November 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v4 1/2] ASoC: Intel - use control components to describe card config
by Jaroslav Kysela 20 Nov '19

20 Nov '19

Use the control interface (field 'components' in the info structure) to pass the I/O configuration details. The goal is to replace the card long name with this. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bytcht_es8316.c | 10 +++++++++- sound/soc/intel/boards/bytcr_rt5640.c | 6 ++++++ sound/soc/intel/boards/bytcr_rt5651.c | 18 +++++++++++------- 3 files changed, 26 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 46612331f5ea..4090550f0fe1 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -361,6 +361,7 @@ static struct snd_soc_dai_link byt_cht_es8316_dais[] = { /* SoC card */ static char codec_name[SND_ACPI_I2C_ID_LEN]; static char long_name[50]; /* = "bytcht-es8316-*-spk-*-mic" */ +static char components_string[32]; /* = "cfg-spk:* cfg-mic:* */ static int byt_cht_es8316_suspend(struct snd_soc_card *card) { @@ -572,17 +573,24 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) } } - /* register the soc card */ + snprintf(components_string, sizeof(components_string), + "cfg-spk:%s cfg-mic:%s", + (quirk & BYT_CHT_ES8316_MONO_SPEAKER) ? "1" : "2", + mic_name[BYT_CHT_ES8316_MAP(quirk)]); + byt_cht_es8316_card.components = components_string; snprintf(long_name, sizeof(long_name), "bytcht-es8316-%s-spk-%s-mic", (quirk & BYT_CHT_ES8316_MONO_SPEAKER) ? "mono" : "stereo", mic_name[BYT_CHT_ES8316_MAP(quirk)]); byt_cht_es8316_card.long_name = long_name; + + /* register the soc card */ snd_soc_card_set_drvdata(&byt_cht_es8316_card, priv); ret = devm_snd_soc_register_card(dev, &byt_cht_es8316_card); if (ret) { gpiod_put(priv->speaker_en_gpio); dev_err(dev, "snd_soc_register_card failed: %d\n", ret); + return ret; } platform_set_drvdata(pdev, &byt_cht_es8316_card); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 9c1aa4ec9cba..0170d31e691a 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1081,6 +1081,7 @@ static char byt_rt5640_codec_name[SND_ACPI_I2C_ID_LEN]; static char byt_rt5640_codec_aif_name[12]; /* = "rt5640-aif[1|2]" */ static char byt_rt5640_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ static char byt_rt5640_long_name[40]; /* = "bytcr-rt5640-*-spk-*-mic" */ +static char byt_rt5640_components[32]; /* = "cfg-spk:* cfg-mic:*" */ static int byt_rt5640_suspend(struct snd_soc_card *card) { @@ -1303,6 +1304,11 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) } } + snprintf(byt_rt5640_components, sizeof(byt_rt5640_components), + "cfg-spk:%s cfg-mic:%s", + (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) ? "1" : "2", + map_name[BYT_RT5640_MAP(byt_rt5640_quirk)]); + byt_rt5640_card.components = byt_rt5640_components; snprintf(byt_rt5640_long_name, sizeof(byt_rt5640_long_name), "bytcr-rt5640-%s-spk-%s-mic", (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) ? diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 4606f6f582d6..80a5674ddb1b 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -798,6 +798,7 @@ static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ static char byt_rt5651_long_name[50]; /* = "bytcr-rt5651-*-spk-*-mic[-swapped-hp]" */ +static char byt_rt5651_components[50]; /* = "cfg-spk:* cfg-mic:*" */ static int byt_rt5651_suspend(struct snd_soc_card *card) { @@ -876,7 +877,6 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) const char *platform_name; struct acpi_device *adev; struct device *codec_dev; - const char *hp_swapped; bool is_bytcr = false; int ret_val = 0; int dai_index = 0; @@ -1080,16 +1080,20 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) } } - if (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) - hp_swapped = "-hp-swapped"; - else - hp_swapped = ""; - + snprintf(byt_rt5651_components, sizeof(byt_rt5651_components), + "cfg-spk:%s cfg-mic:%s%s", + (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ? "1" : "2", + mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], + (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) ? + " cfg-hp:lrswap" : ""); + byt_rt5651_card.components = byt_rt5651_components; snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name), "bytcr-rt5651-%s-spk-%s-mic%s", (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ? "mono" : "stereo", - mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], hp_swapped); + mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], + (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) ? + "-hp-swapped" : ""); byt_rt5651_card.long_name = byt_rt5651_long_name; /* override plaform name, if required */ -- 2.20.1

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Re: [alsa-devel] [PATCH v4 6/7] ALSA: aloop: Support selection of snd_timer instead of jiffies
by Andrew Gabbasov 20 Nov '19

20 Nov '19

Hello Takashi, > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, November 20, 2019 6:59 PM > To: Gabbasov, Andrew > Cc: alsa-devel(a)alsa-project.org; linux-kernel(a)vger.kernel.org; Jaroslav > Kysela; Takashi Iwai; Timo Wischer > Subject: Re: [PATCH v4 6/7] ALSA: aloop: Support selection of snd_timer > instead of jiffies > > On Wed, 20 Nov 2019 16:39:00 +0100, > Andrew Gabbasov wrote: > > > > Hello Takashi, > > > > > -----Original Message----- > > > From: Takashi Iwai [mailto:tiwai@suse.de] > > > Sent: Wednesday, November 20, 2019 6:32 PM > > > To: Gabbasov, Andrew > > > Cc: alsa-devel(a)alsa-project.org; linux-kernel(a)vger.kernel.org; Jaroslav > > > Kysela; Takashi Iwai; Timo Wischer > > > Subject: Re: [PATCH v4 6/7] ALSA: aloop: Support selection of snd_timer > > > instead of jiffies > > > > > > On Wed, 20 Nov 2019 16:21:36 +0100, > > > Andrew Gabbasov wrote: > > > > > > > > Hello Takashi, > > > > > > > > > -----Original Message----- > > > > > From: Takashi Iwai [mailto:tiwai@suse.de] > > > > > Sent: Wednesday, November 20, 2019 5:34 PM > > > > > To: Gabbasov, Andrew > > > > > Cc: alsa-devel(a)alsa-project.org; linux-kernel(a)vger.kernel.org; > > Jaroslav > > > > > Kysela; Takashi Iwai; Timo Wischer > > > > > Subject: Re: [PATCH v4 6/7] ALSA: aloop: Support selection of > > snd_timer > > > > > instead of jiffies > > > > > > > > > > On Wed, 20 Nov 2019 12:58:55 +0100, > > > > > Andrew Gabbasov wrote: > > > > > > +/* call in loopback->cable_lock */ > > > > > > +static int loopback_snd_timer_open(struct loopback_pcm *dpcm) > > > > > > +{ > > > > > > + int err = 0; > > > > > > + struct snd_timer_id tid = { > > > > > > + .dev_class = SNDRV_TIMER_CLASS_PCM, > > > > > > + .dev_sclass = SNDRV_TIMER_SCLASS_APPLICATION, > > > > > > + }; > > > > > > + struct snd_timer_instance *timeri; > > > > > > + struct loopback_cable *cable = dpcm->cable; > > > > > > + > > > > > > + spin_lock_irq(&cable->lock); > > > > > > + > > > > > > + /* check if timer was already opened. It is only opened once > > > > > > + * per playback and capture subdevice (aka cable). > > > > > > + */ > > > > > > + if (cable->snd_timer.instance) > > > > > > + goto unlock; > > > > > > + > > > > > > + err = loopback_parse_timer_id(dpcm->loopback->timer_source, > > > &tid); > > > > > > + if (err < 0) { > > > > > > + pcm_err(dpcm->substream->pcm, > > > > > > + "Parsing timer source \'%s\' failed with > > %d", > > > > > > + dpcm->loopback->timer_source, err); > > > > > > + goto unlock; > > > > > > + } > > > > > > + > > > > > > + cable->snd_timer.stream = dpcm->substream->stream; > > > > > > + cable->snd_timer.id = tid; > > > > > > + > > > > > > + timeri = snd_timer_instance_new(dpcm->loopback->card->id); > > > > > > + if (!timeri) { > > > > > > + err = -ENOMEM; > > > > > > + goto unlock; > > > > > > + } > > > > > > + /* The callback has to be called from another tasklet. If > > > > > > + * SNDRV_TIMER_IFLG_FAST is specified it will be called from > > > the > > > > > > + * snd_pcm_period_elapsed() call of the selected sound card. > > > > > > + * snd_pcm_period_elapsed() helds > > > snd_pcm_stream_lock_irqsave(). > > > > > > + * Due to our callback loopback_snd_timer_function() also > > > calls > > > > > > + * snd_pcm_period_elapsed() which calls > > > > > snd_pcm_stream_lock_irqsave(). > > > > > > + * This would end up in a dead lock. > > > > > > + */ > > > > > > + timeri->flags |= SNDRV_TIMER_IFLG_AUTO; > > > > > > + timeri->callback = loopback_snd_timer_function; > > > > > > + timeri->callback_data = (void *)cable; > > > > > > + timeri->ccallback = loopback_snd_timer_event; > > > > > > + > > > > > > + /* snd_timer_close() and snd_timer_open() should not be > > called > > > with > > > > > > + * locked spinlock because both functions can block on a > > > mutex. The > > > > > > + * mutex loopback->cable_lock is kept locked. Therefore > > > > > snd_timer_open() > > > > > > + * cannot be called a second time by the other device of the > > > same > > > > > cable. > > > > > > + * Therefore the following issue cannot happen: > > > > > > + * [proc1] Call loopback_timer_open() -> > > > > > > + * Unlock cable->lock for snd_timer_close/open() > > call > > > > > > + * [proc2] Call loopback_timer_open() -> snd_timer_open(), > > > > > > + * snd_timer_start() > > > > > > + * [proc1] Call snd_timer_open() and overwrite running timer > > > > > > + * instance > > > > > > + */ > > > > > > + spin_unlock_irq(&cable->lock); > > > > > > + err = snd_timer_open(timeri, &cable->snd_timer.id, current- > > > >pid); > > > > > > + if (err < 0) { > > > > > > + pcm_err(dpcm->substream->pcm, > > > > > > + "snd_timer_open (%d,%d,%d) failed with %d", > > > > > > + cable->snd_timer.id.card, > > > > > > + cable->snd_timer.id.device, > > > > > > + cable->snd_timer.id.subdevice, > > > > > > + err); > > > > > > + snd_timer_instance_free(timeri); > > > > > > + return err; > > > > > > + } > > > > > > + spin_lock_irq(&cable->lock); > > > > > > + > > > > > > + cable->snd_timer.instance = timeri; > > > > > > + > > > > > > + /* initialise a tasklet used for draining */ > > > > > > + tasklet_init(&cable->snd_timer.event_tasklet, > > > > > > + loopback_snd_timer_tasklet, (unsigned > > > long)timeri); > > > > > > > > > > This has to be set before snd_timer_open(). The callback might be > > > > > called immediately after snd_timer_open(). > > > > > > > > This tasklet is used/scheduled only in ccallback (not regular tick > > > > callback), > > > > and only for SNDRV_TIMER_EVENT_MSTOP event. Can this event really > happen > > > > immediately after snd_timer_open()? > > > > > > Why not? The master timer can be stopped at any time, even between > > > these two lines. > > > > > > Beware that there are fuzzer programs that can trigger such racy > > > things, and you're adding the code to the target that is actively > > > slapped by them :) > > > > OK, got it. > > I'll move this initialization to before snd_timer_open() in the next > > update together with the fixes for the other issues you will find > > in this version. > > I have no other issues, so you can just resubmit only that patch, > too. I'm not sure how to correctly format resubmitting of a single patch from a patch set, so I'm submitting the next version v5 of the whole patch set: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-November/158939.h tml Thanks! Best regards, Andrew

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[alsa-devel] [PATCH 1/4] treewide: sys/poll to poll
by Rosen Penev 20 Nov '19

20 Nov '19

Fixes warning on musl: warning redirecting incorrect #include <sys/poll.h> to <poll.h> Signed-off-by: Rosen Penev <rosenp(a)gmail.com> --- amidi/amidi.c | 2 +- amixer/amixer.c | 2 +- aplay/aplay.c | 2 +- seq/aplaymidi/arecordmidi.c | 2 +- seq/aseqdump/aseqdump.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/amidi/amidi.c b/amidi/amidi.c index b6e05e1..c6268e4 100644 --- a/amidi/amidi.c +++ b/amidi/amidi.c @@ -31,7 +31,7 @@ #include <signal.h> #include <sys/timerfd.h> #include <sys/types.h> -#include <sys/poll.h> +#include <poll.h> #include <sys/stat.h> #include <unistd.h> #include <fcntl.h> diff --git a/amixer/amixer.c b/amixer/amixer.c index ad9c482..f7f31f0 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -28,7 +28,7 @@ #include <errno.h> #include <assert.h> #include <alsa/asoundlib.h> -#include <sys/poll.h> +#include <poll.h> #include <stdint.h> #include "amixer.h" #include "../alsamixer/volume_mapping.h" diff --git a/aplay/aplay.c b/aplay/aplay.c index 274bbce..5241068 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -44,7 +44,7 @@ #include <assert.h> #include <termios.h> #include <signal.h> -#include <sys/poll.h> +#include <poll.h> #include <sys/uio.h> #include <sys/time.h> #include <sys/stat.h> diff --git a/seq/aplaymidi/arecordmidi.c b/seq/aplaymidi/arecordmidi.c index f3db65e..604cd0d 100644 --- a/seq/aplaymidi/arecordmidi.c +++ b/seq/aplaymidi/arecordmidi.c @@ -27,7 +27,7 @@ #include <string.h> #include <signal.h> #include <getopt.h> -#include <sys/poll.h> +#include <poll.h> #include <alsa/asoundlib.h> #include "aconfig.h" #include "version.h" diff --git a/seq/aseqdump/aseqdump.c b/seq/aseqdump/aseqdump.c index 7904540..578e06f 100644 --- a/seq/aseqdump/aseqdump.c +++ b/seq/aseqdump/aseqdump.c @@ -25,7 +25,7 @@ #include <string.h> #include <signal.h> #include <getopt.h> -#include <sys/poll.h> +#include <poll.h> #include <alsa/asoundlib.h> #include "aconfig.h" #include "version.h" -- 2.23.0

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[alsa-devel] [PATCH v2 1/3] ASoC: add control components management
by Jaroslav Kysela 20 Nov '19

20 Nov '19

This ASCII string can carry additional information about soundcard components or configuration. Add the possibility to set this string via the ASoC card. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 13 +++++++++++++ 2 files changed, 14 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index e0855dc08d30..bd943b5d7d45 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -982,6 +982,7 @@ struct snd_soc_card { const char *name; const char *long_name; const char *driver_name; + const char *components; char dmi_longname[80]; char topology_shortname[32]; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 55014e7ae0d8..b4683d4588ee 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2121,6 +2121,19 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) soc_setup_card_name(card->snd_card->driver, card->driver_name, card->name, 1); + if (card->components) { + /* the current implementation of snd_component_add() accepts */ + /* multiple components in the string separated by space, */ + /* but the string collision (identical string) check might */ + /* not work correctly */ + ret = snd_component_add(card->snd_card, card->components); + if (ret < 0) { + dev_err(card->dev, "ASoC: %s snd_component_add() failed: %d\n", + card->name, ret); + goto probe_end; + } + } + if (card->late_probe) { ret = card->late_probe(card); if (ret < 0) { -- 2.20.1

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[alsa-devel] [PATCH v5 0/7] ALSA: aloop: Support sound timer as clock source instead of jiffies
by Andrew Gabbasov 20 Nov '19

20 Nov '19

This patch set is an updated version of patches by Timo Wischer: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-March/146871.html This patch set is required for forwarding audio data between a HW sound card and an aloop device without the usage of an asynchronous sample rate converter. Most of sound and timers related code is kept the same as in previous set. The code, related to snd_pcm_link() functionality and its using for timer source setting, is removed (as rejected earlier). The changes in this update are mainly related to the parameters handling and some cleanup. The timer source can be initially selected by "timer_source" kernel module parameter. It is supposed to have the following format: [<pref>:](<card name>|<card idx>)[{.,}<dev idx>[{.,}<subdev idx>]] For example: "hw:I82801AAICH.1.0", or "1.1", or just "I82801AAICH". (Prefix is ignored, just allowed here to be able to use the strings, the user got used to). Although the parsing function recognizes both '.' and ',' as a separator, module parameters handling routines use ',' to separate parameters for different module instances (array elements), so we have to use '.' to separate device and subdevice numbers from the card name or number in module parameters. Empty string indicates using jiffies as a timer source. Besides "static" selection of timer source at module load time, it is possible to dynamically change it via sound "info" interface (using "/proc/asound/<card>/timer_source" file in read-write mode. The contents of this file is used as a timer source string for a particular loopback card, e.g. "hw:0,0,0" (and here ',' can be used as a separator). The timer source string value can be changed at any time, but it is latched by PCM substream open callback "loopback_open()" (the first one for a particular cable). At this point it is actually used, that is the string is parsed, and the timer is looked up and opened. This seems to be a good trade-off between flexibility of updates and synchronizations or racing complexity. The timer source is set for a loopback card (the same as initial setting by module parameter), but every cable uses the value, current at the moment of opening. Theoretically, it's possible to set the timer source for each cable independently (via separate files), but it would be inconsistent with the initial setting via module parameters on a per-card basis. v2: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-November/157961.… v3: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-November/158312.… - Change sound card lookup to use snd_card_ref() and avoid direct access to sound cards array - Squash commits on returning error codes for timer start and stop - Some locking related fixes - Some code cleanup v4: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-November/158896.… - Change to use updated API for snd_timer_open() (separate timer instance) - Change to use snd_timer_close() returning void - Some code cleanup v5: - Change to initialize timer event tasklet before calling snd_timer_open() Andrew Gabbasov (1): ALSA: aloop: Support runtime change of snd_timer via info interface Timo Wischer (6): ALSA: aloop: Describe units of variables ALSA: aloop: Support return of error code for timer start and stop ALSA: aloop: Use callback functions for timer specific implementations ALSA: aloop: Rename all jiffies timer specific functions ALSA: aloop: Move CABLE_VALID_BOTH to the top of file ALSA: aloop: Support selection of snd_timer instead of jiffies sound/drivers/aloop.c | 663 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 628 insertions(+), 35 deletions(-) -- 2.21.0

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[alsa-devel] [PATCH 1/2] ASoC: improve the DMI long card code in asoc-core
by Jaroslav Kysela 20 Nov '19

20 Nov '19

Add append_dmi_string() function and make the code more readable. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 66 +++++++++++++++++--------------------------- 1 file changed, 25 insertions(+), 41 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b4683d4588ee..1d1baf78818c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1755,6 +1755,23 @@ static int is_dmi_valid(const char *field) return 1; } +/* + * + */ +static void append_dmi_string(struct snd_soc_card *card, const char *str) +{ + char *dst = card->dmi_longname; + size_t dst_len = sizeof(card->dmi_longname); + size_t len; + + len = strlen(dst); + snprintf(dst + len, dst_len - len, "-%s", str); + + len++; /* skip the separator "-" */ + if (len < dst_len) + cleanup_dmi_name(dst + len); +} + /** * snd_soc_set_dmi_name() - Register DMI names to card * @card: The card to register DMI names @@ -1789,61 +1806,36 @@ static int is_dmi_valid(const char *field) int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) { const char *vendor, *product, *product_version, *board; - size_t longname_buf_size = sizeof(card->snd_card->longname); - size_t len; if (card->long_name) return 0; /* long name already set by driver or from DMI */ - /* make up dmi long name as: vendor.product.version.board */ + /* make up dmi long name as: vendor-product-version-board */ vendor = dmi_get_system_info(DMI_BOARD_VENDOR); if (!vendor || !is_dmi_valid(vendor)) { dev_warn(card->dev, "ASoC: no DMI vendor name!\n"); return 0; } - snprintf(card->dmi_longname, sizeof(card->snd_card->longname), - "%s", vendor); + snprintf(card->dmi_longname, sizeof(card->dmi_longname), "%s", vendor); cleanup_dmi_name(card->dmi_longname); product = dmi_get_system_info(DMI_PRODUCT_NAME); if (product && is_dmi_valid(product)) { - len = strlen(card->dmi_longname); - snprintf(card->dmi_longname + len, - longname_buf_size - len, - "-%s", product); - - len++; /* skip the separator "-" */ - if (len < longname_buf_size) - cleanup_dmi_name(card->dmi_longname + len); + append_dmi_string(card, product); /* * some vendors like Lenovo may only put a self-explanatory * name in the product version field */ product_version = dmi_get_system_info(DMI_PRODUCT_VERSION); - if (product_version && is_dmi_valid(product_version)) { - len = strlen(card->dmi_longname); - snprintf(card->dmi_longname + len, - longname_buf_size - len, - "-%s", product_version); - - len++; - if (len < longname_buf_size) - cleanup_dmi_name(card->dmi_longname + len); - } + if (product_version && is_dmi_valid(product_version)) + append_dmi_string(card, product_version); } board = dmi_get_system_info(DMI_BOARD_NAME); if (board && is_dmi_valid(board)) { - len = strlen(card->dmi_longname); - snprintf(card->dmi_longname + len, - longname_buf_size - len, - "-%s", board); - - len++; - if (len < longname_buf_size) - cleanup_dmi_name(card->dmi_longname + len); + append_dmi_string(card, board); } else if (!product) { /* fall back to using legacy name */ dev_warn(card->dev, "ASoC: no DMI board/product name!\n"); @@ -1851,16 +1843,8 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) } /* Add flavour to dmi long name */ - if (flavour) { - len = strlen(card->dmi_longname); - snprintf(card->dmi_longname + len, - longname_buf_size - len, - "-%s", flavour); - - len++; - if (len < longname_buf_size) - cleanup_dmi_name(card->dmi_longname + len); - } + if (flavour) + append_dmi_string(card, flavour); /* set the card long name */ card->long_name = card->dmi_longname; -- 2.20.1

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[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt5640: Update quirk for Acer Switch 10 SW5-012 2-in-1
by Hans de Goede 20 Nov '19

20 Nov '19

When the Acer Switch 10 SW5-012 quirk was added we did not have jack-detection support yet; and the builtin microphone selection of the original quirk is wrong too. Fix the microphone-input quirk and add jack-detection info so that the internal-microphone and headphone/set jack on the Switch 10 work properly. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 9c1aa4ec9cba..dd2b5ad08659 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -405,10 +405,12 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Acer"), DMI_MATCH(DMI_PRODUCT_NAME, "Aspire SW5-012"), }, - .driver_data = (void *)(BYT_RT5640_IN1_MAP | - BYT_RT5640_MCLK_EN | - BYT_RT5640_SSP0_AIF1), - + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), }, { .matches = { -- 2.23.0

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[alsa-devel] [PATCH 1/2] ASoC: add control components management
by Jaroslav Kysela 20 Nov '19

20 Nov '19

This ASCII string can carry additional information about soundcard components or configuration. Add the possibility to set this string via the ASoC card. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 13 +++++++++++++ 2 files changed, 14 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index e0855dc08d30..bd943b5d7d45 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -982,6 +982,7 @@ struct snd_soc_card { const char *name; const char *long_name; const char *driver_name; + const char *components; char dmi_longname[80]; char topology_shortname[32]; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 55014e7ae0d8..b4683d4588ee 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2121,6 +2121,19 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) soc_setup_card_name(card->snd_card->driver, card->driver_name, card->name, 1); + if (card->components) { + /* the current implementation of snd_component_add() accepts */ + /* multiple components in the string separated by space, */ + /* but the string collision (identical string) check might */ + /* not work correctly */ + ret = snd_component_add(card->snd_card, card->components); + if (ret < 0) { + dev_err(card->dev, "ASoC: %s snd_component_add() failed: %d\n", + card->name, ret); + goto probe_end; + } + } + if (card->late_probe) { ret = card->late_probe(card); if (ret < 0) { -- 2.20.1

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[alsa-devel] [PATCH 0/2] ASoC: pcm3168a: Update the reset GPIO handling
by Peter Ujfalusi 20 Nov '19

20 Nov '19

Hi, Switch the rst-gpios to generic reset-gpios as suggested by Rob. Based on other threads around GPIOs, revise the GPIO active vs assert/deassert configuration: The RST pin is a reset pin and it is active low. Regards, Peter --- Peter Ujfalusi (2): ASoC: dt-bindings: pcm3168a: Update the optional RST gpio for clarity ASoC: pcm3168a: Update the RST gpio handling to align with documentation .../devicetree/bindings/sound/ti,pcm3168a.txt | 9 +++++---- sound/soc/codecs/pcm3168a.c | 20 ++++++++++++++----- 2 files changed, 20 insertions(+), 9 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] Revert "ASoC: soc-pcm: check symmetry after hw_params"
by Shengjiu Wang 20 Nov '19

20 Nov '19

This reverts commit 957ce0c6b8a1f26559864507ae0bfcba29d924ad. That commit cause soc_pcm_params_symmetry can't take effect. cpu_dai->rate, cpu_dai->channels and cpu_dai->sample_bits are updated in the middle of soc_pcm_hw_params, so move soc_pcm_params_symmetry to the end of soc_pcm_hw_params is not a good solution, for judgement of symmetry in the function is always true. FIXME: According to the comments of that commit, I think the case described in the commit should disable symmetric_rates in Back-End, rather than changing the position of soc_pcm_params_symmetry. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/soc-pcm.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8655df6a6089..b7800c95327a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -861,6 +861,11 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, int i, ret = 0; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + + ret = soc_pcm_params_symmetry(substream, params); + if (ret) + goto out; + if (rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { @@ -940,9 +945,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } component = NULL; - ret = soc_pcm_params_symmetry(substream, params); - if (ret) - goto component_err; out: mutex_unlock(&rtd->card->pcm_mutex); return ret; -- 2.21.0

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