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

[alsa-devel] ASoC legacy i2c client driver probing issue
by Feng Ye 04 May '11

04 May '11

Hello, I have a legacy i2c client driver probing issue. I use kernel version 2.6.27, and wm8750 external codec. The wm8750.c in sound/soc/codecs has a legacy i2c interface (e.g. use attach_adapter() callback). The callback just calls to i2c_probe(). Inside i2c_probe() it checks if the i2c address already been used, and fails there. In our i2c bus driver, we used i2c_register_board_info() with a static i2c_board_info structure. And then i2c_add_numbered_adapter() is called.

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[alsa-devel] [PATCH 1/3] alsa-lib: fixed coverity reported issues under "FORWARD_NULL" checker.
by sudarshan.bisht＠nokia.com 04 May '11

04 May '11

From: Sudarshan Bisht <sudarshan.bisht(a)nokia.com> Coverity Static Analysis helps developers find hard-to-spot, yet potentially crash-causing defects early in the development phase, reducing the cost,time, and risk of software errors. This patch has fix for situations where variable can be NULL but not been checked beforehand --- modules/mixer/simple/sbasedl.c | 2 +- src/conf.c | 1 + src/hwdep/hwdep.c | 1 + src/pcm/pcm_hooks.c | 1 + src/pcm/pcm_simple.c | 2 +- src/rawmidi/rawmidi.c | 1 + src/rawmidi/rawmidi_virt.c | 8 +++++--- 7 files changed, 11 insertions(+), 5 deletions(-) diff --git a/modules/mixer/simple/sbasedl.c b/modules/mixer/simple/sbasedl.c index 0137586..494802f 100644 --- a/modules/mixer/simple/sbasedl.c +++ b/modules/mixer/simple/sbasedl.c @@ -99,7 +99,7 @@ int mixer_simple_basic_dlopen(snd_mixer_class_t *class, __error: if (initflag) free(priv); - if (h == NULL) + if (h) snd_dlclose(h); free(xlib); return -ENXIO; diff --git a/src/conf.c b/src/conf.c index 8939d62..ddefff6 100644 --- a/src/conf.c +++ b/src/conf.c @@ -3268,6 +3268,7 @@ static int snd_config_hooks_call(snd_config_t *root, snd_config_t *config, snd_c snd_config_iterator_t i, next; if (snd_config_get_type(func_conf) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for func %s definition", str); + err = -EINVAL; goto _err; } snd_config_for_each(i, next, func_conf) { diff --git a/src/hwdep/hwdep.c b/src/hwdep/hwdep.c index b882b35..5dc791c 100644 --- a/src/hwdep/hwdep.c +++ b/src/hwdep/hwdep.c @@ -78,6 +78,7 @@ static int snd_hwdep_open_conf(snd_hwdep_t **hwdep, if (err >= 0) { if (snd_config_get_type(type_conf) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for HWDEP type %s definition", str); + err = -EINVAL; goto _err; } snd_config_for_each(i, next, type_conf) { diff --git a/src/pcm/pcm_hooks.c b/src/pcm/pcm_hooks.c index 3a99d55..404d51e 100644 --- a/src/pcm/pcm_hooks.c +++ b/src/pcm/pcm_hooks.c @@ -385,6 +385,7 @@ static int snd_pcm_hook_add_conf(snd_pcm_t *pcm, snd_config_t *root, snd_config_ if (err >= 0) { if (snd_config_get_type(type) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for PCM type %s definition", str); + err = -EINVAL; goto _err; } snd_config_for_each(i, next, type) { diff --git a/src/pcm/pcm_simple.c b/src/pcm/pcm_simple.c index 975f699..f943ec0 100644 --- a/src/pcm/pcm_simple.c +++ b/src/pcm/pcm_simple.c @@ -89,7 +89,7 @@ static int set_hw_params(snd_pcm_t *pcm, return err; if (periods == 1) return -EINVAL; - if (*period_time == 0) { + if (period_time) { err = INTERNAL(snd_pcm_hw_params_get_period_time)(hw_params, period_time, NULL); if (err < 0) return err; diff --git a/src/rawmidi/rawmidi.c b/src/rawmidi/rawmidi.c index b28488a..0bd6b96 100644 --- a/src/rawmidi/rawmidi.c +++ b/src/rawmidi/rawmidi.c @@ -201,6 +201,7 @@ static int snd_rawmidi_open_conf(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp if (err >= 0) { if (snd_config_get_type(type_conf) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for RAWMIDI type %s definition", str); + err = -EINVAL; goto _err; } snd_config_for_each(i, next, type_conf) { diff --git a/src/rawmidi/rawmidi_virt.c b/src/rawmidi/rawmidi_virt.c index 52b8984..e5b17e4 100644 --- a/src/rawmidi/rawmidi_virt.c +++ b/src/rawmidi/rawmidi_virt.c @@ -383,9 +383,11 @@ int snd_rawmidi_virtual_open(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp, _err: if (seq_handle) snd_seq_close(seq_handle); - if (virt->midi_event) - snd_midi_event_free(virt->midi_event); - free(virt); + if (virt) { + if (virt->midi_event) + snd_midi_event_free(virt->midi_event); + free(virt); + } if (inputp) free(*inputp); if (outputp) -- 1.7.0.4

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[alsa-devel] [PATCH] [ALSA][UCM] fix memory leak when executing cset commands. snd_ctl_elem's are allocated, but never free'ed
by baeksan＠ccrma.stanford.edu 04 May '11

04 May '11

From: Baek Chang <breakchange(a)gmail.com> --- src/ucm/main.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 86f29e3..f8ceeeb 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -173,7 +173,8 @@ static int execute_cset(snd_ctl_t *ctl, char *cset) pos = strrchr(cset, ' '); if (pos == NULL) { uc_error("undefined value for cset >%s<", cset); - return -EINVAL; + err = -EINVAL; + goto __fail; } *pos = '\0'; err = snd_ctl_ascii_elem_id_parse(id, cset); @@ -196,6 +197,14 @@ static int execute_cset(snd_ctl_t *ctl, char *cset) err = 0; __fail: *pos = ' '; + + if (id != NULL) + free(id); + if (value != NULL) + free(value); + if (info != NULL) + free(info); + return err; } -- 1.6.0.4

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[alsa-devel] arecord read error: Input/output error
by kshemanth.vasireddi＠wipro.com 04 May '11

04 May '11

Hi, I am new to ALSA, I have written an ALSA driver for ADC voice channels. I am testing ALSA driver using arecord "arecord -c 2 -f s16_LE -r 48000 -t raw test_file.raw"; when capture file reaches 88KB arecord is terminating capture with error "arecord: pcm_read:1529: read error: Input/output error Can someone please help to identify actual reason for this. Thanks and regards, Kshemanth Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[alsa-devel] SNDRV_PCM_TRIGGER_STOP sequence
by Irfan Shaikh 04 May '11

04 May '11

When command SNDRV_PCM_TRIGGER_STOP is called. I am continously reading from capture device. I hav put printk in case SNDRV_PCM_TRIGGER_STOP in usbaudio.c I get SNDRV_PCM_TRIGGER_STOP in between of continous read. Please tell me when this STOP sequence is called from Application->library->driver Regards SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

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[alsa-devel] [PATCH] sst_platform: add hw_free callback
by xingchao 04 May '11

04 May '11

Signed-off-by: xingchao <xingchao.wang(a)intel.com> --- sound/soc/mid-x86/sst_platform.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 9765fb8..5a946b4 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -380,6 +380,11 @@ static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } +static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + static struct snd_pcm_ops sst_platform_ops = { .open = sst_platform_open, .close = sst_platform_close, @@ -388,6 +393,7 @@ static struct snd_pcm_ops sst_platform_ops = { .trigger = sst_platform_pcm_trigger, .pointer = sst_platform_pcm_pointer, .hw_params = sst_platform_pcm_hw_params, + .hw_free = sst_platform_pcm_hw_free, }; static void sst_pcm_free(struct snd_pcm *pcm) -- 1.7.1

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[alsa-devel] [PATCH 1/2] ASoC: Use shared controls for input signal path in WM8915
by Mark Brown 04 May '11

04 May '11

Gives finer grained power management. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8915.c | 45 +++++++++++++++++++++++++-------------------- 1 files changed, 25 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index 4a3c5cc..0dc1dd7 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -991,10 +991,10 @@ SND_SOC_DAPM_MICBIAS("MICB1", WM8915_POWER_MANAGEMENT_1, 8, 0), SND_SOC_DAPM_PGA("IN1L PGA", WM8915_POWER_MANAGEMENT_2, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("IN1R PGA", WM8915_POWER_MANAGEMENT_2, 4, 0, NULL, 0), -SND_SOC_DAPM_PGA("ADC", SND_SOC_NOPM, 0, 0, NULL, 0), - -SND_SOC_DAPM_MUX("IN1 Mux", SND_SOC_NOPM, 0, 0, &in1_mux), -SND_SOC_DAPM_MUX("IN2 Mux", SND_SOC_NOPM, 0, 0, &in2_mux), +SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &in1_mux), +SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &in1_mux), +SND_SOC_DAPM_MUX("IN2L Mux", SND_SOC_NOPM, 0, 0, &in2_mux), +SND_SOC_DAPM_MUX("IN2R Mux", SND_SOC_NOPM, 0, 0, &in2_mux), SND_SOC_DAPM_PGA("IN1L", WM8915_POWER_MANAGEMENT_7, 2, 0, NULL, 0), SND_SOC_DAPM_PGA("IN1R", WM8915_POWER_MANAGEMENT_7, 3, 0, NULL, 0), @@ -1172,28 +1172,33 @@ static const struct snd_soc_dapm_route wm8915_dapm_routes[] = { { "DMIC1L", NULL, "DMIC1" }, { "DMIC1R", NULL, "DMIC1" }, - { "ADC", NULL, "ADCL" }, - { "ADC", NULL, "ADCR" }, + { "IN1L Mux", "ADC", "ADCL" }, + { "IN1L Mux", "DMIC1", "DMIC1L" }, + { "IN1L Mux", "DMIC2", "DMIC2L" }, + + { "IN1R Mux", "ADC", "ADCR" }, + { "IN1R Mux", "DMIC1", "DMIC1R" }, + { "IN1R Mux", "DMIC2", "DMIC2R" }, - { "IN1 Mux", "ADC", "ADC" }, - { "IN1 Mux", "DMIC1", "DMIC1" }, - { "IN1 Mux", "DMIC2", "DMIC2" }, + { "IN2L Mux", "ADC", "ADCL" }, + { "IN2L Mux", "DMIC1", "DMIC1L" }, + { "IN2L Mux", "DMIC2", "DMIC2L" }, - { "IN2 Mux", "ADC", "ADC" }, - { "IN2 Mux", "DMIC1", "DMIC1" }, - { "IN2 Mux", "DMIC2", "DMIC2" }, + { "IN2R Mux", "ADC", "ADCR" }, + { "IN2R Mux", "DMIC1", "DMIC1R" }, + { "IN2R Mux", "DMIC2", "DMIC2R" }, - { "Left Sidetone", "IN1", "IN1 Mux" }, - { "Left Sidetone", "IN2", "IN2 Mux" }, + { "Left Sidetone", "IN1", "IN1L Mux" }, + { "Left Sidetone", "IN2", "IN2L Mux" }, - { "Right Sidetone", "IN1", "IN1 Mux" }, - { "Right Sidetone", "IN2", "IN2 Mux" }, + { "Right Sidetone", "IN1", "IN1R Mux" }, + { "Right Sidetone", "IN2", "IN2R Mux" }, - { "DSP1TXL", "IN1 Switch", "IN1 Mux" }, - { "DSP1TXR", "IN1 Switch", "IN1 Mux" }, + { "DSP1TXL", "IN1 Switch", "IN1L Mux" }, + { "DSP1TXR", "IN1 Switch", "IN1R Mux" }, - { "DSP2TXL", "IN1 Switch", "IN2 Mux" }, - { "DSP2TXR", "IN1 Switch", "IN2 Mux" }, + { "DSP2TXL", "IN1 Switch", "IN2L Mux" }, + { "DSP2TXR", "IN1 Switch", "IN2R Mux" }, { "AIF1TX0", NULL, "DSP1TXL" }, { "AIF1TX1", NULL, "DSP1TXR" }, -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Don't crash on PM operations
by Mark Brown 04 May '11

04 May '11

The move over to exposing snd_soc_register_card() let the initialisation of the driver data we use to find the card in PM operations go AWOL. Fix this by setting the driver data when we register the card. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a823654..f3833f4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3365,6 +3365,8 @@ int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + dev_set_drvdata(card->dev, card); + snd_soc_initialize_card_lists(card); soc_init_card_debugfs(card); -- 1.7.4.1

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[alsa-devel] [PATCH 0/4] Implement control sharing
by Stephen Warren 03 May '11

03 May '11

Control sharing is enabled when two widgets include pointers to the same kcontrol_new in their definition. Specifically: static const struct snd_kcontrol_new adcinput_mux = SOC_DAPM_ENUM("ADC Input", adcinput_enum); static const struct snd_soc_dapm_widget wm8903_dapm_widgets[] = { SND_SOC_DAPM_MUX("Left ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), SND_SOC_DAPM_MUX("Right ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), }; This is useful when a single register bit or field affects multiple muxes at once. The common case is to have separate control bits or fields for each mux (channel). An alternative way of looking at this is that the mux is a stereo (or even n-channel) mux, rather than independant mono muxes. I have tested this on NVIDIA Tegra Seaboard, specifically with the WM8903 "ADC Input" control, and a locally-enabled digital mic setup. I have not compiled nor tested any other platforms. This patchset does not implement control sharing for dapm_new_mixer() yet. I can do so after discussion/review of this patch series, although note that I don't have anything to test the actual sharing on for such controls. Stephen Warren (4): ASoC: s/w->kcontrols/w->kcontrol_news/g ASoC: Add w->kcontrols, and populate it ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol ASoC: Implement mux control sharing include/sound/soc-dapm.h | 75 +++++---- sound/soc/codecs/88pm860x-codec.c | 2 +- sound/soc/soc-dapm.c | 315 ++++++++++++++++++++++++++----------- 3 files changed, 266 insertions(+), 126 deletions(-)

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[alsa-devel] [PATCH] ASoC: JZ4740: Fix i2s shutdown
by Lars-Peter Clausen 03 May '11

03 May '11

The i2s shutdown callback has the check whether it should be disabled reversed. Currently it is disabled if another stream is still active, but kept enabled if the last stream is closed. This patch fixes it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/jz4740-i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 419bf4f..cd22a54 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -133,7 +133,7 @@ static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream, struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); uint32_t conf; - if (!dai->active) + if (dai->active) return; conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); -- 1.7.2.5

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