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

[alsa-devel] Alsa Support for Korg Pandora Mini, Vox ToneLab ST
by Pawel Siemienski 17 May '11

17 May '11

Hi, I am Ubuntu Studio user and I use also ALSA for driving my audio for recording etc... I have tested two devices with ALSA: 1. Vox Tonelab ST - works with ALSA out of the box as USB audio interface with no need of modifications, provides complete functionalities needed for recording and playing audio 2. Korg Pandora Mini - is detected as audio interface, but when selecting it information that no controls are available is displayed, this device can not work as audio interface in fact with Alsa. Could you please advise what modifications in ALSA should be made to enable Korg Pandora Mini as audio interface with ALSA, is there any patch for this device? If you would need any information on Korg Pandora Mini please let know. Thanks Pawel Siemienski

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[alsa-devel] [PATCH alsa-lib emu10k1.conf 1/1] Fix "front" device of SB Live! Platinum
by Raymond Yau 13 May '11

13 May '11

Since commit 148c2b8e7c12b4ab8a039995fa9904e7e1300cc4. Front channels via fxbus 8 and 9 No sound from the front jack of SB Live! platinum CT4760P when the application using "front" device but sound from front jack when application using hw:0,0 and /dev/dsp Error message " I: (alsa-lib)setup.c: Cannot lock ctl elem*"* when Pulseaudio server open "front" device for playback and capture at the same time because of locked "EMU10K1 Send PCM Volume"

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[alsa-devel] Small Regression in midi for M-Audio Fast Track Ultra.
by Grant Diffey 12 May '11

12 May '11

In 2.6.33/34 when support originally was put in alsa mainline the MIDI ports were functional today they're not for me and I get the following in the kernel log when using this card. (Audio all works acceptably.) [ 1846.736062] usb 2-1: new high speed USB device using ehci_hcd and address 4 [ 1846.869790] usb 2-1: config 1 interface 3 altsetting 0 bulk endpoint 0x7 has invalid maxpacket 8 [ 1846.869802] usb 2-1: config 1 interface 3 altsetting 0 bulk endpoint 0x87 has invalid maxpacket 8 [ 1846.870897] usb 2-1: New USB device found, idVendor=0763, idProduct=2080 [ 1846.870905] usb 2-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 1846.870912] usb 2-1: Product: Fast Track Ultra [ 1846.870917] usb 2-1: Manufacturer: M-Audio This issue may be unrelated to the data below but it's included in case it's useful. The midi port is interface 3 and has a comment in quirk-table.h like this /* interface 3 (MIDI) is standard compliant */ { .ifnum = -1 } Which is as I understand it a noop to the quirk stuff so that it just get's processed normally. Which could also be completely wrong. I've verified in lsusb -v of the midi interface (3) of the card is below Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 5 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 63 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) Warning: Descriptor too short bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 8 MIDIStreaming Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) Warning: Descriptor too short bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 97 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x07 EP 7 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 8 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x87 EP 7 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 3

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[alsa-devel] [PATCH] Enable E-mu 0204
by Joseph Teichman 12 May '11

12 May '11

This is a trivial patch to enable the E-mu 0204, which is the newest device in the series 0202USB, 0404USB and Tracker Pre. Signed-off-by: Joseph Teichman <josteich at gmail.com> --- alsa-kmirror/usb/mixer.c 2011-01-27 07:09:20.000000000 -0500 +++ alsa-kmirror-jt/usb/mixer.c 2011-02-05 20:42:00.000000000 -0500 @@ -1566,7 +1566,7 @@ cval->initialized = 1; } else { if (type == USB_XU_CLOCK_RATE) { - /* E-Mu USB 0404/0202/TrackerPre + /* E-Mu USB 0404/0202/TrackerPre/0204 * samplerate control quirk */ cval->min = 0; diff -u ./quirks.c /home/joe/alsa-driver-1.0.24-jt/sound/usb/quirks.c --- ./quirks.c 2011-01-27 07:09:20.000000000 -0500 +++ /home/joe/alsa-driver-1.0.24-jt/sound/usb/quirks.c 2011-02-05 20:40:28.000000000 -0500 @@ -532,7 +532,7 @@ } /* - * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, + * For E-Mu 0404USB/0202USB/TrackerPre/0204USB sample rate should be set for device, * not for interface. */ @@ -589,6 +589,7 @@ case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; } diff -u ./quirks-table.h /home/joe/alsa-driver-1.0.24-jt/sound/usb/quirks-table.h --- ./quirks-table.h 2011-01-27 07:09:20.000000000 -0500 +++ /home/joe/alsa-driver-1.0.24-jt/sound/usb/quirks-table.h 2011-02-05 20:41:02.000000000 -0500 @@ -79,6 +79,13 @@ .idProduct = 0x3f0a, .bInterfaceClass = USB_CLASS_AUDIO, }, +{ + /* E-Mu 0204 USB */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x041e, + .idProduct = 0x3f19, + .bInterfaceClass = USB_CLASS_AUDIO, +}, /* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface

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[alsa-devel] [PATCH 1/7] ASoC: Add control_type in snd_soc_codec
by Dimitris Papastamos 11 May '11

11 May '11

This is mainly used by the soc-cache code to easily determine the currently used underlying serial bus. Set SND_SOC_CUSTOM to 1 so we can distinguish it if it is not initialized or set. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index bfa4836..abb56ef 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -248,7 +248,7 @@ typedef int (*hw_write_t)(void *,const char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; enum snd_soc_control_type { - SND_SOC_CUSTOM, + SND_SOC_CUSTOM = 1, SND_SOC_I2C, SND_SOC_SPI, }; @@ -539,6 +539,7 @@ struct snd_soc_codec { /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ + enum snd_soc_control_type control_type; hw_write_t hw_write; unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); -- 1.7.4.1

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[alsa-devel] buffer_size return by snd_pcm_hw_params_get_buffer_size_max
by Raymond Yau 08 May '11

08 May '11

it seem that the value return by snd_pcm_hw_params_get_buffer_size_max() depend on /proc/asound/Intel/pcm0p/sub0/prealloc Is it a bug in alsa-lib or snd-hda-intel driver refer to pulseaudio/src/modules/alsa/alsa-util.c if ((ret = snd_pcm_hw_params_get_buffer_size_max(hwparams, &max_frames)) < 0) pa_log_warn("snd_pcm_hw_params_get_buffer_size_max() failed: %s", pa_alsa_strerror(ret)); else pa_log_debug("Maximum hw buffer size is %lu ms", (long unsigned) (max_frames * PA_MSEC_PER_SEC / _ss.rate)); cat /proc/asound/Intel/pcm0p/sub0/prealloc 64 D: alsa-util.c: Checking for superset analog-stereo (front:%f) D: alsa-util.c: Trying front:1 with SND_PCM_NO_AUTO_FORMAT ... D: alsa-util.c: Managed to open front:1 I: alsa-util.c: Trying to disable ALSA period wakeups, using timers only D: alsa-util.c: Maximum hw buffer size is 371 ms D: alsa-util.c: Set buffer size first (to 88200 samples), period size second (to 88200 samples). I: alsa-util.c: ALSA period wakeups disabled I: alsa-source.c: Successfully opened device front:1. I: alsa-source.c: Selected mapping 'Analog Stereo' (analog-stereo). I: alsa-source.c: Successfully enabled mmap() mode. I: alsa-source.c: Successfully enabled timer-based scheduling mode. echo 32768 > /proc/asound/Intel/pcm0p/sub0/prealloc D: alsa-util.c: Checking for superset analog-stereo (front:%f) D: alsa-util.c: Trying front:1 with SND_PCM_NO_AUTO_FORMAT ... D: alsa-util.c: Managed to open front:1 I: alsa-util.c: Trying to disable ALSA period wakeups, using timers only D: alsa-util.c: Maximum hw buffer size is 190217 ms D: alsa-util.c: Set buffer size first (to 88200 samples), period size second (to 88200 samples). I: alsa-util.c: ALSA period wakeups disabled I: alsa-sink.c: Successfully opened device front:1. I: alsa-sink.c: Selected mapping 'Analog Stereo' (analog-stereo). I: alsa-sink.c: Successfully enabled mmap() mode. I: alsa-sink.c: Successfully enabled timer-based scheduling mode.

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[alsa-devel] [PATCH] Alsa-plugins: Pulse: Fix snd_pcm_avail returning 0 in some cases
by David Henningsson 07 May '11

07 May '11

Due to a round-off error, snd_pcm_avail could in some cases return 0 even though more data could be written to the stream. This was discovered by Maarten Lankhorst [1], and there is also a test program available that triggers this error [2]. [1] https://tango.0pointer.de/pipermail/pulseaudio-discuss/2011-April/009935.ht… [2] https://tango.0pointer.de/pipermail/pulseaudio-discuss/attachments/20110420… -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[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

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[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

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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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