November 2007 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] issues with pulse plugin
by Mike Gorse 13 Nov '07

13 Nov '07

I have run into what is basically another instance of bug 3470 (pulse_hw_params asserting because the stream is already initialized). The application that I am working with is calling snd_pcm_hw_params each time it plays a sample. The following patch makes things work for me (actually the app still doesn't work because it uses poll() in a way that is incompatible with the pulse plugin, but that is a separate issue, and I don't know if it should be considered a bug in the plugin.) --- pulse/pcm_pulse.c.orig 2007-11-08 15:13:12.000000000 -0500 +++ pulse/pcm_pulse.c 2007-11-08 15:13:12.000000000 -0500 @@ -497,7 +497,7 @@ assert(pcm->p); //Resolving bugtrack ID 0003470 - if(!(base && snd_pcm_state(base) == SND_PCM_STATE_PREPARED)) + if(!(base && (snd_pcm_state(base) == SND_PCM_STATE_PREPARED || snd_pcm_state(base) == SND_PCM_STATE_SETUP))) assert(!pcm->stream); pa_threaded_mainloop_lock(pcm->p->mainloop); -- I couldn't add this to the bug because it was marked as resolved. Any comments? Or is the app behaving in a way that is counter to the specification? Also, I'm still encountering bug 2601 (a separate assert failure that happens if period_size == 0) even with the patch to pcm_ioplug.c. I think that PERIOD_TIME and PERIODS need to be constrained; I added a couple of comments to the bug. Thanks, -- Mike Gorse / AIM:linvortex / http://mgorse.freeshell.org --

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[alsa-devel] HDA Intel SigmaTel STAC9228 in Dell Vostro 1400 defines different pin nids
by Ivan N. Zlatev 13 Nov '07

13 Nov '07

Hello, I've bought a new laptop Dell Vostro 1400 with the following audio card: 00:1b.0 Audio device [Class 0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 02) and with the SigmaTel Codec: Codec: SigmaTel STAC9228 Address: 0 Vendor Id: 0x83847616 Subsystem Id: 0x10280227 Revision Id: 0x100201 I was able to get sound to come out of the internal speakers by explicitly specifying either the 3stack or the dell-3stack models with alsa 1.0.14 . Unfortunately with neither of those does the internal microphone work. Now I am running ALSA HG and have decided to try and fix my problem by adding the proper pinconfig for my card (this is what I've done before for my Mac and it worked). I downloaded the win32 drivers which include all pinconfigs for all Dell machines with intel hda cards. The problem I am facing is that the win32 driver .ini provides a configuration for 13 pins which do not match the ones in patch_sigmatel.c . The ones in ALSA are: static hda_nid_t stac922x_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x15, 0x1b, }; Where as the ones defined by the driver are [1]: 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, 0x13, 0x14, 0x21, 0x22 So I am wondering how should I approach this? I have attached my codec info and mixer output without the model to be specified. [1] win32 driver pinconfigs [HKR\Settings\pin] [HKR\Settings\Pin\0A] CfgDflt = dword: 0x02211210 AltCfg = hex: 20,12,11,02 Action = hex: 80,00,87,00,81,0D [HKR\Settings\Pin\0B] CfgDflt = dword: 0x400003FA [HKR\Settings\Pin\0C] CfgDflt = dword: 0x400003FB [HKR\Settings\Pin\0D] CfgDflt = dword: 0x90170310 ExtAmpCtrl = hex: 02,05,02 [HKR\Settings\Pin\0E] CfgDflt = dword: 0x0281103E Bias = hex: FF,04,02 AltCfg = hex: 31,10,A1,02,22,10,11,02 Action = hex: 0A,0E [HKR\Settings\Pin\0F] CfgDflt = dword: 0x0221122F AltCfg = hex: 21,12,11,02 Action = hex: 80,00,87,00,81,0D [HKR\Settings\Pin\10] CfgDflt = dword: 0x400003FC [HKR\Settings\Pin\11] CfgDflt = dword: 0x400003FD [HKR\Settings\Pin\12] CfgDflt = dword: 0x400003FE [HKR\Settings\Pin\13] CfgDflt = dword: 0x90A60330 [HKR\Settings\Pin\14] CfgDflt = dword: 0x400003FF [HKR\Settings\Pin\21] CfgDflt = dword: 0x04441340 [HKR\Settings\Pin\22] CfgDflt = dword: 0x40C003F1 [HKR\Settings\GPIO] 02 = hex: 0x82 -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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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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Re: [alsa-devel] [linux-usb-devel] [PATCH] ehci-hcd: complete iso urbs ASAP for number_of_packets != (n * 8) also
by Karsten Wiese 09 Nov '07

09 Nov '07

Am Donnerstag, 8. November 2007 schrieb David Brownell: > > Looks interesting, but could you make this apply to *BOTH* types of > ISO transfer descriptor? You did it for ITDs, but not Split-ITDs. Done. Split-ITD changes untested. > > Plus there are two non-technical problems with this patch. First, > it doesn't apply against 2.6.24-rc2-git; second, it triggers lots > of messages from scripts/checkpatch.pl ... both issues need to get > fixed before I can properly review this. Fixed. Karsten -------------------------------------> From: Karsten Wiese <fzu(a)wemgehoertderstaat.de> ehci-hcd: complete iso urbs ASAP for number_of_packets != (n * 8) also Upstream ehci-hcd only completes iso urbs, if the last frame (= 8 uframes) they overlapped with has elapsed. That can be as late as when the following urb emits its interrupt on completion. soundcard drivers tend to work around by only transfering iso urbs with number_of_packets = (n * 8). Patch lets iso urbs complete asap by scanning ITDs always up to the elapsed uframe. An ITD's last scanned uframe is stored in the new struct ehci_itd member uframe_scanned. ITDs stay kept in the hcd's schedule until the complete frame they cover has elapsed, like without patch. To make this possible, ehci->periodic_sched is changed based on the number of active itds instead of the number of active iso urbs. SITD handling is unchanged, only sitd_complete() is renamed to sitd_scan() and the active check is moved to sitd_scan() to make the "case Q_TYPE_ITD:" and "case Q_TYPE_SITD:" sections under scan_periodic() look alike. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- drivers/usb/host/ehci-sched.c | 156 ++++++++++++++++++++--------------------- drivers/usb/host/ehci.h | 3 +- 2 files changed, 76 insertions(+), 83 deletions(-) diff --git a/drivers/usb/host/ehci-sched.c b/drivers/usb/host/ehci-sched.c index 80d99bc..e647842 100644 --- a/drivers/usb/host/ehci-sched.c +++ b/drivers/usb/host/ehci-sched.c @@ -1486,6 +1486,8 @@ itd_link (struct ehci_hcd *ehci, unsigned frame, struct ehci_itd *itd) itd->frame = frame; wmb (); ehci->periodic[frame] = cpu_to_hc32(ehci, itd->itd_dma | Q_TYPE_ITD); + if (unlikely(!ehci->periodic_sched++)) + enable_periodic(ehci); } /* fit urb's itds into the selected schedule slot; activate as needed */ @@ -1558,30 +1560,28 @@ itd_link_urb ( urb->hcpriv = NULL; timer_action (ehci, TIMER_IO_WATCHDOG); - if (unlikely (!ehci->periodic_sched++)) - return enable_periodic (ehci); return 0; } #define ISO_ERRS (EHCI_ISOC_BUF_ERR | EHCI_ISOC_BABBLE | EHCI_ISOC_XACTERR) -static unsigned -itd_complete ( - struct ehci_hcd *ehci, - struct ehci_itd *itd -) { +static bool +itd_scan(struct ehci_hcd *ehci, struct ehci_itd *itd, unsigned uframe_after) +{ struct urb *urb = itd->urb; struct usb_iso_packet_descriptor *desc; u32 t; unsigned uframe; int urb_index = -1; struct ehci_iso_stream *stream = itd->stream; - struct usb_device *dev; /* for each uframe with a packet */ - for (uframe = 0; uframe < 8; uframe++) { + for (uframe = itd->uframe_scanned; uframe < uframe_after; uframe++) { + struct usb_device *dev; + if (likely (itd->index[uframe] == -1)) continue; + urb_index = itd->index[uframe]; desc = &urb->iso_frame_desc [urb_index]; @@ -1608,45 +1608,54 @@ itd_complete ( desc->status = 0; desc->actual_length = EHCI_ITD_LENGTH (t); } - } - usb_put_urb (urb); - itd->urb = NULL; - itd->stream = NULL; - list_move (&itd->itd_list, &stream->free_list); - iso_stream_put (ehci, stream); - - /* handle completion now? */ - if (likely ((urb_index + 1) != urb->number_of_packets)) - return 0; + /* handle completion now? */ + if (urb_index + 1 != urb->number_of_packets) + continue; - /* ASSERT: it's really the last itd for this urb - list_for_each_entry (itd, &stream->td_list, itd_list) - BUG_ON (itd->urb == urb); - */ + usb_put_urb(urb); + /* ASSERT: it's really the last itd for this urb + struct ehci_itd *stritd; + list_for_each_entry(stritd, &stream->td_list, itd_list) + BUG_ON(stritd != itd && stritd->urb == urb); + */ + itd->urb = NULL; - /* give urb back to the driver ... can be out-of-order */ - dev = urb->dev; - ehci_urb_done(ehci, urb, 0); - urb = NULL; + /* give urb back to the driver ... can be out-of-order */ + dev = urb->dev; + ehci_urb_done(ehci, urb, 0); + urb = NULL; - /* defer stopping schedule; completion can submit */ - ehci->periodic_sched--; - if (unlikely (!ehci->periodic_sched)) - (void) disable_periodic (ehci); - ehci_to_hcd(ehci)->self.bandwidth_isoc_reqs--; + ehci_to_hcd(ehci)->self.bandwidth_isoc_reqs--; - if (unlikely (list_empty (&stream->td_list))) { - ehci_to_hcd(ehci)->self.bandwidth_allocated + if (unlikely(list_empty(&stream->td_list))) { + ehci_to_hcd(ehci)->self.bandwidth_allocated -= stream->bandwidth; - ehci_vdbg (ehci, - "deschedule devp %s ep%d%s-iso\n", - dev->devpath, stream->bEndpointAddress & 0x0f, - (stream->bEndpointAddress & USB_DIR_IN) ? "in" : "out"); + ehci_vdbg(ehci, + "deschedule devp %s ep%d%s-iso\n", + dev->devpath, + stream->bEndpointAddress & 0x0f, + (stream->bEndpointAddress & USB_DIR_IN) ? + "in" : "out"); + } + iso_stream_put(ehci, stream); } + + itd->uframe_scanned = uframe_after; + if (uframe_after < 8) + return false; + + itd->stream = NULL; + itd->uframe_scanned = 0; + list_move(&itd->itd_list, &stream->free_list); iso_stream_put (ehci, stream); - return 1; + /* defer stopping schedule; completion can submit */ + ehci->periodic_sched--; + if (unlikely(!ehci->periodic_sched)) + disable_periodic(ehci); + + return true; } /*-------------------------------------------------------------------------*/ @@ -1950,11 +1959,9 @@ sitd_link_urb ( #define SITD_ERRS (SITD_STS_ERR | SITD_STS_DBE | SITD_STS_BABBLE \ | SITD_STS_XACT | SITD_STS_MMF) -static unsigned -sitd_complete ( - struct ehci_hcd *ehci, - struct ehci_sitd *sitd -) { +static bool +sitd_scan(struct ehci_hcd *ehci, struct ehci_sitd *sitd, int live) +{ struct urb *urb = sitd->urb; struct usb_iso_packet_descriptor *desc; u32 t; @@ -1962,9 +1969,12 @@ sitd_complete ( struct ehci_iso_stream *stream = sitd->stream; struct usb_device *dev; + t = hc32_to_cpup(ehci, &sitd->hw_results); + if (t & SITD_STS_ACTIVE && live) + return false; + urb_index = sitd->index; desc = &urb->iso_frame_desc [urb_index]; - t = hc32_to_cpup(ehci, &sitd->hw_results); /* report transfer status */ if (t & SITD_ERRS) { @@ -1991,7 +2001,7 @@ sitd_complete ( /* handle completion now? */ if ((urb_index + 1) != urb->number_of_packets) - return 0; + return true; /* ASSERT: it's really the last sitd for this urb list_for_each_entry (sitd, &stream->td_list, sitd_list) @@ -2019,7 +2029,7 @@ sitd_complete ( } iso_stream_put (ehci, stream); - return 1; + return true; } @@ -2091,13 +2101,11 @@ sitd_submit (struct ehci_hcd *ehci, struct urb *urb, gfp_t mem_flags) return -ENOSYS; } -static inline unsigned -sitd_complete ( - struct ehci_hcd *ehci, - struct ehci_sitd *sitd -) { - ehci_err (ehci, "sitd_complete %p?\n", sitd); - return 0; +static inline bool +sitd_scan(struct ehci_hcd *ehci, struct ehci_sitd *sitd) +{ + ehci_err(ehci, "%s %p?\n", __FUNCTION__, sitd); + return false; } #endif /* USB_EHCI_SPLIT_ISO */ @@ -2175,47 +2183,33 @@ restart: q = q.fstn->fstn_next; break; case Q_TYPE_ITD: - /* skip itds for later in the frame */ rmb (); - for (uf = live ? uframes : 8; uf < 8; uf++) { - if (0 == (q.itd->hw_transaction [uf] - & ITD_ACTIVE(ehci))) - continue; + modified = itd_scan(ehci, q.itd, + uf = live ? uframes : 8); + if (!modified) { q_p = &q.itd->itd_next; hw_p = &q.itd->hw_next; - type = Q_NEXT_TYPE(ehci, - q.itd->hw_next); - q = *q_p; - break; + } else { + *q_p = q.itd->itd_next; + *hw_p = q.itd->hw_next; } - if (uf != 8) - break; - - /* this one's ready ... HC won't cache the - * pointer for much longer, if at all. - */ - *q_p = q.itd->itd_next; - *hw_p = q.itd->hw_next; type = Q_NEXT_TYPE(ehci, q.itd->hw_next); + wmb(); - modified = itd_complete (ehci, q.itd); q = *q_p; break; case Q_TYPE_SITD: - if ((q.sitd->hw_results & SITD_ACTIVE(ehci)) - && live) { + modified = sitd_scan(ehci, q.sitd, live); + if (!modified) { q_p = &q.sitd->sitd_next; hw_p = &q.sitd->hw_next; - type = Q_NEXT_TYPE(ehci, - q.sitd->hw_next); - q = *q_p; - break; + } else { + *q_p = q.sitd->sitd_next; + *hw_p = q.sitd->hw_next; } - *q_p = q.sitd->sitd_next; - *hw_p = q.sitd->hw_next; type = Q_NEXT_TYPE(ehci, q.sitd->hw_next); + wmb(); - modified = sitd_complete (ehci, q.sitd); q = *q_p; break; default: diff --git a/drivers/usb/host/ehci.h b/drivers/usb/host/ehci.h index 951d69f..156c02f 100644 --- a/drivers/usb/host/ehci.h +++ b/drivers/usb/host/ehci.h @@ -581,6 +581,7 @@ struct ehci_itd { struct urb *urb; struct ehci_iso_stream *stream; /* endpoint's queue */ struct list_head itd_list; /* list of stream's itds */ + unsigned uframe_scanned; /* any/all hw_transactions here may be used by that urb */ unsigned frame; /* where scheduled */ @@ -615,8 +616,6 @@ struct ehci_sitd { #define SITD_STS_MMF (1 << 2) /* incomplete split transaction */ #define SITD_STS_STS (1 << 1) /* split transaction state */ -#define SITD_ACTIVE(ehci) cpu_to_hc32(ehci, SITD_STS_ACTIVE) - __hc32 hw_buf [2]; /* EHCI table 3-12 */ __hc32 hw_backpointer; /* EHCI table 3-13 */ __hc32 hw_buf_hi [2]; /* Appendix B */ -- 1.5.3.3

1 0

[alsa-devel] [PATCH] sound - remove dead config symbol from sound code
by Jiri Olsa 09 Nov '07

09 Nov '07

remove dead config symbols from sound code Signed-off-by: Jiri Olsa <olsajiri(a)gmail.com> --- include/sound/cs46xx.h | 3 - include/sound/driver.h | 4 -- sound/pci/cs46xx/cs46xx_lib.c | 99 ----------------------------------------- 3 files changed, 0 insertions(+), 106 deletions(-) diff --git a/include/sound/cs46xx.h b/include/sound/cs46xx.h index 6b40ee6..e3005a6 100644 --- a/include/sound/cs46xx.h +++ b/include/sound/cs46xx.h @@ -1708,9 +1708,6 @@ struct snd_cs46xx { struct gameport *gameport; -#ifdef CONFIG_SND_CS46XX_DEBUG_GPIO - int current_gpio; -#endif #ifdef CONFIG_SND_CS46XX_NEW_DSP struct mutex spos_mutex; diff --git a/include/sound/driver.h b/include/sound/driver.h index 5ccb6c5..1889929 100644 --- a/include/sound/driver.h +++ b/include/sound/driver.h @@ -38,10 +38,6 @@ #define CONFIG_SND_MAJOR 116 #endif -#ifndef CONFIG_SND_DEBUG -#undef CONFIG_SND_DEBUG_MEMORY -#endif - #ifdef ALSA_BUILD #include "adriver.h" #endif diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 2c7bfc9..8c44fef 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2084,71 +2084,6 @@ static int snd_cs46xx_spdif_stream_put(struct snd_kcontrol *kcontrol, #endif /* CONFIG_SND_CS46XX_NEW_DSP */ -#ifdef CONFIG_SND_CS46XX_DEBUG_GPIO -static int snd_cs46xx_egpio_select_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 8; - return 0; -} - -static int snd_cs46xx_egpio_select_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = chip->current_gpio; - - return 0; -} - -static int snd_cs46xx_egpio_select_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - int change = (chip->current_gpio != ucontrol->value.integer.value[0]); - chip->current_gpio = ucontrol->value.integer.value[0]; - - return change; -} - - -static int snd_cs46xx_egpio_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value; - - snd_printdd ("put: reg = %04x, gpio %02x\n",reg,chip->current_gpio); - ucontrol->value.integer.value[0] = - (snd_cs46xx_peekBA0(chip, reg) & (1 << chip->current_gpio)) ? 1 : 0; - - return 0; -} - -static int snd_cs46xx_egpio_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value; - int val = snd_cs46xx_peekBA0(chip, reg); - int oldval = val; - snd_printdd ("put: reg = %04x, gpio %02x\n",reg,chip->current_gpio); - - if (ucontrol->value.integer.value[0]) - val |= (1 << chip->current_gpio); - else - val &= ~(1 << chip->current_gpio); - - snd_cs46xx_pokeBA0(chip, reg,val); - snd_printdd ("put: val %08x oldval %08x\n",val,oldval); - - return (oldval != val); -} -#endif /* CONFIG_SND_CS46XX_DEBUG_GPIO */ - static struct snd_kcontrol_new snd_cs46xx_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -2241,40 +2176,6 @@ static struct snd_kcontrol_new snd_cs46xx_controls[] __devinitdata = { }, #endif -#ifdef CONFIG_SND_CS46XX_DEBUG_GPIO -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "EGPIO select", - .info = snd_cs46xx_egpio_select_info, - .get = snd_cs46xx_egpio_select_get, - .put = snd_cs46xx_egpio_select_put, - .private_value = 0, -}, -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "EGPIO Input/Output", - .info = snd_mixer_boolean_info, - .get = snd_cs46xx_egpio_get, - .put = snd_cs46xx_egpio_put, - .private_value = BA0_EGPIODR, -}, -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "EGPIO CMOS/Open drain", - .info = snd_mixer_boolean_info, - .get = snd_cs46xx_egpio_get, - .put = snd_cs46xx_egpio_put, - .private_value = BA0_EGPIOPTR, -}, -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "EGPIO On/Off", - .info = snd_mixer_boolean_info, - .get = snd_cs46xx_egpio_get, - .put = snd_cs46xx_egpio_put, - .private_value = BA0_EGPIOSR, -}, -#endif }; #ifdef CONFIG_SND_CS46XX_NEW_DSP

1 0

[alsa-devel] When command SNDRV_PCM_TRIGGER_STOP is asked in snd_pcm_ops trigger(), which buffer in the buffer list should be the last one to output by the DMA?
by Sonic Zhang 09 Nov '07

09 Nov '07

In our blackfin ad1981 linux driver, there is an output ring with "periods" buffers. Each buffer is of the size of "period_size". After ALSA starts playing, several buffers ahead of current one running by the DMA controller are usually filled in advance by the ALSA layer. When command SNDRV_PCM_TRIGGER_STOP is asked in snd_pcm_ops trigger(), which buffer should be the last one to output by the DMA? 1. Current buffer, which is running by the DMA controller 2. The last buffer, which the DMA controller just finished outputting and snd_pcm_ops pointer() is notifiyed to ALSA layer. Does this mean current buffer in DMA should be abandoned in the middle? 3. The last buffer, which the ALSA layer has just filled in snd_pcm_ops copy()? Thanks

2 1

[alsa-devel] User-space OSS emulation?
by Timur Tabi 09 Nov '07

09 Nov '07

Can someone tell me more about "user-space OSS emulation"? I've read a few references to it in the mailing list archives, but a google search doesn't turn up anything useful. -- Timur Tabi Linux kernel developer at Freescale

3 4

[alsa-devel] How can OSS applications support multiple streams?
by Timur Tabi 09 Nov '07

09 Nov '07

I've written an ALSA driver that supports only one playback stream, because that's what our hardware supports. I have a customer that has an OSS application that tries to open 32 simultaneous playback streams, which obviously doesn't work. What do I (or the customer) need to do to get 32 streams to work? Is this what a sound server is for? If so, which one should I use? -- Timur Tabi Linux kernel developer at Freescale

1 0

[alsa-devel] Any OSS changes from kernel 2.6.21 to 2.6.23? Something broke.
by Timur Tabi 08 Nov '07

08 Nov '07

I have a customer who says that mplayer (using OSS) does not work when he tries a kernel based on 2.6.23-rc4, but it works when he uses a kernel based on 2.6.21. Everything else is the same (including the version of alsa-lib), so I presume something broke in the OSS emulation in the kernel. Unfortunately, our git repositories for these trees are not compatible, so I can't use git-bisect to narrow the problem down. Is it possible to back-level the ALSA support in the kernel itself? Can I just download ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.XX.tar.bz2 and follow the instructions in the INSTALL file? If you update the kernel, are you supposed to use the latest alsa-lib as well? For a given kernel, how do I know what the right version of alsa-lib is? -- Timur Tabi Linux kernel developer at Freescale

4 9

[alsa-devel] Fwd: [PATCH] ehci-hcd: complete iso urbs ASAP for number_of_packets != (n * 8) also
by Karsten Wiese 08 Nov '07

08 Nov '07

Hi I just sent attached patch to usb-devel. It makes special workaround for iso usb communication using ehci-hcd highspeed needless. Comments, Bug/success reports welcome! kind regards Karsten Upstream (as of 2.6.23) ehci-hcd only completes iso urbs, if the last frame (= 8 uframes) they overlapped with has elapsed. That can be as late as when the following urb emits its interrupt on completion. soundcard drivers tend to work around by only transfering iso urbs with number_of_packets = (n * 8). Patch lets iso urbs complete asap by scanning itd always up to the elapsed uframe. An itd's last scanned uframe is stored in the new struct ehci_itd member uframe_scanned. Itds stay kept in the hcd's schedule until the complete frame they cover has elapsed, like without patch. To make this possible, ehci->periodic_sched is changed based on the number of active itds instead of the number of active iso urbs. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- drivers/usb/host/ehci-sched.c | 108 ++++++++++++++++++++--------------------- drivers/usb/host/ehci.h | 1 + 2 files changed, 54 insertions(+), 55 deletions(-) diff --git a/drivers/usb/host/ehci-sched.c b/drivers/usb/host/ehci-sched.c index e682f23..c456b16 100644 --- a/drivers/usb/host/ehci-sched.c +++ b/drivers/usb/host/ehci-sched.c @@ -1481,6 +1481,8 @@ itd_link (struct ehci_hcd *ehci, unsigned frame, struct ehci_itd *itd) itd->frame = frame; wmb (); ehci->periodic[frame] = cpu_to_hc32(ehci, itd->itd_dma | Q_TYPE_ITD); + if (unlikely (!ehci->periodic_sched++)) + enable_periodic (ehci); } /* fit urb's itds into the selected schedule slot; activate as needed */ @@ -1553,30 +1555,30 @@ itd_link_urb ( urb->hcpriv = NULL; timer_action (ehci, TIMER_IO_WATCHDOG); - if (unlikely (!ehci->periodic_sched++)) - return enable_periodic (ehci); return 0; } #define ISO_ERRS (EHCI_ISOC_BUF_ERR | EHCI_ISOC_BABBLE | EHCI_ISOC_XACTERR) -static unsigned -itd_complete ( - struct ehci_hcd *ehci, - struct ehci_itd *itd -) { +static bool +itd_scan ( + struct ehci_hcd *ehci, + struct ehci_itd *itd, + unsigned uframe_after +) +{ struct urb *urb = itd->urb; struct usb_iso_packet_descriptor *desc; u32 t; unsigned uframe; int urb_index = -1; struct ehci_iso_stream *stream = itd->stream; - struct usb_device *dev; /* for each uframe with a packet */ - for (uframe = 0; uframe < 8; uframe++) { + for (uframe = itd->uframe_scanned; uframe < uframe_after; uframe++) { if (likely (itd->index[uframe] == -1)) continue; + urb_index = itd->index[uframe]; desc = &urb->iso_frame_desc [urb_index]; @@ -1603,45 +1605,53 @@ itd_complete ( desc->status = 0; desc->actual_length = EHCI_ITD_LENGTH (t); } + + /* handle completion now? */ + if (urb_index + 1 == urb->number_of_packets) { + struct usb_device *dev; + + usb_put_urb (urb); + /* ASSERT: it's really the last itd for this urb + struct ehci_itd *stritd; + list_for_each_entry (stritd, &stream->td_list, itd_list) + BUG_ON (stritd != itd && stritd->urb == urb); + */ + itd->urb = NULL; + + /* give urb back to the driver ... can be out-of-order */ + dev = urb->dev; + ehci_urb_done (ehci, urb); + urb = NULL; + + ehci_to_hcd(ehci)->self.bandwidth_isoc_reqs--; + + if (unlikely (list_empty (&stream->td_list))) { + ehci_to_hcd(ehci)->self.bandwidth_allocated + -= stream->bandwidth; + ehci_vdbg (ehci, + "deschedule devp %s ep%d%s-iso\n", + dev->devpath, stream->bEndpointAddress & 0x0f, + (stream->bEndpointAddress & USB_DIR_IN) ? "in" : "out"); + } + iso_stream_put (ehci, stream); + } } - usb_put_urb (urb); - itd->urb = NULL; + itd->uframe_scanned = uframe_after; + if (uframe_after < 8) + return false; + itd->stream = NULL; + itd->uframe_scanned = 0; list_move (&itd->itd_list, &stream->free_list); iso_stream_put (ehci, stream); - /* handle completion now? */ - if (likely ((urb_index + 1) != urb->number_of_packets)) - return 0; - - /* ASSERT: it's really the last itd for this urb - list_for_each_entry (itd, &stream->td_list, itd_list) - BUG_ON (itd->urb == urb); - */ - - /* give urb back to the driver ... can be out-of-order */ - dev = urb->dev; - ehci_urb_done (ehci, urb); - urb = NULL; - /* defer stopping schedule; completion can submit */ ehci->periodic_sched--; if (unlikely (!ehci->periodic_sched)) (void) disable_periodic (ehci); - ehci_to_hcd(ehci)->self.bandwidth_isoc_reqs--; - if (unlikely (list_empty (&stream->td_list))) { - ehci_to_hcd(ehci)->self.bandwidth_allocated - -= stream->bandwidth; - ehci_vdbg (ehci, - "deschedule devp %s ep%d%s-iso\n", - dev->devpath, stream->bEndpointAddress & 0x0f, - (stream->bEndpointAddress & USB_DIR_IN) ? "in" : "out"); - } - iso_stream_put (ehci, stream); - - return 1; + return true; } /*-------------------------------------------------------------------------*/ @@ -2156,30 +2166,18 @@ restart: q = q.fstn->fstn_next; break; case Q_TYPE_ITD: - /* skip itds for later in the frame */ rmb (); - for (uf = live ? uframes : 8; uf < 8; uf++) { - if (0 == (q.itd->hw_transaction [uf] - & ITD_ACTIVE(ehci))) - continue; + modified = itd_scan(ehci, q.itd, uf = live ? uframes : 8); + if (!modified) { q_p = &q.itd->itd_next; hw_p = &q.itd->hw_next; - type = Q_NEXT_TYPE(ehci, - q.itd->hw_next); - q = *q_p; - break; + } else { + *q_p = q.itd->itd_next; + *hw_p = q.itd->hw_next; } - if (uf != 8) - break; - - /* this one's ready ... HC won't cache the - * pointer for much longer, if at all. - */ - *q_p = q.itd->itd_next; - *hw_p = q.itd->hw_next; type = Q_NEXT_TYPE(ehci, q.itd->hw_next); + wmb(); - modified = itd_complete (ehci, q.itd); q = *q_p; break; case Q_TYPE_SITD: diff --git a/drivers/usb/host/ehci.h b/drivers/usb/host/ehci.h index 951d69f..df35964 100644 --- a/drivers/usb/host/ehci.h +++ b/drivers/usb/host/ehci.h @@ -581,6 +581,7 @@ struct ehci_itd { struct urb *urb; struct ehci_iso_stream *stream; /* endpoint's queue */ struct list_head itd_list; /* list of stream's itds */ + unsigned uframe_scanned; /* any/all hw_transactions here may be used by that urb */ unsigned frame; /* where scheduled */ -- 1.5.3.3

2 6

[alsa-devel] Audio quality issues with dmix plugin
by Pharaoh . 08 Nov '07

08 Nov '07

Hi, When I play a 44Khz file without using any ALSA plugin like #aplay -Dhw:0,0 44k.wav the sound quality is awesome but when I play the same file using dmix, clicks are introduced. #aplay -Dplug:dmix 44k.wav changing period size and buffer size doesn't help. I am testing my ALSA driver on an OMAP850 based board. Regards, Pharaoh.

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Re: [alsa-devel] snd_pcm_hw_params_get_xxx question
by Ben C 08 Nov '07

08 Nov '07

Takashi, You were right, it was a linking issue. I'm doing cross-compiling for a target of different processor type. I just had to make sure I'm pointing to the right location for the includes and libs. Thanks, Ben ----- Original Message ---- From: Takashi Iwai <tiwai(a)suse.de> To: Ben C <bchang_2(a)yahoo.com> Cc: alsa-devel(a)alsa-project.org Sent: Wednesday, November 7, 2007 4:24:04 AM Subject: Re: [alsa-devel] snd_pcm_hw_params_get_xxx question At Tue, 6 Nov 2007 18:21:01 -0800 (PST), Ben C wrote: > > Hello, > > I'm trying to develop an external PCM plugin. I'm calling snd_pcm_hw_params_get_xxx functions to retrieve some values. I got everything to work as expected on alsa-lib 1.0.15. However the client's setup is 1.0.13, and I don't think they want to upgrade. > > For 1.0.13, everything compiled and linked fine after some minor changes. But when I called say snd_pcm_hw_params_get_rate(hw, &val, &dir), I was expecting the return value to be a status, and val to contain "44100" which was the case for 1.0.15. However for 1.0.13, val is 0, but returned status = 44100? > > I looked in the header files, and 1.0.13 seems to be the same as 1.0.15 in regards to this function. Am I accidentally using some sort of old API that was intended for backward compatibility? If so, what should I do to use the "new" API's? Yes, it sounds like you're accessing the old API. Do you use libasound from a module? If it's used from a module, the module itself should be built once with link to libasound. For example, $(CC) -shared -o MODULE.so -lasound In such a way, the properly versioned symbol is chosen. The weak link results in the old API. This is irrelveant whether it's ALSA 1.0.13 or 1.0.15. Takashi __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] [HDA Intel/STAC92xx] Alsa 1.0.14
by Greg Jizmagian 07 Nov '07

07 Nov '07

I am trying to fix the sound on the m1330 in OS X. It currently only works out of one of the headphone jacks. I stumbled across the patch you wrote for ubuntu and hoped that I could adapt it for OS X. Or is there a way to see a post patch linux dump that would fix the problem? Any suggestions/help would be greatly appreciated. Thanks. Greg

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[alsa-devel] snd_pcm_hw_params_get_xxx question
by Ben C 07 Nov '07

07 Nov '07

Hello, I'm trying to develop an external PCM plugin. I'm calling snd_pcm_hw_params_get_xxx functions to retrieve some values. I got everything to work as expected on alsa-lib 1.0.15. However the client's setup is 1.0.13, and I don't think they want to upgrade. For 1.0.13, everything compiled and linked fine after some minor changes. But when I called say snd_pcm_hw_params_get_rate(hw, &val, &dir), I was expecting the return value to be a status, and val to contain "44100" which was the case for 1.0.15. However for 1.0.13, val is 0, but returned status = 44100? I looked in the header files, and 1.0.13 seems to be the same as 1.0.15 in regards to this function. Am I accidentally using some sort of old API that was intended for backward compatibility? If so, what should I do to use the "new" API's? Thanks, Ben __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] [PATCH] hda: STAC9228 Subsystem update
by Matthew Ranostay 07 Nov '07

07 Nov '07

Added more laptops subsystem id's that have STAC9228 DMIC support. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 43708ece3f6f pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Nov 06 11:56:17 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Nov 06 14:01:53 2007 -0500 @@ -2921,8 +2921,10 @@ static int patch_stac927x(struct hda_cod } switch (codec->subsystem_id) { - case 0x1028020A: /* STAC 9228 */ - case 0x10280209: /* STAC 9228 */ + case 0x10280242: /* STAC 9228 */ + case 0x102801f3: + case 0x1028020A: + case 0x10280209: spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; spec->dmux_nid = 0x1b;

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[alsa-devel] [PATCH] cmipci - utilize ADC48K44K bit
by Timofei V. Bondarenko 07 Nov '07

07 Nov '07

Hi. Setting the ADC48K44K greatly improves capture quality at 48k sampling rate. With this bit clear ADC does ZOH interpolation of every 22th sample at 48k. At frequencies higher than 48k there ADC performs a little better with ADC48K44K bit set. At 44.1k ADC performs a little better with this bit clear. At frequencies below 44.1k there is no difference. Regards, Tim.

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