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

[alsa-devel] [PATCH] Support for muting channels DAC with CA0106 - SB0570
by Trent Piepho 24 Jul '07

24 Jul '07

This patch adds mute controls for the analog output channels. It does this by sending SPI commands to the DAC on cards like the SB live 24-bit / Audigy SE that have an SPI DAC.

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[alsa-devel] basic workflow when playing some sounddata
by Markus Franke 24 Jul '07

24 Jul '07

Dear Alsa-Developers, I am very new to Alsa driver development and have some problems to get a clue how the process of playing sounddata works. Let me explain my point of view and it would be nice if anybody could give some clarifications and corrections on it. Suppose I want to play a *.wav file from userspace with "aplay". After triggering all the configuration functions for PCM and the DAI at some point the chip_pcm_trigger() function get's called. As per "Writing Alsa Drivers" the TRIGGER_START command should now do something for "enabling the PCM engine". As per my understanding this could be for example to enable the interrupts of my DAI, which is (I2S over GSI) in my case. The interrupt handler by itself should call snd_pcm_period_elapsed() whenever a period of sounddata is copied to the device. Furthermore the so called "hw_pointer" should be updated in the interrupt handler, right? To make it easy I don't want to use DMA in the beginning. I simply want to copy the data to the hardware buffer of my DAI (I2S via GSI) and from there it is copied automatically to my codec. The question is now, at which place in the alsa framework I have to put the copy process of the sounddata. It should not be done in the interrupt handler because this would take to much time. But whereelse could this be done? Also I don't know the meaning of chip_pcm_pointer() callback. As far as I could figure out from the source, this function is called whenever snd_pcm_period_elapsed() is called, right? But what should be done exactly in this callback. Again I want to say that I don't want to use DMA at all by now. I've read already "Writing Alsa Drivers" but the questions above still remain for me and I am quite confused as you maybe noticed already from my questions above. ;-) It would be nice if anybody can give me a basic workflow which functions/callbacks are necessary for the playing part of an Alsa driver. Thanks in advance for any help, Regards, Markus

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[alsa-devel] SuSE 9.1 on Toshiba Satellite A105 - S4004
by Neeraj Vaidya 24 Jul '07

24 Jul '07

Hi, I've installed this distro on my toshiba laptop. In YaST , when I try to configure the sound card , it says no sound card detected. To cut a long story short , I do not get any sounds to work on my laptop. Sound works beautifully on XP , but I am trying to migrate all of my routine activities to linux ;) I am a newbie to the ALSA mailing list. Have a feeling that something to do with ALSA configuration can help me sort out this problem. When I run lspci , I do not see any audio card as being detected. Can someone please help me out ? Also , send me some diagnostic steps , if not the entire solution , so that we can sort this out in a series of emails , if the entire solution can not be provided in one email. Thanks in Advance! -Neeraj --------------------------------- Get the freedom to save as many mails as you wish. Click here to know how.

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[alsa-devel] [PATCH] s3c24xx-pcm: fix hw_params dma handling
by Arnaud Patard 24 Jul '07

24 Jul '07

Hi, While testing the recently merged driver for the neo1973, I found out that it was not working due to dma allocation failure. The problem is that there's only one dma channel available on s3c2410 and there may be more than one call to pcm_ops->hw_params(), resulting in dma request failures. This patch from Harald Welte fix this. Arnaud

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[alsa-devel] Intel HDA digital out distortion
by Ingmar Stieger 24 Jul '07

24 Jul '07

Hi, a couple of weeks ago, I bought an ASUS mainboard with integrated Intel HDA and an AD1988B codec. Unfortunately, the digital / SPDIF output gets distorted with stock 1.0.14 drivers. It does not matter what the volume levels of the various mixers are set to. I think I might have found the source for this problem. Maybe it's only a workaround, but it solved the issue and digital out now works fine for me, so I thought I'd share. In patch_analog.c, in static struct hda_verb ad1988_spdif_init_verbs[], changeing the forth entry from {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, to {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)} mutes what might be an feedback loop from the analog inputs. The distortion is gone if the input to this widget is muted. On a side note, the settings for the ADCs with NID:08, 09, and 0F looks fishy, too. According to the functional diagram, those should have no amps, and thus should not be muteable. The driver mutes them anyway... I'd like to help somebody who really understands this stuff to solve the problem for good. What would the next step be ? Thanks, Ingmar

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[alsa-devel] basic workflow when playing some sounddata
by Markus Franke 24 Jul '07

24 Jul '07

Dear Alsa-Developers, I am very new to Alsa driver development and have some problems to get a clue how the process of playing sounddata works. Let me explain my point of view and it would be nice if anybody could give some clarifications and corrections on it. Suppose I want to play a *.wav file from userspace with "aplay". After triggering all the configuration functions for PCM and the DAI at some point the chip_pcm_trigger() function get's called. As per "Writing Alsa Drivers" the TRIGGER_START command should now do something for "enabling the PCM engine". As per my understanding this could be for example to enable the interrupts of my DAI, which is (I2S over GSI) in my case. The interrupt handler by itself should call snd_pcm_period_elapsed() whenever a period of sounddata is copied to the device. Furthermore the so called "hw_pointer" should be updated in the interrupt handler, right? To make it easy I don't want to use DMA in the beginning. I simply want to copy the data to the hardware buffer of my DAI (I2S via GSI) and from there it is copied automatically to my codec. The question is now, at which place in the alsa framework I have to put the copy process of the sounddata. It should not be done in the interrupt handler because this would take to much time. But whereelse could this be done? Also I don't know the meaning of chip_pcm_pointer() callback. As far as I could figure out from the source, this function is called whenever snd_pcm_period_elapsed() is called, right? But what should be done exactly in this callback. Again I want to say that I don't want to use DMA at all by now. I've read already "Writing Alsa Drivers" but the questions above still remain for me and I am quite confused as you maybe noticed already from my questions above. ;-) It would be nice if anybody can give me a basic workflow which functions/callbacks are necessary for the playing part of an Alsa driver. Thanks in advance for any help, Regards, Markus

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[alsa-devel] basic workflow when playing some sounddata
by Markus Franke 24 Jul '07

24 Jul '07

Dear Alsa-Developers, I am very new to Alsa driver development and have some problems to get a clue how the process of playing sounddata works. Let me explain my point of view and it would be nice if anybody could give some clarifications and corrections on it. Suppose I want to play a *.wav file from userspace with "aplay". After triggering all the configuration functions for PCM and the DAI at some point the chip_pcm_trigger() function get's called. As per "Writing Alsa Drivers" the TRIGGER_START command should now do something for "enabling the PCM engine". As per my understanding this could be for example to enable the interrupts of my DAI, which is (I2S over GSI) in my case. The interrupt handler by itself should call snd_pcm_period_elapsed() whenever a period of sounddata is copied to the device. Furthermore the so called "hw_pointer" should be updated in the interrupt handler, right? To make it easy I don't want to use DMA in the beginning. I simply want to copy the data to the hardware buffer of my DAI (I2S via GSI) and from there it is copied automatically to my codec. The question is now, at which place in the alsa framework I have to put the copy process of the sounddata. It should not be done in the interrupt handler because this would take to much time. But whereelse could this be done? Also I don't know the meaning of chip_pcm_pointer() callback. As far as I could figure out from the source, this function is called whenever snd_pcm_period_elapsed() is called, right? But what should be done exactly in this callback. Again I want to say that I don't want to use DMA at all by now. I've read already "Writing Alsa Drivers" but the questions above still remain for me and I am quite confused as you maybe noticed already from my questions above. ;-) It would be nice if anybody can give me a basic workflow which functions/callbacks are necessary for the playing part of an Alsa driver. Thanks in advance for any help, Regards, Markus

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[alsa-devel] SNDRV_PCM_FMTBIT_S24_LE vs. SNDRV_PCM_FMTBIT_S24_3LE
by Timur Tabi 24 Jul '07

24 Jul '07

What is the difference between SNDRV_PCM_FMTBIT_S24_LE and SNDRV_PCM_FMTBIT_S24_3LE? -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] [PATCH] Fix Kconfig entry for SND_S3C24XX_SOC_NEO1973_WM8753
by Arnaud Patard 23 Jul '07

23 Jul '07

SND_S3C24XX_SOC_NEO1973_WM8753 depends on MACH_GTA01 but the Kconfig entry which is going to be merged is MACH_NEO1973_GTA01. signed-off-by: Arnaud Patard <arnaud.patard(a)rtp-net.org>

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[alsa-devel] [PATCH] Fix tas_suspend/resume build warning
by Stephen Rothwell 23 Jul '07

23 Jul '07

sound/aoa/codecs/snd-aoa-codec-tas.c:750: warning: 'tas_suspend' defined but not used sound/aoa/codecs/snd-aoa-codec-tas.c:760: warning: 'tas_resume' defined but not used Signed-off-by: Stephen Rothwell <sfr(a)canb.auug.org.au> --- sound/aoa/codecs/snd-aoa-codec-tas.c | 9 +++++---- 1 files changed, 5 insertions(+), 4 deletions(-) -- Cheers, Stephen Rothwell sfr(a)canb.auug.org.au diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/snd-aoa-codec-tas.c index 2f771f5..3cbfe46 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas.c +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c @@ -743,6 +743,7 @@ static int tas_switch_clock(struct codec_info_item *cii, enum clock_switch clock return 0; } +#ifdef CONFIG_PM /* we are controlled via i2c and assume that is always up * If that wasn't the case, we'd have to suspend once * our i2c device is suspended, and then take note of that! */ @@ -768,7 +769,6 @@ static int tas_resume(struct tas *tas) return 0; } -#ifdef CONFIG_PM static int _tas_suspend(struct codec_info_item *cii, pm_message_t state) { return tas_suspend(cii->codec_data); @@ -778,7 +778,10 @@ static int _tas_resume(struct codec_info_item *cii) { return tas_resume(cii->codec_data); } -#endif +#else /* CONFIG_PM */ +#define _tas_suspend NULL +#define _tas_resume NULL +#endif /* CONFIG_PM */ static struct codec_info tas_codec_info = { .transfers = tas_transfers, @@ -791,10 +794,8 @@ static struct codec_info tas_codec_info = { .owner = THIS_MODULE, .usable = tas_usable, .switch_clock = tas_switch_clock, -#ifdef CONFIG_PM .suspend = _tas_suspend, .resume = _tas_resume, -#endif }; static int tas_init_codec(struct aoa_codec *codec) -- 1.5.2.3

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[alsa-devel] [PATCH] hda-intel: support for iMac 24'' released on 09/2006
by Nicola Fagnani 23 Jul '07

23 Jul '07

Basic audio support for the iMac 24'' model released on 09/2006, including headphone jack detection with automatic speaker muting. This iMac uses the Realtek ALC885 codec, not a Sigmatel one as in other models. Functionality has been tested for internal speakers, headphone and microphone. Signed-off-by: Nicola Fagnani <nicfagn(a)iol.it> --- diff -ru alsa-driver-1.0.14/alsa-kernel/Documentation/ALSA- Configuration.txt alsa-driver-1.0.14-patched/alsa-kernel/ Documentation/ALSA-Configuration.txt --- alsa-driver-1.0.14/alsa-kernel/Documentation/ALSA- Configuration.txt 2007-06-04 09:28:52.000000000 +0200 +++ alsa-driver-1.0.14-patched/alsa-kernel/Documentation/ALSA- Configuration.txt 2007-07-19 20:16:24.000000000 +0200 @@ -827,6 +827,7 @@ 6stack-dig 6-jack digital with SPDIF I/O arima Arima W820Di1 macpro MacPro support + imac24 iMac 24'' with jack detection w2jc ASUS W2JC auto auto-config reading BIOS (default) diff -ru alsa-driver-1.0.14/alsa-kernel/pci/hda/patch_realtek.c alsa- driver-1.0.14-patched/alsa-kernel/pci/hda/patch_realtek.c --- alsa-driver-1.0.14/alsa-kernel/pci/hda/patch_realtek.c 2007-06-04 09:28:54.000000000 +0200 +++ alsa-driver-1.0.14-patched/alsa-kernel/pci/hda/patch_realtek.c 2007-07-06 12:19:30.000000000 +0200 @@ -144,6 +144,7 @@ ALC882_TARGA, ALC882_ASUS_A7J, ALC885_MACPRO, + ALC885_IMAC24, ALC882_AUTO, ALC882_MODEL_LAST, }; @@ -5054,6 +5055,59 @@ { } }; +/* iMac 24 mixer. */ +static struct snd_kcontrol_new alc885_imac24_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT), + { } /* end */ +}; +/* iMac 24 init verbs. */ +static struct hda_verb alc885_imac24_init_verbs[] = { + /* Internal speakers: output 0 (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Internal speakers: output 0 (0x0c) */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Headphone: output 0 (0x0c) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Front Mic: input vref at 80% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } +}; + +/* Toggle speaker-output according to the hp-jack state */ +static void alc885_imac24_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_update(codec, 0x18, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x18, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); +} +/* Processes unsolicited events. */ +static void alc885_imac24_unsol_event(struct hda_codec *codec, unsigned int res) +{ + /* Headphone insertion or removal. */ + if (((res >> 26) == ALC880_HP_EVENT)) { + alc885_imac24_automute(codec); + } +} + + static struct hda_verb alc882_targa_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -5274,6 +5328,7 @@ [ALC882_ARIMA] = "arima", [ALC882_W2JC] = "w2jc", [ALC885_MACPRO] = "macpro", + [ALC885_IMAC24] = "imac24", [ALC882_AUTO] = "auto", }; @@ -5345,6 +5400,19 @@ .channel_mode = alc882_ch_modes, .input_mux = &alc882_capture_source, }, + [ALC885_IMAC24] = { + .mixers = { alc885_imac24_mixer }, + .init_verbs = { alc885_imac24_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), + .channel_mode = alc882_ch_modes, + .input_mux = &alc882_capture_source, + .unsol_event = alc885_imac24_unsol_event, + .init_hook = alc885_imac24_automute, + }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer, alc882_capture_mixer }, @@ -5494,6 +5562,9 @@ case 0x106b0c00: /* Mac Pro */ board_config = ALC885_MACPRO; break; + case 0x106b1000: /* iMac 24 */ + board_config = ALC885_IMAC24; + break; default: printk(KERN_INFO "hda_codec: Unknown model for ALC882, " "trying auto-probe from BIOS...\n"); @@ -5518,7 +5589,7 @@ if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); - if (board_config == ALC885_MACPRO) { + if (board_config == ALC885_MACPRO || board_config == ALC885_IMAC24) { alc882_gpio_mute(codec, 0, 0); alc882_gpio_mute(codec, 1, 0); }

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[alsa-devel] Close bug 3113?
by Alan Horstmann 22 Jul '07

22 Jul '07

Since bug-track notes don't come to alsa-devel, I'm not sure where/who to send this to. I came across the link to this tidying my bookmarks. Looks like it can be closed now, which = one less. Alan

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[alsa-devel] [BUG] Open Kernel Bugs
by Natalie Protasevich 21 Jul '07

21 Jul '07

This is the first listing of the open bugs that will be sent out and will contain bugs on specific areas. The bugs on the list below are about a year old and most of them confirmed (or claimed) to still be a problem. It would be appreciated if the corresponding maintenance team could take a look, close off any which are fixed and see if we can fix any which aren't. NOTE: when replying to this email, please add the bug number to the Subject in the form [Bug 1234] so that bugzilla will capture the discussion. Thanks. bugzilla: timers [BUG 7967] TIMER_ABSTIME cleared resulting in userland deadlock http://bugzilla.kernel.org/show_bug.cgi?id=7967 bugzilla: parallel [BUG 7492] parport doesn't work if DMA is used http://bugzilla.kernel.org/show_bug.cgi?id=7492 bugzilla: network [BUG 8106] tx_errors and tx_fifo_errors not updated consistantly http://bugzilla.kernel.org/show_bug.cgi?id=8106 - error reporting [BUG 7856] b44 WOL problem (bcm4401) http://bugzilla.kernel.org/show_bug.cgi?id=7856 [BUG 5839] uli526x partially recognizing interface http://bugzilla.kernel.org/show_bug.cgi?id=5839 [BUG 6613] iptables broken on 32-bit PReP (ARCH=ppc) http://bugzilla.kernel.org/show_bug.cgi?id=6613 bugzilla: sound (ALSA) [BUG 7195] pc doesn't correctly shut down because of hda_intel http://bugzilla.kernel.org/show_bug.cgi?id=7195 [BUG 6244] cs46xx in Thinkapad A21m after suspend/resume doesn't play http://bugzilla.kernel.org/process_bug.cgi?id=6244 bugzilla: scheduler [BUG 7372] Unusable system (ie slow) when copying large files. http://bugzilla.kernel.org/show_bug.cgi?id=7372 bugzilla: input [BUG 7746] Support for unicode dead keys http://bugzilla.kernel.org/show_bug.cgi?id=7746 [BUG 7941] The atkbd module makes system unusable http://bugzilla.kernel.org/process_bug.cgi?id=7941 bugzilla: dvb-usb [BUG 8179] Yuan ub300 card not working/tuning http://bugzilla.kernel.org/show_bug.cgi?id=8179 bugzilla: PCI [BUG 7528] pci=assign-busses needed on DELL Latitude D810 http://bugzilla.kernel.org/show_bug.cgi?id=7528 [BUG 8187] 2.6.20 "PCI: Quirks" patch breaks X11 on I82801 http://bugzilla.kernel.org/show_bug.cgi?id=8187 [BUG 7917] "PCI: Failed to allocate mem resource" for PCI E-to-PCI Bridge http://bugzilla.kernel.org/show_bug.cgi?id=7917 bugzilla: i386 [BUG 5565] Guess of i386 APIC PTE area scribble http://bugzilla.kernel.org/show_bug.cgi?id=5565 [BUG 7589] libata: does not work anymore on VIA VT8251 AHCI/SATA 4-Port Controller http://bugzilla.kernel.org/show_bug.cgi?id=7589 bugzilla: Interrupts [BUG 7673] Memory card reader 5-in-1 don't working on Lenovo N100 3000 http://bugzilla.kernel.org/show_bug.cgi?id=7673 bugzilla: I/O - Storage [BUG 7921] oops in 2.6.20-rc6-git3 &&udf && pktcdvd http://bugzilla.kernel.org/show_bug.cgi?id=7921 [BUG 7507] pata_it821x: system freezes http://bugzilla.kernel.org/show_bug.cgi?id=7507 [BUG 7506] it821x: long wait time due to dma_timer_expiry http://bugzilla.kernel.org/show_bug.cgi?id=7506 [BUG 7703] hpt366 does not find drives http://bugzilla.kernel.org/show_bug.cgi?id=7703 [BUG 6516] wrong value for avgqu-sz http://bugzilla.kernel.org/show_bug.cgi?id=6516 bugzilla: serial [BUG 7750] open/close loop causes disabled irq http://bugzilla.kernel.org/show_bug.cgi?id=7750 bugzilla: PCMCIA [BUG 7788] System freeze as soon as PCMCIA card is inserted http://bugzilla.kernel.org/show_bug.cgi?id=7788 [BUG 8118] Wrong device identifiers returned in lspci for pccard with RL5c476 II (rev 80) http://bugzilla.kernel.org/show_bug.cgi?id=8118 [BUG 8262] PCMCIA: socket *** DANGER *** unable to remove socket power http://bugzilla.kernel.org/show_bug.cgi?id=8262 [BUG 7215] PCMCIA network card causes either X or kernel to freeze http://bugzilla.kernel.org/show_bug.cgi?id=7215 bugzilla: SCSI [BUG 4880] dpt_i2o.c does not register itself with sysfs http://bugzilla.kernel.org/show_bug.cgi?id=4880 bugzilla: File Systems [BUG 6124] mount of UDF fs ignores UID and GID options http://bugzilla.kernel.org/show_bug.cgi?id=6124 bugzilla: USB [BUG 8104] kernel floods console with "connect-debounce failed" http://bugzilla.kernel.org/show_bug.cgi?id=8104 ----------------------------------------------------------------------------- ----------------------------------------------------------------------------- ----------------------------- -------------------------------------------------------

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[alsa-devel] ALSA-USB driver issues with C-Media chipset
by Saurav SAHU 20 Jul '07

20 Jul '07

We have encountered some audio recording quality issues on the recent ALSA-USB drivers. These problems are faced on C-Media chipset based external USB audio devices. The recorded audio with 'arecord' is quite noisy, unclear and jerky. The version of ALSA I encountered are 1.0.12 and beyond. It seemed to work well previously on ALSA version 1.0.8. If I record with a Philips webcam mic (Philips chipset), the audio recording is very clear on across all ALSA versions. In case you have observed similar issues with C-Media drivers and aware of them, please let me know. Best Regards Saurav

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[alsa-devel] Noise Problem with rate convert plugins
by Andreas Rumpler 20 Jul '07

20 Jul '07

Hello, I'm new to this list.I hope someone could help me with a problem concerning the samplerate converter plugins of the alsa driver. Firstly some facts about my application. - Envy24 (VT1270) pci audio chip on a custom mainboard - playback of sounds/music with different sample rates from 8kHz to 48kHz (sample rate of the files) - the hardware (Envy24 chip) must always run with 44.1kHz or 48kHz (selectable by user), because the audio is outputed by a DAC and SPDIF and the SPDIF clock must not be changed according to the sample rate of the files. - Kernel is 2.6.22 and alsa is 1.0.14 So I'm using the samplerate (libsample rate based) plugin from the alsa-plugin package. Generaly it works very good. My problem is: Every time a playback is started a short plop noise is heard. The noise is on analog and SPDIF output. If I don't use the samplerate converter the noise is not heard at all. So it comes definitely from the rate converter plugin, I think. Changing the quality level of the plugin don't solve the problem. I've also switched to the new Speex rate converter plugin. With this plugin there is no noise at the start of playback. But here I have trouble with awful noise at some rate conversions (6kHz(source) -> 48kHz(output); 11.025kHz -> 44.1kHz; 22.05kHz -> 44.1kHz). So it's even worse than using the libsamplerate plugin. Finaly I've tried the libavcodec plugin, which is the worst according to noise. Could any comment this problem, please? This is my asound.conf: ############################################## # Playback over Line-Out and SPDIF simultaniously as default pcm.!default { type plug slave.pcm { type multi slaves.a.pcm "plughw:0,0" slaves.a.channels 2 slaves.b.pcm "hw:0,1" slaves.b.channels 2 bindings.0.slave a bindings.0.channel 0 bindings.1.slave a bindings.1.channel 1 bindings.2.slave b bindings.2.channel 0 bindings.3.slave b bindings.3.channel 1 } ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 } pcm_slave.s1 { pcm default rate 44100 } pcm_slave.s2 { pcm default rate 48000 } pcm.rate_convert_44100 { type rate slave s1 converter samplerate } pcm.rate_convert_48000 { type rate slave s2 converter samplerate } ##################################################### Regards Andreas Rumpler

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[alsa-devel] The users view of an audio card
by David Webb 20 Jul '07

20 Jul '07

I've been editing a couple of the unofficial wiki pages and a number of issues have arisen. My own system has the NVidia/Realtek motherboard chips so I may be in the dark about what is available on other chips. Some of the questions are background to ensure that what I write later in the wiki is correct. Takashi Iwai suggested I raise the issues here: 1. A large number of users seem to be reporting problems with Nvidia and Via soundcards. As far as I can tell Nvidia and Via do not produce soundcards but they do produce chipsets with southbridges which can connect to a motherboard audio chip. The when alsamixer (or Kmix) runs it reports something like: Card: NVidia CK804 Chip: Realtek AL650 rev 0 Presumably because the southbridge is the primary PCI device. So: 1.1 Is the above always correct (so I can add it to the wiki)? 1.2 Could alsamixer etc be modified to give a more sensible answer for non-experts. Something like: Card: Motherboard via NVidia CK804 Chip: Realtek AL650 rev 0 might work. I presume a look-up table is used to convert the PCI 2-byte integers to the above names so presumably the extra information needed could be obtained at this level. 2. On different motherboards a Realtek 650 may be connected using a NVidia CK804 with the intel8x0 alsa driver or using a VIA VT8233 with the vt82xx driver. 2.1 Given that the audio chip registers are the same in both cases why are two different drivers needed? (Are they?) 2.2 If it is something to do with the southbridges, where may the manuals that describe the differences be found? 2.3 What would happen if the motherboard manufacturers kept the same southbridges but started using an audio chip with a different set of registers? (2.2a Where can an AD'97 manual be found? - I could not find one on the intel site) 3. The Realtek 650 datasheet implies that independently of the mixing in the main input -> line-out pathway, the input to the ADC (i.e. on capture) may include more than one source (i.e. both audio and microphone). However alsamixer (and kmix) only allows me one input stream. So: 3.1 Is the manual misleading or have I misread it? 3.2 Is the manual correct but the driver is based on an earlier intel chip which did not have this facility? Related to this the manual seems to show a number of options which I cannot identify in alsamixer. For example the manual shows two surroundsound volume stages between which there are switched links, one to and two from the main stero path. Alsamixer shows a single amplifier. It has one link 'from' the main path but I could not find a link 'to' the main path. 3.3 Is the driver assuming a old layout which fits most of the new audio chip layout but not all? (This is really a question about what sort of warnings/advice should my write up include). 3.4 At which stage does alsa relate the registers to the descriptions, aux-volume, aux-mute etc? Again does it assume a layout which fits most aspects of the chips listed? Thats enough for the moment. I presume I could sit down for a year and read and understand the code - but in the meantime any help will be welcome. Regards, David Webb.

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[alsa-devel] eliminating sequencer and midi from my alsa module dependencies
by John L. Utz III 20 Jul '07

20 Jul '07

Hello; What arguments should give to the current version of alsa to eliminate the aforementioned things? here is my current conf string from hgcompile, am i missing something? args="--with-debug=full --with-isapnp=no --with-sequencer=no --with-oss=no --with-cards=via82xx" tnx! johnu

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[alsa-devel] Problem and workaround for Intel HDA
by Yann Le Brech 19 Jul '07

19 Jul '07

The problem : missing controls of the sound cards for an Intel HDA, Toshiba Satellite Pro U200 Full description : I was using the kernel 2.6.19 for a while, and I got some interesting functions like controlling the sound level on headphones or speakers separately. Then I upgraded to the kernel 2.6.21.1 and suddently the level controls changed, to useless level controls, like Master and PCM only (and some other useless). I wanted to get back to the previous control set, and I've found a solution by checking what was done in debugging mode. The previous kernel was not detecting the model config, but the new one did. So I simply strapped the detection routine by a return -1 and it did the trick. I know its not the best thing to do, but it was the simplest thing I thought about. Outputs : 2.6.19's Alsa output : ALSA sound/pci/hda/hda_intel.c:695: codec_mask = 0x3 2.6.21.1 : ALSA sound/pci/hda/hda_intel.c:695: codec_mask = 0x3 ALSA sound/pci/hda/hda_codec.c:1757: hda_codec: model 'toshiba' is selected for config 1179:1 (Toshiba U205) How to see the problem : Just see the list of controls in any mixer software. Environment : Toshiba Satellite Pro U200 Software : KateOS 3.2 with Linux 2.6.19 (working) and 2.6.21.1 (not working) Workaround : int __devinit snd_hda_check_board_config(struct hda_codec *codec, int num_configs, const char **models, const struct snd_pci_quirk *tbl) { return -1; // WORKAROUND if (codec->bus->modelname && models) { int i; for (i = 0; i < num_configs; i++) { if (models[i] && -- a++ Y.

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[alsa-devel] [PATCH] Add quirk for Asus P5LD2
by Claudio Matsuoka 19 Jul '07

19 Jul '07

Another simple patch that adds a system to the ALC882 quirk table. Tested with a week-old HG version, buildsystem of the current alsa-driver seems to be broken.

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[alsa-devel] a new synthesizer driver
by Steve Longerbeam 19 Jul '07

19 Jul '07

Hi all, I'm embarking on a big task: writing a new synthesizer driver. My device has all the ingredients for a MIDI wavetable sythesizer: 64 voices, each voice with envelope, pitch, and LFO controls. Also global effects like reverb. It's similar to the emu10k1. So my first question would be, is the synth code under synth/emux/ general enough to use for a different synthesizer chip? Ie, can I implement my own operators in 'struct snd_emux_operators' and pass that to snd_emux_register() ? Would it be a simple matter to make synth/emux/ a generic synth layer, by moving all the source in synth/emux/ up one level to synth/, and then rename all functions and data structures starting with "snd_emux" to "snd_synth" ? For instance, snd_emux_operators becomes snd_synth_operators, snd_emux_control() becomes snd_synth_control(), etc. It seems this path is already partially completed, since synth/emux/soundfont.c is a generic soundfont loader. The thing I want to avoid doing is creating a synth/my_synth/ directory and duplicating lots of code from synth/emux/ (including just copying soundfont.c verbatim!). Steve

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[alsa-devel] Re-uploaded patch for bug #2708
by Nicola Fagnani 19 Jul '07

19 Jul '07

I re-uploaded the patch with the requested changes some time ago, and I was wondering if there is still something wrong with it. Thanks Nicola Fagnani

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[alsa-devel] underrun observed when aplay is used with mmap
by Pharaoh . 19 Jul '07

19 Jul '07

Hi , Finally I have got my driver working and now testing it using aplay. I am able to play a song without getting any underruns when I use read/write mode but when I use mmapped mode I get underrun warnings. After this if I use read/write mode again then I observe 1 warning of underrun. Any idea what could be wrong? I know why underruns occur etc. Does it have to do with mmapped mode? I dont think so... Here is the log. # aplay -Dplughw:0,0 --rate=44100 --format=s16 --channels=2 --mmap 44.wav Playing WAVE '44.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo underrun!!! (at least 0.384 ms long) underrun!!! (at least 0.118 ms long) underrun!!! (at least 0.124 ms long) underrun!!! (at least 0.117 ms long) underrun!!! (at least 0.121 ms long) underrun!!! (at least 0.128 ms long) underrun!!! (at least 0.116 ms long) underrun!!! (at least 0.121 ms long) underrun!!! (at least 0.121 ms long) underrun!!! (at least 0.125 ms long) ....... and it goes on till the song finishes. And after this I play without using mmap and log says.. # aplay -Dplughw:0,0 --rate=44100 --format=s16 --channels=2 44.wav Playing WAVE '44.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo underrun!!! (at least 516.424 ms long) and then the song plays without any underrun. -pharaoh.

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[alsa-devel] cs46xx thinkpad resume fix - #bug1946
by Pavel Kysilka 19 Jul '07

19 Jul '07

Hi, attached patch fix resume on thinkpad notebooks with cs46xx chip. Tested with Thinkpad A21m. Patch is opposite latest linus git 2.6.22. Bug #1946 may be closed. Signed-off-by: Pavel Kysilka <pavelk(a)bsys.cz> Pavel Kysilka ________ Information from NOD32 ________ This message was checked by NOD32 Antivirus System for Linux Mail Servers. http://www.eset.com

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[alsa-devel] PS3 - Surround sound on SPDIF set-up
by gt bradley 19 Jul '07

19 Jul '07

Hi, I've been beating my had against this for several weeks now. (started with fedora 6, tried adding in the plugings etc (version 1.0.12) -- also tried upgrading to the current 1.0.14 version, without success (couldn't produce surround-sound on teh spdif output). I have now switched to fedorea 7 and was able to have rythmbox play surround sound via spdif, but have since lost that ability. (but I'm convinced the hardware and low-level drivers support it). In order to limit the number of moving parts, I have reduced to just trying to get speaker-test to play on all 6 speakers. from pirut & I have alsa-lib 1.0.14-0.4.rc3.fc7.ppc(64) and alsa-utils 1.0.14-0.7.rc2.fc7.ppc The default install has no asound.conf file. (attempts to create a "better" one didn't improve the situation). I do not see a file matching the ps3 device in etc/alsa/cards nor etc/alsa/pcm, but I have not found any documentation on adding entries to the above directories. [root@ps3 pcm0p]# cat /proc/asound/card0/pcm0p/info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: SPDIF name: SPDIF subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 [root@ps3 ~]# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PS3 [PS3], device 0: SPDIF [SPDIF] Subdevices: 1/1 Subdevice #0: subdevice #0 [root@ps3 ~]# aplay -L null Discard all samples (playback) or generate zero samples (capture) [root@ps3 ~]# amixer [root@ps3 ~]# amixer scontrols [root@ps3 ~]# speaker-test -c 6 speaker-test 1.0.14rc2 Playback device is default Stream parameters are 48000Hz, S16_BE, 6 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 4096 to 8192 Period size range from 256 to 256 Using max buffer size 8192 Periods = 4 was set period_size = 256 was set buffer_size = 8192 0 - Front Left 4 - Center 1 - Front Right 3 - Rear Right 2 - Rear Left 5 - LFE Time per period = 17.821324 * above test only produces sound on channels 0 & 1. xmms produces no sound and emits the following: ** WARNING **: alsa_setup_mixer(): Failed to find mixer element: PCM Is it possible to get some help/pointers in the right direction? thanks GT* *

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Re: [alsa-devel] ALSA driver for RealTek ALC268 chip
by Jonathan Woithe 18 Jul '07

18 Jul '07

Hi Amitay Amitay Isaacs (A.Isaacs(a)adfa.edu.au) wrote: > I found your names from the ALSA RealTek driver available for download > on the realtek website. Unfortunately the driver package does not have > support for the ALC268 chip. > > I have an ACER laptop with ALC268 HDA sound and would like to run linux > on it. > > Is the ALC268 chip driver available yet? Even the alpha/test version? Firstly, please submit queries like this to the ALSA development list. While I'm not fussed, other developers may not respond to emails sent only to their private addresses. It also means you don't have to guess who would be the most appropriate person to ask. Consequently I have added alsa-devel to the CC of this email. It would probably not take a lot of effort to add support for the ALC268 chip given that the ALC260 and ALC262 are supported. Furthermore, it seems that some work has already gone into this - see for example https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3104 So I would say that ALC268 support is being worked on. Pshou can probably give more details or provide further assistance. Note that my involvement in the ALSA RealTek driver was limited to the extended ALC260 support I added to take full advantage of this chip in my laptop. I don't have access to machines with other Realtek chips in them and so I'm not in the best position to provide direct assistance for other chips. Regards jonathan

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