_PERIOD_TIME)) + dmix->slave_appl_ptr = dmix->slave_hw_ptr = + ((dmix->slave_hw_ptr / dmix->slave_period_size) * + dmix->slave_period_size); +} diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index da5e280..a71aab1 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -24,6 +24,11 @@ #define DIRECT_IPC_SEMS 1 #define DIRECT_IPC_SEM_CLIENT 0 +/* Seconds representing in Milli seconds */ +#define SEC_TO_MS 1000 +/* slave_period time for low latency requirements in ms */ +#define LOW_LATENCY_PERIOD_TIME 10 + typedef void (mix_areas_t)(unsigned int size, volatile void *dst, void *src, @@ -257,6 +262,8 @@ struct snd_pcm_direct { snd1_pcm_direct_get_chmap #define snd_pcm_direct_set_chmap \ snd1_pcm_direct_set_chmap +#define snd_pcm_direct_reset_slave_ptr \ + snd1_pcm_direct_reset_slave_ptr int snd_pcm_direct_semaphore_create_or_connect(snd_pcm_direct_t *dmix); @@ -341,6 +348,7 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct); int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm); int snd_timer_async(snd_timer_t *timer, int sig, pid_t pid); struct timespec snd_pcm_hw_fast_tstamp(snd_pcm_t *pcm); +void snd_pcm_direct_reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix); struct snd_pcm_direct_open_conf { key_t ipc_key; diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index c5592cd..dcde40d 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -55,9 +55,6 @@ const char *_snd_module_pcm_dmix = ""; #define STATE_RUN_PENDING 1024 #endif -#define SEC_TO_MS 1000 /* Seconds representing in Milli seconds */ -#define LOW_LATENCY_PERIOD_TIME 10 /* slave_period time for low latency requirements in ms */ - /* * */ @@ -560,30 +557,12 @@ static int snd_pcm_dmix_hwsync(snd_pcm_t *pcm) } } -static void reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) -{ - dmix->slave_appl_ptr = dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; - - if (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_ROUNDUP || - (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_AUTO && - pcm->buffer_size <= pcm->period_size * 2)) - dmix->slave_appl_ptr = - ((dmix->slave_appl_ptr + dmix->slave_period_size - 1) - / dmix->slave_period_size) * dmix->slave_period_size; - else if (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_ROUNDDOWN || - (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_AUTO && - (dmix->slave_period_size * SEC_TO_MS) / pcm->rate < LOW_LATENCY_PERIOD_TIME)) - dmix->slave_appl_ptr = dmix->slave_hw_ptr = - ((dmix->slave_hw_ptr / dmix->slave_period_size) * - dmix->slave_period_size); -} - static int snd_pcm_dmix_reset(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; dmix->hw_ptr %= pcm->period_size; dmix->appl_ptr = dmix->last_appl_ptr = dmix->hw_ptr; - reset_slave_ptr(pcm, dmix); + snd_pcm_direct_reset_slave_ptr(pcm, dmix); return 0; } @@ -592,7 +571,7 @@ static int snd_pcm_dmix_start_timer(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) int err; snd_pcm_hwsync(dmix->spcm); - reset_slave_ptr(pcm, dmix); + snd_pcm_direct_reset_slave_ptr(pcm, dmix); err = snd_timer_start(dmix->timer); if (err < 0) return err; -- 2.7.4

2 9

[alsa-devel] [PATCH v4 00/22] soundwire: code cleanup
by Pierre-Louis Bossart 02 May '19

02 May '19

SoundWire support will be provided in Linux with the Sound Open Firmware (SOF) on Intel platforms. Before we start adding the missing pieces, there are a number of warnings and style issues reported by checkpatch, cppcheck and Coccinelle that need to be cleaned-up. Changes since v3: patch 1,3,4 were merged for 5.2-rc1 No code change, only split patch2 in 21 patches to make Vinod happy. Each patch only fixes a specific issue. patch 5 is now patch22 and wasn't changed. Not sure why Vinod reported the patch didn't apply. Added Takashi's Reviewed-by tag in all patches since the code is exactly the same as in v3. Changes since v2: fixed inversion of devm_kcalloc parameters, detected while rebasing additional patches. Changes since v1: added missing newlines in new patch (suggested by Joe Perches) Pierre-Louis Bossart (22): soundwire: Kconfig: fix help format soundwire: fix SPDX license for header files soundwire: fix alignment issues in header files soundwire: bus: fix alignment issues soundwire: bus: fix typos in comments soundwire: bus: remove useless parentheses soundwire: bus: fix boolean comparisons soundwire: bus: remove spurious newline soundwire: bus_type: fix alignment issues soundwire: mipi_disco: fix alignment issues soundwire: mipi_disco: fix boolean comparisons soundwire: stream: fix alignment issues soundwire: slave: fix alignment issues soundwire: intel_init: fix alignment issues soundwire: intel: fix alignment issues soundwire: intel: protect macro parameters soundwire: intel: fix boolean comparison soundwire: cadence_master: fix alignment issues soundwire: cadence_master: balance parentheses soundwire: cadence_master: fix boolean comparisons soundwire: cadence_master: remove spurious newline soundwire: add missing newlines in dynamic debug logs drivers/soundwire/Kconfig | 2 +- drivers/soundwire/bus.c | 135 ++++++++------- drivers/soundwire/bus.h | 16 +- drivers/soundwire/bus_type.c | 4 +- drivers/soundwire/cadence_master.c | 99 +++++------ drivers/soundwire/cadence_master.h | 22 +-- drivers/soundwire/intel.c | 99 ++++++----- drivers/soundwire/intel.h | 4 +- drivers/soundwire/intel_init.c | 12 +- drivers/soundwire/mipi_disco.c | 116 +++++++------ drivers/soundwire/slave.c | 10 +- drivers/soundwire/stream.c | 265 +++++++++++++++-------------- 12 files changed, 401 insertions(+), 383 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH][next] ASoC: SOF: remove redundant null checks of dai
by Colin King 02 May '19

02 May '19

From: Colin Ian King <colin.king(a)canonical.com> Currently there are two null checks of pointer dai in function sof_connect_dai_widget and yet there is no null check of dai at the end of the function when checking !dai->name. The latter would be a null pointer deference if dai is null (as picked up by static analysis), however the function is only ever called when dai is successfully allocated, so the null checks are redundant. Clean up the code by removing the null checks. Addresses-Coverity: ("Dereference after null check") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sof/topology.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 2b9de1b97447..1f71857298a9 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1127,15 +1127,13 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp, switch (w->id) { case snd_soc_dapm_dai_out: rtd->cpu_dai->capture_widget = w; - if (dai) - dai->name = rtd->dai_link->name; + dai->name = rtd->dai_link->name; dev_dbg(sdev->dev, "tplg: connected widget %s -> DAI link %s\n", w->name, rtd->dai_link->name); break; case snd_soc_dapm_dai_in: rtd->cpu_dai->playback_widget = w; - if (dai) - dai->name = rtd->dai_link->name; + dai->name = rtd->dai_link->name; dev_dbg(sdev->dev, "tplg: connected widget %s -> DAI link %s\n", w->name, rtd->dai_link->name); break; -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: da7219: Update the support rate list
by Yu-Hsuan Hsu 02 May '19

02 May '19

If we want to set rate to 64000 on da7219, it fails and returns "snd_pcm_hw_params: Invalid argument". We should remove 64000 from support rate list because it is not available. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/da7219.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 5f5fa3416af3..7497457cf3d4 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1658,20 +1658,26 @@ static const struct snd_soc_dai_ops da7219_dai_ops = { #define DA7219_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +#define DA7219_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + static struct snd_soc_dai_driver da7219_dai = { .name = "da7219-hifi", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = DA7219_DAI_CH_NUM_MAX, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = DA7219_RATES, .formats = DA7219_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = DA7219_DAI_CH_NUM_MAX, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = DA7219_RATES, .formats = DA7219_FORMATS, }, .ops = &da7219_dai_ops, -- 2.21.0.593.g511ec345e18-goog

2 1

[alsa-devel] [PATCH] ASoC: Intel: kbl_da7219_max98357a: Map BTN_0 to KEY_PLAYPAUSE
by mac.chiang＠intel.com 02 May '19

02 May '19

From: Mac Chiang <mac.chiang(a)intel.com> On kbl_rt5663_max98927, commit 38a5882e4292 ("ASoC: Intel: kbl_rt5663_max98927: Map BTN_0 to KEY_PLAYPAUSE") This key pair mapping to play/pause when playing Youtube The Android 3.5mm Headset jack specification mentions that BTN_0 should be mapped to KEY_MEDIA, but this is less logical than KEY_PLAYPAUSE, which has much broader userspace support. For example, the Chrome OS userspace now supports KEY_PLAYPAUSE to toggle play/pause of videos and audio, but does not handle KEY_MEDIA. Furthermore, Android itself now supports KEY_PLAYPAUSE equivalently, as the new USB headset spec requires KEY_PLAYPAUSE for BTN_0. https://source.android.com/devices/accessories/headset/usb-headset-spec The same fix is required on Chrome kbl_da7219_max98357a. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Reviewed-by: Benson Leung <bleung(a)chromium.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 38f6ab7..07491a0 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -188,7 +188,7 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); -- 2.7.4

1 0

[alsa-devel] [PATCH v2 0/2] ASoC: CX2072X codec support (revised)
by Takashi Iwai 02 May '19

02 May '19

Hi, this is a revised patchset for ASoC CX2072X codec support on Intel platforms. v1->v2: Uncomment SOF entries in ACPI binding Move snd_soc_dai_set_bclk_ratio() call into init callback === It's been long time ago the first version of CX2072X codec patch was submitted, and I hoped that it'd be resubmitted, but never happened, as the original author seems to have left, unfortunately. Since I have such a laptop (ASUS E200HA), I took some time to cook and brush up the patches. The patches are for 5.2, should be applicable to ASoC tree for-next branch. Will post the corresponding UCM profile patch once when this gets accepted. thanks, Takashi === Simon Ho (1): ASoC: Add support for Conexant CX2072X CODEC Takashi Iwai (1): ASoC: Intel: Add machine driver for CX2072X on BYT/CHT platforms sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2072x.c | 2197 +++++++++++++++++++++ sound/soc/codecs/cx2072x.h | 317 +++ sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcht_cx2072x.c | 286 +++ sound/soc/intel/common/soc-acpi-intel-byt-match.c | 8 + sound/soc/intel/common/soc-acpi-intel-cht-match.c | 8 + 9 files changed, 2838 insertions(+) create mode 100644 sound/soc/codecs/cx2072x.c create mode 100644 sound/soc/codecs/cx2072x.h create mode 100644 sound/soc/intel/boards/bytcht_cx2072x.c -- 2.16.4

3 5

[alsa-devel] [PATCH] ASoC: rt5645: Headphone Jack sense inverts on the LattePanda board
by Hui Wang 02 May '19

02 May '19

The LattePanda board has a sound card chtrt5645, when there is nothing plugged in the headphone jack, the system thinks the headphone is plugged in, while we plug a headphone in the jack, the system thinks the headphone is unplugged. If adding quirk=0x21 in the module parameter, the headphone jack can work well. So let us fix it via platform_data. https://bugs.launchpad.net/bugs/182459 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/codecs/rt5645.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9a0751978090..a6064606424a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3631,6 +3631,11 @@ static const struct rt5645_platform_data jd_mode3_platform_data = { .jd_mode = 3, }; +static const struct rt5645_platform_data lattepanda_board_platform_data = { + .jd_mode = 2, + .inv_jd1_1 = true +}; + static const struct dmi_system_id dmi_platform_data[] = { { .ident = "Chrome Buddy", @@ -3728,6 +3733,15 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&intel_braswell_platform_data, }, + { + .ident = "LattePanda board", + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"), + DMI_EXACT_MATCH(DMI_BOARD_VERSION, "Default string"), + }, + .driver_data = (void *)&lattepanda_board_platform_data, + }, { } }; -- 2.17.1

2 1

[alsa-devel] [PATCH v2] ASoC: da7219: Use clk_round_rate to handle enabled bclk/wclk case
by Adam Thomson 02 May '19

02 May '19

For some platforms where DA7219 is the DAI clock master, BCLK/WCLK will be set and enabled prior to the codec's hw_params() function being called. It is possible the platform requires a different BCLK configuration than would be chosen by hw_params(), for example S16_LE format needed with a 64-bit frame to satisfy certain devices using the clocks. To handle those kinds of scenarios, the use of clk_round_rate() is now employed as part of hw_params(). If BCLK is already enabled then this function will just return the currently set rate, if it is valid for the desired frame size, so the subsequent call to clk_set_rate() will succeed and nothing changes with regards to clocking. In addition the specific BCLK & WCLK recalc_rate() implementations needed updating to always give back a real value, as those functions are called as part of the clk init code and a real value is needed for the clk_round_rate() call to work as expected. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- Changes in v2: - Removed clk_round_rate() call for WCLK/SR as this is not sensible. - Added checking after clk_round_rate() call for BCLK to verify returne rate is suitable for the hw_params() desired frame size. sound/soc/codecs/da7219.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 5f5fa34..206d01c 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1621,6 +1621,21 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, if (bclk) { bclk_rate = frame_size * sr; + /* + * Rounding the rate here avoids failure trying to set a + * new rate on an already enabled bclk. In that + * instance this will just set the same rate as is + * currently in use, and so should continue without + * problem, as long as the BCLK rate is suitable for the + * desired frame size. + */ + bclk_rate = clk_round_rate(bclk, bclk_rate); + if ((bclk_rate / sr) < frame_size) { + dev_err(component->dev, + "BCLK rate mismatch against frame size"); + return -EINVAL; + } + ret = clk_set_rate(bclk, bclk_rate); if (ret) { dev_err(component->dev, @@ -1927,9 +1942,6 @@ static unsigned long da7219_wclk_recalc_rate(struct clk_hw *hw, struct snd_soc_component *component = da7219->component; u8 fs = snd_soc_component_read32(component, DA7219_SR); - if (!da7219->master) - return 0; - switch (fs & DA7219_SR_MASK) { case DA7219_SR_8000: return 8000; @@ -2016,9 +2028,6 @@ static unsigned long da7219_bclk_recalc_rate(struct clk_hw *hw, u8 bclks_per_wclk = snd_soc_component_read32(component, DA7219_DAI_CLK_MODE); - if (!da7219->master) - return 0; - switch (bclks_per_wclk & DA7219_DAI_BCLKS_PER_WCLK_MASK) { case DA7219_DAI_BCLKS_PER_WCLK_32: return parent_rate * 32; -- 1.9.1

2 1

[alsa-devel] [PATCH] spi: AD ASoC: declare missing of table
by Daniel Gomez 02 May '19

02 May '19

Add missing <of_device_id> table for SPI driver relying on SPI device match since compatible is in a DT binding or in a DTS. Before this patch: modinfo sound/soc/codecs/snd-soc-adau1977-spi.ko | grep alias alias: spi:adau1979 alias: spi:adau1978 alias: spi:adau1977 After this patch: modinfo sound/soc/codecs/snd-soc-adau1977-spi.ko | grep alias alias: of:N*T*Cadi,adau1979C* alias: of:N*T*Cadi,adau1979 alias: of:N*T*Cadi,adau1978C* alias: of:N*T*Cadi,adau1978 alias: of:N*T*Cadi,adau1977C* alias: of:N*T*Cadi,adau1977 alias: spi:adau1979 alias: spi:adau1978 alias: spi:adau1977 Reported-by: Javier Martinez Canillas <javier(a)dowhile0.org> Signed-off-by: Daniel Gomez <dagmcr(a)gmail.com> --- sound/soc/codecs/adau1977-spi.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/adau1977-spi.c b/sound/soc/codecs/adau1977-spi.c index 84ffbde..2baf615 100644 --- a/sound/soc/codecs/adau1977-spi.c +++ b/sound/soc/codecs/adau1977-spi.c @@ -10,6 +10,8 @@ #include <linux/mod_devicetable.h> #include <linux/module.h> #include <linux/regmap.h> +#include <linux/of.h> +#include <linux/of_device.h> #include <linux/spi/spi.h> #include <sound/soc.h> @@ -54,9 +56,18 @@ static const struct spi_device_id adau1977_spi_ids[] = { }; MODULE_DEVICE_TABLE(spi, adau1977_spi_ids); +static const struct of_device_id adau1977_spi_of_match[] = { + { .compatible = "adi,adau1977" }, + { .compatible = "adi,adau1978" }, + { .compatible = "adi,adau1979" }, + { }, +}; +MODULE_DEVICE_TABLE(of, adau1977_spi_of_match); + static struct spi_driver adau1977_spi_driver = { .driver = { .name = "adau1977", + .of_match_table = of_match_ptr(adau1977_spi_of_match), }, .probe = adau1977_spi_probe, .id_table = adau1977_spi_ids, -- 2.7.4

2 3

[alsa-devel] Applied "ASoC: hdmi-codec: fix spelling mistake "plalform" -> "platform"" to the asoc tree
by Mark Brown 02 May '19

02 May '19

The patch ASoC: hdmi-codec: fix spelling mistake "plalform" -> "platform" 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 7b6531c5054e7804ccce25f389a2d4810357f5c9 Mon Sep 17 00:00:00 2001 From: Colin Ian King <colin.king(a)canonical.com> Date: Wed, 17 Apr 2019 17:26:28 +0100 Subject: [PATCH] ASoC: hdmi-codec: fix spelling mistake "plalform" -> "platform" There is a spelling mistake in a dev_err message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 35df73e42cbc..b9d9dde9fbaf 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -757,7 +757,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) dev_dbg(dev, "%s()\n", __func__); if (!hcd) { - dev_err(dev, "%s: No plalform data\n", __func__); + dev_err(dev, "%s: No platform data\n", __func__); return -EINVAL; } -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: Mediatek: MT8183: Fix build err while CONFIG_I2C set to module
by Yue Haibing 02 May '19

02 May '19

From: YueHaibing <yuehaibing(a)huawei.com> During randconfig builds, I occasionally run into an invalid configuration WARNING: unmet direct dependencies detected for SND_SOC_TS3A227E Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=m] Selected by [y]: - SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_MT8183 [=y] sound/soc/codecs/ts3a227e.o: In function `ts3a227e_i2c_probe': ts3a227e.c:(.text+0x684): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/ts3a227e.o: In function `ts3a227e_driver_init': ts3a227e.c:(.init.text+0x18): undefined reference to `i2c_register_driver' sound/soc/codecs/ts3a227e.o: In function `ts3a227e_driver_exit': ts3a227e.c:(.exit.text+0x14): undefined reference to `i2c_del_driver' This patch add I2C dependency to fix this. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: ebbddc75bbe8 ("ASoC: Mediatek: MT8183: Add machine driver with DA7219") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/mediatek/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 874404b..f70b710 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -118,6 +118,7 @@ config SND_SOC_MT8183 config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A tristate "ASoC Audio driver for MT8183 with MT6358 TS3A227E MAX98357A codec" + depends on I2C depends on SND_SOC_MT8183 select SND_SOC_MT6358 select SND_SOC_MAX98357A -- 2.7.4

2 2

[alsa-devel] [v3] ASoC: AMD: Configure wclk and bclk of master codec
by Agrawal, Akshu 02 May '19

02 May '19

With CCF support in da7219, we can now set the correct rate of wclk and bclk. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Removed clk set rate and enable/disable from da7219 ops as da7219 codec takes care of them internally. Clock configuration kept for those codecs where da7219 acts as master of clock. As suggested by Adam Thomson. v3: As per spec, adau7002 bclk to be minimum of 64x of wclk. Hence, limiting bclk for all codecs. Also, because of this changes done in v2 of this patch have to be brought back. sound/soc/amd/acp-da7219-max98357a.c | 66 ++++++++++++++++++---------- 1 file changed, 42 insertions(+), 24 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 8a619a75b3a9..16b0ea3a3d72 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -46,7 +46,8 @@ #define DUAL_CHANNEL 2 static struct snd_soc_jack cz_jack; -static struct clk *da7219_dai_clk; +static struct clk *da7219_dai_wclk; +static struct clk *da7219_dai_bclk; extern bool bt_uart_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) @@ -72,7 +73,8 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } - da7219_dai_clk = clk_get(component->dev, "da7219-dai-bclk"); + da7219_dai_wclk = clk_get(component->dev, "da7219-dai-wclk"); + da7219_dai_bclk = clk_get(component->dev, "da7219-dai-bclk"); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | @@ -94,12 +96,15 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int da7219_clk_enable(struct snd_pcm_substream *substream) +static int da7219_clk_enable(struct snd_pcm_substream *substream, + int wclk_rate, int bclk_rate) { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - ret = clk_prepare_enable(da7219_dai_clk); + clk_set_rate(da7219_dai_wclk, wclk_rate); + clk_set_rate(da7219_dai_bclk, bclk_rate); + ret = clk_prepare_enable(da7219_dai_bclk); if (ret < 0) { dev_err(rtd->dev, "can't enable master clock %d\n", ret); return ret; @@ -110,7 +115,7 @@ static int da7219_clk_enable(struct snd_pcm_substream *substream) static void da7219_clk_disable(void) { - clk_disable_unprepare(da7219_dai_clk); + clk_disable_unprepare(da7219_dai_bclk); } static const unsigned int channels[] = { @@ -151,7 +156,7 @@ static int cz_da7219_play_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_SP_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) @@ -173,12 +178,7 @@ static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; - return da7219_clk_enable(substream); -} - -static void cz_da7219_shutdown(struct snd_pcm_substream *substream) -{ - da7219_clk_disable(); + return 0; } static int cz_max_startup(struct snd_pcm_substream *substream) @@ -199,12 +199,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_BT_INSTANCE; - return da7219_clk_enable(substream); -} - -static void cz_max_shutdown(struct snd_pcm_substream *substream) -{ - da7219_clk_disable(); + return 0; } static int cz_dmic0_startup(struct snd_pcm_substream *substream) @@ -225,7 +220,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->cap_i2s_instance = I2S_BT_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static int cz_dmic1_startup(struct snd_pcm_substream *substream) @@ -247,10 +242,28 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; - return da7219_clk_enable(substream); + return 0; } -static void cz_dmic_shutdown(struct snd_pcm_substream *substream) +static int cz_da7219_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int wclk, bclk; + + wclk = params_rate(params); + bclk = wclk * params_channels(params) * + snd_pcm_format_width(params_format(params)); + /* ADAU7002 spec: "The ADAU7002 requires a BCLK rate + * that is minimum of 64x the LRCLK sample rate." + * DA7219 is the only clk source so for all codecs + * we have to limit bclk to 64X lrclk. + */ + if (bclk < (wclk * 64)) + bclk = wclk * 64; + return da7219_clk_enable(substream, wclk, bclk); +} + +static void cz_da7219_shutdown(struct snd_pcm_substream *substream) { da7219_clk_disable(); } @@ -258,26 +271,31 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_da7219_cap_ops = { .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, - .shutdown = cz_max_shutdown, + .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, - .shutdown = cz_dmic_shutdown, + .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, - .shutdown = cz_dmic_shutdown, + .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static struct snd_soc_dai_link cz_dai_7219_98357[] = { -- 2.17.1

2 2

[alsa-devel] [PATCH] ASoC: pcm3168a: Enable TDM support for DSP_A/B modes
by Peter Ujfalusi 02 May '19

02 May '19

The 24-bit TDM mode also applies to DSP_A and DSP_B modes. Most dais on the SoC side can not interpret I2S/Left_j with other than 2 channels of audio. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/pcm3168a.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 08d3fe192e65..e0d5839fe1a7 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -457,13 +457,16 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, if (chan > 2) { switch (fmt) { case PCM3168A_FMT_I2S: + case PCM3168A_FMT_DSP_A: fmt = PCM3168A_FMT_I2S_TDM; break; case PCM3168A_FMT_LEFT_J: + case PCM3168A_FMT_DSP_B: fmt = PCM3168A_FMT_LEFT_J_TDM; break; default: - dev_err(component->dev, "TDM is supported under I2S/Left_J only\n"); + dev_err(component->dev, + "TDM is supported under DSP/I2S/Left_J only\n"); return -EINVAL; } } @@ -526,6 +529,8 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, break; case PCM3168A_FMT_LEFT_J: case PCM3168A_FMT_I2S: + case PCM3168A_FMT_DSP_A: + case PCM3168A_FMT_DSP_B: sample_min = 24; channel_max = channel_maxs[tx]; break; -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 2

[alsa-devel] [PATCH -next] ASoC: fsl_micfil: Remove set but not used variable 'osr'
by Yue Haibing 02 May '19

02 May '19

From: YueHaibing <yuehaibing(a)huawei.com> Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/fsl/fsl_micfil.c: In function 'get_clk_div': sound/soc/fsl/fsl_micfil.c:154:6: warning: variable 'osr' set but not used [-Wunused-but-set-variable] It is never used since introduction in commit 47a70e6fc9a8 ("ASoC: Add MICFIL SoC Digital Audio Interface driver.") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/fsl_micfil.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 40c07e7..f7f2d29 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -151,12 +151,9 @@ static inline int get_clk_div(struct fsl_micfil *micfil, { u32 ctrl2_reg; long mclk_rate; - int osr; int clk_div; regmap_read(micfil->regmap, REG_MICFIL_CTRL2, &ctrl2_reg); - osr = 16 - ((ctrl2_reg & MICFIL_CTRL2_CICOSR_MASK) - >> MICFIL_CTRL2_CICOSR_SHIFT); mclk_rate = clk_get_rate(micfil->mclk); -- 2.7.4

5 6

[alsa-devel] [PATCH -next] ASoC: tlv320aic32x4: Remove set but not used variable 'mclk_rate'
by Yue Haibing 02 May '19

02 May '19

From: YueHaibing <yuehaibing(a)huawei.com> Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_setup_clocks': sound/soc/codecs/tlv320aic32x4.c:669:16: warning: variable 'mclk_rate' set but not used [-Wunused-but-set-variable] It is not used since introduction in commit 96c3bb00239d ("ASoC: tlv320aic32x4: Dynamically Determine Clocking") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/tlv320aic32x4.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 6edee05..83608f3 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -684,9 +684,8 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, u8 madc, nadc, mdac, ndac, max_nadc, min_mdac, max_ndac; u8 dosr_increment; u16 max_dosr, min_dosr; - unsigned long mclk_rate, adc_clock_rate, dac_clock_rate; + unsigned long adc_clock_rate, dac_clock_rate; int ret; - struct clk *mclk; struct clk_bulk_data clocks[] = { { .id = "pll" }, @@ -700,9 +699,6 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, if (ret) return ret; - mclk = clk_get_parent(clocks[1].clk); - mclk_rate = clk_get_rate(mclk); - if (sample_rate <= 48000) { aosr = 128; adc_resource_class = 6; -- 2.7.4

2 2

[alsa-devel] [PATCH -next] ASoC: amd: acp3x: Make acp3x_dai_i2s_ops static
by Yue Haibing 02 May '19

02 May '19

From: YueHaibing <yuehaibing(a)huawei.com> Fix sparse warning: sound/soc/amd/raven/acp3x-pcm-dma.c:561:24: warning: symbol 'acp3x_dai_i2s_ops' was not declared. Should it be static? Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 1a2e15f..9775bda 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -558,7 +558,7 @@ static int acp3x_dai_i2s_trigger(struct snd_pcm_substream *substream, return ret; } -struct snd_soc_dai_ops acp3x_dai_i2s_ops = { +static struct snd_soc_dai_ops acp3x_dai_i2s_ops = { .hw_params = acp3x_dai_i2s_hwparams, .trigger = acp3x_dai_i2s_trigger, .set_fmt = acp3x_dai_i2s_set_fmt, -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: imx: fix fiq dependencies
by Arnd Bergmann 02 May '19

02 May '19

During randconfig builds, I occasionally run into an invalid configuration of the freescale FIQ sound support: WARNING: unmet direct dependencies detected for SND_SOC_IMX_PCM_FIQ Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m] Selected by [y]: - SND_SOC_FSL_SPDIF [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m]!=n && (MXC_TZIC [=n] || MXC_AVIC [=y]) sound/soc/fsl/imx-ssi.o: In function `imx_ssi_remove': imx-ssi.c:(.text+0x28): undefined reference to `imx_pcm_fiq_exit' sound/soc/fsl/imx-ssi.o: In function `imx_ssi_probe': imx-ssi.c:(.text+0xa64): undefined reference to `imx_pcm_fiq_init' The Kconfig warning is a result of the symbol being defined inside of the "if SND_IMX_SOC" block, and is otherwise harmless. The link error is more tricky and happens with SND_SOC_IMX_SSI=y, which may or may not imply FIQ support. However, if SND_SOC_FSL_SSI is set to =m at the same time, that selects SND_SOC_IMX_PCM_FIQ as a loadable module dependency, which then causes a link failure from imx-ssi. The solution here is to make SND_SOC_IMX_PCM_FIQ built-in whenever one of its potential users is built-in. Fixes: ff40260f79dc ("ASoC: fsl: refine DMA/FIQ dependencies") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/fsl/Kconfig | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7b1d9970be8b..1f65cf555ebe 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -182,16 +182,17 @@ config SND_MPC52xx_SOC_EFIKA endif # SND_POWERPC_SOC +config SND_SOC_IMX_PCM_FIQ + tristate + default y if SND_SOC_IMX_SSI=y && (SND_SOC_FSL_SSI=m || SND_SOC_FSL_SPDIF=m) && (MXC_TZIC || MXC_AVIC) + select FIQ + if SND_IMX_SOC config SND_SOC_IMX_SSI tristate select SND_SOC_FSL_UTILS -config SND_SOC_IMX_PCM_FIQ - tristate - select FIQ - comment "SoC Audio support for Freescale i.MX boards:" config SND_MXC_SOC_WM1133_EV1 -- 2.20.0

2 2

[alsa-devel] [PATCH -next] ASoC: atmel: tse850: Make some functions static
by Yue Haibing 02 May '19

02 May '19

From: YueHaibing <yuehaibing(a)huawei.com> Fix sparse warnings: sound/soc/atmel/tse850-pcm5142.c:120:5: warning: symbol 'tse850_get_mix' was not declared. Should it be static? sound/soc/atmel/tse850-pcm5142.c:132:5: warning: symbol 'tse850_put_mix' was not declared. Should it be static? sound/soc/atmel/tse850-pcm5142.c:154:5: warning: symbol 'tse850_get_ana' was not declared. Should it be static? sound/soc/atmel/tse850-pcm5142.c:187:5: warning: symbol 'tse850_put_ana' was not declared. Should it be static? Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/atmel/tse850-pcm5142.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 214adca..ae44518 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -117,8 +117,8 @@ static int tse850_put_mux2(struct snd_kcontrol *kctrl, return snd_soc_dapm_put_enum_double(kctrl, ucontrol); } -int tse850_get_mix(struct snd_kcontrol *kctrl, - struct snd_ctl_elem_value *ucontrol) +static int tse850_get_mix(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kctrl); struct snd_soc_card *card = dapm->card; @@ -129,8 +129,8 @@ int tse850_get_mix(struct snd_kcontrol *kctrl, return 0; } -int tse850_put_mix(struct snd_kcontrol *kctrl, - struct snd_ctl_elem_value *ucontrol) +static int tse850_put_mix(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kctrl); struct snd_soc_card *card = dapm->card; @@ -151,8 +151,8 @@ int tse850_put_mix(struct snd_kcontrol *kctrl, return 1; } -int tse850_get_ana(struct snd_kcontrol *kctrl, - struct snd_ctl_elem_value *ucontrol) +static int tse850_get_ana(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kctrl); struct snd_soc_card *card = dapm->card; @@ -184,8 +184,8 @@ int tse850_get_ana(struct snd_kcontrol *kctrl, return 0; } -int tse850_put_ana(struct snd_kcontrol *kctrl, - struct snd_ctl_elem_value *ucontrol) +static int tse850_put_ana(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kctrl); struct snd_soc_card *card = dapm->card; -- 2.7.4

2 2

[alsa-devel] [PATCH 0/2] Fix invalid license IDS
by Andra Danciu 02 May '19

02 May '19

This series of patches solves two issues reported by Thomas Gleixner Andra Danciu (2): ASoC: mpc5200_dma: Fix invalid license ID ASoC: mpc5200_psc_i2s: Fix invalid license ID sound/soc/fsl/mpc5200_dma.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 2.11.0

5 11

[alsa-devel] Applied "ASoC: rsnd: Document r8a77470 bindings" to the asoc tree
by Mark Brown 02 May '19

02 May '19

The patch ASoC: rsnd: Document r8a77470 bindings 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 21c7f3e31ec582bf7f0fa4068333acd22b8cb445 Mon Sep 17 00:00:00 2001 From: Cao Van Dong <cv-dong(a)jinso.co.jp> Date: Thu, 11 Apr 2019 16:54:18 +0900 Subject: [PATCH] ASoC: rsnd: Document r8a77470 bindings Document SoC specific bindings for R-Car RZ/G1C(r8a77470) SoC. Signed-off-by: Cao Van Dong <cv-dong(a)jinso.co.jp> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index b34afa66fb00..5c52182f7dcf 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -266,6 +266,7 @@ Required properties: - "renesas,rcar_sound-r8a7743" (RZ/G1M) - "renesas,rcar_sound-r8a7744" (RZ/G1N) - "renesas,rcar_sound-r8a7745" (RZ/G1E) + - "renesas,rcar_sound-r8a77470" (RZ/G1C) - "renesas,rcar_sound-r8a774a1" (RZ/G2M) - "renesas,rcar_sound-r8a774c0" (RZ/G2E) - "renesas,rcar_sound-r8a7778" (R-Car M1A) -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: Intel: Make boards more available for compile test
by Mark Brown 02 May '19

02 May '19

The Intel boards have very strict dependencies which make them less available for compile test than is desirable, with requirements for specific drivers that are only needed at runtime but not at build time. Relax this a bit if COMPILE_TEST is enabled to improve build coverage for these drivers. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/Kconfig | 79 +++++++++++++++++++++++----------- 1 file changed, 54 insertions(+), 25 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 12d6b73e9531..07b1a02934f1 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -16,7 +16,9 @@ if SND_SOC_INTEL_HASWELL config SND_SOC_INTEL_HASWELL_MACH tristate "Haswell Lynxpoint" - depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM + depends on I2C + depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT5640 help This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell @@ -26,7 +28,10 @@ config SND_SOC_INTEL_HASWELL_MACH config SND_SOC_INTEL_BDW_RT5677_MACH tristate "Broadwell with RT5677 codec" - depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM && GPIOLIB + depends on I2C + depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST + depends on GPIOLIB || COMPILE_TEST + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT5677 help This adds support for Intel Broadwell platform based boards with @@ -36,7 +41,9 @@ config SND_SOC_INTEL_BDW_RT5677_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "Broadwell Wildcatpoint" - depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM + depends on I2C + depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT286 help This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell @@ -49,7 +56,8 @@ if SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "Baytrail with MAX98090 codec" - depends on X86_INTEL_LPSS && I2C + depends on I2C + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_MAX98090 help This adds audio driver for Intel Baytrail platform based boards @@ -59,7 +67,8 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "Baytrail with RT5640 codec" - depends on X86_INTEL_LPSS && I2C + depends on I2C + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT5640 help This adds audio driver for Intel Baytrail platform based boards @@ -72,7 +81,8 @@ if SND_SST_ATOM_HIFI2_PLATFORM config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "Baytrail and Baytrail-CR with RT5640 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5640 help @@ -83,7 +93,8 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_BYTCR_RT5651_MACH tristate "Baytrail and Baytrail-CR with RT5651 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5651 help @@ -94,7 +105,8 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "Cherrytrail & Braswell with RT5672 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5670 help @@ -105,7 +117,8 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "Cherrytrail & Braswell with RT5645/5650 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5645 help @@ -116,7 +129,8 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH tristate "Cherrytrail & Braswell with MAX98090 & TI codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_MAX98090 select SND_SOC_TS3A227E help @@ -127,7 +141,8 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH config SND_SOC_INTEL_CHT_BSW_NAU8824_MACH tristate "Cherrytrail & Braswell with NAU88L24 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_NAU8824 help @@ -138,7 +153,8 @@ config SND_SOC_INTEL_CHT_BSW_NAU8824_MACH config SND_SOC_INTEL_BYT_CHT_DA7213_MACH tristate "Baytrail & Cherrytrail with DA7212/7213 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_DA7213 help @@ -149,7 +165,8 @@ config SND_SOC_INTEL_BYT_CHT_DA7213_MACH config SND_SOC_INTEL_BYT_CHT_ES8316_MACH tristate "Baytrail & Cherrytrail with ES8316 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_ES8316 help @@ -160,7 +177,8 @@ config SND_SOC_INTEL_BYT_CHT_ES8316_MACH config SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH tristate "Baytrail & Cherrytrail platform with no codec (MinnowBoard MAX, Up)" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST help This adds support for ASoC machine driver for the MinnowBoard Max or Up boards and provides access to I2S signals on the Low-Speed @@ -176,7 +194,8 @@ if SND_SOC_INTEL_SKL config SND_SOC_INTEL_SKL_RT286_MACH tristate "SKL with RT286 I2S mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT286 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI @@ -188,7 +207,8 @@ config SND_SOC_INTEL_SKL_RT286_MACH config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH tristate "SKL with NAU88L25 and SSM4567 in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_NAU8825 select SND_SOC_SSM4567 select SND_SOC_DMIC @@ -201,7 +221,8 @@ config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH tristate "SKL with NAU88L25 and MAX98357A in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_NAU8825 select SND_SOC_MAX98357A select SND_SOC_DMIC @@ -218,7 +239,8 @@ if SND_SOC_INTEL_APL config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH tristate "Broxton with DA7219 and MAX98357A in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC @@ -232,7 +254,8 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH config SND_SOC_INTEL_BXT_RT298_MACH tristate "Broxton with RT298 I2S mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT298 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI @@ -249,7 +272,8 @@ if SND_SOC_INTEL_KBL config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH tristate "KBL with RT5663 and MAX98927 in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT5663 select SND_SOC_MAX98927 select SND_SOC_DMIC @@ -263,7 +287,8 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SPI select SND_SOC_RT5663 select SND_SOC_RT5514 @@ -278,7 +303,8 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH tristate "KBL with DA7219 and MAX98357A in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC @@ -290,7 +316,8 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH tristate "KBL with DA7219 and MAX98927 in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_DA7219 select SND_SOC_MAX98927 select SND_SOC_MAX98373 @@ -304,7 +331,8 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH config SND_SOC_INTEL_KBL_RT5660_MACH tristate "KBL with RT5660 in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT5660 select SND_SOC_HDAC_HDMI help @@ -318,7 +346,8 @@ if SND_SOC_INTEL_GLK config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT5682 select SND_SOC_MAX98357A select SND_SOC_DMIC -- 2.20.1

2 4

[alsa-devel] [PATCH v2] ASoC: Intel: common: add Geminilake Realtek+Maxim machine driver entry
by Yong Zhi 02 May '19

02 May '19

This patch adds glk_rt5682_max98357a_i2s machine driver entry into machine table. Both Skylake and SOF platform drivers can use this machine drivers. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Hi, Pierre, This is an update to https://patchwork.kernel.org/patch/10472523/ with SOF files prefix and asoc_plat_name removed, tested with SOF v1.1, thanks!! sound/soc/intel/common/soc-acpi-intel-glk-match.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index 3f2061475ae4..616eb09e78a0 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -31,6 +31,15 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { .sof_fw_filename = "sof-glk.ri", .sof_tplg_filename = "sof-glk-da7219.tplg", }, + { + .id = "10EC5682", + .drv_name = "glk_rt5682_max98357a", + .fw_filename = "intel/dsp_fw_glk.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &glk_codecs, + .sof_fw_filename = "sof-glk.ri", + .sof_tplg_filename = "sof-glk-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_glk_machines); -- 2.7.4

3 3

[alsa-devel] [PATCH] sound: rt5645: fix a NULL pointer dereference
by Kangjie Lu 02 May '19

02 May '19

devm_kcalloc() may fail and return NULL. The fix returns ENOMEM in case it fails to avoid NULL pointer dereference. Signed-off-by: Kangjie Lu <kjlu(a)umn.edu> --- sound/soc/codecs/rt5645.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9a0751978090..f842775dbf2c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3419,6 +3419,9 @@ static int rt5645_probe(struct snd_soc_component *component) RT5645_HWEQ_NUM, sizeof(struct rt5645_eq_param_s), GFP_KERNEL); + if (!rt5645->eq_param) + return -ENOMEM; + return 0; } -- 2.17.1

3 3

[alsa-devel] [PATCH] sound: cs43130: fix a NULL pointer dereference
by Kangjie Lu 02 May '19

02 May '19

In case create_singlethread_workqueue fails, the fix returns -ENOMEM to avoid potential NULL pointer dereference. Signed-off-by: Kangjie Lu <kjlu(a)umn.edu> --- sound/soc/codecs/cs43130.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 3f7b255587e6..80d672710eae 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -2322,6 +2322,8 @@ static int cs43130_probe(struct snd_soc_component *component) return ret; cs43130->wq = create_singlethread_workqueue("cs43130_hp"); + if (!cs43130->wq) + return -ENOMEM; INIT_WORK(&cs43130->work, cs43130_imp_meas); } -- 2.17.1

2 3

[alsa-devel] Applied "ASoC: imx-audmix: fix object reference leaks in probe" to the asoc tree
by Mark Brown 02 May '19

02 May '19

The patch ASoC: imx-audmix: fix object reference leaks in probe 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 8bb678d75954e9e5b9ecabfe95b20f650fd91edb Mon Sep 17 00:00:00 2001 From: Viorel Suman <viorel.suman(a)nxp.com> Date: Wed, 10 Apr 2019 11:06:38 +0000 Subject: [PATCH] ASoC: imx-audmix: fix object reference leaks in probe Release the reference to the underlying device taken by of_find_device_by_node() call. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Reported-by: Julia Lawall <Julia.Lawall(a)lip6.fr> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/imx-audmix.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 7983bd339c01..9aaf3e5b45b9 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -171,6 +171,7 @@ static int imx_audmix_probe(struct platform_device *pdev) np->full_name); return -EINVAL; } + put_device(&audmix_pdev->dev); num_dai = of_count_phandle_with_args(audmix_np, "dais", NULL); if (num_dai != FSL_AUDMIX_MAX_DAIS) { @@ -216,6 +217,7 @@ static int imx_audmix_probe(struct platform_device *pdev) dev_err(&pdev->dev, "failed to find SAI platform device\n"); return -EINVAL; } + put_device(&cpu_pdev->dev); dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s%s", fe_name_pref, args.np->full_name + 1); @@ -280,6 +282,8 @@ static int imx_audmix_probe(struct platform_device *pdev) dev_err(&pdev->dev, "failed to find SAI platform device\n"); return -EINVAL; } + put_device(&cpu_pdev->dev); + priv->cpu_mclk = devm_clk_get(&cpu_pdev->dev, "mclk1"); if (IS_ERR(priv->cpu_mclk)) { ret = PTR_ERR(priv->cpu_mclk); -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: fsl_audmix: remove "model" attribute from DT document" to the asoc tree
by Mark Brown 02 May '19

02 May '19

The patch ASoC: fsl_audmix: remove "model" attribute from DT document 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 d04195b0a351b6c7bc086bd8590e5f303122a0be Mon Sep 17 00:00:00 2001 From: Viorel Suman <viorel.suman(a)nxp.com> Date: Wed, 10 Apr 2019 11:06:37 +0000 Subject: [PATCH] ASoC: fsl_audmix: remove "model" attribute from DT document Remove "model" attribute from fsl_audmix DT document. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/fsl,audmix.txt | 4 ---- 1 file changed, 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.txt b/Documentation/devicetree/bindings/sound/fsl,audmix.txt index 45f807ec21a5..840b7e0d6a63 100644 --- a/Documentation/devicetree/bindings/sound/fsl,audmix.txt +++ b/Documentation/devicetree/bindings/sound/fsl,audmix.txt @@ -38,9 +38,6 @@ Device driver required properties: to SAI interfaces to be provided, the first SAI in the list being used to route the AUDMIX output. - - model : Must contain machine driver name which will configure - and instantiate the appropriate audio card. - Device driver configuration example: ====================================== audmix: audmix@59840000 { @@ -50,5 +47,4 @@ Device driver configuration example: clock-names = "ipg"; power-domains = <&pd_audmix>; dais = <&sai4>, <&sai5>; - model = "imx-audmix"; }; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: fsl_audmix: remove "model" attribute" to the asoc tree
by Mark Brown 02 May '19

02 May '19

The patch ASoC: fsl_audmix: remove "model" attribute 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 f2a36a78423ee83b1aac79956891fcad0e833885 Mon Sep 17 00:00:00 2001 From: Viorel Suman <viorel.suman(a)nxp.com> Date: Wed, 10 Apr 2019 11:06:36 +0000 Subject: [PATCH] ASoC: fsl_audmix: remove "model" attribute Use "of_device_id.data" to specify the machine driver instead of "model" DTS attribute. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_audmix.c | 43 ++++++++++++++++++++------------------ 1 file changed, 23 insertions(+), 20 deletions(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index dabde0342c95..dc802d5c4ccd 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -445,13 +445,29 @@ static const struct regmap_config fsl_audmix_regmap_config = { .cache_type = REGCACHE_FLAT, }; +static const struct of_device_id fsl_audmix_ids[] = { + { + .compatible = "fsl,imx8qm-audmix", + .data = "imx-audmix", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, fsl_audmix_ids); + static int fsl_audmix_probe(struct platform_device *pdev) { struct fsl_audmix *priv; struct resource *res; + const char *mdrv; + const struct of_device_id *of_id; void __iomem *regs; int ret; - const char *sprop; + + of_id = of_match_device(fsl_audmix_ids, &pdev->dev); + if (!of_id || !of_id->data) + return -EINVAL; + + mdrv = of_id->data; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -487,19 +503,12 @@ static int fsl_audmix_probe(struct platform_device *pdev) return ret; } - sprop = of_get_property(pdev->dev.of_node, "model", NULL); - if (sprop) { - priv->pdev = platform_device_register_data(&pdev->dev, sprop, 0, - NULL, 0); - if (IS_ERR(priv->pdev)) { - ret = PTR_ERR(priv->pdev); - dev_err(&pdev->dev, - "failed to register platform %s: %d\n", sprop, - ret); - } - } else { - dev_err(&pdev->dev, "[model] attribute missing.\n"); - ret = -EINVAL; + priv->pdev = platform_device_register_data(&pdev->dev, mdrv, 0, NULL, + 0); + if (IS_ERR(priv->pdev)) { + ret = PTR_ERR(priv->pdev); + dev_err(&pdev->dev, "failed to register platform %s: %d\n", + mdrv, ret); } return ret; @@ -553,12 +562,6 @@ static const struct dev_pm_ops fsl_audmix_pm = { pm_runtime_force_resume) }; -static const struct of_device_id fsl_audmix_ids[] = { - { .compatible = "fsl,imx8qm-audmix", }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(of, fsl_audmix_ids); - static struct platform_driver fsl_audmix_driver = { .probe = fsl_audmix_probe, .remove = fsl_audmix_remove, -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: Mediatek: MT8183: change supported formats of DL2 and UL1
by Tzung-Bi Shih 02 May '19

02 May '19

DL2 and UL1 are for BTSCO. They should only provide 16-bit, mono, 8kHz and 16kHz to userspace. Change the formats accordingly. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 42 +++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 681bedb524f9..4e44e5689d6f 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -69,6 +69,46 @@ mt8183_mt6358_ts3a227_max98357_dapm_routes[] = { {"IT6505_8CH", NULL, "TDM"}, }; +static int +mt8183_mt6358_ts3a227_max98357_bt_sco_startup( + struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 8000, 16000 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + static const unsigned int channels[] = { + 1, + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + + return 0; +} + +static const struct snd_soc_ops mt8183_mt6358_ts3a227_max98357_bt_sco_ops = { + .startup = mt8183_mt6358_ts3a227_max98357_bt_sco_startup, +}; + static struct snd_soc_dai_link mt8183_mt6358_ts3a227_max98357_dai_links[] = { /* FE */ @@ -93,6 +133,7 @@ mt8183_mt6358_ts3a227_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8183_mt6358_ts3a227_max98357_bt_sco_ops, }, { .name = "Playback_3", @@ -115,6 +156,7 @@ mt8183_mt6358_ts3a227_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8183_mt6358_ts3a227_max98357_bt_sco_ops, }, { .name = "Capture_2", -- 2.21.0.392.gf8f6787159e-goog

2 2

[alsa-devel] [PATCH v2 0/1] ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3
by Shunli Wang 02 May '19

02 May '19

ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3 Shunli Wang (1): ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3 sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 38 ++++++++++++++++++++++ 1 file changed, 38 insertions(+) -- 2.18.0

2 4

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Fix potential uninitialized variable
by Annaliese McDermond 02 May '19

02 May '19

Fix compiler warning about uninitialized variable reported by Stephen Rothwell <sfr(a)canb.auug.org.au>. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4-clk.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 9e4899eb1d8e..156c153c12ab 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -82,7 +82,7 @@ static int clk_aic32x4_pll_get_muldiv(struct clk_aic32x4 *pll, int ret; ret = regmap_read(pll->regmap, AIC32X4_PLLPR, &val); - if (ret) + if (ret < 0) return ret; settings->r = val & AIC32X4_PLL_R_MASK; settings->p = (val & AIC32X4_PLL_P_MASK) >> AIC32X4_PLL_P_SHIFT; -- 2.19.1

2 2

[alsa-devel] [PATCH v4 0/4] ASoC: fsl: audmix: remove "model" attribute and fix ref leaks
by Viorel Suman 02 May '19

02 May '19

The latest audmix patch-set (v5) had the "model" attribute removed as requested by Nicolin Chen, but looks like (v4) version of DAI driver reached "for-next" branch - fix this by removing "model" attribute. Asside of this fix object reference leaks in machine probe reported by Julia Lawall. Viorel Suman (4): ASoC: fsl_audmix: remove "model" attribute ASoC: fsl_audmix: remove "model" attribute from DT document ASoC: imx-audmix: fix object reference leaks in probe ASoC: fsl_audmix: cache pdev->dev pointer Changes since V1: a) Removed "model" attribute from dt-bindings documentation b) Adressed Daniel's comments Changes since V2: a) Cache pdev->dev pointer in fsl_audmix probe as suggested by Nicolin Changes since V3: a) Use subject lines matching the style for the subsystem. .../devicetree/bindings/sound/fsl,audmix.txt | 4 -- sound/soc/fsl/fsl_audmix.c | 60 +++++++++++----------- sound/soc/fsl/imx-audmix.c | 4 ++ 3 files changed, 35 insertions(+), 33 deletions(-) -- 2.7.4

4 10

[alsa-devel] [PATCH] ASoC: dapm: fix kcontrols for effect widgets
by Pierre-Louis Bossart 02 May '19

02 May '19

From: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> This patch adds the handling of snd_soc_dapm_effect that was missing. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 65ee0bb5dd0b..81a7a12196ff 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -883,6 +883,7 @@ static int dapm_create_or_share_kcontrol(struct snd_soc_dapm_widget *w, case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_pga: + case snd_soc_dapm_effect: case snd_soc_dapm_out_drv: wname_in_long_name = true; kcname_in_long_name = true; @@ -2370,6 +2371,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, case snd_soc_dapm_dac: case snd_soc_dapm_adc: case snd_soc_dapm_pga: + case snd_soc_dapm_effect: case snd_soc_dapm_out_drv: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: @@ -3197,6 +3199,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) dapm_new_mux(w); break; case snd_soc_dapm_pga: + case snd_soc_dapm_effect: case snd_soc_dapm_out_drv: dapm_new_pga(w); break; -- 2.17.1

2 1

[alsa-devel] [PATCH v2] ASoC: da7213: fix DAI_CLK_EN register bit overwrite
by Logesh Kolandavel 02 May '19

02 May '19

From: Logesh <logesh.kolandavel(a)timesys.com> If the da7213 codec is configured as Master with the DAPM power down delay time set, 'snd_soc_component_write' function overwrites the DAI_CLK_EN bit of DAI_CLK_MODE register which leads to audio play only once until it re-initialize after codec power up. Signed-off-by: Logesh <logesh.kolandavel(a)timesys.com> --- Changes in v2: -Include the mask DA7213_DAI_BCLKS_PER_WCLK_MASK sound/soc/codecs/da7213.c | 5 ++++- sound/soc/codecs/da7213.h | 2 ++ 2 files changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 92d006a5283e..425c11d63e49 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1305,7 +1305,10 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* By default only 64 BCLK per WCLK is supported */ dai_clk_mode |= DA7213_DAI_BCLKS_PER_WCLK_64; - snd_soc_component_write(component, DA7213_DAI_CLK_MODE, dai_clk_mode); + snd_soc_component_update_bits(component, DA7213_DAI_CLK_MODE, + DA7213_DAI_BCLKS_PER_WCLK_MASK | + DA7213_DAI_CLK_POL_MASK | DA7213_DAI_WCLK_POL_MASK, + dai_clk_mode); snd_soc_component_update_bits(component, DA7213_DAI_CTRL, DA7213_DAI_FORMAT_MASK, dai_ctrl); snd_soc_component_write(component, DA7213_DAI_OFFSET, dai_offset); diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 5a78dba1dcb5..9d31efc3cfe5 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -181,7 +181,9 @@ #define DA7213_DAI_BCLKS_PER_WCLK_256 (0x3 << 0) #define DA7213_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) #define DA7213_DAI_CLK_POL_INV (0x1 << 2) +#define DA7213_DAI_CLK_POL_MASK (0x1 << 2) #define DA7213_DAI_WCLK_POL_INV (0x1 << 3) +#define DA7213_DAI_WCLK_POL_MASK (0x1 << 3) #define DA7213_DAI_CLK_EN_MASK (0x1 << 7) /* DA7213_DAI_CTRL = 0x29 */ -- 2.21.0

3 2

[alsa-devel] [PATCH -next] ASoC: sprd: Fix return value check in sprd_mcdt_probe()
by Wei Yongjun 02 May '19

02 May '19

In case of error, the function devm_ioremap_resource() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Fixes: d7bff893e04f ("ASoC: sprd: Add Spreadtrum multi-channel data transfer support") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sprd/sprd-mcdt.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index 28f5e649733d..e9318d7a4810 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -951,8 +951,8 @@ static int sprd_mcdt_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); mcdt->base = devm_ioremap_resource(&pdev->dev, res); - if (!mcdt->base) - return -ENOMEM; + if (IS_ERR(mcdt->base)) + return PTR_ERR(mcdt->base); mcdt->dev = &pdev->dev; spin_lock_init(&mcdt->lock);

3 2

[alsa-devel] [PATCH -next] ASoC: sprd: Fix to use list_for_each_entry_safe() when delete items
by Wei Yongjun 02 May '19

02 May '19

Since we will remove items off the list using list_del() we need to use a safe version of the list_for_each_entry() macro aptly named list_for_each_entry_safe(). Fixes: d7bff893e04f ("ASoC: sprd: Add Spreadtrum multi-channel data transfer support") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sprd/sprd-mcdt.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index 28f5e649733d..df250f7f2b6f 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -978,12 +978,12 @@ static int sprd_mcdt_probe(struct platform_device *pdev) static int sprd_mcdt_remove(struct platform_device *pdev) { - struct sprd_mcdt_chan *temp; + struct sprd_mcdt_chan *chan, *temp; mutex_lock(&sprd_mcdt_list_mutex); - list_for_each_entry(temp, &sprd_mcdt_chan_list, list) - list_del(&temp->list); + list_for_each_entry_safe(chan, temp, &sprd_mcdt_chan_list, list) + list_del(&chan->list); mutex_unlock(&sprd_mcdt_list_mutex);

3 2

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Correct region base typo in wm_halo_setup_algs
by Charles Keepax 02 May '19

02 May '19

Due to a typo the wrong base is being supplied for the primary algorithm on Halo firmwares, which will cause the controls to not function. Fixes: 170b1e123f38 ("ASoC: wm_adsp: Add support for new Halo core DSPs") Reported-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index a9298bfddd9c7..f3c28e54d6d6e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2445,7 +2445,7 @@ static int wm_halo_setup_algs(struct wm_adsp *dsp) wmfw_v3_parse_id_header(dsp, &halo_id.fw, n_algs); ret = wm_halo_create_regions(dsp, halo_id.fw.id, - halo_id.ym_base, halo_id.ym_base); + halo_id.xm_base, halo_id.ym_base); if (ret) return ret; -- 2.11.0

2 3

[alsa-devel] [PATCH][next] ASoC: SOF: Intel: fix spelling mistake "incompatble" -> "incompatible"
by Colin King 02 May '19

02 May '19

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a hda_dsp_rom_msg message, fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sof/intel/hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index b8fc19790f3b..84baf275b467 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -54,7 +54,7 @@ static const struct hda_dsp_msg_code hda_dsp_rom_msg[] = { {HDA_DSP_ROM_L2_CACHE_ERROR, "error: L2 cache error"}, {HDA_DSP_ROM_LOAD_OFFSET_TO_SMALL, "error: load offset too small"}, {HDA_DSP_ROM_API_PTR_INVALID, "error: API ptr invalid"}, - {HDA_DSP_ROM_BASEFW_INCOMPAT, "error: base fw incompatble"}, + {HDA_DSP_ROM_BASEFW_INCOMPAT, "error: base fw incompatible"}, {HDA_DSP_ROM_UNHANDLED_INTERRUPT, "error: unhandled interrupt"}, {HDA_DSP_ROM_MEMORY_HOLE_ECC, "error: ECC memory hole"}, {HDA_DSP_ROM_KERNEL_EXCEPTION, "error: kernel exception"}, -- 2.20.1

4 3

[alsa-devel] [PATCH V6] ASoC: fsl_esai: Fix missing break in switch statement
by S.j. Wang 02 May '19

02 May '19

case ESAI_HCKT_EXTAL and case ESAI_HCKR_EXTAL should be independent of each other, so replace fall-through with break. Fixes: 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: <stable(a)vger.kernel.org> --- Changes in V6 - resend base one for-5.2 Changes in v5 - remove new line after Fixes Changes in v4 - Add acked-by Changes in v3 - Update subject line and cc stable Changes in v2 - Fix "Fixes" tag sound/soc/fsl/fsl_esai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c7410bbfd2af..bad0dfed6b68 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -251,7 +251,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, break; case ESAI_HCKT_EXTAL: ecr |= ESAI_ECR_ETI; - /* fall through */ + break; case ESAI_HCKR_EXTAL: ecr |= esai_priv->synchronous ? ESAI_ECR_ETI : ESAI_ECR_ERI; break; -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: tlv320aic3x: Add support for high power analog output
by Saravanan Sekar 02 May '19

02 May '19

Add support to output level control for the analog high power output drivers HPOUT and HPCOM. Signed-off-by: Saravanan Sekar <sravanhome(a)gmail.com> --- Notes: Changes in V2: - Removed power control as it is handled by DAPM - Added level control for left channel sound/soc/codecs/tlv320aic3x.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 516d17cb2182..90f53f9b5c2f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -324,6 +324,9 @@ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 50, 0); */ static DECLARE_TLV_DB_SCALE(output_stage_tlv, -5900, 50, 1); +/* HP/HPCOM volumes. From 0 to 9 dB in 1 dB steps */ +static DECLARE_TLV_DB_SCALE(hp_tlv, 0, 100, 0); + static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Output */ SOC_DOUBLE_R_TLV("PCM Playback Volume", @@ -419,6 +422,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Pop reduction */ SOC_ENUM("Output Driver Power-On time", aic3x_poweron_time_enum), SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum), + + /* Analog HPOUT, HPCOM output level controls */ + SOC_DOUBLE_R_TLV("HP Playback Volume", HPLOUT_CTRL, HPROUT_CTRL, + 4, 9, 0, hp_tlv); + SOC_DOUBLE_R_TLV("HPCOM Playback Volume", HPLCOM_CTRL, HPRCOM_CTRL, + 4, 9, 0, hp_tlv); }; /* For other than tlv320aic3104 */ -- 2.17.1

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[alsa-devel] Correct git tree for ALSA development
by Ross Zwisler 02 May '19

02 May '19

Hi, I'm trying to ramp up a bit on ALSA development, and I noticed that many sound-related entries in MAINTAINERS still refer to alsa-kernel.git hosted on alsa-project.org, i.e.: BT87X AUDIO DRIVER M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) T: git git://git.alsa-project.org/alsa-kernel.git S: Maintained F: Documentation/sound/cards/bt87x.rst F: sound/pci/bt87x.c >From the alsa-project.org pages: https://alsa-project.org/wiki/Download#Developers:_GIT_access https://alsa-project.org/wiki/GIT_Server#For_developers_-_kernel_drivers it seems like current development/maintenance actually happens somewhere else, possibly on Mark Brown's kernel.org tree? https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/ If so, should we update those stale MAINTAINERS entries? Thanks, - Ross

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[alsa-devel] [PATCH] ASoC: da7219: Use clk_round_rate to handle enabled bclk/wclk case
by Adam Thomson 02 May '19

02 May '19

For some platforms where DA7219 is the DAI clock master, BCLK/WCLK will be set and enabled prior to the codec's hw_params() function being called. It is possible the platform requires a different BCLK/WCLK configuration than would be chosen by hw_params(), for example S16_LE format needed with a 64-bit frame to satisfy certain devices using the clocks. To handle those kinds of scenarios, the use of clk_round_rate() is now employed as part of hw_params(). If BCLK/WCLk are already enabled then this function will just return the currently set rates, so the subsequent calls to clk_set_rate() will succeed and nothing changes with regards to clocking. In addition the specific recalc_rate() implementations needed updating to always give back a real value, as those functions are called as part of the clk init code and a real value is needed for the clk_round_rate() calls to work as expected. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7219.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 5f5fa34..72268f5 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1593,6 +1593,13 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, sr = params_rate(params); if (da7219->master && wclk) { + /* + * Rounding the rate here avoids failure trying to set a new + * rate on an already enabled wclk. In that instance this will + * just set the same rate as is currently in use, and so should + * continue without problem. + */ + sr = clk_round_rate(wclk, sr); ret = clk_set_rate(wclk, sr); if (ret) { dev_err(component->dev, @@ -1621,6 +1628,14 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, if (bclk) { bclk_rate = frame_size * sr; + /* + * Rounding the rate here avoids failure trying to set a + * new rate on an already enabled bclk. In that + * instance this will just set the same rate as is + * currently in use, and so should continue without + * problem. + */ + bclk_rate = clk_round_rate(bclk, bclk_rate); ret = clk_set_rate(bclk, bclk_rate); if (ret) { dev_err(component->dev, @@ -1927,9 +1942,6 @@ static unsigned long da7219_wclk_recalc_rate(struct clk_hw *hw, struct snd_soc_component *component = da7219->component; u8 fs = snd_soc_component_read32(component, DA7219_SR); - if (!da7219->master) - return 0; - switch (fs & DA7219_SR_MASK) { case DA7219_SR_8000: return 8000; @@ -2016,9 +2028,6 @@ static unsigned long da7219_bclk_recalc_rate(struct clk_hw *hw, u8 bclks_per_wclk = snd_soc_component_read32(component, DA7219_DAI_CLK_MODE); - if (!da7219->master) - return 0; - switch (bclks_per_wclk & DA7219_DAI_BCLKS_PER_WCLK_MASK) { case DA7219_DAI_BCLKS_PER_WCLK_32: return parent_rate * 32; -- 1.9.1

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[alsa-devel] [PATCH] ACPI / device_sysfs: change _ADR representation to 64 bits
by Pierre-Louis Bossart 01 May '19

01 May '19

Standards such as the MIPI DisCo for SoundWire 1.0 specification assume the _ADR field is 64 bits. _ADR is defined as an "Integer" represented as 64 bits since ACPI 2.0 released in 2002. The low levels already use _ADR as 64 bits, e.g. in struct acpi_device_info. This patch bumps the representation used for sysfs to 64 bits. Example with a SoundWire device, the results show the complete vendorID and linkID which were omitted before: Before: $ more /sys/bus/acpi/devices/device\:38/adr 0x5d070000 After: $ more /sys/bus/acpi/devices/device\:38/adr 0x000010025d070000 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/acpi/device_sysfs.c | 3 +-- include/acpi/acpi_bus.h | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/drivers/acpi/device_sysfs.c b/drivers/acpi/device_sysfs.c index 8940054d6250..f8d73ae42529 100644 --- a/drivers/acpi/device_sysfs.c +++ b/drivers/acpi/device_sysfs.c @@ -428,8 +428,7 @@ static ssize_t acpi_device_adr_show(struct device *dev, { struct acpi_device *acpi_dev = to_acpi_device(dev); - return sprintf(buf, "0x%08x\n", - (unsigned int)(acpi_dev->pnp.bus_address)); + return sprintf(buf, "0x%016llx\n", acpi_dev->pnp.bus_address); } static DEVICE_ATTR(adr, 0444, acpi_device_adr_show, NULL); diff --git a/include/acpi/acpi_bus.h b/include/acpi/acpi_bus.h index f7981751ac77..9075e28ea60a 100644 --- a/include/acpi/acpi_bus.h +++ b/include/acpi/acpi_bus.h @@ -230,7 +230,7 @@ struct acpi_device_dir { /* Plug and Play */ typedef char acpi_bus_id[8]; -typedef unsigned long acpi_bus_address; +typedef u64 acpi_bus_address; typedef char acpi_device_name[40]; typedef char acpi_device_class[20]; -- 2.17.1

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[alsa-devel] [PATCH 00/19] ASoC: SOF: improvements for ABI checks and Intel code
by Pierre-Louis Bossart 01 May '19

01 May '19

This series is a set of relatively small SOF changes after the big batch merged last week (Thanks!). Since we are very close to the merge window and in May where most of the world takes time off, it'd be perfectly understandable if those patches were queued for 5.3, after feedback and corrections as needed. First we added optional stricter ABI checks when the firmware, topology and kernel ABI levels differ, which can happen when patches are merged at different times on Github. This is mainly for developers and the Github CI to track disconnects, should they happen despite our new process to keep evolutions under control. This option has no impact on the usual problem of updating the kernel without updating the firmware files. The Intel changes are mainly a) fixes for the slave mode suppport b) clean-ups such as removal of a static index for HDAudio support or removal of unneeded include file c) use of a workqueue to defer period elapsed events after the IPC completion. d) simplification of the IRQ code e) fixes to deal with resume on HDaudio links. In the previous patchset we removed the RESUME_INFO flag but missed the need to explicitly set the hw_params on resume. f) a set of routines to dump the Intel IPC registers, mainly for early platform enablement. Keyon Jie (7): ASoC: SOF: Intel: cnl: add pointer ops to use DPIB position ASoC: SOF: PCM: add period_elapsed work to fix race condition in interrupt context ASoC: SOF: Intel: use snd_sof_pcm_period_elapsed ASoC: SOF: ipc: use snd_sof_pcm_period_elapsed ASoC: SOF: Intel: hda-ipc: simplify handling of IPC IRQ ASoC: SOF: Intel: hda-stream: store stream capabilities ASoC: SOF: Intel: hda-stream: handle real stream interrupts only Pan Xiuli (3): ASoC: SOF: IPC: add ipc dump function ASoC: SOF: Intel: APL: add ipc dump function ASoC: SOF: Intel: CNL: add ipc dump function Pierre-Louis Bossart (4): ASoC: SOF: add Kconfig option for strict ABI checks ASOC: SOF: ipc: add support for stricter ABI checks ASoC: SOF: topology: add support for stricter ABI checks ASoC: SOF: Intel: hda-pcm: remove useless dependency on hdac_ext Ranjani Sridharan (1): ASoC: SOF: intel: hda: add hw_params_upon_resume flag for hda stream Zhu Yingjiang (4): ASoC: SOF: Intel: hda: add the SSP Host Device memory space ASoC: SOF: Intel: hda: add SSP info to the chip info struct ASoC: SOF: Intel: hda: set I2S slave before enabling DSP ASoC: SOF: Intel: hda: set bus->idx as 0 sound/soc/sof/Kconfig | 15 ++++++++++ sound/soc/sof/intel/apl.c | 4 +++ sound/soc/sof/intel/cnl.c | 27 +++++++++++++++--- sound/soc/sof/intel/hda-bus.c | 9 ++++-- sound/soc/sof/intel/hda-dai.c | 23 +++++++++------ sound/soc/sof/intel/hda-dsp.c | 16 +++++++++++ sound/soc/sof/intel/hda-ipc.c | 13 ++++----- sound/soc/sof/intel/hda-loader.c | 11 ++++++++ sound/soc/sof/intel/hda-pcm.c | 1 - sound/soc/sof/intel/hda-stream.c | 15 ++++++++-- sound/soc/sof/intel/hda.c | 18 ++++++++++++ sound/soc/sof/intel/hda.h | 23 +++++++++++++++ sound/soc/sof/intel/shim.h | 2 ++ sound/soc/sof/ipc.c | 12 +++++++- sound/soc/sof/ops.h | 12 ++++++++ sound/soc/sof/pcm.c | 48 ++++++++++++++++++++++++++++++++ sound/soc/sof/pm.c | 3 ++ sound/soc/sof/sof-priv.h | 5 +++- sound/soc/sof/topology.c | 43 +++++++++++++++++++--------- 19 files changed, 257 insertions(+), 43 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH 0/3] ASoC: Intel: machine drivers for SOF
by Pierre-Louis Bossart 01 May '19

01 May '19

Add support for the two main machine drivers used by SOF (not tested with the Skylake driver due to lack of hardware and/or firmware signature/ missing topology) The DMIC support for HDaudio-based devices is very much needed for a number of WhiskyLake+ platforms designed for Windows. SOF will be the only Intel driver supporting this configuration. A firmware binary signed with the required production key should be available shortly. The rt5682 codec is the main validation vehicle of the Intel SOF team, with eval boards connected to the Baytrail MinnowBoard as well as more recent devices. The main difference with traditional Intel machine drivers is that the dai links are added dynamically at probe time to avoid multiple table copy/pastes. Bard liao (1): ASoC: Intel: add sof-rt5682 machine driver Rander Wang (2): ASoC: Intel: skl_hda_dsp_generic: add DMIC support ASoC: Intel: skl_hda_dsp_generic:refine code style sound/soc/intel/boards/Kconfig | 16 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 22 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 19 +- sound/soc/intel/boards/sof_rt5682.c | 563 ++++++++++++++++++ .../intel/common/soc-acpi-intel-byt-match.c | 6 + .../intel/common/soc-acpi-intel-cht-match.c | 6 + .../intel/common/soc-acpi-intel-cnl-match.c | 19 + .../intel/common/soc-acpi-intel-icl-match.c | 6 + 10 files changed, 656 insertions(+), 5 deletions(-) create mode 100644 sound/soc/intel/boards/sof_rt5682.c -- 2.17.1

1 3

[alsa-devel] [PATCH v3 0/5] soundwire: code cleanup
by Pierre-Louis Bossart 30 Apr '19

30 Apr '19

SoundWire support will be provided in Linux with the Sound Open Firmware (SOF) on Intel platforms. Before we start adding the missing pieces, there are a number of warnings and style issues reported by checkpatch, cppcheck and Coccinelle that need to be cleaned-up. Changes since v2: fixed inversion of devm_kcalloc parameters, detected while rebasing additional patches. Changes since v1: added missing newlines in new patch (suggested by Joe Perches) Pierre-Louis Bossart (5): soundwire: intel: fix inversion in devm_kcalloc parameters soundwire: fix style issues soundwire: bus: remove useless initializations soundwire: stream: remove useless initialization of local variable soundwire: add missing newlines in dynamic debug logs drivers/soundwire/Kconfig | 2 +- drivers/soundwire/bus.c | 137 ++++++++------- drivers/soundwire/bus.h | 16 +- drivers/soundwire/bus_type.c | 4 +- drivers/soundwire/cadence_master.c | 99 +++++------ drivers/soundwire/cadence_master.h | 22 +-- drivers/soundwire/intel.c | 103 ++++++----- drivers/soundwire/intel.h | 4 +- drivers/soundwire/intel_init.c | 12 +- drivers/soundwire/mipi_disco.c | 116 +++++++------ drivers/soundwire/slave.c | 10 +- drivers/soundwire/stream.c | 267 +++++++++++++++-------------- 12 files changed, 404 insertions(+), 388 deletions(-) -- 2.17.1

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[alsa-devel] snd-usb-audio Buffer Sizes and Round Trip Latency
by Jonathan Liu 30 Apr '19

30 Apr '19

Hi, I want to start a discussion regarding round trip latency for class compliant USB audio interfaces on Linux. In particular, I am noticing with my USB 2.0 RME Babyface Pro audio interface that the round trip latency is considerably higher on Linux than on macOS High Sierra and Windows 10. I tested the round trip latency using a loopback audio cable and the ReaInsert plugin included with Reaper DAW (www.reaper.fm) that can be downloaded for Windows/macOS/Linux to calculate the additional delay. Here are the results for 48000 Hz, 24-bit on my RME Babyface Pro: === block_size/periods block_size*periods + additional_delay ~ round_trip_latency round_trip_latency = (block_size*periods + additional_delay) / 48000 * 1000 Linux 4.17.14, Class Compliant Mode (snd-usb-audio, ALSA backend): 16/2 32 + 80 ~ 2.333 ms 16/3 48 + 109 ~ 3.271 ms 32/2 64 + 129 ~ 4.021 ms 32/3 96 + 166 ~ 5.458 ms 64/2 128 + 205 ~ 6.938 ms 64/3 192 + 242 ~ 9.042 ms 128/2 256 + 352 ~ 12.667 ms 128/3 384 + 496 ~ 18.334 ms 256/2 512 + 650 ~ 24.208 ms 256/3 768 + 650 ~ 29.542 ms 512/2 1024 + 634 ~ 34.542 ms 512/3 1536 + 634 ~ 45.208 ms 1024/2 2048 + 650 ~ 56.208 ms 1024/3 3072 + 650 ~ 77.542 ms 2048/2 4096 + 633 ~ 98.521 ms 2048/3 6144 + 633 ~ 141.188 ms macOS High Sierra, Class Compliant Mode (Apple Driver): 16/2 32 + 205 ~ 4.938 ms 32/2 64 + 205 ~ 5.604 ms 64/2 128 + 205 ~ 6.938 ms 128/2 256 + 205 ~ 9.604 ms 256/2 512 + 205 ~ 14.938 ms 512/2 1024 + 205 ~ 25.604 ms 1024/2 2048 + 205 ~ 46.938 ms 2048/2 4096 + 205 ~ 89.604 ms macOS High Sierra, PC Mode (RME Driver v3.08): 16/2 32 + 59 ~ 1.896 ms 32/2 64 + 59 ~ 2.563 ms 64/2 128 + 59 ~ 3.896 ms 128/2 256 + 59 ~ 6.563 ms 256/2 512 + 59 ~ 11.596 ms 512/2 1024 + 59 ~ 22.563 ms 1024/2 2048 + 59 ~ 43.896 ms 2048/2 4096 + 59 ~ 86.563 ms Windows 10, PC Mode (RME Driver 1.099): 48/2 96 + 63 ~ 3.313 ms 64/2 128 + 63 ~ 3.979 ms 96/2 192 + 63 ~ 5.313 ms 128/2 256 + 63 ~ 6.646 ms 256/2 512 + 63 ~ 11.979 ms 512/2 1024 + 63 ~ 22.646 ms 1024/2 2048 + 63 ~ 43.979 ms 2048/2 4096 + 63 ~ 86.646 ms === Some things in particular I noticed on Linux: - additional_delay varies a bit if I close and open the audio device again - additional_delay seems to increase as the block_size increases. I can make the additional_delay stay about the same rather than increasing by setting MAX_PACKS and MAX_PACKS_HS to 1 in sound/usb/card.h. In Linux versions before 3.13 there was a nrpacks parameter for snd-usb-audio to control this but it was removed with commit https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… - additional_delay is not constant as block_size is increased like on macOS and Windows I made a patch to snd-usb-audio to expose the snd-usb-audio constants as runtime adjustable module parameters (/sys/module/snd_usb_audio/parameters/) for testing (takes effect when the device is disconnected+reconnected and logs the parameter values to dmesg): https://aur.archlinux.org/cgit/aur.git/plain/parameters.patch?h=snd-usb-aud… The patch is used in my Arch Linux AUR package for convenience (using DKMS to avoid having to recompile entire kernel): https://aur.archlinux.org/packages/snd-usb-audio-lowlatency-dkms/ Can snd-usb-audio be improved so the additional_delay is always the same when closing/opening/reconfiguring the audio device and does not increase as the block_size increases? I noticed using USB audio on Linux at lower latencies (block_size <= 128) is more prone to audio dropouts under load compared to macOS and Windows, even with CPU power management disabled (writing 0 to /dev/cpu_dma_latency). What can be done about this? Thanks. Regards, Jonathan

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[alsa-devel] [PATCH] ALSA: hda - Register irq handler after the chip initialization
by Takashi Iwai 30 Apr '19

30 Apr '19

Currently the IRQ handler in HD-audio controller driver is registered before the chip initialization. That is, we have some window opened between the azx_acquire_irq() call and the CORB/RIRB setup. If an interrupt is triggered in this small window, the IRQ handler may access to the uninitialized RIRB buffer, which leads to a NULL dereference Oops. This is usually no big problem since most of Intel chips do register the IRQ via MSI, and we've already fixed the order of the IRQ enablement and the CORB/RIRB setup in the former commit b61749a89f82 ("sound: enable interrupt after dma buffer initialization"), hence the IRQ won't be triggered in that room. However, some platforms use a shared IRQ, and this may allow the IRQ trigger by another source. Another possibility is the kdump environment: a stale interrupt might be present in there, the IRQ handler can be falsely triggered as well. For covering this small race, let's move the azx_acquire_irq() call after hda_intel_init_chip() call. Although this is a bit radical change, it can cover more widely than checking the CORB/RIRB setup locally in the callee side. Reported-by: Liwei Song <liwei.song(a)windriver.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2ec91085fa3e..789308f54785 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1788,9 +1788,6 @@ static int azx_first_init(struct azx *chip) chip->msi = 0; } - if (azx_acquire_irq(chip, 0) < 0) - return -EBUSY; - pci_set_master(pci); synchronize_irq(bus->irq); @@ -1904,6 +1901,9 @@ static int azx_first_init(struct azx *chip) return -ENODEV; } + if (azx_acquire_irq(chip, 0) < 0) + return -EBUSY; + strcpy(card->driver, "HDA-Intel"); strlcpy(card->shortname, driver_short_names[chip->driver_type], sizeof(card->shortname)); -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: hda: check RIRB to avoid use NULL pointer
by Song liwei 30 Apr '19

30 Apr '19

From: Liwei Song <liwei.song(a)windriver.com> Fix the following BUG: BUG: unable to handle kernel NULL pointer dereference at 000000000000000c Workqueue: events azx_probe_work [snd_hda_intel] RIP: 0010:snd_hdac_bus_update_rirb+0x80/0x160 [snd_hda_core] Call Trace: <IRQ> azx_interrupt+0x78/0x140 [snd_hda_codec] __handle_irq_event_percpu+0x49/0x300 handle_irq_event_percpu+0x23/0x60 handle_irq_event+0x3c/0x60 handle_edge_irq+0xdb/0x180 handle_irq+0x23/0x30 do_IRQ+0x6a/0x140 common_interrupt+0xf/0xf The Call Trace happened when run kdump on a NFS rootfs system. Exist the following calling sequence when boot the second kernel: azx_first_init() --> azx_acquire_irq() <-- interrupt come in, azx_interrupt() was called --> hda_intel_init_chip() --> azx_init_chip() --> snd_hdac_bus_init_chip() --> snd_hdac_bus_init_cmd_io(); --> init rirb.buf and corb.buf Interrupt happened after azx_acquire_irq() while RIRB still didn't got initialized, then NULL pointer will be used when process the interrupt. Check the value of RIRB to ensure it is not NULL, to aviod some special case may hang the system. Fixes: 14752412721c ("ALSA: hda - Add the controller helper codes to hda-core module") Signed-off-by: Liwei Song <liwei.song(a)windriver.com> --- sound/hda/hdac_controller.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 74244d8e2909..2f0fa5353361 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -195,6 +195,9 @@ void snd_hdac_bus_update_rirb(struct hdac_bus *bus) return; bus->rirb.wp = wp; + if (!bus->rirb.buf) + return; + while (bus->rirb.rp != wp) { bus->rirb.rp++; bus->rirb.rp %= AZX_MAX_RIRB_ENTRIES; -- 2.7.4

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[alsa-devel] [GIT PULL] soundwire updates for v5.2-rc1
by Vinod Koul 30 Apr '19

30 Apr '19

Hey Greg, Please PULL to receive updates to soundwire subsystem The following changes since commit bfeffd155283772bbe78c6a05dec7c0128ee500c: Linux 5.0-rc1 (2019-01-06 17:08:20 -0800) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-5.2-rc1 for you to fetch changes up to 4abbd783d126cabfc20d1b8d50b1d5026b5cba09: soundwire: intel: fix implicit header use of module.h/export.h (2019-04-14 15:52:50 +0530) ---------------------------------------------------------------- soundwire updates for v5.2-rc1 This round saw interest from Intel, Linaro and Cadence in Soundwire. More patches are in pipeline (maybe next cycle) - removal of useless initialzation in core - couple of header and kcalloc inversion fixes on Intel driver ---------------------------------------------------------------- Paul Gortmaker (1): soundwire: intel: fix implicit header use of module.h/export.h Pierre-Louis Bossart (3): soundwire: intel: fix inversion in devm_kcalloc parameters soundwire: remove useless initializations soundwire: stream: remove useless initialization of local variable drivers/soundwire/bus.c | 2 -- drivers/soundwire/intel.c | 5 +++-- drivers/soundwire/intel_init.c | 2 ++ drivers/soundwire/stream.c | 2 +- 4 files changed, 6 insertions(+), 5 deletions(-) Thanks -- ~Vinod

3 4

[alsa-devel] move eapd coef function before ACT_PRE_PROBE state
by Kailang 30 Apr '19

30 Apr '19

Hi Takashi, alc_fill_eapd_coef(),this function was change EAPD control to default. Default was set EAPD by verb control. This function was run in ACT_INIT state. Move it to ACT_PRE_PROBE above. It will have a chance to change EAPD control on ACT_PRE_PROBE state. It could change control by 0x20 coef bit. BR, Kailang

2 1

Re: [alsa-devel] ALC892 acquisition stutter
by Takashi Iwai 30 Apr '19

30 Apr '19

On Tue, 30 Apr 2019 08:27:54 +0200, rodomar705(a)protonmail.com wrote: > > Nope, neither adding AZX_DCAPS_PM_RUNTIME as seen on other AMD codecs below seems to affect it, unfortunately. Please don't drop Cc to ML and others. And, did you try other position_fix option value, too? Otherwise I have out of idea. It must be platform-specific issue only the hardware vendors know. Takashi > > Sent with ProtonMail Secure Email. > > ‐‐‐‐‐‐‐ Original Message ‐‐‐‐‐‐‐ > Il lunedì 29 aprile 2019 21:28, Takashi Iwai <tiwai(a)suse.de> ha scritto: > > > On Mon, 29 Apr 2019 20:21:22 +0200, > > rodomar705(a)protonmail.com wrote: > > > > > Whoops, sorry. Here it is. > > > > OK, it's an AMD chip and not listed explicitly in the driver config. > > Does the patch below have any influence? > > > > Takashi > > > > --- a/sound/pci/hda/hda_intel.c > > +++ b/sound/pci/hda/hda_intel.c > > @@ -2452,6 +2452,9 @@ static const struct pci_device_id azx_ids[] = { > > /* AMD Hudson */ > > { PCI_DEVICE(0x1022, 0x780d), > > .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, > > > > - /* AMD family 17h */ > > - { PCI_DEVICE(0x1022, 0x1457), > > - .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, > > > > > > /* AMD Stoney */ > > { PCI_DEVICE(0x1022, 0x157a), > > .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | > > > >

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Re: [alsa-devel] ALC892 acquisition stutter
by Takashi Iwai 29 Apr '19

29 Apr '19

On Mon, 29 Apr 2019 20:21:22 +0200, rodomar705(a)protonmail.com wrote: > > Whoops, sorry. Here it is. OK, it's an AMD chip and not listed explicitly in the driver config. Does the patch below have any influence? Takashi --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2452,6 +2452,9 @@ static const struct pci_device_id azx_ids[] = { /* AMD Hudson */ { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + /* AMD family 17h */ + { PCI_DEVICE(0x1022, 0x1457), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, /* AMD Stoney */ { PCI_DEVICE(0x1022, 0x157a), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |

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[alsa-devel] [PATCH] ASoC: Intel: avoid Oops if DMA setup fails
by Ross Zwisler 29 Apr '19

29 Apr '19

Currently in sst_dsp_new() if we get an error return from sst_dma_new() we just print an error message and then still complete the function successfully. This means that we are trying to run without sst->dma properly set up, which will result in NULL pointer dereference when sst->dma is later used. This was happening for me in sst_dsp_dma_get_channel(): struct sst_dma *dma = dsp->dma; ... dma->ch = dma_request_channel(mask, dma_chan_filter, dsp); This resulted in: BUG: unable to handle kernel NULL pointer dereference at 0000000000000018 IP: sst_dsp_dma_get_channel+0x4f/0x125 [snd_soc_sst_firmware] Fix this by adding proper error handling for the case where we fail to set up DMA. Signed-off-by: Ross Zwisler <zwisler(a)google.com> Cc: stable(a)vger.kernel.org --- sound/soc/intel/common/sst-firmware.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 1e067504b6043..9be3a793a55e3 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -1251,11 +1251,15 @@ struct sst_dsp *sst_dsp_new(struct device *dev, goto irq_err; err = sst_dma_new(sst); - if (err) + if (err) { dev_warn(dev, "sst_dma_new failed %d\n", err); + goto dma_err; + } return sst; +dma_err: + free_irq(sst->irq, sst); irq_err: if (sst->ops->free) sst->ops->free(sst); -- 2.21.0.593.g511ec345e18-goog

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Re: [alsa-devel] ALC892 acquisition stutter
by Takashi Iwai 29 Apr '19

29 Apr '19

On Mon, 29 Apr 2019 18:25:18 +0200, rodomar705(a)protonmail.com wrote: > > Sample formats or sample rates wasn't yielding anything useful, > unfortunately; nor the -D on the hardware. However, after setting > --buffer-size 8192 and --period-size to 32768 the clicking are > finally completely gone in arecord. The question remains on how to > set this two parameters permanently to force ALSA to use them, if > it's possible at all. (and I hope that they will not influence too > much the delay of the acquisition). This is a too big period size for normal use case. > On pulseaudio, no luck in any way. tsched=0 make the acquisition 10 > times worse, regardless of the latency_msec passed to the loopback > module that was the only "half fix", since to have clean audio I > need a latency of 1,25s, that for doing real time communication is > not really the ideal solution. If the smaller period size in arecord causes a problem, it's rather other way round; tsched=0 is for small periods but works with the inaccurate position reporting. OTOH, tsched=1 is basically with a huge single period in the buffer. So, if it were only about the period size, PA tsched=1 should have worked better. It means that something else plays another role. > snoop=0 changed nothing, > unfortunately, nor passing model=HDMI, generic to the codec (the > first codec on my system is an AMD GPU card, the second is the > integrated codec). For these kernel module parameters, double-check that they are properly applied. The snoop=0 should be seen in the dmesg output where the driver will tell the snoop mode change. And the generic model would lead to a different set of mixer elements, so you can compare the difference in "amixer contents" output. The fact that the period size matters implies that it *might* be a HD-audio controller problem, not the codec. Could you give alsa-info.sh output? Run with --no-upload option, and attach the output. If it's about the controller side, position_fix option might have some influence, too. Also, with any options, try to disable the runtime PM. At easiest, do echo 0 > /sys/module/snd_hda_intel/parameters/power_save" should disable the runtime PM temporarily. Takashi > Thanks a lot for the detailed email, > > I hope that someone will figure out what's causing this, > > Marco. > > Sent with ProtonMail Secure Email. > > ‐‐‐‐‐‐‐ Original Message ‐‐‐‐‐‐‐ > Il lunedì 29 aprile 2019 17:06, Takashi Iwai <tiwai(a)suse.de> ha scritto: > > > On Sun, 28 Apr 2019 11:34:34 +0200, > > rodomar705(a)protonmail.com wrote: > > > > > Maybe, just maybe, I found a probable cause of what's happening here. Since it was working with ALSA directly only with audacity, I follow the instruction on tracing the HDA commands sent to the codec while acquiring audio with Audacity. One with ALSA directly (just with Audacity thought, arecord show the same problem from Pulse), when is working correctly, one without it, using pulse. The logs are very interesting. > > > The working one just simply sent two command to the audio codec, and then the stream started. > > > The other one spammed a lot of more data, and it appears that the problem (and the continuous popping of the audio) derives from the fact that the codec keep closing the stream after 100 to 300 ms, that force pulse to ask the audio controller to restart the stream and cause an actual noise coming from the microphone (since the stream is not instantly restarted). > > > There is another little difference beetween the second command sent to the codec directly with ALSA (from audacity) and the command sent from Pulse: > > > ALSA with Audacity: > > > 1106.051129: hda_send_cmd: [0000:09:00.3:0] val=0x00820031 > > > Pulse with Audacity: > > > 1231.103757: hda_send_cmd: [0000:09:00.3:0] val=0x00820011 > > > However with arecord -d 10 -fS16_LE -r44100 /tmp/test-mic.wav as a parameter I still get almost the same as Pulseaudo: > > > arecord with Audacity: > > > 2891.230874: hda_send_cmd: [0000:09:00.3:0] val=0x00824011 > > > However, no start stop behaviour like on pulseaudio, however the audio clicking are still there. > > > The log are attached to this email, hoping that will help fix the issue. > > > > Basically there can be three or four different parameters that may > > bring difference easily. The most suspected ones are the sample rates > > and the sample formats. So, check the combination of -fS16_LE, > > -fS32_LE, -r44110 and -r48000 with arecord. At best, pass -Dhw:0 > > option (or hw:1, depending on the machine) to assure that no > > conversion is done in alsa-lib, and pass -c2 option for stereo. > > > > arecord -Dhw:0 -c2 -fS32_LE -r44100 foo.wav > > > > The next would be to check the period size and buffer size. You can > > specify them via --period-size and --buffer-size options of arecord. > > They are in frames unit, not in bytes. Try to use a small and a large > > period / buffer size to see whether they give any difference. > > > > Also, for PA side, you can try the following: > > > > - Disable timer-scheduling: if the HD-audio controller is buggy and > > the pointer updates aren't accurate, it's a problem with the timer > > scheduling mode used as default in PA. You can pass tsched=0 option > > somehow, see the PA documentation. > > > > - Try to not load Realtek codec driver but use the generic codec > > driver. For that, pass model=generic option to snd-hda-intel > > module. (If there are two HD-audio controllers, pass > > model=generic,generic instead.) > > > > - Try to disable snooping in HD-audio driver, pass snoop=0 option to > > snd-hda-intel module. I don't think this would matter, but it's > > been often a problem on a new chip on AMD or other platforms. > > > > HTH, > > > > Takashi > > > > > > > Sent with ProtonMail Secure Email. > > > ‐‐‐‐‐‐‐ Original Message ‐‐‐‐‐‐‐ > > > Il domenica 28 aprile 2019 10:52, rodomar705(a)protonmail.com ha scritto: > > > > > > > Already sent to alsa-user, however a developer asked to send this also on devel, so here it is: > > > > I have audio stutters with an ALC892 Realtek Chipset on a B450 GAMING ELITE board from Gigabyte during acquisition (at least with microphone input, both front and back. Line in not tested). No issues with Windows. > > > > After opening an issue and testing with a PulseAudio developer, he has determined that the issue is not within PulseAudio, but on ALSA. > > > > There are already two issues about that on the bug tracker on Linux, namely this two: > > > > ALC892: https://bugzilla.kernel.org/show_bug.cgi?id=203351 > > > > ALC1220: https://bugzilla.kernel.org/show_bug.cgi?id=195303 > > > > To me both appears to be identical, for this reason even if I have a ALC892 I’ve posted on the second one, just because it had more posts. > > > > The particular thing is that acquiring with PulseAudio on Audacity show the problem, however using ALSA directly the issue is not present. Trying using arecord still present the issue from PulseAudio. > > > > Tried to play with the parameters /sys/devices/audio to no avail. Compared the pinconfigs from Windows, they are identical. > > > > Can someone try to help me fix the issue? > > > > Sent with ProtonMail Secure Email. > > > > [2 alsaaudacitylog <application/octet-stream (base64)>] > > > > > > [3 pulseaudacitylog <application/octet-stream (base64)>] > > > [4 arecordlog <application/octet-stream (base64)>] > > > > > > Alsa-devel mailing list > > > Alsa-devel(a)alsa-project.org > > > https://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

1 0

[alsa-devel] [PATCH v2] ALSA: usx2y: fix a double free bug
by Wenwen Wang 29 Apr '19

29 Apr '19

In usX2Y_In04_init(), a new urb is firstly created through usb_alloc_urb() and saved to 'usX2Y->In04urb'. Then, a buffer is allocated through kmalloc() and saved to 'usX2Y->In04Buf'. If the allocation of the buffer fails, the error code ENOMEM is returned after usb_free_urb(), which frees the created urb. However, the urb is actually freed at card->private_free callback, i.e., snd_usX2Y_card_private_free(). So the free operation here leads to a double free bug. To fix the above issue, simply remove usb_free_urb(). Signed-off-by: Wenwen Wang <wang6495(a)umn.edu> --- sound/usb/usx2y/usbusx2y.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index da4a5a5..1f0b0100 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -293,10 +293,8 @@ int usX2Y_In04_init(struct usX2Ydev *usX2Y) if (! (usX2Y->In04urb = usb_alloc_urb(0, GFP_KERNEL))) return -ENOMEM; - if (! (usX2Y->In04Buf = kmalloc(21, GFP_KERNEL))) { - usb_free_urb(usX2Y->In04urb); + if (! (usX2Y->In04Buf = kmalloc(21, GFP_KERNEL))) return -ENOMEM; - } init_waitqueue_head(&usX2Y->In04WaitQueue); usb_fill_int_urb(usX2Y->In04urb, usX2Y->dev, usb_rcvintpipe(usX2Y->dev, 0x4), -- 2.7.4

2 1

[alsa-devel] [PATCH] spi: Madera codecs: add missing of table registration
by Daniel Gomez 29 Apr '19

29 Apr '19

MODULE_DEVICE_TABLE(of, <of_match_table> should be called to complete DT OF mathing mechanism and register it. Before this patch: modinfo drivers/mfd/madera-spi.ko | grep alias alias: spi:wm1840 alias: spi:cs47l91 alias: spi:cs47l90 alias: spi:cs47l85 alias: spi:cs47l35 After this patch: modinfo drivers/mfd/madera-spi.ko | grep alias alias: spi:wm1840 alias: spi:cs47l91 alias: spi:cs47l90 alias: spi:cs47l85 alias: spi:cs47l35 alias: of:N*T*Ccirrus,wm1840C* alias: of:N*T*Ccirrus,wm1840 alias: of:N*T*Ccirrus,cs47l91C* alias: of:N*T*Ccirrus,cs47l91 alias: of:N*T*Ccirrus,cs47l90C* alias: of:N*T*Ccirrus,cs47l90 alias: of:N*T*Ccirrus,cs47l85C* alias: of:N*T*Ccirrus,cs47l85 alias: of:N*T*Ccirrus,cs47l35C* alias: of:N*T*Ccirrus,cs47l35 Reported-by: Javier Martinez Canillas <javier(a)dowhile0.org> Signed-off-by: Daniel Gomez <dagmcr(a)gmail.com> --- drivers/mfd/madera-spi.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/drivers/mfd/madera-spi.c b/drivers/mfd/madera-spi.c index 4c398b2..1128584 100644 --- a/drivers/mfd/madera-spi.c +++ b/drivers/mfd/madera-spi.c @@ -121,6 +121,16 @@ static const struct spi_device_id madera_spi_ids[] = { }; MODULE_DEVICE_TABLE(spi, madera_spi_ids); +const struct of_device_id madera_of_match[] = { + { .compatible = "cirrus,cs47l35", .data = (void *)CS47L35 }, + { .compatible = "cirrus,cs47l85", .data = (void *)CS47L85 }, + { .compatible = "cirrus,cs47l90", .data = (void *)CS47L90 }, + { .compatible = "cirrus,cs47l91", .data = (void *)CS47L91 }, + { .compatible = "cirrus,wm1840", .data = (void *)WM1840 }, + {} +}; +MODULE_DEVICE_TABLE(of, madera_of_match); + static struct spi_driver madera_spi_driver = { .driver = { .name = "madera", -- 2.7.4

2 1

Re: [alsa-devel] ALC892 acquisition stutter
by rodomar705＠protonmail.com 29 Apr '19

29 Apr '19

Maybe, just maybe, I found a probable cause of what's happening here. Since it was working with ALSA directly only with audacity, I follow the instruction on tracing the HDA commands sent to the codec while acquiring audio with Audacity. One with ALSA directly (just with Audacity thought, arecord show the same problem from Pulse), when is working correctly, one without it, using pulse. The logs are very interesting. The working one just simply sent two command to the audio codec, and then the stream started. The other one spammed a lot of more data, and it appears that the problem (and the continuous popping of the audio) derives from the fact that the codec keep closing the stream after 100 to 300 ms, that force pulse to ask the audio controller to restart the stream and cause an actual noise coming from the microphone (since the stream is not instantly restarted). There is another little difference beetween the second command sent to the codec directly with ALSA (from audacity) and the command sent from Pulse: ALSA with Audacity: 1106.051129: hda_send_cmd: [0000:09:00.3:0] val=0x00820031 Pulse with Audacity: 1231.103757: hda_send_cmd: [0000:09:00.3:0] val=0x00820011 However with arecord -d 10 -fS16_LE -r44100 /tmp/test-mic.wav as a parameter I still get almost the same as Pulseaudo: arecord with Audacity: 2891.230874: hda_send_cmd: [0000:09:00.3:0] val=0x00824011 However, no start stop behaviour like on pulseaudio, however the audio clicking are still there. The log are attached to this email, hoping that will help fix the issue. Sent with [ProtonMail](https://protonmail.com) Secure Email. ‐‐‐‐‐‐‐ Original Message ‐‐‐‐‐‐‐ Il domenica 28 aprile 2019 10:52, <rodomar705(a)protonmail.com> ha scritto: > Already sent to alsa-user, however a developer asked to send this also on devel, so here it is: > > I have audio stutters with an ALC892 Realtek Chipset on a B450 GAMING ELITE board from Gigabyte during acquisition (at least with microphone input, both front and back. Line in not tested). No issues with Windows. > > After opening an issue and testing with a PulseAudio developer, he has determined that the issue is not within PulseAudio, but on ALSA. > > There are already two issues about that on the bug tracker on Linux, namely this two: > > ALC892: https://bugzilla.kernel.org/show_bug.cgi?id=203351 > ALC1220: https://bugzilla.kernel.org/show_bug.cgi?id=195303 > > To me both appears to be identical, for this reason even if I have a ALC892 I’ve posted on the second one, just because it had more posts. > > The particular thing is that acquiring with PulseAudio on Audacity show the problem, however using ALSA directly the issue is not present. Trying using arecord still present the issue from PulseAudio. > > Tried to play with the parameters /sys/devices/audio to no avail. Compared the pinconfigs from Windows, they are identical. > > Can someone try to help me fix the issue? > > Sent with [ProtonMail](https://protonmail.com) Secure Email.

2 1

[alsa-devel] [PATCH RFC V2] ALSA: usb-audio: Scarlett Gen 2 mixer interface
by Geoffrey D. Bennett 29 Apr '19

29 Apr '19

Add mixer quirk for the Focusrite Scarlett 18i8 and 18i20 Gen 2 audio interfaces. Although the interfaces are USB compliant, additional input/output level controls and hardware routing/mixing functionality are available using proprietary USB requests. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- Hi all, This patch adds the following controls for the Scarlett 18i8 and 18i20 Gen 2: - Master volume knob indicator (18i20 only) - Volume controls for the analogue HW outputs - HW/SW volume switches for the 10 analogue HW outputs (18i20 only) - Output mux (where the sound for the HW outputs comes from; defaults to PCM outputs) - Capture mux (where the sound for PCM recording comes from; defaults to HW inputs) - Matrix mux (where the sound going into the mixer comes from; 18 inputs default off) - Mixer matrix (18 inputs * 10 outputs = 180 controls) - Mute and dim indicators (18i20 only) - Line Level/Instrument Level and Pad controls (18i8 only) - Level meters Changes since v1: - Add support for the Scarlett 18i8 Gen 2 - Save configuration parameters to NVRAM - Implemented feedback from Takashi's email 24/Apr/2019 - Moved private field from struct snd_usb_audio to struct usb_mixer_interface - Added timer and buffer fields to struct usb_mixer_interface - Other small code fixes/cleanups/improvements Outstanding issues/questions: - Adding fields to struct usb_mixer_interface seemed a better place than struct snd_usb_audio as in version 1 of this patch (I think these new fields are used in the same sort of context as the "Sound Blaster remote control stuff" fields, and I am reusing rc_urb and rc_setup_packet). I wasn't sure of the naming; should they be named generically ("timer", "private") or specifically ("scarlett2_buffer")? Should the Sound Blaster RC fields that I'm reusing be renamed to have generic names? - There are software controllable pads on the 18i8 line-inputs. I called the controls "Line In %d Pad Capture Switch". They work correctly in alsamixer (shown in the capture view as e.g. "Line In 1 Pad") but the "on" state is displayed as "L R CAPTURE" which looks unusual. I could use an Enum with Off/On values, but I thought this is what "Switch" is for (as per control-names.rst). Should I leave this as-is, or is there a better name that will make this show up as a regular on/off switch? amixer contents shows: numid=15,iface=MIXER,name='Line In 1 Pad Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off amixer scontents shows: Simple mixer control 'Line In 1 Pad',0 Capabilities: cswitch cswitch-joined Capture channels: Mono Mono: Capture [off] Thanks, Geoffrey. diff --git a/sound/usb/Makefile b/sound/usb/Makefile index d330f74c90e6..8cf7081e17cb 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -11,6 +11,7 @@ snd-usb-audio-objs := card.o \ mixer.o \ mixer_quirks.o \ mixer_scarlett.o \ + mixer_scarlett_gen2.o \ mixer_us16x08.o \ pcm.o \ power.o \ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e7d441d0e839..8cd1fc2f7580 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3521,6 +3521,9 @@ void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer) usb_kill_urb(mixer->urb); if (mixer->rc_urb) usb_kill_urb(mixer->rc_urb); + kfree(mixer->scarlett2_buffer); + del_timer_sync(&mixer->timer); + kfree(mixer->private); mixer->disconnected = true; } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 3d12af8bf191..d009f6cb74b9 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -25,6 +25,11 @@ struct usb_mixer_interface { u8 rc_buffer[6]; bool disconnected; + + /* Scarlett Gen 2 */ + void *scarlett2_buffer; + struct timer_list timer; + void *private; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 85ae0ff2382a..faa5e149ef10 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -45,6 +45,7 @@ #include "mixer.h" #include "mixer_quirks.h" #include "mixer_scarlett.h" +#include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" #include "helper.h" @@ -2273,6 +2274,11 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_scarlett_controls_create(mixer); break; + case USB_ID(0x1235, 0x8204): /* Focusrite Scarlett 18i8 2nd Gen */ + case USB_ID(0x1235, 0x8201): /* Focusrite Scarlett 18i20 2nd Gen */ + err = snd_scarlett_gen2_controls_create(mixer); + break; + case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */ err = snd_soundblaster_e1_switch_create(mixer); break; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c new file mode 100644 index 000000000000..c19a520c0d69 --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.c @@ -0,0 +1,2032 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Focusrite Scarlett 18i8/18i20 Gen 2 Driver for ALSA + * + * Copyright (c) 2018-2019 by Geoffrey D. Bennett <g at b4.vu> + * + * Based on the Scarlett (Gen 1) Driver for ALSA: + * + * Copyright (c) 2013 by Tobias Hoffmann + * Copyright (c) 2013 by Robin Gareus <robin at gareus.org> + * Copyright (c) 2002 by Takashi Iwai <tiwai at suse.de> + * Copyright (c) 2014 by Chris J Arges <chris.j.arges at canonical.com> + * + * Many codes borrowed from audio.c by + * Alan Cox (alan at lxorguk.ukuu.org.uk) + * Thomas Sailer (sailer at ife.ee.ethz.ch) + * + * Code cleanup: + * David Henningsson <david.henningsson at canonical.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +/* Mixer Interface for the Focusrite Scarlett 18i8/18i20 Gen 2 audio + * interface. Based on the Gen 1 driver and rewritten. + */ + +/* The protocol was reverse engineered by looking at the communication + * between Focusrite Control 2.3.4 and the Focusrite(R) Scarlett 18i20 + * (firmware 1083) using usbmon in July-August 2018. + * + * Scarlett 18i8 support added in April 2019. + * + * This ALSA mixer gives access to: + * - input, output, mixer-matrix muxes + * - 18x10 mixer-matrix gain stages + * - gain/volume controls + * - level meters + * - line/inst level and pad controls + * + * <ditaa> + * /--------------\ 18chn 20chn /--------------\ + * | Hardware in +--+------\ /-------------+--+ ALSA PCM out | + * \--------------/ | | | | \--------------/ + * | | | /-----\ | + * | | | | | | + * | v v v | | + * | +---------------+ | | + * | \ Matrix Mux / | | + * | +-----+-----+ | | + * | | | | + * | |18chn | | + * | | | | + * | | 10chn| | + * | v | | + * | +------------+ | | + * | | Mixer | | | + * | | Matrix | | | + * | | | | | + * | | 18x10 Gain | | | + * | | stages | | | + * | +-----+------+ | | + * | | | | + * |18chn |10chn | |20chn + * | | | | + * | +----------/ | + * | | | + * v v v + * =========================== + * +---------------+ +--—------------+ + * \ Output Mux / \ Capture Mux / + * +---+---+---+ +-----+-----+ + * | | | + * 10chn| | |18chn + * | | | + * /--------------\ | | | /--------------\ + * | S/PDIF, ADAT |<--/ |10chn \-->| ALSA PCM in | + * | Hardware out | | \--------------/ + * \--------------/ | + * v + * +-------------+ switch per channel to select + * | Master Gain |<-- front panel volume knob or + * +------+------+ individual software gain + * | (18i20 only) + * |10chn + * /--------------\ | + * | Analogue |<------/ + * | Hardware out | + * \--------------/ + * </ditaa> + * + */ + +#include <linux/slab.h> +#include <linux/usb.h> + +#include <sound/control.h> +#include <sound/tlv.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "helper.h" + +#include "mixer_scarlett_gen2.h" + +/* some gui mixers can't handle negative ctl values */ +#define SCARLETT_VOLUME_BIAS 127 + +/* mixer range from -80dB to +6dB in 0.5dB steps */ +#define SCARLETT_MIXER_MIN_DB -80 +#define SCARLETT_MIXER_BIAS (-SCARLETT_MIXER_MIN_DB * 2) +#define SCARLETT_MIXER_MAX_DB 6 +#define SCARLETT_MIXER_MAX_VALUE \ + ((SCARLETT_MIXER_MAX_DB - SCARLETT_MIXER_MIN_DB) * 2) + +/* map from (dB + 80) * 2 to mixer value + * for dB in 0 .. 172: int(8192 * pow(10, ((dB - 160) / 2 / 20))) + */ +static const u16 scarlett2_mixer_values[173] = { + 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, + 2, 2, 3, 3, 3, 3, 3, 4, 4, 4, 4, 5, 5, 5, 6, 6, 6, 7, 7, 8, 8, + 9, 9, 10, 10, 11, 12, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, + 23, 24, 25, 27, 29, 30, 32, 34, 36, 38, 41, 43, 46, 48, 51, + 54, 57, 61, 65, 68, 73, 77, 81, 86, 91, 97, 103, 109, 115, + 122, 129, 137, 145, 154, 163, 173, 183, 194, 205, 217, 230, + 244, 259, 274, 290, 307, 326, 345, 365, 387, 410, 434, 460, + 487, 516, 547, 579, 614, 650, 689, 730, 773, 819, 867, 919, + 973, 1031, 1092, 1157, 1225, 1298, 1375, 1456, 1543, 1634, + 1731, 1833, 1942, 2057, 2179, 2308, 2445, 2590, 2744, 2906, + 3078, 3261, 3454, 3659, 3876, 4105, 4349, 4606, 4879, 5168, + 5475, 5799, 6143, 6507, 6892, 7301, 7733, 8192, 8677, 9191, + 9736, 10313, 10924, 11571, 12257, 12983, 13752, 14567, 15430, + 16345 +}; + +/* Maximum number of analogue outputs */ +#define SCARLETT2_ANALOGUE_MAX 10 + +/* Maximum number of level and pad switches */ +#define SCARLETT2_LEVEL_SWITCH_MAX 2 +#define SCARLETT2_PAD_SWITCH_MAX 4 + +/* Maximum number of inputs to the mixer */ +#define SCARLETT2_INPUT_MIX_MAX 18 + +/* Maximum number of outputs from the mixer */ +#define SCARLETT2_OUTPUT_MIX_MAX 10 + +/* Maximum size of the data in the USB mux assignment message: + * 18 inputs, 20 outputs, 18 matrix inputs, 8 spare + */ +#define SCARLETT2_MUX_MAX 64 + +/* Number of meters: + * 18 inputs, 20 outputs, 18 matrix inputs + */ +#define SCARLETT2_NUM_METERS 56 + +/* Hardware port types: + * - None (no input to mux) + * - Analogue I/O + * - S/PDIF I/O + * - ADAT I/O + * - Mixer I/O + * - PCM I/O + */ +enum { + SCARLETT2_PORT_TYPE_NONE = 0, + SCARLETT2_PORT_TYPE_ANALOGUE = 1, + SCARLETT2_PORT_TYPE_SPDIF = 2, + SCARLETT2_PORT_TYPE_ADAT = 3, + SCARLETT2_PORT_TYPE_MIX = 4, + SCARLETT2_PORT_TYPE_PCM = 5, + SCARLETT2_PORT_TYPE_COUNT = 6, +}; + +/* Count of total I/O and number available at each sample rate */ +enum { + SCARLETT2_PORT_IN = 0, + SCARLETT2_PORT_OUT = 1, + SCARLETT2_PORT_OUT_44 = 2, + SCARLETT2_PORT_OUT_88 = 3, + SCARLETT2_PORT_OUT_176 = 4, + SCARLETT2_PORT_DIRECTIONS = 5, +}; + +/* Hardware buttons on the 18i20 */ +#define SCARLETT2_BUTTON_MAX 2 + +static const char *const scarlett2_button_names[SCARLETT2_BUTTON_MAX] = { + "Mute", "Dim" +}; + +/* Description of each hardware port type: + * - id: hardware ID for this port type + * - num: number of sources/destinations of this port type + * - src_descr: printf format string for mux input selections + * - src_num_offset: added to channel number for the fprintf + * - dst_descr: printf format string for mixer controls + */ +struct scarlett2_ports { + u16 id; + int num[SCARLETT2_PORT_DIRECTIONS]; + const char * const src_descr; + int src_num_offset; + const char * const dst_descr; +}; + +struct scarlett2_device_info { + u8 line_out_hw_vol; /* line out hw volume is sw controlled */ + u8 button_count; /* number of buttons */ + u8 level_input_count; /* inputs with level selectable */ + u8 pad_input_count; /* inputs with pad selectable */ + const char * const line_out_descrs[SCARLETT2_ANALOGUE_MAX]; + struct scarlett2_ports ports[SCARLETT2_PORT_TYPE_COUNT]; +}; + +struct scarlett2_mixer_data { + struct mutex usb_mutex; /* prevent sending concurrent USB requests */ + struct mutex data_mutex; /* lock access to this data */ + const struct scarlett2_device_info *info; + int num_mux_srcs; + u16 scarlett2_seq; + u8 vol_updated; + u8 master_vol; + u8 vol[SCARLETT2_ANALOGUE_MAX]; + u8 vol_sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; + u8 level_switch[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX]; + u8 buttons[SCARLETT2_BUTTON_MAX]; + struct snd_kcontrol *master_vol_ctl; + struct snd_kcontrol *vol_ctls[SCARLETT2_ANALOGUE_MAX]; + struct snd_kcontrol *button_ctls[SCARLETT2_ANALOGUE_MAX]; + u8 mux[SCARLETT2_MUX_MAX]; + u8 mix[SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX]; +}; + +struct scarlett2_usb_volume_status { + u8 buttons[SCARLETT2_BUTTON_MAX]; /* mute & dim buttons */ + u8 pad1; + s16 sw_vol[10]; /* software volume setting */ + s16 hw_vol[10]; /* actual volume of output inc. dim (-18dB) */ + u8 pad2[10]; + u8 sw_hw_switch[10]; /* sw (0) or hw (1) controlled */ + u8 pad3[6]; + s16 master_vol; /* front panel volume knob */ +} __packed; + +/*** Model-specific data ***/ + +static const struct scarlett2_device_info s18i8_gen2_info = { + /* The first two analogue inputs can be switched between line + * and instrument levels. + */ + .level_input_count = 2, + + /* The first four analogue inputs have an optional pad. */ + .pad_input_count = 4, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 4 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 6, 6, 6, 6 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 0, 0, 0, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +static const struct scarlett2_device_info s18i20_gen2_info = { + /* The analogue line outputs on the 18i20 can be switched + * between software and hardware volume control + */ + .line_out_hw_vol = 1, + + /* Mute and dim buttons */ + .button_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + NULL, + NULL, + NULL, + NULL, + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 6 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 10, 10, 10, 10 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 8, 8, 4, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + .dst_descr = "ADAT Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +/* get the starting port index number for a given port type/direction */ +static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, + int direction, int port_type) +{ + int i, num = 0; + + for (i = 0; i < port_type; i++) + num += ports[i].num[direction]; + + return num; +} + +/*** USB Interactions ***/ + +/* Vendor-Specific Interface, Endpoint, MaxPacketSize, Interval */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE 5 +#define SCARLETT2_USB_INTERRUPT_ENDPOINT 4 +#define SCARLETT2_USB_INTERRUPT_MAX_DATA 64 +#define SCARLETT2_USB_INTERRUPT_INTERVAL 3 + +/* Interrupt flags for volume and mute/dim button changes */ +#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x400000 +#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x200000 + +/* Commands for sending/receiving requests/responses */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP 3 + +#define SCARLETT2_USB_INIT_SEQ 0x00000000 +#define SCARLETT2_USB_GET_METER_LEVELS 0x00001001 +#define SCARLETT2_USB_SET_MIX 0x00002002 +#define SCARLETT2_USB_SET_MUX 0x00003002 +#define SCARLETT2_USB_GET_DATA 0x00800000 +#define SCARLETT2_USB_SET_DATA 0x00800001 +#define SCARLETT2_USB_DATA_CMD 0x00800002 +#define SCARLETT2_USB_CONFIG_SAVE 6 + +#define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31 +#define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1 + +/* Configuration parameters that can be read and written */ +enum { + SCARLETT2_CONFIG_LINE_OUT_VOLUME = 0, + SCARLETT2_CONFIG_SW_HW_SWITCH = 1, + SCARLETT2_CONFIG_LEVEL_SWITCH = 2, + SCARLETT2_CONFIG_PAD_SWITCH = 3, + SCARLETT2_CONFIG_COUNT = 4 +}; + +/* Location, size, and activation command number for the configuration + * parameters + */ +struct scarlett2_config { + u8 offset; + u8 size; + u8 activate; +}; + +static const struct scarlett2_config + scarlett2_config_items[SCARLETT2_CONFIG_COUNT] = { + /* Line Out Volume */ + { + .offset = 0x34, + .size = 2, + .activate = 1 + }, + + /* SW/HW Volume Switch */ + { + .offset = 0x66, + .size = 1, + .activate = 3 + }, + + /* Level Switch */ + { + .offset = 0x7c, + .size = 1, + .activate = 7 + }, + + /* Pad Switch */ + { + .offset = 0x84, + .size = 1, + .activate = 8 + } +}; + +/* proprietary request/response format */ +struct scarlett2_usb_packet { + u32 cmd; + u16 size; + u16 seq; + u32 error; + u32 pad; + u8 data[]; +}; + +#define SCARLETT2_USB_PACKET_LEN (sizeof(struct scarlett2_usb_packet)) + +static void scarlett2_fill_request_header(struct scarlett2_mixer_data *private, + struct scarlett2_usb_packet *req, + u32 cmd, u16 req_size) +{ + /* sequence must go up by 1 for each request */ + u16 seq = private->scarlett2_seq++; + + req->cmd = cpu_to_le32(cmd); + req->size = cpu_to_le16(req_size); + req->seq = cpu_to_le16(seq); + req->error = 0; + req->pad = 0; +} + +/* Send a proprietary format request to the Scarlett interface */ +static int scarlett2_usb( + struct usb_mixer_interface *mixer, u32 cmd, + void *req_data, u16 req_size, void *resp_data, u16 resp_size) +{ + struct scarlett2_mixer_data *private = mixer->private; + + u16 req_buf_size = sizeof(struct scarlett2_usb_packet) + req_size; + u16 resp_buf_size = sizeof(struct scarlett2_usb_packet) + resp_size; + + struct scarlett2_usb_packet *req = 0, *resp = 0; + + int err = 0; + + mutex_lock(&private->usb_mutex); + + /* build request message and send it */ + + req = kmalloc(req_buf_size, GFP_KERNEL); + if (!req) { + err = -ENOMEM; + goto unlock; + } + + scarlett2_fill_request_header(private, req, cmd, req_size); + + if (req_size) + memcpy(req->data, req_data, req_size); + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + req, + req_buf_size); + + if (err != req_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB request result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + /* send a second message to get the response */ + + resp = kmalloc(resp_buf_size, GFP_KERNEL); + if (!resp) { + err = -ENOMEM; + goto unlock; + } + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + resp, + resp_buf_size); + + /* validate the response */ + + if (err != resp_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB response result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + if (resp->cmd != req->cmd || + resp->seq != req->seq || + resp_size != le16_to_cpu(resp->size) || + resp->error || + resp->pad) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB invalid response; " + "cmd tx/rx %d/%d seq %d/%d size %d/%d " + "error %d pad %d\n", + le16_to_cpu(req->cmd), le16_to_cpu(resp->cmd), + le16_to_cpu(req->seq), le16_to_cpu(resp->seq), + resp_size, le16_to_cpu(resp->size), + le16_to_cpu(resp->error), + le16_to_cpu(resp->pad)); + err = -EINVAL; + goto unlock; + } + + if (resp_size > 0) + memcpy(resp_data, resp->data, resp_size); + +unlock: + kfree(req); + kfree(resp); + mutex_unlock(&private->usb_mutex); + return err; +} + +/* Clean up after SCARLETT2_USB_CONFIG_SAVE response is received */ +static void scarlett2_config_save_response_complete(struct urb *urb) +{ + struct usb_mixer_interface *mixer = urb->context; + struct scarlett2_usb_packet *resp = mixer->scarlett2_buffer; + + if (urb->status < 0) { + usb_audio_err(mixer->chip, "%s: status %d\n", + __func__, urb->status); + } else if (resp->error) { + usb_audio_err(mixer->chip, "%s: error %d\n", + __func__, le16_to_cpu(resp->error)); + } + + usb_free_urb(mixer->rc_urb); + mixer->rc_urb = 0; + kfree(mixer->rc_setup_packet); + mixer->rc_setup_packet = 0; +} + +/* Fill mixer->rc_setup_packet and mixer->urb then submit it */ +static void scarlett2_send_mixer_control_urb( + struct usb_mixer_interface *mixer, u8 dirn, u8 req, + int len, usb_complete_t complete_fn) +{ + struct usb_ctrlrequest *ctrl = mixer->rc_setup_packet; + + ctrl->bRequestType = USB_RECIP_INTERFACE | USB_TYPE_CLASS | dirn; + ctrl->bRequest = req; + ctrl->wValue = cpu_to_le16(0); + ctrl->wIndex = cpu_to_le16(SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE); + ctrl->wLength = cpu_to_le16(len); + + usb_fill_control_urb(mixer->rc_urb, mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + (u8 *)ctrl, mixer->scarlett2_buffer, len, + complete_fn, mixer); + + usb_submit_urb(mixer->rc_urb, GFP_ATOMIC); +} + +/* SCARLETT2_USB_CONFIG_SAVE is complete; send a request for the response */ +static void scarlett2_config_save_complete(struct urb *urb) +{ + struct usb_mixer_interface *mixer = urb->context; + int len = SCARLETT2_USB_PACKET_LEN; + + if (urb->status < 0) { + usb_audio_err(mixer->chip, + "%s: status %d\n", __func__, urb->status); + + usb_free_urb(mixer->rc_urb); + mixer->rc_urb = 0; + kfree(mixer->rc_setup_packet); + mixer->rc_setup_packet = 0; + + return; + } + + scarlett2_send_mixer_control_urb( + mixer, USB_DIR_IN, SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP, + len, scarlett2_config_save_response_complete); +} + +/* Send SCARLETT2_USB_DATA_CMD SCARLETT2_USB_CONFIG_SAVE */ +static void scarlett2_config_save(struct timer_list *t) +{ + struct usb_mixer_interface *mixer = from_timer(mixer, t, timer); + struct scarlett2_usb_packet *req = mixer->scarlett2_buffer; + struct scarlett2_mixer_data *private = mixer->private; + int len = SCARLETT2_USB_PACKET_LEN + sizeof(u32); + + if (mixer->rc_urb) { + usb_audio_err(mixer->chip, + "%s: mixer rc_urb already in use!\n", __func__); + return; + } + mixer->rc_urb = usb_alloc_urb(0, GFP_ATOMIC); + if (!mixer->rc_urb) { + usb_audio_err(mixer->chip, + "%s: usb_alloc_urb() failed!\n", __func__); + return; + } + mixer->rc_setup_packet = + kmalloc(sizeof(*mixer->rc_setup_packet), GFP_KERNEL); + if (!mixer->rc_setup_packet) { + usb_free_urb(mixer->rc_urb); + mixer->rc_urb = NULL; + usb_audio_err(mixer->chip, + "%s: kmalloc() failed!\n", __func__); + return; + } + + scarlett2_fill_request_header(private, mixer->scarlett2_buffer, + SCARLETT2_USB_DATA_CMD, sizeof(u32)); + *(u32 *)(req->data) = cpu_to_le32(SCARLETT2_USB_CONFIG_SAVE); + + scarlett2_send_mixer_control_urb( + mixer, USB_DIR_OUT, SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ, + len, scarlett2_config_save_complete); +} + +/* Send a USB message to set a configuration parameter (volume level, + * sw/hw volume switch, line/inst level switch, or pad switch) + */ +static int scarlett2_usb_set_config( + struct usb_mixer_interface *mixer, + int config_item_num, int index, int value) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + struct { + u32 offset; + u32 bytes; + s32 value; + } __packed req; + u32 req2; + int err; + + /* Send the configuration parameter data */ + req.offset = cpu_to_le32(config_item.offset + index * config_item.size); + req.bytes = cpu_to_le32(config_item.size); + req.value = cpu_to_le32(value); + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_DATA, + &req, sizeof(u32) * 2 + config_item.size, + NULL, 0); + if (err < 0) + return err; + + /* Activate the change */ + req2 = cpu_to_le32(config_item.activate); + err = scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req2, sizeof(req2), NULL, 0); + if (err < 0) + return err; + + /* Schedule the change to be written to NVRAM */ + mod_timer(&mixer->timer, jiffies + msecs_to_jiffies(2000)); + + return 0; +} + +/* Send a USB message to get data; result placed in *buf */ +static int scarlett2_usb_get( + struct usb_mixer_interface *mixer, + int offset, void *buf, int size) +{ + struct { + u32 offset; + u32 size; + } __packed req; + + req.offset = cpu_to_le32(offset); + req.size = cpu_to_le32(size); + return scarlett2_usb(mixer, SCARLETT2_USB_GET_DATA, + &req, sizeof(req), buf, size); +} + +/* Send a USB message to get configuration parameters; result placed in *buf */ +static int scarlett2_usb_get_config( + struct usb_mixer_interface *mixer, + int config_item_num, int count, void *buf) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + int size = config_item.size * count; + + return scarlett2_usb_get(mixer, config_item.offset, buf, size); +} + +/* Send a USB message to get volume status; result placed in *buf */ +static int scarlett2_usb_get_volume_status( + struct usb_mixer_interface *mixer, + struct scarlett2_usb_volume_status *buf) +{ + return scarlett2_usb_get(mixer, SCARLETT2_USB_VOLUME_STATUS_OFFSET, + buf, sizeof(*buf)); +} + +/* Send a USB message to set the volumes for all inputs of one mix + * (values obtained from private->mix[]) + */ +static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, + int mix_num) +{ + struct scarlett2_mixer_data *private = mixer->private; + const struct scarlett2_device_info *info = private->info; + + struct { + u16 mix_num; + u16 data[SCARLETT2_INPUT_MIX_MAX]; + } __packed req; + + int i, j; + int num_mixer_in = + info->ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + + req.mix_num = cpu_to_le16(mix_num); + + for (i = 0, j = mix_num * num_mixer_in; i < num_mixer_in; i++, j++) + req.data[i] = cpu_to_le16( + scarlett2_mixer_values[private->mix[j]] + ); + + return scarlett2_usb(mixer, SCARLETT2_USB_SET_MIX, + &req, (num_mixer_in + 1) * sizeof(u16), + NULL, 0); +} + +/* Convert a port number index (per info->ports) to a hardware ID */ +static u32 scarlett2_mux_src_num_to_id(const struct scarlett2_ports *ports, + int num) +{ + int port_type; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (num < ports[port_type].num[SCARLETT2_PORT_IN]) + return ports[port_type].id | num; + num -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + /* Oops */ + return 0; +} + +/* Send USB messages to set mux inputs */ +static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int rate, port_dir_rate; + + static const int assignment_order[SCARLETT2_PORT_TYPE_COUNT] = { + SCARLETT2_PORT_TYPE_PCM, + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + SCARLETT2_PORT_TYPE_MIX, + SCARLETT2_PORT_TYPE_NONE, + }; + + struct { + u16 pad; + u16 num; + u32 data[SCARLETT2_MUX_MAX]; + } __packed req; + + req.pad = 0; + + /* mux settings for each rate */ + for (rate = 0, port_dir_rate = SCARLETT2_PORT_OUT_44; + port_dir_rate <= SCARLETT2_PORT_OUT_176; + rate++, port_dir_rate++) { + int order_num, i, err; + + req.num = cpu_to_le16(rate); + + for (order_num = 0, i = 0; + order_num < SCARLETT2_PORT_TYPE_COUNT; + order_num++) { + int port_type = assignment_order[order_num]; + int j = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + port_type); + int port_id = ports[port_type].id; + int channel; + + for (channel = 0; + channel < ports[port_type].num[port_dir_rate]; + channel++, i++, j++) + /* lower 12 bits for the destination and + * next 12 bits for the source + */ + req.data[i] = !port_id + ? 0 + : cpu_to_le32( + port_id | + channel | + scarlett2_mux_src_num_to_id( + ports, private->mux[j] + ) << 12 + ); + + /* skip private->mux[j] entries not output */ + j += ports[port_type].num[SCARLETT2_PORT_OUT] - + ports[port_type].num[port_dir_rate]; + } + + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_MUX, + &req, (i + 1) * sizeof(u32), + NULL, 0); + if (err < 0) + return err; + } + + return 0; +} + +/* Send USB message to get meter levels */ +static int scarlett2_usb_get_meter_levels(struct usb_mixer_interface *mixer, + u16 *levels) +{ + struct { + u16 pad; + u16 num_meters; + u32 magic; + } __packed req; + u32 resp[SCARLETT2_NUM_METERS]; + int i, err; + + req.pad = 0; + req.num_meters = cpu_to_le16(SCARLETT2_NUM_METERS); + req.magic = cpu_to_le32(SCARLETT2_USB_METER_LEVELS_GET_MAGIC); + err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER_LEVELS, + &req, sizeof(req), resp, sizeof(resp)); + if (err < 0) + return err; + + /* copy, convert to u16 */ + for (i = 0; i < SCARLETT2_NUM_METERS; i++) + levels[i] = resp[i]; + + return 0; +} + +/*** Control Functions ***/ + +/* helper function to create a new control */ +static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *ncontrol, + int index, int channels, const char *name, + struct snd_kcontrol **kctl_return) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + int err; + + elem = kzalloc(sizeof(*elem), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = index; + elem->head.id = index; + elem->channels = channels; + + kctl = snd_ctl_new1(ncontrol, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + + err = snd_usb_mixer_add_control(&elem->head, kctl); + if (err < 0) + return err; + + if (kctl_return) + *kctl_return = kctl; + + return 0; +} + +/*** Analogue Line Out Volume Controls ***/ + +/* Update hardware volume controls after receiving notification that + * they have changed + */ +static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private; + const struct scarlett2_ports *ports = private->info->ports; + struct scarlett2_usb_volume_status volume_status; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + + private->vol_updated = 0; + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + if (private->vol_sw_hw_switch[i]) + private->vol[i] = private->master_vol; + } + + for (i = 0; i < private->info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +static int scarlett2_volume_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT_VOLUME_BIAS; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->master_vol; + return 0; +} + +static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private; + int index = elem->control; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->vol[index]; + return 0; +} + +static int scarlett2_volume_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private; + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + del_timer_sync(&mixer->timer); + + oval = private->vol[index]; + val = ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol[index] = val; + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, val - SCARLETT_VOLUME_BIAS); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett_gain, -SCARLETT_VOLUME_BIAS * 100, 0 +); + +static const struct snd_kcontrol_new scarlett2_master_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_master_volume_ctl_get, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_volume_ctl_get, + .put = scarlett2_volume_ctl_put, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +/*** HW/SW Volume Switch Controls ***/ + +static int scarlett2_sw_hw_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "SW", "HW" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_sw_hw_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private; + + ucontrol->value.enumerated.item[0] = + private->vol_sw_hw_switch[elem->control]; + return 0; +} + +static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + del_timer_sync(&mixer->timer); + + oval = private->vol_sw_hw_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol_sw_hw_switch[index] = val; + + /* Change access mode to RO (hardware controlled volume) + * or RW (software controlled volume) + */ + if (val) + private->vol_ctls[index]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + else + private->vol_ctls[index]->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* Reset volume to master volume */ + private->vol[index] = private->master_vol; + + /* Set SW volume to current HW volume */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, private->master_vol - SCARLETT_VOLUME_BIAS); + if (err < 0) + goto unlock; + + /* Notify of RO/RW change */ + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_INFO, + &private->vol_ctls[index]->id); + + /* Send SW/HW switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_sw_hw_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_sw_hw_enum_ctl_info, + .get = scarlett2_sw_hw_enum_ctl_get, + .put = scarlett2_sw_hw_enum_ctl_put, +}; + +/*** Line Level/Instrument Level Switch Controls ***/ + +static int scarlett2_level_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "Line", "Inst" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private; + + ucontrol->value.enumerated.item[0] = + private->level_switch[elem->control]; + return 0; +} + +static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + del_timer_sync(&mixer->timer); + + oval = private->level_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->level_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_level_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_level_enum_ctl_info, + .get = scarlett2_level_enum_ctl_get, + .put = scarlett2_level_enum_ctl_put, +}; + +/*** Pad Switch Controls ***/ + +static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private; + + ucontrol->value.enumerated.item[0] = + private->pad_switch[elem->control]; + return 0; +} + +static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + del_timer_sync(&mixer->timer); + + oval = private->pad_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->pad_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PAD_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_pad_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_pad_ctl_get, + .put = scarlett2_pad_ctl_put, +}; + +/*** Mute/Dim Controls ***/ + +static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.enumerated.item[0] = private->buttons[elem->control]; + return 0; +} + +static const struct snd_kcontrol_new scarlett2_button_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_button_ctl_get +}; + +/*** Create the analogue output controls ***/ + +static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add R/O HW volume control */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), "Master HW Playback Volume"); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_master_volume_ctl, + 0, 1, s, &private->master_vol_ctl); + if (err < 0) + return err; + } + + /* Add volume controls */ + for (i = 0; i < num_line_out; i++) { + + /* Fader */ + if (info->line_out_descrs[i]) + snprintf(s, sizeof(s), + "Line %02d (%s) Playback Volume", + i + 1, info->line_out_descrs[i]); + else + snprintf(s, sizeof(s), + "Line %02d Playback Volume", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_line_out_volume_ctl, + i, 1, s, &private->vol_ctls[i]); + if (err < 0) + return err; + + /* Make the fader read-only if the SW/HW switch is set to HW */ + if (private->vol_sw_hw_switch[i]) + private->vol_ctls[i]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* SW/HW Switch */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), + "Line Out %02d Volume Control Playback Enum", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_sw_hw_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + /* Add HW button controls */ + for (i = 0; i < private->info->button_count; i++) { + err = scarlett2_add_new_ctl(mixer, &scarlett2_button_ctl, + i, 1, scarlett2_button_names[i], + &private->button_ctls[i]); + if (err < 0) + return err; + } + + return 0; +} + +/*** Create the analogue input controls ***/ + +static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private; + const struct scarlett2_device_info *info = private->info; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add input level (line/inst) controls */ + for (i = 0; i < info->level_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Level Capture Enum", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_level_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + /* Add input pad controls */ + for (i = 0; i < info->pad_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Pad Capture Switch", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_pad_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + return 0; +} + +/*** Mixer Volume Controls ***/ + +static int scarlett2_mixer_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT_MIXER_MAX_VALUE; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_mixer_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private; + + ucontrol->value.integer.value[0] = private->mix[elem->control]; + return 0; +} + +static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int oval, val, num_mixer_in, mix_num, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mix[elem->control]; + val = ucontrol->value.integer.value[0]; + num_mixer_in = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + mix_num = elem->control / num_mixer_in; + + if (oval == val) + goto unlock; + + private->mix[elem->control] = val; + err = scarlett2_usb_set_mix(mixer, mix_num); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett_mixer, + SCARLETT_MIXER_MIN_DB * 100, + SCARLETT_MIXER_MAX_DB * 100 +); + +static const struct snd_kcontrol_new scarlett2_mixer_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_mixer_ctl_info, + .get = scarlett2_mixer_ctl_get, + .put = scarlett2_mixer_ctl_put, + .private_value = SCARLETT_MIXER_MAX_DB, /* max value */ + .tlv = { .p = db_scale_scarlett_mixer } +}; + +static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private; + const struct scarlett2_ports *ports = private->info->ports; + int err, i, j; + int index; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + int num_inputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + int num_outputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; + + for (i = 0, index = 0; i < num_outputs; i++) { + for (j = 0; j < num_inputs; j++, index++) { + snprintf(s, sizeof(s), + "Mix %c Input %02d Playback Volume", + 'A' + i, j + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_mixer_ctl, + index, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Mux Source Selection Controls ***/ + +static int scarlett2_mux_src_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private; + const struct scarlett2_ports *ports = private->info->ports; + unsigned int item = uinfo->value.enumerated.item; + int items = private->num_mux_srcs; + int port_type; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = elem->channels; + uinfo->value.enumerated.items = items; + + if (item >= items) + item = uinfo->value.enumerated.item = items - 1; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (item < ports[port_type].num[SCARLETT2_PORT_IN]) { + sprintf(uinfo->value.enumerated.name, + ports[port_type].src_descr, + item + ports[port_type].src_num_offset); + return 0; + } + item -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + return -EINVAL; +} + +static int scarlett2_mux_src_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private; + + ucontrol->value.enumerated.item[0] = private->mux[elem->control]; + return 0; +} + +static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private; + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mux[index]; + val = clamp(ucontrol->value.integer.value[0], + 0L, private->num_mux_srcs - 1L); + + if (oval == val) + goto unlock; + + private->mux[index] = val; + err = scarlett2_usb_set_mux(mixer); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_mux_src_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_mux_src_enum_ctl_info, + .get = scarlett2_mux_src_enum_ctl_get, + .put = scarlett2_mux_src_enum_ctl_put, +}; + +static int scarlett2_add_mux_enums(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private; + const struct scarlett2_ports *ports = private->info->ports; + int port_type, channel, i; + + for (i = 0, port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + for (channel = 0; + channel < ports[port_type].num[SCARLETT2_PORT_OUT]; + channel++, i++) { + int err; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + const char *const descr = ports[port_type].dst_descr; + + snprintf(s, sizeof(s) - 5, descr, channel + 1); + strcat(s, " Enum"); + + err = scarlett2_add_new_ctl(mixer, + &scarlett2_mux_src_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Meter Controls ***/ + +static int scarlett2_meter_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 4095; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + u16 meter_levels[SCARLETT2_NUM_METERS]; + int i, err; + + err = scarlett2_usb_get_meter_levels(elem->head.mixer, meter_levels); + if (err < 0) + return err; + + for (i = 0; i < elem->channels; i++) + ucontrol->value.integer.value[i] = meter_levels[i]; + + return 0; +} + +static const struct snd_kcontrol_new scarlett2_meter_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .name = "", + .info = scarlett2_meter_ctl_info, + .get = scarlett2_meter_ctl_get +}; + +static int scarlett2_add_meter_ctl(struct usb_mixer_interface *mixer) +{ + return scarlett2_add_new_ctl(mixer, &scarlett2_meter_ctl, + 0, SCARLETT2_NUM_METERS, + "Level Meter", NULL); +} + +/*** Initialisation ***/ + +static int scarlett2_count_mux_srcs(const struct scarlett2_ports *ports) +{ + int port_type, count = 0; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) + count += ports[port_type].num[SCARLETT2_PORT_IN]; + + return count; +} + +/* Default routing connects PCM outputs and inputs to Analogue, + * S/PDIF, then ADAT + */ +static void scarlett2_init_routing(u8 *mux, + const struct scarlett2_ports *ports) +{ + int i, input_num, input_count, port_type; + int output_num, output_count, port_type_connect_num; + + static const int connect_order[] = { + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + -1 + }; + + /* Assign PCM inputs (routing outputs) */ + output_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + SCARLETT2_PORT_TYPE_PCM); + output_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_OUT]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + input_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_IN, port_type); + input_count = ports[port_type].num[SCARLETT2_PORT_IN]; + for (i = 0; + i < input_count && output_count; + i++, output_count--) + mux[output_num++] = input_num++; + } + + /* Assign PCM outputs (routing inputs) */ + input_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_IN, + SCARLETT2_PORT_TYPE_PCM); + input_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_IN]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + output_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_OUT, port_type); + output_count = ports[port_type].num[SCARLETT2_PORT_OUT]; + for (i = 0; + i < output_count && input_count; + i++, input_count--) + mux[output_num++] = input_num++; + } +} + +/* Initialise private data, routing, sequence number */ +static int scarlett2_init_private(struct usb_mixer_interface *mixer, + const struct scarlett2_device_info *info) +{ + struct scarlett2_mixer_data *private = + kzalloc(sizeof(struct scarlett2_mixer_data), GFP_KERNEL); + + if (!private) + return -ENOMEM; + + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + timer_setup(&mixer->timer, scarlett2_config_save, 0); + private->info = info; + private->num_mux_srcs = scarlett2_count_mux_srcs(info->ports); + private->scarlett2_seq = 0; + mixer->private = private; + + /* enough for the SCARLETT2_USB_DATA_CMD request sent by the timer + * callback + */ + mixer->scarlett2_buffer = kmalloc( + SCARLETT2_USB_PACKET_LEN + sizeof(u32), GFP_KERNEL); + + /* Setup default routing */ + scarlett2_init_routing(private->mux, info->ports); + + /* Initialise the sequence number used for the proprietary commands */ + return scarlett2_usb(mixer, SCARLETT2_USB_INIT_SEQ, + NULL, 0, NULL, 0); +} + +/* Read line-in config and line-out volume settings on start */ +static int scarlett2_read_configs(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + u8 level_switches[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switches[SCARLETT2_PAD_SWITCH_MAX]; + struct scarlett2_usb_volume_status volume_status; + int err, i; + + if (info->level_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_LEVEL_SWITCH, + info->level_input_count, + level_switches); + if (err < 0) + return err; + for (i = 0; i < info->level_input_count; i++) + private->level_switch[i] = level_switches[i]; + } + + if (info->pad_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_PAD_SWITCH, + info->pad_input_count, + pad_switches); + if (err < 0) + return err; + for (i = 0; i < info->pad_input_count; i++) + private->pad_switch[i] = pad_switches[i]; + } + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + int volume; + + private->vol_sw_hw_switch[i] = + info->line_out_hw_vol + && volume_status.sw_hw_switch[i]; + + volume = private->vol_sw_hw_switch[i] + ? volume_status.master_vol + : volume_status.sw_vol[i]; + volume = clamp(volume + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + private->vol[i] = volume; + } + + for (i = 0; i < info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +/* Notify on volume change */ +static void scarlett2_mixer_interrupt_vol_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private; + const struct scarlett2_ports *ports = private->info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int i; + + private->vol_updated = 1; + + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->master_vol_ctl->id); + + for (i = 0; i < num_line_out; i++) { + if (!private->vol_sw_hw_switch[i]) + continue; + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->vol_ctls[i]->id); + } +} + +/* Notify on button change */ +static void scarlett2_mixer_interrupt_button_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private; + int i; + + private->vol_updated = 1; + + for (i = 0; i < private->info->button_count; i++) + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->button_ctls[i]->id); +} + +/* Interrupt callback */ +static void scarlett2_mixer_interrupt(struct urb *urb) +{ + struct usb_mixer_interface *mixer = urb->context; + int len = urb->actual_length; + int ustatus = urb->status; + u32 data; + + if (ustatus != 0) + goto requeue; + + if (len == 8) { + data = le32_to_cpu(*(u32 *)urb->transfer_buffer); + if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) + scarlett2_mixer_interrupt_vol_change(mixer); + if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) + scarlett2_mixer_interrupt_button_change(mixer); + } else { + usb_audio_err(mixer->chip, + "scarlett mixer interrupt length %d\n", len); + } + +requeue: + if (ustatus != -ENOENT && + ustatus != -ECONNRESET && + ustatus != -ESHUTDOWN) { + urb->dev = mixer->chip->dev; + usb_submit_urb(urb, GFP_ATOMIC); + } +} + +static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer) +{ + void *transfer_buffer; + + if (mixer->urb) { + usb_audio_err(mixer->chip, + "%s: mixer urb already in use!\n", __func__); + return 0; + } + + mixer->urb = usb_alloc_urb(0, GFP_KERNEL); + if (!mixer->urb) + return -ENOMEM; + + transfer_buffer = kmalloc(SCARLETT2_USB_INTERRUPT_MAX_DATA, GFP_KERNEL); + if (!transfer_buffer) + return -ENOMEM; + + usb_fill_int_urb( + mixer->urb, + mixer->chip->dev, + usb_rcvintpipe(mixer->chip->dev, + SCARLETT2_USB_INTERRUPT_ENDPOINT), + transfer_buffer, + SCARLETT2_USB_INTERRUPT_MAX_DATA, + scarlett2_mixer_interrupt, + mixer, + SCARLETT2_USB_INTERRUPT_INTERVAL); + + return usb_submit_urb(mixer->urb, GFP_KERNEL); +} + +/* Entry point */ +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer) +{ + const struct scarlett2_device_info *info; + int err; + + /* only use UAC_VERSION_2 */ + if (!mixer->protocol) + return 0; + + switch (mixer->chip->usb_id) { + case USB_ID(0x1235, 0x8204): + info = &s18i8_gen2_info; + break; + case USB_ID(0x1235, 0x8201): + info = &s18i20_gen2_info; + break; + default: /* device not (yet) supported */ + return -EINVAL; + } + + /* Initialise private data, routing, sequence number */ + err = scarlett2_init_private(mixer, info); + if (err < 0) + return err; + + /* Read volume levels and controls from the interface */ + err = scarlett2_read_configs(mixer); + if (err < 0) + return err; + + /* Create the analogue output controls */ + err = scarlett2_add_line_out_ctls(mixer); + if (err < 0) + return err; + + /* Create the analogue input controls */ + err = scarlett2_add_line_in_ctls(mixer); + if (err < 0) + return err; + + /* Create the input, output, and mixer mux input selections */ + err = scarlett2_add_mux_enums(mixer); + if (err < 0) + return err; + + /* Create the matrix mixer controls */ + err = scarlett2_add_mixer_ctls(mixer); + if (err < 0) + return err; + + /* Create the level meter controls */ + err = scarlett2_add_meter_ctl(mixer); + if (err < 0) + return err; + + /* Set up the interrupt polling if there are hardware buttons */ + if (info->button_count) { + err = scarlett2_mixer_status_create(mixer); + if (err < 0) + return err; + } + + return 0; +} diff --git a/sound/usb/mixer_scarlett_gen2.h b/sound/usb/mixer_scarlett_gen2.h new file mode 100644 index 000000000000..52e1dad77afd --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __USB_MIXER_SCARLETT_GEN2_H +#define __USB_MIXER_SCARLETT_GEN2_H + +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer); + +#endif /* __USB_MIXER_SCARLETT_GEN2_H */

2 1

[alsa-devel] [PATCH] quirks for Roland MX-1
by Julian Wiesener 29 Apr '19

29 Apr '19

Hello, the attached Patch workarrounds the fact that the device appears to be working ASYNC, however it will be hanging, not playing or capturing a single byte. The 2 channel Altset kind of works, but sounds garbage. The 18 channel Altset works as intended. It needs to be cutted down to be usefull, channels 16 and 17 are the default "PC" outputs, i attached a asoundrc for the device, providing 9 stereo interfaces. In the default configuration with no other devices attached, mxout4 to mxout9 will be working stereoi outputs bound to faders on the mixer. I have not yet tested the MIDI interface. Kind regards, Julian

2 2

[alsa-devel] [PATCH 0/6] ASoC: hdmi-codec: fixes and improvements
by Jerome Brunet 29 Apr '19

29 Apr '19

This patchset is a collection of fixes and improvement on the hdmi codec driver. It should be completely transparent for the current users of the codec. The most important change is removal the current_substream pointer which allows the codec to be used on codec-to-codec links. I plan to use this for the HDMI sound support of the Amlogic g12a SoC family. Jerome Brunet (6): ASoC: hdmi-codec: remove function name debug traces ASoC: hdmi-codec: unlock the device on startup errors ASoC: hdmi-codec: stream is already locked in hw_params ASoC: hdmi-codec: remove reference to the current substream ASoC: hdmi-codec: remove reference to the dai drivers in the private data ASoC: hdmi-codec: remove ops dependency on the dai id sound/soc/codecs/hdmi-codec.c | 259 ++++++++++++++++------------------ 1 file changed, 125 insertions(+), 134 deletions(-) -- 2.20.1

1 6

[alsa-devel] [PATCH 1/1] ASoC: Intel: Skylake: change list empty check to warning
by Patrick Waterlander 29 Apr '19

29 Apr '19

NULL is a valid state for the module list when querying unique ID. Downgrade the empty condition from error to warning. Signed-off-by: Patrick Waterlander <patrick.waterlander(a)intel.com> diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 5951bbdf1f1a..95e6a23538cd 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -324,7 +324,7 @@ static struct skl_module_table *skl_module_get_from_id( struct skl_module_table *module; if (list_empty(&ctx->module_list)) { - dev_err(ctx->dev, "Module list is empty\n"); + dev_warn(ctx->dev, "Module list is empty\n"); return NULL; } -- 2.17.1

2 1

[alsa-devel] [PATCH v7 0/6] drm/bridge: sii902x: HDMI-audio support and some fixes
by Jyri Sarha 29 Apr '19

29 Apr '19

The already reviewed patches are in front. Not reviewed still patches 5-6. Changes since v6: - "dt-bindings: display: sii902x: Add HDMI audio bindings" - Replace references to ASoC (a Linux term) with more generic terms - Add "sil,"-prefix to "i2s-data-lanes"-property in the example node - "drm/bridge: sii902x: Implement HDMI audio support" - Do not use unsigned in sii902x_select_mclk_div()'s "distance" calculations to avoid bad type promotions to unsigned causing unexpected results - Always return the clock divisor by sii902x_select_mclk_div() - Disable regmap locking since the driver specific locking should be enough - Add missing mutex locking to the register access in the irq handler The previous round: https://patchwork.kernel.org/cover/10881169/ Jyri Sarha (5): drm/bridge: sii902x: Set output mode to HDMI or DVI according to EDID drm/bridge: sii902x: pixel clock unit is 10kHz instead of 1kHz dt-bindings: display: sii902x: Remove trailing white space dt-bindings: display: sii902x: Add HDMI audio bindings drm/bridge: sii902x: Implement HDMI audio support Tomi Valkeinen (1): drm/bridge: sii902x: add input_bus_flags .../bindings/display/bridge/sii902x.txt | 43 +- drivers/gpu/drm/bridge/sii902x.c | 488 +++++++++++++++++- 2 files changed, 523 insertions(+), 8 deletions(-) -- Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 8

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Add support for high power analog output
by Saravanan Sekar 29 Apr '19

29 Apr '19

Add support to power and output level control for the analog high power output drivers HPOUT and HPCOM. Signed-off-by: Saravanan Sekar <sravanhome(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 516d17cb2182..d4bafac802eb 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -419,6 +419,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Pop reduction */ SOC_ENUM("Output Driver Power-On time", aic3x_poweron_time_enum), SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum), + + /* Analog HPOUT, HPCOM power and output level controls */ + SOC_DOUBLE_R("Analog output power control", HPROUT_CTRL, + HPRCOM_CTRL, 0, 1, 0), + SOC_DOUBLE_R("Analog output level control", HPROUT_CTRL, + HPRCOM_CTRL, 4, 9, 0), }; /* For other than tlv320aic3104 */ -- 2.17.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: fix hw_params/hw_free validity checks
by Jerome Brunet 29 Apr '19

29 Apr '19

While working on a codec, I have found an odd situation. The hw_params callback which was provided by the codec dai was not called as expected. However, the hw_free callback was called, even if hw_params was skipped. After debugging, I found out that the dai was considered invalid because it did not specify any rates, only rate_min and rate_max. That's a first problem. If a codec dai is considered invalid, hw_free should probably not be called either, that's a 2nd issue. This patchset is meant to solve both issues. Jerome Brunet (2): ASoC: fix valid stream condition ASoC: skip hw_free on codec dai for which the stream is invalid sound/soc/soc-pcm.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) -- 2.20.1

1 2

[alsa-devel] [PATCH 1/2] ALSA: hda: fix unregister device twice on ASoC driver
by Bard liao 29 Apr '19

29 Apr '19

snd_hda_codec_device_new() is used by both legacy HDA and ASoC driver. However, we will call snd_hdac_device_unregister() in snd_hdac_ext_bus_device_remove() for ASoC device. This patch uses the type flag in hdac_device struct to determine is it a ASoC device or legacy HDA device and call snd_hdac_device_unregister() in snd_hda_codec_dev_free() only if it is a legacy HDA device. Signed-off-by: Bard liao <yung-chuan.liao(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 701a69d856f5..b20eb7fc83eb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -832,7 +832,13 @@ static int snd_hda_codec_dev_free(struct snd_device *device) struct hda_codec *codec = device->device_data; codec->in_freeing = 1; - snd_hdac_device_unregister(&codec->core); + /* + * snd_hda_codec_device_new() is used by legacy HDA and ASoC driver. + * We can't unregister ASoC device since it will be unregistered in + * snd_hdac_ext_bus_device_remove(). + */ + if (codec->core.type == HDA_DEV_LEGACY) + snd_hdac_device_unregister(&codec->core); codec_display_power(codec, false); put_device(hda_codec_dev(codec)); return 0; -- 2.17.1

2 3

[alsa-devel] [PATCH] ALSA: usx2y: fix a memory leak bug
by Wenwen Wang 29 Apr '19

29 Apr '19

In usX2Y_In04_init(), a new urb is firstly created through usb_alloc_urb() and saved to 'usX2Y->In04urb'. Then, a buffer is allocated through kmalloc() and saved to 'usX2Y->In04Buf'. After the urb is initialized, a sanity check is performed for the endpoint in the urb by invoking usb_urb_ep_type_check(). If the check fails, the error code EINVAL will be returned. In that case, however, the created urb and the allocated buffer are not freed, leading to memory leaks. To fix the above issue, free the urb and the buffer if the check fails. Signed-off-by: Wenwen Wang <wang6495(a)umn.edu> --- sound/usb/usx2y/usbusx2y.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index da4a5a5..0817018 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -303,8 +303,11 @@ int usX2Y_In04_init(struct usX2Ydev *usX2Y) usX2Y->In04Buf, 21, i_usX2Y_In04Int, usX2Y, 10); - if (usb_urb_ep_type_check(usX2Y->In04urb)) + if (usb_urb_ep_type_check(usX2Y->In04urb)) { + kfree(usX2Y->In04Buf); + usb_put_urb(usX2Y->In04urb); return -EINVAL; + } return usb_submit_urb(usX2Y->In04urb, GFP_KERNEL); } -- 2.7.4

2 5

[alsa-devel] [PATCH] sound: USB: line6: use dynamic buffers
by Greg Kroah-Hartman 28 Apr '19

28 Apr '19

The line6 driver uses a lot of USB buffers off of the stack, which is not allowed on many systems, causing the driver to crash on some of them. Fix this up by dynamically allocating the buffers with kmalloc() which allows for proper DMA-able memory. Reported-by: Christo Gouws <gouws.christo(a)gmail.com> Reported-by: Alan Stern <stern(a)rowland.harvard.edu> Tested-by: Christo Gouws <gouws.christo(a)gmail.com> Cc: stable <stable(a)vger.kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/usb/line6/driver.c | 60 ++++++++++++++++++++++++++------------------- sound/usb/line6/podhd.c | 21 +++++++++------ sound/usb/line6/toneport.c | 23 ++++++++++++----- 3 files changed, 64 insertions(+), 40 deletions(-) --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -351,12 +351,16 @@ int line6_read_data(struct usb_line6 *li { struct usb_device *usbdev = line6->usbdev; int ret; - unsigned char len; + unsigned char *len; unsigned count; if (address > 0xffff || datalen > 0xff) return -EINVAL; + len = kmalloc(sizeof(*len), GFP_KERNEL); + if (!len) + return -ENOMEM; + /* query the serial number: */ ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, @@ -365,7 +369,7 @@ int line6_read_data(struct usb_line6 *li if (ret < 0) { dev_err(line6->ifcdev, "read request failed (error %d)\n", ret); - return ret; + goto exit; } /* Wait for data length. We'll get 0xff until length arrives. */ @@ -375,28 +379,29 @@ int line6_read_data(struct usb_line6 *li ret = usb_control_msg(usbdev, usb_rcvctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, - 0x0012, 0x0000, &len, 1, + 0x0012, 0x0000, len, 1, LINE6_TIMEOUT * HZ); if (ret < 0) { dev_err(line6->ifcdev, "receive length failed (error %d)\n", ret); - return ret; + goto exit; } - if (len != 0xff) + if (*len != 0xff) break; } - if (len == 0xff) { + ret = -EIO; + if (*len == 0xff) { dev_err(line6->ifcdev, "read failed after %d retries\n", count); - return -EIO; - } else if (len != datalen) { + goto exit; + } else if (*len != datalen) { /* should be equal or something went wrong */ dev_err(line6->ifcdev, "length mismatch (expected %d, got %d)\n", - (int)datalen, (int)len); - return -EIO; + (int)datalen, (int)*len); + goto exit; } /* receive the result: */ @@ -405,12 +410,12 @@ int line6_read_data(struct usb_line6 *li 0x0013, 0x0000, data, datalen, LINE6_TIMEOUT * HZ); - if (ret < 0) { + if (ret < 0) dev_err(line6->ifcdev, "read failed (error %d)\n", ret); - return ret; - } - return 0; +exit: + kfree(len); + return ret; } EXPORT_SYMBOL_GPL(line6_read_data); @@ -422,12 +427,16 @@ int line6_write_data(struct usb_line6 *l { struct usb_device *usbdev = line6->usbdev; int ret; - unsigned char status; + unsigned char *status; int count; if (address > 0xffff || datalen > 0xffff) return -EINVAL; + status = kmalloc(sizeof(*status), GFP_KERNEL); + if (!status) + return -ENOMEM; + ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, 0x0022, address, data, datalen, @@ -436,7 +445,7 @@ int line6_write_data(struct usb_line6 *l if (ret < 0) { dev_err(line6->ifcdev, "write request failed (error %d)\n", ret); - return ret; + goto exit; } for (count = 0; count < LINE6_READ_WRITE_MAX_RETRIES; count++) { @@ -447,28 +456,29 @@ int line6_write_data(struct usb_line6 *l USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, 0x0012, 0x0000, - &status, 1, LINE6_TIMEOUT * HZ); + status, 1, LINE6_TIMEOUT * HZ); if (ret < 0) { dev_err(line6->ifcdev, "receiving status failed (error %d)\n", ret); - return ret; + goto exit; } - if (status != 0xff) + if (*status != 0xff) break; } - if (status == 0xff) { + if (*status == 0xff) { dev_err(line6->ifcdev, "write failed after %d retries\n", count); - return -EIO; - } else if (status != 0) { + ret = -EIO; + } else if (*status != 0) { dev_err(line6->ifcdev, "write failed (error %d)\n", ret); - return -EIO; + ret = -EIO; } - - return 0; +exit: + kfree(status); + return ret; } EXPORT_SYMBOL_GPL(line6_write_data); --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -225,28 +225,32 @@ static void podhd_startup_start_workqueu static int podhd_dev_start(struct usb_line6_podhd *pod) { int ret; - u8 init_bytes[8]; + u8 *init_bytes; int i; struct usb_device *usbdev = pod->line6.usbdev; + init_bytes = kmalloc(8, GFP_KERNEL); + if (!init_bytes) + return -ENOMEM; + ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, 0x11, 0, NULL, 0, LINE6_TIMEOUT * HZ); if (ret < 0) { dev_err(pod->line6.ifcdev, "read request failed (error %d)\n", ret); - return ret; + goto exit; } /* NOTE: looks like some kind of ping message */ ret = usb_control_msg(usbdev, usb_rcvctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, 0x11, 0x0, - &init_bytes, 3, LINE6_TIMEOUT * HZ); + init_bytes, 3, LINE6_TIMEOUT * HZ); if (ret < 0) { dev_err(pod->line6.ifcdev, "receive length failed (error %d)\n", ret); - return ret; + goto exit; } pod->firmware_version = @@ -255,7 +259,7 @@ static int podhd_dev_start(struct usb_li for (i = 0; i <= 16; i++) { ret = line6_read_data(&pod->line6, 0xf000 + 0x08 * i, init_bytes, 8); if (ret < 0) - return ret; + goto exit; } ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), @@ -263,10 +267,9 @@ static int podhd_dev_start(struct usb_li USB_TYPE_STANDARD | USB_RECIP_DEVICE | USB_DIR_OUT, 1, 0, NULL, 0, LINE6_TIMEOUT * HZ); - if (ret < 0) - return ret; - - return 0; +exit: + kfree(init_bytes); + return ret; } static void podhd_startup_workqueue(struct work_struct *work) --- a/sound/usb/line6/toneport.c +++ b/sound/usb/line6/toneport.c @@ -365,16 +365,21 @@ static bool toneport_has_source_select(s /* Setup Toneport device. */ -static void toneport_setup(struct usb_line6_toneport *toneport) +static int toneport_setup(struct usb_line6_toneport *toneport) { - u32 ticks; + u32 *ticks; struct usb_line6 *line6 = &toneport->line6; struct usb_device *usbdev = line6->usbdev; + ticks = kmalloc(sizeof(*ticks), GFP_KERNEL); + if (!ticks) + return -ENOMEM; + /* sync time on device with host: */ /* note: 32-bit timestamps overflow in year 2106 */ - ticks = (u32)ktime_get_real_seconds(); - line6_write_data(line6, 0x80c6, &ticks, 4); + *ticks = (u32)ktime_get_real_seconds(); + line6_write_data(line6, 0x80c6, ticks, 4); + kfree(ticks); /* enable device: */ toneport_send_cmd(usbdev, 0x0301, 0x0000); @@ -451,7 +456,9 @@ static int toneport_init(struct usb_line return err; } - toneport_setup(toneport); + err = toneport_setup(toneport); + if (err) + return err; /* register audio system: */ return snd_card_register(line6->card); @@ -463,7 +470,11 @@ static int toneport_init(struct usb_line */ static int toneport_reset_resume(struct usb_interface *interface) { - toneport_setup(usb_get_intfdata(interface)); + int err; + + err = toneport_setup(usb_get_intfdata(interface)); + if (err) + return err; return line6_resume(interface); } #endif

3 3

[alsa-devel] (No Subject)
by rodomar705＠protonmail.com 28 Apr '19

28 Apr '19

Already sent to alsa-user, however a developer asked to send this also on devel, so here it is: I have audio stutters with an ALC892 Realtek Chipset on a B450 GAMING ELITE board from Gigabyte during acquisition (at least with microphone input, both front and back. Line in not tested). No issues with Windows. After opening an issue and testing with a PulseAudio developer, he has determined that the issue is not within PulseAudio, but on ALSA. There are already two issues about that on the bug tracker on Linux, namely this two: ALC892: https://bugzilla.kernel.org/show_bug.cgi?id=203351 ALC1220: https://bugzilla.kernel.org/show_bug.cgi?id=195303 To me both appears to be identical, for this reason even if I have a ALC892 I’ve posted on the second one, just because it had more posts. The particular thing is that acquiring with PulseAudio on Audacity show the problem, however using ALSA directly the issue is not present. Trying using arecord still present the issue from PulseAudio. Tried to play with the parameters /sys/devices/audio to no avail. Compared the pinconfigs from Windows, they are identical. Can someone try to help me fix the issue? Sent with [ProtonMail](https://protonmail.com) Secure Email.

1 0

[alsa-devel] [PATCH v2] ALSA: hda/realtek: Enable audio output of ASUS UX362FA with ALC294
by Ladislav Michl 28 Apr '19

28 Apr '19

The ASUS UX362FA with ALC294 cannot detect the outputs through the internal speaker and the headphone until ALC294_FIXUP_ASUS_SPK quirk applied. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=203443 Suggested-by: Petr Nosek <petr.nosek(a)gmail.com> Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- CHANGES: - v2: adjusted commit message 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 070749b50eff..490b42e3e3dc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6896,6 +6896,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x13e1, "ASUS UX362FA", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), -- 2.20.1

1 0

[alsa-devel] HDAudio and pulse audio
by rander.wang 28 Apr '19

28 Apr '19

Hi all, I am working on sound open firmware in ASoC to enable HDAudio with pulse audio in ubuntu. There is a HDAudio codec driver hdac_hda.c in ASoC, which invokes HDAudio functions defined in sound/pci/hda, for ASoC to utilize the HDAudio codec. Now I get a issue: pulse audio can't detect headphone pluged or not. Could anyone help to tell me about Jack detection flow in HDAudio and how does HDAudio work with pulse audio ? Does HDAudio report a event to pulse audio when headphone detected ? As I know, ASoC codec driver would invoke snd_soc_jack_report to notify user mode, how about HDAudio driver ? Thanks! Rander

2 1

Re: [alsa-devel] [PATCH] ALSA: hda/realtek: Enable internal speaker of ASUS UX362FA with ALC294
by Ladislav Michl 28 Apr '19

28 Apr '19

Hi Petr, please keep list Cced. On Sun, Apr 28, 2019 at 12:48:06AM +0100, Petr Nosek wrote: > Hi Ladislav, > > it repairs both. Inernal speaker and headphones. Ok, will adjust commit message. > I'm sending output of dmesg - I think it helps you to see information, > when is computer booting. Your patch looks like good. Assuming this part of your boot log comes from patched kernel, right? snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC294: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 snd_hda_codec_realtek hdaudioC0D0: inputs: snd_hda_codec_realtek hdaudioC0D0: Headset Mic=0x19 snd_hda_codec_realtek hdaudioC0D0: Internal Mic=0x12 > Can I do something else to help with repair? I'm not sure, what you mean > by the last sentence. If I have anything else to do. I'll send v2 shortly, Thank you, ladis

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Fix a memory leak bug
by Wenwen Wang 28 Apr '19

28 Apr '19

In parse_audio_selector_unit(), the string array 'namelist' is allocated through kmalloc_array(), and each string pointer in this array, i.e., 'namelist[]', is allocated through kmalloc() in the following for loop. Then, a control instance 'kctl' is created by invoking snd_ctl_new1(). If an error occurs during the creation process, the string array 'namelist', including all string pointers in the array 'namelist[]', should be freed, before the error code ENOMEM is returned. However, the current code does not free 'namelist[]', resulting in memory leaks. To fix the above issue, free all string pointers 'namelist[]' in a loop. Signed-off-by: Wenwen Wang <wang6495(a)umn.edu> --- sound/usb/mixer.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 73d7dff..53dccbf 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2675,6 +2675,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, kctl = snd_ctl_new1(&mixer_selectunit_ctl, cval); if (! kctl) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); + for (i = 0; i < desc->bNrInPins; i++) + kfree(namelist[i]); kfree(namelist); kfree(cval); return -ENOMEM; -- 2.7.4

2 1

[alsa-devel] [PATCH v3] sound: isa: gus: fix misuse of %x
by Fuqian Huang 28 Apr '19

28 Apr '19

Pointers should be printed with %p or %px rather than cast to long type and printed with %lx. Drop the address printing. Signed-off-by: Fuqian Huang <huangfq.daxian(a)gmail.com> --- sound/isa/gus/gus_mem.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 4ac76f46dd76..d708ae1525e4 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -306,7 +306,7 @@ static void snd_gf1_mem_info_read(struct snd_info_entry *entry, used = 0; for (block = alloc->first, i = 0; block; block = block->next, i++) { used += block->size; - snd_iprintf(buffer, "Block %i at 0x%lx onboard 0x%x size %i (0x%x):\n", i, (long) block, block->ptr, block->size, block->size); + snd_iprintf(buffer, "Block %i onboard 0x%x size %i (0x%x):\n", i, block->ptr, block->size, block->size); if (block->share || block->share_id[0] || block->share_id[1] || block->share_id[2] || block->share_id[3]) -- 2.11.0

2 1

[alsa-devel] EAPD turn on later
by Kailang 28 Apr '19

28 Apr '19

Hi Takashi, EAPD need to turn on after set pin output. This will avoid pop noise for more machine via speaker. BR, Kailang

2 1

[alsa-devel] Dell AIO speaker pop noise
by Kailang 28 Apr '19

28 Apr '19

Hi Takashi, Issue 1: The noise keeps occurring via speaker when Headset is plugged in. Issue 2: The humming noise is occuring when switching to sound of settings. The noise will be stopped when switching to other options of settings. Issue 3: The popping noise occurred for one second when plugged in and removed the headset. This patch will solve this three issues. BR, Kailang

4 18

[alsa-devel] snd_hdac_device_unregister is called twice on ASoC driver
by Liao, Bard 28 Apr '19

28 Apr '19

Hi Takashi and all, We meet a problem that snd_hdac_device_unregister will be called twice on ASoC driver. We will call snd_hdac_ext_bus_device_init -> snd_hdac_device_register and snd_hdac_ext_bus_device_remove -> snd_hdac_device_unregister on ASoC driver. It looks even to me. However, hdac_hda_codec_probe will call snd_hda_codec_device_new to create a hda codec device. And it will assign snd_hda_codec_dev_free to the .dev_free ops where snd_hdac_device_unregister will be called. That's why snd_hdac_device_unregister will be call twice on ASoC driver. Below patch can "fix" this issue, but I don't know if it is the right way to fix it. Could you take a look and give me some advice? --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -841,7 +841,13 @@ static int snd_hda_codec_dev_free(struct snd_device *device) struct hda_codec *codec = device->device_data; codec->in_freeing = 1; - snd_hdac_device_unregister(&codec->core); + /* + * snd_hda_codec_device_new() is used by legacy HDA and ASoC driver. + * We can't unregister ASoC device since it will be unregistered in + * snd_hdac_ext_bus_device_remove(). + */ + if (codec->core.type == HDA_DEV_LEGACY) + snd_hdac_device_unregister(&codec->core); codec_display_power(codec, false); put_device(hda_codec_dev(codec)); return 0; --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -328,6 +328,12 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) dev_err(&hdev->dev, "failed to create hda codec %d\n", ret); goto error_no_pm; } + /* + * Overwrite type to HDA_DEV_ASOC since it is a ASoC driver + * hda_codec.c will check this flag to determine if unregister + * device is needed. + */ + hdev->type = HDA_DEV_ASOC; /* * snd_hda_codec_device_new decrements the usage count so call get pm

2 1

Re: [alsa-devel] [PATCH V3] ASoC: fsl_esai: Add pm runtime function
by S.j. Wang 28 Apr '19

28 Apr '19

Hi Mark > On Fri, Apr 26, 2019 at 10:51:15AM +0000, S.j. Wang wrote: > > > On Mon, Apr 22, 2019 at 02:31:55AM +0000, S.j. Wang wrote: > > > > Add pm runtime support and move clock handling there. > > > > Close the clocks at suspend to reduce the power consumption. > > > > > fsl_esai_suspend is replaced by pm_runtime_force_suspend. > > > > fsl_esai_resume is replaced by pm_runtime_force_resume. > > > > This doesn't apply against current code, please check and resend. > > > Which branch are you using? I tried for-next and for-linus, both Are > > successful applied. > > I'm applying against for-5.2, though if it depends on a patch queued for > 5.1 that's fine, I can just merge that up - please just resend. I think I did try > merging 5.1 though... I think may be caused by the patch " ASoC: fsl_esai: Fix missing break in switch statement", so I resend them both base on for-5.2. best regards wang shengjiu

1 0

[alsa-devel] no sound in laptop ASUS UX362FA - repair
by Petr Nosek 27 Apr '19

27 Apr '19

Hi, I bought laptop ASUS UX362FA: https://www.asus.com/2-in-1-PCs/ASUS-ZenBook-Flip-13-UX362FA/ The sound didn't work. I was looking for solution and I found this article: https://forum.manjaro.org/t/alc294-analog-audio-card-detected-but-no-sound-… So I added this line into ( https://github.com/torvalds/linux/blob/master/sound/pci/hda/patch_realtek.c ): SND_PCI_QUIRK(0x1043, 0x13e1, "ASUS UX362FA", ALC294_FIXUP_ASUS_SPK), behind this line: SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK) And recompile kernel and it works. I'm sending this report for others, who have the same problem with this laptop version. Please add this repair to kernel for the others. Thank you. my hwinfo: root@zenbook:~$ hwinfo --sound 15: PCI 1f.3: 0403 Audio device [Created at pci.378] Unique ID: nS1_.jsQALdVXpeB SysFS ID: /devices/pci0000:00/0000:00:1f.3 SysFS BusID: 0000:00:1f.3 Hardware Class: sound Model: "Intel Audio device" Vendor: pci 0x8086 "Intel Corporation" Device: pci 0x9dc8 SubVendor: pci 0x1043 "ASUSTeK Computer Inc." SubDevice: pci 0x13e1 Revision: 0x30 Driver: "snd_hda_intel" Driver Modules: "snd_hda_intel" Memory Range: 0xb1218000-0xb121bfff (rw,non-prefetchable) Memory Range: 0xb1000000-0xb10fffff (rw,non-prefetchable) IRQ: 127 (100 events) Module Alias: "pci:v00008086d00009DC8sv00001043sd000013E1bc04sc03i80" Driver Info #0: Driver Status: snd_hda_intel is active Driver Activation Cmd: "modprobe snd_hda_intel" Driver Info #1: Driver Status: snd_soc_skl is active Driver Activation Cmd: "modprobe snd_soc_skl" Config Status: cfg=new, avail=yes, need=no, active=unknown For repairing it is neccessary to compile kernel. I'm sending solution for other and if is it possible, please add it in ubuntu kernel. I reported bug (and solution) also there: https://bugzilla.kernel.org/show_bug.cgi?id=203443 additional information: root@zenbook:~$ lsb_release -a No LSB modules are available. Distributor ID: Ubuntu Description: Ubuntu 18.10 Release: 18.10 Codename: cosmic root@zenbook:~$ uname -a Linux zenbook 4.20.10-custom #1 SMP Sat Apr 6 21:21:30 CEST 2019 x86_64 x86_64 x86_64 GNU/Linux kernel compiled and patched from this source: https://kernel.ubuntu.com/~kernel-ppa/mainline/v4.20/ and added line: SND_PCI_QUIRK(0x1043, 0x13e1, "ASUS UX362FA", ALC294_FIXUP_ASUS_SPK), into /sound/pci/hda/patch_realtek.c file Petr

2 1

[alsa-devel] [PATCH v6 00/14] ASoC: Sound Open Firmware (SOF) core
by Pierre-Louis Bossart 27 Apr '19

27 Apr '19

Sound Open Firmware (SOF) is a host and DSP architecture agnostic audio DSP firmware. SOF is not tied to any specific host architecture or any specific physical IO communication type (it will work with on SoC DSPs, or DSP connected via SPI/I2C). SOF is also not coupled to any particular DSP architecture and has abstraction similar to Linux to allow porting to other DSP architectures. https://www.sofproject.org/ This patch series introduces the SOF core and utilities. Support for Intel devices is provided as a follow-up series. The SOF core manages all the core DSP services and ALSA/ASoC IO. The core is responsible for loading firmware, parsing topology, exposing PCMs and kcontrols, providing debug and trace mechanisms and performing IPC between host and DSP. The SOF core also has logic to allow reuse of existing machine drivers for other platforms/machines without any code modification. i.e. DAI links can be modified at runtime to bind with SOF and SOF topologies instead of existing hard coded DAI links and topology. Changes since v5: In addition to addressing feedback from Takashi, the major change in this series is the IPC rework. We reverted some optimizations to help Daniel Baluta @ NXP submit support for i.MX platforms, additional IPC core refactorization will start once we have two platforms supported on github. Hopefully by then the code will be accepted in the ASoC tree and we can submit delta patches, hint hint wink wink. All feedback from Takashi addressed: Tabify/align Kconfigs Remove INFO_RESUME flag Add const char* for core prototypes Fix snd_sof_get_status return value Use smd_pcm_format_physical_width() Fix hw_constraints and BUFFER_BYTES/SIZE confusion Remove useless fallthrough statements Fix typos and duplicate words in comments and commit messages Remove unnecessary casts, use container_of Remove superfluous initializations Fix potential race condition in debugfs Check for wrap-arounds in loader Fix alignment in debugfs read Don't use devm_kasprintf for firmware name string. Other updates: Major refactor of IPC handling (support for large messages, code optimization, better abstraction to push Intel-specific code out of the core). Regenerate compressed page table only if needed Support for multiple DAI hw_config (from <dragos_tarcatu(a)mentor.com>) Use iopoll macros (with small change due to indirection) Fix PM refcount handling on errors Rename/unify all definitions for effects (now 'process') Added basic support for keyword/hotword detection and binding widgets with external events Fix release of topology resources Support new IPC_PANIC message Clean-up mandatory/optional callbacks, simplify wrappers Add topology ABI check Fix acpi/machine driver handling Fix memory leak in debugfs error handling Alignment with firmware header files Fixed topology endianness issues Changes since v4: feedback from Mark: Removed need for private field in soc_runtime Completely reworked IPC, now much simpler and robust. IPC refactoring work will continue to e.g. allow for different types of IPC hardware and a better split between core and platform-specific drivers. Feedback from Takashi: Clarified endianness and alignment issues for page tables Removed useless mutexes Removed unneeded initializations, return types and wrappers Use snd_mask_set_format() Trace code flow optimization Removed checks on preallocate_pages() Test status of open/close functions. Released reference when pm_runtime_get_sync fails Fixed multiple memory allocation issues Fixed static analysis warnings Fixed error checks with copy_to_user() Added support for multi-core DSPs Fixed leaked references with firmware handling on resume Multiple topology fixes/improvements (enum/switch, mux, removal of duplicate connections, widget power not reported as ON) Code simplifications (removed set-but-unused, typecasts, etc). Changes since v3: Addressed dozens ofcomments from Takashi Iwai, Mark Brown, Andy Shevchenko, Daniel Baluta (Thanks!) Hardened memory allocation, fixed module load/unload oops or errors (now at hundreds of cycles on ApolloLake devices) Fixed suspend/resume and removed use of suspend_late, now using snd_soc_pm_ops and standard flow Fixed topology free (multiple patches already contributed for ASoC-core) Fixed debugfs/pm_runtime interaction Removed error checks on debugfs, changed to EXPORT_SYMBOL_GPL Added filenames in debug messages, ues hex_dump_to_buffer Removed "sof-audio" platform device and added better error checks on probe Reworked data structures to remove mix of const/variable fields and duplication of pointers for platform-specific changes Removed redundant ops tables Fixed trace/logger issues Fixed issues with loader (alignment and block size errors) Fixed release_firmware handling (was all over the place) Simplified PCI handling Added comments on non-atomic triggers Removed enum controls (not supported in topology) Simplified include file dependencies Fix cppcheck warnings Fixed Kconfigs to deal with Kbuild warnings on exotic architectures Precisions: The code in this patchset is directly squashed from the SOF development branch [1], which tracks Mark Brown's for-next branch on a weekly basis and the configurations used for testing are based on the defconfigs at [2]. The patches are also used backported for Chromebook devices. Full disclosure on known limitation and issues (full list at [3]) a) the interaction with ASoC is not always perfect, there are a couple of points where a better solution is desired (use of private data, support for link DMA, etc). The known points are explicitly documented in the code and will be updated. To be clearer, it's our belief that SOF does not cripple ASoC in any way, and that merging this SOF core does not harm others. It's just complicated to get things right, especially on a couple of cases where the topology and DPCM frameworks generate complex flows that very few people understand. b) The get/put methods for controls generate an IPC and possibly a wake-up. This is not optimal and is being fixed. c) runtime_pm is being hardened and the use of SMART_SUSPEND was suggested. Thank you for reviews and comments, we appreciate the time spent commenting on this large patchset. Thanks in particular to Alan Cox and Andy Shevchenko for their comments on an earlier version. This patchset also includes contributions from Daniel Baluta for loading code on non-Intel platforms. Pierre [1] https://github.com/thesofproject/linux [2] https://github.com/thesofproject/kconfig [3] https://github.com/thesofproject/linux/issues Liam Girdwood (12): ASoC: SOF: Add Sound Open Firmware driver core ASoC: SOF: Add Sound Open Firmware KControl support ASoC: SOF: Add driver debug support. ASoC: SOF: Add support for IPC IO between DSP and Host ASoC: SOF: Add PCM operations support ASoC: SOF: Add support for loading topologies ASoC: SOF: Add DSP firmware logger support ASoC: SOF: Add DSP HW abstraction operations ASoC: SOF: Add firmware loader support ASoC: SOF: Add userspace ABI support ASoC: SOF: Add PM support ASoC: SOF: Add Nocodec machine driver support Pierre-Louis Bossart (2): ASoC: SOF: Add xtensa support ASoC: SOF: Add utils include/sound/sof.h | 100 + include/sound/sof/control.h | 158 ++ include/sound/sof/dai-intel.h | 178 ++ include/sound/sof/dai.h | 75 + include/sound/sof/header.h | 158 ++ include/sound/sof/info.h | 118 ++ include/sound/sof/pm.h | 48 + include/sound/sof/stream.h | 148 ++ include/sound/sof/topology.h | 256 +++ include/sound/sof/trace.h | 67 + include/sound/sof/xtensa.h | 44 + include/uapi/sound/sof/abi.h | 62 + include/uapi/sound/sof/eq.h | 172 ++ include/uapi/sound/sof/fw.h | 78 + include/uapi/sound/sof/header.h | 27 + include/uapi/sound/sof/manifest.h | 188 ++ include/uapi/sound/sof/tokens.h | 107 + include/uapi/sound/sof/tone.h | 21 + include/uapi/sound/sof/trace.h | 66 + sound/soc/sof/control.c | 552 +++++ sound/soc/sof/core.c | 506 +++++ sound/soc/sof/debug.c | 232 +++ sound/soc/sof/ipc.c | 832 ++++++++ sound/soc/sof/loader.c | 400 ++++ sound/soc/sof/nocodec.c | 109 + sound/soc/sof/ops.c | 163 ++ sound/soc/sof/ops.h | 399 ++++ sound/soc/sof/pcm.c | 719 +++++++ sound/soc/sof/pm.c | 385 ++++ sound/soc/sof/sof-priv.h | 632 ++++++ sound/soc/sof/topology.c | 3164 +++++++++++++++++++++++++++++ sound/soc/sof/trace.c | 297 +++ sound/soc/sof/utils.c | 112 + sound/soc/sof/xtensa/Kconfig | 2 + sound/soc/sof/xtensa/Makefile | 5 + sound/soc/sof/xtensa/core.c | 138 ++ 36 files changed, 10718 insertions(+) create mode 100644 include/sound/sof.h create mode 100644 include/sound/sof/control.h create mode 100644 include/sound/sof/dai-intel.h create mode 100644 include/sound/sof/dai.h create mode 100644 include/sound/sof/header.h create mode 100644 include/sound/sof/info.h create mode 100644 include/sound/sof/pm.h create mode 100644 include/sound/sof/stream.h create mode 100644 include/sound/sof/topology.h create mode 100644 include/sound/sof/trace.h create mode 100644 include/sound/sof/xtensa.h create mode 100644 include/uapi/sound/sof/abi.h create mode 100644 include/uapi/sound/sof/eq.h create mode 100644 include/uapi/sound/sof/fw.h create mode 100644 include/uapi/sound/sof/header.h create mode 100644 include/uapi/sound/sof/manifest.h create mode 100644 include/uapi/sound/sof/tokens.h create mode 100644 include/uapi/sound/sof/tone.h create mode 100644 include/uapi/sound/sof/trace.h create mode 100644 sound/soc/sof/control.c create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/debug.c create mode 100644 sound/soc/sof/ipc.c create mode 100644 sound/soc/sof/loader.c create mode 100644 sound/soc/sof/nocodec.c create mode 100644 sound/soc/sof/ops.c create mode 100644 sound/soc/sof/ops.h create mode 100644 sound/soc/sof/pcm.c create mode 100644 sound/soc/sof/pm.c create mode 100644 sound/soc/sof/sof-priv.h create mode 100644 sound/soc/sof/topology.c create mode 100644 sound/soc/sof/trace.c create mode 100644 sound/soc/sof/utils.c create mode 100644 sound/soc/sof/xtensa/Kconfig create mode 100644 sound/soc/sof/xtensa/Makefile create mode 100644 sound/soc/sof/xtensa/core.c base-commit: 5f2ab35fe853f4886da490de8bc47637f5fc2921 -- 2.17.1

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[alsa-devel] [PATCH v5 00/21] ASoC: Sound Open Firmware (SOF) - Intel support
by Pierre-Louis Bossart 27 Apr '19

27 Apr '19

This patch series introduces the support for Intel devices (Baytrail, CherryTrail, Broadwell, ApolloLake, GeminiLake, CannonLake, IceLake). Support for Haswell, Skylake and KabyLake is incomplete at the moment and is not shared for now. The series starts with the 'legacy' devices, then all the more recent ones and concludes with ACPI/PCI hooks. SKL+ platforms can be compiled without HDaudio link/codec support, in which case the hdac library is not used. When support for HDMI or HDaudio codecs is selected, parts of this library are used. A small set of machine drivers supported by SOF is now provided as part of this series, more to follow. Support for SPI-based devices, compressed streams is also not provided for now since it's not ready for upstream just yet. Thank you for reviews and comments, we appreciate the time spent commenting on this large patchset. Thanks in particular to Alan Cox and Andy Shevchenko for their comments on an earlier version. Changes since v4: Feedback from Andy: Rename EDISON as MERRIFIELD, byt_pci_probe as tng_pci_probe Feedback from Takashi: Remove superfluous initializations and casts Tabify Kconfigs Fix dependency on SND_DMA_SGBUFF Other updates: Major IPC rework (Intel-specific parts moved out of the core) Fix NOCODEC/HDA compatibility Fix period bytes calculation Fix CTX_SAVE handling (do not read mailbox which can be powered off) Fix ROM message handling Drop unexpected IPC replies Add prepare for BE dailink Fix typos in comments Changes since v3: Removed Haswell, Skylake and Kabylake support (code is not tested or is still under development). These devices will be re-added when there is functional firmware and for Skylake/Kabylake a signed version of the firmware usable on existing platforms. Added support for existing machine drivers when relevant (more machine drivers will be added in next batch) Fixed hardware programming sequences (ROM init timeout, HDaudio capture, simultaneous capture and playback, DMA preload) Avoid reading IPC reply on CTX_SAVE reply since memory windows might be powered off. Simplified NOCODEC support, now mutually exclusive with HDaudio link to avoid suspend/resume issues. Changes since v2: Addressed dozens of comments from Takashi Iwai, Mark Brown, Andy Shevchenko, Daniel Baluta (Thanks!) Transition to new display power management Fixed suspend issues Optimized register polling, better handling of timeouts and firmware download retries Detect number of HDaudio codec and select machine driver accordingly Added workqueue based probe for HDaudio (dependency on modules) Fixed platform name override and conflicts with DT deferred probe changes Added prefix for firmware and topology files and module arguments Split Baytrail and Edison Fixed warnings (smatch, cppcheck) Guennadi Liakhovetski (1): ASoC: SOF: Intel: Add legacy IPC support Keyon Jie (2): ASoC: SOF: Intel: Add hda-bus support and initialization ASoC: SOF: Intel: add SKL+ platform DAIs Liam Girdwood (15): ASoC: SOF: Intel: Add BYT, CHT and BSW DSP HW support. ASoC: SOF: Intel: Add BDW HW DSP support ASoC: SOF: Intel: Add APL/CNL HW DSP support ASoC: SOF: Intel: Add HDA controller for Intel DSP ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations ASoC: SOF: Intel: Add Intel specific HDA IPC mechanisms. ASoC: SOF: Intel: Add Intel specific HDA firmware loader ASoC: SOF: Intel: Add Intel specific HDA PCM operations ASoC: SOF: Intel: Add Intel specific HDA stream operations ASoC: SOF: Intel: Add Intel specific HDA trace operations ASoC: SOF: Intel: Add platform differentiation for APL and CNL ASoC: SOF: Add ACPI device support ASoC: SOF: Add PCI device support ASoC: SOF: Add Build support for SOF core and Intel drivers ASoC: Intel: Make sure BDW based machine drivers build for SOF Pierre-Louis Bossart (3): ASoC: SOF: Intel: Add support for HDAudio codecs ASoC: Intel: Kconfig: expose common option between SST and SOF drivers ASoC: Intel: select relevant machine drivers for SOF sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/intel/Kconfig | 7 +- sound/soc/intel/boards/Kconfig | 22 +- sound/soc/intel/boards/bdw-rt5677.c | 4 + sound/soc/intel/boards/broadwell.c | 4 + sound/soc/sof/Kconfig | 141 +++++ sound/soc/sof/Makefile | 18 + sound/soc/sof/intel/Kconfig | 230 ++++++++ sound/soc/sof/intel/Makefile | 19 + sound/soc/sof/intel/apl.c | 109 ++++ sound/soc/sof/intel/bdw.c | 713 +++++++++++++++++++++++ sound/soc/sof/intel/byt.c | 874 ++++++++++++++++++++++++++++ sound/soc/sof/intel/cnl.c | 249 ++++++++ sound/soc/sof/intel/hda-bus.c | 108 ++++ sound/soc/sof/intel/hda-codec.c | 171 ++++++ sound/soc/sof/intel/hda-ctrl.c | 181 ++++++ sound/soc/sof/intel/hda-dai.c | 351 +++++++++++ sound/soc/sof/intel/hda-dsp.c | 455 +++++++++++++++ sound/soc/sof/intel/hda-ipc.c | 458 +++++++++++++++ sound/soc/sof/intel/hda-loader.c | 371 ++++++++++++ sound/soc/sof/intel/hda-pcm.c | 240 ++++++++ sound/soc/sof/intel/hda-stream.c | 692 ++++++++++++++++++++++ sound/soc/sof/intel/hda-trace.c | 94 +++ sound/soc/sof/intel/hda.c | 671 +++++++++++++++++++++ sound/soc/sof/intel/hda.h | 560 ++++++++++++++++++ sound/soc/sof/intel/intel-ipc.c | 92 +++ sound/soc/sof/intel/shim.h | 183 ++++++ sound/soc/sof/sof-acpi-dev.c | 312 ++++++++++ sound/soc/sof/sof-pci-dev.c | 373 ++++++++++++ 30 files changed, 7696 insertions(+), 8 deletions(-) create mode 100644 sound/soc/sof/Kconfig create mode 100644 sound/soc/sof/Makefile create mode 100644 sound/soc/sof/intel/Kconfig create mode 100644 sound/soc/sof/intel/Makefile create mode 100644 sound/soc/sof/intel/apl.c create mode 100644 sound/soc/sof/intel/bdw.c create mode 100644 sound/soc/sof/intel/byt.c create mode 100644 sound/soc/sof/intel/cnl.c create mode 100644 sound/soc/sof/intel/hda-bus.c create mode 100644 sound/soc/sof/intel/hda-codec.c create mode 100644 sound/soc/sof/intel/hda-ctrl.c create mode 100644 sound/soc/sof/intel/hda-dai.c create mode 100644 sound/soc/sof/intel/hda-dsp.c create mode 100644 sound/soc/sof/intel/hda-ipc.c create mode 100644 sound/soc/sof/intel/hda-loader.c create mode 100644 sound/soc/sof/intel/hda-pcm.c create mode 100644 sound/soc/sof/intel/hda-stream.c create mode 100644 sound/soc/sof/intel/hda-trace.c create mode 100644 sound/soc/sof/intel/hda.c create mode 100644 sound/soc/sof/intel/hda.h create mode 100644 sound/soc/sof/intel/intel-ipc.c create mode 100644 sound/soc/sof/intel/shim.h create mode 100644 sound/soc/sof/sof-acpi-dev.c create mode 100644 sound/soc/sof/sof-pci-dev.c -- 2.17.1

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[alsa-devel] [PATCH 0/3] Add runtime PM for SAI digital audio interface
by Daniel Baluta 27 Apr '19

27 Apr '19

First patch fixes a bug by correctly setting is_slave_mode, then second patch adds support for runtime PM and finally 3rd patch moves clock handling from startup/shtudown function to runtime PM handlers. Changes since v1: - added patch 1 - added fsl_sai_remove in order to call pm_runtime_disable - only disable/enable mclk when SAI in master mode. Daniel Baluta (2): ASoC: fsl_sai: Update is_slave_mode with correct value ASoC: fsl_sai: Add support for runtime pm Shengjiu Wang (1): ASoC: fsl_sai: Move clock operation to PM runtime sound/soc/fsl/fsl_sai.c | 84 +++++++++++++++++++++++++++++++++-------- 1 file changed, 69 insertions(+), 15 deletions(-) -- 2.17.1

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Re: [alsa-devel] [PATCH V3] ASoC: fsl_esai: Add pm runtime function
by S.j. Wang 27 Apr '19

27 Apr '19

Hi Mark > > On Mon, Apr 22, 2019 at 02:31:55AM +0000, S.j. Wang wrote: > > Add pm runtime support and move clock handling there. > > Close the clocks at suspend to reduce the power consumption. > > > > fsl_esai_suspend is replaced by pm_runtime_force_suspend. > > fsl_esai_resume is replaced by pm_runtime_force_resume. > > This doesn't apply against current code, please check and resend. Which branch are you using? I tried for-next and for-linus, both Are successful applied. Best regards Wang shengjiu

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[alsa-devel] Help with cross compilation of alsa-lib.
by Mathias Nielsen 27 Apr '19

27 Apr '19

Hi. I have made a small Go app (https://github.com/Lynges/susimup). It is a simple music player for the console and one of the libraries I use has alsa as a dependency on linux. I would really like to have cross compilation working so that I might provide binaries for the various platforms. I have tried getting it working using xgo (https://github.com/karalabe/xgo) and it solves the problem for both windows and macos, but on linux I have to get alsa-lib injected and compiled. I have created an SO question with some more details here: ( https://stackoverflow.com/questions/55656194/xgo-cross-compile-not-working-… ) I realize that I am asking no small thing unless someone on this list is very familiar with both xgo and alsa, but I could really use some help getting this working and I have spent quite a lot of time searching for other solutions and found nothing so I guess it would be helpful to others as well as my self. I do not have a whole lot of experience getting cross compilation for C working and the fact that the process is somewhat obscured by xgo does not make it any easier. One thing I have not tried yet is just forking xgo and using a different version of their docker image that then has the alsa packages installed with multiarch, but I have no idea how to make that work or if it even could work. Any more information I can provide, please dont hold back. -- Best regards Mathias Nielsen

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[alsa-devel] Will we ever see a low latency/high performance PCM api for usb, firewire, avb ?
by Ralf Beck 26 Apr '19

26 Apr '19

Subject says it all. During the Dec/Jan period i did some USB patches for the Motu AVB series and a trivial alsa driver to connect my Motu via AVB for a single i/o stream. So i discovered how totally unsuitable the mmapped ringbuffer model of alsa is for the above types of interfaces. Is there any intention to change this ? I've read that this has already been discussed during miniconf 2017 (USB) and 2018 (AVB). Any results?

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[alsa-devel] [PATCH] ASoC: da7213: fix DAI_CLK_EN register bit overwrite
by Logesh Kolandavel 26 Apr '19

26 Apr '19

From: Logesh <logesh.kolandavel(a)timesys.com> If the da7213 codec is configured as Master with the DAPM power down delay time set, 'snd_soc_component_write' function overwrites the DAI_CLK_EN bit of DAI_CLK_MODE register which leads to audio play only once until it re-initialize after codec power up. Signed-off-by: Logesh <logesh.kolandavel(a)timesys.com> --- sound/soc/codecs/da7213.c | 4 +++- sound/soc/codecs/da7213.h | 2 ++ 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 92d006a5283e..306e01684bda 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1305,7 +1305,9 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* By default only 64 BCLK per WCLK is supported */ dai_clk_mode |= DA7213_DAI_BCLKS_PER_WCLK_64; - snd_soc_component_write(component, DA7213_DAI_CLK_MODE, dai_clk_mode); + snd_soc_component_update_bits(component, DA7213_DAI_CLK_MODE, + DA7213_DAI_CLK_POL_MASK | DA7213_DAI_WCLK_POL_MASK, + dai_clk_mode); snd_soc_component_update_bits(component, DA7213_DAI_CTRL, DA7213_DAI_FORMAT_MASK, dai_ctrl); snd_soc_component_write(component, DA7213_DAI_OFFSET, dai_offset); diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 5a78dba1dcb5..9d31efc3cfe5 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -181,7 +181,9 @@ #define DA7213_DAI_BCLKS_PER_WCLK_256 (0x3 << 0) #define DA7213_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) #define DA7213_DAI_CLK_POL_INV (0x1 << 2) +#define DA7213_DAI_CLK_POL_MASK (0x1 << 2) #define DA7213_DAI_WCLK_POL_INV (0x1 << 3) +#define DA7213_DAI_WCLK_POL_MASK (0x1 << 3) #define DA7213_DAI_CLK_EN_MASK (0x1 << 7) /* DA7213_DAI_CTRL = 0x29 */ -- 2.21.0

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[alsa-devel] [PATCH v3 0/3] Add runtime PM for SAI digital audio interface
by Daniel Baluta 26 Apr '19

26 Apr '19

First patch fixes a bug by correctly setting is_slave_mode, then second patch adds support for runtime PM and finally 3rd patch moves clock handling from startup/shtudown function to runtime PM handlers. Changes since v2: (after Viorel's comments) - no need to check for is_slave_mode when enabling/disabling the clocks because sai->mclk_streams is only set when SAI is in master mode. Changes since v1: (after Nicolin's comments) - added patch 1 - added fsl_sai_remove in order to call pm_runtime_disable - only disable/enable mclk when SAI in master mode. Daniel Baluta (2): ASoC: fsl_sai: Update is_slave_mode with correct value ASoC: fsl_sai: Add support for runtime pm Shengjiu Wang (1): ASoC: fsl_sai: Move clock operation to PM runtime sound/soc/fsl/fsl_sai.c | 78 +++++++++++++++++++++++++++++++++-------- 1 file changed, 63 insertions(+), 15 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH] ASoC: mediatek: btcvsd fix irq mask
by KaiChieh Chuang 26 Apr '19

26 Apr '19

disable = 1 enable = 0 prevent AP wake up, when not AP BTSCO scenario Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- sound/soc/mediatek/common/mtk-btcvsd.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index 6784e35..b918f79 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -194,13 +194,13 @@ struct mtk_btcvsd_snd_time_buffer_info { static void mtk_btcvsd_snd_irq_enable(struct mtk_btcvsd_snd *bt) { regmap_update_bits(bt->infra, bt->infra_misc_offset, - bt->conn_bt_cvsd_mask, bt->conn_bt_cvsd_mask); + bt->conn_bt_cvsd_mask, 0); } static void mtk_btcvsd_snd_irq_disable(struct mtk_btcvsd_snd *bt) { regmap_update_bits(bt->infra, bt->infra_misc_offset, - bt->conn_bt_cvsd_mask, 0); + bt->conn_bt_cvsd_mask, bt->conn_bt_cvsd_mask); } static void mtk_btcvsd_snd_set_state(struct mtk_btcvsd_snd *bt, -- 1.9.1

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[alsa-devel] [PATCH v2 1/2] ASoC: simple-card-utils: add asoc_simple_parse_pin_switches()
by Paul Cercueil 26 Apr '19

26 Apr '19

This function is a helper that permits to create pin switch controls for a list of widgets whose names are listed in the PREFIX "pin-switches" devicetree property. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> --- Notes: v2: - Rebase on top of for-next - Rename to asoc_simple_parse_pin_switches() include/sound/simple_card_utils.h | 2 + sound/soc/generic/simple-card-utils.c | 57 +++++++++++++++++++++++++++ 2 files changed, 59 insertions(+) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 67dc3ee6ed01..3429888347e7 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -133,6 +133,8 @@ int asoc_simple_parse_routing(struct snd_soc_card *card, char *prefix); int asoc_simple_parse_widgets(struct snd_soc_card *card, char *prefix); +int asoc_simple_parse_pin_switches(struct snd_soc_card *card, + char *prefix); int asoc_simple_init_jack(struct snd_soc_card *card, struct asoc_simple_jack *sjack, diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index db1458a19985..f4c6375d11c7 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -430,6 +430,63 @@ int asoc_simple_parse_widgets(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_parse_widgets); +int asoc_simple_parse_pin_switches(struct snd_soc_card *card, + char *prefix) +{ + const unsigned int nb_controls_max = 16; + const char **strings, *control_name; + struct snd_kcontrol_new *controls; + struct device *dev = card->dev; + unsigned int i, nb_controls; + char prop[128]; + int ret; + + if (!prefix) + prefix = ""; + + snprintf(prop, sizeof(prop), "%s%s", prefix, "pin-switches"); + + if (!of_property_read_bool(dev->of_node, prop)) + return 0; + + strings = devm_kcalloc(dev, nb_controls_max, + sizeof(*strings), GFP_KERNEL); + if (!strings) + return -ENOMEM; + + ret = of_property_read_string_array(dev->of_node, prop, + strings, nb_controls_max); + if (ret < 0) + return ret; + + nb_controls = (unsigned int)ret; + + controls = devm_kcalloc(dev, nb_controls, + sizeof(*controls), GFP_KERNEL); + if (!controls) + return -ENOMEM; + + for (i = 0; i < nb_controls; i++) { + control_name = devm_kasprintf(dev, GFP_KERNEL, + "%s Switch", strings[i]); + if (!control_name) + return -ENOMEM; + + controls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + controls[i].name = control_name; + controls[i].info = snd_soc_dapm_info_pin_switch; + controls[i].get = snd_soc_dapm_get_pin_switch; + controls[i].put = snd_soc_dapm_put_pin_switch; + controls[i].private_value = (unsigned long)strings[i]; + } + + card->controls = controls; + card->num_controls = nb_controls; + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_parse_pin_switches); + int asoc_simple_init_jack(struct snd_soc_card *card, struct asoc_simple_jack *sjack, int is_hp, char *prefix) -- 2.21.0.593.g511ec345e18

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[alsa-devel] [PATCH] ASoC: fsl_sai: Add missing return 0 in remove()
by Nicolin Chen 26 Apr '19

26 Apr '19

Build warning being reported: sound/soc/fsl/fsl_sai.c: In function 'fsl_sai_remove': sound/soc/fsl/fsl_sai.c:921:1: warning: no return statement in function returning non-void [-Wreturn-type] So this patch just adds a "return 0" to fix it. Fixes: 812ad463e089 ("ASoC: fsl_sai: Add support for runtime pm") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_sai.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 26c27dc..8593269 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -918,6 +918,8 @@ static int fsl_sai_probe(struct platform_device *pdev) static int fsl_sai_remove(struct platform_device *pdev) { pm_runtime_disable(&pdev->dev); + + return 0; } static const struct of_device_id fsl_sai_ids[] = { -- 2.7.4

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[alsa-devel] Applied "ASoC: stm32: sai: simplify dai driver initialisation" to the asoc tree
by Mark Brown 26 Apr '19

26 Apr '19

The patch ASoC: stm32: sai: simplify dai driver initialisation 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 8f8a54884dfd3d756693b7f174a06397bf19d8a5 Mon Sep 17 00:00:00 2001 From: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Date: Tue, 23 Apr 2019 17:58:08 +0200 Subject: [PATCH] ASoC: stm32: sai: simplify dai driver initialisation Suppress the useless dynamic allocation of the dai driver structure. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 43 ++++++++--------------------------- 1 file changed, 9 insertions(+), 34 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3dd54bc54fa1..e3b021c9e8d0 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -109,7 +109,7 @@ struct stm32_sai_sub_data { struct regmap *regmap; const struct regmap_config *regmap_config; struct snd_dmaengine_dai_dma_data dma_params; - struct snd_soc_dai_driver *cpu_dai_drv; + struct snd_soc_dai_driver cpu_dai_drv; struct snd_soc_dai *cpu_dai; struct snd_pcm_substream *substream; struct stm32_sai_data *pdata; @@ -1204,8 +1204,7 @@ static const struct snd_pcm_hardware stm32_sai_pcm_hw = { .periods_max = 8, }; -static struct snd_soc_dai_driver stm32_sai_playback_dai[] = { -{ +static struct snd_soc_dai_driver stm32_sai_playback_dai = { .probe = stm32_sai_dai_probe, .pcm_new = stm32_sai_pcm_new, .id = 1, /* avoid call to fmt_single_name() */ @@ -1222,11 +1221,9 @@ static struct snd_soc_dai_driver stm32_sai_playback_dai[] = { SNDRV_PCM_FMTBIT_S32_LE, }, .ops = &stm32_sai_pcm_dai_ops, - } }; -static struct snd_soc_dai_driver stm32_sai_capture_dai[] = { -{ +static struct snd_soc_dai_driver stm32_sai_capture_dai = { .probe = stm32_sai_dai_probe, .id = 1, /* avoid call to fmt_single_name() */ .capture = { @@ -1242,7 +1239,6 @@ static struct snd_soc_dai_driver stm32_sai_capture_dai[] = { SNDRV_PCM_FMTBIT_S32_LE, }, .ops = &stm32_sai_pcm_dai_ops, - } }; static const struct snd_dmaengine_pcm_config stm32_sai_pcm_config = { @@ -1411,29 +1407,6 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, return 0; } -static int stm32_sai_sub_dais_init(struct platform_device *pdev, - struct stm32_sai_sub_data *sai) -{ - sai->cpu_dai_drv = devm_kzalloc(&pdev->dev, - sizeof(struct snd_soc_dai_driver), - GFP_KERNEL); - if (!sai->cpu_dai_drv) - return -ENOMEM; - - if (STM_SAI_IS_PLAYBACK(sai)) { - memcpy(sai->cpu_dai_drv, &stm32_sai_playback_dai, - sizeof(stm32_sai_playback_dai)); - sai->cpu_dai_drv->playback.stream_name = sai->cpu_dai_drv->name; - } else { - memcpy(sai->cpu_dai_drv, &stm32_sai_capture_dai, - sizeof(stm32_sai_capture_dai)); - sai->cpu_dai_drv->capture.stream_name = sai->cpu_dai_drv->name; - } - sai->cpu_dai_drv->name = dev_name(&pdev->dev); - - return 0; -} - static int stm32_sai_sub_probe(struct platform_device *pdev) { struct stm32_sai_sub_data *sai; @@ -1465,9 +1438,11 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) if (ret) return ret; - ret = stm32_sai_sub_dais_init(pdev, sai); - if (ret) - return ret; + if (STM_SAI_IS_PLAYBACK(sai)) + sai->cpu_dai_drv = stm32_sai_playback_dai; + else + sai->cpu_dai_drv = stm32_sai_capture_dai; + sai->cpu_dai_drv.name = dev_name(&pdev->dev); ret = devm_request_irq(&pdev->dev, sai->pdata->irq, stm32_sai_isr, IRQF_SHARED, dev_name(&pdev->dev), sai); @@ -1477,7 +1452,7 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) } ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component, - sai->cpu_dai_drv, 1); + &sai->cpu_dai_drv, 1); if (ret) return ret; -- 2.20.1

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