September 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] conf: Allow for a directory to be given as a config file.
by gmane＠colin.guthr.ie 20 Sep '11

20 Sep '11

From: Colin Guthrie <colin(a)mageia.org> When this is done, *.conf files can be placed in that directory and they will be processed by as if they were included directly. A directory (typically /usr/share/alsa/alsa.conf.d/) has been added into the distribution. v2: Used existing conf syntax rather than processing via autotools v3: Split file loading into separate function and made error handling more consistent. Signed-off-by: Colin Guthrie <colin(a)mageia.org> --- configure.in | 3 +- src/conf.c | 78 ++++++++++++++++++++++++++++++++----- src/conf/Makefile.am | 2 +- src/conf/alsa.conf | 7 +++ src/conf/alsa.conf.d/Makefile.am | 8 ++++ src/conf/alsa.conf.d/README | 2 + 6 files changed, 87 insertions(+), 13 deletions(-) create mode 100644 src/conf/alsa.conf.d/Makefile.am create mode 100644 src/conf/alsa.conf.d/README diff --git a/configure.in b/configure.in index 7ee0ccc..13e38b8 100644 --- a/configure.in +++ b/configure.in @@ -616,7 +616,8 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/pcm/Makefile src/pcm/scopes/Makefile \ src/rawmidi/Makefile src/timer/Makefile \ src/hwdep/Makefile src/seq/Makefile src/ucm/Makefile \ - src/compat/Makefile src/alisp/Makefile src/conf/Makefile \ + src/compat/Makefile src/alisp/Makefile \ + src/conf/Makefile src/conf/alsa.conf.d/Makefile \ src/conf/cards/Makefile \ src/conf/pcm/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ diff --git a/src/conf.c b/src/conf.c index ddefff6..5b1b5a6 100644 --- a/src/conf.c +++ b/src/conf.c @@ -417,6 +417,7 @@ beginning:</P> #include <stdarg.h> #include <limits.h> #include <sys/stat.h> +#include <dirent.h> #include <locale.h> #include "local.h" #ifdef HAVE_LIBPTHREAD @@ -3373,6 +3374,42 @@ static int snd_config_hooks(snd_config_t *config, snd_config_t *private_data) return err; } +static int config_filename_filter(const struct dirent *dirent) +{ + size_t flen; + + if (dirent == NULL) + return 0; + if (dirent->d_type == DT_DIR) + return 0; + + flen = strlen(dirent->d_name); + if (flen <= 5) + return 0; + + if (strncmp(&dirent->d_name[flen-5], ".conf", 5) == 0) + return 1; + + return 0; +} + +static int config_file_open(snd_config_t *root, const char *filename) +{ + snd_input_t *in; + int err; + + err = snd_input_stdio_open(&in, filename, "r"); + if (err >= 0) { + err = snd_config_load(root, in); + snd_input_close(in); + if (err < 0) + SNDERR("%s may be old or corrupted: consider to remove or fix it", filename); + } else + SNDERR("cannot access file %s", filename); + + return err; +} + /** * \brief Loads and parses the given configurations files. * \param[in] root Handle to the root configuration node. @@ -3457,20 +3494,39 @@ int snd_config_hook_load(snd_config_t *root, snd_config_t *config, snd_config_t } } while (hit); for (idx = 0; idx < fi_count; idx++) { - snd_input_t *in; + struct stat st; if (!errors && access(fi[idx].name, R_OK) < 0) continue; - err = snd_input_stdio_open(&in, fi[idx].name, "r"); - if (err >= 0) { - err = snd_config_load(root, in); - snd_input_close(in); - if (err < 0) { - SNDERR("%s may be old or corrupted: consider to remove or fix it", fi[idx].name); - goto _err; - } - } else { - SNDERR("cannot access file %s", fi[idx].name); + if (stat(fi[idx].name, &st) < 0) { + SNDERR("cannot stat file/directory %s", fi[idx].name); + continue; } + if (S_ISDIR(st.st_mode)) { + struct dirent **namelist; + int n; + + n = scandir(fi[idx].name, &namelist, config_filename_filter, versionsort); + if (n > 0) { + int j; + err = 0; + for (j = 0; j < n; ++j) { + if (err >= 0) { + int sl = strlen(fi[idx].name) + strlen(namelist[j]->d_name) + 2; + char *filename = malloc(sl); + snprintf(filename, sl, "%s/%s", fi[idx].name, namelist[j]->d_name); + filename[sl-1] = '\0'; + + err = config_file_open(root, filename); + free(filename); + } + free(namelist[j]); + } + free(namelist); + if (err < 0) + goto _err; + } + } else if (config_file_open(root, fi[idx].name) < 0) + goto _err; } *dst = NULL; err = 0; diff --git a/src/conf/Makefile.am b/src/conf/Makefile.am index 2e5d0bf..456454f 100644 --- a/src/conf/Makefile.am +++ b/src/conf/Makefile.am @@ -1,4 +1,4 @@ -SUBDIRS=cards pcm +SUBDIRS=cards pcm alsa.conf.d cfg_files = alsa.conf if BUILD_ALISP diff --git a/src/conf/alsa.conf b/src/conf/alsa.conf index a33c24e..bc91df3 100644 --- a/src/conf/alsa.conf +++ b/src/conf/alsa.conf @@ -8,6 +8,13 @@ { func load files [ + { + @func concat + strings [ + { @func datadir } + "/alsa.conf.d/" + ] + } "/etc/asound.conf" "~/.asoundrc" ] diff --git a/src/conf/alsa.conf.d/Makefile.am b/src/conf/alsa.conf.d/Makefile.am new file mode 100644 index 0000000..c91661e --- /dev/null +++ b/src/conf/alsa.conf.d/Makefile.am @@ -0,0 +1,8 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +alsadir = $(alsaconfigdir)/alsa.conf.d +cfg_files = README + +alsa_DATA = $(cfg_files) + +EXTRA_DIST = \ + $(cfg_files) diff --git a/src/conf/alsa.conf.d/README b/src/conf/alsa.conf.d/README new file mode 100644 index 0000000..9997884 --- /dev/null +++ b/src/conf/alsa.conf.d/README @@ -0,0 +1,2 @@ +You can place files named *.conf in this folder and they will be processed +when initialising alsa-lib. -- 1.7.6

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[alsa-devel] [PATCH] ALSA: HDA: Add support for IDT 92HD93
by David Henningsson 20 Sep '11

20 Sep '11

Cc: stable(a)kernel.org BugLink: http://bugs.launchpad.net/bugs/854468 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 5145b66..7d8efab 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -6573,6 +6573,7 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx }, { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx }, { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx}, -- 1.7.5.4

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[alsa-devel] [PATCH 1/1] ymfpci - enable rear device and dup4ch control when ac97 support 4 channels
by Raymond Yau 20 Sep '11

20 Sep '11

ymfpci - enable rear device and dup4ch control when ac97 support 4 channels

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Re: [alsa-devel] hda- Disable Sticky PCM
by Takashi Iwai 20 Sep '11

20 Sep '11

At Tue, 20 Sep 2011 08:15:09 +0800, Raymond Yau wrote: > > 2011/9/18 Takashi Iwai <tiwai(a)suse.de>: > > > > Well, I can't reproduce it in hda-emu, so I don't know now. > > Try to get a stacktrace to see what's calling there actually. > > > > > set stream active : active stream 1 > ALSA hda_intel.c:1757: azx_pcm_prepare: bufsize=0xd750, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x4, stream=0x5, > channel=0, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x3, stream=0x5, > channel=0, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x6, stream=0x5, > channel=0, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x5, stream=0x5, > channel=0, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0xa, stream=0x5, > channel=0, format=0x4111 > hda-intel: IRQ timing workaround is activated for card #1. Suggest a > bigger bdl_pos_adj. > ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x4 > ------------[ cut here ]------------ (snip) Thanks, so it's called from hw_free. This is a call just to be sure, and should be harmless. If you mind, the verbs are suppressed by a patch like below. Takashi --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6b611d5..e3db196 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1491,8 +1491,11 @@ static void really_cleanup_stream(struct hda_codec *codec, struct hda_cvt_setup *q) { hda_nid_t nid = q->nid; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); + if (q->stream_tag || q->channel_id) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); + if (q->format_id) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0 +); memset(q, 0, sizeof(*q)); q->nid = nid; }

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[alsa-devel] [PATCH] ASoC: Fix unused variable warning in WM8996
by Mark Brown 19 Sep '11

19 Sep '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index f455997..bede1ac 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -670,8 +670,6 @@ SOC_SINGLE_TLV("DSP2 EQ B5 Volume", WM8996_DSP2_RX_EQ_GAINS_2, 6, 31, 0, static int bg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; - struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); int ret = 0; switch (event) { -- 1.7.6.3

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[PATCH 2/2] [ASoC] 88pm860x-codec - reset the codec correctly
by Bas Vermeulen 19 Sep '11

19 Sep '11

Reset the codec according to the Audio power-up delay errata for the 88PM8607. Signed-off-by: Bas Vermeulen <bas.vermeulen(a)novero.com> --- sound/soc/codecs/88pm860x-codec.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 0198dbb..df7b4a0 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1179,6 +1179,9 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ + data = AUDIO_PLL | AUDIO_SECTION_ON; + pm860x_reg_write(codec->control_data, REG_MISC2, data); + udelay(300); data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; pm860x_reg_write(codec->control_data, REG_MISC2, data); -- 1.7.0.4 --------------080107070107020807050104--

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[PATCH 1/2] [ASoC] 88pm860x-codec - Allow independent use of both I2S playback and capture
by Bas Vermeulen 19 Sep '11

19 Sep '11

Introduce a I2S CLK supply so playback and capture can operate independently. Signed-off-by: Bas Vermeulen <bas.vermeulen(a)novero.com> --- sound/soc/codecs/88pm860x-codec.c | 10 ++++++++-- 1 files changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 1924157..0198dbb 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -772,11 +772,12 @@ static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0, - PM860X_DAC_EN_2, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0, - PM860X_DAC_EN_2, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0, PM860X_I2S_IFACE_3, 5, 1), + SND_SOC_DAPM_SUPPLY("I2S CLK", PM860X_DAC_EN_2, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux), SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux), SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux), @@ -868,6 +869,11 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Left ADC", NULL, "Left ADC MOD"}, {"Right ADC", NULL, "Right ADC MOD"}, + /* I2S Clock */ + {"I2S DIN", NULL, "I2S CLK"}, + {"I2S DIN1", NULL, "I2S CLK"}, + {"I2S DOUT", NULL, "I2S CLK"}, + /* PCM/AIF1 Inputs */ {"PCM SDO", NULL, "ADC Left Mux"}, {"PCM SDO", NULL, "ADCR EC Mux"}, -- 1.7.0.4 --------------030808060208020409080204--

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Re: [alsa-devel] [PATCH] ASoC: sn95031: Staticize sn95031_pcm_hw_params
by Mark Brown 19 Sep '11

19 Sep '11

On Mon, Sep 19, 2011 at 04:34:28PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks.

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[alsa-devel] [RFC PATCH] Fixup automute for Realtek auto parser
by David Henningsson 19 Sep '11

19 Sep '11

The ~6 months old functionality for configurable automuting is broken for the case that the user has only HP and LO (no speakers) - basically there is an "Automute mode" control, but enabling it does nothing. While fixing that, I also took the liberty of refactoring/rewriting some of that code, in order to make it easier to understand and maintain. Takashi, given you approve/like these changes, I'll go ahead and fix up the quirk files to fit the new variable names. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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Re: [alsa-devel] [PATCH] ASoC: sn95031: Staticize sn95031_pcm_hw_params
by Vinod Koul 19 Sep '11

19 Sep '11

On Mon, 2011-09-19 at 16:34 +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/codecs/sn95031.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c > index b4f1cb4..29945b0 100644 > --- a/sound/soc/codecs/sn95031.c > +++ b/sound/soc/codecs/sn95031.c > @@ -658,7 +658,7 @@ static int sn95031_pcm_spkr_mute(struct snd_soc_dai *dai, int mute) > return 0; > } > > -int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, > +static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, > struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) > { > unsigned int format, rate; Acked-by: Vinod Koul <vinod.koul(a)linux.intel.com> -- ~Vinod

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