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

[alsa-devel] [PATCH V2] ASoC: max98373: Added max98373_reset for stable amp reset
by Ryan Lee 30 Nov '18

30 Nov '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes since v1 : Removed unusual repeat for amp software reset and verification. Amp software reset will be performed once and it repeats verification maximum 3 times if it is failed. Wait 10ms before every verification trial. Maximum 30ms delay will be applied to wait AMP idle state. Changes : Created max98373_reset function to minimize code duplication. Changed regmap_write to regmap_update_bits. Other bits except LSB need to be masked. Added reset verification step to make sure software reset is completed well. Software reset is done in 10ms in normal case. Revision ID is available when the amp is in the idle state which means software reset is completed. Software reset will be performed maximum 3 times to avoid amp reset failure. Generally it is done in the first trial. sleep time after software reset is increased + 30ms for every retrial. Maximum possible msleep time is 100 ms (initial 10 ms + 30 ms * 3 times). sound/soc/codecs/max98373.c | 35 +++++++++++++++++++++++++++++------ 1 file changed, 29 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a09d013..9c8616a 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -724,14 +724,39 @@ static struct snd_soc_dai_driver max98373_dai[] = { } }; +static void max98373_reset(struct max98373_priv *max98373, struct device *dev) +{ + int ret, reg, count; + + /* Software Reset */ + ret = regmap_update_bits(max98373->regmap, + MAX98373_R2000_SW_RESET, + MAX98373_SOFT_RESET, + MAX98373_SOFT_RESET); + if (ret) + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + + count = 0; + while (count < 3) { + usleep_range(10000, 11000); + /* Software Reset Verification */ + ret = regmap_read(max98373->regmap, + MAX98373_R21FF_REV_ID, &reg); + if (!ret) { + dev_info(dev, "Reset completed (retry:%d)\n", count); + return; + } + count++; + } + dev_err(dev, "Reset failed. (ret:%d)\n", ret); +} + static int max98373_probe(struct snd_soc_component *component) { struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); /* Software Reset */ - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, component->dev); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -818,9 +843,7 @@ static int max98373_resume(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, dev); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; -- 2.7.4

3 6

[alsa-devel] [PATCH 0/4] ASoC: merge audio-graph-scu into audio-graph - step3
by Kuninori Morimoto 30 Nov '18

30 Nov '18

Hi Mark audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. So, this is the time to merge audio-graph and audio-graph-scu. The patch-set are a little bit huge (and few patches are depend on posted OF patch. I'm now waiting...), thus, I separated these into few steps. This is step3 patches. After step3, the remaining patch is "merge audio-graph and audio-graph-scu" and "remove audio-graph-scu" patches, and, +1 more patch which is based on OF graph patches. I'm posting it to OF ML, but no response so far. Thus, step4 patch will be *after* OF graph patch. Kuninori Morimoto (4): ASoC: audio-graph-scu-card: care link / dai count ASoC: audio-graph-scu-card: use cpu/codec pointer on graph_dai_props ASoC: audio-graph-scu-card: care multi DPCM codec_conf ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props sound/soc/generic/audio-graph-card.c | 57 ++++++----- sound/soc/generic/audio-graph-scu-card.c | 163 ++++++++++++++++++++++--------- 2 files changed, 153 insertions(+), 67 deletions(-) -- 2.7.4

2 7

[alsa-devel] [PATCH] ALSA: pcm: Fix starvation on down_write_nonblock()
by Chanho Min 30 Nov '18

30 Nov '18

Commit 67ec1072b053 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream") fixes deadlock for non-atomic PCM stream. But, This patch causes antother stuck. If writer is RT thread and reader is a normal thread, the reader thread will be difficult to get scheduled. It may not give chance to release readlocks and writer gets stuck for a long time if they are pinned to single cpu. The deadlock described in the previous commit is because the linux rwsem queues like a FIFO. So, we might need non-FIFO writelock, not non-block one. My suggestion is that the writer gives reader a chance to be scheduled by using the minimum msleep() instaed of spinning without blocking by writer. Also, The *_nonblock may be changed to *_nonfifo appropriately to this concept. In terms of performance, when trylock is failed, this minimum periodic msleep will have the same performance as the tick-based schedule()/wake_up_q(). Suggested-by: Wonmin Jung <wonmin.jung(a)lge.com> Signed-off-by: Chanho Min <chanho.min(a)lge.com> --- sound/core/pcm_native.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 66c90f4..bdca0e1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -36,6 +36,7 @@ #include <sound/timer.h> #include <sound/minors.h> #include <linux/uio.h> +#include <linux/delay.h> #include "pcm_local.h" @@ -91,12 +92,12 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem); * and this may lead to a deadlock when the code path takes read sem * twice (e.g. one in snd_pcm_action_nonatomic() and another in * snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to - * spin until it gets the lock. + * sleep until all the readers are completed without blocking by writer. */ -static inline void down_write_nonblock(struct rw_semaphore *lock) +static inline void down_write_nonfifo(struct rw_semaphore *lock) { while (!down_write_trylock(lock)) - cond_resched(); + msleep(1); } #define PCM_LOCK_DEFAULT 0 @@ -1967,7 +1968,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) res = -ENOMEM; goto _nolock; } - down_write_nonblock(&snd_pcm_link_rwsem); + down_write_nonfifo(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN || substream->runtime->status->state != substream1->runtime->status->state || @@ -2014,7 +2015,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) struct snd_pcm_substream *s; int res = 0; - down_write_nonblock(&snd_pcm_link_rwsem); + down_write_nonfifo(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (!snd_pcm_stream_linked(substream)) { res = -EALREADY; -- 2.1.4

2 7

[alsa-devel] [PATCH] ALSA: pcm: Fix interval evaluation with openmin/max
by Takashi Iwai 29 Nov '18

29 Nov '18

As addressed in alsa-lib (commit b420056604f0), we need to fix the case where the evaluation of PCM interval "(x x+1]" leading to -EINVAL. After applying rules, such an interval may be translated as "(x x+1)". Fixes: ff2d6acdf6f1 ("ALSA: pcm: Fix snd_interval_refine first/last with open min/max") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm_params.h | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 2dd37cada7c0..888a833d3b00 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -254,11 +254,13 @@ static inline int snd_interval_empty(const struct snd_interval *i) static inline int snd_interval_single(const struct snd_interval *i) { return (i->min == i->max || - (i->min + 1 == i->max && i->openmax)); + (i->min + 1 == i->max && (i->openmin || i->openmax))); } static inline int snd_interval_value(const struct snd_interval *i) { + if (i->openmin && !i->openmax) + return i->max; return i->min; } -- 2.19.1

1 0

[alsa-devel] [PATCH v2 0/6] Introduce audio-mute LED trigger (and conversions to it)
by Takashi Iwai 29 Nov '18

29 Nov '18

Hi, this is revised patch series for adding a new LED trigger audio-mute and audio-micmute, and converting the HD-audio driver and the platform drivers to use the LED trigger instead of the ugly direct dynamic symbol binding. v2 series addressed a couple of issues Andy suggested, and changed the led class dev names to "platform::xxx" as discussed in the previous thread. The latest version of patches are found in topic/leds-trigger branch in my sound git tree. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git As these are cross-tree patches, the branch above is based cleanly on v4.20-rc3. From now on, it'll stay as an immutable branch, so feel free to merge into other trees. thanks, Takashi === Takashi Iwai (6): leds: trigger: Introduce audio mute LED trigger platform/x86: dell-laptop: Add micmute LED trigger support platform/x86: thinkpad_acpi: Add audio mute LED classdev support ALSA: hda - Support led audio trigger platform/x86: dell-laptop: Drop superfluous exported function platform/x86: thinkpad_acpi: Drop superfluous exported function drivers/leds/trigger/Kconfig | 7 +++ drivers/leds/trigger/Makefile | 1 + drivers/leds/trigger/ledtrig-audio.c | 44 ++++++++++++++++++ drivers/platform/x86/Kconfig | 4 ++ drivers/platform/x86/dell-laptop.c | 27 ++++++++--- drivers/platform/x86/thinkpad_acpi.c | 67 +++++++++++++++++++++------- include/linux/dell-led.h | 7 --- include/linux/leds.h | 20 +++++++++ include/linux/thinkpad_acpi.h | 16 ------- sound/pci/hda/dell_wmi_helper.c | 48 -------------------- sound/pci/hda/hda_generic.c | 31 +++++++++++++ sound/pci/hda/hda_generic.h | 2 + sound/pci/hda/patch_realtek.c | 17 +++---- sound/pci/hda/thinkpad_helper.c | 43 +++--------------- 14 files changed, 194 insertions(+), 140 deletions(-) create mode 100644 drivers/leds/trigger/ledtrig-audio.c delete mode 100644 include/linux/dell-led.h delete mode 100644 include/linux/thinkpad_acpi.h delete mode 100644 sound/pci/hda/dell_wmi_helper.c -- 2.19.1

1 7

[alsa-devel] [PATCH] ASoc: au8540: use 64-bit arithmetic instead of 32-bit
by Yang Xiao 29 Nov '18

29 Nov '18

From: Young_X <YangX92(a)hotmail.com> Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: 256 * fs * 2 * mclk_src_scaling[i].param Signed-off-by: Young_X <YangX92(a)hotmail.com> --- sound/soc/codecs/nau8540.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index e3c8cd1..4dd1a60 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -585,7 +585,7 @@ static int nau8540_calc_fll_param(unsigned int fll_in, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: adau1761: Ensure DSP_RUN and DSP_ENABLE are disabled
by niklas.morberg＠axis.com 29 Nov '18

29 Nov '18

From: Niklas Carlsson <niklasc(a)axis.com> DSP_RUN and DSP_ENABLE needs to be disabled during FW load. This is not a problem after power-cycling but for soft reboots. Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/adau1761.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..3075c13bcbbc 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -460,8 +460,35 @@ static int adau1761_set_bias_level(struct snd_soc_component *component, regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN); - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + /* + * When going OFF -> STANDBY, the device can be in two + * states: + * 1) Power-cycled and reset + * 2) Soft reboot + * In the case of soft reboot, we need to sync the HW + * registers even if our regmap reports default values. + * Marking the cache as dirty ensures that both cases + * are handled. + */ + regcache_mark_dirty(adau->regmap); regcache_sync(adau->regmap); + /* + * In order for FW to load correctly, the device needs + * DSP_RUN and DSP_ENABLE to be 0. This might not be the + * case for soft reboots. Ensure that DSP_RUN and + * DSP_ENABLE is 0 by bypassing the cache and write + * directly to HW when going OFF -> STANDBY. + */ + if (adau17x1_has_dsp(adau)) { + regcache_cache_bypass(adau->regmap, true); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0); + regmap_write(adau->regmap, + ADAU17X1_DSP_ENABLE, 0); + regcache_cache_bypass(adau->regmap, false); + } + } break; case SND_SOC_BIAS_OFF: regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, -- 2.11.0

2 2

[alsa-devel] [PATCH] ALSA: hda: Add support for AMD Stoney Ridge
by Kai-Heng Feng 29 Nov '18

29 Nov '18

It's similar to other AMD audio devices, it also supports D3, which can save some power drain. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d8eb2b5f51ae..3f5659064585 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2496,6 +2496,10 @@ static const struct pci_device_id azx_ids[] = { /* AMD Hudson */ { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + /* AMD Stoney */ + { PCI_DEVICE(0x1022, 0x157a), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | + AZX_DCAPS_PM_RUNTIME }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | -- 2.17.1

2 1

[alsa-devel] User-friendly dynamic HDA pin reconfiguration
by Daniel Drake 29 Nov '18

29 Nov '18

Hi, We are working with a new consumer All-in-One platform with HDA audio (Realtek codec). Upon plugging in an audio device into one of the audio jacks, under Windows, a prompt appears which says: Which device did you plug in? And you can select from several options: Mic In / Headphone / Speaker Out Screenshot: https://imgur.com/a/mrUolHN The system vendor is asking if we can implement the same under Linux. At first glance, this sounds quite similar to what happens on other models when you plug in a headset under GNOME: a dialog pops up asking if you connected a mic, headphones, or a headset. Under the surface, such headset jacks seems to correspond to 2 separate HDA pins (one input and one output) and this dialog seems to manage which streams are enabled or muted. However, this case is different. We have experimented, and the jack corresponds to just a single HDA pin. We can manually quirk or retask this pin to be an input or an output and it works in the direction that we choose. But that's a far cry from having a nice GUI that automatically comes up asking the user how the pin should be configured. Is there any current solution or work in progress with the goal of offering this functionality? If not, brainstorming, possible solutions could include: 1. Taking the hdajackretask idea but making it user-accessible and user-friendly (currently it goes through debugfs and must be done by root). This would require the creation of a new kernel interface to expose pin reconfiguration in a nicer way, which would have to be plumbed through pulseaudio and GNOME. 2. Could ALSA ignore the pin config here and provide both input and output streams for this pin? Then pulseaudio + GUI could pick which one to use - similar to the existing Headset dialog. This would still require some work on pulseaudio and the corresponding UI, but it might be a less invasive project. Thanks, Daniel

2 2

[alsa-devel] [PATCH] ucm: Set default include path
by Takashi Iwai 29 Nov '18

29 Nov '18

Many UCM profiles include the UCM profile components under ucm/* subdirectories and thusly put <searchdir:ucm> at each place. This is rather cumbersome. This patch makes the UCM parser to set the default include path, so that each profile no longer needs to set searchdir. All the <searchdir:ucm> lines currently found in the profiles are removed gracefully, too. For the needed implementation, a new helper, _snd_config_load_with_include() is introduced. It's not exported, only for the use inside alsa-lib. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/local.h | 3 ++ src/conf.c | 32 +++++++++++++++++-- .../HiFi.conf | 1 - .../bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf | 1 - .../bytcr-rt5640-mono-spk-in1-mic/HiFi.conf | 1 - .../bytcr-rt5640-mono-spk-in3-mic/HiFi.conf | 1 - .../HiFi.conf | 1 - .../bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf | 1 - .../bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf | 1 - src/conf/ucm/bytcr-rt5640/HiFi.conf | 1 - .../bytcr-rt5651-mono-spk-in1-mic/HiFi.conf | 1 - .../HiFi.conf | 1 - .../bytcr-rt5651-mono-spk-in2-mic/HiFi.conf | 1 - .../bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf | 1 - .../HiFi.conf | 1 - .../bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf | 1 - src/conf/ucm/bytcr-rt5651/HiFi.conf | 1 - src/conf/ucm/chtnau8824/HiFi.conf | 1 - src/conf/ucm/chtrt5645/HiFi.conf | 1 - .../HiFi.conf | 1 - src/ucm/parser.c | 3 -- src/ucm/ucm_local.h | 3 ++ src/ucm/utils.c | 7 +++- 23 files changed, 41 insertions(+), 25 deletions(-) diff --git a/include/local.h b/include/local.h index b5e7b6f00ff5..03d8f0ccdc58 100644 --- a/include/local.h +++ b/include/local.h @@ -354,6 +354,9 @@ int snd_config_search_alias_hooks(snd_config_t *config, int _snd_conf_generic_id(const char *id); +int _snd_config_load_with_include(snd_config_t *config, snd_input_t *in, + const char *default_include_path); + /* convenience macros */ #define ARRAY_SIZE(x) (sizeof(x) / sizeof(x[0])) diff --git a/src/conf.c b/src/conf.c index 254c485868b5..1f9bad7b3a97 100644 --- a/src/conf.c +++ b/src/conf.c @@ -1832,7 +1832,8 @@ int snd_config_top(snd_config_t **config) return _snd_config_make(config, 0, SND_CONFIG_TYPE_COMPOUND); } -static int snd_config_load1(snd_config_t *config, snd_input_t *in, int override) +static int snd_config_load1(snd_config_t *config, snd_input_t *in, int override, + char *default_include_path) { int err; input_t input; @@ -1847,6 +1848,11 @@ static int snd_config_load1(snd_config_t *config, snd_input_t *in, int override) fd->column = 0; fd->next = NULL; INIT_LIST_HEAD(&fd->include_paths); + if (default_include_path) { + err = add_include_path(fd, default_include_path); + if (err < 0) + goto _end; + } input.current = fd; input.unget = 0; err = parse_defs(config, &input, 0, override); @@ -1915,9 +1921,29 @@ static int snd_config_load1(snd_config_t *config, snd_input_t *in, int override) */ int snd_config_load(snd_config_t *config, snd_input_t *in) { - return snd_config_load1(config, in, 0); + return snd_config_load1(config, in, 0, NULL); } +#ifndef DOC_HIDDEN +/* load config with the default include path; used internally for UCM parser */ +int _snd_config_load_with_include(snd_config_t *config, snd_input_t *in, + const char *default_include_path) +{ + int err; + char *s = NULL; + + if (default_include_path) { + s = strdup(default_include_path); + if (!s) + return -ENOMEM; + } + err = snd_config_load1(config, in, 0, s); + if (err < 0) + free(s); + return err; +} +#endif + /** * \brief Loads a configuration tree and overrides existing configuration nodes. * \param config Handle to a top level configuration node. @@ -1930,7 +1956,7 @@ int snd_config_load(snd_config_t *config, snd_input_t *in) */ int snd_config_load_override(snd_config_t *config, snd_input_t *in) { - return snd_config_load1(config, in, 1); + return snd_config_load1(config, in, 1, NULL); } /** diff --git a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf index facc73ac3b2b..b931b7914ce7 100644 --- a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf @@ -1,4 +1,3 @@ -<searchdir:ucm> SectionVerb { diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf index c78cdfb80a44..0e21d6e1abf9 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf index 34a5d53d363c..d8ca499a5799 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf index 0aa0c0266412..e4fb28df546f 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf index e358d4f23f12..bb31bf531633 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf index 097560d02d12..9474a64e84ff 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf index 8a018f63c9c6..478130e67ae4 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640/HiFi.conf b/src/conf/ucm/bytcr-rt5640/HiFi.conf index 20ebe2dd36e2..54dd2517dbd7 100644 --- a/src/conf/ucm/bytcr-rt5640/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf index ca8a7d9078fb..b93e0be33cbd 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf index 60ef6b332ea2..ef048626732e 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf index e75210d4aa7e..c324c14ad542 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf index 2b7675bce831..aa1a1cde7856 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf index 7a8c94bc5d6b..343aed211bb4 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf index 45c7837c6fb9..6f6d74cf2e68 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651/HiFi.conf b/src/conf/ucm/bytcr-rt5651/HiFi.conf index d3928e9b9976..fddd8a3ca19a 100644 --- a/src/conf/ucm/bytcr-rt5651/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/chtnau8824/HiFi.conf b/src/conf/ucm/chtnau8824/HiFi.conf index 97a7ac7ba366..845cbcaffe5f 100644 --- a/src/conf/ucm/chtnau8824/HiFi.conf +++ b/src/conf/ucm/chtnau8824/HiFi.conf @@ -1,4 +1,3 @@ -<searchdir:ucm> SectionVerb { diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf index 6a3c6875f301..422f3b7eef4f 100644 --- a/src/conf/ucm/chtrt5645/HiFi.conf +++ b/src/conf/ucm/chtrt5645/HiFi.conf @@ -1,4 +1,3 @@ -<searchdir:ucm> SectionVerb { # ALSA PCM diff --git a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf index facc73ac3b2b..b931b7914ce7 100644 --- a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf @@ -1,4 +1,3 @@ -<searchdir:ucm> SectionVerb { diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 2b6f1159aa7d..5e1a8862fac1 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -34,9 +34,6 @@ #include <dirent.h> #include <limits.h> -/** The name of the environment variable containing the UCM directory */ -#define ALSA_CONFIG_UCM_VAR "ALSA_CONFIG_UCM" - /* Directories to store UCM configuration files for components, like * off-soc codecs or embedded DSPs. Components can define their own * devices and sequences, to be reused by sound cards/machines. UCM diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index ce2fc6ea18eb..8e7ac7ea48e9 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -243,3 +243,6 @@ void uc_mgr_free_sequence_element(struct sequence_element *seq); void uc_mgr_free_transition_element(struct transition_sequence *seq); void uc_mgr_free_verb(snd_use_case_mgr_t *uc_mgr); void uc_mgr_free(snd_use_case_mgr_t *uc_mgr); + +/** The name of the environment variable containing the UCM directory */ +#define ALSA_CONFIG_UCM_VAR "ALSA_CONFIG_UCM" diff --git a/src/ucm/utils.c b/src/ucm/utils.c index 18785488db07..ea5ac5dd1b01 100644 --- a/src/ucm/utils.c +++ b/src/ucm/utils.c @@ -54,6 +54,7 @@ int uc_mgr_config_load(const char *file, snd_config_t **cfg) FILE *fp; snd_input_t *in; snd_config_t *top; + const char *default_path; int err; fp = fopen(file, "r"); @@ -70,7 +71,11 @@ int uc_mgr_config_load(const char *file, snd_config_t **cfg) err = snd_config_top(&top); if (err < 0) return err; - err = snd_config_load(top, in); + + default_path = getenv(ALSA_CONFIG_UCM_VAR); + if (!default_path || !*default_path) + default_path = ALSA_CONFIG_DIR "/ucm"; + err = _snd_config_load_with_include(top, in, default_path); if (err < 0) { uc_error("could not load configuration file %s", file); snd_config_delete(top); -- 2.19.1

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