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

[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

1 0

[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

1 0

[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

4 3

[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>

3 4

[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

3 3

[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); }

3 3

[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

1 0

[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 ----------------------------------------------------------------------------- ----------------------------------------------------------------------------- ----------------------------- -------------------------------------------------------

1 0

[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

2 1

[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

2 3