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[alsa-devel] aplay: xrun:1104: read/write error ???
by 13 Nov '07

13 Nov '07

Hi All: im using aplay from ALSA-1.0.15 to play music through bluetooth a2dp. im using montavista 2.6.18 on ARM9 platform. I got an "aplay: xrun:1104: read/write error, state = RUNNING" error after the music has been played for one second or two. Could someone tell me how this happened? Here's the command that i use : aplay -v -D bluetooth /1.wav And the message log is as follows: /$ aplay -v -D bluetooth /1.wav audio[842]: Accepted new client connection on unix socket (fd=8) audio[842]: Package PKT_TYPE_CFG_REQ:0 audio[842]: avdtp_ref(0x3c990): ref=2 audio[842]: a2dp_source_request_stream: selected SEP 0x3c218 audio[842]: avdtp_ref(0x3c990): ref=3 hcid[828]: link_key_request (sba=00:80:E1:00:AB:BA, dba=00:0D:3C:EB:53:F0) audio[842]: AVDTP: connected signaling channel to 00:0D:3C:EB:53:F0 audio[842]: session_cb audio[842]: DISCOVER request succeeded audio[842]: seid 1 type 1 media 0 in use 0 audio[842]: seid 2 type 1 media 0 in use 0 audio[842]: session_cb audio[842]: GET_CAPABILITIES request succeeded audio[842]: seid 1 type 1 media 0 hcid[828]: link_key_request (sba=00:80:E1:00:AB:BA, dba=00:0D:3C:EB:53:F0) hcid[828]: link_key_request (sba=00:80:E1:00:AB:BA, dba=00:0D:3C:EB:53:F0) audio[842]: session_cb audio[842]: GET_CAPABILITIES request succeeded audio[842]: seid 2 type 1 media 0 audio[842]: Discovery complete audio[842]: avdtp_set_configuration(0x3c990): int_seid=1, acp_seid=1 audio[842]: session_cb audio[842]: SET_CONFIGURATION request succeeded audio[842]: SBC Source: Set_Configuration_Cfm audio[842]: avdtp_ref(0x3c990): ref=4 audio[842]: stream state changed: IDLE -> CONFIGURED audio[842]: session_cb audio[842]: OPEN request succeeded audio[842]: stream state changed: CONFIGURED -> OPEN audio[842]: AVDTP: connected transport channel to 00:0D:3C:EB:53:F0 audio[842]: SBC Source: Open_Cfm audio[842]: session_cb audio[842]: START request succeeded audio[842]: SBC Source: Start_Cfm audio[842]: SBC Source SEP 0x3c218 locked audio[842]: fd=10, fd_opt=1, pkt_len=672, sample_size=2, rate=44100 audio[842]: 34 bytes sent audio[842]: 1 bytes sent audio[842]: avdtp_unref(0x3c990): ref=3 audio[842]: stream state changed: OPEN -> STREAMING Playing WAVE '/1.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Bluetooth Audio Device Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 6400 period_size : 128 period_time : 2902 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 128 xfer_align : 128 start_threshold : 6400 stop_threshold : 6400 silence_threshold: 0 silence_size : 0 boundary : 1677721600 hcid[828]: link_key_request (sba=00:80:E1:00:AB:BA, dba=00:0D:3C:EB:53:F0) Status(R/W): state : RUNNING trigger_time: 24.586883000 tstamp : 0.000000 delay : 0 avail : 6912 avail_max : 6912 aplay: xrun:1104: read/write error, state = RUNNING ---------------->>>>> this is where this error happens audio[842]: Unix client disconnected (fd=8) audio[842]: SBC Source SEP 0x3c218 unlocked audio[842]: avdtp_unref(0x3c990): ref=2 thanks semiyd _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

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[alsa-devel] [PATCH] cmipci - allow capture of raw spdif subframes
by Timofei V. Bondarenko 13 Nov '07

13 Nov '07

Hi. Enable capturing of raw 32bit IEC958_SUBFRAME. The 24-bits PCM data can be obtained using iec958 plugin. Known problem: captured stream may begin with either left or right subframe. Since the iec958 plugin doesn't decode preamble it may swap the channels sometime. Regards, Tim.

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[alsa-devel] patch for Dell 1210 which using stac9221 codec
by zhejiang 13 Nov '07

13 Nov '07

Hi, Dell 1210 have three jacks on the front. Two of them serve as output jack and one serve as microphone. The microphone can be switched to output model too. There are some problems with current dell 1210 driver: 1.Unable to use the front microphone 2.Unable to switch the microphone 3.If plug the left hp,another hp will be muted,so it's impossible to work as 6-channel According to the issue 0003427 on alsa-bugzilla, I wrote a patch and the reporter helped me to test it on his machine. It can support the front microphone. When switch the microphone to output model, it can mute/unmute the internal speaker with jack-sensing. diff -Nur a/alsa-driver-hg20071017/alsa-kernel/pci/hda/patch_sigmatel.c b/alsa-driver-hg20071017/alsa-kernel/pci/hda/patch_sigmatel.c --- a/alsa-driver-hg20071017/alsa-kernel/pci/hda/patch_sigmatel.c 2007-10-17 08:00:08.000000000 +0800 +++ b/alsa-driver-hg20071017/alsa-kernel/pci/hda/patch_sigmatel.c 2007-10-17 17:07:27.000000000 +0800 @@ -171,6 +171,10 @@ 0x02, }; +static hda_nid_t dell_m82_1210_dac_nids[3] = { + 0x02, 0x05, 0x03, +}; + static hda_nid_t stac925x_adc_nids[1] = { 0x03, }; @@ -874,10 +878,10 @@ 102801D7 (Dell XPS M1210) */ static unsigned int dell_922x_m82_pin_configs[10] = { - 0x0221121f, 0x408103ff, 0x02111212, 0x90100310, - 0x408003f1, 0x02111211, 0x03451340, 0x40c003f2, + 0x02211211, 0x408103ff, 0x02a1123e, 0x90100310, + 0x408003f1, 0x0221121f, 0x03451340, 0x40c003f2, 0x508003f3, 0x405003f4, -}; +}; static unsigned int d945gtp3_pin_configs[10] = { 0x0221401f, 0x01a19022, 0x01813021, 0x01014010, @@ -1512,6 +1516,15 @@ AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); } +static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, + unsigned int event) +{ + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | event)); +} + #define stac92xx_io_switch_info snd_ctl_boolean_mono_info static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1542,6 +1555,18 @@ pinctl |= stac92xx_get_vref(codec, nid); stac92xx_auto_set_pinctl(codec, nid, pinctl); } + + if (spec->board_config == STAC_922X_DELL_M82) { + if (val) { + /*If front mic serves as output,let it mute the internal speaker*/ + spec->autocfg.hp_outs = 3; + spec->autocfg.hp_pins[2] = 0x0c; + enable_pin_detect(codec, 0x0c, STAC_HP_EVENT); + } else { + spec->autocfg.hp_outs = 2; + } + codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + } return 1; } @@ -1811,9 +1836,15 @@ struct sigmatel_spec *spec = codec->spec; unsigned int wid_caps; + int max_controls; + if (spec->board_config == STAC_922X_DELL_M82) { + max_controls = spec->multiout.num_dacs; + } else { + max_controls = cfg->line_outs; + } - for (i = 0; i < cfg->line_outs; i++) { + for (i = 0; i < max_controls; i++) { if (!spec->multiout.dac_nids[i]) continue; @@ -2268,15 +2299,6 @@ AC_VERB_SET_GPIO_DATA, gpiostate); } -static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, - unsigned int event) -{ - if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | event)); -} - static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -2677,6 +2699,11 @@ spec->multiout.dac_nids = spec->dac_nids; + if (spec->board_config == STAC_922X_DELL_M82) { + spec->multiout.num_dacs = 3; + spec->multiout.dac_nids = dell_m82_1210_dac_nids; + } + err = stac92xx_parse_auto_config(codec, 0x08, 0x09); if (!err) { if (spec->board_config < 0) { @@ -2687,6 +2714,14 @@ } err = -EINVAL; } + + if ((err >= 0) && (spec->board_config == STAC_922X_DELL_M82)) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Front Mic as Output Switch", (spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC] << 8) | 1); + /*In case that add_controls realloc the spec->kctl_alloc*/ + if ((err >= 0) && (spec->kctl_alloc != spec->mixers[spec->num_mixers - 1])) + spec->mixers[spec->num_mixers - 1] = spec->kctl_alloc; + } + if (err < 0) { stac92xx_free(codec); return err;

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[alsa-devel] hardware-dependent features
by Daniel Mack 13 Nov '07

13 Nov '07

Hi, some of the devices the caiaq driver supports have special hardware features such as switchable ground lift, input impedance selection or other, hardware-specific features. On both Windows and MacOSX there are special tools to set those features but the ALSA driver still lacks support for them which I want to change soon. However, I couldn't find a generic way to do that. I was thinking about a sysfs entry which is attached to the sound card instance but didn't find a driver which did similar things. Also, I'd like to support the LEDs and displays on the devices and was wondering which interface to choose for that. I've got a patch ready which uses the generic led subsystem, but that approach seems to be a bit ugly as I end up having one entry per led on each device in one single folder. Any idea of a better way? Thanks, Daniel

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Re: [alsa-devel] No sound from usb on Motorola C350 cellphone
by Giovanni Maruzzelli 13 Nov '07

13 Nov '07

With an old copy of motorola driver, the usb sound device was recognized by windows XP, with mixer elements only for the playback. I was not able to record from it, playing give no errors, but no sound either With the last copy of driver from motorola developer devsite, on XP a whole new "Motorola Device" is created, with the audio interface listed on it in the usb control panel, but no audio device is available for normal usage. Needs more info, tests? On Nov 12, 2007 2:50 PM, Alexander E. Patrakov <patrakov(a)gmail.com> wrote: > Giovanni Maruzzelli wrote: > > Maybe there is a way to not using soundcards, that would be *so* much > > better: I found that Motorola c350 phones after receiving an AT+MODE=8 > > command via USB, make available an usb sound device. > > Confirmed with my Motorola C350 cellphone. It appears to accept S16_LE, MU_LAW > and A_LAW formats, 8 kHz sample rate, but actually records no data and plays no > sounds (even if a cal is in progress). > > One question: what playback and recording functionality is available on this > cellphone under Windows? > > -- > Alexander E. Patrakov >

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[alsa-devel] No sound from usb on Motorola C350 cellphone
by Giovanni Maruzzelli 12 Nov '07

12 Nov '07

Hi list participants, I am using cellphones connected to soundcards as cheap gsm gateways (more complex than that, actually. You can see the whole GPL project at www.celliax.org). Maybe there is a way to not using soundcards, that would be *so* much better: I found that Motorola c350 phones after receiving an AT+MODE=8 command via USB, make available an usb sound device. That would be wonderful (less cables, more robustness, etc). Problem is: the usb sound device do not plays or captures samples. I suspect that the usb audio has to be activated, maybe by an AT or a P2K command. Or that a quirk has to be added to the alsa driver (snd_usb_audio). Do you know anything about (maybe the quirk to be added to ALSA driver...)? Do you have any hint? I'm available to do all the research-testing-debugging that may be useful. What happens is as follow: 1) attach the c350 to usb 2) is recognized as acm0 3) send at+mode=8 to c350 (via minicom, for eg) 4) it changes its usbid, disconnect from usb, reconnect, is recognized as acm1, makes available a sound device 5) at commands work on the new acm1 6) trying to use the audio device directly (eg aplay -Dhw:1) do not give errors on play (but no sound is head on the phone), gives error pcm_read 1349 on record (eg arecord -Dhw:1) 7) trying to use the audiodevice through alsalib does not gives errors at all (eg -Dplughw:1) but does not caputeres samples, and no sound is heard on the phone I'm attaching: lusb_without -> result of lusb -vv with no device attached lsusb_with_ACM0 -> result of lusb -vv with c350 just attached, recognized as acm modem lsusb_with_ACM1 -> result of lusb -vv with c350 after sending at+mode=8 to him, he has disconnected and was recognized again, this time as ACM1, with a lot of additional usb interfaces dmesg-acm0 -> result of dmesg after first c350 attachment dmesg-acm1 -> result of dmesg after at+mode=8 was sent to the c350 aplay-l -> result of aplay -l showing the c350 as an alsa device stream0_pstop_cstop -> result of cat /proc/asound/C350/stream0 while not active stream0_pstop_crun -> result of cat /proc/asound/C350/stream0 while arecord -vv -Dplughw:1 -fS16_LE ciapa.raw (if using -Dhw:1 it fails with error: "pcm_read:1349: read error: Input/output error"). arecord does not gives errors, but do not write samples stream0_prun_crun -> result of cat /proc/asound/C350/stream0 while aplay and arecord -vv -Dplughw:1 -fS16_LE ciapa.raw (if using -Dhw:1 arecord fails with error: "pcm_read:1349: read error: Input/output error", aplay works but no audio) arecord does not gives errors, but do not write samples aplay_play -> results of aplay arecord_rec -> results of arecord Thank you all in advance, Giovanni Maruzzelli

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[alsa-devel] Non standard sound card
by Ning Zhang 12 Nov '07

12 Nov '07

Hi Dear, I am a fresh man in this community. Now I plan to develope a sound driver for my embedded platform. The problem is that my audio chip is a DSP ,not a standard codec chip. The MCU communicates with this DSP via a shared ram. So at first , I have to make the DSP as a platform_device and use the platform_driver_register() to register the DSP , right? Then I can use alsa driver API to develop the sound card. Is my understanding right? Thanks. Where can I find some information about ALSA architecture? David.

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[alsa-devel] [PATCH] use convenient treble scale on WM8750
by Stanislav Brabec 12 Nov '07

12 Nov '07

On Zaurus SL-C3200 (terrier/spitz) based on WM8750, treble scale is inconveniently reverted (increase level = decrease treble), in opposite to bass scale, which uses convenient scale. Fix ALSA WM8750 mixer treble to use convenient treble scale (increase = increase treble level): --- linux-2.6.23/sound/soc/codecs/wm8750.c 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23/sound/soc/codecs/wm8750.c 2007-11-02 16:47:35.000000000 +0100 @@ -189,7 +189,7 @@ SOC_ENUM("Bass Filter", wm8750_enum[1]), SOC_SINGLE("Bass Volume", WM8750_BASS, 0, 15, 1), -SOC_SINGLE("Treble Volume", WM8750_TREBLE, 0, 15, 0), +SOC_SINGLE("Treble Volume", WM8750_TREBLE, 0, 15, 1), SOC_ENUM("Treble Cut-off", wm8750_enum[2]), SOC_SINGLE("3D Switch", WM8750_3D, 0, 1, 0), ________________________________________________________________________ Stanislav Brabec http://www.penguin.cz/~utx

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[alsa-devel] I could maybe help for Creative Labs SB X-Fi integration
by Frederic Bouy 10 Nov '07

10 Nov '07

Hello everybody, I have just installed a distribution of Ubuntu (7.10 Gutsy Gibbon) on my computer. The issue is that my sound card is not working. According to your website, this is because my SB X-Fi is not yet supported by ALSA : "Card delivered to developers. Completely new architecture. Creative actively preventing support due to no datasheets being released to ALSA developers. Reverse engineering work not started due to lack of time." I would like to know if there is a way for me to help you integrating this sound card. What can I do to help? Thanks. Frédéric

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[alsa-devel] Getting XRUN after writei in RUNNING state?
by Piers O'Hanlon 10 Nov '07

10 Nov '07

Hi folks, I've trying to improve the ALSA driver for the Robust Audio Tool (RAT) tool: http://frostie.cs.ucl.ac.uk/nets/mmedia/wiki/RatWiki Specifically the code I'm talking about is: http://frostie.cs.ucl.ac.uk/nets/mmedia/browser/rat/trunk/auddev_alsa.c I've added some status dumps in the code that I'm running which aren't in the repository version but basically the code is doing the same thing. However I'm seeing weird XRUNs (under runs on the playback side) - There's as couple of instances I don't understand: (The pcm device setup is shown at the bottom); 1) RAT is happily writing audio to the playback device for while then the device just seems to go in XRUN state after being in RUNNING state and doing a writei()??: [+945 auddev_alsa.c] Time since last write: 42734us, Trying to write 640 frames state : RUNNING trigger_time: 1194622844.333930000 tstamp : 0.000000 delay : 121 avail : 1927 avail_max : 2043 [pid/9901 +954 auddev_alsa.c] Wrote 1280 bytes, frames: 640 state : XRUN trigger_time: 1194622844.333930000 tstamp : 0.000000 delay : 758 avail : 1290 avail_max : 2046 2) RAT is writing audio to the playback device for while then the device just seems to go in XRUN state when there's still positive delay in the buffer. Here's last write that went ok - showing the snd_pcm_status_dump before doing the write and afterwards: [+945 auddev_alsa.c] Time since last write: 42635us, Trying to write 640 frames state : RUNNING trigger_time: 1194622843.886113000 tstamp : 0.000000 delay : 275 avail : 1773 avail_max : 2041 [+954 auddev_alsa.c] Wrote 1280 bytes, frames: 640 state : RUNNING trigger_time: 1194622843.886113000 tstamp : 0.000000 delay : 907 avail : 1141 avail_max : 2043 The next time the audio write is called the state is checked and found to be in XRUN with 225 delay in the buffer:??? (And yet I've seen other writes find the device still running with alot less delay in the buffer??): [ +936 auddev_alsa.c] In SND_PCM_STATE_XRUN - preparing audio [pid/9901 +945 auddev_alsa.c] Frames avail to be written=2048, time diff=43182, Trying to write 640 frames, Curr delay 0 state : XRUN trigger_time: 1194622843.886113000 tstamp : 0.000000 delay : 225 avail : 1823 avail_max : 2049 [pid/9901 +954 auddev_alsa.c] Wrote 1280 bytes, frames: 640 state : RUNNING trigger_time: 1194622844.142516000 tstamp : 0.000000 delay : 637 avail : 1411 avail_max : 1708 Currently I'm stumped at to solve these problems.... And basically ALSA sound quality as worse than the OSS driver for RAT as a result. Any suggestions are to how improve or sort these problems is welcome. Thanks, Piers O'Hanlon. University College London ===snd_pcm_dump_setup:=============== stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 16000 exact rate : 16000 (16000/1) msbits : 16 buffer_size : 2048 period_size : 341 period_time : 21333 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 341 xfer_align : 1 start_threshold : 341 stop_threshold : 2048 silence_threshold: 0 silence_size : 0 boundary : 536870912

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