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

[alsa-devel] [PATCH 4/6] alsa-lib:pcm: fix the hw_ptr update until the boundary available.
by sutar.mounesh＠gmail.com 19 Jan '18

19 Jan '18

From: mahendran.k <mahendran.kuppusamy(a)in.bosch.com> For long time test case, the slave_hw_ptr will exceed the boundary and wraparound the slave_hw_ptr. This slave boundary wraparound will cause the rate->hw_ptr to wraparound irrespective of the rate->boundary availability and due to that the available size goes wrong. Hence, To get the correct available size, - Its necessary to increment the rate->hw_ptr upto the rate->boundary and then wraparound the rate->hw_ptr. - While handling fraction part of slave period, rounded value will be introduced by input_frames(). To eliminate rounding issue on rate->hw_ptr, subtract last rounded value from rate->hw_ptr and add new rounded value of present slave_hw_ptr fraction part to rate->hw_ptr. Signed-off-by: mahendran.k <mahendran.kuppusamy(a)in.bosch.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 1f830dd..5bee77c 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -50,7 +50,7 @@ typedef struct _snd_pcm_rate snd_pcm_rate_t; struct _snd_pcm_rate { snd_pcm_generic_t gen; - snd_pcm_uframes_t appl_ptr, hw_ptr; + snd_pcm_uframes_t appl_ptr, hw_ptr, last_slave_hw_ptr; snd_pcm_uframes_t last_commit_ptr; snd_pcm_uframes_t orig_avail_min; snd_pcm_sw_params_t sw_params; @@ -563,14 +563,28 @@ static inline void snd_pcm_rate_sync_hwptr0(snd_pcm_t *pcm, snd_pcm_uframes_t sl { snd_pcm_rate_t *rate = pcm->private_data; + snd_pcm_sframes_t slave_hw_ptr_diff = slave_hw_ptr - rate->last_slave_hw_ptr; + snd_pcm_sframes_t last_slave_hw_ptr_frac; + if (pcm->stream != SND_PCM_STREAM_PLAYBACK) return; - /* FIXME: boundary overlap of slave hw_ptr isn't evaluated here! - * e.g. if slave rate is small... - */ - rate->hw_ptr = - (slave_hw_ptr / rate->gen.slave->period_size) * pcm->period_size + - rate->ops.input_frames(rate->obj, slave_hw_ptr % rate->gen.slave->period_size); + + if (slave_hw_ptr_diff < 0) + slave_hw_ptr_diff += rate->gen.slave->boundary; /* slave boundary wraparound */ + else if (slave_hw_ptr_diff == 0) + return; + last_slave_hw_ptr_frac = rate->last_slave_hw_ptr % rate->gen.slave->period_size; + /* While handling fraction part fo slave period, rounded value will be introduced by input_frames(). + * To eliminate rounding issue on rate->hw_ptr, subtract last rounded value from rate->hw_ptr and + * add new rounded value of present slave_hw_ptr fraction part to rate->hw_ptr. Hence, + * rate->hw_ptr += [ (no. of updated slave periods * pcm rate period size) - + * fractional part of last_slave_hw_ptr rounded value + + * fractional part of updated slave hw ptr's rounded value ] */ + rate->hw_ptr += ( + (((last_slave_hw_ptr_frac + slave_hw_ptr_diff) / rate->gen.slave->period_size) * pcm->period_size) - + rate->ops.input_frames(rate->obj, last_slave_hw_ptr_frac) + + rate->ops.input_frames(rate->obj, (last_slave_hw_ptr_frac + slave_hw_ptr_diff) % rate->gen.slave->period_size)); + rate->last_slave_hw_ptr = slave_hw_ptr; rate->hw_ptr %= pcm->boundary; } @@ -635,6 +649,7 @@ static int snd_pcm_rate_prepare(snd_pcm_t *pcm) return err; *pcm->hw.ptr = 0; *pcm->appl.ptr = 0; + rate->last_slave_hw_ptr = 0; err = snd_pcm_rate_init(pcm); if (err < 0) return err; @@ -650,6 +665,7 @@ static int snd_pcm_rate_reset(snd_pcm_t *pcm) return err; *pcm->hw.ptr = 0; *pcm->appl.ptr = 0; + rate->last_slave_hw_ptr = 0; err = snd_pcm_rate_init(pcm); if (err < 0) return err; -- 1.7.9.5

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[alsa-devel] [PATCH 1/4] ASoC: max98373: Changed SPDX header in C++ comments style
by Ryan Lee 18 Jan '18

18 Jan '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changelog: Changed SPDX header in C comment-style to C++ comment-style sound/soc/codecs/max98373.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 9af0d98..e040b03 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +// SPDX-License-Identifier: GPL-2.0 /* Copyright (c) 2017, Maxim Integrated */ #include <linux/acpi.h> -- 2.7.4

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[alsa-devel] [PATCH] ASoC: soc-core: add missing EXPORT_SYMBOL_GPL() for snd_soc_rtdcom_lookup
by Kuninori Morimoto 18 Jan '18

18 Jan '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reported-by: Atsushi Nemoto <anemo(a)mba.ocn.ne.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 971574c..10f808e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -366,6 +366,7 @@ struct snd_soc_component *snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd, return NULL; } +EXPORT_SYMBOL_GPL(snd_soc_rtdcom_lookup); struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: use seq_file to dump the contents of dai_list, platoform_list and codec_list
by Peng Donglin 18 Jan '18

18 Jan '18

Now the debugfs files dais/codecs/platforms have a size limit PAGE_SIZE and the user can not see the whole contents of dai_list/platform_list/codec_list when they are larger than this limit. This patch uses seq_file instead to make sure dais/platforms/codecs show the full contents of dai_list/platform_list/codec_list. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/soc-core.c | 111 +++++++++++++++++---------------------------------- 1 file changed, 37 insertions(+), 74 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c0edac8..7b58211 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -349,120 +349,84 @@ static void soc_init_codec_debugfs(struct snd_soc_component *component) "ASoC: Failed to create codec register debugfs file\n"); } -static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) +static int codec_list_seq_show(struct seq_file *m, void *v) { - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); - ssize_t len, ret = 0; struct snd_soc_codec *codec; - if (!buf) - return -ENOMEM; - mutex_lock(&client_mutex); - list_for_each_entry(codec, &codec_list, list) { - len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", - codec->component.name); - if (len >= 0) - ret += len; - if (ret > PAGE_SIZE) { - ret = PAGE_SIZE; - break; - } - } + list_for_each_entry(codec, &codec_list, list) + seq_printf(m, "%s\n", codec->component.name); mutex_unlock(&client_mutex); - if (ret >= 0) - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); - - kfree(buf); + return 0; +} - return ret; +static int codec_list_seq_open(struct inode *inode, struct file *file) +{ + return single_open(file, codec_list_seq_show, NULL); } static const struct file_operations codec_list_fops = { - .read = codec_list_read_file, - .llseek = default_llseek,/* read accesses f_pos */ + .open = codec_list_seq_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, }; -static ssize_t dai_list_read_file(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) +static int dai_list_seq_show(struct seq_file *m, void *v) { - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); - ssize_t len, ret = 0; struct snd_soc_component *component; struct snd_soc_dai *dai; - if (!buf) - return -ENOMEM; - mutex_lock(&client_mutex); - list_for_each_entry(component, &component_list, list) { - list_for_each_entry(dai, &component->dai_list, list) { - len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", - dai->name); - if (len >= 0) - ret += len; - if (ret > PAGE_SIZE) { - ret = PAGE_SIZE; - break; - } - } - } + list_for_each_entry(component, &component_list, list) + list_for_each_entry(dai, &component->dai_list, list) + seq_printf(m, "%s\n", dai->name); mutex_unlock(&client_mutex); - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); - - kfree(buf); + return 0; +} - return ret; +static int dai_list_seq_open(struct inode *inode, struct file *file) +{ + return single_open(file, dai_list_seq_show, NULL); } static const struct file_operations dai_list_fops = { - .read = dai_list_read_file, - .llseek = default_llseek,/* read accesses f_pos */ + .open = dai_list_seq_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, }; -static ssize_t platform_list_read_file(struct file *file, - char __user *user_buf, - size_t count, loff_t *ppos) +static int platform_list_seq_show(struct seq_file *m, void *v) { - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); - ssize_t len, ret = 0; struct snd_soc_platform *platform; - if (!buf) - return -ENOMEM; - mutex_lock(&client_mutex); - list_for_each_entry(platform, &platform_list, list) { - len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", - platform->component.name); - if (len >= 0) - ret += len; - if (ret > PAGE_SIZE) { - ret = PAGE_SIZE; - break; - } - } + list_for_each_entry(platform, &platform_list, list) + seq_printf(m, "%s\n", platform->component.name); mutex_unlock(&client_mutex); - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); - - kfree(buf); + return 0; +} - return ret; +static int platform_list_seq_open(struct inode *inode, struct file *file) +{ + return single_open(file, platform_list_seq_show, NULL); } static const struct file_operations platform_list_fops = { - .read = platform_list_read_file, - .llseek = default_llseek,/* read accesses f_pos */ + .open = platform_list_seq_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, }; static void soc_init_card_debugfs(struct snd_soc_card *card) @@ -491,7 +455,6 @@ static void soc_cleanup_card_debugfs(struct snd_soc_card *card) debugfs_remove_recursive(card->debugfs_card_root); } - static void snd_soc_debugfs_init(void) { snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL); -- 2.7.4

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[alsa-devel] [PATCH] ASoC: Intel: remove select on non-existing SND_SOC_INTEL_COMMON
by Corentin Labbe 18 Jan '18

18 Jan '18

SND_SST_ATOM_HIFI2_PLATFORM_PCI select SND_SOC_INTEL_COMMON which do not exists anymore. So remove this select. Fixes: c6059879be29 ("ASoC: Intel: Fix Kconfig with top-level selector") Signed-off-by: Corentin Labbe <clabbe.montjoie(a)gmail.com> --- sound/soc/intel/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b0bd1938b71e..f2c9e8c5970a 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -77,7 +77,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI depends on X86 && PCI select SND_SST_IPC_PCI select SND_SOC_COMPRESS - select SND_SOC_INTEL_COMMON help If you have a Intel Medfield or Merrifield/Edison platform, then enable this option by saying Y or m. Distros will typically not -- 2.13.6

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[alsa-devel] [PATCH] ASoC: sgtl5000: Do not print error on probe deferral
by Fabio Estevam 18 Jan '18

18 Jan '18

When the MCLK is not yet available when the codec is probed, probe deferral will happen and in this case we should not print an error message. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/sgtl5000.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f2bb4fe..633cdcf 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1332,10 +1332,13 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, sgtl5000->mclk = devm_clk_get(&client->dev, NULL); if (IS_ERR(sgtl5000->mclk)) { ret = PTR_ERR(sgtl5000->mclk); - dev_err(&client->dev, "Failed to get mclock: %d\n", ret); /* Defer the probe to see if the clk will be provided later */ if (ret == -ENOENT) ret = -EPROBE_DEFER; + + if (ret != -EPROBE_DEFER) + dev_err(&client->dev, "Failed to get mclock: %d\n", + ret); goto disable_regs; } -- 2.7.4

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[alsa-devel] [PATCH] ASoC: bcm2835: fix hw_params error when device is in prepared state
by Matthias Reichl 18 Jan '18

18 Jan '18

If bcm2835 is configured as bitclock master calling hw_params() after prepare() fails with EBUSY. This also makes it impossible to use bcm2835 in full duplex mode. The error is caused by the split clock setup: clk_set_rate is called in hw_params, clk_prepare_enable in prepare. As hw_params doesn't check if the clock was already enabled clk_set_rate fails with EBUSY. Fix this by moving clock startup from prepare to hw_params and let hw_params properly deal with an already set up or enabled clock. Signed-off-by: Matthias Reichl <hias(a)horus.com> --- sound/soc/bcm/bcm2835-i2s.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 2e449d7173fc..d5f73a8ab893 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -130,6 +130,7 @@ struct bcm2835_i2s_dev { struct regmap *i2s_regmap; struct clk *clk; bool clk_prepared; + int clk_rate; }; static void bcm2835_i2s_start_clock(struct bcm2835_i2s_dev *dev) @@ -419,10 +420,19 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, } /* Clock should only be set up here if CPU is clock master */ - if (bit_clock_master) { - ret = clk_set_rate(dev->clk, bclk_rate); - if (ret) - return ret; + if (bit_clock_master && + (!dev->clk_prepared || dev->clk_rate != bclk_rate)) { + if (dev->clk_prepared) + bcm2835_i2s_stop_clock(dev); + + if (dev->clk_rate != bclk_rate) { + ret = clk_set_rate(dev->clk, bclk_rate); + if (ret) + return ret; + dev->clk_rate = bclk_rate; + } + + bcm2835_i2s_start_clock(dev); } /* Setup the frame format */ @@ -618,8 +628,6 @@ static int bcm2835_i2s_prepare(struct snd_pcm_substream *substream, struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); uint32_t cs_reg; - bcm2835_i2s_start_clock(dev); - /* * Clear both FIFOs if the one that should be started * is not empty at the moment. This should only happen -- 2.11.0

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[alsa-devel] Applied "IIO: ADC: stm32-dfsdm: remove unused variable again" to the asoc tree
by Mark Brown 18 Jan '18

18 Jan '18

The patch IIO: ADC: stm32-dfsdm: remove unused variable again 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 166a5a33d5a7bfa62c039eb10e69589b09fd0557 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann <arnd(a)arndb.de> Date: Thu, 18 Jan 2018 12:17:47 +0100 Subject: [PATCH] IIO: ADC: stm32-dfsdm: remove unused variable again The merge between commit abaca806fd13 ("IIO: ADC: stm32-dfsdm: code optimization") and commit 2353758bc2d4 ("IIO: ADC: stm32-dfsdm: avoid unused-variable warning") left one variable behind that is no longer needed and can be removed, as shown by the gcc warning: drivers/iio/adc/stm32-dfsdm-core.c: In function 'stm32_dfsdm_probe': drivers/iio/adc/stm32-dfsdm-core.c:245:29: error: unused variable 'of_id' [-Werror=unused-variable] Fixes: d84b4c7c706f ("Merge branch 'topic/iio' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-st-dfsdm") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/iio/adc/stm32-dfsdm-core.c | 1 - 1 file changed, 1 deletion(-) diff --git a/drivers/iio/adc/stm32-dfsdm-core.c b/drivers/iio/adc/stm32-dfsdm-core.c index 84277bcc465f..6290332cfd3f 100644 --- a/drivers/iio/adc/stm32-dfsdm-core.c +++ b/drivers/iio/adc/stm32-dfsdm-core.c @@ -242,7 +242,6 @@ MODULE_DEVICE_TABLE(of, stm32_dfsdm_of_match); static int stm32_dfsdm_probe(struct platform_device *pdev) { struct dfsdm_priv *priv; - const struct of_device_id *of_id; const struct stm32_dfsdm_dev_data *dev_data; struct stm32_dfsdm *dfsdm; int ret; -- 2.15.1

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[alsa-devel] [PATCH] ALSA: usb-audio: Support changing input on Sound Blaster E1
by Ian Douglas Scott 18 Jan '18

18 Jan '18

The E1 has two headphone jacks, one of which can be set as a microphone input. In the default mode, it uses the built-in microphone as an input. By sending a special command, the second headphone jack is instead used as an input. This might work with the E3 as well, but I don't have one of those to test it. Signed-off-by: Ian Douglas Scott <ian(a)iandouglasscott.com> --- sound/usb/mixer_quirks.c | 79 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 79 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index e1e7ce9ab217..a91a0b75f6db 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -27,6 +27,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/hid.h> #include <linux/init.h> #include <linux/slab.h> #include <linux/usb.h> @@ -1721,6 +1722,80 @@ static int snd_microii_controls_create(struct usb_mixer_interface *mixer) return 0; } +static int snd_soundblaster_e1_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = kcontrol->private_value; + return 0; +} + +static int snd_soundblaster_e1_switch_update(struct usb_mixer_interface *mixer, + unsigned char state) +{ + struct snd_usb_audio *chip = mixer->chip; + int err; + unsigned char buff[2]; + + buff[0] = 0x02; + buff[1] = state ? 0x02 : 0x00; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), + HID_REQ_SET_REPORT, + USB_TYPE_CLASS | USB_RECIP_INTERFACE + | USB_DIR_OUT, + 0x0202, 3, + buff, 2); + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_soundblaster_e1_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + unsigned char value = ucontrol->value.integer.value[0]; + + kcontrol->private_value = value; + return snd_soundblaster_e1_switch_update(list->mixer, value); +} + +static int snd_soundblaster_e1_switch_resume(struct usb_mixer_elem_list *list) +{ + return snd_soundblaster_e1_switch_update(list->mixer, + list->kctl->private_value); +} + +static int snd_soundblaster_e1_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "Mic", "Aux" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static struct snd_kcontrol_new snd_soundblaster_e1_input_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .info = snd_soundblaster_e1_switch_info, + .get = snd_soundblaster_e1_switch_get, + .put = snd_soundblaster_e1_switch_put, + .private_value = 0, +}; + +static int snd_soundblaster_e1_switch_create(struct usb_mixer_interface *mixer) +{ + return add_single_ctl_with_resume(mixer, 0, + snd_soundblaster_e1_switch_resume, + &snd_soundblaster_e1_input_switch, + NULL); +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1802,6 +1877,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x1235, 0x800c): /* Focusrite Scarlett 18i20 */ err = snd_scarlett_controls_create(mixer); break; + + case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */ + err = snd_soundblaster_e1_switch_create(mixer); + break; } return err; -- 2.15.1

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[alsa-devel] Bad PCM stream after a suspend/resume cycle
by Mirza Krak 18 Jan '18

18 Jan '18

Hi. I have a quite simple problem really (simple test-case at least). Following describes the test case: $ aplay test.wav # While the .wav is playing suspend the system $ systemctl suspend # When system is resumed I get the following error on my aplay command aplay: pcm_write:2030: write error: File descriptor in bad state I was expecting for the playback to resume. I did some debugging using "aplay" and what I can see that happens before the EBADFD error is that the "writei_func()" returns a positive value once which results in a call to "snd_pcm_wait()" and on the next "writei_func()" call -EBADFD is returned. I would expect a -ESTRPIPE error which should then result in that the PCM stream to be "resumed" (according to documentation at least). I have tried "forcing" a call to "suspend()" on the first write error in aplay after system is resumed and it actually works, kinda. The playback is resumed even-though the "snd_pcm_resume()" call returns an error. I have not worked much with the sound subsystem before and I am having a hard time following all the call paths to see who/what is to blame for this behavior. Maybe it expected to work like this? And I do not know if this is only on my SoC or if this is a generic sound problem. Information about my system: Using a RK3288 SoC (RK3288-FireFly board) with 4.14.13 Linux kernel (latest stable). alsa-lib version 1.1.4.1 Playback using I2S $ cat /proc/asound/cards 0 [I2S ]: I2S - I2S I2S $ cat /proc/asound/card0/pcm0p/info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: Audio es8328-hifi-analog-0 name: subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 Let me know if there is additional information that I need to provide. -- Med Vänliga Hälsningar / Best Regards Mirza Krak

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