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

[alsa-devel] [PATCH] ASoC: soc-pcm: don't call flush_delayed_work() many times in soc_pcm_private_free()
by Kuninori Morimoto 24 Jan '18

24 Jan '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit f523acebbb74 ("ASoC: add Component level pcm_new/pcm_free v2") added component level pcm_new/pcm_free, but flush_delayed_work() on soc_pcm_private_free() is called in for_each_rtdcom() loop. It doesn't need to be called many times. This patch moves it out of loop. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8075856..998800c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2831,10 +2831,9 @@ static void soc_pcm_private_free(struct snd_pcm *pcm) struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_component *component; + /* need to sync the delayed work before releasing resources */ + flush_delayed_work(&rtd->delayed_work); for_each_rtdcom(rtd, rtdcom) { - /* need to sync the delayed work before releasing resources */ - - flush_delayed_work(&rtd->delayed_work); component = rtdcom->component; if (component->pcm_free) -- 1.9.1

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[alsa-devel] [PATCH v5] ASoC: ak4613: call dummy write for PW_MGMT1/3 when Playback
by Kuninori Morimoto 24 Jan '18

24 Jan '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Power Down Release Command (PMVR, PMDAC, RSTN, PMDA1-PMDA6) which are located on PW_MGMT1 / PW_MGMT3 register must be write again after at least 5 LRCK cycle or later on each command. Otherwise, Playback volume will be 0dB. Basically, it should be 1. PowerDownRelease by Power Management1 <= call 1.x after 5LRCK 1.x Dummy write to Power Management1 2. PowerDownRelease by Power Management3 <= call 2.x after 5LRCK 2.x Dummy write to Power Management3 To avoid too many dummy write, this patch is merging these. 1. PowerDownRelease by Power Management1 2. PowerDownRelease by Power Management3 <= call after 5LRCK 2.x Dummy write to Power Management1/3 <= merge dummy write This patch adds dummy write when Playback Start timing. Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v4 -> v5 - added Note on comment - tidyup comment (delayed work -> schedule_work) sound/soc/codecs/ak4613.c | 78 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 78 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index ee9e822..d85c61e 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -15,6 +15,7 @@ */ #include <linux/clk.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -95,6 +96,9 @@ struct ak4613_priv { struct mutex lock; const struct ak4613_interface *iface; struct snd_pcm_hw_constraint_list constraint; + struct work_struct dummy_write_work; + struct snd_soc_component *component; + unsigned int rate; unsigned int sysclk; unsigned int fmt; @@ -392,6 +396,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } + priv->rate = rate; /* * FIXME @@ -467,11 +472,83 @@ static int ak4613_set_bias_level(struct snd_soc_component *component, return 0; } +static void ak4613_dummy_write(struct work_struct *work) +{ + struct ak4613_priv *priv = container_of(work, + struct ak4613_priv, + dummy_write_work); + struct snd_soc_component *component = priv->component; + unsigned int mgmt1; + unsigned int mgmt3; + + /* + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * + * Note + * + * To avoid extra delay, we want to avoid preemption here, + * but we can't. Because it uses I2C access which is using IRQ + * and sleep. Thus, delay might be more than 5 LR clocks + * see also + * ak4613_dai_trigger() + */ + udelay(5000000 / priv->rate); + + snd_soc_component_read(component, PW_MGMT1, &mgmt1); + snd_soc_component_read(component, PW_MGMT3, &mgmt3); + + snd_soc_component_write(component, PW_MGMT1, mgmt1); + snd_soc_component_write(component, PW_MGMT3, mgmt3); +} + +static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + /* + * FIXME + * + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * from Power Down Release. Otherwise, Playback volume will be 0dB. + * To avoid complex multiple delay/dummy_write method from + * ak4613_set_bias_level() / SND_SOC_DAPM_DAC_E("DACx", ...), + * call it once here. + * + * But, unfortunately, we can't "write" here because here is atomic + * context (It uses I2C access for writing). + * Thus, use schedule_work() to switching to normal context + * immediately. + * + * Note + * + * Calling ak4613_dummy_write() function might be delayed. + * In such case, ak4613 volume might be temporarily 0dB when + * beggining of playback. + * see also + * ak4613_dummy_write() + */ + + if ((cmd != SNDRV_PCM_TRIGGER_START) && + (cmd != SNDRV_PCM_TRIGGER_RESUME)) + return 0; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + + priv->component = &codec->component; + schedule_work(&priv->dummy_write_work); + + return 0; +} + static const struct snd_soc_dai_ops ak4613_dai_ops = { .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, + .trigger = ak4613_dai_trigger, .hw_params = ak4613_dai_hw_params, }; @@ -591,6 +668,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, priv->iface = NULL; priv->cnt = 0; priv->sysclk = 0; + INIT_WORK(&priv->dummy_write_work, ak4613_dummy_write); mutex_init(&priv->lock); -- 1.9.1

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[alsa-devel] [PATCH - JACK PCM plugin] jack: Support snd_pcm_drain()
by twischer＠de.adit-jv.com 24 Jan '18

24 Jan '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Block on drain till available samples played Without this commit the JACK thread will be stopped before the ALSA buffer was completely forwarded to the JACK daemon. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- This patch is depending on http://mailman.alsa-project.org/pipermail/alsa-devel/2018-January/130991.ht… diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index e75aadc..ec94583 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -28,6 +28,17 @@ #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> +/* ALSA supports up to 64 periods per buffer. + * Therefore at least 64 retries are valid and + * should not be handled as an error case + */ +#define MAX_DRAIN_RETRIES 100 +/* ALSA supports a a period with 8192 frames. + * This would result in ~170ms at 48kHz. + * Therefore a time out of 1 second is sufficient + */ +#define DRAIN_TIMEOUT 1000 + typedef enum _jack_format { SND_PCM_JACK_FORMAT_RAW } snd_pcm_jack_format_t; @@ -146,7 +157,13 @@ static int pcm_poll_unblock_check(snd_pcm_ioplug_t *io) snd_pcm_jack_t *jack = io->private_data; avail = snd_pcm_avail_update(io->pcm); - if (avail < 0 || avail >= jack->min_avail) { + /* In draining state poll_fd is used to wait + * till all pending frames are played. + * Therefore it has to be guarantee that a poll event is also generated + * if the buffer contains less than min_avail frames + */ + if (avail < 0 || avail >= jack->min_avail || + jack->state == SND_PCM_STATE_DRAINING) { write(jack->fd, &buf, 1); return 1; } @@ -242,7 +259,8 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) } hw_ptr = jack->hw_ptr; - if (jack->state == SND_PCM_STATE_RUNNING) { + if (jack->state == SND_PCM_STATE_RUNNING || + jack->state == SND_PCM_STATE_DRAINING) { const snd_pcm_channel_area_t *areas = snd_pcm_ioplug_mmap_areas(io); while (xfer < nframes) { @@ -402,6 +420,61 @@ static int snd_pcm_jack_start(snd_pcm_ioplug_t *io) return err; } +static int snd_pcm_jack_drain(snd_pcm_ioplug_t *io) +{ + snd_pcm_jack_t *jack = io->private_data; + const snd_pcm_state_t state = jack->state; + unsigned int retries = MAX_DRAIN_RETRIES; + char buf[32]; + + /* Immediately stop on capture device. + * snd_pcm_jack_stop() will be automatically called + * by snd_pcm_ioplug_drain() + */ + if (io->stream == SND_PCM_STREAM_CAPTURE) { + return 0; + } + + if (snd_pcm_jack_hw_avail(io, jack->hw_ptr, io->appl_ptr) <= 0) { + /* No data pending. Nothing to drain. */ + return 0; + } + + /* start device if not yet done */ + if (state == SND_PCM_STATE_PREPARED) { + snd_pcm_jack_start(io); + } + + /* FIXME: io->state will not be set to SND_PCM_STATE_DRAINING by the + * ALSA library before calling this function. + * Therefore this state has to be stored internally. + */ + jack->state = SND_PCM_STATE_DRAINING; + + struct pollfd pfd; + pfd.fd = io->poll_fd; + pfd.events = io->poll_events | POLLERR | POLLNVAL; + + while (snd_pcm_jack_hw_avail(io, jack->hw_ptr, io->appl_ptr) > 0) { + if (retries <= 0) { + SNDERR("Pending frames not yet processed."); + return -ETIMEDOUT; + } + + if (poll(&pfd, 1, DRAIN_TIMEOUT) < 0) { + SNDERR("Waiting for next JACK process callback failed (err %d)", + -errno); + return -errno; + } + + /* clean pending events. */ + while (read(io->poll_fd, &buf, sizeof(buf)) == sizeof(buf)) + ; + } + + return 0; +} + static int snd_pcm_jack_stop(snd_pcm_ioplug_t *io) { snd_pcm_jack_t *jack = io->private_data; @@ -433,6 +506,7 @@ static snd_pcm_ioplug_callback_t jack_pcm_callback = { .stop = snd_pcm_jack_stop, .pointer = snd_pcm_jack_pointer, .prepare = snd_pcm_jack_prepare, + .drain = snd_pcm_jack_drain, .poll_revents = snd_pcm_jack_poll_revents, }; -- 2.7.4

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[alsa-devel] (no subject)
by twischer＠de.adit-jv.com 24 Jan '18

24 Jan '18

The following patch set will extend the JACK plugin to detect Xruns and signal them to the user application. This patch set is depending on http://mailman.alsa-project.org/pipermail/alsa-devel/2018-January/130987.ht…

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[alsa-devel] Bug message caught in syslog
by Pavel Řezníček 24 Jan '18

24 Jan '18

Dear developers, I'm sorry if I write to the wrong list but I just catched several log messages that I was advised to report as a bug to you, the ALSA developers. I'm not experiencing any sound problems by ear, just found these messages in the syslog (via logcheck). Ran the alsa-info_alsa-info.sh shell script and its output can be found here: http://www.alsa-project.org/db/?f=d9b579a048b94cdc71e619c4deac60e4c8228dbb The log messages in question read as follows: Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: snd_pcm_avail() navrátil hodnotu, která je nezvykle vysoká: 570108 bajtů (3231 ms). Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: S největší pravděpodobností se jedná o chybu v ovladači ALSA "snd_hda_intel". Nahlaste prosím tento problém vývojářům ALSA. Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: snd_pcm_dump(): Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: Soft volume PCM Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: Control: PCM Playback Volume Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: min_dB: -51 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: max_dB: 0 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: resolution: 256 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: Its setup is: Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: stream : PLAYBACK Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: access : MMAP_INTERLEAVED Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: format : S16_LE Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: subformat : STD Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: channels : 2 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: rate : 44100 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: exact rate : 44100 (44100/1) Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: msbits : 16 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: buffer_size : 16384 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_size : 8192 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_time : 185759 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: tstamp_mode : ENABLE Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: tstamp_type : MONOTONIC Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_step : 1 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: avail_min : 15802 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_event : 0 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: start_threshold : -1 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: stop_threshold : 4611686018427387904 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: silence_threshold: 0 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: silence_size : 0 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: boundary : 4611686018427387904 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: Its setup is: Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: stream : PLAYBACK Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: access : MMAP_INTERLEAVED Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: format : S16_LE Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: subformat : STD Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: channels : 2 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: rate : 44100 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: exact rate : 44100 (44100/1) Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: msbits : 16 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: buffer_size : 16384 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_size : 8192 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_time : 185759 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: tstamp_mode : ENABLE Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: tstamp_type : MONOTONIC Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_step : 1 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: avail_min : 15802 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_event : 0 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: start_threshold : -1 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: stop_threshold : 4611686018427387904 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: silence_threshold: 0 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: silence_size : 0 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: boundary : 4611686018427387904 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: appl_ptr : 95068169 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: hw_ptr : 95194312 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: snd_pcm_delay() navrátil hodnotu, která je nezvykle vysoká: -454064 bajtů (-2574 ms). Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: S největší pravděpodobností se jedná o chybu v ovladači ALSA "snd_hda_intel". Nahlaste prosím tento problém vývojářům ALSA. Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: snd_pcm_dump(): Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: Soft volume PCM Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: Control: PCM Playback Volume Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: min_dB: -51 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: max_dB: 0 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: resolution: 256 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: Its setup is: Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: stream : PLAYBACK Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: access : MMAP_INTERLEAVED Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: format : S16_LE Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: subformat : STD Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: channels : 2 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: rate : 44100 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: exact rate : 44100 (44100/1) Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: msbits : 16 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: buffer_size : 16384 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_size : 8192 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_time : 185759 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: tstamp_mode : ENABLE Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: tstamp_type : MONOTONIC Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_step : 1 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: avail_min : 15802 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_event : 0 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: start_threshold : -1 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: stop_threshold : 4611686018427387904 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: silence_threshold: 0 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: silence_size : 0 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: boundary : 4611686018427387904 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: Its setup is: Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: stream : PLAYBACK Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: access : MMAP_INTERLEAVED Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: format : S16_LE Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: subformat : STD Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: channels : 2 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: rate : 44100 Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: exact rate : 44100 (44100/1) Jan 24 07:01:25 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: msbits : 16 Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: buffer_size : 16384 Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_size : 8192 Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_time : 185759 Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: tstamp_mode : ENABLE Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: tstamp_type : MONOTONIC Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_step : 1 Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: avail_min : 15802 Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: period_event : 0 Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: start_threshold : -1 Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: stop_threshold : 4611686018427387904 Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: silence_threshold: 0 Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: silence_size : 0 Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: boundary : 4611686018427387904 Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: appl_ptr : 95099921 Jan 24 07:01:26 ideapad-310 pulseaudio[2724]: [alsa-sink-CX20751/2 Analog] alsa-util.c: hw_ptr : 95213480 I'd be happy to help you solving this issue if not known already. Please feel free to ask for whatever information you'd need. Pavel Řezníček A. K. A. Cigydd

1 0

[alsa-devel] [PATCH][resend] ASoC: soc-pcm: don't call flush_delayed_work() many times in soc_pcm_private_free()
by Kuninori Morimoto 24 Jan '18

24 Jan '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit f523acebbb74 ("ASoC: add Component level pcm_new/pcm_free v2") added component level pcm_new/pcm_free, but flush_delayed_work() on soc_pcm_private_free() is called in for_each_rtdcom() loop. It doesn't need to be called many times. This patch moves it out of loop. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- 2weeks passed. resend sound/soc/soc-pcm.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8075856..998800c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2831,10 +2831,9 @@ static void soc_pcm_private_free(struct snd_pcm *pcm) struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_component *component; + /* need to sync the delayed work before releasing resources */ + flush_delayed_work(&rtd->delayed_work); for_each_rtdcom(rtd, rtdcom) { - /* need to sync the delayed work before releasing resources */ - - flush_delayed_work(&rtd->delayed_work); component = rtdcom->component; if (component->pcm_free) -- 1.9.1

1 0

[alsa-devel] [PATCH v5][RESEND v2] ASoC: ak4613: call dummy write for PW_MGMT1/3 when Playback
by Kuninori Morimoto 24 Jan '18

24 Jan '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Power Down Release Command (PMVR, PMDAC, RSTN, PMDA1-PMDA6) which are located on PW_MGMT1 / PW_MGMT3 register must be write again after at least 5 LRCK cycle or later on each command. Otherwise, Playback volume will be 0dB. Basically, it should be 1. PowerDownRelease by Power Management1 <= call 1.x after 5LRCK 1.x Dummy write to Power Management1 2. PowerDownRelease by Power Management3 <= call 2.x after 5LRCK 2.x Dummy write to Power Management3 To avoid too many dummy write, this patch is merging these. 1. PowerDownRelease by Power Management1 2. PowerDownRelease by Power Management3 <= call after 5LRCK 2.x Dummy write to Power Management1/3 <= merge dummy write This patch adds dummy write when Playback Start timing. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- 2week passed. resend sound/soc/codecs/ak4613.c | 78 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 78 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index ee9e822..d85c61e 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -15,6 +15,7 @@ */ #include <linux/clk.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -95,6 +96,9 @@ struct ak4613_priv { struct mutex lock; const struct ak4613_interface *iface; struct snd_pcm_hw_constraint_list constraint; + struct work_struct dummy_write_work; + struct snd_soc_component *component; + unsigned int rate; unsigned int sysclk; unsigned int fmt; @@ -392,6 +396,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } + priv->rate = rate; /* * FIXME @@ -467,11 +472,83 @@ static int ak4613_set_bias_level(struct snd_soc_component *component, return 0; } +static void ak4613_dummy_write(struct work_struct *work) +{ + struct ak4613_priv *priv = container_of(work, + struct ak4613_priv, + dummy_write_work); + struct snd_soc_component *component = priv->component; + unsigned int mgmt1; + unsigned int mgmt3; + + /* + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * + * Note + * + * To avoid extra delay, we want to avoid preemption here, + * but we can't. Because it uses I2C access which is using IRQ + * and sleep. Thus, delay might be more than 5 LR clocks + * see also + * ak4613_dai_trigger() + */ + udelay(5000000 / priv->rate); + + snd_soc_component_read(component, PW_MGMT1, &mgmt1); + snd_soc_component_read(component, PW_MGMT3, &mgmt3); + + snd_soc_component_write(component, PW_MGMT1, mgmt1); + snd_soc_component_write(component, PW_MGMT3, mgmt3); +} + +static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + /* + * FIXME + * + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * from Power Down Release. Otherwise, Playback volume will be 0dB. + * To avoid complex multiple delay/dummy_write method from + * ak4613_set_bias_level() / SND_SOC_DAPM_DAC_E("DACx", ...), + * call it once here. + * + * But, unfortunately, we can't "write" here because here is atomic + * context (It uses I2C access for writing). + * Thus, use schedule_work() to switching to normal context + * immediately. + * + * Note + * + * Calling ak4613_dummy_write() function might be delayed. + * In such case, ak4613 volume might be temporarily 0dB when + * beggining of playback. + * see also + * ak4613_dummy_write() + */ + + if ((cmd != SNDRV_PCM_TRIGGER_START) && + (cmd != SNDRV_PCM_TRIGGER_RESUME)) + return 0; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + + priv->component = &codec->component; + schedule_work(&priv->dummy_write_work); + + return 0; +} + static const struct snd_soc_dai_ops ak4613_dai_ops = { .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, + .trigger = ak4613_dai_trigger, .hw_params = ak4613_dai_hw_params, }; @@ -591,6 +668,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, priv->iface = NULL; priv->cnt = 0; priv->sysclk = 0; + INIT_WORK(&priv->dummy_write_work, ak4613_dummy_write); mutex_init(&priv->lock); -- 1.9.1 _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 0

[alsa-devel] [PATCH] ASoC: soc-core: snd_soc_rtdcom_lookup() cares component driver name
by Kuninori Morimoto 23 Jan '18

23 Jan '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_rtdcom_lookup() look up component by uisng driver name. Then, it uses component->driver->name. Some driver might doesn't have it, thus it should care NULL pointer. This patch solve this issue. Reported-by: Mukunda,Vijendar <vijendar.mukunda(a)amd.com> Reported-by: Manuel Lauss <manuel.lauss(a)gmail.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Mukunda,Vijendar <vijendar.mukunda(a)amd.com> --- sound/soc/soc-core.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 10f808e..c9fdcbb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -358,9 +358,17 @@ struct snd_soc_component *snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd, { struct snd_soc_rtdcom_list *rtdcom; + if (!driver_name) + return NULL; + for_each_rtdcom(rtd, rtdcom) { - if ((rtdcom->component->driver->name == driver_name) || - strcmp(rtdcom->component->driver->name, driver_name) == 0) + const char *component_name = rtdcom->component->driver->name; + + if (!component_name) + continue; + + if ((component_name == driver_name) || + strcmp(component_name, driver_name) == 0) return rtdcom->component; } -- 1.9.1

2 1

[alsa-devel] [PATCH] sound: emu10k1: Replace GFP_ATOMIC with GFP_KERNEL in synth_alloc_pages
by Jia-Ju Bai 23 Jan '18

23 Jan '18

The function synth_alloc_pages is not called in atomic context. Thus GFP_ATOMIC is not necessary, and it can be replaced with GFP_KERNEL. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/pci/emu10k1/memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 4f1f69b..a4dbae1 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -477,7 +477,7 @@ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk if (p) __free_page(p); /* try to allocate from <16MB zone */ - p = alloc_page(GFP_ATOMIC | GFP_DMA | + p = alloc_page(GFP_KERNEL | GFP_DMA | __GFP_NORETRY | /* no OOM-killer */ __GFP_NOWARN); } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: intel: clean up CONFIG_SND_SST_IPC handling
by Arnd Bergmann 22 Jan '18

22 Jan '18

In a configuration with SND_SST_ATOM_HIFI2_PLATFORM_PCI=y and SND_SST_ATOM_HIFI2_PLATFORM=m, we get this module link failure: ERROR: "sst_context_init" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_context_cleanup" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_alloc_drv_context" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "intel_sst_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_configure_runtime_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! The problem is that the sound/soc/intel/atom/ directory only gets entered by Kbuild when SND_SST_ATOM_HIFI2_PLATFORM is set, so we only build modules (obj-m) under here but not built-in files (obj-y) including the snd-intel-sst-core that we clearly want need here. Before commit 4772c16ede52 ("ASoC: Intel: Kconfig: Simplify-clarify ACPI/PCI dependencies"), this could not happen as all files in sound/soc/intel/atom/ depended on CONFIG_SND_SST_ATOM_HIFI2_PLATFORM anyway. Slightly later, commit 05f4434bc130 ("ASoC: Intel: remove mfld_machine") was added, which then removed the Merrifield machine code that happened to be the only user of SND_SST_ATOM_HIFI2_PLATFORM_PCI. With that gone, it appears that we can completely remove that symbol as well, along with the SND_SST_IPC_PCI code and everything associated with it that is now unused. For clarify, this also removes the SND_SST_IPC_ACPI symbol that is now synonymous with SND_SST_IPC, and links both into a single loadable module. Fixes: 4772c16ede52 ("ASoC: Intel: Kconfig: Simplify-clarify ACPI/PCI dependencies") Fixes: 05f4434bc130 ("ASoC: Intel: remove mfld_machine") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- I checked that this patch leads to a clean compilation, and that the code I'm removing is only used for the now-obsolete Merrifield PCI ID. If I got something else wrong, or you want a different fix, please just send a replacement patch and treat this as a bug report. --- sound/soc/intel/Kconfig | 27 +---- sound/soc/intel/atom/sst/Makefile | 6 +- sound/soc/intel/atom/sst/sst_pci.c | 209 ------------------------------------- sound/soc/intel/common/sst-acpi.c | 4 +- 4 files changed, 5 insertions(+), 241 deletions(-) delete mode 100644 sound/soc/intel/atom/sst/sst_pci.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f2c9e8c5970a..2dc8cda7a7cd 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -17,21 +17,11 @@ if SND_SOC_INTEL_SST_TOPLEVEL config SND_SST_IPC tristate # This option controls the IPC core for HiFi2 platforms - -config SND_SST_IPC_PCI - tristate - select SND_SST_IPC - # This option controls the PCI-based IPC for HiFi2 platforms - # (Medfield, Merrifield). - -config SND_SST_IPC_ACPI - tristate - select SND_SST_IPC - # This option controls the ACPI-based IPC for HiFi2 platforms # (Baytrail, Cherrytrail) config SND_SOC_INTEL_SST_ACPI tristate + select SND_SST_IPC # This option controls ACPI-based probing on # Haswell/Broadwell/Baytrail legacy and will be set # when these platforms are enabled @@ -72,23 +62,10 @@ config SND_SOC_INTEL_BAYTRAIL for Baytrail Chromebooks but this option is now deprecated and is not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead. -config SND_SST_ATOM_HIFI2_PLATFORM_PCI - tristate "PCI HiFi2 (Medfield, Merrifield) Platforms" - depends on X86 && PCI - select SND_SST_IPC_PCI - select SND_SOC_COMPRESS - help - If you have a Intel Medfield or Merrifield/Edison platform, then - enable this option by saying Y or m. Distros will typically not - enable this option: Medfield devices are not available to - developers and while Merrifield/Edison can run a mainline kernel with - limited functionality it will require a firmware file which - is not in the standard firmware tree - config SND_SST_ATOM_HIFI2_PLATFORM tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" depends on X86 && ACPI - select SND_SST_IPC_ACPI + select SND_SST_IPC select SND_SOC_COMPRESS select SND_SOC_ACPI_INTEL_MATCH select IOSF_MBI diff --git a/sound/soc/intel/atom/sst/Makefile b/sound/soc/intel/atom/sst/Makefile index 795d1cf8f386..f5b7008b4cea 100644 --- a/sound/soc/intel/atom/sst/Makefile +++ b/sound/soc/intel/atom/sst/Makefile @@ -1,8 +1,4 @@ # SPDX-License-Identifier: GPL-2.0 -snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o -snd-intel-sst-pci-objs += sst_pci.o -snd-intel-sst-acpi-objs += sst_acpi.o +snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o sst_acpi.o obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o -obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o -obj-$(CONFIG_SND_SST_IPC_ACPI) += snd-intel-sst-acpi.o diff --git a/sound/soc/intel/atom/sst/sst_pci.c b/sound/soc/intel/atom/sst/sst_pci.c deleted file mode 100644 index 6906ee624cf6..000000000000 --- a/sound/soc/intel/atom/sst/sst_pci.c +++ /dev/null @@ -1,209 +0,0 @@ -/* - * sst_pci.c - SST (LPE) driver init file for pci enumeration. - * - * Copyright (C) 2008-14 Intel Corp - * Authors: Vinod Koul <vinod.koul(a)intel.com> - * Harsha Priya <priya.harsha(a)intel.com> - * Dharageswari R <dharageswari.r(a)intel.com> - * KP Jeeja <jeeja.kp(a)intel.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; version 2 of the License. - * - * 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. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ -#include <linux/module.h> -#include <linux/pci.h> -#include <linux/fs.h> -#include <linux/firmware.h> -#include <linux/pm_runtime.h> -#include <sound/core.h> -#include <sound/soc.h> -#include <asm/platform_sst_audio.h> -#include "../sst-mfld-platform.h" -#include "sst.h" - -static int sst_platform_get_resources(struct intel_sst_drv *ctx) -{ - int ddr_base, ret = 0; - struct pci_dev *pci = ctx->pci; - - ret = pci_request_regions(pci, SST_DRV_NAME); - if (ret) - return ret; - - /* map registers */ - /* DDR base */ - if (ctx->dev_id == SST_MRFLD_PCI_ID) { - ctx->ddr_base = pci_resource_start(pci, 0); - /* check that the relocated IMR base matches with FW Binary */ - ddr_base = relocate_imr_addr_mrfld(ctx->ddr_base); - if (!ctx->pdata->lib_info) { - dev_err(ctx->dev, "lib_info pointer NULL\n"); - ret = -EINVAL; - goto do_release_regions; - } - if (ddr_base != ctx->pdata->lib_info->mod_base) { - dev_err(ctx->dev, - "FW LSP DDR BASE does not match with IFWI\n"); - ret = -EINVAL; - goto do_release_regions; - } - ctx->ddr_end = pci_resource_end(pci, 0); - - ctx->ddr = pcim_iomap(pci, 0, - pci_resource_len(pci, 0)); - if (!ctx->ddr) { - ret = -EINVAL; - goto do_release_regions; - } - dev_dbg(ctx->dev, "sst: DDR Ptr %p\n", ctx->ddr); - } else { - ctx->ddr = NULL; - } - /* SHIM */ - ctx->shim_phy_add = pci_resource_start(pci, 1); - ctx->shim = pcim_iomap(pci, 1, pci_resource_len(pci, 1)); - if (!ctx->shim) { - ret = -EINVAL; - goto do_release_regions; - } - dev_dbg(ctx->dev, "SST Shim Ptr %p\n", ctx->shim); - - /* Shared SRAM */ - ctx->mailbox_add = pci_resource_start(pci, 2); - ctx->mailbox = pcim_iomap(pci, 2, pci_resource_len(pci, 2)); - if (!ctx->mailbox) { - ret = -EINVAL; - goto do_release_regions; - } - dev_dbg(ctx->dev, "SRAM Ptr %p\n", ctx->mailbox); - - /* IRAM */ - ctx->iram_end = pci_resource_end(pci, 3); - ctx->iram_base = pci_resource_start(pci, 3); - ctx->iram = pcim_iomap(pci, 3, pci_resource_len(pci, 3)); - if (!ctx->iram) { - ret = -EINVAL; - goto do_release_regions; - } - dev_dbg(ctx->dev, "IRAM Ptr %p\n", ctx->iram); - - /* DRAM */ - ctx->dram_end = pci_resource_end(pci, 4); - ctx->dram_base = pci_resource_start(pci, 4); - ctx->dram = pcim_iomap(pci, 4, pci_resource_len(pci, 4)); - if (!ctx->dram) { - ret = -EINVAL; - goto do_release_regions; - } - dev_dbg(ctx->dev, "DRAM Ptr %p\n", ctx->dram); -do_release_regions: - pci_release_regions(pci); - return 0; -} - -/* - * intel_sst_probe - PCI probe function - * - * @pci: PCI device structure - * @pci_id: PCI device ID structure - * - */ -static int intel_sst_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) -{ - int ret = 0; - struct intel_sst_drv *sst_drv_ctx; - struct sst_platform_info *sst_pdata = pci->dev.platform_data; - - dev_dbg(&pci->dev, "Probe for DID %x\n", pci->device); - ret = sst_alloc_drv_context(&sst_drv_ctx, &pci->dev, pci->device); - if (ret < 0) - return ret; - - sst_drv_ctx->pdata = sst_pdata; - sst_drv_ctx->irq_num = pci->irq; - snprintf(sst_drv_ctx->firmware_name, sizeof(sst_drv_ctx->firmware_name), - "%s%04x%s", "fw_sst_", - sst_drv_ctx->dev_id, ".bin"); - - ret = sst_context_init(sst_drv_ctx); - if (ret < 0) - return ret; - - /* Init the device */ - ret = pcim_enable_device(pci); - if (ret) { - dev_err(sst_drv_ctx->dev, - "device can't be enabled. Returned err: %d\n", ret); - goto do_free_drv_ctx; - } - sst_drv_ctx->pci = pci_dev_get(pci); - ret = sst_platform_get_resources(sst_drv_ctx); - if (ret < 0) - goto do_free_drv_ctx; - - pci_set_drvdata(pci, sst_drv_ctx); - sst_configure_runtime_pm(sst_drv_ctx); - - return ret; - -do_free_drv_ctx: - sst_context_cleanup(sst_drv_ctx); - dev_err(sst_drv_ctx->dev, "Probe failed with %d\n", ret); - return ret; -} - -/** - * intel_sst_remove - PCI remove function - * - * @pci: PCI device structure - * - * This function is called by OS when a device is unloaded - * This frees the interrupt etc - */ -static void intel_sst_remove(struct pci_dev *pci) -{ - struct intel_sst_drv *sst_drv_ctx = pci_get_drvdata(pci); - - sst_context_cleanup(sst_drv_ctx); - pci_dev_put(sst_drv_ctx->pci); - pci_release_regions(pci); - pci_set_drvdata(pci, NULL); -} - -/* PCI Routines */ -static const struct pci_device_id intel_sst_ids[] = { - { PCI_VDEVICE(INTEL, SST_MRFLD_PCI_ID), 0}, - { 0, } -}; - -static struct pci_driver sst_driver = { - .name = SST_DRV_NAME, - .id_table = intel_sst_ids, - .probe = intel_sst_probe, - .remove = intel_sst_remove, -#ifdef CONFIG_PM - .driver = { - .pm = &intel_sst_pm, - }, -#endif -}; - -module_pci_driver(sst_driver); - -MODULE_DESCRIPTION("Intel (R) SST(R) Audio Engine PCI Driver"); -MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); -MODULE_AUTHOR("Harsha Priya <priya.harsha(a)intel.com>"); -MODULE_AUTHOR("Dharageswari R <dharageswari.r(a)intel.com>"); -MODULE_AUTHOR("KP Jeeja <jeeja.kp(a)intel.com>"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("sst"); diff --git a/sound/soc/intel/common/sst-acpi.c b/sound/soc/intel/common/sst-acpi.c index cf6fbbd4e378..3b669d5adbb6 100644 --- a/sound/soc/intel/common/sst-acpi.c +++ b/sound/soc/intel/common/sst-acpi.c @@ -206,7 +206,7 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { .dma_size = SST_LPT_DSP_DMA_SIZE, }; -#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI) +#if !IS_ENABLED(CONFIG_SND_SST_IPC) static struct sst_acpi_desc sst_acpi_baytrail_desc = { .drv_name = "baytrail-pcm-audio", .machines = snd_soc_acpi_intel_baytrail_legacy_machines, @@ -222,7 +222,7 @@ static struct sst_acpi_desc sst_acpi_baytrail_desc = { static const struct acpi_device_id sst_acpi_match[] = { { "INT33C8", (unsigned long)&sst_acpi_haswell_desc }, { "INT3438", (unsigned long)&sst_acpi_broadwell_desc }, -#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI) +#if !IS_ENABLED(CONFIG_SND_SST_IPC) { "80860F28", (unsigned long)&sst_acpi_baytrail_desc }, #endif { } -- 2.9.0

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