August 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RESENT PATCH] ASoC: hdmi-codec: Use different name for playback streams
by Jeffy Chen 30 Aug '17

30 Aug '17

Currently the hdmi i2s playback stream and hdmi spdif playback stream are using the same name. So when they are enabled at the same time, kernel will print this warning: [ 2.201835] hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: Failed to create Playback debugfs file Assign different names to them to avoid that. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- sound/soc/codecs/hdmi-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 509ab51..a2af440 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -696,7 +696,7 @@ static struct snd_soc_dai_driver hdmi_i2s_dai = { .name = "i2s-hifi", .id = DAI_ID_I2S, .playback = { - .stream_name = "Playback", + .stream_name = "I2S Playback", .channels_min = 2, .channels_max = 8, .rates = HDMI_RATES, @@ -711,7 +711,7 @@ static const struct snd_soc_dai_driver hdmi_spdif_dai = { .name = "spdif-hifi", .id = DAI_ID_SPDIF, .playback = { - .stream_name = "Playback", + .stream_name = "SPDIF Playback", .channels_min = 2, .channels_max = 2, .rates = HDMI_RATES, -- 2.1.4

2 1

[alsa-devel] USB dB data
by Jack Bates 30 Aug '17

30 Aug '17

I confirmed with Wireshark that my USB sound card responds to GET_MIN with 0x0000 and GET_MAX with 0x6300 .. The audio device class definition [1] says the units are 1/256 dB (section 5.2.2.2.3: Mixer Control) ... So by my calculation, the min and max are 0 and 99 dB (0x6300 / 256 == 99) ... Why then does alsactl print: > range '0 - 99' > dbmin 0 > dbmax 38 (/usr/sbin/aslactl -f - store) I originally asked about this on the pulseaudio-discuss list [2] because it uses "dB data" [3]. The "0 - 99" makes sense -- it lines up with my calculation ... Where does the 38 come from? (I expect it's obvious, I just haven't found the explanation?) Background: I just got a Bose SoundLink Revolve ... When I connect it with USB, the volume is silent from 0 to 99% and deafening at 100% (using PulseAudio). Directly adjusting the volume with alsamixer works properly: It increases smoothly from 0 to 100% -- changes that are reflected in the PulseAudio volume as fine adjustments in between 99 and 100% (64587 to 65536 PulseAudio volume). From what I've read there's a common problem where especially USB devices report the wrong dB data (don't follow the spec [4]) -- this has previously been handled by patching the USB audio driver [5] ... However I haven't figured out how my device deviates from the spec (0x0000 to 0x6300 or 0 to 99 dB seem plausible) -- my understanding is incomplete: I haven't figured out where the value 38 comes from ... [1] http://www.usb.org/developers/docs/devclass_docs/audio10.pdf [2] https://lists.freedesktop.org/archives/pulseaudio-discuss/2017-August/02860… [3] http://0pointer.de/blog/projects/decibel-data [4] https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/637066 [5] https://bugs.freedesktop.org/show_bug.cgi?id=93193#c14

1 0

[alsa-devel] Unable to play MP3 using TinyCompress and bytcr-rt5640
by Lambdadroid 30 Aug '17

30 Aug '17

Hi, I'm attempting to use TinyCompress to play a MP3 file on a Baytrail tablet (using bytcr-rt5640 on Linux 4.13-rc7) with ALSA's Compress-Offload API. After finding the right quirks and setting up the UCM files normal audio output is working, but I'm unable to play a MP3 file using the `cplay` tool included with TinyCompress. Note that `cplay` appears to be broken as of commit 50c2b03 "cplay: check the values of fread". fread returns the number of items that were read but the commit assumes it returns the number of bytes. I've simply changed the check to `read != 1` to fix it. When I play a MP3 file after fixing that I get a cryptic SST firmware error in dmesg and there is nothing playing on the speakers: [ 29.080173] intel_sst_acpi 80860F28:00: FW Version 01.0c.00.01 [ 39.575366] intel_sst_acpi 80860F28:00: FW sent async error msg: [ 39.578059] (null)00040009 00000000 [ 39.578068] (null)00040009 00000000 [ 53.872092] intel_sst_acpi 80860F28:00: FW sent async error msg: [ 53.874999] (null)0002000b 00000000 [ 53.875006] (null)0002000b 00000000 I've tried various different MP3 files with different bit rates but this happens for all of them. Is this an indication that it's not supported on my device or is something else wrong? Other related dmesg messages (but these look all fine to me): [ 14.085226] intel_sst_acpi 80860F28:00: Detected Baytrail-CR platform [ 14.085422] intel_sst_acpi 80860F28:00: LPE base: 0x51000000 size:0x200000 [ 14.085425] intel_sst_acpi 80860F28:00: IRAM base: 0x510c0000 [ 14.085475] intel_sst_acpi 80860F28:00: DRAM base: 0x51100000 [ 14.085484] intel_sst_acpi 80860F28:00: SHIM base: 0x51140000 [ 14.085492] intel_sst_acpi 80860F28:00: Mailbox base: 0x51144000 [ 14.085499] intel_sst_acpi 80860F28:00: DDR base: 0x20000000 [ 14.085730] intel_sst_acpi 80860F28:00: Got drv data max stream 25 [ 14.976784] bytcr_rt5640 bytcr_rt5640: Overriding quirk 0x640002 => 0x510000 [ 14.976788] bytcr_rt5640 bytcr_rt5640: quirk DMIC1_MAP enabled [ 14.976791] bytcr_rt5640 bytcr_rt5640: quirk DMIC enabled [ 14.976793] bytcr_rt5640 bytcr_rt5640: quirk SSP0_AIF1 enabled [ 14.976795] bytcr_rt5640 bytcr_rt5640: quirk MCLK_EN enabled [ 14.980650] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> media-cpu-dai mapping ok [ 14.980710] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> deepbuffer-cpu-dai mapping ok [ 14.980740] compress asoc: snd-soc-dummy-dai <-> compress-cpu-dai mapping ok [ 14.981233] bytcr_rt5640 bytcr_rt5640: rt5640-aif1 <-> ssp0-port mapping ok The full dmesg and alsa-info.txt is quite big so I've uploaded it to gist. I hope that's okay. https://gist.github.com/lambdadroid/d46675e54720292ed7718c74247753b5 Thanks in advance! Lambdadroid

2 3

[alsa-devel] [PATCH 1/2] ALSA: Get rid of card power_lock
by Takashi Iwai 30 Aug '17

30 Aug '17

Currently we're taking power_lock at each card component for assuring the power-up sequence, but it doesn't help anything in the implementation at the moment: it just serializes unnecessarily the callers, but it doesn't protect about the power state change itself. It used to have some usefulness in the early days where we managed the PM manually. But now the suspend/resume core procedure is beyond our hands, and power_lock lost its meaning. This patch drops the power_lock from allover the places. There shouldn't be any issues by this change, as it's no helper regarding the power state change. Rather we'll get better performance by removing the serialization; which is the only slight concern of any behavior change, but it can't be a showstopper, after all. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/core.h | 13 ---------- sound/core/control.c | 58 ++++++++++++++++++++++++--------------------- sound/core/control_compat.c | 34 ++++++++++++-------------- sound/core/init.c | 5 ---- sound/core/pcm_native.c | 30 ++++++----------------- sound/soc/soc-core.c | 2 -- 6 files changed, 53 insertions(+), 89 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 357f36b5ee80..4104a9d1001f 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -136,7 +136,6 @@ struct snd_card { #ifdef CONFIG_PM unsigned int power_state; /* power state */ - struct mutex power_lock; /* power lock */ wait_queue_head_t power_sleep; #endif @@ -149,16 +148,6 @@ struct snd_card { #define dev_to_snd_card(p) container_of(p, struct snd_card, card_dev) #ifdef CONFIG_PM -static inline void snd_power_lock(struct snd_card *card) -{ - mutex_lock(&card->power_lock); -} - -static inline void snd_power_unlock(struct snd_card *card) -{ - mutex_unlock(&card->power_lock); -} - static inline unsigned int snd_power_get_state(struct snd_card *card) { return card->power_state; @@ -175,8 +164,6 @@ int snd_power_wait(struct snd_card *card, unsigned int power_state); #else /* ! CONFIG_PM */ -#define snd_power_lock(card) do { (void)(card); } while (0) -#define snd_power_unlock(card) do { (void)(card); } while (0) static inline int snd_power_wait(struct snd_card *card, unsigned int state) { return 0; } #define snd_power_get_state(card) ({ (void)(card); SNDRV_CTL_POWER_D0; }) #define snd_power_change_state(card, state) do { (void)(card); } while (0) diff --git a/sound/core/control.c b/sound/core/control.c index 51d4b4ad3e1d..56b3e2d49c82 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -864,14 +864,14 @@ static int snd_ctl_elem_info_user(struct snd_ctl_file *ctl, if (copy_from_user(&info, _info, sizeof(info))) return -EFAULT; - snd_power_lock(ctl->card); result = snd_power_wait(ctl->card, SNDRV_CTL_POWER_D0); - if (result >= 0) - result = snd_ctl_elem_info(ctl, &info); - snd_power_unlock(ctl->card); - if (result >= 0) - if (copy_to_user(_info, &info, sizeof(info))) - return -EFAULT; + if (result < 0) + return result; + result = snd_ctl_elem_info(ctl, &info); + if (result < 0) + return result; + if (copy_to_user(_info, &info, sizeof(info))) + return -EFAULT; return result; } @@ -905,17 +905,19 @@ static int snd_ctl_elem_read_user(struct snd_card *card, if (IS_ERR(control)) return PTR_ERR(control); - snd_power_lock(card); result = snd_power_wait(card, SNDRV_CTL_POWER_D0); - if (result >= 0) { - down_read(&card->controls_rwsem); - result = snd_ctl_elem_read(card, control); - up_read(&card->controls_rwsem); - } - snd_power_unlock(card); - if (result >= 0) - if (copy_to_user(_control, control, sizeof(*control))) - result = -EFAULT; + if (result < 0) + goto error; + + down_read(&card->controls_rwsem); + result = snd_ctl_elem_read(card, control); + up_read(&card->controls_rwsem); + if (result < 0) + goto error; + + if (copy_to_user(_control, control, sizeof(*control))) + result = -EFAULT; + error: kfree(control); return result; } @@ -964,17 +966,19 @@ static int snd_ctl_elem_write_user(struct snd_ctl_file *file, return PTR_ERR(control); card = file->card; - snd_power_lock(card); result = snd_power_wait(card, SNDRV_CTL_POWER_D0); - if (result >= 0) { - down_write(&card->controls_rwsem); - result = snd_ctl_elem_write(card, file, control); - up_write(&card->controls_rwsem); - } - snd_power_unlock(card); - if (result >= 0) - if (copy_to_user(_control, control, sizeof(*control))) - result = -EFAULT; + if (result < 0) + goto error; + + down_write(&card->controls_rwsem); + result = snd_ctl_elem_write(card, file, control); + up_write(&card->controls_rwsem); + if (result < 0) + goto error; + + if (copy_to_user(_control, control, sizeof(*control))) + result = -EFAULT; + error: kfree(control); return result; } diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 1fa70766ffab..a848836a5de0 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -111,12 +111,10 @@ static int snd_ctl_elem_info_compat(struct snd_ctl_file *ctl, if (get_user(data->value.enumerated.item, &data32->value.enumerated.item)) goto error; - snd_power_lock(ctl->card); err = snd_power_wait(ctl->card, SNDRV_CTL_POWER_D0); - if (err >= 0) - err = snd_ctl_elem_info(ctl, data); - snd_power_unlock(ctl->card); - + if (err < 0) + goto error; + err = snd_ctl_elem_info(ctl, data); if (err < 0) goto error; /* restore info to 32bit */ @@ -315,14 +313,13 @@ static int ctl_elem_read_user(struct snd_card *card, if (err < 0) goto error; - snd_power_lock(card); err = snd_power_wait(card, SNDRV_CTL_POWER_D0); - if (err >= 0) - err = snd_ctl_elem_read(card, data); - snd_power_unlock(card); - if (err >= 0) - err = copy_ctl_value_to_user(userdata, valuep, data, - type, count); + if (err < 0) + goto error; + err = snd_ctl_elem_read(card, data); + if (err < 0) + goto error; + err = copy_ctl_value_to_user(userdata, valuep, data, type, count); error: kfree(data); return err; @@ -344,14 +341,13 @@ static int ctl_elem_write_user(struct snd_ctl_file *file, if (err < 0) goto error; - snd_power_lock(card); err = snd_power_wait(card, SNDRV_CTL_POWER_D0); - if (err >= 0) - err = snd_ctl_elem_write(card, file, data); - snd_power_unlock(card); - if (err >= 0) - err = copy_ctl_value_to_user(userdata, valuep, data, - type, count); + if (err < 0) + goto error; + err = snd_ctl_elem_write(card, file, data); + if (err < 0) + goto error; + err = copy_ctl_value_to_user(userdata, valuep, data, type, count); error: kfree(data); return err; diff --git a/sound/core/init.c b/sound/core/init.c index 6e219dc23f96..32ebe2f6bc59 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -253,7 +253,6 @@ int snd_card_new(struct device *parent, int idx, const char *xid, spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); #ifdef CONFIG_PM - mutex_init(&card->power_lock); init_waitqueue_head(&card->power_sleep); #endif @@ -978,8 +977,6 @@ EXPORT_SYMBOL(snd_card_file_remove); * Waits until the power-state is changed. * * Return: Zero if successful, or a negative error code. - * - * Note: the power lock must be active before call. */ int snd_power_wait(struct snd_card *card, unsigned int power_state) { @@ -999,9 +996,7 @@ int snd_power_wait(struct snd_card *card, unsigned int power_state) if (snd_power_get_state(card) == power_state) break; set_current_state(TASK_UNINTERRUPTIBLE); - snd_power_unlock(card); schedule_timeout(30 * HZ); - snd_power_lock(card); } remove_wait_queue(&card->power_sleep, &wait); return result; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index cf0433f80067..621142ea9ec6 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1830,7 +1830,6 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, add_wait_queue(&to_check->sleep, &wait); snd_pcm_stream_unlock_irq(substream); up_read(&snd_pcm_link_rwsem); - snd_power_unlock(card); if (runtime->no_period_wakeup) tout = MAX_SCHEDULE_TIMEOUT; else { @@ -1842,7 +1841,6 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, tout = msecs_to_jiffies(tout * 1000); } tout = schedule_timeout_interruptible(tout); - snd_power_lock(card); down_read(&snd_pcm_link_rwsem); snd_pcm_stream_lock_irq(substream); remove_wait_queue(&to_check->sleep, &wait); @@ -2764,11 +2762,16 @@ static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg) return 0; } -static int snd_pcm_common_ioctl(struct file *file, +static int snd_pcm_common_ioctl1(struct file *file, struct snd_pcm_substream *substream, unsigned int cmd, void __user *arg) { struct snd_pcm_file *pcm_file = file->private_data; + int res; + + res = snd_power_wait(substream->pcm->card, SNDRV_CTL_POWER_D0); + if (res < 0) + return res; switch (cmd) { case SNDRV_PCM_IOCTL_PVERSION: @@ -2846,21 +2849,6 @@ static int snd_pcm_common_ioctl(struct file *file, return -ENOTTY; } -static int snd_pcm_common_ioctl1(struct file *file, - struct snd_pcm_substream *substream, - unsigned int cmd, void __user *arg) -{ - struct snd_card *card = substream->pcm->card; - int res; - - snd_power_lock(card); - res = snd_power_wait(card, SNDRV_CTL_POWER_D0); - if (res >= 0) - res = snd_pcm_common_ioctl(file, substream, cmd, arg); - snd_power_unlock(card); - return res; -} - static int snd_pcm_playback_ioctl1(struct file *file, struct snd_pcm_substream *substream, unsigned int cmd, void __user *arg) @@ -3064,7 +3052,6 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, { snd_pcm_uframes_t *frames = arg; snd_pcm_sframes_t result; - int err; switch (cmd) { case SNDRV_PCM_IOCTL_FORWARD: @@ -3084,10 +3071,7 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, case SNDRV_PCM_IOCTL_START: return snd_pcm_start_lock_irq(substream); case SNDRV_PCM_IOCTL_DRAIN: - snd_power_lock(substream->pcm->card); - err = snd_pcm_drain(substream, NULL); - snd_power_unlock(substream->pcm->card); - return err; + return snd_pcm_drain(substream, NULL); case SNDRV_PCM_IOCTL_DROP: return snd_pcm_drop(substream); case SNDRV_PCM_IOCTL_DELAY: diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 13c875e2392a..62c11e26ce5c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -653,9 +653,7 @@ int snd_soc_suspend(struct device *dev) /* Due to the resume being scheduled into a workqueue we could * suspend before that's finished - wait for it to complete. */ - snd_power_lock(card->snd_card); snd_power_wait(card->snd_card, SNDRV_CTL_POWER_D0); - snd_power_unlock(card->snd_card); /* we're going to block userspace touching us until resume completes */ snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot); -- 2.14.1

1 1

[alsa-devel] [asoc:topic/component 7/8] include/sound/soc.h:811:8: error: duplicate member 'set_pll'
by kbuild test robot 30 Aug '17

30 Aug '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/component head: 880f02224c624fa2d6e4f8b2d37410fdc98b35fe commit: 47bdbaa2c9a6dd7b112fb403b05dd9293b35a41c [7/8] ASoC: add Component level set_pll config: xtensa-allmodconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 4.9.0 reproduce: wget https://raw.githubusercontent.com/01org/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 47bdbaa2c9a6dd7b112fb403b05dd9293b35a41c # save the attached .config to linux build tree make.cross ARCH=xtensa All errors (new ones prefixed by >>): In file included from sound/soc/adi/axi-i2s.c:20:0: include/sound/soc.h:809:8: error: duplicate member 'set_sysclk' int (*set_sysclk)(struct snd_soc_component *component, ^ >> include/sound/soc.h:811:8: error: duplicate member 'set_pll' int (*set_pll)(struct snd_soc_component *component, int pll_id, ^ include/sound/soc.h:891:8: error: duplicate member 'set_sysclk' int (*set_sysclk)(struct snd_soc_component *component, ^ include/sound/soc.h:893:8: error: duplicate member 'set_pll' int (*set_pll)(struct snd_soc_component *component, int pll_id, ^ vim +/set_pll +811 include/sound/soc.h 780 781 /* component interface */ 782 struct snd_soc_component_driver { 783 const char *name; 784 785 /* Default control and setup, added after probe() is run */ 786 const struct snd_kcontrol_new *controls; 787 unsigned int num_controls; 788 const struct snd_soc_dapm_widget *dapm_widgets; 789 unsigned int num_dapm_widgets; 790 const struct snd_soc_dapm_route *dapm_routes; 791 unsigned int num_dapm_routes; 792 793 int (*probe)(struct snd_soc_component *); 794 void (*remove)(struct snd_soc_component *); 795 int (*suspend)(struct snd_soc_component *); 796 int (*resume)(struct snd_soc_component *); 797 798 /* pcm creation and destruction */ 799 int (*pcm_new)(struct snd_soc_pcm_runtime *); 800 void (*pcm_free)(struct snd_pcm *); 801 802 /* component wide operations */ 803 int (*set_sysclk)(struct snd_soc_component *component, 804 int clk_id, int source, unsigned int freq, int dir); 805 int (*set_pll)(struct snd_soc_component *component, int pll_id, 806 int source, unsigned int freq_in, unsigned int freq_out); 807 808 /* component wide operations */ > 809 int (*set_sysclk)(struct snd_soc_component *component, 810 int clk_id, int source, unsigned int freq, int dir); > 811 int (*set_pll)(struct snd_soc_component *component, int pll_id, 812 int source, unsigned int freq_in, unsigned int freq_out); 813 814 /* DT */ 815 int (*of_xlate_dai_name)(struct snd_soc_component *component, 816 struct of_phandle_args *args, 817 const char **dai_name); 818 int (*of_xlate_dai_id)(struct snd_soc_component *comment, 819 struct device_node *endpoint); 820 void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type, 821 int subseq); 822 int (*stream_event)(struct snd_soc_component *, int event); 823 824 /* probe ordering - for components with runtime dependencies */ 825 int probe_order; 826 int remove_order; 827 }; 828 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [asoc:topic/component 6/8] include/sound/soc.h:809:8: error: duplicate member 'set_sysclk'
by kbuild test robot 30 Aug '17

30 Aug '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/component head: 880f02224c624fa2d6e4f8b2d37410fdc98b35fe commit: 9cfeb53411be7c08d7dd19bc4bf0599a6d211390 [6/8] ASoC: add Component level set_sysclk config: xtensa-allmodconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 4.9.0 reproduce: wget https://raw.githubusercontent.com/01org/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 9cfeb53411be7c08d7dd19bc4bf0599a6d211390 # save the attached .config to linux build tree make.cross ARCH=xtensa All errors (new ones prefixed by >>): In file included from sound//soc/adi/axi-i2s.c:20:0: >> include/sound/soc.h:809:8: error: duplicate member 'set_sysclk' int (*set_sysclk)(struct snd_soc_component *component, ^ include/sound/soc.h:889:8: error: duplicate member 'set_sysclk' int (*set_sysclk)(struct snd_soc_component *component, ^ vim +/set_sysclk +809 include/sound/soc.h 780 781 /* component interface */ 782 struct snd_soc_component_driver { 783 const char *name; 784 785 /* Default control and setup, added after probe() is run */ 786 const struct snd_kcontrol_new *controls; 787 unsigned int num_controls; 788 const struct snd_soc_dapm_widget *dapm_widgets; 789 unsigned int num_dapm_widgets; 790 const struct snd_soc_dapm_route *dapm_routes; 791 unsigned int num_dapm_routes; 792 793 int (*probe)(struct snd_soc_component *); 794 void (*remove)(struct snd_soc_component *); 795 int (*suspend)(struct snd_soc_component *); 796 int (*resume)(struct snd_soc_component *); 797 798 /* pcm creation and destruction */ 799 int (*pcm_new)(struct snd_soc_pcm_runtime *); 800 void (*pcm_free)(struct snd_pcm *); 801 802 /* component wide operations */ 803 int (*set_sysclk)(struct snd_soc_component *component, 804 int clk_id, int source, unsigned int freq, int dir); 805 int (*set_pll)(struct snd_soc_component *component, int pll_id, 806 int source, unsigned int freq_in, unsigned int freq_out); 807 808 /* component wide operations */ > 809 int (*set_sysclk)(struct snd_soc_component *component, 810 int clk_id, int source, unsigned int freq, int dir); 811 812 /* DT */ 813 int (*of_xlate_dai_name)(struct snd_soc_component *component, 814 struct of_phandle_args *args, 815 const char **dai_name); 816 int (*of_xlate_dai_id)(struct snd_soc_component *comment, 817 struct device_node *endpoint); 818 void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type, 819 int subseq); 820 int (*stream_event)(struct snd_soc_component *, int event); 821 822 /* probe ordering - for components with runtime dependencies */ 823 int probe_order; 824 int remove_order; 825 }; 826 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: simple_card_utils: fix fallback when "label" property isn't present
by Lucas Stach 30 Aug '17

30 Aug '17

snd_soc_of_parse_card_name() doesn't return an error if the requested property isn't present, but silently fails to fill the card name. This can not be changed, as it is a backwards compatibility measure itself. We can not rely on the return value of this function alone, but must check if the card name has been filled sucessfully when deciding to skip the fallback path, which is in place for existing users. Fixes: dedfaa1eedc7 (ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name) Signed-off-by: Lucas Stach <l.stach(a)pengutronix.de> --- This may be applied instead of the revert I just sent. After examining the function more closely, the fix seems easy enough to go into 4.13 final. --- sound/soc/generic/simple-card-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 7d7ab4aee42e..d72f7d58102f 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -132,7 +132,7 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, /* Parse the card name from DT */ ret = snd_soc_of_parse_card_name(card, "label"); - if (ret < 0) { + if (ret < 0 || !card->name) { char prop[128]; snprintf(prop, sizeof(prop), "%sname", prefix); -- 2.11.0

3 2

[alsa-devel] Applied "ASoC: add Component level set_sysclk" to the asoc tree
by Mark Brown 30 Aug '17

30 Aug '17

The patch ASoC: add Component level set_sysclk has been applied to the asoc tree at git://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 9cfeb53411be7c08d7dd19bc4bf0599a6d211390 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 24 Aug 2017 00:57:35 +0000 Subject: [PATCH] ASoC: add Component level set_sysclk In current ALSA SoC, Codec only has set_sysclk feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 29 +++++++++++++++++++++++++++++ sound/soc/soc-core.c | 45 ++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 69 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2bcb38045dc1..e7c476febf82 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -799,6 +799,16 @@ struct snd_soc_component_driver { int (*pcm_new)(struct snd_soc_pcm_runtime *); void (*pcm_free)(struct snd_pcm *); + /* component wide operations */ + int (*set_sysclk)(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir); + int (*set_pll)(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out); + + /* component wide operations */ + int (*set_sysclk)(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir); + /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, struct of_phandle_args *args, @@ -871,6 +881,14 @@ struct snd_soc_component { int (*pcm_new)(struct snd_soc_pcm_runtime *); void (*pcm_free)(struct snd_pcm *); + int (*set_sysclk)(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir); + int (*set_pll)(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out); + + int (*set_sysclk)(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir); + /* machine specific init */ int (*init)(struct snd_soc_component *component); @@ -1465,6 +1483,17 @@ void snd_soc_component_async_complete(struct snd_soc_component *component); int snd_soc_component_test_bits(struct snd_soc_component *component, unsigned int reg, unsigned int mask, unsigned int value); +/* component wide operations */ +int snd_soc_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir); +int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out); + +/* component wide operations */ +int snd_soc_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir); + #ifdef CONFIG_REGMAP void snd_soc_component_init_regmap(struct snd_soc_component *component, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b8f96a41b594..046ab9189b36 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2590,11 +2590,9 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, { if (dai->driver && dai->driver->ops->set_sysclk) return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); - else if (dai->codec && dai->codec->driver->set_sysclk) - return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0, - freq, dir); - else - return -ENOTSUPP; + + return snd_soc_component_set_sysclk(dai->component, clk_id, 0, + freq, dir); } EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); @@ -2619,6 +2617,32 @@ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, } EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); +/** + * snd_soc_component_set_sysclk - configure COMPONENT system or master clock. + * @component: COMPONENT + * @clk_id: DAI specific clock ID + * @source: Source for the clock + * @freq: new clock frequency in Hz + * @dir: new clock direction - input/output. + * + * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. + */ +int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id, + int source, unsigned int freq, int dir) +{ + /* will be removed */ + if (component->set_sysclk) + return component->set_sysclk(component, clk_id, source, + freq, dir); + + if (component->driver->set_sysclk) + return component->driver->set_sysclk(component, clk_id, source, + freq, dir); + + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_component_set_sysclk); + /** * snd_soc_dai_set_clkdiv - configure DAI clock dividers. * @dai: DAI @@ -3176,6 +3200,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->resume = component->driver->resume; component->pcm_new = component->driver->pcm_new; component->pcm_free = component->driver->pcm_free; + component->set_sysclk = component->driver->set_sysclk; dapm = &component->dapm; dapm->dev = dev; @@ -3582,6 +3607,14 @@ static int snd_soc_codec_drv_read(struct snd_soc_component *component, return 0; } +static int snd_soc_codec_set_sysclk_(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + return snd_soc_codec_set_sysclk(codec, clk_id, source, freq, dir); +} + static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { @@ -3633,6 +3666,8 @@ int snd_soc_register_codec(struct device *dev, codec->component.write = snd_soc_codec_drv_write; if (codec_drv->read) codec->component.read = snd_soc_codec_drv_read; + if (codec_drv->set_sysclk) + codec->component.set_sysclk = snd_soc_codec_set_sysclk_; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; dapm = snd_soc_codec_get_dapm(codec); -- 2.14.1

1 0

[alsa-devel] Applied "ASoC: add Component level set_pll" to the asoc tree
by Mark Brown 30 Aug '17

30 Aug '17

The patch ASoC: add Component level set_pll has been applied to the asoc tree at git://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 47bdbaa2c9a6dd7b112fb403b05dd9293b35a41c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 24 Aug 2017 00:57:51 +0000 Subject: [PATCH] ASoC: add Component level set_pll In current ALSA SoC, Codec only has set_pll feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 7 +++++++ sound/soc/soc-core.c | 47 ++++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 49 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e7c476febf82..02a510183ab4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -808,6 +808,8 @@ struct snd_soc_component_driver { /* component wide operations */ int (*set_sysclk)(struct snd_soc_component *component, int clk_id, int source, unsigned int freq, int dir); + int (*set_pll)(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out); /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, @@ -888,6 +890,8 @@ struct snd_soc_component { int (*set_sysclk)(struct snd_soc_component *component, int clk_id, int source, unsigned int freq, int dir); + int (*set_pll)(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out); /* machine specific init */ int (*init)(struct snd_soc_component *component); @@ -1493,6 +1497,9 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, /* component wide operations */ int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id, int source, unsigned int freq, int dir); +int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out); #ifdef CONFIG_REGMAP diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 046ab9189b36..299d23e832ea 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2679,11 +2679,9 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, if (dai->driver && dai->driver->ops->set_pll) return dai->driver->ops->set_pll(dai, pll_id, source, freq_in, freq_out); - else if (dai->codec && dai->codec->driver->set_pll) - return dai->codec->driver->set_pll(dai->codec, pll_id, source, - freq_in, freq_out); - else - return -EINVAL; + + return snd_soc_component_set_pll(dai->component, pll_id, source, + freq_in, freq_out); } EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); @@ -2708,6 +2706,33 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, } EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); +/* + * snd_soc_component_set_pll - configure component PLL. + * @component: COMPONENT + * @pll_id: DAI specific PLL ID + * @source: DAI specific source for the PLL + * @freq_in: PLL input clock frequency in Hz + * @freq_out: requested PLL output clock frequency in Hz + * + * Configures and enables PLL to generate output clock based on input clock. + */ +int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + /* will be removed */ + if (component->set_pll) + return component->set_pll(component, pll_id, source, + freq_in, freq_out); + + if (component->driver->set_pll) + return component->driver->set_pll(component, pll_id, source, + freq_in, freq_out); + + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_component_set_pll); + /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI @@ -3201,6 +3226,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->pcm_new = component->driver->pcm_new; component->pcm_free = component->driver->pcm_free; component->set_sysclk = component->driver->set_sysclk; + component->set_pll = component->driver->set_pll; dapm = &component->dapm; dapm->dev = dev; @@ -3615,6 +3641,15 @@ static int snd_soc_codec_set_sysclk_(struct snd_soc_component *component, return snd_soc_codec_set_sysclk(codec, clk_id, source, freq, dir); } +static int snd_soc_codec_set_pll_(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + return snd_soc_codec_set_pll(codec, pll_id, source, freq_in, freq_out); +} + static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { @@ -3668,6 +3703,8 @@ int snd_soc_register_codec(struct device *dev, codec->component.read = snd_soc_codec_drv_read; if (codec_drv->set_sysclk) codec->component.set_sysclk = snd_soc_codec_set_sysclk_; + if (codec_drv->set_pll) + codec->component.set_pll = snd_soc_codec_set_pll_; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; dapm = snd_soc_codec_get_dapm(codec); -- 2.14.1

1 0

[alsa-devel] Applied "ASoC: add Component level set_jack" to the asoc tree
by Mark Brown 30 Aug '17

30 Aug '17

The patch ASoC: add Component level set_jack has been applied to the asoc tree at git://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 880f02224c624fa2d6e4f8b2d37410fdc98b35fe Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 24 Aug 2017 00:58:07 +0000 Subject: [PATCH] ASoC: add Component level set_jack In current ALSA SoC, Codec only has set_jack feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 4 ++++ sound/soc/soc-core.c | 11 +++++++++++ sound/soc/soc-jack.c | 22 ++++++++++++++++++++++ 3 files changed, 37 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 02a510183ab4..18676ab4f87c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -810,6 +810,8 @@ struct snd_soc_component_driver { int clk_id, int source, unsigned int freq, int dir); int (*set_pll)(struct snd_soc_component *component, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); + int (*set_jack)(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data); /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, @@ -892,6 +894,8 @@ struct snd_soc_component { int clk_id, int source, unsigned int freq, int dir); int (*set_pll)(struct snd_soc_component *component, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); + int (*set_jack)(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data); /* machine specific init */ int (*init)(struct snd_soc_component *component); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 299d23e832ea..78b8886a26d6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3227,6 +3227,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->pcm_free = component->driver->pcm_free; component->set_sysclk = component->driver->set_sysclk; component->set_pll = component->driver->set_pll; + component->set_jack = component->driver->set_jack; dapm = &component->dapm; dapm->dev = dev; @@ -3650,6 +3651,14 @@ static int snd_soc_codec_set_pll_(struct snd_soc_component *component, return snd_soc_codec_set_pll(codec, pll_id, source, freq_in, freq_out); } +static int snd_soc_codec_set_jack_(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + return snd_soc_codec_set_jack(codec, jack, data); +} + static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { @@ -3705,6 +3714,8 @@ int snd_soc_register_codec(struct device *dev, codec->component.set_sysclk = snd_soc_codec_set_sysclk_; if (codec_drv->set_pll) codec->component.set_pll = snd_soc_codec_set_pll_; + if (codec_drv->set_jack) + codec->component.set_jack = snd_soc_codec_set_jack_; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; dapm = snd_soc_codec_get_dapm(codec); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7daf21fee355..2f9f496ab14f 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -40,6 +40,28 @@ int snd_soc_codec_set_jack(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_codec_set_jack); +/** + * snd_soc_component_set_jack - configure component jack. + * @component: COMPONENTs + * @jack: structure to use for the jack + * @data: can be used if codec driver need extra data for configuring jack + * + * Configures and enables jack detection function. + */ +int snd_soc_component_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + /* will be removed */ + if (component->set_jack) + return component->set_jack(component, jack, data); + + if (component->driver->set_jack) + return component->driver->set_jack(component, jack, data); + + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_component_set_jack); + /** * snd_soc_card_jack_new - Create a new jack * @card: ASoC card -- 2.14.1

1 0