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

[alsa-devel] [PATCH] ALSA: Write outside array bounds
by Roel Kluin 30 Jul '09

30 Jul '09

e->sad[] is declared with size ELD_MAX_SAD (16), but the guard allows the range 0-31 Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with Parfait, http://research.sun.com/projects/parfait/ diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index fcad5ec..ec04e58 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -539,7 +539,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry, sname++; n = 10 * n + name[4] - '0'; } - if (n < 0 || n > 31) /* double the CEA limit */ + if (n < 0 || n > ELD_MAX_SAD) continue; if (!strcmp(sname, "_coding_type")) e->sad[n].format = val;

4 3

[alsa-devel] [PATCH] hda: add HP automute support to Intel ALC889/ALC889A models
by Wu Fengguang 30 Jul '09

30 Jul '09

It auto mutes all 8-channel outputs at rear panel when the front panel headphone is connected. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_realtek.c | 42 ++++++++++++++++++++++++-------- 1 file changed, 32 insertions(+), 10 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_realtek.c +++ sound-2.6/sound/pci/hda/patch_realtek.c @@ -1436,6 +1436,25 @@ static void alc_automute_amp_unsol_event alc_automute_amp(codec); } +static void alc889_automute_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x17; + spec->autocfg.speaker_pins[3] = 0x19; + spec->autocfg.speaker_pins[4] = 0x1a; + alc_automute_amp(codec); +} + +static void alc889_intel_init_hook(struct hda_codec *codec) +{ + alc889_coef_init(codec); + alc889_automute_init(codec); +} + static void alc888_fujitsu_xa3530_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -6994,6 +7013,11 @@ static struct hda_verb alc889_eapd_verbs { } }; +static struct hda_verb alc262_hippo_unsol_verbs[] = { + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} +}; static struct hda_verb alc885_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -7019,7 +7043,7 @@ static struct hda_verb alc885_init_verbs {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* Front HP Pin: output 0 (0x0c) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front Pin: output 0 (0x0c) */ @@ -8861,7 +8885,8 @@ static struct alc_config_preset alc882_p }, [ALC889A_INTEL] = { .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer }, - .init_verbs = { alc885_init_verbs, alc885_init_input_verbs }, + .init_verbs = { alc885_init_verbs, alc885_init_input_verbs, + alc262_hippo_unsol_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), @@ -8873,12 +8898,14 @@ static struct alc_config_preset alc882_p .channel_mode = alc889_8ch_intel_modes, .capsrc_nids = alc889_capsrc_nids, .input_mux = &alc889_capture_source, + .init_hook = alc889_automute_init, + .unsol_event = alc_automute_amp_unsol_event, .need_dac_fix = 1, }, [ALC889_INTEL] = { .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer }, .init_verbs = { alc885_init_verbs, alc889_init_input_verbs, - alc889_eapd_verbs }, + alc889_eapd_verbs, alc262_hippo_unsol_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), @@ -8890,7 +8917,8 @@ static struct alc_config_preset alc882_p .channel_mode = alc889_8ch_intel_modes, .capsrc_nids = alc889_capsrc_nids, .input_mux = &alc889_capture_source, - .init_hook = alc889_coef_init, + .init_hook = alc889_intel_init_hook, + .unsol_event = alc_automute_amp_unsol_event, .need_dac_fix = 1, }, [ALC883_6ST_DIG] = { @@ -10123,12 +10151,6 @@ static struct hda_verb alc262_eapd_verbs { } }; -static struct hda_verb alc262_hippo_unsol_verbs[] = { - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {} -}; - static struct hda_verb alc262_hippo1_unsol_verbs[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},

2 3

[alsa-devel] [PATCH] hda: add 2-channel mode to Intel ALC889/ALC889A models
by Wu Fengguang 30 Jul '09

30 Jul '09

This 2-channel mode is useful in that it will broadcast a 2-channel audio stream to all front/side/... ports. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_realtek.c | 44 ++++++++++++++++++-------------- 1 file changed, 25 insertions(+), 19 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_realtek.c +++ sound-2.6/sound/pci/hda/patch_realtek.c @@ -6682,18 +6682,27 @@ static struct hda_channel_mode alc883_3S }; /* + * 2ch mode + */ +static struct hda_verb alc889_ch2_intel_init[] = { + { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* * 6ch mode */ static struct hda_verb alc889_ch6_intel_init[] = { - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { } /* end */ }; @@ -6702,21 +6711,18 @@ static struct hda_verb alc889_ch6_intel_ * 8ch mode */ static struct hda_verb alc889_ch8_intel_init[] = { - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 }, { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 }, { } /* end */ }; -static struct hda_channel_mode alc889_8ch_intel_modes[2] = { +static struct hda_channel_mode alc889_8ch_intel_modes[3] = { + { 2, alc889_ch2_intel_init }, { 6, alc889_ch6_intel_init }, { 8, alc889_ch8_intel_init }, };

2 1

[alsa-devel] Chat 50 at usb-0000:00:1a.7-4.3 audio issues
by Jason Bell 30 Jul '09

30 Jul '09

G'day ASLA Developers I am hoping that someone might be able to assist me with an audio issue I am having. Recently I have begun using the Dell T3500 precisions as an Video Conferencing machine (using the Access Grid and EVO Software). Information on hardware specs, etc can be found at http://www.alsa-project.org/db/?f=7f33770c89802b0ab3132f21914cc6ca7f6626 37 Unfortunately, I have issues when using either a clearone Chat50 or Chat150 (these are essentially speaker phone devices [See http://www.clearone.com/products/product.php?cat=9&prod=94 for device description]) with the speaker audio being distorted and static, cracks and popping type sounds being introduced. The microphone audio quality is fine. These issues were seen using Fedora 10 and Fedora 11. I tried upgrading to the latest ALSA drivers and "testing" kernels without any success (on a previous Fedora 10 install). It was found (still in the initial stages of testings) that if I plugged the Chat50 device into a USB hub, rather than directly into the Machine, the audio issues disappeared (at least for some of the software). It should also be noted that I have a number of Dell T3400 series machines that don't experience these issues. Additionally is should be mentioned that the T3400 uses the ICH9 chipset family whereas the T3500 uses the ICH10 chipset family which includes the USB controller. Just wondering if there is anything I can do to resolve this issue? Many thanks in advance, Jason. -------------------------------------------- Jason Bell, B.I.T. (Honours) Research Systems Support Officer Information Technology Division CQ University Australia Australian Research Collaboration Service http://www.arcs.org.au/ E-mail : j.bell(a)cqu.edu.au jason.bell(a)arcs.org.au Work : +61 7 4930 9229 Mobile : 0409 630897 Postal : Building 19 Central Queensland University Bruce Highway Rockhampton, Queensland, Australia, 4702 -------------------------------------------- Patience is a virtue. But if I wanted Patience, I would have become a Doctor. --------------------------------------------

1 1

[alsa-devel] [PATCH] Fix checkpatch issues and typos of ad1938 codec and bf5xx-tdm dai
by Barry Song 29 Jul '09

29 Jul '09

1. fix "line over 80 characters" checkpatch warnings 2. ‘DMA_nnBIT_MASK’ is deprecated, use DMA_BIT_MASK instead 3. fix typos Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/bf5xx-tdm-pcm.c | 4 +- sound/soc/blackfin/bf5xx-tdm-pcm.h | 2 +- sound/soc/blackfin/bf5xx-tdm.h | 2 +- sound/soc/codecs/ad1938.c | 50 +++++++++++++++++++++++------------ 4 files changed, 37 insertions(+), 21 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index af7fa8d..ccb5e82 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -276,7 +276,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) sport_done(sport_handle); } -static u64 bf5xx_pcm_dmamask = DMA_32BIT_MASK; +static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) @@ -286,7 +286,7 @@ static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->dma_mask) card->dev->dma_mask = &bf5xx_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (dai->playback.channels_min) { ret = bf5xx_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.h b/sound/soc/blackfin/bf5xx-tdm-pcm.h index 9e8473d..ddc5047 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.h +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin + * sound/soc/blackfin/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin * * Copyright 2009 Analog Device Inc. * diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h index e5157cd..618ec3d 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-i2s.h + * sound/soc/blackfin/bf5xx-tdm.h * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index ffc97c9..aafda74 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -72,14 +72,20 @@ static const struct snd_kcontrol_new ad1938_snd_controls[] = { AD1938_DAC_R4_VOL, 0, 0xFF, 1), /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD1938_ADC_CTRL0, AD1938_ADCL1_MUTE, AD1938_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD1938_ADC_CTRL0, AD1938_ADCL2_MUTE, AD1938_ADCR2_MUTE, 1, 1), + SOC_DOUBLE("ADC1 Switch", AD1938_ADC_CTRL0, AD1938_ADCL1_MUTE, + AD1938_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD1938_ADC_CTRL0, AD1938_ADCL2_MUTE, + AD1938_ADCR2_MUTE, 1, 1), /* DAC switch control */ - SOC_DOUBLE("DAC1 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL1_MUTE, AD1938_DACR1_MUTE, 1, 1), - SOC_DOUBLE("DAC2 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL2_MUTE, AD1938_DACR2_MUTE, 1, 1), - SOC_DOUBLE("DAC3 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL3_MUTE, AD1938_DACR3_MUTE, 1, 1), - SOC_DOUBLE("DAC4 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL4_MUTE, AD1938_DACR4_MUTE, 1, 1), + SOC_DOUBLE("DAC1 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL1_MUTE, + AD1938_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL2_MUTE, + AD1938_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL3_MUTE, + AD1938_DACR3_MUTE, 1, 1), + SOC_DOUBLE("DAC4 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL4_MUTE, + AD1938_DACR4_MUTE, 1, 1), /* ADC high-pass filter */ SOC_SINGLE("ADC High Pass Filter Switch", AD1938_ADC_CTRL0, @@ -110,7 +116,8 @@ static int ad1938_mute(struct snd_soc_dai *dai, int mute) int reg; reg = codec->read(codec, AD1938_DAC_CTRL2); - reg = (mute > 0) ? reg | AD1938_DAC_MASTER_MUTE : reg & (~AD1938_DAC_MASTER_MUTE); + reg = (mute > 0) ? reg | AD1938_DAC_MASTER_MUTE : reg & + (~AD1938_DAC_MASTER_MUTE); codec->write(codec, AD1938_DAC_CTRL2, reg); return 0; @@ -119,7 +126,8 @@ static int ad1938_mute(struct snd_soc_dai *dai, int mute) static inline int ad1938_pll_powerctrl(struct snd_soc_codec *codec, int cmd) { int reg = codec->read(codec, AD1938_PLL_CLK_CTRL0); - reg = (cmd > 0) ? reg & (~AD1938_PLL_POWERDOWN) : reg | AD1938_PLL_POWERDOWN; + reg = (cmd > 0) ? reg & (~AD1938_PLL_POWERDOWN) : reg | + AD1938_PLL_POWERDOWN; codec->write(codec, AD1938_PLL_CLK_CTRL0, reg); return 0; @@ -171,8 +179,8 @@ static int ad1938_set_dai_fmt(struct snd_soc_dai *codec_dai, adc_reg = codec->read(codec, AD1938_ADC_CTRL2); dac_reg = codec->read(codec, AD1938_DAC_CTRL1); - /* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S with TDM) - * and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A) + /* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S + * with TDM) and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A) */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -363,7 +371,8 @@ static unsigned int ad1938_read_reg_cache(struct snd_soc_codec *codec, * read from the ad1938 register space */ -static unsigned int ad1938_read_reg(struct snd_soc_codec *codec, unsigned int reg) +static unsigned int ad1938_read_reg(struct snd_soc_codec *codec, + unsigned int reg) { char w_buf[AD1938_SPI_BUFLEN]; char r_buf[AD1938_SPI_BUFLEN]; @@ -502,12 +511,19 @@ static int ad1938_register(struct ad1938_priv *ad1938) ad1938_codec = codec; /* default setting for ad1938 */ - codec->write(codec, AD1938_DAC_CHNL_MUTE, 0x0); /* unmute dac channels */ - codec->write(codec, AD1938_DAC_CTRL2, 0x1A); /* de-emphasis: 48kHz, powedown dac */ - codec->write(codec, AD1938_DAC_CTRL0, 0x21); /* powerdown dac, dac tdm mode */ - codec->write(codec, AD1938_ADC_CTRL0, 0x3); /* high-pass filter enable */ - codec->write(codec, AD1938_ADC_CTRL1, 0x43); /* sata delay=1, adc aux mode */ - codec->write(codec, AD1938_PLL_CLK_CTRL0, 0x9D); /* pll input:mclki/xi */ + + /* unmute dac channels */ + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0x0); + /* de-emphasis: 48kHz, powedown dac */ + codec->write(codec, AD1938_DAC_CTRL2, 0x1A); + /* powerdown dac, dac in tdm mode */ + codec->write(codec, AD1938_DAC_CTRL0, 0x41); + /* high-pass filter enable */ + codec->write(codec, AD1938_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + codec->write(codec, AD1938_ADC_CTRL1, 0x43); + /* pll input: mclki/xi */ + codec->write(codec, AD1938_PLL_CLK_CTRL0, 0x9D); codec->write(codec, AD1938_PLL_CLK_CTRL1, 0x04); ad1938_fill_cache(codec); -- 1.5.6.3

2 1

[alsa-devel] how to configure .asoundrc to make arecord use a default device
by Jelle de Jong 29 Jul '09

29 Jul '09

Hello everybody, I am having some issues making arecord use the default device for recording. Could somebody help me get a working .asoundrc configuration? I attached all information in the attachment and also made temporary hyperlink: http://debian.pastebin.com/m52601230 It works with: arecord -v -f S16_LE -r 44100 -D audiodevice0 ~/record0.wav but it does not work with: arecord -v -f S16_LE -r 44100 ~/record0.wav background: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-November/012366.h… Thanks in advance, Best regards, Jelle de Jong

2 6

Re: [alsa-devel] clicking noise with Sigmatel STAC9202
by Christian Steffen 29 Jul '09

29 Jul '09

Here I send the two also-info files as attachments. Initial problem was: Under kernel 2.6.30, Sound is working, but has constant clicking noise in the background, also if the sound is completely muted. First I thought this sound would come from some hdd or whatever, but the device has no mechanical parts. But this is what it sounds like - Irregular clicking noise. This noise immediately starts during boot up as soon as the module is loaded and before X11 has started. I already tried options like "position_fix=1 or 2" and different "model="-options with no success. "position_fix" didn't do anything at all and different model-options have muted the device completely. Kernel 2.6.24 with the same files has no issues. >> Then give as attachments later (and don't forget Cc to alsa-devel ML) >> these two files, not URL. It's important because ML is archived >> while pastebin can be removed at any time. Thanks, Christian Steffen -----Ursprüngliche Nachricht----- Von: Takashi Iwai [mailto:tiwai@suse.de] Gesendet: Mittwoch, 29. Juli 2009 17:32 An: Christian Steffen Cc: 'Takashi Iwai' Betreff: Re: AW: [alsa-devel] clicking noise with Sigmatel STAC9202 At Wed, 29 Jul 2009 17:11:18 +0200, Christian Steffen wrote: > > Thanks, > > >> Well, then at least, the contents of /proc/asound/card0/codec#*, > >> /sys/class/sound/hwC0D*/*_pin_configs files (if present), and the file > >> generated via "alsactl -f somefile store". These are the most > >> important information. > > There is no pinconfig file and there is also no alsactl tool present. > I could obtain the output from the codec stuff: > > http://pastebin.com/m1b81f47d (old kernel 2.6.24, only the codec info) > http://pastebin.com/m5baeca9a (new 2.6.30) > > that's all I could get for the older kernel as I only have access using > telnet. Then give as attachments later (and don't forget Cc to alsa-devel ML) these two files, not URL. It's important because ML is archived while pastebin can be removed at any time. Takashi

1 0

[alsa-devel] backward/forward binary compatibility checker
by Andrey Ponomarenko 29 Jul '09

29 Jul '09

Colleagues, I'm software engineer from Institute for System Programing of Russian Academy of Sciences and we are developing a free lightweight tool for checking backward/forward binary compatibility of shared C/C++ libraries in OS Linux. It checks interface signatures and data type definitions in two library versions (headers and shared objects) and searches differences that may lead to incompatibility according to ABI standards. We have released beta version of this tool and we'd like you to consider its usefulness for your project. The package with the latest version of tool is in the attachment. The wiki-page with the latest release of binary compatibility checker is http://ispras.linux-foundation.org/index.php/ABI_compliance_checker Andrey Ponomarenko <susanin(a)ispras.ru>

1 0

[alsa-devel] Master channel is only mono
by Sune Mølgaard 29 Jul '09

29 Jul '09

As per gnubien's advice on #alsa on irc.freenode.net, I have run alsa-info.sh - the URL is http://www.alsa-project.org/db/?f=fbe93f301ca214007e66c521cd6c41a17a604c79 . My problem is that the master channel for my card is shown to be mono only, which I don't think seems right... Also, sound in general is rather low. My system is a very new Dell Studio laptop - I have loaded snd-hda-intel with mode=dell-m6 as per the kernel documentation. Best regards, Sune Mølgaard

2 2

[alsa-devel] pulseaudio workaround for alsa-driver bug
by jack craig 29 Jul '09

29 Jul '09

Hi alsa-developers, I wanted to report my recent issues with audio on fc11. after an update from fc10, my audio started failing. not the failure to year any audio, but instead, it would start off fine and then tail spin. i am using mplayer to play an internet asf stream to the local audio. the stream processing failure is described in https://bugzilla.redhat.com/show_bug.cgi?id=514213 i was able to get a workaround from the #pulseaudio guys, but! they encouraged me to report my experience and its fix to you to encourage the fix of this snd_pcm_delay() defect. if you get a driver fix for this, i am happy to try it out when its ready. thx, jackc... -- jack craig jackc(a)LinuxLightHouse.com 831-684-1375 (Office) 831-596-6924 (cell) IM: jackcraigaptos (AIM) _________________________________ This email has been ClamScanned ! www.LinuxLightHouse.com

3 3