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

[alsa-devel] multi channels input and single channel out
by hong zhang 06 May '11

06 May '11

List, I have 8 channels of hdmi input and one channel output on my board. When arecord -f dat hdmi-rx.wav to store 8 channels data, aplay -f dat hdmi-rx.wav releases noise. Ia there any way arecord stores each channel data to 8 different files? Or I have to do beam forming to sort 8 channels out by custom application? Thanks. ---henry

3 4

[alsa-devel] STAC92xx model problem on HP EliteBook 8740w
by Johannes Kimmel 06 May '11

06 May '11

Hi, I hope this is the right location for my problem. When don't specify the model for my soundcard, everything works except spdif output over displayport. Correct led status, control names etc. working fine. But when I want to use sound over displayport, one working option is model=rev, but then my laptop-speakers don't work anymore. I think there is maybe a right model missing, or at least I didn't find the right. Here are 2 alsa-infos: spdif working: http://paste.pocoo.org/show/380474/ laptop-speaker working: http://paste.pocoo.org/show/380473/ I compile my kernel myself and I know the way to the alsa-drivers. So if there's anything to try that might help solve the problem, just ask. Johannes Kimmel

2 4

[alsa-devel] Hang in snd_pcm_writei with alsa-pulse plugin
by Matthew Gregan 06 May '11

06 May '11

Hi, I think I'm seeing a bug in the alsa-pulse plugin where the buffer management ends up corrupt and results in a deadlock waiting for free buffer space. This occurs when resuming from pause using snd_pcm_pause. After resuming, my application tries to write a fixed block of data, expecting snd_pcm_writei to block if the data is larger than the available buffer size (the result of snd_pcm_avail_update). I originally observed this in the wild in Firefox, which pauses and resumes the sound device whenever network buffering occurs. I'm planning to include the workaround mentioned below in the next Firefox release (Mozilla bug 573924). What happens is that, after resuming with snd_pcm_pause, a call to snd_pcm_writei never returns. This happens on the write call that would have exceeded the available buffer size, which I would expect to block only until sufficient buffer space became available. It's possible to get into a similar situation using SND_PCM_NONBLOCK and waiting on the sound device if it returns EAGAIN, except that snd_pcm_writei always returns EAGAIN and snd_pcm_wait returns 1 immediately, resulting in a tight loop in the calling code. I discovered that I can reliably workaround the problem by ensuring the first writes after resuming from pause are never larger than what snd_pcm_avail_update returns. After writing enough to fill (but not exceed) the available buffer size, the code returns to the fixed buffer size per write strategy and continues as normal. The problem occurs with the following stack: #0 __poll (fds=<value optimized out>, nfds=<value optimized out>, timeout=<value optimized out>) at ../sysdeps/unix/sysv/linux/poll.c:87 #1 snd1_pcm_wait_nocheck (pcm=0x1b9a780, timeout=-1) at pcm.c:2367 #2 snd1_pcm_write_areas (pcm=0x1b9a780, areas=0x7fff4ce9b890, offset=<value optimized out>, size=30000, func=0x339ba91d10 <ioplug_priv_transfer_areas>) at pcm.c:6655 #3 snd_pcm_ioplug_writei (pcm=0x1b9a780, buffer=<value optimized out>, size=30000) at pcm_ioplug.c:561 #4 bwrite (pcm=0x1b9a780, towrite=30000) at atest2.c:29 #5 main (argc=1, argv=0x7fff4ce9ba68) at atest2.c:86 I'm Fedora 13 x86_64 with all updates from updates-testing. ALSA is 1.0.22-1, PulseAudio is 0.9.21-6, and the kernel is 2.6.34.7-61. I've also tested against the current git versions of alsa-libs and alsa-plugins and can still reproduce the problem. I've attached a simple test program that reproduces this problem reliably on my machine. It writes a period sized buffer in a loop, waiting half a period until the next attempt. Every few iterations, it pauses the sound device for half a period and then resumes it. It usually hangs within 2-3 pause/resume cycles. Running the test with "-r" enables the recovery code I mentioned above. It never hangs when tested using the hardware ALSA backend with alsa-pulse disabled, but my sound hardware doesn't seem to support snd_pcm_pause. Cheers, -mjg -- Matthew Gregan |/ /| kinetik(a)flim.org

3 9

[alsa-devel] [PATCH 00/19] ASoC for moorestown
by Lu Guanqun 06 May '11

06 May '11

This series of patches provides the ASoC implementation of moorestown audio. This series adds a new codec called upd9976 and a new machine driver called mrst_machine. The platform driver is mostly re-used. With this patchset, it can do: - playback via headphone - playback via internal speaker - capture via DMIC - capture via headset microphone - jack detection However, other functionalities such as voice support in uPD9976 codec is not implemented yet. This patchset mainly focuses on the audio support in uPD9976 codec. Any comments are welcome. --- Lu Guanqun (17): ASoC: upd9976: Add Renesas uPD9976 codec driver ASoC: sst_platform: add cpu dai driver for moorestown platform ASoC: mrst_machine: add moorestown machine driver ASoC: mrst_machine: add speaker widget to moorestown machine driver ASoC: mrst_machine: enable user to select different output ASoC: upd9976: add DMIC support ASoC: upd9976: add Analog MIC support ASoC: upd9976: add microphone bias support ASoC: upd9976: add jack detection function ASoC: mrst_machine: add capture functionality ASoC: mrst_machine: add jack detection support ASoC: upd9976: add mute switch for DMIC ASoC: upd9976: add mute switch for analog ASoC: mrst_machine: make DMIC's output to PCM2 mono ASoC: mrst_machine: make MIC2 pseudo-differential ASoC: upd9976: add capture volume for analog inputs ASoC: upd9976: add capture volume for DMIC Wang Xingchao (2): ASoC: upd9976: add capture ability for dai driver ASoC: sst_platform: add capture capability for cpu dai driver sound/soc/codecs/Kconfig | 4 sound/soc/codecs/Makefile | 2 sound/soc/codecs/upd9976.c | 569 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/upd9976.h | 85 ++++++ sound/soc/mid-x86/Kconfig | 12 + sound/soc/mid-x86/Makefile | 2 sound/soc/mid-x86/mrst_machine.c | 365 ++++++++++++++++++++++++ sound/soc/mid-x86/sst_platform.c | 20 + 8 files changed, 1059 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/upd9976.c create mode 100644 sound/soc/codecs/upd9976.h create mode 100644 sound/soc/mid-x86/mrst_machine.c -- guanqun

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[alsa-devel] Disable Sticky PCM stream assignment for AD codecs
by Raymond Yau 06 May '11

06 May '11

Refer to 0e7adbe263f89ea2ef15b5af5e80a812b2a85025 "Disable sticky PCM stream assignment for AD codecs" Is it normal for "hda_codec_cleanup_stream" appear twice in system log when stick PCM stream is disabled arecord -v -Dhw:CARD=Intel -f cd /dev/null ALSA hda_intel.c:1678: azx_pcm_prepare: bufsize=0x10000, format=0x4011 ALSA hda_codec.c:1295: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x4011 ALSA hda_codec.c:1358: hda_codec_cleanup_stream: NID=0x8 ALSA hda_codec.c:1358: hda_codec_cleanup_stream: NID=0x8

1 0

[alsa-devel] snd_pcm_avail_update() returning large values
by Baek Chang 06 May '11

06 May '11

Hi, Occasionally, when using pulseaudio I see the following message: snd_pcm_avail_update() returned a value that is exceptionally large. This seems to be an issue with the ALSA driver itself. My question is how does snd_pcm_avail_update(), get updated? It uses hw buffer positions correct? Is there an example driver that implements snd_pcm_avail_update() correctly and updates buffer positions correctly? Thanks Baek

2 1

[alsa-devel] [PATCH 2/2] alsa-lib: fixed coverity reported issues under "RESOURCE_LEAK" checker.
by sudarshan.bisht＠nokia.com 05 May '11

05 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 got fixes for potential memory leaks in error cases. --- modules/mixer/simple/sbase.c | 1 + src/conf.c | 2 ++ src/control/control_hw.c | 4 +++- src/pcm/pcm_file.c | 6 +++++- src/pcm/pcm_ladspa.c | 22 +++++++++++++++------- src/pcm/pcm_plug.c | 9 +++++++-- src/pcm/pcm_rate.c | 4 ++++ src/rawmidi/rawmidi.c | 4 +++- 8 files changed, 40 insertions(+), 12 deletions(-) diff --git a/modules/mixer/simple/sbase.c b/modules/mixer/simple/sbase.c index 97feee8..bb2f59d 100644 --- a/modules/mixer/simple/sbase.c +++ b/modules/mixer/simple/sbase.c @@ -377,6 +377,7 @@ static int simple_event_add1(snd_mixer_class_t *class, if (ctype != SND_CTL_ELEM_TYPE_BOOLEAN) { __invalid_type: snd_mixer_selem_id_free(id); + free(hsimple); return -EINVAL; } break; diff --git a/src/conf.c b/src/conf.c index ddefff6..66b0a51 100644 --- a/src/conf.c +++ b/src/conf.c @@ -4590,6 +4590,8 @@ static int parse_args(snd_config_t *subs, const char *str, snd_config_t *defs) if (err < 0) { _err: free(val); + if (sub) + snd_config_delete(sub); return err; } free(val); diff --git a/src/control/control_hw.c b/src/control/control_hw.c index cf258b4..29cbbc6 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -230,8 +230,10 @@ static int snd_ctl_hw_elem_tlv(snd_ctl_t *handle, int op_flag, return -errno; } if (op_flag == 0) { - if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) + if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) { + free(xtlv); return -EFAULT; + } memcpy(tlv, xtlv->tlv, xtlv->tlv[1] + 2 * sizeof(unsigned int)); } free(xtlv); diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index bfa1cc8..c74e8ab 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -226,7 +226,11 @@ static int snd_pcm_file_open_output_file(snd_pcm_file_t *file) file->final_fname); return -errno; } - fd = fileno(pipe); + fd = dup(fileno(pipe)); + if (fd >= 0) + fclose(pipe); + else + return -errno; } else { if (file->trunc) fd = open(file->final_fname, O_WRONLY|O_CREAT|O_TRUNC, diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index c413c10..41ad06e 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -750,8 +750,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads if (instance->input.data == NULL || instance->input.m_data == NULL || instance->output.data == NULL || - instance->output.m_data == NULL) + instance->output.m_data == NULL) { + free(pchannels); return -ENOMEM; + } for (idx = 0; idx < instance->input.channels.size; idx++) { chn = instance->output.channels.array[idx]; if (pchannels[chn] == NULL && chn < ichannels) { @@ -761,8 +763,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads instance->input.data[idx] = pchannels[chn]; if (instance->input.data[idx] == NULL) { instance->input.data[idx] = snd_pcm_ladspa_allocate_zero(ladspa, 0); - if (instance->input.data[idx] == NULL) - return -ENOMEM; + if (instance->input.data[idx] == NULL) { + free(pchannels); + return -ENOMEM; + } } } for (idx = 0; idx < instance->output.channels.size; idx++) { @@ -770,8 +774,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads /* FIXME/OPTIMIZE: check if we can remove double alloc */ /* if LADSPA plugin has no broken inplace */ instance->output.data[idx] = malloc(sizeof(LADSPA_Data) * ladspa->allocated); - if (instance->output.data[idx] == NULL) - return -ENOMEM; + if (instance->output.data[idx] == NULL) { + free(pchannels); + return -ENOMEM; + } pchannels[chn] = instance->output.m_data[idx] = instance->output.data[idx]; } } @@ -793,8 +799,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads instance->output.data[idx] = NULL; } else { instance->output.data[idx] = snd_pcm_ladspa_allocate_zero(ladspa, 1); - if (instance->output.data[idx] == NULL) - return -ENOMEM; + if (instance->output.data[idx] == NULL) { + free(pchannels); + return -ENOMEM; + } } } } diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index e9d2923..e80bd3e 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -1298,6 +1298,7 @@ int _snd_pcm_plug_open(snd_pcm_t **pcmp, const char *name, err = snd_pcm_route_load_ttable(tt, ttable, csize, ssize, &cused, &sused, -1); if (err < 0) { snd_config_delete(sconf); + free(ttable); return err; } } @@ -1310,12 +1311,16 @@ int _snd_pcm_plug_open(snd_pcm_t **pcmp, const char *name, err = snd_pcm_open_slave(&spcm, root, sconf, stream, mode, conf); snd_config_delete(sconf); - if (err < 0) + if (err < 0) { + free(ttable); return err; + } err = snd_pcm_plug_open(pcmp, name, sformat, schannels, srate, rate_converter, route_policy, ttable, ssize, cused, sused, spcm, 1); - if (err < 0) + if (err < 0) { + free(ttable); snd_pcm_close(spcm); + } return err; } #ifndef DOC_HIDDEN diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 70e30e5..eb35e4a 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -1392,11 +1392,13 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, } else { SNDERR("Invalid type for rate converter"); snd_pcm_close(pcm); + free(rate); return -EINVAL; } if (err < 0) { SNDERR("Cannot find rate converter"); snd_pcm_close(pcm); + free(rate); return -ENOENT; } #else @@ -1405,6 +1407,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, err = open_func(SND_PCM_RATE_PLUGIN_VERSION, &rate->obj, &rate->ops); if (err < 0) { snd_pcm_close(pcm); + free(rate); return err; } #endif @@ -1413,6 +1416,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, ! rate->ops.input_frames || ! rate->ops.output_frames) { SNDERR("Inproper rate plugin %s initialization", type); snd_pcm_close(pcm); + free(rate); return err; } diff --git a/src/rawmidi/rawmidi.c b/src/rawmidi/rawmidi.c index 0bd6b96..53cd068 100644 --- a/src/rawmidi/rawmidi.c +++ b/src/rawmidi/rawmidi.c @@ -256,8 +256,10 @@ static int snd_rawmidi_open_conf(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp snd_config_delete(type_conf); if (err >= 0) err = open_func(inputp, outputp, name, rawmidi_root, rawmidi_conf, mode); - if (err < 0) + if (err < 0) { + snd_dlclose(h); return err; + } if (inputp) { (*inputp)->dl_handle = h; h = NULL; snd_rawmidi_params_default(*inputp, &params); -- 1.7.0.4

1 1

[alsa-devel] [PATCH 2/2] alsa-lib: fixed coverity reported issues under "RESOURCE_LEAK" checker.
by sudarshan.bisht＠nokia.com 05 May '11

05 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 got fixes for potential memory leaks in error cases. --- modules/mixer/simple/sbase.c | 1 + src/conf.c | 3 +++ src/control/control_hw.c | 4 +++- src/pcm/pcm_file.c | 4 +++- src/pcm/pcm_ladspa.c | 22 +++++++++++++++------- src/pcm/pcm_plug.c | 9 +++++++-- src/pcm/pcm_rate.c | 4 ++++ src/rawmidi/rawmidi.c | 4 +++- 8 files changed, 39 insertions(+), 12 deletions(-) diff --git a/modules/mixer/simple/sbase.c b/modules/mixer/simple/sbase.c index 97feee8..bb2f59d 100644 --- a/modules/mixer/simple/sbase.c +++ b/modules/mixer/simple/sbase.c @@ -377,6 +377,7 @@ static int simple_event_add1(snd_mixer_class_t *class, if (ctype != SND_CTL_ELEM_TYPE_BOOLEAN) { __invalid_type: snd_mixer_selem_id_free(id); + free(hsimple); return -EINVAL; } break; diff --git a/src/conf.c b/src/conf.c index ddefff6..842e5e3 100644 --- a/src/conf.c +++ b/src/conf.c @@ -638,6 +638,7 @@ static int get_char_skip_comments(input_t *input) fd = malloc(sizeof(*fd)); if (!fd) { free(str); + snd_input_close(in); return -ENOMEM; } fd->name = str; @@ -4590,6 +4591,8 @@ static int parse_args(snd_config_t *subs, const char *str, snd_config_t *defs) if (err < 0) { _err: free(val); + if (sub) + snd_config_delete(sub); return err; } free(val); diff --git a/src/control/control_hw.c b/src/control/control_hw.c index cf258b4..29cbbc6 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -230,8 +230,10 @@ static int snd_ctl_hw_elem_tlv(snd_ctl_t *handle, int op_flag, return -errno; } if (op_flag == 0) { - if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) + if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) { + free(xtlv); return -EFAULT; + } memcpy(tlv, xtlv->tlv, xtlv->tlv[1] + 2 * sizeof(unsigned int)); } free(xtlv); diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index bfa1cc8..adba7c0 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -226,7 +226,9 @@ static int snd_pcm_file_open_output_file(snd_pcm_file_t *file) file->final_fname); return -errno; } - fd = fileno(pipe); + fd = dup(fileno(pipe)); + if (fd >= 0) + fclose(pipe); } else { if (file->trunc) fd = open(file->final_fname, O_WRONLY|O_CREAT|O_TRUNC, diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index c413c10..41ad06e 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -750,8 +750,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads if (instance->input.data == NULL || instance->input.m_data == NULL || instance->output.data == NULL || - instance->output.m_data == NULL) + instance->output.m_data == NULL) { + free(pchannels); return -ENOMEM; + } for (idx = 0; idx < instance->input.channels.size; idx++) { chn = instance->output.channels.array[idx]; if (pchannels[chn] == NULL && chn < ichannels) { @@ -761,8 +763,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads instance->input.data[idx] = pchannels[chn]; if (instance->input.data[idx] == NULL) { instance->input.data[idx] = snd_pcm_ladspa_allocate_zero(ladspa, 0); - if (instance->input.data[idx] == NULL) - return -ENOMEM; + if (instance->input.data[idx] == NULL) { + free(pchannels); + return -ENOMEM; + } } } for (idx = 0; idx < instance->output.channels.size; idx++) { @@ -770,8 +774,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads /* FIXME/OPTIMIZE: check if we can remove double alloc */ /* if LADSPA plugin has no broken inplace */ instance->output.data[idx] = malloc(sizeof(LADSPA_Data) * ladspa->allocated); - if (instance->output.data[idx] == NULL) - return -ENOMEM; + if (instance->output.data[idx] == NULL) { + free(pchannels); + return -ENOMEM; + } pchannels[chn] = instance->output.m_data[idx] = instance->output.data[idx]; } } @@ -793,8 +799,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads instance->output.data[idx] = NULL; } else { instance->output.data[idx] = snd_pcm_ladspa_allocate_zero(ladspa, 1); - if (instance->output.data[idx] == NULL) - return -ENOMEM; + if (instance->output.data[idx] == NULL) { + free(pchannels); + return -ENOMEM; + } } } } diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index e9d2923..e80bd3e 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -1298,6 +1298,7 @@ int _snd_pcm_plug_open(snd_pcm_t **pcmp, const char *name, err = snd_pcm_route_load_ttable(tt, ttable, csize, ssize, &cused, &sused, -1); if (err < 0) { snd_config_delete(sconf); + free(ttable); return err; } } @@ -1310,12 +1311,16 @@ int _snd_pcm_plug_open(snd_pcm_t **pcmp, const char *name, err = snd_pcm_open_slave(&spcm, root, sconf, stream, mode, conf); snd_config_delete(sconf); - if (err < 0) + if (err < 0) { + free(ttable); return err; + } err = snd_pcm_plug_open(pcmp, name, sformat, schannels, srate, rate_converter, route_policy, ttable, ssize, cused, sused, spcm, 1); - if (err < 0) + if (err < 0) { + free(ttable); snd_pcm_close(spcm); + } return err; } #ifndef DOC_HIDDEN diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 70e30e5..eb35e4a 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -1392,11 +1392,13 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, } else { SNDERR("Invalid type for rate converter"); snd_pcm_close(pcm); + free(rate); return -EINVAL; } if (err < 0) { SNDERR("Cannot find rate converter"); snd_pcm_close(pcm); + free(rate); return -ENOENT; } #else @@ -1405,6 +1407,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, err = open_func(SND_PCM_RATE_PLUGIN_VERSION, &rate->obj, &rate->ops); if (err < 0) { snd_pcm_close(pcm); + free(rate); return err; } #endif @@ -1413,6 +1416,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, ! rate->ops.input_frames || ! rate->ops.output_frames) { SNDERR("Inproper rate plugin %s initialization", type); snd_pcm_close(pcm); + free(rate); return err; } diff --git a/src/rawmidi/rawmidi.c b/src/rawmidi/rawmidi.c index 0bd6b96..53cd068 100644 --- a/src/rawmidi/rawmidi.c +++ b/src/rawmidi/rawmidi.c @@ -256,8 +256,10 @@ static int snd_rawmidi_open_conf(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp snd_config_delete(type_conf); if (err >= 0) err = open_func(inputp, outputp, name, rawmidi_root, rawmidi_conf, mode); - if (err < 0) + if (err < 0) { + snd_dlclose(h); return err; + } if (inputp) { (*inputp)->dl_handle = h; h = NULL; snd_rawmidi_params_default(*inputp, &params); -- 1.7.0.4

3 5

[alsa-devel] platform driver clarification in twl6040 codec driver
by preetam 05 May '11

05 May '11

Hai, I am building an ALSA ASoC driver based on the original twl6040 driver on omap 4 panda board. I notice that there is a platform_device being created in the machine driver which is getting bound to the platform_driver created by the alsa soc core. i.e "soc-audio". sound/soc/omap/sd4430.c ================ sdp4430_snd_device = platform_device_alloc("soc-audio", -1); if (!sdp4430_snd_device) { printk(KERN_ERR "Platform device allocation failed\n"); return -ENOMEM; } snd_soc_register_dais(&sdp4430_snd_device->dev, dai, ARRAY_SIZE(dai)); platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430); ret = platform_device_add(sdp4430_snd_device); soc-core.c ====== static struct platform_driver soc_driver = { .driver = { .name = "soc-audio", .owner = THIS_MODULE, .pm = &soc_pm_ops, }, .probe = soc_probe, .remove = soc_remove, }; However, I fail to understand, how the platform_driver in the twl6040.c file gets bound and who calls its probe. sound/soc/codecs/twl6040.c ================= static struct platform_driver twl6040_codec_driver = { .driver = { .name = "twl6040-codec", .owner = THIS_MODULE, }, .probe = twl6040_codec_probe, .remove = __devexit_p(twl6040_codec_remove), }; I have used the same model for my codec device driver and the probe function in the codec driver under platform_driver never gets invoked. Can anyone give me a clarification on this? Regards, Preetam

1 0

[alsa-devel] trigger function is constantly called
by hong zhang 04 May '11

04 May '11

List, My code in driver looks like: static struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .startup = davinci_mcasp_startup, .trigger = davinci_mcasp_trigger, .hw_params = davinci_mcasp_hw_params, .set_fmt = davinci_mcasp_set_dai_fmt, }; Trigger function, davinci_mcasp_trigger(), is called only once when do arecord or aplay from command line. Works perfect. When combine both together and use ping pong buffers, if no (AEC) is introduced, sound is good. After adding (AEC- Acoustic Echo Canceler), davinci_mcasp_trigger() is constantly called for start/close on playback side respectively and complain underrun (CPU is %90 usage). Increase ping-pong buffers do not stop triggering. Any help will be appreciated! ---henry

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