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

[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 21 Jul '07

21 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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