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

[alsa-devel] Altec Lansing FX5051 mixer
by Jeremy Nickurak 04 Apr '08

04 Apr '08

I recently purchased an Altec Lansing USB Surround speaker set. (It's a set of 5.1 speakers, with a built-in USB sound card). I've got about a week before the return policy expires and I'd have to return it. The mixer doesn't work at all w/linux kernel 2.6.22, everything is locked at 100% volume, which is pretty unhelpful, as it's the only way to adjust the volume. (no analog controls). Also, the digital control panel locks up when the speakers power up, further preventing any access to the volume controls. The device works out-of-the-box without any drivers under XP, so I have to so-far assume that alsa's usb-audio is missing something. Any thoughts as to what could be wrong? I'm more than happy to prod the device and submit any information I can extract about it, or even potentially opening the device up for remote access if a reputable developer is interested in examining it. -- Jeremy Nickurak -= Email/Jabber: atrus(a)rifetech.com =-

3 4

[alsa-devel] ALSA driver for Native Instruments sound hardware
by Daniel Mack 23 Mar '08

23 Mar '08

Hi, some months after the last posting, we finally managed to finish a new version of the ALSA driver for Native Instruments' external USB sound cards such as "Rig Kontrol 2", "Kore Kontroller", "Audio Kontrol 1" and "Audio 8 DJ". Compared to the last release, the following things were changed: - the streaming logic was completely reworked so that multiple substreams are handled correctly - a Linux input device layer was added to support the knobs, buttons and pedals on "Rig Kontrol 2" and "AudioKontrol 1" - workqueues were entirely removed from the code - most data structures are now allocated within one big struct to avoid multiple, smaller allocations - many more cleanups according to Clemens' reply on my last posting[1] The driver was properly tested with all the mentioned hardware on both record and playback channels. The patchset should apply to the current mecurial alsa-kernel tree. Please let me know what you think. Best regards, Daniel [1] http://article.gmane.org/gmane.linux.alsa.devel/34324/ Signed-off-by: Daniel Mack <daniel(a)caiaq.org> Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de>

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Re: [alsa-devel] asihpi driver -> kernel
by Takashi Iwai 20 Dec '07

20 Dec '07

Hi Eliot, [it seems that you Cc'ed to the old alsa-devel ML so the last one seemed missing.] At Tue, 16 Oct 2007 14:47:15 +1300, Eliot Blennerhassett wrote: > > Hi Takashi, > > still trying to move forward on this... > > Looking for "what is the first step" change to make in your opinion? The first step would be to get closer to linux kernel coding style. Namely, - get rid of C++ style function names, variables and field names (no hungarian!) - remove API wrappers Eventually, we may get rid of unneeded typedefs, too. Then, we can begin with simplifying the driver structure. Currently, asihpi and hpi are almost indepedent implementations. This makes life harder. > Takashi Iwai wrote: > > At Wed, 05 Sep 2007 13:00:35 +1200, > > Eliot Blennerhassett wrote: > >> Hi Takashi, > >> > >> I'd like to talk more about what you think is needed to get asihpi driver > >> ready for kernel tree. > >> > >> I think for now we should put aside cosmetic coding style changes (eg. comment > >> style, whitespace etc.) these are fixable with combination of automated tools > >> and some monkey work. > >> > >> So, lets start with the hard part. If we can't solve that there is no point in > >> cosmetics! > >> > >> A while ago you said "The specific problem I'm concerned is the "glue-code" > >> style in HPI driver. The OS-independent wrapper code is often disliked by > >> kernel maintainers. " > >> > >> Can you point to specific examples. Even better suggest how you think it > >> should be done better. Either way I need to consider how your concerns could > >> be addressed without breaking our codebase for other OSes etc. > >> > >> thanks and regards > > > > The problem is that the whole "wrapping" is superfluous when you > > concentrate on Linux driver. Merging into the linux kernel tree means > > that the driver would be dedicated to Linux. It's no longer your > > child, and it's no longer for every other OS. Thus, "common codebase" > > doesn't exist per se. > > > > For example, the whole definitions in hpios_linux_kernel.h should go > > away. The driver should use the native kernel API instead. > > Even this is not so hard, as it a mechanical process to replace wrapped > calls with linux native calls. OK, then it's fine. > > I know it's a very hard decision. Supporting other OSes is very > > important in most cases. OTOH, the slim-down is the best way to make > > the driver code maintainable for other people. Honestly, I find it > > also hard to maintain/debug the current ASIHPI code. It's too > > different from the normal codes, and understanding it alone takes too > > much energy before analyzing the problem. > > This is where I need more input, because to _me_ (of course) the code is > understandable. > > Is it the whole source or just parts that are hard. > E.g toplevel asihpi.c Ok/No? The code in asihpi.c is readable enough. But, other parts are different. The difficulty is that the driver structure isn't straight (apart from the coding style issues). We have an entry point for PCI driver _and_ indepdendent HPI initialization. This is already puzzling because HPI is also assigned to the corresponding PCI device. Also, the message union/struct is another stuff that is pretty hard to follow. Since we use no C++, union is often bad for readability. Could it be a bit simpler? > > The coding-style issues are similar problems. When C++-style codes > > appear in the tree of C-codes, they are strange. Simply so. It makes > > difficult to find the problematic part. "A buggy part looks buggy" -- > > it's the whole point of having a uniform coding style. > > Again, a mechanical process to change comments and style etc. Then it's fine with me. Let's try and see the progress. > > These problems (different from others) make it impossible to merge to > > the upstream tree, so far. > > > > > > > >> Eliot > >> BTW I'm happy to conduct this discussion publically on alsa-devel, some other > >> people might learn something too... > > > > I think it's a good thing, too. Feel free to add alsa-devel to Cc in > > the next reply with citation. > > > >> For reference here is the source tree diagram. > >> > >> asihpi.c alsa driver toplevel > >> | > >> hpifunc.c message/response pack/unpack > >> | > >> hpimsgx.c hpi stream ownership, backend selection, some init > >> | > >> hpi6205.c hpi6000.c hpi4000.c card family drivers > >> | > >> hpicmn.c, hpios_linux.c hpidspcd.c hpidebug.c library code > >> > >> hpimod.c probing, HPI ioctl > >> > > > > > > thanks, > > > > Takashi thanks, Takashi

2 13

Re: [alsa-devel] [Pkg-alsa-devel] Bug#438118: alsa-utils: aplay non-blocking mode isn't working
by Elimar Riesebieter 14 Dec '07

14 Dec '07

forwarded 438118 alsa-devel(a)alsa-project.org stop Hi all, can one please have a lokk at this? aplay -N isn't working for the OP. On Thu, 16 Aug 2007 the mental interface of Anders Boström told: > >>>>> "ER" == Elimar Riesebieter <riesebie(a)lxtec.de> writes: > > >> strace aplay -N : > >> > >> ... > >> open("/usr/lib/alsa-lib/libasound_module_rate_speexrate.so", O_RDONLY) = -1 ENOENT (No such file or directory) > > ER> This file is provided by > ER> libasound2-plugins: /usr/lib/alsa-lib/libasound_module_rate_speexrate.so > > ER> Please install libasound2-plugins and try again. > > Thanks for the suggestion. > > Unfortunately, it didn't solve my problem, no sound when non-blocking > mode is used. New strace: > > strace aplay -N : > ... > open("/usr/lib/alsa-lib/libasound_module_rate_speexrate.so", O_RDONLY) = 5 > read(5, "\177ELF\2\1\1\0\0\0\0\0\0\0\0\0\3\0>\0\1\0\0\0\20\22\0"..., 832) = 832 > fstat(5, {st_mode=S_IFREG|0644, st_size=20848, ...}) = 0 > mmap(NULL, 2116136, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 5, 0) = 0x2b9f792fe000 > mprotect(0x2b9f79302000, 2097152, PROT_NONE) = 0 > mmap(0x2b9f79502000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 5, 0x4000) = 0x2b9f79502000 > close(5) = 0 > ioctl(4, 0xc2604110, 0x7fff3269dec0) = 0 > ioctl(4, 0xc2604110, 0x7fff3269dec0) = 0 > ioctl(4, 0xc2604110, 0x7fff3269de00) = 0 > ioctl(4, 0xc2604110, 0x7fff3269de00) = 0 > ioctl(4, 0xc2604110, 0x7fff3269ddd0) = 0 > ioctl(4, 0xc2604110, 0x7fff3269ddc0) = 0 > ioctl(4, 0xc2604111, 0x7fff3269ddc0) = 0 > ioctl(4, 0xc0884113, 0x7fff3269dd10) = 0 > ioctl(4, 0x80184132, 0x7fff3269dc70) = 0 > mmap(NULL, 49152, PROT_READ|PROT_WRITE, MAP_SHARED, 4, 0) = 0x2b9f79503000 > ioctl(4, 0xc0884113, 0x624e88) = 0 > ioctl(4, 0x4140, 0) = 0 > ioctl(4, 0xc0884113, 0x624e88) = 0 > read(3, "\377eO\333\337\370\344}\335_\377\330^l\356\361`j\336_l"..., 1001) = 1001 > read(3, "n{u^\342zl~e~_[\365_y\362_\363\350\337n[\347nmi`\367y\344"..., 604) = 604 > ioctl(4, 0x4144, 0x2b9f785d8000) = -1 EAGAIN (Resource temporarily unavailable) > close(3) = 0 > ioctl(4, 0x4143, 0x3) = 0 > munmap(0x2b9f79503000, 49152) = 0 > ioctl(4, 0x4112, 0) = 0 > close(4) = 0 > munmap(0x2b9f785d7000, 4096) = 0 > munmap(0x2b9f785d8000, 4096) = 0 > munmap(0x2b9f792fe000, 2116136) = 0 > exit_group(0) = ? I have no idea, so forwarded to alsa-dev then. THX in advance Elimar -- Experience is something you don't get until just after you need it!

5 8

[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;

4 23

[alsa-devel] Wyse Winterm CS5535 Driver
by Darryl Bond 05 Nov '07

05 Nov '07

I have a Wyse S50 terminal that I am booting diskless. I have been trying to get sound working. The box has a cs5535 chipset. After much fiddling in the kernel source (2.6.22.5 <http://2.6.22.5> ) I have been able to make it work. I'm not sure if it is a cs5535 problem as the code that fixes it is in ac97_pcm.c. It would appear that the card is getting passed an unexpected rate. The function sound/pci/ac97/ac97_pcm.c snd_ac97_set_rate() has a section: case AC97_PCM_FRONT_DAC_RATE: case AC97_PCM_LR_ADC_RATE: if ((ac97->regs[AC97_EXTENDED_STATUS] & AC97_EA_VRA) == 0) /* VRA */ if (rate != 48000 && rate != 96000) return -EINVAL; break; If I use 'aplay wavfile' the rate passed to this function is 44100. I I changethe code to: if (rate != 48000 && rate != 96000 && rate != 44100) it plays but at a slightly incorrect rate. If I use the Citrix ICA client (which is what I really want to work) the rate passed in is 8000. If I add 8000 to the test, it works, but is far too fast. Any hints as to what I should try? Regards Darryl Bond __________________________________________________________________________________ DISCLAIMER The contents of this electronic message and any attachments are intended only for the addressee and may contain legally privileged, personal, sensitive or confidential information. If you are not the intended addressee, and have received this email, any transmission, distribution, downloading, printing or photocopying of the contents of this message or attachments is strictly prohibited. Any legal privilege or confidentiality attached to this message and attachments is not waived, lost or destroyed by reason of delivery to any person other than intended addressee. If you have received this message and are not the intended addressee you should notify the sender by return email and destroy all copies of the message and any attachments. Unless expressly attributed, the views expressed in this email do not necessarily represent the views of the company.

2 4

[alsa-devel] [PATCH] cmipci at 96kHz
by Timofei V. Bondarenko 31 Oct '07

31 Oct '07

Hi. This patch adds support for 88.2k, 96k, and 128k samplerates on cmi8738-55 chip. Analog playback works fine on all channels. Analog capture works well too, though the extra samples seems interpolated by hardware. spdif playback and capture works fine. Regards, Tim.

3 6

[alsa-devel] via82xx DC offset at -30dB
by plessas＠mur.at 31 Oct '07

31 Oct '07

Dear List, having a Compaq Presario notebook in front of me, running Debian testing and a 2.6.18-4-686 kernel, i have a strange DC offset when trying to record audio via the mic/line input. As the audacity level meters suggest, it resides at -30dBFS. Audio comes through and gets recorded though, but the offset severely limits the headroom of course. The sensitivity of the line/mic input seems to be quite high, although i have the mic boost setting turned off of course. Setting the module option ac97_quirk=yes did not help either. Can it be that a soundcard of this strange behavior is manufactured, or is it an alsa/driver issue? I really don't want to install another M$ operating system to check if it works there, so any help is deeply appreciated. lspci lists the device as: 00:07.5 Multimedia audio controller: VIA Technologies, Inc. VT82C686 AC97 Audio Controller (rev 20) regards, PP PS: please CC: me, as i am not subscribed to alsa-devel (yet)

3 2

[alsa-devel] Testers Wanted: OPL3 hwdep loader interface
by Takashi Iwai 31 Oct '07

31 Oct '07

Hi, one of big pending patches on my local tree is the removal of seq-instr interface. The only real user of instrument layer is OPL3 patch loader, and I'd like to move this to hwdep interface. The patch below is to move the OPL3 patch loading to hwdep device. Two more patches will follow in other mails for alsa-lib addition and sbiload change. Can someone test whether these patches work? Unfortunately, I have no hardware with OPL3 at this moment around here. Thanks, Takashi diff -r 88d273f360dd drivers/opl3/opl3_lib.c --- a/drivers/opl3/opl3_lib.c Fri Oct 26 15:48:22 2007 +0200 +++ b/drivers/opl3/opl3_lib.c Fri Oct 26 16:09:56 2007 +0200 @@ -327,6 +327,7 @@ static int snd_opl3_free(struct snd_opl3 snd_assert(opl3 != NULL, return -ENXIO); if (opl3->private_free) opl3->private_free(opl3); + snd_opl3_clear_patches(opl3); release_and_free_resource(opl3->res_l_port); release_and_free_resource(opl3->res_r_port); kfree(opl3); @@ -521,6 +522,7 @@ int snd_opl3_hwdep_new(struct snd_opl3 * /* operators - only ioctl */ hw->ops.open = snd_opl3_open; hw->ops.ioctl = snd_opl3_ioctl; + hw->ops.write = snd_opl3_write; hw->ops.release = snd_opl3_release; opl3->seq_dev_num = seq_device; diff -r 88d273f360dd drivers/opl3/opl3_midi.c --- a/drivers/opl3/opl3_midi.c Fri Oct 26 15:48:22 2007 +0200 +++ b/drivers/opl3/opl3_midi.c Fri Oct 26 16:09:56 2007 +0200 @@ -289,8 +289,6 @@ void snd_opl3_note_on(void *p, int note, void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) { struct snd_opl3 *opl3; - struct snd_seq_instr wanted; - struct snd_seq_kinstr *kinstr; int instr_4op; int voice; @@ -306,11 +304,13 @@ void snd_opl3_note_on(void *p, int note, unsigned char voice_offset; unsigned short opl3_reg; unsigned char reg_val; + unsigned char prg, bank; int key = note; unsigned char fnum, blocknum; int i; + struct fm_patch *patch; struct fm_instrument *fm; unsigned long flags; @@ -320,19 +320,17 @@ void snd_opl3_note_on(void *p, int note, snd_printk("Note on, ch %i, inst %i, note %i, vel %i\n", chan->number, chan->midi_program, note, vel); #endif - wanted.cluster = 0; - wanted.std = SNDRV_SEQ_INSTR_TYPE2_OPL2_3; /* in SYNTH mode, application takes care of voices */ /* in SEQ mode, drum voice numbers are notes on drum channel */ if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) { if (chan->drum_channel) { /* percussion instruments are located in bank 128 */ - wanted.bank = 128; - wanted.prg = note; + bank = 128; + prg = note; } else { - wanted.bank = chan->gm_bank_select; - wanted.prg = chan->midi_program; + bank = chan->gm_bank_select; + prg = chan->midi_program; } } else { /* Prepare for OSS mode */ @@ -340,8 +338,8 @@ void snd_opl3_note_on(void *p, int note, return; /* OSS instruments are located in bank 127 */ - wanted.bank = 127; - wanted.prg = chan->midi_program; + bank = 127; + prg = chan->midi_program; } spin_lock_irqsave(&opl3->voice_lock, flags); @@ -353,15 +351,14 @@ void snd_opl3_note_on(void *p, int note, } __extra_prg: - kinstr = snd_seq_instr_find(opl3->ilist, &wanted, 1, 0); - if (kinstr == NULL) { + patch = snd_opl3_find_patch(opl3, prg, bank, 0); + if (!patch) { spin_unlock_irqrestore(&opl3->voice_lock, flags); return; } - fm = KINSTR_DATA(kinstr); - - switch (fm->type) { + fm = &patch->inst; + switch (patch->type) { case FM_PATCH_OPL2: instr_4op = 0; break; @@ -371,14 +368,12 @@ void snd_opl3_note_on(void *p, int note, break; } default: - snd_seq_instr_free_use(opl3->ilist, kinstr); spin_unlock_irqrestore(&opl3->voice_lock, flags); return; } - #ifdef DEBUG_MIDI snd_printk(" --> OPL%i instrument: %s\n", - instr_4op ? 3 : 2, kinstr->name); + instr_4op ? 3 : 2, patch->name); #endif /* in SYNTH mode, application takes care of voices */ /* in SEQ mode, allocate voice on free OPL3 channel */ @@ -569,8 +564,6 @@ void snd_opl3_note_on(void *p, int note, /* get extra pgm, but avoid possible loops */ extra_prg = (extra_prg) ? 0 : fm->modes; - snd_seq_instr_free_use(opl3->ilist, kinstr); - /* do the bookkeeping */ vp->time = opl3->use_time++; vp->note = key; @@ -601,12 +594,12 @@ void snd_opl3_note_on(void *p, int note, /* allocate extra program if specified in patch library */ if (extra_prg) { if (extra_prg > 128) { - wanted.bank = 128; + bank = 128; /* percussions start at 35 */ - wanted.prg = extra_prg - 128 + 35 - 1; + prg = extra_prg - 128 + 35 - 1; } else { - wanted.bank = 0; - wanted.prg = extra_prg - 1; + bank = 0; + prg = extra_prg - 1; } #ifdef DEBUG_MIDI snd_printk(" *** allocating extra program\n"); diff -r 88d273f360dd drivers/opl3/opl3_oss.c --- a/drivers/opl3/opl3_oss.c Fri Oct 26 15:48:22 2007 +0200 +++ b/drivers/opl3/opl3_oss.c Fri Oct 26 16:09:56 2007 +0200 @@ -195,17 +195,6 @@ static int snd_opl3_close_seq_oss(struct /* load patch */ -/* offsets for SBI params */ -#define AM_VIB 0 -#define KSL_LEVEL 2 -#define ATTACK_DECAY 4 -#define SUSTAIN_RELEASE 6 -#define WAVE_SELECT 8 - -/* offset for SBI instrument */ -#define CONNECTION 10 -#define OFFSET_4OP 11 - /* from sound_config.h */ #define SBFM_MAXINSTR 256 @@ -213,112 +202,42 @@ static int snd_opl3_load_patch_seq_oss(s const char __user *buf, int offs, int count) { struct snd_opl3 *opl3; - int err = -EINVAL; + struct sbi_instrument sbi; + char name[32]; + int err, type; snd_assert(arg != NULL, return -ENXIO); opl3 = arg->private_data; - if ((format == FM_PATCH) || (format == OPL3_PATCH)) { - struct sbi_instrument sbi; + if (format == FM_PATCH) + type = FM_PATCH_OPL2; + else if (format == OPL3_PATCH) + type = FM_PATCH_OPL3; + else + return -EINVAL; - size_t size; - struct snd_seq_instr_header *put; - struct snd_seq_instr_data *data; - struct fm_xinstrument *xinstr; + if (count < (int)sizeof(sbi)) { + snd_printk("FM Error: Patch record too short\n"); + return -EINVAL; + } + if (copy_from_user(&sbi, buf, sizeof(sbi))) + return -EFAULT; - struct snd_seq_event ev; - int i; + if (sbi.channel < 0 || sbi.channel >= SBFM_MAXINSTR) { + snd_printk("FM Error: Invalid instrument number %d\n", + sbi.channel); + return -EINVAL; + } - mm_segment_t fs; + memset(name, 0, sizeof(name)); + sprintf(name, "Chan%d", sbi.channel); - if (count < (int)sizeof(sbi)) { - snd_printk("FM Error: Patch record too short\n"); - return -EINVAL; - } - if (copy_from_user(&sbi, buf, sizeof(sbi))) - return -EFAULT; + err = snd_opl3_load_patch(opl3, sbi.channel, 127, type, name, NULL, + sbi.operators); + if (err < 0) + return err; - if (sbi.channel < 0 || sbi.channel >= SBFM_MAXINSTR) { - snd_printk("FM Error: Invalid instrument number %d\n", sbi.channel); - return -EINVAL; - } - - size = sizeof(*put) + sizeof(struct fm_xinstrument); - put = kzalloc(size, GFP_KERNEL); - if (put == NULL) - return -ENOMEM; - /* build header */ - data = &put->data; - data->type = SNDRV_SEQ_INSTR_ATYPE_DATA; - strcpy(data->data.format, SNDRV_SEQ_INSTR_ID_OPL2_3); - /* build data section */ - xinstr = (struct fm_xinstrument *)(data + 1); - xinstr->stype = FM_STRU_INSTR; - - for (i = 0; i < 2; i++) { - xinstr->op[i].am_vib = sbi.operators[AM_VIB + i]; - xinstr->op[i].ksl_level = sbi.operators[KSL_LEVEL + i]; - xinstr->op[i].attack_decay = sbi.operators[ATTACK_DECAY + i]; - xinstr->op[i].sustain_release = sbi.operators[SUSTAIN_RELEASE + i]; - xinstr->op[i].wave_select = sbi.operators[WAVE_SELECT + i]; - } - xinstr->feedback_connection[0] = sbi.operators[CONNECTION]; - - if (format == OPL3_PATCH) { - xinstr->type = FM_PATCH_OPL3; - for (i = 0; i < 2; i++) { - xinstr->op[i+2].am_vib = sbi.operators[OFFSET_4OP + AM_VIB + i]; - xinstr->op[i+2].ksl_level = sbi.operators[OFFSET_4OP + KSL_LEVEL + i]; - xinstr->op[i+2].attack_decay = sbi.operators[OFFSET_4OP + ATTACK_DECAY + i]; - xinstr->op[i+2].sustain_release = sbi.operators[OFFSET_4OP + SUSTAIN_RELEASE + i]; - xinstr->op[i+2].wave_select = sbi.operators[OFFSET_4OP + WAVE_SELECT + i]; - } - xinstr->feedback_connection[1] = sbi.operators[OFFSET_4OP + CONNECTION]; - } else { - xinstr->type = FM_PATCH_OPL2; - } - - put->id.instr.std = SNDRV_SEQ_INSTR_TYPE2_OPL2_3; - put->id.instr.bank = 127; - put->id.instr.prg = sbi.channel; - put->cmd = SNDRV_SEQ_INSTR_PUT_CMD_CREATE; - - memset (&ev, 0, sizeof(ev)); - ev.source.client = SNDRV_SEQ_CLIENT_OSS; - ev.dest = arg->addr; - - ev.flags = SNDRV_SEQ_EVENT_LENGTH_VARUSR; - ev.queue = SNDRV_SEQ_QUEUE_DIRECT; - - fs = snd_enter_user(); - __again: - ev.type = SNDRV_SEQ_EVENT_INSTR_PUT; - ev.data.ext.len = size; - ev.data.ext.ptr = put; - - err = snd_seq_instr_event(&opl3->fm_ops, opl3->ilist, &ev, - opl3->seq_client, 0, 0); - if (err == -EBUSY) { - struct snd_seq_instr_header remove; - - memset (&remove, 0, sizeof(remove)); - remove.cmd = SNDRV_SEQ_INSTR_FREE_CMD_SINGLE; - remove.id.instr = put->id.instr; - - /* remove instrument */ - ev.type = SNDRV_SEQ_EVENT_INSTR_FREE; - ev.data.ext.len = sizeof(remove); - ev.data.ext.ptr = &remove; - - snd_seq_instr_event(&opl3->fm_ops, opl3->ilist, &ev, - opl3->seq_client, 0, 0); - goto __again; - } - snd_leave_user(fs); - - kfree(put); - } - return err; + return sizeof(sbi); } /* ioctl */ diff -r 88d273f360dd drivers/opl3/opl3_seq.c --- a/drivers/opl3/opl3_seq.c Fri Oct 26 15:48:22 2007 +0200 +++ b/drivers/opl3/opl3_seq.c Fri Oct 26 16:09:56 2007 +0200 @@ -152,15 +152,7 @@ static int snd_opl3_synth_event_input(st { struct snd_opl3 *opl3 = private_data; - if (ev->type >= SNDRV_SEQ_EVENT_INSTR_BEGIN && - ev->type <= SNDRV_SEQ_EVENT_INSTR_CHANGE) { - if (direct) { - snd_seq_instr_event(&opl3->fm_ops, opl3->ilist, ev, - opl3->seq_client, atomic, hop); - } - } else { - snd_midi_process_event(&opl3_ops, ev, opl3->chset); - } + snd_midi_process_event(&opl3_ops, ev, opl3->chset); return 0; } @@ -249,16 +241,6 @@ static int snd_opl3_seq_new_device(struc return err; } - /* initialize instrument list */ - opl3->ilist = snd_seq_instr_list_new(); - if (opl3->ilist == NULL) { - snd_seq_delete_kernel_client(client); - opl3->seq_client = -1; - return -ENOMEM; - } - opl3->ilist->flags = SNDRV_SEQ_INSTR_FLG_DIRECT; - snd_seq_fm_init(&opl3->fm_ops, NULL); - /* setup system timer */ init_timer(&opl3->tlist); opl3->tlist.function = snd_opl3_timer_func; @@ -287,8 +269,6 @@ static int snd_opl3_seq_delete_device(st snd_seq_delete_kernel_client(opl3->seq_client); opl3->seq_client = -1; } - if (opl3->ilist) - snd_seq_instr_list_free(&opl3->ilist); return 0; } diff -r 88d273f360dd drivers/opl3/opl3_synth.c --- a/drivers/opl3/opl3_synth.c Fri Oct 26 15:48:22 2007 +0200 +++ b/drivers/opl3/opl3_synth.c Fri Oct 26 16:09:56 2007 +0200 @@ -186,6 +186,170 @@ int snd_opl3_release(struct snd_hwdep * mutex_unlock(&opl3->access_mutex); return 0; +} + +/* + * write the device - load patches + */ +long snd_opl3_write(struct snd_hwdep *hw, const char __user *buf, long count, + loff_t *offset) +{ + struct snd_opl3 *opl3 = hw->private_data; + long result = 0; + int err = 0; + struct sbi_patch inst; + + while (count >= sizeof(inst)) { + unsigned char type; + if (copy_from_user(&inst, buf, sizeof(inst))) + return -EFAULT; + if (!memcmp(inst.key, FM_KEY_SBI, 4) || + !memcmp(inst.key, FM_KEY_2OP, 4)) + type = FM_PATCH_OPL2; + else if (!memcmp(inst.key, FM_KEY_4OP, 4)) + type = FM_PATCH_OPL3; + else /* invalid type */ + break; + err = snd_opl3_load_patch(opl3, inst.prog, inst.bank, type, + inst.name, inst.extension, + inst.data); + if (err < 0) + break; + result += sizeof(inst); + count -= sizeof(inst); + } + return result > 0 ? result : err; +} + + +/* + * Patch management + */ + +/* offsets for SBI params */ +#define AM_VIB 0 +#define KSL_LEVEL 2 +#define ATTACK_DECAY 4 +#define SUSTAIN_RELEASE 6 +#define WAVE_SELECT 8 + +/* offset for SBI instrument */ +#define CONNECTION 10 +#define OFFSET_4OP 11 + +/* + * load a patch, obviously. + * + * loaded on the given program and bank numbers with the given type + * (FM_PATCH_OPLx). + * data is the pointer of SBI record _without_ header (key and name). + * name is the name string of the patch. + * ext is the extension data of 7 bytes long (stored in name of SBI + * data up to offset 25), or NULL to skip. + * return 0 if successful or a negative error code. + */ +int snd_opl3_load_patch(struct snd_opl3 *opl3, + int prog, int bank, int type, + const char *name, + const unsigned char *ext, + const unsigned char *data) +{ + struct fm_patch *patch; + int i; + + patch = snd_opl3_find_patch(opl3, prog, bank, 1); + if (!patch) + return -ENOMEM; + + patch->type = type; + + for (i = 0; i < 2; i++) { + patch->inst.op[i].am_vib = data[AM_VIB + i]; + patch->inst.op[i].ksl_level = data[KSL_LEVEL + i]; + patch->inst.op[i].attack_decay = data[ATTACK_DECAY + i]; + patch->inst.op[i].sustain_release = data[SUSTAIN_RELEASE + i]; + patch->inst.op[i].wave_select = data[WAVE_SELECT + i]; + } + patch->inst.feedback_connection[0] = data[CONNECTION]; + + if (type == FM_PATCH_OPL3) { + for (i = 0; i < 2; i++) { + patch->inst.op[i+2].am_vib = + data[OFFSET_4OP + AM_VIB + i]; + patch->inst.op[i+2].ksl_level = + data[OFFSET_4OP + KSL_LEVEL + i]; + patch->inst.op[i+2].attack_decay = + data[OFFSET_4OP + ATTACK_DECAY + i]; + patch->inst.op[i+2].sustain_release = + data[OFFSET_4OP + SUSTAIN_RELEASE + i]; + patch->inst.op[i+2].wave_select = + data[OFFSET_4OP + WAVE_SELECT + i]; + } + patch->inst.feedback_connection[1] = + data[OFFSET_4OP + CONNECTION]; + } + + if (ext) { + patch->inst.echo_delay = ext[0]; + patch->inst.echo_atten = ext[1]; + patch->inst.chorus_spread = ext[2]; + patch->inst.trnsps = ext[3]; + patch->inst.fix_dur = ext[4]; + patch->inst.modes = ext[5]; + patch->inst.fix_key = ext[6]; + } + + if (name) + strlcpy(patch->name, name, sizeof(patch->name)); + + return 0; +} +EXPORT_SYMBOL(snd_opl3_load_patch); + +/* + * find a patch with the given program and bank numbers, returns its pointer + * if no matching patch is found and create_patch is set, it creates a + * new patch object. + */ +struct fm_patch *snd_opl3_find_patch(struct snd_opl3 *opl3, int prog, int bank, + int create_patch) +{ + /* pretty dumb hash key */ + unsigned int key = (prog + bank) % OPL3_PATCH_HASH_SIZE; + struct fm_patch *patch; + + for (patch = opl3->patch_table[key]; patch; patch = patch->next) { + if (patch->prog == prog && patch->bank == bank) + return patch; + } + if (!create_patch) + return NULL; + + patch = kzalloc(sizeof(*patch), GFP_KERNEL); + if (!patch) + return NULL; + patch->prog = prog; + patch->bank = bank; + patch->next = opl3->patch_table[key]; + opl3->patch_table[key] = patch; + return patch; +} +EXPORT_SYMBOL(snd_opl3_find_patch); + +/* + * Clear all patches of the given OPL3 instance + */ +void snd_opl3_clear_patches(struct snd_opl3 *opl3) +{ + int i; + for (i = 0; i < OPL3_PATCH_HASH_SIZE; i++) { + struct fm_patch *patch, *next; + for (patch = opl3->patch_table[i]; patch; patch = next) { + next = patch->next; + kfree(patch); + } + } + memset(opl3->patch_table, 0, sizeof(opl3->patch_table)); } /* ------------------------------ */ diff -r 88d273f360dd include/asound_fm.h --- a/include/asound_fm.h Fri Oct 26 15:48:22 2007 +0200 +++ b/include/asound_fm.h Fri Oct 26 16:09:56 2007 +0200 @@ -112,4 +112,21 @@ struct snd_dm_fm_params { #define SNDRV_DM_FM_OSS_IOCTL_SET_MODE 0x24 #define SNDRV_DM_FM_OSS_IOCTL_SET_OPL 0x25 +/* + * Patch Record - fixed size for write + */ + +#define FM_KEY_SBI "SBI\032" +#define FM_KEY_2OP "2OP\032" +#define FM_KEY_4OP "4OP\032" + +struct sbi_patch { + unsigned char prog; + unsigned char bank; + char key[4]; + char name[25]; + char extension[7]; + unsigned char data[32]; +}; + #endif /* __SOUND_ASOUND_FM_H */ diff -r 88d273f360dd include/opl3.h --- a/include/opl3.h Fri Oct 26 15:48:22 2007 +0200 +++ b/include/opl3.h Fri Oct 26 16:09:56 2007 +0200 @@ -63,7 +63,7 @@ #include "seq_oss_legacy.h" #endif #include "seq_device.h" -#include "ainstr_fm.h" +#include "asound_fm.h" /* * Register numbers for the global registers @@ -240,6 +240,47 @@ struct snd_opl3; struct snd_opl3; /* + * Instrument record, aka "Patch" + */ + +/* FM operator */ +struct fm_operator { + unsigned char am_vib; + unsigned char ksl_level; + unsigned char attack_decay; + unsigned char sustain_release; + unsigned char wave_select; +} __attribute__((packed)); + +/* Instrument data */ +struct fm_instrument { + struct fm_operator op[4]; + unsigned char feedback_connection[2]; + unsigned char echo_delay; + unsigned char echo_atten; + unsigned char chorus_spread; + unsigned char trnsps; + unsigned char fix_dur; + unsigned char modes; + unsigned char fix_key; +}; + +/* type */ +#define FM_PATCH_OPL2 0x01 /* OPL2 2 operators FM instrument */ +#define FM_PATCH_OPL3 0x02 /* OPL3 4 operators FM instrument */ + +/* Instrument record */ +struct fm_patch { + unsigned char prog; + unsigned char bank; + unsigned char type; + struct fm_instrument inst; + char name[24]; + struct fm_patch *next; +}; + + +/* * A structure to keep track of each hardware voice */ struct snd_opl3_voice { @@ -297,8 +338,8 @@ struct snd_opl3 { struct snd_midi_channel_set * oss_chset; #endif - struct snd_seq_kinstr_ops fm_ops; - struct snd_seq_kinstr_list *ilist; +#define OPL3_PATCH_HASH_SIZE 32 + struct fm_patch *patch_table[OPL3_PATCH_HASH_SIZE]; struct snd_opl3_voice voices[MAX_OPL3_VOICES]; /* Voices (OPL3 'channel') */ int use_time; /* allocation counter */ @@ -333,8 +374,19 @@ int snd_opl3_open(struct snd_hwdep * hw, int snd_opl3_open(struct snd_hwdep * hw, struct file *file); int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file, unsigned int cmd, unsigned long arg); +long snd_opl3_write(struct snd_hwdep *hw, const char __user *buf, long count, + loff_t *offset); int snd_opl3_release(struct snd_hwdep * hw, struct file *file); void snd_opl3_reset(struct snd_opl3 * opl3); +int snd_opl3_load_patch(struct snd_opl3 *opl3, + int prog, int bank, int type, + const char *name, + const unsigned char *ext, + const unsigned char *data); +struct fm_patch *snd_opl3_find_patch(struct snd_opl3 *opl3, int prog, int bank, + int create_patch); +void snd_opl3_clear_patches(struct snd_opl3 *opl3); + #endif /* __SOUND_OPL3_H */

3 11

[alsa-devel] Sample format coversion bug?
by Alexander E. Patrakov 31 Oct '07

31 Oct '07

Hello, I am writing an application that uses ALSA, and I don't want to write code to support all possible sample formats. On IRC, I asked whether it is OK to support only S24 format internally and rely on the "plug" to do the required conversion. I got a positive answer, but, when testing my application against my SAA7134-based tuner as a capture device, found that it doesn't work. The problem is also reproducible with "arecord". Given that plughw:1 refers to a SAA7134-based tuner (which obviously doesn't support S24_LE natively), running the following command arecord -vvv -f S24_LE -c 2 -r 32000 -D plughw:1 /dev/null produces the following output: ALSA lib pcm_mmap.c:369:(snd_pcm_mmap) mmap failed: Invalid argument arecord: set_params:961: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S24_LE SUBFORMAT: STD SAMPLE_BITS: 32 FRAME_BITS: 64 CHANNELS: 2 RATE: 32000 PERIOD_TIME: 125000 PERIOD_SIZE: 4000 PERIOD_BYTES: 32000 PERIODS: 4 BUFFER_TIME: 500000 BUFFER_SIZE: 16000 BUFFER_BYTES: 128000 TICK_TIME: 4000 This is on Debian, the kernel is 2.6.22-2-amd64, libasound2 is 1.0.14a-2. Same for S32_LE. -- Alexander E. Patrakov

3 8

[alsa-devel] [PATCH] Free after memory allocation failure in emu10k1
by Roel Kluin 31 Oct '07

31 Oct '07

I can confirm that it builds. -- Free after memory allocation failure Signed-off-by: Roel Kluin <12o3l(a)tiscali.nl> --- diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 97c41d7..88f8fb3 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1894,25 +1894,31 @@ static unsigned char saved_regs_audigy[] = { static int __devinit alloc_pm_buffer(struct snd_emu10k1 *emu) { int size; size = ARRAY_SIZE(saved_regs); if (emu->audigy) size += ARRAY_SIZE(saved_regs_audigy); emu->saved_ptr = vmalloc(4 * NUM_G * size); if (! emu->saved_ptr) return -ENOMEM; - if (snd_emu10k1_efx_alloc_pm_buffer(emu) < 0) + + if (snd_emu10k1_efx_alloc_pm_buffer(emu) < 0) { + vfree(emu->saved_ptr); return -ENOMEM; + } + if (emu->card_capabilities->ca0151_chip && - snd_p16v_alloc_pm_buffer(emu) < 0) + snd_p16v_alloc_pm_buffer(emu) < 0) { + vfree(emu->saved_ptr); return -ENOMEM; + } return 0; } static void free_pm_buffer(struct snd_emu10k1 *emu) { vfree(emu->saved_ptr); snd_emu10k1_efx_free_pm_buffer(emu); if (emu->card_capabilities->ca0151_chip) snd_p16v_free_pm_buffer(emu); } diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 9bf1cd5..a31ad02 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -2634,30 +2634,43 @@ int __devinit snd_emu10k1_fx8010_new(struct snd_emu10k1 *emu, int device, struct #ifdef CONFIG_PM int __devinit snd_emu10k1_efx_alloc_pm_buffer(struct snd_emu10k1 *emu) { int len; len = emu->audigy ? 0x200 : 0x100; emu->saved_gpr = kmalloc(len * 4, GFP_KERNEL); if (! emu->saved_gpr) return -ENOMEM; + len = emu->audigy ? 0x100 : 0xa0; emu->tram_val_saved = kmalloc(len * 4, GFP_KERNEL); + if (! emu->tram_val_saved) + goto free_gpr; + emu->tram_addr_saved = kmalloc(len * 4, GFP_KERNEL); - if (! emu->tram_val_saved || ! emu->tram_addr_saved) - return -ENOMEM; + if (! emu->tram_addr_saved) + goto free_tram_val; + len = emu->audigy ? 2 * 1024 : 2 * 512; emu->saved_icode = vmalloc(len * 4); if (! emu->saved_icode) - return -ENOMEM; + goto free_tram_addr; + return 0; +free_gpr: + kfree(emu->saved_gpr); +free_tram_val: + kfree(emu->tram_val_saved); +free_tram_addr: + kfree(emu->tram_addr_saved); + return -ENOMEM; } void snd_emu10k1_efx_free_pm_buffer(struct snd_emu10k1 *emu) { kfree(emu->saved_gpr); kfree(emu->tram_val_saved); kfree(emu->tram_addr_saved); vfree(emu->saved_icode); }

2 1

[alsa-devel] how to handle bt848 audio driver
by Trent Piepho 30 Oct '07

30 Oct '07

I've made some more patches for the ALSA bt87x audio driver to support more features of the Osprey 440 card. In order to support certain features, the ALSA driver needs access to the bt878's GPIO lines and I2C bus. These are controlled by PCI function 0, which is driven by the V4L driver. The bt878 DVB driver (which drives PCI function 1 just like the ALSA driver) needs to do the same thing and there is already some code to handle it. So, the ALSA driver (snd-bt87x) needs to use functions from the V4L driver (bttv). As it is now, the two drivers are completely separate and don't share any code and can be loaded individually. The DVB driver in the same tree as bttv, so this isn't a problem. But the snd-bt87x is in another tree, so it's somewhat harder, and I'm not sure what to do about it. The driver will need some header files from the bttv directory. I guess this can be handled by adding a -I flag in the Makefile, though that seems a little ugly. The bigger problem is that changes to snd-bt87x will depend on changes to bttv. How can the ALSA build system handle this? For instance, my patch to add a digital volume control to snd-bt87x requires a new function to be exported from bttv. How will ALSA build snd-bt87x if the kernel source doesn't yet have this function?

3 3

[alsa-devel] External Sound Card
by John Dey 30 Oct '07

30 Oct '07

I am looking for a external sound card that has spdif (7.1) functioning. The driver Soundblaster nx usb seems like it is under active development. The alsa documentation page says that spdif is not working. Is the documentation page out of date or is someone working on that capability? If spdif capability is not in the near future, can someone suggest a sound device (spdif capable) that I could look at. Thanks. John

2 3

Re: [alsa-devel] sysfs: WARNING: at fs/sysfs/dir.c:424 sysfs_add_one() - with ALSA
by Kamalesh Babulal 30 Oct '07

30 Oct '07

On Fri, Oct 26, 2007 at 12:44:38PM +0530, Amit Shah wrote: > On 2.6.24-rc1, I get this for all my eth interfaces: > > [ 150.389821] sysfs: duplicate filename 'eth1_rename' can not be created > [ 150.389824] WARNING: at fs/sysfs/dir.c:424 sysfs_add_one() > [ 150.389826] > [ 150.389826] Call Trace: > [ 150.389840] [<ffffffff802ee9cf>] sysfs_add_one+0xaf/0xf0 > [ 150.389844] [<ffffffff802ef9fe>] sysfs_create_link+0x9e/0x150 > [ 150.389849] [<ffffffff8038a418>] device_rename+0x1d8/0x230 > [ 150.389854] [<ffffffff803c2626>] dev_change_name+0xb6/0x230 > [ 150.389857] [<ffffffff803c2d5b>] dev_ioctl+0x2bb/0x5a0 > [ 150.389862] [<ffffffff80437436>] _spin_unlock_irqrestore+0x16/0x40 > [ 150.389864] [<ffffffff8043947a>] do_page_fault+0x24a/0x930 > [ 150.389869] [<ffffffff803b341d>] sock_ioctl+0x7d/0x250 > [ 150.389872] [<ffffffff802aafef>] do_ioctl+0x2f/0xa0 > [ 150.389875] [<ffffffff802ab0d4>] vfs_ioctl+0x74/0x2d0 > [ 150.389878] [<ffffffff802ab3c1>] sys_ioctl+0x91/0xb0 > [ 150.389881] [<ffffffff8020c32e>] system_call+0x7e/0x83 > [ 150.389884] > [ 150.389886] net eth1_rename: device_rename: sysfs_create_symlink failed (-17) > [ 151.169923] sysfs: duplicate filename 'eth2' can not be created > [ 151.169927] WARNING: at fs/sysfs/dir.c:424 sysfs_add_one() > [ 151.169929] > [ 151.169930] Call Trace: > [ 151.169946] [<ffffffff802ee9cf>] sysfs_add_one+0xaf/0xf0 > [ 151.169949] [<ffffffff802ef9fe>] sysfs_create_link+0x9e/0x150 > [ 151.169955] [<ffffffff8038a418>] device_rename+0x1d8/0x230 > [ 151.169960] [<ffffffff803c2626>] dev_change_name+0xb6/0x230 > [ 151.169963] [<ffffffff803c2d5b>] dev_ioctl+0x2bb/0x5a0 > [ 151.169967] [<ffffffff80437436>] _spin_unlock_irqrestore+0x16/0x40 > [ 151.169970] [<ffffffff8043947a>] do_page_fault+0x24a/0x930 > [ 151.169974] [<ffffffff803b341d>] sock_ioctl+0x7d/0x250 > [ 151.169978] [<ffffffff802aafef>] do_ioctl+0x2f/0xa0 > [ 151.169980] [<ffffffff802ab0d4>] vfs_ioctl+0x74/0x2d0 > [ 151.169983] [<ffffffff802ab3c1>] sys_ioctl+0x91/0xb0 > [ 151.169987] [<ffffffff8020c32e>] system_call+0x7e/0x83 > [ 151.169990] > [ 151.169992] net eth2: device_rename: sysfs_create_symlink failed (-17) > [ 151.176762] e1000: eth2: e1000_watchdog: NIC Link is Up 100 Mbps > Full Duplex, Flow Control: RX/TX > [ 151.209396] sysfs: duplicate filename 'eth0' can not be created > Hi, I get similar warning, while loading the alsa driver [ 191.933548] Advanced Linux Sound Architecture Driver Version 1.0.15 (Tue Oct 23 06:09:18 2007 UTC). [ 191.996323] ALSA sound/core/seq/oss/seq_oss.c:232: can't register device seq [ 192.046004] sysfs: duplicate filename 'audio' can not be created [ 192.086048] WARNING: at fs/sysfs/dir.c:424 sysfs_add_one() [ 192.122882] [ 192.122883] Call Trace: [ 192.154111] [<ffffffff802fbe74>] sysfs_add_one+0x5c/0xc9 [ 192.190144] [<ffffffff802fc498>] create_dir+0x58/0x93 [ 192.224571] [<ffffffff802fc50b>] sysfs_create_dir+0x38/0x4f [ 192.262068] [<ffffffff813a622b>] _spin_unlock+0x26/0x2a [ 192.297448] [<ffffffff8048eb92>] kobject_add+0x13e/0x1f9 [ 192.333292] [<ffffffff8065c746>] device_add+0xac/0x610 [ 192.367959] [<ffffffff8065ccc3>] device_register+0x19/0x1d [ 192.404621] [<ffffffff8065cda6>] device_create+0xdf/0x110 [ 192.440760] [<ffffffff8049301c>] sprintf+0x68/0x6a [ 192.473288] [<ffffffff80ff9b4a>] sound_insert_unit+0xcf/0x172 [ 192.511443] [<ffffffff80ff9ba4>] sound_insert_unit+0x129/0x172 [ 192.549996] [<ffffffff80ff9deb>] register_sound_special_device+0x14a/0x159 [ 192.594734] [<ffffffff8101ed06>] snd_register_oss_device+0x11d/0x18f [ 192.636281] [<ffffffff81034470>] register_oss_dsp+0x54/0x8a [ 192.673031] [<ffffffff802604e6>] mark_held_locks+0x4a/0x6a [ 192.709275] [<ffffffff813a50f1>] mutex_lock_nested+0x2a8/0x2c3 [ 192.747533] [<ffffffff802604e6>] mark_held_locks+0x4a/0x6a [ 192.783587] [<ffffffff813a4e32>] __mutex_unlock_slowpath+0x12e/0x13a [ 192.824764] [<ffffffff802606d0>] trace_hardirqs_on+0x11c/0x147 [ 192.862739] [<ffffffff810378d4>] snd_pcm_oss_register_minor+0x35/0x16b [ 192.904816] [<ffffffff813a4e47>] mutex_unlock+0x9/0xb [ 192.938016] [<ffffffff810208de>] snd_timer_dev_register+0x125/0x12c [ 192.978469] [<ffffffff8101e27c>] snd_device_register+0x73/0xe0 [ 193.016307] [<ffffffff8102e03d>] snd_pcm_timer_init+0xba/0xf5 [ 193.053660] [<ffffffff810245bd>] snd_pcm_dev_register+0x1d7/0x211 [ 193.093096] [<ffffffff8101e1e1>] snd_device_register_all+0x5d/0x85 [ 193.133041] [<ffffffff810197ac>] snd_card_register+0x4e/0x2ea [ 193.170427] [<ffffffff8104f66d>] snd_dummy_probe+0x201/0x21a [ 193.207240] [<ffffffff806601b5>] platform_drv_probe+0x12/0x14 [ 193.244615] [<ffffffff8065ea16>] driver_probe_device+0x132/0x1d2 [ 193.283580] [<ffffffff8065eab6>] __device_attach+0x0/0xb [ 193.318382] [<ffffffff8065eabf>] __device_attach+0x9/0xb [ 193.353173] [<ffffffff8065df69>] bus_for_each_drv+0x47/0x7d [ 193.389534] [<ffffffff8065eb5a>] device_attach+0x75/0x8d [ 193.424319] [<ffffffff8065dd78>] bus_attach_device+0x2d/0x79 [ 193.461202] [<ffffffff8065ca2d>] device_add+0x393/0x610 [ 193.495910] [<ffffffff806605a5>] platform_device_add+0x133/0x16a [ 193.534870] [<ffffffff806607a8>] platform_device_register_simple+0x46/0x64 [ 193.579013] [<ffffffff8208d029>] alsa_card_dummy_init+0x38/0x7e [ 193.617437] [<ffffffff82028678>] kernel_init+0x176/0x2ed [ 193.652173] [<ffffffff813a5d27>] trace_hardirqs_on_thunk+0x35/0x3a [ 193.692147] [<ffffffff802606d0>] trace_hardirqs_on+0x11c/0x147 [ 193.730030] [<ffffffff8020cef8>] child_rip+0xa/0x12 [ 193.762145] [<ffffffff8020c60f>] restore_args+0x0/0x30 [ 193.795821] [<ffffffff8023fc7f>] release_console_sem+0x4d/0x1f5 [ 193.834171] [<ffffffff82028502>] kernel_init+0x0/0x2ed [ 193.867870] [<ffffffff8020ceee>] child_rip+0x0/0x12 [ 193.899987] [ 193.911270] kobject_add failed for audio with -EEXIST, don't try to register things with the same name in the same directory. [ 193.983989] [ 193.983990] Call Trace: [ 194.012668] [<ffffffff8048e440>] kobject_put+0x19/0x1b [ 194.046571] [<ffffffff8048ec17>] kobject_add+0x1c3/0x1f9 [ 194.081445] [<ffffffff8065c746>] device_add+0xac/0x610 [ 194.115311] [<ffffffff8065ccc3>] device_register+0x19/0x1d [ 194.151267] [<ffffffff8065cda6>] device_create+0xdf/0x110 [ 194.186663] [<ffffffff8049301c>] sprintf+0x68/0x6a [ 194.218463] [<ffffffff80ff9b4a>] sound_insert_unit+0xcf/0x172 [ 194.255953] [<ffffffff80ff9ba4>] sound_insert_unit+0x129/0x172 [ 194.293932] [<ffffffff80ff9deb>] register_sound_special_device+0x14a/0x159 [ 194.338208] [<ffffffff8101ed06>] snd_register_oss_device+0x11d/0x18f [ 194.379363] [<ffffffff81034470>] register_oss_dsp+0x54/0x8a [ 194.415853] [<ffffffff802604e6>] mark_held_locks+0x4a/0x6a [ 194.451820] [<ffffffff813a50f1>] mutex_lock_nested+0x2a8/0x2c3 [ 194.489889] [<ffffffff802604e6>] mark_held_locks+0x4a/0x6a [ 194.525880] [<ffffffff813a4e32>] __mutex_unlock_slowpath+0x12e/0x13a [ 194.567098] [<ffffffff802606d0>] trace_hardirqs_on+0x11c/0x147 [ 194.605551] [<ffffffff810378d4>] snd_pcm_oss_register_minor+0x35/0x16b [ 194.647848] [<ffffffff813a4e47>] mutex_unlock+0x9/0xb [ 194.681271] [<ffffffff810208de>] snd_timer_dev_register+0x125/0x12c [ 194.721968] [<ffffffff8101e27c>] snd_device_register+0x73/0xe0 [ 194.760041] [<ffffffff8102e03d>] snd_pcm_timer_init+0xba/0xf5 [ 194.797490] [<ffffffff810245bd>] snd_pcm_dev_register+0x1d7/0x211 [ 194.836921] [<ffffffff8101e1e1>] snd_device_register_all+0x5d/0x85 [ 194.876879] [<ffffffff810197ac>] snd_card_register+0x4e/0x2ea [ 194.914262] [<ffffffff8104f66d>] snd_dummy_probe+0x201/0x21a [ 194.951096] [<ffffffff806601b5>] platform_drv_probe+0x12/0x14 [ 194.988480] [<ffffffff8065ea16>] driver_probe_device+0x132/0x1d2 [ 195.027449] [<ffffffff8065eab6>] __device_attach+0x0/0xb [ 195.062267] [<ffffffff8065eabf>] __device_attach+0x9/0xb [ 195.097053] [<ffffffff8065df69>] bus_for_each_drv+0x47/0x7d [ 195.133408] [<ffffffff8065eb5a>] device_attach+0x75/0x8d [ 195.168203] [<ffffffff8065dd78>] bus_attach_device+0x2d/0x79 [ 195.205133] [<ffffffff8065ca2d>] device_add+0x393/0x610 [ 195.239455] [<ffffffff806605a5>] platform_device_add+0x133/0x16a [ 195.278432] [<ffffffff806607a8>] platform_device_register_simple+0x46/0x64 [ 195.322586] [<ffffffff8208d029>] alsa_card_dummy_init+0x38/0x7e [ 195.361032] [<ffffffff82028678>] kernel_init+0x176/0x2ed [ 195.395798] [<ffffffff813a5d27>] trace_hardirqs_on_thunk+0x35/0x3a [ 195.435764] [<ffffffff802606d0>] trace_hardirqs_on+0x11c/0x147 [ 195.473657] [<ffffffff8020cef8>] child_rip+0xa/0x12 [ 195.505771] [<ffffffff8020c60f>] restore_args+0x0/0x30 [ 195.539459] [<ffffffff8023fc7f>] release_console_sem+0x4d/0x1f5 [ 195.577825] [<ffffffff82028502>] kernel_init+0x0/0x2ed [ 195.611533] [<ffffffff8020ceee>] child_rip+0x0/0x12 [ 195.643690] [ 195.659326] ALSA sound/core/rawmidi.c:1542: unable to register OSS rawmidi device 1:0 [ 195.712879] no UART detected at 0x1 [ 195.739880] Motu MidiTimePiece on parallel port irq: 7 ioport: 0x378 [ 195.804370] usbcore: registered new interface driver snd-usb-audio [ 195.844370] usbcore: registered new interface driver snd-usb-usx2y [ 195.884286] usbcore: registered new interface driver snd-usb-caiaq [ 195.924293] ASoC version 0.13.1 [ 195.946107] ALSA device list: [ 195.966550] #0: Dummy 1 [ 195.984913] #1: Virtual MIDI Card 1 [ 196.009489] #2: MTPAV on parallel port at 0x378 -- Thanks & Regards, Kamalesh Babulal, Linux Technology Center, IBM, ISTL.

5 9

[alsa-devel] [PATCH] SB Live24-External better handling
by Timofei V. Bondarenko 30 Oct '07

30 Oct '07

Hi. This patch improves support for 'SB Live 24-bit Extarnal' USB card. 1) This card can go into muted state when a headphones connected or disconnected. So notify mixer about changes in headphone jack. 2) Add LED controls and procfs support just as in similar Audigy 2 NX card. 3) Rename 'PCM Capture' conrol to 'Mic Capture' to reflect reality: the card may adjust microphone input level only. Regards. Tim.

2 3

[alsa-devel] Removal of sequencer instrument layer
by Takashi Iwai 30 Oct '07

30 Oct '07

Hi, JFYI, together with OPL3 hwdep loader, $SUBJECT is done as I mentioned earlier. This resulted in removal of 5600 LOC in alsa-kernel tree. Yeah, a good number. Takashi

1 0

[alsa-devel] cmipci CM_FLINKOFF disables multinchannel playback
by Timofei V. Bondarenko 30 Oct '07

30 Oct '07

Hi. I found the latest cmipci driver cannot playback in multichannel mode anymore. My hardware is 'CMI8738 model 55' attempting to play 2 or 4 channels through the 2nd DAC i get no sound at all. 6 ch produces sound only at center/bass while front and rear still mute. I found the root of problem in redefinition of some bits of the CM_REG_MISC_CTRL (0x18) register. Was: CM_FLINKON=0x40, CM_FLINKOFF=0x20 Now it is: CM_FLINKON=0x80 CM_FLINKOFF=0x40 Note the CM_FLINKOFF does collide with the CM_MUTECH1=0x40 bit. This bit disables playback on 2nd DAC. The CM_FLINKOFF/CM_FLINKON bits used in the 'Modem' mixer control only. During startup the mixer turns the modem off somewhere and this disables 2nd DAC as well. Complete removal of this 'Modem' control fixes the problem. Is it an acceptable solution? I guess the snd_cmipci_control_switches could be removed too. Regards. Tim. --- alsa-driver-hg20071027/alsa-kernel/pci/cmipci.c 2007-09-18 04:00:10.000000000 +0400 +++ alsa-driver/alsa-kernel/pci/cmipci.c 2007-10-29 17:09:31.000000000 +0300 @@ -2389,7 +2389,6 @@ DEFINE_BIT_SWITCH_ARG(fourch, CM_REG_MIS // DEFINE_BIT_SWITCH_ARG(line_rear, CM_REG_MIXER1, CM_REAR2LIN, 1, 0); // DEFINE_BIT_SWITCH_ARG(line_bass, CM_REG_LEGACY_CTRL, CM_CENTR2LIN|CM_BASE2LIN, 0, 0); // DEFINE_BIT_SWITCH_ARG(joystick, CM_REG_FUNCTRL1, CM_JYSTK_EN, 0, 0); /* now module option */ -DEFINE_SWITCH_ARG(modem, CM_REG_MISC_CTRL, CM_FLINKON|CM_FLINKOFF, CM_FLINKON, 0, 0); #define DEFINE_SWITCH(sname, stype, sarg) \ { .name = sname, \ @@ -2597,7 +2596,7 @@ static struct snd_kcontrol_new snd_cmipc /* card control switches */ static struct snd_kcontrol_new snd_cmipci_control_switches[] __devinitdata = { // DEFINE_CARD_SWITCH("Joystick", joystick), /* now module option */ - DEFINE_CARD_SWITCH("Modem", modem), +// DEFINE_CARD_SWITCH("Modem", modem), };

2 3

[alsa-devel] PATCH: ln2440sbc ac97 support
by Ivan Kuten 29 Oct '07

29 Oct '07

Hello, This patch adds ac97 support for ln2440sbc board from LittleChips. This board is based on s3c2440 SoC + AC97 Realtek ALC650 codec. Existing s3c2443 implementation is slightly modified because s3c2440 and s3c2443 have different AC97 interrupts. Regards, Ivan Signed-off-by: Ivan Kuten <ivan.kuten(a)promwad.com> diff -urN linux-2.6.23/sound/soc/s3c24xx/Kconfig linux-2.6.23-2440/sound/soc/s3c24xx/Kconfig --- linux-2.6.23/sound/soc/s3c24xx/Kconfig 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23-2440/sound/soc/s3c24xx/Kconfig 2007-10-28 01:18:23.000000000 +0200 @@ -34,4 +34,12 @@ Say Y if you want to add support for SoC audio on smdk2443 with the WM9710. +config SND_S3C24XX_SOC_LN2440SBC_ALC650 + tristate "SoC AC97 Audio support for LN2440SBC - ALC650" + depends on SND_S3C24XX_SOC + select SND_S3C2443_SOC_AC97 + select SND_SOC_AC97_CODEC + help + Say Y if you want to add support for SoC audio on ln2440sbc + with the ALC650. diff -urN linux-2.6.23/sound/soc/s3c24xx/Makefile linux-2.6.23-2440/sound/soc/s3c24xx/Makefile --- linux-2.6.23/sound/soc/s3c24xx/Makefile 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23-2440/sound/soc/s3c24xx/Makefile 2007-10-25 01:28:18.000000000 +0200 @@ -10,6 +10,8 @@ # S3C24XX Machine Support snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o +snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o +obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o diff -urN linux-2.6.23/sound/soc/s3c24xx/ln2440sbc_alc650.c linux-2.6.23-2440/sound/soc/s3c24xx/ln2440sbc_alc650.c --- linux-2.6.23/sound/soc/s3c24xx/ln2440sbc_alc650.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.23-2440/sound/soc/s3c24xx/ln2440sbc_alc650.c 2007-10-28 00:27:41.000000000 +0200 @@ -0,0 +1,86 @@ +/* + * SoC audio for ln2440sbc + * + * Copyright 2007 KonekTel, a.s. + * Author: Ivan Kuten + * ivan.kuten(a)promwad.com + * + * Heavily based on smdk2443_wm9710.c + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/ac97.h" +#include "s3c24xx-pcm.h" +#include "s3c24xx-ac97.h" + +static struct snd_soc_machine ln2440sbc; + +static struct snd_soc_dai_link ln2440sbc_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &s3c2443_ac97_dai[0], + .codec_dai = &ac97_dai, +}, +}; + +static struct snd_soc_machine ln2440sbc = { + .name = "LN2440SBC", + .dai_link = ln2440sbc_dai, + .num_links = ARRAY_SIZE(ln2440sbc_dai), +}; + +static struct snd_soc_device ln2440sbc_snd_ac97_devdata = { + .machine = &ln2440sbc, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *ln2440sbc_snd_ac97_device; + +static int __init ln2440sbc_init(void) +{ + int ret; + + ln2440sbc_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!ln2440sbc_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(ln2440sbc_snd_ac97_device, + &ln2440sbc_snd_ac97_devdata); + ln2440sbc_snd_ac97_devdata.dev = &ln2440sbc_snd_ac97_device->dev; + ret = platform_device_add(ln2440sbc_snd_ac97_device); + + if (ret) + platform_device_put(ln2440sbc_snd_ac97_device); + + return ret; +} + +static void __exit ln2440sbc_exit(void) +{ + platform_device_unregister(ln2440sbc_snd_ac97_device); +} + +module_init(ln2440sbc_init); +module_exit(ln2440sbc_exit); + +/* Module information */ +MODULE_AUTHOR("Ivan Kuten"); +MODULE_DESCRIPTION("ALSA SoC ALC650 LN2440SBC"); +MODULE_LICENSE("GPL"); diff -urN linux-2.6.23/sound/soc/s3c24xx/s3c2443-ac97.c linux-2.6.23-2440/sound/soc/s3c24xx/s3c2443-ac97.c --- linux-2.6.23/sound/soc/s3c24xx/s3c2443-ac97.c 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23-2440/sound/soc/s3c24xx/s3c2443-ac97.c 2007-10-28 00:55:57.000000000 +0200 @@ -32,7 +32,7 @@ #include <asm/hardware.h> #include <asm/io.h> -#include <asm/arch/regs-ac97.h> +#include <asm/plat-s3c/regs-ac97.h> #include <asm/arch/regs-gpio.h> #include <asm/arch/regs-clock.h> #include <asm/arch/audio.h> @@ -253,7 +253,7 @@ ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ret = request_irq(IRQ_S3C2443_AC97, s3c2443_ac97_irq, + ret = request_irq(IRQ_S3C244x_AC97, s3c2443_ac97_irq, IRQF_DISABLED, "AC97", NULL); if (ret < 0) { printk(KERN_ERR "s3c24xx-ac97: interrupt request failed.\n"); @@ -266,7 +266,7 @@ static void s3c2443_ac97_remove(struct platform_device *pdev) { - free_irq(IRQ_S3C2443_AC97, NULL); + free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); clk_put(s3c24xx_ac97.ac97_clk); iounmap(s3c24xx_ac97.regs); diff -urN linux-2.6.23/sound/soc/s3c24xx/s3c24xx-ac97.h linux-2.6.23-2440/sound/soc/s3c24xx/s3c24xx-ac97.h --- linux-2.6.23/sound/soc/s3c24xx/s3c24xx-ac97.h 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23-2440/sound/soc/s3c24xx/s3c24xx-ac97.h 2007-10-28 00:32:04.000000000 +0200 @@ -20,6 +20,12 @@ #define AC_CMD_ADDR(x) (x << 16) #define AC_CMD_DATA(x) (x & 0xffff) +#ifdef CONFIG_CPU_S3C2440 +#define IRQ_S3C244x_AC97 IRQ_S3C2440_AC97 +#else +#define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97 +#endif + extern struct snd_soc_cpu_dai s3c2443_ac97_dai[]; #endif /*S3C24XXAC97_H_*/

3 3

[alsa-devel] [PATCH] ac97_patch: compilation warning fix
by Krzysztof Helt 29 Oct '07

29 Oct '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch kills these two compilation warnings if power management is disabled: sound/pci/ac97/ac97_patch.h:86: warning: 'snd_ac97_restore_status' declared 'static' but never defined sound/pci/ac97/ac97_patch.h:87: warning: 'snd_ac97_restore_iec958' declared 'static' but never defined Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The patch was done against kernel 2.6.24-rc1-git4. diff -urp linux-ref/sound/pci/ac97/ac97_patch.h linux-new/sound/pci/ac97/ac97_patch.h --- linux-ref/sound/pci/ac97/ac97_patch.h 2007-10-28 11:11:20.000000000 +0100 +++ linux-new/sound/pci/ac97/ac97_patch.h 2007-10-28 12:46:34.000000000 +0100 @@ -83,8 +83,10 @@ static int snd_ac97_swap_ctl(struct snd_ const char *s2, const char *suffix); static void snd_ac97_rename_vol_ctl(struct snd_ac97 *ac97, const char *src, const char *dst); +#ifdef CONFIG_PM static void snd_ac97_restore_status(struct snd_ac97 *ac97); static void snd_ac97_restore_iec958(struct snd_ac97 *ac97); +#endif static int snd_ac97_info_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); static int snd_ac97_get_enum_double(struct snd_kcontrol *kcontrol,

2 1

[alsa-devel] [PATCH] Dreamcast AICA sound - Get rid of annoying compiler warning
by Adrian McMenamin 29 Oct '07

29 Oct '07

This patch supresses an annoying compiler warning that the variable err may be used uninitialised. Signed-off by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 88dc840..8861d2f 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -237,6 +237,7 @@ static int aica_dma_transfer(int channels, int buffer_size, struct snd_card_aica *dreamcastcard; struct snd_pcm_runtime *runtime; unsigned long flags; + err = 0; dreamcastcard = substream->pcm->private_data; period_offset = dreamcastcard->clicks; period_offset %= (AICA_PERIOD_NUMBER / channels);

2 1

[alsa-devel] [PATCH] Fix for assertion in pulse plugin when calling hw_params on a prepared stream
by Sean McNamara 29 Oct '07

29 Oct '07

Hello, I have opened a bug and recently submitted a patch for it on bugtrack: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3470 All the details are there. But here is a 5-minute summary for you: --The Problem-- (1) All plugins implementing the PCM interface, that I've tested so far, allow calling snd_*_hw_params on a stream when snd_*_state returns SND_PCM_STATE_PREPARED. (2) Applications have used this fact to their advantage, and sometimes employ lazy initialization of the hw params (initialize it once, get to another path in code, oh no wait! we need to initialize them differently...) (3) Per the spec, calling snd_*_hw_params also makes the stream PREPARED, so calling snd_*_hw_params on a stream that has _either_ been explicitly prepared, _or_ had its hw_params previously set, will trigger this problem. (4) The ALSA<->PulseAudio plugin is an exception to (1) above; i.e., it triggers an assertion when exercising this functionality, as if to say, "I can't do that!" (5) Indeed, the PulseAudio client API forbids re-initialization of the parameters in question on a PulseAudio stream, after instantiation. --The Solution-- Only perform the assertion if snd_pcm_state(io->pcm) != SND_PCM_STATE_PREPARED. The assertion is still useful for cases where the stream is mysteriously non-NULL and the stream has not already been prepared; I can't think of a valid case where: (1) The state is not SND_PCM_STATE_PREPARED, and (2) The underlying PulseAudio stream (pcm->stream) is non-NULL, and (3) snd_pcm_hw_params would be expected to do anything meaningful or correct. For this reason, I kept the assertion in, rather than just removing it (which by itself "fixes" the problem with alsaplayer and my test case, which is submitted with the patch.) --Why It Works-- (1) This assertion might be a throwback from before ioplug was so smart; but apparently, ioplug will now call pulse_prepare, which reinitializes the underlying pulseaudio stream, after pulse_hw_params. (2) There is nothing in pulse_hw_params that actually requires any particular state out of pcm->stream. The underlying pulse stream is simply not used in this call. (3) ioplug catches the fact that the hw_params have been changed, and calls back to pulse_prepare to handle the underlying stream teardown and reinitialization (which pulse_prepare does correctly). --Impact-- (1) At least one open source application, alsaplayer, can now play through the ALSA<->PulseAudio plugin, where it could never do so before. (2) The ALSA<->PulseAudio plugin is now more tightly compatible with the other alsa-lib and alsa-plugin backends (dmix, hw, asym, oss, jack, ...) (3) Older versions of sox might have also suffered from this bug, according to information here: http://www.pulseaudio.org/ticket/120 (although the Ubuntu 7.10 version of sox does not exhibit this behavior). --Notes-- (1) I submitted a first-attempt patch that fixed the problem, but it was inefficient: it needlessly called pulse_prepare _prior_ to the main pulse_hw_params function body. This was a result of my lack of understanding of how ioplug was actually saving me from disaster in the background. Be sure to grab 2_pcm_pulse.c.patch rather than pcm_pulse.c.patch. (2) I tested both positive and negative cases; the ALSA<->PulseAudio plugin works with applications that hit the modified code, as well as applications that don't. I've tested both branches of the conditional I added. Thanks, Sean

2 1

[alsa-devel] snd-hdsp in 2.6.22 crashes badly
by ilya .d 29 Oct '07

29 Oct '07

Hello. i'm using RME Hammerfall HDSP9632.. it worked fine with 2.6.17 and 2.6.20 (-gentoo), i don't think gentoo team modifies alsa tree .. as soon as i upgraded to 2.6.22[-gentoo-r8], it booted fine, but when i started gnome-settigs-daemon, kernell generated OOPS, and i couldn't inload the module with modprobe. i disabled gnome-settings-daemon, and reboot. i managed to use hdspconf , hdspmixer + a couple of applications (puredata and snd editor), it was all good. as soon as i ran alsamixer (well , normaly you don't need it for snd-hdsp, but i wanted to configure my second soundcard, which is SB_live, but forgot to give it '-c 1') - so here the same crash heppends. i tried 2.6.20 back - and all absolutely fine with it. looking at hdsp.c in both kernels i spoted quite a few changes , however i don't know much about it ;[ so i attached the details below. -- cheers, ilya .d

2 2

[alsa-devel] ALSA dma ring buffer issue on 2.6.14
by Filipe Braz 27 Oct '07

27 Oct '07

Hi, I am trying to use ALSA with the OMAP2430 embedded platform, using kernel 2.6.14 for low latency voip applications. The problem seems to be that the driver ring buffer is not filled by the alsa kernel/core on time to be sent to the hw codec. So, If I use the following large buffer size / period-size it works fine: aplay --period-size=1024 --buffer-size=8192 -f S16_LE -c 2 -r 8000 pattern.pcm But if I use smaller buffer / period sizes such as the following it does not work: aplay --period-size=32 --buffer-size=128 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=32 --buffer-size=256 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=64 --buffer-size=512 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=128 --buffer-size=512 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=256 --buffer-size=1024 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=512 --buffer-size=2048 -f S16_LE -c 2 -r 8000 pattern.pcm I am playing a special pattern sound file that I verify before each period is transfered to the codec via dma (on the driver). On smaller buffer sizes. the 1st period AFTER the ringbuffer rotates to the beginning, appears to not contain the pattern data (instead is filled with zeros). This does not happens with larger buffer / periods. On all cases the pattern file is large enough so that the dma ring buffer rotates several times until transfer is complete. I have tested this on the pc and it works just fine. Also if I use the OSS driver for the OMAP2430 instead of the ALSA it works fine. ALSA with OSS emulation works significantly better but there is still some distortion. Any ideas why this happens? Is this a known issue or is there a solution for this problem? Thanks in advance, Filipe

1 0

[alsa-devel] Confused about 'default' device
by Timur Tabi 26 Oct '07

26 Oct '07

I'm having a hard time understand when ALSA does sample rate and format conversion, and when it doesn't. When I issue the following command: mplayer -ao alsa filename.mov -v I get this output: Building audio filter chain for 48000Hz/2ch/s24le -> 0Hz/0ch/s8... [libaf] Adding filter dummy [dummy] Was reinitialized: 48000Hz/2ch/s24le [dummy] Was reinitialized: 48000Hz/2ch/s24le alsa-init: requested format: 48000 Hz, 2 channels, 11 alsa-init: using ALSA 1.0.8 alsa-init: setup for 1/2 channel(s) alsa-init: using device default alsa-init: pcm opend in blocking mode alsa-init: format s24le are not supported by hardware, trying default alsa-lib: pcm_plug.c:729:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for 'default' alsa-init: unable to set format: Invalid argument Sure enough, my device can't handle s24le. It can only handle s16be. I was under the impression that ALSA can do format and sample rate conversion if necessary, but I guess that's not always the case. I know ALSA has a bunch of plug-ins for that, but I guess they're not used by default. What do I need to do to tell ALSA to convert s24le to s16be?

4 14

[alsa-devel] New alsa lowlevel driver for the SE-200PCI and SE-90PCI
by sh_okada＠d4.dion.ne.jp 26 Oct '07

26 Oct '07

HI. I mail from Japan. I wrote a driver for the SE-200PCI and SE-90PCI what are salling in Japan. And I want to merge it to current source tree. How should I do? Thanks.

4 16

[alsa-devel] [PATCH] make sbiload to use hwdep interface
by Takashi Iwai 26 Oct '07

26 Oct '07

This patch makes sbiload using the hwdep interface instead of seq instr layer. This simplifies the logic a lot. Takashi diff -r 9b306261208b seq/sbiload/README --- a/seq/sbiload/README Mon Oct 22 15:16:35 2007 +0200 +++ b/seq/sbiload/README Fri Oct 26 16:10:50 2007 +0200 @@ -1,4 +1,4 @@ This is sbiload - an OPL2/3 FM instrumen -This is sbiload - an OPL2/3 FM instrument loader for ALSA sequencer +This is sbiload - an OPL2/3 FM instrument loader for ALSA hwdep Written by Uros Bizjak <uros(a)kss-loka.si> Web: http://www.kss-loka.si/~uros/sbiload.html @@ -14,62 +14,42 @@ By default it will install into /usr/loc By default it will install into /usr/local/bin, change INSTDIR in make.conf to change this. -Before use ----------- - -Please check that the correct sequencer modules are loaded for your -sound card. You need to load snd-synth-opl3 module. - Usage ----- - loadsbi [-p client:port] [-4] [-l] [-P path] [-v level] instfile drumfile + loadsbi [-D device] [-4] [-P path] [-v level] instfile drumfile - -p, --port=client:port - An ALSA client and port number to use + -D, --device=name - An ALSA hwdep name to use -4 --opl3 - Four operators (OPL3) file type - -l, --list - List possible output ports -P, --path=path - Specify the patch path -v, --verbose=level - Verbose level -Because you may have more than one sound card and each sound -card may have several MIDI connections, you have to tell sbiload -which one to use. +When you have multiple sound cards and/or there may be multiple OPL2/3 +synth devices, you have to tell sbiload which one to use, and pass its +hwdep name via -D option. Otherwise, sbiload will load the data onto +the first found device. -First find out what the possibilities are for your system: +To find out the hwdep name of the OPL devices, check +/proc/asound/hwdep file first. It has the list of hwdep devices +currently available on your system. -If you run: - - sbiload -l + % cat /proc/asound/hwdep + 00-00: Emux WaveTable + 00-01: OPL3 FM + 01-00: OPL3 FM -it will give you a list of the devices. On my system I get: - - Port Client name Port name - 64:0 0: MIDI Synth MIDI 0-0 - 65:0 OPL3 FM Synth OPL3 FM Synth port +In the example above, you have two OPL3 devices. To the first one +(00-01), which means the first card (00-) and the second device (01), +pass "-D hw:0,1". For the second OPL3 device (01-00), pass "-D +hw:1,0" instead. - -Only OPLx FM Synth port can be used with sbiload. In the example above -the first one is the external Midi port, and the second is the internal -four-operator FM synthesizer (OPL3). - -If you don't see anything listed then check that the correct ALSA -modules are loaded as in the section "Before use". Running sbiload --------------- -You specify the port to use with the -p option, or by setting the -environment variable ALSA_OUT_PORT. +Pass the instrument and drum file data as arguments. -For example to use port 0 on client 65 to load four-operator FM -instrument - - sbiload -p 65:0 --opl3 std.o3 drums.o3 - -or - - export ALSA_OUT_PORT=65:0 - sbiload --opl3 std.o3 drums.o3 + % sbiload std.o3 drums.o3 Acknowledgements diff -r 9b306261208b seq/sbiload/sbiload.c --- a/seq/sbiload/sbiload.c Mon Oct 22 15:16:35 2007 +0200 +++ b/seq/sbiload/sbiload.c Fri Oct 26 16:10:50 2007 +0200 @@ -1,5 +1,5 @@ /* - * ALSA sequencer SBI FM instrument loader + * ALSA hwdep SBI FM instrument loader * Copyright (c) 2000 Uros Bizjak <uros(a)kss-loka.si> * * This program is free software; you can redistribute it and/or modify @@ -16,11 +16,13 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * + * + * Oct. 2007 - Takashi Iwai <tiwai(a)suse.de> + * Changed to use hwdep instead of obsoleted seq-instr interface */ #include <errno.h> #include <getopt.h> -#include <alsa/sound/ainstr_fm.h> #include <stdio.h> #include <stdlib.h> #include <string.h> @@ -29,24 +31,12 @@ #include <fcntl.h> #include <errno.h> #include <alsa/asoundlib.h> +#include <alsa/sound/asound_fm.h> #include <sys/time.h> #include <unistd.h> -typedef struct sbi_header -{ - char key[4]; - char name[32]; -} -sbi_header_t; - -typedef struct sbi_inst -{ - sbi_header_t header; #define DATA_LEN_2OP 16 #define DATA_LEN_4OP 24 - char data[DATA_LEN_4OP]; -} -sbi_inst_t; /* offsets for SBI params */ #define AM_VIB 0 @@ -59,21 +49,20 @@ sbi_inst_t; #define CONNECTION 10 #define OFFSET_4OP 11 -/* offsets in sbi_header.name for SBI extensions */ -#define ECHO_DELAY 25 -#define ECHO_ATTEN 26 -#define CHORUS_SPREAD 27 -#define TRNSPS 28 -#define FIX_DUR 29 -#define MODES 30 -#define FIX_KEY 31 +/* offsets for SBI extensions */ +#define ECHO_DELAY 0 +#define ECHO_ATTEN 1 +#define CHORUS_SPREAD 2 +#define TRNSPS 3 +#define FIX_DUR 4 +#define MODES 5 +#define FIX_KEY 6 /* Options for the command */ #define HAS_ARG 1 static struct option long_opts[] = { - {"port", HAS_ARG, NULL, 'p'}, + {"device", HAS_ARG, NULL, 'D'}, {"opl3", 0, NULL, '4'}, - {"list", 0, NULL, 'l'}, {"path", HAS_ARG, NULL, 'P'}, {"verbose", HAS_ARG, NULL, 'v'}, {"version", 0, NULL, 'V'}, @@ -83,87 +72,35 @@ static struct option long_opts[] = { /* Number of elements in an array */ #define NELEM(a) ( sizeof(a)/sizeof((a)[0]) ) -#define ADDR_PARTS 4 /* Number of part in a port description addr 1:2:3:4 */ -#define SEP ", \t" /* Separators for port description */ - -#define SBI_FILE_TYPE_2OP 0 -#define SBI_FILE_TYPE_4OP 1 +enum { + FM_PATCH_UNKNOWN, + FM_PATCH_OPL2, + FM_PATCH_OPL3 +}; /* Default file type */ -int file_type = SBI_FILE_TYPE_2OP; +static int file_type = FM_PATCH_OPL2; /* Default verbose level */ -int verbose = 0; +static int verbose = 0; /* Global declarations */ -snd_seq_t *seq_handle; - -int seq_client; -int seq_port; -int seq_dest_client; -int seq_dest_port; +static snd_hwdep_t *handle; #ifndef PATCHDIR #define PATCHDIR "/usr/share/sounds/opl3" #endif -char *patchdir = PATCHDIR; +static char *patchdir = PATCHDIR; /* Function prototypes */ -static void show_list (); -static void show_usage (); -static void show_op (fm_instrument_t * fm_instr); +static void show_usage (void); +static void show_op (struct sbi_patch * instr, int type); -static int load_patch (fm_instrument_t * fm_instr, int bank, int prg, char *name); +static int load_patch (struct sbi_patch * instr); static int load_file (int bank, char *filename); -static int parse_portdesc (char *portdesc); -static int init_client (); - -static void -ignore_errors (const char *file, int line, const char *function, int err, const char *fmt, ...) -{ - /* ignore */ -} - -/* - * Show a list of possible output ports that midi could be sent to. - */ -static void -show_list () { - snd_seq_client_info_t *cinfo; - snd_seq_port_info_t *pinfo; - - int client, err; - - snd_lib_error_set_handler (ignore_errors); - if ((err = snd_seq_open (&seq_handle, "hw", SND_SEQ_OPEN_DUPLEX, 0)) < 0) { - fprintf (stderr, "Could not open sequencer: %s\n", snd_strerror (err)); - return; - } - - printf (" Port %-30.30s %s\n", "Client name", "Port name"); - snd_seq_client_info_alloca(&cinfo); - snd_seq_client_info_set_client(cinfo, -1); - while (snd_seq_query_next_client(seq_handle, cinfo) >= 0) { - client = snd_seq_client_info_get_client(cinfo); - snd_seq_port_info_alloca(&pinfo); - snd_seq_port_info_set_client(pinfo, client); - snd_seq_port_info_set_port(pinfo, -1); - while (snd_seq_query_next_port(seq_handle, pinfo) >= 0) { - unsigned int cap; - - cap = (SND_SEQ_PORT_CAP_SUBS_WRITE | SND_SEQ_PORT_CAP_WRITE); - if ((snd_seq_port_info_get_capability(pinfo) & cap) == cap) { - printf ("%3d:%-3d %-30.30s %s\n", - client, snd_seq_port_info_get_port(pinfo), - snd_seq_client_info_get_name(cinfo), - snd_seq_port_info_get_name(pinfo)); - } - } - } - snd_seq_close (seq_handle); -} +static int init_hwdep (const char *name); /* * Show usage message @@ -172,11 +109,10 @@ show_usage () { show_usage () { char **cpp; static char *msg[] = { - "Usage: sbiload [-p client:port] [-4] [-l] [-P path] [-v level] instfile drumfile", + "Usage: sbiload [-D device] [-4] [-P path] [-v level] instfile drumfile", "", - " -p client:port - A alsa client and port number to send midi to", + " -D device - hwdep device string", " -4 - four operators file type (default = two ops)", - " -l - List possible output ports that could be used", " -P path - Specify the patch path", " -v level - Verbose level (default = 0)", " -V - Show version", @@ -199,169 +135,86 @@ show_version () { * Show instrument FM operators */ static void -show_op (fm_instrument_t * fm_instr) { +show_op (struct sbi_patch * inst, int type) { int i = 0; + int ofs = 0; do { - printf (" OP%i: flags: %s %s %s %s\011OP%i: flags: %s %s %s %s\n", - i, - fm_instr->op[i].am_vib & (1 << 7) ? "AM" : " ", - fm_instr->op[i].am_vib & (1 << 6) ? "VIB" : " ", - fm_instr->op[i].am_vib & (1 << 5) ? "EGT" : " ", - fm_instr->op[i].am_vib & (1 << 4) ? "KSR" : " ", - i + 1, - fm_instr->op[i + 1].am_vib & (1 << 7) ? "AM" : " ", - fm_instr->op[i + 1].am_vib & (1 << 6) ? "VIB" : " ", - fm_instr->op[i + 1].am_vib & (1 << 5) ? "EGT" : " ", - fm_instr->op[i + 1].am_vib & (1 << 4) ? "KSR" : ""); - printf (" OP%i: MULT = 0x%x" "\011\011OP%i: MULT = 0x%x\n", - i, fm_instr->op[i].am_vib & 0x0f, - i + 1, fm_instr->op[i + 1].am_vib & 0x0f); - printf (" OP%i: KSL = 0x%x TL = 0x%.2x\011OP%i: KSL = 0x%x TL = 0x%.2x\n", - i, (fm_instr->op[i].ksl_level >> 6) & 0x03, fm_instr->op[i].ksl_level & 0x3f, - i + 1, (fm_instr->op[i + 1].ksl_level >> 6) & 0x03, fm_instr->op[i + 1].ksl_level & 0x3f); - printf (" OP%i: AR = 0x%x DL = 0x%x\011OP%i: AR = 0x%x DL = 0x%x\n", - i, (fm_instr->op[i].attack_decay >> 4) & 0x0f, fm_instr->op[i].attack_decay & 0x0f, - i + 1, (fm_instr->op[i + 1].attack_decay >> 4) & 0x0f, fm_instr->op[i + 1].attack_decay & 0x0f); - printf (" OP%i: SL = 0x%x RR = 0x%x\011OP%i: SL = 0x%x RR = 0x%x\n", - i, (fm_instr->op[i].sustain_release >> 4) & 0x0f, fm_instr->op[i].sustain_release & 0x0f, - i + 1, (fm_instr->op[i + 1].sustain_release >> 4) & 0x0f, fm_instr->op[i + 1].sustain_release & 0x0f); - printf (" OP%i: WS = 0x%x\011\011OP%i: WS = 0x%x\n", - i, fm_instr->op[i].wave_select & 0x07, - i + 1, fm_instr->op[i + 1].wave_select & 0x07); - printf (" FB = 0x%x, %s\n", - (fm_instr->feedback_connection[i / 2] >> 1) & 0x07, - fm_instr->feedback_connection[i / 2] & (1 << 0) ? "parallel" : "serial"); - i += 2; + unsigned char val; + val = inst->data[AM_VIB + ofs]; + printf (" OP%i: flags: %s %s %s %s", i, + val & (1 << 7) ? "AM" : " ", + val & (1 << 6) ? "VIB" : " ", + val & (1 << 5) ? "EGT" : " ", + val & (1 << 4) ? "KSR" : " "); + val = inst->data[AM_VIB + ofs + 1]; + printf ("\011OP%i: flags: %s %s %s %s\n", i + 1, + val & (1 << 7) ? "AM" : " ", + val & (1 << 6) ? "VIB" : " ", + val & (1 << 5) ? "EGT" : " ", + val & (1 << 4) ? "KSR" : " "); + val = inst->data[AM_VIB + ofs]; + printf (" OP%i: MULT = 0x%x", i, val & 0x0f); + val = inst->data[AM_VIB + ofs + 1]; + printf ("\011\011OP%i: MULT = 0x%x\n", i + 1, val & 0x0f); + + val = inst->data[KSL_LEVEL + ofs]; + printf (" OP%i: KSL = 0x%x TL = 0x%.2x", i, + (val >> 6) & 0x03, val & 0x3f); + val = inst->data[KSL_LEVEL + ofs + 1]; + printf ("\011OP%i: KSL = 0x%x TL = 0x%.2x\n", i + 1, + (val >> 6) & 0x03, val & 0x3f); + val = inst->data[ATTACK_DECAY + ofs]; + printf (" OP%i: AR = 0x%x DL = 0x%x", i, + (val >> 4) & 0x0f, val & 0x0f); + val = inst->data[ATTACK_DECAY + ofs + 1]; + printf ("\011OP%i: AR = 0x%x DL = 0x%x\n", i + 1, + (val >> 4) & 0x0f, val & 0x0f); + val = inst->data[SUSTAIN_RELEASE + ofs]; + printf (" OP%i: SL = 0x%x RR = 0x%x", i, + (val >> 4) & 0x0f, val & 0x0f); + val = inst->data[SUSTAIN_RELEASE + ofs + 1]; + printf ("\011OP%i: SL = 0x%x RR = 0x%x\n", i + 1, + (val >> 4) & 0x0f, val & 0x0f); + val = inst->data[WAVE_SELECT + ofs]; + printf (" OP%i: WS = 0x%x", i, val & 0x07); + val = inst->data[WAVE_SELECT + ofs + 1]; + printf ("\011\011OP%i: WS = 0x%x\n", i + 1, val & 0x07); + val = inst->data[CONNECTION + ofs]; + printf (" FB = 0x%x, %s\n", (val >> 1) & 0x07, + val & (1 << 0) ? "parallel" : "serial"); + i += 2; + ofs += OFFSET_4OP; } - while (i == (fm_instr->type == FM_PATCH_OPL3) << 1); + while (i == (type == FM_PATCH_OPL3) << 1); printf (" Extended data:\n" " ED = %.3i EA = %.3i CS = %.3i TR = %.3i\n" " FD = %.3i MO = %.3i FK = %.3i\n", - fm_instr->echo_delay, fm_instr->echo_atten, fm_instr->chorus_spread, fm_instr->trnsps, - fm_instr->fix_dur, fm_instr->modes, fm_instr->fix_key); -} - -/* - * Check the result of the previous instr event - */ -static int -check_result (int evtype) { - snd_seq_event_t *pev; - int err; - - for (;;) { - if ((err = snd_seq_event_input (seq_handle, &pev)) < 0) { - fprintf (stderr, "Unable to read event input: %s\n", - snd_strerror (err)); - return -ENXIO; - } - if (pev->type == SND_SEQ_EVENT_RESULT && pev->data.result.event == evtype) - break; - snd_seq_free_event (pev); - } - err = pev->data.result.result; - snd_seq_free_event (pev); - - return err; + inst->extension[ECHO_DELAY], inst->extension[ECHO_ATTEN], + inst->extension[CHORUS_SPREAD], inst->extension[TRNSPS], + inst->extension[FIX_DUR], inst->extension[MODES], + inst->extension[FIX_KEY]); } /* * Send patch to destination port */ static int -load_patch (fm_instrument_t * fm_instr, int bank, int prg, char *name) { - snd_instr_header_t *put; - snd_seq_instr_t id; - snd_seq_event_t ev; +load_patch (struct sbi_patch * inst) { - size_t size; - int err; - - if (verbose > 1) { - printf ("%.3i: [OPL%i] %s\n", prg, fm_instr->type == FM_PATCH_OPL3 ? 3 : 2, name); - show_op (fm_instr); - } - - if ((err = snd_instr_fm_convert_to_stream (fm_instr, name, &put, &size)) < 0) { - fprintf (stderr, "Unable to convert instrument %.3i to stream: %s\n", - prg, snd_strerror (err)); - return -1; - } - memset(&id, 0, sizeof(id)); - id.std = SND_SEQ_INSTR_TYPE2_OPL2_3; - id.prg = prg; - id.bank = bank; - snd_instr_header_set_id(put, &id); - - /* build event */ - memset (&ev, 0, sizeof (ev)); - ev.source.client = seq_client; - ev.source.port = seq_port; - ev.dest.client = seq_dest_client; - ev.dest.port = seq_dest_port; - - ev.flags = SND_SEQ_EVENT_LENGTH_VARUSR; - ev.queue = SND_SEQ_QUEUE_DIRECT; - -__again: - ev.type = SND_SEQ_EVENT_INSTR_PUT; - ev.data.ext.len = size; - ev.data.ext.ptr = put; - - if ((err = snd_seq_event_output (seq_handle, &ev)) < 0) { + ssize_t ret; + ret = snd_hwdep_write(handle, inst, sizeof(*inst)); + if (ret != sizeof(*inst)) { fprintf (stderr, "Unable to write an instrument %.3i put event: %s\n", - prg, snd_strerror (err)); + inst->prog, snd_strerror (ret)); return -1; } - if ((err = snd_seq_drain_output (seq_handle)) < 0) { - fprintf (stderr, "Unable to write instrument %.3i data: %s\n", prg, - snd_strerror (err)); - return -1; - } - - err = check_result (SND_SEQ_EVENT_INSTR_PUT); - if (err >= 0) { - if (verbose) - printf ("Loaded instrument %.3i, bank %.3i: %s\n", prg, bank, name); - return 0; - } else if (err == -EBUSY) { - snd_instr_header_t *remove; - - snd_instr_header_alloca(&remove); - snd_instr_header_set_cmd(remove, SND_SEQ_INSTR_FREE_CMD_SINGLE); - snd_instr_header_set_id(remove, snd_instr_header_get_id(put)); - - /* remove instrument */ - ev.type = SND_SEQ_EVENT_INSTR_FREE; - ev.data.ext.len = snd_instr_header_sizeof(); - ev.data.ext.ptr = remove; - - if ((err = snd_seq_event_output (seq_handle, &ev)) < 0) { - fprintf (stderr, "Unable to write an instrument %.3i free event: %s\n", - prg, snd_strerror (err)); - return -1; - } - - if ((err = snd_seq_drain_output (seq_handle)) < 0) { - fprintf (stderr, "Unable to write instrument %.3i data: %s\n", prg, - snd_strerror (err)); - return -1; - } - - if ((err = check_result (SND_SEQ_EVENT_INSTR_FREE)) < 0) { - fprintf (stderr, "Instrument %.3i, bank %.3i - free error: %s\n", - prg, bank, snd_strerror (err)); - return -1; - } - goto __again; - } - - fprintf (stderr, "Instrument %.3i, bank %.3i - put error: %s\n", - prg, bank, snd_strerror (err)); - return -1; + if (verbose) + printf ("Loaded instrument %.3i, bank %.3i: %s\n", + inst->prog, inst->bank, inst->name); + return 0; } /* @@ -369,66 +222,45 @@ __again: */ static void load_sb (int bank, int fd) { - int len, i; + int len; int prg; - - sbi_inst_t sbi_instr; - fm_instrument_t fm_instr; + struct sbi_patch inst; int fm_instr_type; - len = (file_type == SBI_FILE_TYPE_4OP) ? DATA_LEN_4OP : DATA_LEN_2OP; + len = (file_type == FM_PATCH_OPL2) ? DATA_LEN_4OP : DATA_LEN_2OP; for (prg = 0;; prg++) { - if (read (fd, &sbi_instr.header, sizeof (sbi_header_t)) < (ssize_t)sizeof (sbi_header_t)) + inst.prog = prg; + inst.bank = bank; + + if (read (fd, inst.key, 4) != 4) break; - if (!strncmp (sbi_instr.header.key, "SBI\032", 4) || !strncmp (sbi_instr.header.key, "2OP\032", 4)) { + if (!memcmp (inst.key, "SBI\032", 4) || !memcmp (inst.key, "2OP\032", 4)) { + len = DATA_LEN_2OP; fm_instr_type = FM_PATCH_OPL2; - } else if (!strncmp (sbi_instr.header.key, "4OP\032", 4)) { + } else if (!strncmp (inst.key, "4OP\032", 4)) { + len = DATA_LEN_4OP; fm_instr_type = FM_PATCH_OPL3; } else { - fm_instr_type = 0; if (verbose) printf ("%.3i: wrong instrument key!\n", prg); + fm_instr_type = FM_PATCH_UNKNOWN; } - if (read (fd, &sbi_instr.data, len) < len) + if (read (fd, &inst.name, sizeof(inst.name)) != sizeof(inst.name) || + read (fd, &inst.extension, sizeof(inst.extension)) != sizeof(inst.extension) || + read (fd, &inst.data, len) != len) break; - if (fm_instr_type == 0) + if (fm_instr_type == FM_PATCH_UNKNOWN) continue; - memset (&fm_instr, 0, sizeof (fm_instr)); - fm_instr.type = fm_instr_type; - - for (i = 0; i < 2; i++) { - fm_instr.op[i].am_vib = sbi_instr.data[AM_VIB + i]; - fm_instr.op[i].ksl_level = sbi_instr.data[KSL_LEVEL + i]; - fm_instr.op[i].attack_decay = sbi_instr.data[ATTACK_DECAY + i]; - fm_instr.op[i].sustain_release = sbi_instr.data[SUSTAIN_RELEASE + i]; - fm_instr.op[i].wave_select = sbi_instr.data[WAVE_SELECT + i]; - } - fm_instr.feedback_connection[0] = sbi_instr.data[CONNECTION]; - - if (fm_instr_type == FM_PATCH_OPL3) { - for (i = 0; i < 2; i++) { - fm_instr.op[i + 2].am_vib = sbi_instr.data[OFFSET_4OP + AM_VIB + i]; - fm_instr.op[i + 2].ksl_level = sbi_instr.data[OFFSET_4OP + KSL_LEVEL + i]; - fm_instr.op[i + 2].attack_decay = sbi_instr.data[OFFSET_4OP + ATTACK_DECAY + i]; - fm_instr.op[i + 2].sustain_release = sbi_instr.data[OFFSET_4OP + SUSTAIN_RELEASE + i]; - fm_instr.op[i + 2].wave_select = sbi_instr.data[OFFSET_4OP + WAVE_SELECT + i]; - } - fm_instr.feedback_connection[1] = sbi_instr.data[OFFSET_4OP + CONNECTION]; + if (verbose > 1) { + printf ("%.3i: %s\n", inst.prog, inst.name); + show_op (&inst, fm_instr_type); } - fm_instr.echo_delay = sbi_instr.header.name[ECHO_DELAY]; - fm_instr.echo_atten = sbi_instr.header.name[ECHO_ATTEN]; - fm_instr.chorus_spread = sbi_instr.header.name[CHORUS_SPREAD]; - fm_instr.trnsps = sbi_instr.header.name[TRNSPS]; - fm_instr.fix_dur = sbi_instr.header.name[FIX_DUR]; - fm_instr.modes = sbi_instr.header.name[MODES]; - fm_instr.fix_key = sbi_instr.header.name[FIX_KEY]; - - if (load_patch (&fm_instr, bank, prg, sbi_instr.header.name) < 0) + if (load_patch (&inst) < 0) break; } return; @@ -461,92 +293,57 @@ load_file (int bank, char *filename) { } /* - * Parse port description + * Open a hwdep device */ -static int -parse_portdesc (char *portdesc) { - char *astr; - char *cp; - int a[ADDR_PARTS]; - int count; +static int open_hwdep (const char *name) +{ + int err; + snd_hwdep_info_t *info; - if (portdesc == NULL) - return -1; + if ((err = snd_hwdep_open (&handle, name, SND_HWDEP_OPEN_WRITE)) < 0) + return err; - for (astr = strtok (portdesc, SEP); astr; astr = strtok (NULL, SEP)) { - for (cp = astr, count = 0; cp && *cp; cp++) { - if (count < ADDR_PARTS) - a[count++] = atoi (cp); - cp = strchr (cp, ':'); - if (cp == NULL) - break; - } - - if (count == 2) { - seq_dest_client = a[0]; - seq_dest_port = a[1]; - } else { - printf ("Addresses in %d parts not supported yet\n", count); - return -1; - } + snd_hwdep_info_alloca(&info); + if (!snd_hwdep_info (handle, info)) { + snd_hwdep_iface_t iface = snd_hwdep_info_get_iface(info); + if (iface == SND_HWDEP_IFACE_OPL2 || iface == SND_HWDEP_IFACE_OPL3) + return 0; } - return 0; + snd_hwdep_close(handle); + handle = NULL; + return -EINVAL; } -/* - * Open sequencer, create client port and - * subscribe client to destination port - */ static int -init_client () { - char name[64]; - snd_seq_port_subscribe_t *sub; - snd_seq_addr_t addr; +init_hwdep (const char *name) { + int err; + char tmpname[16]; - if ((err = snd_seq_open (&seq_handle, "hw", SND_SEQ_OPEN_DUPLEX, 0)) < 0) { - fprintf (stderr, "Could not open sequencer: %s\n", snd_strerror (err)); + if (!name || !*name) { + /* auto probe */ + int card = -1; + snd_ctl_t *ctl; + + while (!snd_card_next(&card) && card >= 0) { + int dev; + sprintf(tmpname, "hw:%d", card); + if (snd_ctl_open(&ctl, tmpname, 0) < 0) + continue; + dev = -1; + while (!snd_ctl_hwdep_next_device(ctl, &dev) && dev >= 0) { + sprintf(tmpname, "hw:%d,%d", card, dev); + if (!open_hwdep(tmpname)) + return 0; + } + snd_ctl_close(ctl); + } + fprintf (stderr, "Can't find any OPL3 hwdep device\n"); return -1; } - seq_client = snd_seq_client_id (seq_handle); - if (seq_client < 0) { - snd_seq_close (seq_handle); - fprintf (stderr, "Unable to determine my client number: %s\n", - snd_strerror (err)); - return -1; - } - - sprintf (name, "sbiload - %i", getpid ()); - if ((err = snd_seq_set_client_name (seq_handle, name)) < 0) { - snd_seq_close (seq_handle); - fprintf (stderr, "Unable to set client info: %s\n", - snd_strerror (err)); - return -1; - } - - if ((seq_port = snd_seq_create_simple_port (seq_handle, "Output", - SND_SEQ_PORT_CAP_READ | SND_SEQ_PORT_CAP_WRITE, - SND_SEQ_PORT_TYPE_SPECIFIC)) < 0) { - snd_seq_close (seq_handle); - fprintf (stderr, "Unable to create a client port: %s\n", - snd_strerror (seq_port)); - return -1; - } - - snd_seq_port_subscribe_alloca(&sub); - addr.client = seq_client; - addr.port = seq_port; - snd_seq_port_subscribe_set_sender(sub, &addr); - addr.client = seq_dest_client; - addr.port = seq_dest_port; - snd_seq_port_subscribe_set_dest(sub, &addr); - snd_seq_port_subscribe_set_exclusive(sub, 1); - - if ((err = snd_seq_subscribe_port (seq_handle, sub)) < 0) { - snd_seq_close (seq_handle); - fprintf (stderr, "Unable to subscribe destination port: %s\n", - snd_strerror (errno)); + if ((err = open_hwdep (name)) < 0) { + fprintf (stderr, "Could not open hwdep %s: %s\n", name, snd_strerror (err)); return -1; } @@ -558,40 +355,25 @@ init_client () { * and close sequencer */ static void -finish_client () +finish_hwdep () { - snd_seq_port_subscribe_t *sub; - snd_seq_addr_t addr; - int err; - - snd_seq_port_subscribe_alloca(&sub); - addr.client = seq_client; - addr.port = seq_port; - snd_seq_port_subscribe_set_sender(sub, &addr); - addr.client = seq_dest_client; - addr.port = seq_dest_port; - snd_seq_port_subscribe_set_dest(sub, &addr); - if ((err = snd_seq_unsubscribe_port (seq_handle, sub)) < 0) { - fprintf (stderr, "Unable to unsubscribe destination port: %s\n", - snd_strerror (errno)); - } - snd_seq_close (seq_handle); + snd_hwdep_close(handle); + handle = NULL; } /* * Load a .SBI FM instrument patch * sbiload [-p client:port] [-l] [-P path] [-v level] instfile drumfile * - * -p, --port=client:port - An ALSA client and port number to use + * -D, --device=name - An ALSA hwdep name to use * -4 --opl3 - four operators file type - * -l, --list - List possible output ports that could be used * -P, --path=path - Specify the patch path * -v, --verbose=level - Verbose level */ int main (int argc, char **argv) { char opts[NELEM (long_opts) * 2 + 1]; - char *portdesc; + char *name; char *cp; int c; struct option *op; @@ -604,7 +386,7 @@ main (int argc, char **argv) { *cp++ = ':'; } - portdesc = NULL; + name = NULL; /* Deal with the options */ for (;;) { @@ -613,11 +395,11 @@ main (int argc, char **argv) { break; switch (c) { - case 'p': - portdesc = optarg; + case 'D': + name = optarg; break; case '4': - file_type = SBI_FILE_TYPE_4OP; + file_type = FM_PATCH_OPL3; break; case 'v': verbose = atoi (optarg); @@ -625,9 +407,6 @@ main (int argc, char **argv) { case 'V': show_version(); exit (1); - case 'l': - show_list (); - exit (0); case 'P': patchdir = optarg; break; @@ -637,30 +416,14 @@ main (int argc, char **argv) { } } - if (portdesc == NULL) { - portdesc = getenv ("ALSA_OUT_PORT"); - if (portdesc == NULL) { - fprintf (stderr, "No client/port specified.\n" - "You must supply one with the -p option or with the\n" - "environment variable ALSA_OUT_PORT\n"); - exit (1); - } - } - - /* Parse port description to dest_client and dest_port */ - if (parse_portdesc (portdesc) < 0) { - return 1; - } - - /* Initialize client and subscribe to destination port */ - if (init_client () < 0) { + if (init_hwdep (name) < 0) { return 1; } /* Process instrument and drum file */ if (optind < argc) { if (load_file (0, argv[optind++]) < 0) { - finish_client(); + finish_hwdep(); return 1; } } else { @@ -668,15 +431,14 @@ main (int argc, char **argv) { } if (optind < argc) { if (load_file (128, argv[optind]) < 0) { - finish_client(); + finish_hwdep(); return 1; } } else { fprintf(stderr, "Warning: drum file was not specified\n"); } - /* Unsubscribe destination port and close client */ - finish_client(); + finish_hwdep(); return 0; }

1 0

[alsa-devel] [PATCH] Change asound_fm.h in alsa-lib for OPL3 hwdep loader
by Takashi Iwai 26 Oct '07

26 Oct '07

The change for alsa-lib to make sbiload compile. Takashi diff -r f6057aeb5008 include/sound/asound_fm.h --- a/include/sound/asound_fm.h Thu Oct 25 15:36:03 2007 +0200 +++ b/include/sound/asound_fm.h Fri Oct 26 13:09:29 2007 +0200 @@ -29,16 +29,16 @@ #define SNDRV_DM_FM_MODE_OPL2 0x00 #define SNDRV_DM_FM_MODE_OPL3 0x01 -typedef struct snd_dm_fm_info { +struct snd_dm_fm_info { unsigned char fm_mode; /* OPL mode, see SNDRV_DM_FM_MODE_XXX */ unsigned char rhythm; /* percussion mode flag */ -} snd_dm_fm_info_t; +}; /* * Data structure composing an FM "note" or sound event. */ -typedef struct snd_dm_fm_voice { +struct snd_dm_fm_voice { unsigned char op; /* operator cell (0 or 1) */ unsigned char voice; /* FM voice (0 to 17) */ @@ -60,25 +60,25 @@ typedef struct snd_dm_fm_voice { unsigned char left; /* stereo left */ unsigned char right; /* stereo right */ unsigned char waveform; /* 3 bits: waveform shape */ -} snd_dm_fm_voice_t; +}; /* * This describes an FM note by its voice, octave, frequency number (10bit) * and key on/off. */ -typedef struct snd_dm_fm_note { +struct snd_dm_fm_note { unsigned char voice; /* 0-17 voice channel */ unsigned char octave; /* 3 bits: what octave to play */ unsigned int fnum; /* 10 bits: frequency number */ unsigned char key_on; /* set for active, clear for silent */ -} snd_dm_fm_note_t; +}; /* * FM parameters that apply globally to all voices, and thus are not "notes" */ -typedef struct snd_dm_fm_params { +struct snd_dm_fm_params { unsigned char am_depth; /* amplitude modulation depth (1=hi) */ unsigned char vib_depth; /* vibrato depth (1=hi) */ unsigned char kbd_split; /* keyboard split */ @@ -90,17 +90,17 @@ typedef struct snd_dm_fm_params { unsigned char tomtom; unsigned char cymbal; unsigned char hihat; -} snd_dm_fm_params_t; +}; /* * FM mode ioctl settings */ -#define SNDRV_DM_FM_IOCTL_INFO _IOR('H', 0x20, snd_dm_fm_info_t) +#define SNDRV_DM_FM_IOCTL_INFO _IOR('H', 0x20, struct snd_dm_fm_info) #define SNDRV_DM_FM_IOCTL_RESET _IO ('H', 0x21) -#define SNDRV_DM_FM_IOCTL_PLAY_NOTE _IOW('H', 0x22, snd_dm_fm_note_t) -#define SNDRV_DM_FM_IOCTL_SET_VOICE _IOW('H', 0x23, snd_dm_fm_voice_t) -#define SNDRV_DM_FM_IOCTL_SET_PARAMS _IOW('H', 0x24, snd_dm_fm_params_t) +#define SNDRV_DM_FM_IOCTL_PLAY_NOTE _IOW('H', 0x22, struct snd_dm_fm_note) +#define SNDRV_DM_FM_IOCTL_SET_VOICE _IOW('H', 0x23, struct snd_dm_fm_voice) +#define SNDRV_DM_FM_IOCTL_SET_PARAMS _IOW('H', 0x24, struct snd_dm_fm_params) #define SNDRV_DM_FM_IOCTL_SET_MODE _IOW('H', 0x25, int) /* for OPL3 only */ #define SNDRV_DM_FM_IOCTL_SET_CONNECTION _IOW('H', 0x26, int) @@ -112,4 +112,21 @@ typedef struct snd_dm_fm_params { #define SNDRV_DM_FM_OSS_IOCTL_SET_MODE 0x24 #define SNDRV_DM_FM_OSS_IOCTL_SET_OPL 0x25 +/* + * Patch Record - fixed size for write + */ + +#define FM_KEY_SBI "SBI\032" +#define FM_KEY_2OP "2OP\032" +#define FM_KEY_4OP "4OP\032" + +struct sbi_patch { + unsigned char prog; + unsigned char bank; + char key[4]; + char name[25]; + char extension[7]; + unsigned char data[32]; +}; + #endif /* __SOUND_ASOUND_FM_H */

1 0

[alsa-devel] HDA: Add support for Samsung Q1 Ultra Vista edition.
by Tobin Davis 26 Oct '07

26 Oct '07

This patch adds full record and playback support for the Samsung Q1 Ultra - Vista model (different codec than the earlier Q1 Ultra models). Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] [PATCH] AD1986A 6-channel bugs
by zhejiang 26 Oct '07

26 Oct '07

Hi, I have a ICH8M + AD1986A machine. In 6 channel modes, if adjust the volumes when playing 2-channel or 4-channel music, codec will be muted and all widget's Amp-Out vals will be reset. If you play 2-channel files in 6 channel modes,there is noise in the jacks. By doing experiments, I found that the 6-channel bug was caused by sharing channels. If play 2-channel audio file,the rest channels will share channels with the first 2-channel. The HDA spec doesn’t define the exact behavior under such circumstances. Majority of the codec chips support sharing channel, but seem that AD1986A doesn’t support. I wrote a patch and tested it in my hardware,now it works well. diff -Nur a/alsa-driver-1.0.15/alsa-kernel/pci/hda/hda_codec.c b/alsa-driver-1.0.15/alsa-kernel/pci/hda/hda_codec.c --- a/alsa-driver-1.0.15/alsa-kernel/pci/hda/hda_codec.c 2007-10-25 10:03:29.000000000 -0400 +++ b/alsa-driver-1.0.15/alsa-kernel/pci/hda/hda_codec.c 2007-10-25 10:02:46.000000000 -0400 @@ -2506,7 +2506,8 @@ if (chs >= (i + 1) * 2) /* independent out */ snd_hda_codec_setup_stream(codec, nids[i], stream_tag, i * 2, format); - else /* copy front */ + else if (codec->vendor_id != 0x11d41986) + /* copy front if it is NOT AD1986A*/ snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0, format); } In alsa bugzilla and maillist, I found a lot of issues about AD1986A 6-channel .Anybody know the detail about AD1986A codec chip and give any comments? Here is a old description of the issue. > > Hi, > > I have a ICH8M + AD1986A machine and run alsa-driver-1.0.15rc. > > I found that : > In 6 channel mode,if adjust the front or pcm volume when playing music, > all channel will be muted,but if adjust the volume alone(don't play > music at the same time),every thing is ok. > I use printk to display the verb in the azx_send_cmd(), > when adjusting the volume in either case, there are just two verb > (0x300) sent to the nid(0x1b) [correct action]. > But in the first case, the dac widget amp-out vals are all affected. > > > Some other interesting thing, > If i adjust the Sur or CLFE volume when playing music, the printk show > that correct verb is sent, but the codec status don't change. > If I adjust them alone,every thing is ok.

2 3

[alsa-devel] aplay appication failed to open the alsa device
by Tharmarajan G 24 Oct '07

24 Oct '07

Hi, I am working on a Development Board which has the PXA270 processor and TSC2301 Audio Codec Chip. The TSC2301 Chip is interfaced with the processor by SPI interface for reading/writing the TSC2301's registers. The TSC2301 Chip is interfaced with the processor by I2S interface for sending/receiving Audio Data to/from the TSC2301 Audio Codec Chip. I have downloaded the pxa-uda1380.c alsa driver source for uda1380 audio codec chip and modified it for tsc2301 . I am using the kernel version 2.6.22 and cross-compiler version arm-linux-gcc 3.4.1. I have built the kernel image with ALSA audio support and the applications aplay and arecord for playing and recording audio data using ALSA api. I have got the alsa related entries in the /proc filesystem . But when playing the aplay, it failed to open the device. [root@SIRIUS /flash2]#./aplay /usr/local/Songs/AjRaj-Anthimanthirai-Theme4.mp3 ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:545: audio open error: No such file or directory [root@SIRIUS /flash2]# /aplay -D /dev/snd/pcmC0D0p /usr/local/Songs/AjRaj-Anthimanthirai-Theme4.mp3 ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM /dev/snd/pcmC0D0p aplay: main:545: audio open error: No such file or directory What could be the issue ? Am I missing anyting in the driver for alsa ? Regards, Tharmarajan G

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[alsa-devel] New hardware report
by Alain DANGER 24 Oct '07

24 Oct '07

Hi. I managed to get the sound work with ALSA on the following MB : Shuttle AK32VN. OS : Kubuntu 7.10. Regards. Alain Danger

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[alsa-devel] mplayer plays video twice as fast with Alsa than with OSS
by Timur Tabi 24 Oct '07

24 Oct '07

Hi, I wrote an Alsa device driver that supposedly works with the OSS emulation, but one of our customers is reporting that mplayer with the "-ao oss" option plays video twice at double speed. If they select "-ao alsa" instead, then it plays normally. Is this a known problem?

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Re: [alsa-devel] [Jackit-devel] RME HDSPe floating point question
by Didier Villevalois 24 Oct '07

24 Oct '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, (Cross-posting my report - last time i do this) Great news (almost for me in anyways). RME HDSPe PCI *works as is* with the current driver. Firmware uploads as well. I had nothing to do to make it work! (I think the ExpressCard should work as well...) Maybe the alsa-project wiki page can be updated. Little by little in the next weeks, I will delve into the driver's code to identify parts that needs to be enhanced in order to take profit of the 32bit floating point format that the HDSPe card can provide. When ready, I'll contact RME to request them for informations about the activation of this mode. Didier. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHHcG6VE1mALQL3JsRAjeZAKCJqicClHgy1UDc1YvAywuw1FokyQCeJO2d BhWpkxL8xLGxKB3zmbWFQpk= =wpbm -----END PGP SIGNATURE-----

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[alsa-devel] [RFC] ALSA scenario API
by Liam Girdwood 23 Oct '07

23 Oct '07

A little late with this but "better late than never".....I did promise a few people that I would post what was imho needed for ALSA scenarios. Also, apologies for cross posting, but it some of this probably applies to both lists... ALSA probably needs functionality to define and set audio usage scenarios. A scenario is simply a list of sound card mixer settings for a given use case. This is quite important for mobile devices (e.g. Smartphones, GPS, PDA, etc) with complex audio routing and power requirements. Some example scenarios could be :- "HiFi Headphone Playback" "HiFi Speaker Playback" "Handset Phone Call" "Headset Phone Call" etc... Scenarios are changed atm by either manually setting each control (e.g. using alsamixer or alsa-lib in apps) or by using alsactl to restore a scenario. The above two methods are not very portable between devices and applications. We would ideally like to just say "set Phone Call Handset" for a regular handset phone call or "set HiFi Headphones" for MP3 playback. ALSA also needs to have functionality to alias mixer controls. This would allow a single master volume/mute control to be exposed to user space and have it aliased to different hardware controls based upon the current scenario. e.g. Many mobile devices have different physical volume controls for speaker and headphones. This makes it somewhat difficult for user space apps to change the correct volume (without being bound to the hardware). We would like to be able to just say "set volume to 0dB" and have the alias take care of the correct physical CODEC mixer for the current scenario. Proposal: ======== 1. Add some new scenario functions to (s)alsa-lib. e.g. snd_scenario_set(); snd_scenario_get(); snd_scenario_list(); _set() would set the scenario e.g. snd_scenario_set(card, SND_VOICE_CALL_HANDSET); We would need to have a list of standard scenarios so that applications could be more portable. Vendor scenario's should also be supported. _get() would return the current scenario. _list() would return a list of supported scenarios for the given device. 2. Add some new mixer alias functions to set/get the current "master" volume/mute. e.g. snd_kcontrol *snd_scenario_volume_set(card, type, volume); snd_kcontrol *snd_scenario_switch_set(card, type, switch); where type is playback, capture, sidetone, bypass, etc This would set the volume for the current scenario and would guarantee always to get/set the _correct_ volume/switch. 3. Add a new function to return the active PCM for a given scenario. e.g. snd_scenario_get_pcm(card, scenario); Would return the given PCM for a particular scenario. i.e. the WM9713 driver has PCM 0.0 for high quality (higher power) audio (MP3), PCM 0.1 for lower power voice and PCM 0.2 for lowest power and quality system sounds. Note: scenarios may need to be more tightly bound to PCM's rather than cards. 4. Define a file format to represent each scenario. Probably base this upon alsactl's format as alsamixer could be used to quickly generate good settings and then save them with alsactl. Alias, scenario names and PCM would need to be added later somehow. 5. (Optional) Allow scenarios to define active kcontrols. This would allow tools like alsamixer to only display mixer controls for the given scenario. Controls that don't effect the scenario would not be rendered. (similar to alsamixers Playback|Capture|All tab) 6. (OHM list) Add input driver (or similar) for sound card user events. We need to be able to send events like "Headphone Jack Insertion", "Mic Jack Insertion", etc to user space. This would allow changing of scenario and power configuration depending on policy and configuration. Just my 2p Unfortunately I'm not in a position to implement any of this functionality. However, it's probably useful to folks working on mobile devices. Maybe a good project for Google SoC. HTH Liam

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Re: [alsa-devel] [PATCH] asihpi checkpatch clean plus control name refactor
by Takashi Iwai 23 Oct '07

23 Oct '07

At Tue, 23 Oct 2007 16:48:20 +1300, Eliot Blennerhassett wrote: > > checkpatch.pl --no-tree now reports no errors on asihpi.c > Redo control names to be strings only (not using control index) > > Signed-off-by: Eliot Blennerhassett <linux(a)audioscience.com> Thanks, applied to HG tree now. (BTW, I couldn't access the ftp. Anon-ftp isn't allowed?) Takashi

1 0

[alsa-devel] [PATCH] asihpi checkpatch clean plus control name refactor
by Eliot Blennerhassett 23 Oct '07

23 Oct '07

checkpatch.pl --no-tree now reports no errors on asihpi.c Redo control names to be strings only (not using control index) Signed-off-by: Eliot Blennerhassett <linux(a)audioscience.com> Patch here; (70k) ftp://ftp.audioscience.com/internet/download/beta/asihpi_checkpatch_clean.d… regards Eliot

1 0

[alsa-devel] USB device gives wrong data?
by william estrada 23 Oct '07

23 Oct '07

Hi group, I think I have found a problem with the USB sound driver? It appears that the current drive returns a x'81' for silence but it should be a x'7f'. When I use my laptop's sound device and record without a mic, I get a file full of x'7f's. If I do the same with my cheap-o USB device I get a file full of x'81's. The sound quality from the laptop is unacceptable, so that is way I want to use the USB device. I have written a recording program that I have a 'silence' filter that filters out segments of dead air. It works with the laptop's sound device but not the USB device. The filter works by remove x'7f's after allowing a selectable amount. I get the results using arecord which does not have the filtering. The USB device is a C -Media Electronics, Inc. I am running FC6. > [root@laptop ~]# lsmod |grep snd > snd_usb_audio 81985 0 > snd_usb_lib 21057 1 snd_usb_audio > snd_rawmidi 28481 1 snd_usb_lib > snd_hwdep 13637 1 snd_usb_audio > snd_ali5451 27853 1 > snd_ac97_codec 99813 1 snd_ali5451 > snd_ac97_bus 6593 1 snd_ac97_codec > snd_seq_dummy 8133 0 > snd_seq_oss 37057 0 > snd_seq_midi_event 11841 1 snd_seq_oss > snd_seq 57009 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event > snd_seq_device 12365 4 snd_rawmidi,snd_seq_dummy,snd_seq_oss,snd_seq > snd_pcm_oss 46433 0 > snd_mixer_oss 20545 1 snd_pcm_oss > snd_pcm 80965 4 snd_usb_audio,snd_ali5451,snd_ac97_codec,snd_pcm_oss > snd_timer 26693 2 snd_seq,snd_pcm > snd 58181 140 snd_usb_audio,snd_rawmidi,snd_hwdep,snd_ali5451,snd_ac97_codec,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer > soundcore 12193 1 snd > snd_page_alloc 14281 1 snd_pcm Where do I report this 'bug' or how do I fix it? Thanks for your time. Sorry for the multiple posting, not sure where is should have gone. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

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Re: [alsa-devel] WM9712 monoout
by Mike Rapoport 22 Oct '07

22 Oct '07

Liam Girdwood wrote: > On Sun, 2007-10-21 at 09:35 +0200, Mike Rapoport wrote: >> Hello, >> >> I'm trying to enable phone on a board with WM9715 codec. >> I have not found a control responsible for mute/unmute the MONOUT. >> My question is whether I should add this control as platform specific control, >> or it would be correct to patch sound/soc/codecs/wm9712.c? > > It should probably go in the codec driver. It looks like an omission. > >> Also, "Mono Playback Volume" control is not inverted, which seems to be a bug. >> > > Could you send a patch ? Sure. A also found that treble and bass need to be inverted. > Many Thanks > > Liam > Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> sound/soc/codecs/wm9712.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 986b5d5..427cb61 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -102,7 +102,8 @@ SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0), SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0), SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0), SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0), -SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 0), +SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), +SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), @@ -145,8 +146,8 @@ SOC_ENUM("Bass Control", wm9712_enum[5]), SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1), SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1), SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0), -SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 0), -SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 0), +SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1), +SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),

2 3

[alsa-devel] snd-pcsp doesn't accept the index parameter
by Alexander E. Patrakov 22 Oct '07

22 Oct '07

It is impossible to put pcsp at a stable index without the trivial patch below. Signed-Off-By: Alexander E. Patrakov <patrakov(a)ums.usu.ru> --- pcsp.c +++ pcsp.c @@ -27,6 +27,8 @@ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for pcsp soundcard."); module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for pcsp soundcard."); module_param(enable, bool, 0444); -- Alexander E. Patrakov

2 1

[alsa-devel] [PATCH] hda: STAC9228 updated DMUX nid.
by Matthew Ranostay 22 Oct '07

22 Oct '07

Changed the dmux for STAC9228 from ADC1MUX to ADC0MUX to avoid confusion. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 7cf5e23f804e pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Oct 19 08:23:00 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Fri Oct 19 13:47:35 2007 -0400 @@ -2777,7 +2777,7 @@ static int patch_stac927x(struct hda_cod case 0x10280209: /* STAC 9228 */ spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; - spec->dmux_nid = 0x1c; + spec->dmux_nid = 0x1b; break; default: spec->num_dmics = 0;

2 1

[alsa-devel] SALSA Lib - TLV Deprecated warnings
by J. Scott Merritt 22 Oct '07

22 Oct '07

It appears that if SALSA Lib 0.0.12 is built -without- TLV enabled in the configuration options, then simply -including- asoundlib.h from an application program will trigger deprecated warnings on some of the TLV read/write function definitions (even though they are not called from the application). Is there something we can do to the header files to eliminate these warnings ? Thanks, Scott.

2 3

[alsa-devel] alsa-lib not installing plugins
by Larry D. Pyeatt 22 Oct '07

22 Oct '07

Hi, I wrote a new plugin for alsa-lib. After running make, there are .o and .lo files in src/pcm. However, when I run make install, nothing gets put into /usr/lib/alsa-lib. I first tried installing alsa-lib-1.0.15rc3. Then I tried alsa-lib-1.0.13rc3 and alsa-lib-1.0.14rc4. All of them did the same thing. Am I missing something obvious? Since they are all doing the same thing, it must be either that I just don't know what I'm doing, or something else in my system is not working. I would appreciate any suggestions on what could cause this behavior. Thanks, -- Larry D. Pyeatt, PhD. pyeatt(a)cs.ttu.edu http://www.cs.ttu.edu/~pyeatt

2 1

Re: [alsa-devel] hda-intel: no soundcard with current linus' git tree
by Maxim Levitsky 21 Oct '07

21 Oct '07

Hi, I understand why this happens. Your sound card has a single "headphone" output, and since it is single, it is called "Master" the stac you have also has a real master volume control, that controls all the DACs, not just the headphones. I added support for it, thus two controls collided. The following patch will rename the "Master" output to "Headphone" Note that on the stac 9205/9204, the only one that has no master volume (the volume knob) but has a "Master Volume" amplifier widget, this will probably rename the "Master Volume" to Headphones. This chip is quite different, thus I think this should be handled in its patch function. I see what I can do later. Meanwhile the rename from "Master" to Headphones" is the only regression. Best regards, Maxim Levitsky PS: I reply to this mail with the patch.

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Re: [alsa-devel] hda-intel: no soundcard with current linus' git tree
by Jan-Simon Möller 21 Oct '07

21 Oct '07

I've also no sound. Had it working using the alsa.git tree _before_ the last merge (>48h). Needed to use model=asus for this fujitsu v5505 here. Now it doesn't work, even with other model-options (fujitsu, laptop). legolas:/proc/asound # cat cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc400000 irq 22 legolas:/proc/asound # cat pcm 00-01: Conexant Digital : Conexant Digital : playback 1 00-00: CONEXANT Analog : CONEXANT Analog : playback 1 : capture 1 legolas:/proc/asound # cat hwdep 00-00: HDA Codec 0 legolas:/proc/asound # cat devices 2: : timer 3: [ 0- 1]: digital audio playback 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0- 0]: hardware dependent 7: [ 0] : control 8: : sequencer legolas:/proc/asound # cat card0/codec#0 Codec: Conexant CX20549 (Venice) Address: 0 Vendor Id: 0x14f15045 Subsystem Id: 0x1734110e Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x140]: 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x10 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x2b 0x2b] Pincap 0x0810014: OUT EAPD Detect Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 2 0x19 0x17* Node 0x11 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x2b 0x2b] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x0121401f: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Power: 0x0 Connection: 2 0x19 0x17* Node 0x12 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x2b 0x2b] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x01a19020: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x20: IN Power: 0x0 Connection: 2 0x19 0x17* Node 0x13 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x214571f0: [Jack] SPDIF Out at Sep Rear Conn = Optical, Color = Yellow Pin-ctls: 0x00: Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081124: IN Detect Pin Default 0x97a7012e: [Fixed] Mic at Int Riser Conn = Analog, Color = Unknown Pin-ctls: 0x24: IN Node 0x15 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Node 0x16 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0b, mute=1 Amp-Out vals: [0x06] Node 0x17 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x14, nsteps=0x2b, stepsize=0x05, mute=1 Amp-In vals: [0x94 0x94] [0x94 0x94] [0x94 0x94] [0x94 0x94] [0x94 0x94] Power: 0x0 Connection: 5 0x19 0x14 0x12 0x11 0x15 Node 0x18 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x40]: 48000 bits [0x6]: 16 20 formats [0x5]: PCM AC3 Node 0x19 [Audio Output] wcaps 0xc11: Stereo PCM: rates [0x540]: 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: 0x0 Node 0x1a [Audio Input] wcaps 0x100d0b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x17, stepsize=0x05, mute=1 Amp-In vals: [0x17 0x17] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Power: 0x0 Connection: 5 0x17 0x14* 0x12 0x11 0x15 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Best regards Jan-Simon

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Re: [alsa-devel] [Jackit-devel] RME HDSP driver enhancements for HDSPe cards
by John Rigg 19 Oct '07

19 Oct '07

On Fri, Oct 19, 2007 at 08:58:00AM +0100, Ed W wrote: > RME are also helpful to the extent that you ask them specific questions > and in the name of writing drivers. They should give you hardware > details, etc > > > As an aside, apart from CPU load I will be suprised if turning floats > into 32bit ints makes any difference to the audio, but I agree that it > would be a nice feature to have! IIRC the new RME cards convert 24 bit integers to 32 bit floats in hardware (and vice versa), thus saving apps like jackd (and the CPU) the trouble. jackd doesn't support that yet, but AFAIK it shouldn't be too much effort to change that. John

1 0

[alsa-devel] [PATCH] hda: Add dmux to STAC 9228
by Matthew Ranostay 19 Oct '07

19 Oct '07

Added a dmux to the STAC9228 cards with DMIC support. And added a STAC_DIGITAL_INPUT_SOURCE macro for repeating mixer code. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 821a4558dd47 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Oct 18 17:38:17 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Thu Oct 18 17:35:56 2007 -0400 @@ -430,6 +430,16 @@ static struct hda_verb stac9205_core_ini {} }; +#define STAC_DIGITAL_INPUT_SOURCE(cnt) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Digital Input Source", \ + .count = cnt, \ + .info = stac92xx_dmux_enum_info, \ + .get = stac92xx_dmux_enum_get, \ + .put = stac92xx_dmux_enum_put,\ + } + #define STAC_INPUT_SOURCE(cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -482,14 +492,7 @@ static struct snd_kcontrol_new stac925x_ }; static struct snd_kcontrol_new stac9205_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Input Source", - .count = 1, - .info = stac92xx_dmux_enum_info, - .get = stac92xx_dmux_enum_get, - .put = stac92xx_dmux_enum_put, - }, + STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0), STAC_VOLKNOB(0x24), @@ -521,6 +524,7 @@ static struct snd_kcontrol_new stac922x_ static struct snd_kcontrol_new stac927x_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(3), STAC_VOLKNOB(0x24), STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), @@ -2773,6 +2777,7 @@ static int patch_stac927x(struct hda_cod case 0x10280209: /* STAC 9228 */ spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; + spec->dmux_nid = 0x1c; break; default: spec->num_dmics = 0;

3 5

[alsa-devel] ALSA Bluetooth connection issue
by 19 Oct '07

19 Oct '07

Hi All: Im using ALSA 1.0.14 and bluez 3.20 and linux 2.6.18.The linux 2.6.18 (montavista) is running on an embeded development board(using S3C2410). When i tried to use 'aplay' to connect my bluetooth headset, i got the following error from ALSA: $ aplay -D bluetooth /1.wav audio[326]: Accepted new client connection on unix socket (fd=8) audio[326]: Package PKT_TYPE_CFG_REQ:0 hcid[318]: /org/bluez: org.bluez.Manager.FindAdapter() audio[326]: Got path /org/bluez/hci0 for adapter with address 00:80:E1:00:00:00 hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceHandles() hcid[318]: no matching session found. creating a new one hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceHandles() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceHandles() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache audio[326]: Audio service discovery completed with success hcid[318]: /org/bluez/hci0: org.bluez.Adapter.FinishRemoteServiceTransaction() hcid[318]: found matching session, removing from list audio[326]: Found Hansfree record audio[326]: Discovered Headset service on RFCOMM channel 1 audio[326]: Found Headset record audio[326]: Found Audio Sink audio[326]: Selecting default device audio[326]: avdtp_ref(0x3b018): ref=2 audio[326]: a2dp_source_request_stream: selected SEP 0x37250 audio[326]: avdtp_ref(0x3b018): ref=3 hcid[318]: link_key_request (sba=00:80:E1:00:00:00, dba=00:0D:3C:EB:53:F0) audio[326]: AVDTP: connected signaling channel to 00:0D:3C:EB:53:F0 audio[326]: session_cb audio[326]: DISCOVER request succeeded audio[326]: seid 1 type 1 media 0 in use 0 audio[326]: seid 2 type 1 media 0 in use 0 audio[326]: session_cb audio[326]: GET_CAPABILITIES request succeeded audio[326]: seid 1 type 1 media 0 hcid[318]: link_key_request (sba=00:80:E1:00:00:00, dba=00:0D:3C:EB:53:F0) hcid[318]: link_key_request (sba=00:80:E1:00:00:00, dba=00:0D:3C:EB:53:F0) audio[326]: session_cb audio[326]: GET_CAPABILITIES request succeeded audio[326]: seid 2 type 1 media 0 audio[326]: Discovery complete audio[326]: avdtp_set_configuration(0x3b018): int_seid=1, acp_seid=1 audio[326]: session_cb audio[326]: SET_CONFIGURATION request succeeded audio[326]: SBC Source: Set_Configuration_Cfm audio[326]: avdtp_ref(0x3b018): ref=4 audio[326]: stream state changed: IDLE -> CONFIGURED audio[326]: session_cb audio[326]: OPEN request succeeded audio[326]: stream state changed: CONFIGURED -> OPEN audio[326]: AVDTP: connected transport channel to 00:0D:3C:EB:53:F0 audio[326]: SBC Source: Open_Cfm audio[326]: session_cb audio[326]: START request succeeded audio[326]: SBC Source: Start_CAlignment trap: bluetoothd-serv (326) PC=0x0001be44 Instr=0xe1c320b0 Address=0xbec85c2d FSR 0x813 fm audio[326]: SBC Source SEP 0x37250 locked hcid[318]: Service owner exited: :1.2 aplay: main:550: audio open error: Input/output error hcid[318]: Audio service (audio) was killed by signal 7 ALSA's aplay reports an "Input/output error". Later i digged into alsa-lib-1.0.14/src/pcm/pcm.c and found that snd_config_search_definition() returns an error. It seems ALSA didn't find the proper conf definition or something like that. My ".asoundrc" is as followed: pcm.a2dpd { type a2dpd } pcm.bluetooth { type bluetooth device 00:0D:3C:EB:53:F0 } pcm.!default { type hw card 0 device 0 } pcm.card0 { type hw card 0 } ctl.card0 { type hw card 0 } Does someone have any idea about what's going wrong in this? Thanks in advance! semiyd _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

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[alsa-devel] ALSA-for custom Operating System????
by Anoop Thomas 19 Oct '07

19 Oct '07

First of all don't make fun of me if i am asking a foolishness.Could you plz tell me can i use alsa for my custom OS which concentrates more on audio side.It doesn't uses linux kernel.Does it make a difference???? (ie. is it enough to give sufficient data to ALSA to help me to make the audio out of the system).If I can plz give me a brief description.... Help me plz................. --------------------------------- Forgot the famous last words? Access your message archive online. Click here.

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[alsa-devel] [PATCH] s3c2443-ac97: compilation fix
by Krzysztof Helt 18 Oct '07

18 Oct '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The Samsung S3C24xx uses new architecture file layout in the post 2.6.23 kernel. This patch fixes include path for the s3c2443-ac97.c. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The patch was done against the 2.6.23-git13 kernel. diff -urp linux-ref/sound/soc/s3c24xx/s3c2443-ac97.c linux-new/sound/soc/s3c24xx/s3c2443-ac97.c --- linux-ref/sound/soc/s3c24xx/s3c2443-ac97.c 2007-10-09 22:31:38.000000000 +0200 +++ linux-new/sound/soc/s3c24xx/s3c2443-ac97.c 2007-10-18 22:27:43.547470033 +0200 @@ -32,7 +32,7 @@ #include <asm/hardware.h> #include <asm/io.h> -#include <asm/arch/regs-ac97.h> +#include <asm/plat-s3c/regs-ac97.h> #include <asm/arch/regs-gpio.h> #include <asm/arch/regs-clock.h> #include <asm/arch/audio.h> ---------------------------------------------------------------------- Jaros�aw Drob, Przyjdziemy i Spie***ymy - poznaj kandydata! Kliknij >>> http://link.interia.pl/f1bef

1 0

[alsa-devel] Kernel hierarchy
by Adrian McMenamin 18 Oct '07

18 Oct '07

Why are sound drivers in /sound and not /drivers/sound? Ok, not the world's most pressing problem, but enquiring minds want to know! Adrian

2 1

[alsa-devel] [PATCH] hda: STAC9228 DMIC Support
by Matthew Ranostay 18 Oct '07

18 Oct '07

Added DMIC support for STAC9228 cards, as well as adding terminators to other hda_nid DMIC arrays, and added a new Vendor ID tag. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r e91f57f8127a pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Tue Oct 09 11:58:41 2007 +0200 +++ b/pci/hda/hda_codec.c Wed Oct 17 11:59:47 2007 -0400 @@ -55,6 +55,7 @@ static struct hda_vendor_id hda_vendor_i { 0x10ec, "Realtek" }, { 0x1057, "Motorola" }, { 0x1106, "VIA" }, + { 0x111d, "IDT" }, { 0x11d4, "Analog Devices" }, { 0x13f6, "C-Media" }, { 0x14f1, "Conexant" }, diff -r e91f57f8127a pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Oct 09 11:58:41 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Wed Oct 17 13:06:58 2007 -0400 @@ -182,8 +182,9 @@ static hda_nid_t stac925x_dac_nids[1] = 0x02, }; -static hda_nid_t stac925x_dmic_nids[1] = { - 0x15, +static hda_nid_t stac925x_dmic_nids[2] = { + 0x15, + 0x00, /* terminator */ }; static hda_nid_t stac922x_adc_nids[2] = { @@ -202,6 +203,11 @@ static hda_nid_t stac927x_mux_nids[3] = 0x15, 0x16, 0x17 }; +static hda_nid_t stac927x_dmic_nids[3] = { + 0x13, 0x14, + 0x00, /* terminator */ +}; + static hda_nid_t stac9205_adc_nids[2] = { 0x12, 0x13 }; @@ -210,8 +216,9 @@ static hda_nid_t stac9205_mux_nids[2] = 0x19, 0x1a }; -static hda_nid_t stac9205_dmic_nids[2] = { +static hda_nid_t stac9205_dmic_nids[3] = { 0x17, 0x18, + 0x00, /* terminator */ }; static hda_nid_t stac9200_pin_nids[8] = { @@ -2716,7 +2723,6 @@ static int patch_stac927x(struct hda_cod spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->num_dmics = 0; spec->init = d965_core_init; spec->mixer = stac927x_mixer; break; @@ -2725,7 +2731,6 @@ static int patch_stac927x(struct hda_cod spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->num_dmics = 0; spec->init = d965_core_init; spec->mixer = stac927x_mixer; break; @@ -2734,9 +2739,18 @@ static int patch_stac927x(struct hda_cod spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->num_dmics = 0; spec->init = stac927x_core_init; spec->mixer = stac927x_mixer; + } + + switch (codec->subsystem_id) { + case 0x1028020A: /* STAC 9228 */ + case 0x10280209: /* STAC 9228 */ + spec->dmic_nids = stac927x_dmic_nids; + spec->num_dmics = 2; + break; + default: + spec->num_dmics = 0; } spec->multiout.dac_nids = spec->dac_nids; @@ -2798,7 +2812,7 @@ static int patch_stac9205(struct hda_cod spec->mux_nids = stac9205_mux_nids; spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids); spec->dmic_nids = stac9205_dmic_nids; - spec->num_dmics = ARRAY_SIZE(stac9205_dmic_nids); + spec->num_dmics = 2; spec->dmux_nid = 0x1d; spec->init = stac9205_core_init;

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[alsa-devel] alsa-driver-1.0.15rc2 build issues on RHEL4/CentOS4
by James Pearson 18 Oct '07

18 Oct '07

alsa-driver-1.0.15rc2 won't build cleanly on RHEL4/CentOS4 (2.6.9 based kernel). I've included the relevant errors below after running: configure --with-redhat make -i The problem with misc/ac97_bus.c, I've already reported at: <http://mailman.alsa-project.org/pipermail/alsa-devel/2007-September/003119.…> James Pearson Build errors: CC [M] /tmp/alsa-driver-1.0.15rc2/misc/ac97_bus.o In file included from /tmp/alsa-driver-1.0.15rc2/misc/ac97_bus.c:3: /tmp/alsa-driver-1.0.15rc2/misc/../alsa-kernel/ac97_bus.c: In function `ac97_bus_suspend': /tmp/alsa-driver-1.0.15rc2/misc/../alsa-kernel/ac97_bus.c:35: error: too few arguments to function /tmp/alsa-driver-1.0.15rc2/misc/../alsa-kernel/ac97_bus.c: In function `ac97_bus_resume': /tmp/alsa-driver-1.0.15rc2/misc/../alsa-kernel/ac97_bus.c:45: error: too few arguments to function make[4]: [/tmp/alsa-driver-1.0.15rc2/misc/ac97_bus.o] Error 1 (ignored) CC [M] /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/cs5535audio_pm.o In file included from /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/cs5535audio_pm.c:2: /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/../../alsa-kernel/pci/cs5535audio/cs5535audio_pm.c: In function `snd_cs5535audio_suspend': /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/../../alsa-kernel/pci/cs5535audio/cs5535audio_pm.c:77: error: void value not ignored as it ought to be /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/../../alsa-kernel/pci/cs5535audio/cs5535audio_pm.c: In function `snd_cs5535audio_resume': /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/../../alsa-kernel/pci/cs5535audio/cs5535audio_pm.c:95: error: void value not ignored as it ought to be make[5]: [/tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/cs5535audio_pm.o] Error 1 (ignored) LD [M] /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/snd-cs5535audio.o ld: /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/cs5535audio_pm.o: No such file: No such file or directory make[5]: [/tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/snd-cs5535audio.o] Error 1 (ignored) Building modules, stage 2. MODPOST *** Warning: "class_device_destroy" [/tmp/alsa-driver-1.0.15rc2/pci/asihpi/snd-asihpi.ko] undefined! *** Warning: "ac97_bus_type" [/tmp/alsa-driver-1.0.15rc2/pci/ac97/snd-ac97-codec.ko] undefined! *** Warning: "platform_device_alloc" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-portman2x4.ko] undefined! *** Warning: "platform_device_add" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-portman2x4.ko] undefined! *** Warning: "platform_device_put" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-portman2x4.ko] undefined! *** Warning: "platform_device_alloc" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-mts64.ko] undefined! *** Warning: "platform_device_add" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-mts64.ko] undefined! *** Warning: "platform_device_put" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-mts64.ko] undefined!

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[alsa-devel] [patch] snd_pcm_dmix_close: raise semaphore if unable to discard
by Mike Gorse 18 Oct '07

18 Oct '07

This patch causes snd_pcm_dmix_close() to up a semaphore after downing it if it is unable to discard it. It prevents some deadlock that I am getting when a couple of applications interact and one of them closes the device and later re-opens it. -- --- src/pcm/pcm_dmix.c.orig 2007-10-17 20:19:47.000000000 -0400 +++ src/pcm/pcm_dmix.c 2007-10-17 20:50:46.000000000 -0400 @@ -652,8 +652,11 @@ if (dmix->client) snd_pcm_direct_client_discard(dmix); shm_sum_discard(dmix); - if (snd_pcm_direct_shm_discard(dmix)) - snd_pcm_direct_semaphore_discard(dmix); + if (snd_pcm_direct_shm_discard(dmix)) { + if (snd_pcm_direct_semaphore_discard(dmix)) { + snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); + } + } else snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); free(dmix->bindings);

2 1

[alsa-devel] [patch] Fix bug with bt87x driver's load_all option
by Trent Piepho 18 Oct '07

18 Oct '07

snd-bt87x: Make the load_all option work correctly If the load_all option was turned on all cards would be treated as unknown, even those which are in the database. Of course, if the card is in the database there is no reason to use the load_all option. It's there to force loading when the card isn't in the database. But there are out of date wikis that say to do this and some distros might turn this option on by default. So, we keep the load_all option from turning known cards into unknown cards. Signed-off-by: Trent Piepho <xyzzy(a)speakeasy.org> diff --git a/pci/bt87x.c b/pci/bt87x.c --- a/pci/bt87x.c +++ b/pci/bt87x.c @@ -1021,7 +1021,7 @@ static int __devinit snd_bt87x_detect_ca int i; const struct pci_device_id *supported; - supported = pci_match_device(&driver, pci); + supported = pci_match_id(snd_bt87x_ids, pci); if (supported && supported->driver_data > 0) return supported->driver_data;

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[alsa-devel] [PATCH] ALSA - protect Dreamcast PCM driver (AICA) from G2 bus effects
by Adrian McMenamin 18 Oct '07

18 Oct '07

The G2 bus on the SEGA Dreamcast connects both the maple peripheral bus and the AICA sound memory. DMA requests on one can cause the other to timeout on memory operations. This patch prevents maple interrupts from causing hiccoughs in the AICA sound (maple bus code will land in 2.6.24). There are other cleanups for this (AICA) code - but this is in effect a regression fix rather than a cleanup. Signed-off-by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 7397865..407981d 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -106,11 +106,14 @@ static void spu_write_wait(void) static void spu_memset(u32 toi, u32 what, int length) { int i; + unsigned long flags; snd_assert(length % 4 == 0, return); for (i = 0; i < length; i++) { if (!(i % 8)) spu_write_wait(); + local_irq_save(flags); writel(what, toi + SPU_MEMORY_BASE); + local_irq_restore(flags); toi++; } } @@ -118,6 +121,7 @@ static void spu_memset(u32 toi, u32 what, int length) /* spu_memload - write to SPU address space */ static void spu_memload(u32 toi, void *from, int length) { + unsigned long flags; u32 *froml = from; u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi); int i; @@ -128,7 +132,9 @@ static void spu_memload(u32 toi, void *from, int length) if (!(i % 8)) spu_write_wait(); val = *froml; + local_irq_save(flags); writel(val, to); + local_irq_restore(flags); froml++; to++; } @@ -138,28 +144,36 @@ static void spu_memload(u32 toi, void *from, int length) static void spu_disable(void) { int i; + unsigned long flags; u32 regval; spu_write_wait(); regval = readl(ARM_RESET_REGISTER); regval |= 1; spu_write_wait(); + local_irq_save(flags); writel(regval, ARM_RESET_REGISTER); + local_irq_restore(flags); for (i = 0; i < 64; i++) { spu_write_wait(); regval = readl(SPU_REGISTER_BASE + (i * 0x80)); regval = (regval & ~0x4000) | 0x8000; spu_write_wait(); + local_irq_save(flags); writel(regval, SPU_REGISTER_BASE + (i * 0x80)); + local_irq_restore(flags); } } /* spu_enable - set spu registers to enable sound output */ static void spu_enable(void) { + unsigned long flags; u32 regval = readl(ARM_RESET_REGISTER); regval &= ~1; spu_write_wait(); + local_irq_save(flags); writel(regval, ARM_RESET_REGISTER); + local_irq_restore(flags); } /* @@ -168,25 +182,34 @@ static void spu_enable(void) */ static void spu_reset(void) { + unsigned long flags; spu_disable(); spu_memset(0, 0, 0x200000 / 4); /* Put ARM7 in endless loop */ + local_irq_save(flags); ctrl_outl(0xea000002, SPU_MEMORY_BASE); + local_irq_restore(flags); spu_enable(); } /* aica_chn_start - write to spu to start playback */ static void aica_chn_start(void) { + unsigned long flags; spu_write_wait(); + local_irq_save(flags); writel(AICA_CMD_KICK | AICA_CMD_START, (u32 *) AICA_CONTROL_POINT); + local_irq_restore(flags); } /* aica_chn_halt - write to spu to halt playback */ static void aica_chn_halt(void) { + unsigned long flags; spu_write_wait(); + local_irq_save(flags); writel(AICA_CMD_KICK | AICA_CMD_STOP, (u32 *) AICA_CONTROL_POINT); + local_irq_restore(flags); } /* ALSA code below */ @@ -213,12 +236,13 @@ static int aica_dma_transfer(int channels, int buffer_size, int q, err, period_offset; struct snd_card_aica *dreamcastcard; struct snd_pcm_runtime *runtime; - err = 0; + unsigned long flags; dreamcastcard = substream->pcm->private_data; period_offset = dreamcastcard->clicks; period_offset %= (AICA_PERIOD_NUMBER / channels); runtime = substream->runtime; for (q = 0; q < channels; q++) { + local_irq_save(flags); err = dma_xfer(AICA_DMA_CHANNEL, (unsigned long) (runtime->dma_area + (AICA_BUFFER_SIZE * q) / @@ -228,9 +252,12 @@ static int aica_dma_transfer(int channels, int buffer_size, AICA_CHANNEL0_OFFSET + q * CHANNEL_OFFSET + AICA_PERIOD_SIZE * period_offset, buffer_size / channels, AICA_DMA_MODE); - if (unlikely(err < 0)) + if (unlikely(err < 0)) { + local_irq_restore(flags); break; + } dma_wait_for_completion(AICA_DMA_CHANNEL); + local_irq_restore(flags); } return err; }

2 1

[alsa-devel] RME HDSP driver enhancements for HDSPe cards
by Ptitjes 18 Oct '07

18 Oct '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, First, sorry for the cross-post, but as i did not know where to start, i wanted to ask to both lists. I bought a RME HDSPe PCI card. I should receive it in the following days. My plan is to: 1) Determine whether the current driver works as is with the HDSPe cards and make my report to both lists. 2) Enhance the current driver in order to take profit of the new feature (32bit float format) However, i am new to Alsa, Jack and HDSP driver internal code structure. So i wanted to collect informations and hints as where to start from and be the most efficient in starting the work as i can. Could you please give me links as to: - - General Alsa driver structure and data flow - - Work done by Alsa HDSP driver and work done by Jack (I ever thought most of the work is done by Jack as my current HDSP PCCard do not work with Alsa only) - - Parts of the Alsa HDSP driver that would a priori need changes. - - Parts of Jack that would a priori need changes. Thank you very much for your help. I hope i did not bother everyone with my newbie questions. Best regards, Didier. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHFdRLVE1mALQL3JsRAvp1AKCV7L7DdX1doenajmXFX8vnYkbbqQCgkbuS GZgsAKdtGyiwfP4vNYYdZb4= =vvgU -----END PGP SIGNATURE-----

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[alsa-devel] 1.0.15rc3 patch_analog.s bug + fix
by Peter Skensved 17 Oct '07

17 Oct '07

Hi, I've come across a problem with the 1.0.15rc3 alsa driver on my Lenovo 3000 N100 laptop using the auto mute feature - it appears to work backwards ! Plugging in the headphones unmutes the external speakers and unplugging the heaphones mutes them ... Changing the if statement in line ~600 in patch_analog.c in ad1986a_update_hp to if (!spec->jack_preset) makes it work the intended way. ( spec->jack_present changes state _before_ the call to ad1986a_hp_automute ) peter ---- Peter Skensved Email : peter(a)SNO.Phy.QueensU.CA Dept. of Physics, Queen's University, Kingston, Ontario, Canada

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[alsa-devel] Recent ALSA version needs special configuration for Terratec Phase 26 USB
by Janne Himanka 17 Oct '07

17 Oct '07

Hello, I run Debian unstable on my Athlon XP1800 machine, and updating libasound2 package from version 1.0.13-2 to 1.0.14 (and later) made my Terratec Phase 26 USB soundcard useless. Whenever I tried to play something, the program would crash with a segmentation fault. When I downgraded to 1.0.13, everything would work again. It took a lot of head scratching and experimentation to find out that adding cards.USB-Audio.pcm.use_dmix."USB2448" no to my .asoundrc (it is otherwise empty) is needed for it to work. After adding that, it seems to work as it did before. The string in hyphens varies according to the mode one selects from the Terratec front panel. There are four modes with different word length and bitrates. The Phase 26 is adaptive; it doesn't need converting for a specific format. Bitrates from 32 KHz to 96 KHz and word lengths from 16 to 32 bits just work. I'm writing you this because a comment in USB-Audio.conf says: # If your device requires such a definition to work correctly, please report it # to <alsa-devel(a)alsa-project.org>. If you need any additional info, I'll be glad to help. Regards, Janne Himanka

2 1

[alsa-devel] Asus Eeepc missing in alsa models list?
by Herton Ronaldo Krzesinski 17 Oct '07

17 Oct '07

Looking at the change 291702f017efdfe556cb87b8530eb7d1ff08cbae, may be the addition of the Asus Eeepc model was missed? The following patch fixes it if it was not intended originally: diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 4b48c2e..e2a1985 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -843,6 +843,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-6ch-dig 3-stack (6-channel) with SPDIF 6stack-dig 6-stack with SPDIF lenovo-101e Lenovo laptop + eeepc-p701 ASUS Eeepc auto auto-config reading BIOS (default) ALC882/885 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 53b0428..2ceaf4c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12058,6 +12058,7 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC662_3ST_6ch] = "3stack-6ch", [ALC662_5ST_DIG] = "6stack-dig", [ALC662_LENOVO_101E] = "lenovo-101e", + [ALC662_ASUS_EEEPC_P701] = "eeepc-p701", [ALC662_AUTO] = "auto", }; -- []'s Herton

2 1

Re: [alsa-devel] hda-intel: no soundcard with current linus' git tree
by Maxim Levitsky 17 Oct '07

17 Oct '07

On Wednesday 17 October 2007 00:18:42 Thomas Meyer wrote: > $ dmesg > > [schnipp] > > ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 21 > PCI: Enabling bus mastering for device 0000:00:1b.0 > PCI: Setting latency timer of device 0000:00:1b.0 to 64 > hda_codec: STAC922x, Apple subsys_id=106b0200 > ACPI: PCI interrupt for device 0000:00:1b.0 disabled > HDA Intel: probe of 0000:00:1b.0 failed with error -16 > > [schnipp] > > > - > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ > Please bisect this (the alsa merge happened just short time ago) Which kernel did work? Best regards, Maxim Levitsky

1 0

[alsa-devel] About the buffer_time and period_time
by Fengwei Yin 17 Oct '07

17 Oct '07

Hi all, I met a problem about period_time/buffer_time on PXA with ASOC enabled. The kernel version is 2.6.21. The Audio stream is 32k which is not supported by PXA ASOC driver. Please refer following RATES definition for PXA2xx. #define PXA2XX_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) So I set plughw as default ALSA device enable auto samplerate conversion. The code to set the period_time/buffer_time is as following: err = snd_pcm_hw_params_set_rate_near(hPCM, params, &rate, 0); if (err < 0) { error("Rate not available: %d\n", rate); return err; } err = snd_pcm_hw_params_get_buffer_time_max(params, &buffer_time, 0); if (buffer_time > 500000) buffer_time = 500000; buffer_time = 400000; period_time = buffer_time/4; err = snd_pcm_hw_params_set_buffer_time_near(hPCM, params, &buffer_time, 0); if (err < 0) { error("Buffer time not available: %d, error number: %d\n", buffer_time, err); return err; } err = snd_pcm_hw_params_set_period_time_near(hPCM, params, &period_time, 0); if (err < 0) { error("Period time not available: %d\n", period_time); return err; } If I set buffer_time to 400000, I will get: " Buffer time not available: 400000, error number: -22 Cannot init audio driver. Floating point exception " If I set buffer time to 500000, buffer_time can be set correctly. But audio playback failed. If I set period_time before setting buffer_time, Everything is OK. Any thought about this problem. Thanks. Regards Yin, Fengwei

2 2

[alsa-devel] Intel HDA digital out distortion
by pr0t0cod7 16 Oct '07

16 Oct '07

Hi, On Monday 23 July 2007 23:09:24 Ingmar Stieger wrote: > 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 have the same issues. Any sound is distorted through spdif output (seems more pronounced with low frequencies.) Sound is not distorted: - using passthough (playing AC3) - when volume is set to the very minimal (no perceptible distortion) - with analog output My motherboard: ASUS P5B deluxe with integrated soundchip Kernel: 2.6.22 Alsa: Intel HD audio kernel built module asound.conf: http://pastebin.com/f2c54b47b > 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. The patch worked for me, no more distortion :) Thanks for sharing, Alex.

2 2

[alsa-devel] fsc esprimo v5505 (snd_intel_hda) - output in /var/log/messages
by Jan-Simon Möller 16 Oct '07

16 Oct '07

Hi! What do these messages mean ? Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_intel.c:1185: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x0, channel=0, format=0x0 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x0, channel=0, format=0x0 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_intel.c:1185: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x0, channel=0, format=0x0 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x0, channel=0, format=0x0 Thanks ! Jan

2 1

Re: [alsa-devel] [Alsa-user] How to get set/fixed sample rate of ALSA device?
by Rene Herman 16 Oct '07

16 Oct '07

On 10/07/2007 07:04 PM, Peteris Krisjanis wrote: >> "Current sample rate" only has meaning with the device open , and if I >> understand you right, a "cat >> /proc/asound/card0/pcm0{p,c}/sub0/hw_params will then get you what you >> want as to the "get" part (substitute for 0 in the above as required). > > I actually meant what kind of sample rate is set in sound card. For > example, my onboard sound card has fixed Hz. I would like to know such > information _before_ I touch sound card :) Okay, you want the information that the driver author supplied in the struct snd_pcm_hardware for the stream. I've in fact also wanted that information on a number of occasions and just looked at the driver source then. > So you suggest to open device via any means and then read that file? I > did this and it works. > I guess I can even try to implement it in app and check for sample > rate everytime before I play. > > But is this only solution? It's rather hack, and I would like to have > more reliable method. You might consider this less of of a hack -- it uses the ALSA API to query the information from the PCM handle directly. Please see the library docoumentation for other information you can extract from a hw_params struct. (I don't suppose you can ever get anything but dir == 0 for "hw" devices). Introducing a /proc/asound/card0/pcm0{p,c}/hw_limits or similar might not be a bad idea though? Rene /* gcc -W -Wall -o snd_rate snd_rate.c -l asound */ #include <stdio.h> #include <alsa/asoundlib.h> int main(void) { snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; int err; snd_pcm_hw_params_alloca(&params); err = snd_pcm_open(&handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0); if (err < 0) { fprintf(stderr, "snd_pcm_open: %s\n", snd_strerror(err)); return -1; } err = snd_pcm_hw_params_any(handle, params); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_any: %s\n", snd_strerror(err)); return -1; } err = snd_pcm_hw_params_get_rate_min(params, &val, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_min: %s\n", snd_strerror(err)); return -1; } printf("min: (%c) %u\n", "<=>"[dir + 1], val); err = snd_pcm_hw_params_get_rate_max(params, &val, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_max: %s\n", snd_strerror(err)); return -1; } printf("max: (%c) %u\n", "<=>"[dir + 1], val); snd_pcm_close(handle); return 0; }

4 7

[alsa-devel] intel-hda patch for Dell Latitude D531
by glogow＠fbihome.de 16 Oct '07

16 Oct '07

Hi I have attached three files to this mail: 1. patch_sigmatel.c.diff I have a Latitude D531 which has a ATI SB600 with a STAC 9205. The chip is detected, but no config was read with the latest drivers. The problems are the dmic_nids arrays, which are ignored for auto config lookup. This calles is_in_nid_list, which expects a 0 terminated list, but these arrays are not. Additionally I updated the dell_9205_m43_pin_configs with values from the latest win drivers. 2. create_dell_tables.pl When looking into the ini files and comparing the nids and config values of the various tables, I got really confused, so I wrote a litte script to extract the values and build the table. This is the extended version, after I got my sound working. It extracts all pin infos from all ini files, generates the apropriate tables and prepends the Version.txt from the driver, resultiung in a kind of C header. Maybe it's usefull for you. 3. patch_sigmatel_dell.h Output from running create_dell_tables.pl on the latest drivers for my D531 from Dell. HTH Jan-Marek Glogowski ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

3 3

[alsa-devel] ALSA 1.0.15 final
by Jaroslav Kysela 16 Oct '07

16 Oct '07

Hello all, delayed but ... all ALSA packages 1.0.15 final are available for download on http://www.alsa-project.org . The changelog is available on web for 1.0.14 -> 1.0.15. I will make also available 1.0.15rc3 -> 1.0.15 changelog in next few hours. Have fun, Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project

1 0

[alsa-devel] How to detect 'dead air' while recording?
by william estrada 15 Oct '07

15 Oct '07

Hi group, I want to filter out 'dead air' while recording. What is the best way to do this? I don't want to use JACK, just ALSA. Any URLs? -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

1 0

[alsa-devel] LGPL Licensing
by WOODEN John 15 Oct '07

15 Oct '07

I wish to use either the ALSA or SALSA API within my application. I also intend to commercially distribute my application and sell it. What affect does ALSA's and/or SALSA's Lesser GPL license have upon applications that utilize their API's? Do I have to make the source code of my application freely available if I choose to utilize the ALSA or SALSA API? Thanks, John. IMPORTANT NOTICE: This e-mail and any attachment to it are intended only to be read or used by the named addressee. It is confidential and may contain legally privileged information. No confidentiality or privilege is waived or lost by any mistaken transmission to you. The RTA is not responsible for any unauthorised alterations to this e-mail or attachment to it. Views expressed in this message are those of the individual sender, and are not necessarily the views of the RTA. If you receive this e-mail in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or use any part of this e-mail if you are not the intended recipient.

2 1

Re: [alsa-devel] [Pkg-alsa-devel] Bug#429326: Bug#437827: libasound2: Sound creaks badly when playing DVDs or OGG files
by Elimar Riesebieter 15 Oct '07

15 Oct '07

forwarded 429326 alsa-devel(a)alsa-project.org thanks On Sun, 14 Oct 2007 the mental interface of Loïc Minier told: > On Sun, Oct 14, 2007, Elimar Riesebieter wrote: > > From my point of view it is not reasonable to patch Debian packages, > > which of course will break 32bit only sound cards. So we've to find > > a way to force a solution directly on alsa-devel. Who will forward > > the _Debian_bug to alsa-devel(a)alsa-project.org ? > > I don't understand your remark about patching the Debian package? Why > would reverting the bit shift macros break anything? > > The attached patch for alsa-lib should be enough to fix this, upstream > should revert the broken change in the bit shifts. Yeah, we can provide it. But shouldn't it be a solution for all ALSA users? This is Debian #429326. As well reported as https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3360. We have to look for a general solution as the distributors don't need to patch sources. > Alsa #3360; GNOME #436192; LP #116990; Mandriva #33908; Revert HG r2264 > (changeset 23c4c0f5de40) as the bit shifts overflow when the volume is high > for 32 bits to 16 bits resampling > > diff -r 23c4c0f5de40 -r 883ae69471d2 src/pcm/plugin_ops.h > --- a/src/pcm/plugin_ops.h Fri Mar 09 17:06:23 2007 +0100 > +++ b/src/pcm/plugin_ops.h Fri Mar 09 15:30:38 2007 +0100 > @@ -85,8 +85,6 @@ static inline u_int32_t sx24s(u_int32_t > #define _put_triple(ptr,val) _put_triple_be(ptr,val) > #define _put_triple_s(ptr,val) _put_triple_le(ptr,val) > #endif > - > -#define shift_down(val, bits) (((val) + (1 << ((bits) - 1))) >> (bits)) > > #ifdef COPY_LABELS > static void *copy_labels[5] = { > @@ -258,14 +256,14 @@ conv_xxx1_009x: as_u32(dst) = sx24s((u_i > conv_xxx1_009x: as_u32(dst) = sx24s((u_int32_t)(as_u8c(src) ^ 0x80) << 8); goto CONV_END; > conv_xxx1_9000: as_u32(dst) = (u_int32_t)(as_u8c(src) ^ 0x80) << 24; goto CONV_END; > conv_xxx1_0009: as_u32(dst) = (u_int32_t)(as_u8c(src) ^ 0x80); goto CONV_END; > -conv_xx12_xxx1: as_u8(dst) = shift_down(as_u16c(src), 8); goto CONV_END; > +conv_xx12_xxx1: as_u8(dst) = as_u16c(src) >> 8; goto CONV_END; > conv_xx12_xx12: as_u16(dst) = as_u16c(src); goto CONV_END; > conv_xx12_xx21: as_u16(dst) = bswap_16(as_u16c(src)); goto CONV_END; > conv_xx12_x120: as_u32(dst) = sx24((u_int32_t)as_u16c(src) << 8); goto CONV_END; > conv_xx12_021x: as_u32(dst) = sx24s((u_int32_t)bswap_16(as_u16c(src)) << 8); goto CONV_END; > conv_xx12_1200: as_u32(dst) = (u_int32_t)as_u16c(src) << 16; goto CONV_END; > conv_xx12_0021: as_u32(dst) = (u_int32_t)bswap_16(as_u16c(src)); goto CONV_END; > -conv_xx12_xxx9: as_u8(dst) = shift_down(as_u16c(src), 8) ^ 0x80; goto CONV_END; > +conv_xx12_xxx9: as_u8(dst) = (as_u16c(src) >> 8) ^ 0x80; goto CONV_END; > conv_xx12_xx92: as_u16(dst) = as_u16c(src) ^ 0x8000; goto CONV_END; > conv_xx12_xx29: as_u16(dst) = bswap_16(as_u16c(src)) ^ 0x80; goto CONV_END; > conv_xx12_x920: as_u32(dst) = sx24((u_int32_t)(as_u16c(src) ^ 0x8000) << 8); goto CONV_END; > @@ -284,57 +282,57 @@ conv_xx12_01Ax: as_u32(dst) = sx24s((u_i > conv_xx12_01Ax: as_u32(dst) = sx24s((u_int32_t)(as_u16c(src) ^ 0x80) << 8); goto CONV_END; > conv_xx12_A100: as_u32(dst) = (u_int32_t)bswap_16(as_u16c(src) ^ 0x80) << 16; goto CONV_END; > conv_xx12_001A: as_u32(dst) = (u_int32_t)(as_u16c(src) ^ 0x80); goto CONV_END; > -conv_x123_xxx1: as_u8(dst) = shift_down(as_u32c(src), 16); goto CONV_END; > -conv_x123_xx12: as_u16(dst) = shift_down(as_u32c(src), 8); goto CONV_END; > -conv_x123_xx21: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 8)); goto CONV_END; > +conv_x123_xxx1: as_u8(dst) = as_u32c(src) >> 16; goto CONV_END; > +conv_x123_xx12: as_u16(dst) = as_u32c(src) >> 8; goto CONV_END; > +conv_x123_xx21: as_u16(dst) = bswap_16(as_u32c(src) >> 8); goto CONV_END; > conv_x123_x123: as_u32(dst) = sx24(as_u32c(src)); goto CONV_END; > conv_x123_321x: as_u32(dst) = sx24s(bswap_32(as_u32c(src))); goto CONV_END; > conv_x123_1230: as_u32(dst) = as_u32c(src) << 8; goto CONV_END; > -conv_x123_0321: as_u32(dst) = shift_down(bswap_32(as_u32c(src)), 8); goto CONV_END; > - conv_x123_xxx9: as_u8(dst) = shift_down(as_u32c(src), 16) ^ 0x80; goto CONV_END; > -conv_x123_xx92: as_u16(dst) = shift_down(as_u32c(src), 8) ^ 0x8000; goto CONV_END; > -conv_x123_xx29: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 8)) ^ 0x80; goto CONV_END; > +conv_x123_0321: as_u32(dst) = bswap_32(as_u32c(src)) >> 8; goto CONV_END; > +conv_x123_xxx9: as_u8(dst) = (as_u32c(src) >> 16) ^ 0x80; goto CONV_END; > +conv_x123_xx92: as_u16(dst) = (as_u32c(src) >> 8) ^ 0x8000; goto CONV_END; > +conv_x123_xx29: as_u16(dst) = bswap_16(as_u32c(src) >> 8) ^ 0x80; goto CONV_END; > conv_x123_x923: as_u32(dst) = sx24(as_u32c(src) ^ 0x800000); goto CONV_END; > conv_x123_329x: as_u32(dst) = sx24s(bswap_32(as_u32c(src)) ^ 0x8000); goto CONV_END; > conv_x123_9230: as_u32(dst) = (as_u32c(src) ^ 0x800000) << 8; goto CONV_END; > -conv_x123_0329: as_u32(dst) = shift_down(bswap_32(as_u32c(src)), 8) ^ 0x80; goto CONV_END; > -conv_123x_xxx3: as_u8(dst) = shift_down(as_u32c(src), 8) & 0xff; goto CONV_END; > -conv_123x_xx32: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 8)); goto CONV_END; > -conv_123x_xx23: as_u16(dst) = shift_down(as_u32c(src), 8) & 0xffff; goto CONV_END; > +conv_x123_0329: as_u32(dst) = (bswap_32(as_u32c(src)) >> 8) ^ 0x80; goto CONV_END; > +conv_123x_xxx3: as_u8(dst) = (as_u32c(src) >> 8) & 0xff; goto CONV_END; > +conv_123x_xx32: as_u16(dst) = bswap_16(as_u32c(src) >> 8); goto CONV_END; > +conv_123x_xx23: as_u16(dst) = (as_u32c(src) >> 8) & 0xffff; goto CONV_END; > conv_123x_x321: as_u32(dst) = sx24(bswap_32(as_u32c(src))); goto CONV_END; > conv_123x_123x: as_u32(dst) = sx24s(as_u32c(src)); goto CONV_END; > conv_123x_3210: as_u32(dst) = bswap_32(as_u32c(src)) << 8; goto CONV_END; > -conv_123x_0123: as_u32(dst) = shift_down(as_u32c(src), 8); goto CONV_END; > -conv_123x_xxxB: as_u8(dst) = (shift_down(as_u32c(src), 8) & 0xff) ^ 0x80; goto CONV_END; > -conv_123x_xxB2: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 8) ^ 0x80); goto CONV_END; > -conv_123x_xx2B: as_u16(dst) = (shift_down(as_u32c(src), 8) & 0xffff) ^ 0x80; goto CONV_END; > +conv_123x_0123: as_u32(dst) = as_u32c(src) >> 8; goto CONV_END; > +conv_123x_xxxB: as_u8(dst) = ((as_u32c(src) >> 8) & 0xff) ^ 0x80; goto CONV_END; > +conv_123x_xxB2: as_u16(dst) = bswap_16((as_u32c(src) >> 8) ^ 0x80); goto CONV_END; > +conv_123x_xx2B: as_u16(dst) = ((as_u32c(src) >> 8) & 0xffff) ^ 0x80; goto CONV_END; > conv_123x_xB21: as_u32(dst) = sx24(bswap_32(as_u32c(src)) ^ 0x800000); goto CONV_END; > conv_123x_12Bx: as_u32(dst) = sx24s(as_u32c(src) ^ 0x8000); goto CONV_END; > conv_123x_B210: as_u32(dst) = bswap_32(as_u32c(src) ^ 0x8000) << 8; goto CONV_END; > -conv_123x_012B: as_u32(dst) = shift_down(as_u32c(src), 8) ^ 0x80; goto CONV_END; > -conv_1234_xxx1: as_u8(dst) = shift_down(as_u32c(src), 24); goto CONV_END; > -conv_1234_xx12: as_u16(dst) = shift_down(as_u32c(src), 16); goto CONV_END; > -conv_1234_xx21: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 16)); goto CONV_END; > -conv_1234_x123: as_u32(dst) = sx24(shift_down(as_u32c(src), 8)); goto CONV_END; > +conv_123x_012B: as_u32(dst) = (as_u32c(src) >> 8) ^ 0x80; goto CONV_END; > +conv_1234_xxx1: as_u8(dst) = as_u32c(src) >> 24; goto CONV_END; > +conv_1234_xx12: as_u16(dst) = as_u32c(src) >> 16; goto CONV_END; > +conv_1234_xx21: as_u16(dst) = bswap_16(as_u32c(src) >> 16); goto CONV_END; > +conv_1234_x123: as_u32(dst) = sx24(as_u32c(src) >> 8); goto CONV_END; > conv_1234_321x: as_u32(dst) = sx24s(bswap_32(as_u32c(src)) << 8); goto CONV_END; > conv_1234_1234: as_u32(dst) = as_u32c(src); goto CONV_END; > conv_1234_4321: as_u32(dst) = bswap_32(as_u32c(src)); goto CONV_END; > -conv_1234_xxx9: as_u8(dst) = shift_down(as_u32c(src), 24) ^ 0x80; goto CONV_END; > -conv_1234_xx92: as_u16(dst) = shift_down(as_u32c(src), 16) ^ 0x8000; goto CONV_END; > -conv_1234_xx29: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 16)) ^ 0x80; goto CONV_END; > -conv_1234_x923: as_u32(dst) = sx24(shift_down(as_u32c(src), 8) ^ 0x800000); goto CONV_END; > +conv_1234_xxx9: as_u8(dst) = (as_u32c(src) >> 24) ^ 0x80; goto CONV_END; > +conv_1234_xx92: as_u16(dst) = (as_u32c(src) >> 16) ^ 0x8000; goto CONV_END; > +conv_1234_xx29: as_u16(dst) = bswap_16(as_u32c(src) >> 16) ^ 0x80; goto CONV_END; > +conv_1234_x923: as_u32(dst) = sx24((as_u32c(src) >> 8) ^ 0x800000); goto CONV_END; > conv_1234_329x: as_u32(dst) = sx24s((bswap_32(as_u32c(src)) ^ 0x80) << 8); goto CONV_END; > conv_1234_9234: as_u32(dst) = as_u32c(src) ^ 0x80000000; goto CONV_END; > conv_1234_4329: as_u32(dst) = bswap_32(as_u32c(src)) ^ 0x80; goto CONV_END; > conv_1234_xxx4: as_u8(dst) = as_u32c(src) & 0xff; goto CONV_END; > conv_1234_xx43: as_u16(dst) = bswap_16(as_u32c(src)); goto CONV_END; > conv_1234_xx34: as_u16(dst) = as_u32c(src) & 0xffff; goto CONV_END; > -conv_1234_x432: as_u32(dst) = sx24(shift_down(bswap_32(as_u32c(src)), 8)); goto CONV_END; > +conv_1234_x432: as_u32(dst) = sx24(bswap_32(as_u32c(src)) >> 8); goto CONV_END; > conv_1234_234x: as_u32(dst) = sx24s(as_u32c(src) << 8); goto CONV_END; > conv_1234_xxxC: as_u8(dst) = (as_u32c(src) & 0xff) ^ 0x80; goto CONV_END; > conv_1234_xxC3: as_u16(dst) = bswap_16(as_u32c(src) ^ 0x80); goto CONV_END; > conv_1234_xx3C: as_u16(dst) = (as_u32c(src) & 0xffff) ^ 0x80; goto CONV_END; > -conv_1234_xC32: as_u32(dst) = sx24(shift_down(bswap_32(as_u32c(src)), 8) ^ 0x800000); goto CONV_END; > +conv_1234_xC32: as_u32(dst) = sx24((bswap_32(as_u32c(src)) >> 8) ^ 0x800000); goto CONV_END; > conv_1234_23Cx: as_u32(dst) = sx24s((as_u32c(src) ^ 0x80) << 8); goto CONV_END; > conv_1234_C321: as_u32(dst) = bswap_32(as_u32c(src) ^ 0x80); goto CONV_END; > conv_1234_123C: as_u32(dst) = as_u32c(src) ^ 0x80; goto CONV_END; > @@ -384,26 +382,26 @@ get16_12_92: sample = as_u16c(src) ^ 0x8 > get16_12_92: sample = as_u16c(src) ^ 0x8000; goto GET16_END; > get16_12_21: sample = bswap_16(as_u16c(src)); goto GET16_END; > get16_12_A1: sample = bswap_16(as_u16c(src) ^ 0x80); goto GET16_END; > -get16_0123_12: sample = shift_down(as_u32c(src), 8); goto GET16_END; > -get16_0123_92: sample = shift_down(as_u32c(src), 8) ^ 0x8000; goto GET16_END; > -get16_1230_32: sample = bswap_16(shift_down(as_u32c(src), 8)); goto GET16_END; > -get16_1230_B2: sample = bswap_16(shift_down(as_u32c(src), 8) ^ 0x80); goto GET16_END; > -get16_1234_12: sample = shift_down(as_u32c(src), 16); goto GET16_END; > -get16_1234_92: sample = shift_down(as_u32c(src), 16) ^ 0x8000; goto GET16_END; > +get16_0123_12: sample = as_u32c(src) >> 8; goto GET16_END; > +get16_0123_92: sample = (as_u32c(src) >> 8) ^ 0x8000; goto GET16_END; > +get16_1230_32: sample = bswap_16(as_u32c(src) >> 8); goto GET16_END; > +get16_1230_B2: sample = bswap_16((as_u32c(src) >> 8) ^ 0x80); goto GET16_END; > +get16_1234_12: sample = as_u32c(src) >> 16; goto GET16_END; > +get16_1234_92: sample = (as_u32c(src) >> 16) ^ 0x8000; goto GET16_END; > get16_1234_43: sample = bswap_16(as_u32c(src)); goto GET16_END; > get16_1234_C3: sample = bswap_16(as_u32c(src) ^ 0x80); goto GET16_END; > -get16_123_12: sample = shift_down(_get_triple(src), 8); goto GET16_END; > -get16_123_92: sample = shift_down(_get_triple(src), 8) ^ 0x8000; goto GET16_END; > -get16_123_32: sample = shift_down(_get_triple_s(src), 8); goto GET16_END; > -get16_123_B2: sample = shift_down(_get_triple_s(src), 8) ^ 0x8000; goto GET16_END; > -get16_123_12_20: sample = shift_down(_get_triple(src), 4); goto GET16_END; > -get16_123_92_20: sample = shift_down(_get_triple(src), 4) ^ 0x8000; goto GET16_END; > -get16_123_32_20: sample = shift_down(_get_triple_s(src), 4); goto GET16_END; > -get16_123_B2_20: sample = shift_down(_get_triple_s(src), 4) ^ 0x8000; goto GET16_END; > -get16_123_12_18: sample = shift_down(_get_triple(src), 2); goto GET16_END; > -get16_123_92_18: sample = shift_down(_get_triple(src), 2) ^ 0x8000; goto GET16_END; > -get16_123_32_18: sample = shift_down(_get_triple_s(src), 2); goto GET16_END; > -get16_123_B2_18: sample = shift_down(_get_triple_s(src), 2) ^ 0x8000; goto GET16_END; > +get16_123_12: sample = _get_triple(src) >> 8; goto GET16_END; > +get16_123_92: sample = (_get_triple(src) >> 8) ^ 0x8000; goto GET16_END; > +get16_123_32: sample = _get_triple_s(src) >> 8; goto GET16_END; > +get16_123_B2: sample = (_get_triple_s(src) >> 8) ^ 0x8000; goto GET16_END; > +get16_123_12_20: sample = _get_triple(src) >> 4; goto GET16_END; > +get16_123_92_20: sample = (_get_triple(src) >> 4) ^ 0x8000; goto GET16_END; > +get16_123_32_20: sample = _get_triple_s(src) >> 4; goto GET16_END; > +get16_123_B2_20: sample = (_get_triple_s(src) >> 4) ^ 0x8000; goto GET16_END; > +get16_123_12_18: sample = _get_triple(src) >> 2; goto GET16_END; > +get16_123_92_18: sample = (_get_triple(src) >> 2) ^ 0x8000; goto GET16_END; > +get16_123_32_18: sample = _get_triple_s(src) >> 2; goto GET16_END; > +get16_123_B2_18: sample = (_get_triple_s(src) >> 2) ^ 0x8000; goto GET16_END; > } > #endif > > @@ -431,8 +429,8 @@ static void *put16_labels[4 * 2 * 2] = { > > #ifdef PUT16_END > while (0) { > -put16_12_1: as_u8(dst) = shift_down(sample, 8); goto PUT16_END; > -put16_12_9: as_u8(dst) = shift_down(sample, 8) ^ 0x80; goto PUT16_END; > +put16_12_1: as_u8(dst) = sample >> 8; goto PUT16_END; > +put16_12_9: as_u8(dst) = (sample >> 8) ^ 0x80; goto PUT16_END; > put16_12_12: as_u16(dst) = sample; goto PUT16_END; > put16_12_92: as_u16(dst) = sample ^ 0x8000; goto PUT16_END; > put16_12_21: as_u16(dst) = bswap_16(sample); goto PUT16_END; > @@ -502,8 +500,8 @@ get32_12_A100: sample = (u_int32_t)bswap > get32_12_A100: sample = (u_int32_t)bswap_16(as_u16c(src) ^ 0x80) << 16; goto GET32_END; > get32_0123_1230: sample = as_u32c(src) << 8; goto GET32_END; > get32_0123_9230: sample = (as_u32c(src) << 8) ^ 0x80000000; goto GET32_END; > -get32_1230_3210: sample = bswap_32(shift_down(as_u32c(src), 8)); goto GET32_END; > -get32_1230_B210: sample = bswap_32(shift_down(as_u32c(src), 8) ^ 0x80); goto GET32_END; > +get32_1230_3210: sample = bswap_32(as_u32c(src) >> 8); goto GET32_END; > +get32_1230_B210: sample = bswap_32((as_u32c(src) >> 8) ^ 0x80); goto GET32_END; > get32_1234_1234: sample = as_u32c(src); goto GET32_END; > get32_1234_9234: sample = as_u32c(src) ^ 0x80000000; goto GET32_END; > get32_1234_4321: sample = bswap_32(as_u32c(src)); goto GET32_END; > @@ -570,32 +568,32 @@ static void *put32_labels[4 * 2 * 2 + 4 > > #ifdef PUT32_END > while (0) { > -put32_1234_1: as_u8(dst) = shift_down(sample, 24); goto PUT32_END; > -put32_1234_9: as_u8(dst) = shift_down(sample, 24) ^ 0x80; goto PUT32_END; > -put32_1234_12: as_u16(dst) = shift_down(sample, 16); goto PUT32_END; > -put32_1234_92: as_u16(dst) = shift_down(sample, 16) ^ 0x8000; goto PUT32_END; > -put32_1234_21: as_u16(dst) = bswap_16(shift_down(sample, 16)); goto PUT32_END; > -put32_1234_29: as_u16(dst) = bswap_16(shift_down(sample, 16)) ^ 0x80; goto PUT32_END; > -put32_1234_0123: as_u32(dst) = sx24(shift_down(sample, 8)); goto PUT32_END; > -put32_1234_0923: as_u32(dst) = sx24(shift_down(sample, 8) ^ 0x800000); goto PUT32_END; > +put32_1234_1: as_u8(dst) = sample >> 24; goto PUT32_END; > +put32_1234_9: as_u8(dst) = (sample >> 24) ^ 0x80; goto PUT32_END; > +put32_1234_12: as_u16(dst) = sample >> 16; goto PUT32_END; > +put32_1234_92: as_u16(dst) = (sample >> 16) ^ 0x8000; goto PUT32_END; > +put32_1234_21: as_u16(dst) = bswap_16(sample >> 16); goto PUT32_END; > +put32_1234_29: as_u16(dst) = bswap_16(sample >> 16) ^ 0x80; goto PUT32_END; > +put32_1234_0123: as_u32(dst) = sx24(sample >> 8); goto PUT32_END; > +put32_1234_0923: as_u32(dst) = sx24((sample >> 8) ^ 0x800000); goto PUT32_END; > put32_1234_3210: as_u32(dst) = sx24s(bswap_32(sample) << 8); goto PUT32_END; > put32_1234_3290: as_u32(dst) = sx24s((bswap_32(sample) ^ 0x80) << 8); goto PUT32_END; > put32_1234_1234: as_u32(dst) = sample; goto PUT32_END; > put32_1234_9234: as_u32(dst) = sample ^ 0x80000000; goto PUT32_END; > put32_1234_4321: as_u32(dst) = bswap_32(sample); goto PUT32_END; > put32_1234_4329: as_u32(dst) = bswap_32(sample) ^ 0x80; goto PUT32_END; > -put32_1234_123: _put_triple(dst, shift_down(sample, 8)); goto PUT32_END; > -put32_1234_923: _put_triple(dst, shift_down(sample ^ 0x80000000, 8)); goto PUT32_END; > -put32_1234_321: _put_triple_s(dst, shift_down(sample, 8)); goto PUT32_END; > -put32_1234_329: _put_triple_s(dst, shift_down(sample ^ 0x80000000, 8)); goto PUT32_END; > -put32_1234_123_20: _put_triple(dst, shift_down(sample, 12)); goto PUT32_END; > -put32_1234_923_20: _put_triple(dst, shift_down(sample ^ 0x80000000, 12)); goto PUT32_END; > -put32_1234_321_20: _put_triple_s(dst, shift_down(sample, 12)); goto PUT32_END; > -put32_1234_329_20: _put_triple_s(dst, shift_down(sample ^ 0x80000000, 12)); goto PUT32_END; > -put32_1234_123_18: _put_triple(dst, shift_down(sample, 14)); goto PUT32_END; > -put32_1234_923_18: _put_triple(dst, shift_down(sample ^ 0x80000000, 14)); goto PUT32_END; > -put32_1234_321_18: _put_triple_s(dst, shift_down(sample, 14)); goto PUT32_END; > -put32_1234_329_18: _put_triple_s(dst, shift_down(sample ^ 0x80000000, 14)); goto PUT32_END; > +put32_1234_123: _put_triple(dst, sample >> 8); goto PUT32_END; > +put32_1234_923: _put_triple(dst, (sample ^ 0x80000000) >> 8); goto PUT32_END; > +put32_1234_321: _put_triple_s(dst, sample >> 8); goto PUT32_END; > +put32_1234_329: _put_triple_s(dst, (sample ^ 0x80000000) >> 8); goto PUT32_END; > +put32_1234_123_20: _put_triple(dst, sample >> 12); goto PUT32_END; > +put32_1234_923_20: _put_triple(dst, (sample ^ 0x80000000) >> 12); goto PUT32_END; > +put32_1234_321_20: _put_triple_s(dst, sample >> 12); goto PUT32_END; > +put32_1234_329_20: _put_triple_s(dst, (sample ^ 0x80000000) >> 12); goto PUT32_END; > +put32_1234_123_18: _put_triple(dst, sample >> 14); goto PUT32_END; > +put32_1234_923_18: _put_triple(dst, (sample ^ 0x80000000) >> 14); goto PUT32_END; > +put32_1234_321_18: _put_triple_s(dst, sample >> 14); goto PUT32_END; > +put32_1234_329_18: _put_triple_s(dst, (sample ^ 0x80000000) >> 14); goto PUT32_END; > } > #endif > > @@ -1081,4 +1079,3 @@ norms_32_s32s: _norms(src, dst, 32, 1, > #undef _put_triple_le > #undef _put_triple_be > > -#undef shift_down -- Excellent day for drinking heavily. Spike the office water cooler;-)

2 1

[alsa-devel] intel-hda patch for sigmatel codec volume knob
by zhejiang 15 Oct '07

15 Oct '07

Hi, I found that STAC_VOLKNOB hardwired the KNOB nid to 0x24. It is okay for stac9205 and stac927x. But the VolumeKnob nid of stac9220-9221 is 0x16. diff -Nur a/alsa-driver-1.0.15rc3/alsa-kernel/pci/hda/patch_sigmatel.c b/alsa-driver-1.0.15rc3/alsa-kernel/pci/hda/patch_sigmatel.c --- a/alsa-driver-1.0.15rc3/alsa-kernel/pci/hda/patch_sigmatel.c 2007-09-21 16:41:28.000000000 +0800 +++ b/alsa-driver-1.0.15rc3/alsa-kernel/pci/hda/patch_sigmatel.c 2007-10-15 15:13:14.000000000 +0800 @@ -351,7 +351,7 @@ static int stac92xx_volknob_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = kcontrol->private_value; + ucontrol->value.integer.value[0] = kcontrol->private_value & 0xFFFF; return 0; } @@ -360,14 +360,15 @@ { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - if (kcontrol->private_value == ucontrol->value.integer.value[0]) + if ((kcontrol->private_value & 0xFFFF) == ucontrol->value.integer.value[0]) return 0; - kcontrol->private_value = ucontrol->value.integer.value[0]; + kcontrol->private_value &= 0xFFFF0000; + kcontrol->private_value += ucontrol->value.integer.value[0]; - snd_hda_codec_write_cache(codec, 0x24, 0, + snd_hda_codec_write_cache(codec, kcontrol->private_value >> 16, 0, AC_VERB_SET_VOLUME_KNOB_CONTROL, - kcontrol->private_value | 0x80); + (kcontrol->private_value & 0xFFFF) | 0x80); return 1; } @@ -433,7 +434,7 @@ .private_value = verb_read | (verb_write << 16), \ } -#define STAC_VOLKNOB \ +#define STAC_VOLKNOB(knob_nid) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = "Master Playback Volume", \ @@ -441,7 +442,7 @@ .info = stac92xx_volknob_info, \ .get = stac92xx_volknob_get, \ .put = stac92xx_volknob_put, \ - .private_value = 127, \ + .private_value = 127 | (knob_nid << 16), \ } @@ -474,7 +475,7 @@ }, STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0), - STAC_VOLKNOB, + STAC_VOLKNOB(0x24), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), @@ -490,7 +491,7 @@ /* This needs to be generated dynamically based on sequence */ static struct snd_kcontrol_new stac922x_mixer[] = { STAC_INPUT_SOURCE(2), - STAC_VOLKNOB, + STAC_VOLKNOB(0x16), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x12, 0x0, HDA_OUTPUT), @@ -504,7 +505,7 @@ static struct snd_kcontrol_new stac927x_mixer[] = { STAC_INPUT_SOURCE(3), - STAC_VOLKNOB, + STAC_VOLKNOB(0x24), STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT),

3 2

[alsa-devel] Fwd: FSC Esprimo Mobile V5505 - info, partly working
by Jan-Simon Möller 13 Oct '07

13 Oct '07

Hi ! I got a new FSC Esprimo Mobile V5505 (full system info attached). When i use "options snd-hda-intel model=fujitsu-c1410 enable=1 index=0" sound works, but it sounds not good and smooth. Chip is a conexant afaik. Attachements: * Full lspci -vv -nn Whats needed to improve sound and make the mixer available ? If any other data is needed -> please ping ! Sincerely yours Jan-Simon Möller dl9pf(a)gmx.de

2 3

[alsa-devel] Patch for Intel-HDA ALC262 on FSC Lifebook C1410
by Veli-Matti Lintu 12 Oct '07

12 Oct '07

Hello, After error and trial I got sound working on Fujitsu-Siemens Computers Lifebook C1410. Attached is a patch that works for me, but probably needs some cleaning. The laptop uses an ALC262 chip that is also used on other FSC models, but the needed settings seem to be somewhat different from the existing Fujitsu-settings in patch_realtek.c. If I understand correctly, the chips use same subsystem ids, so the patch does not automatically detect the correct chipset. Maybe someone more knowledgeable can help with this? The patch names the internal mic as Int Mic and external mic that can be plugged in as Ext Mic. There's some more background information in Ubuntu's Launchpad: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/125790 The patch is tested with kernel 2.6.22.9. I am not really familiar with alsa development, so I cannot say which version of the alsa drivers that kernel version uses. The environment where the testing was done is running Ubuntu 7.04 with alsa version 1.0.13. Only updated component is the kernel. The patch needs model=fujitsu-c1410 parameter to be passed to the module: /etc/modprobe.d/alsa-base: options snd-hda-intel model=fujitsu-c1410 The subsystem 10cf:1397 is mapped in patch_realtek.c alc262_cfg_tbl to ALC262_FUJITSU and ALC262_FUJITSU_C1410 would need to be mapped to same ids: ------------------------------------------------------------------- sound/pci/hda/patch_realtek.c: SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), ------------------------------------------------------------------- lspci -nv (on C1410) 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 10cf:1397 Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at f0640000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) ------------------------------------------------------------------- I hope someone with more knowledge about alsa drivers could have a look at this. The patch has been in internal use for a few weeks and people seem to be happy with it, so I hope also others can get their sound working. The patch and hw information follow: --------------------------------------------------------------------------------------------- --- linux-2.6.22.9/sound/pci/hda/patch_realtek.c.orig 2007-09-26 21:03:01.000000000 +0300 +++ linux-2.6.22.9/sound/pci/hda/patch_realtek.c 2007-10-07 13:28:12.000000000 +0300 @@ -89,6 +89,7 @@ ALC262_HIPPO, ALC262_HIPPO_1, ALC262_FUJITSU, + ALC262_FUJITSU_C1410, ALC262_HP_BPC, ALC262_HP_BPC_D7000_WL, ALC262_HP_BPC_D7000_WF, @@ -7068,6 +7069,15 @@ }, }; +static struct hda_input_mux alc262_fujitsu_c1410_capture_source = { + .num_items = 3, + .items = { + { "Ext Mic", 0x0 }, + { "Int Mic", 0x1 }, + { "CD", 0x4 }, + }, +}; + static struct hda_input_mux alc262_HP_capture_source = { .num_items = 5, .items = { @@ -7182,6 +7192,43 @@ { } /* end */ }; +static struct snd_kcontrol_new alc262_fujitsu_c1410_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = alc262_fujitsu_master_vol_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, + .private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc262_fujitsu_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT), + }, + + HDA_CODEC_VOLUME("Headphones Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphones Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + { } /* end */ +}; + /* additional init verbs for Benq laptops */ static struct hda_verb alc262_EAPD_verbs[] = { {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -7581,6 +7628,7 @@ [ALC262_HIPPO] = "hippo", [ALC262_HIPPO_1] = "hippo_1", [ALC262_FUJITSU] = "fujitsu", + [ALC262_FUJITSU_C1410] = "fujitsu-c1410", [ALC262_HP_BPC] = "hp-bpc", [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000", [ALC262_BENQ_ED8] = "benq", @@ -7659,6 +7707,18 @@ .input_mux = &alc262_fujitsu_capture_source, .unsol_event = alc262_fujitsu_unsol_event, }, + [ALC262_FUJITSU_C1410] = { + .mixers = { alc262_fujitsu_c1410_mixer }, + .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs, alc262_fujitsu_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_fujitsu_c1410_capture_source, + .unsol_event = alc262_fujitsu_unsol_event, + }, [ALC262_HP_BPC] = { .mixers = { alc262_HP_BPC_mixer }, .init_verbs = { alc262_HP_BPC_init_verbs }, --------------------------------------------------------------------------------------------- ALSA Audio Debug v0.1.0 - su 7.10.2007 15:13:09 +0300 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/gpl.txt Kernel ---------------------------------------------------- Linux oppilas 2.6.22.9-c1410 #2 SMP Sun Oct 7 11:57:15 EEST 2007 i686 GNU/Linux Loaded Modules -------------------------------------------- snd_hda_intel 257560 1 snd_pcm_oss 44928 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm 80772 2 snd_hda_intel,snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 33408 0 snd_seq_midi 9600 0 snd_rawmidi 25856 1 snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 53360 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24196 2 snd_pcm,snd_seq snd_seq_device 9356 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54916 11 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device snd_page_alloc 11400 2 snd_hda_intel,snd_pcm Modprobe Conf --------------------------------------------- Warning: module config file does not exist This means any kernel modules will not be auto loaded See your linux distro docs on how to create this file Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 1.0.14 (Thu May 31 09:03:25 2007 UTC). 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf0640000 irq 22 2: : timer 3: : sequencer 4: [ 0- 6]: digital audio playback 5: [ 0- 6]: digital audio capture 6: [ 0- 2]: digital audio capture 7: [ 0- 1]: digital audio playback 8: [ 0- 0]: digital audio playback 9: [ 0- 0]: digital audio capture 10: [ 0] : control 00-06: Si3054 Modem : Si3054 Modem : playback 1 : capture 1 00-02: ALC262 Analog : ALC262 Analog : capture 2 00-01: ALC262 Digital : ALC262 Digital : playback 1 00-00: ALC262 Analog : ALC262 Analog : playback 1 : capture 2 Client info cur clients : 3 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Connecting To: 15:0 Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 15 : "OSS sequencer" [Kernel] Port 0 : "Receiver" (-we-) Connected From: 0:1 Dev Snd --------------------------------------------------- controlC0 pcmC0D0p pcmC0D2c pcmC0D6p timer pcmC0D0c pcmC0D1p pcmC0D6c seq CPU ------------------------------------------------------- model name : Intel(R) Core(TM)2 CPU T5600 @ 1.83GHz cpu MHz : 1000.000 model name : Intel(R) Core(TM)2 CPU T5600 @ 1.83GHz cpu MHz : 1000.000 RAM ------------------------------------------------------- MemTotal: 1026252 kB SwapTotal: 2441872 kB Hardware -------------------------------------------------- 00:00.0 Host bridge: Intel Corporation Mobile 945GM/PM/GMS/940GML and 945GT Express Memory Controller Hub (rev 03) 00:02.0 VGA compatible controller: Intel Corporation Mobile 945GM/GMS/940GML Express Integrated Graphics Controller (rev 03) 00:02.1 Display controller: Intel Corporation Mobile 945GM/GMS/940GML Express Integrated Graphics Controller (rev 03) 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 1 (rev 02) 00:1c.1 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 2 (rev 02) 00:1c.2 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 3 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #3 (rev 02) 00:1d.3 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #4 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801G (ICH7 Family) USB2 EHCI Controller (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 Mobile PCI Bridge (rev e2) 00:1f.0 ISA bridge: Intel Corporation 82801GBM (ICH7-M) LPC Interface Bridge (rev 02) 00:1f.1 IDE interface: Intel Corporation 82801G (ICH7 Family) IDE Controller (rev 02) 00:1f.2 SATA controller: Intel Corporation 82801GBM/GHM (ICH7 Family) Serial ATA Storage Controller AHCI (rev 02) 00:1f.3 SMBus: Intel Corporation 82801G (ICH7 Family) SMBus Controller (rev 02) 02:00.0 Ethernet controller: Marvell Technology Group Ltd. 88E8055 PCI-E Gigabit Ethernet Controller (rev 12) 05:00.0 Network controller: Intel Corporation PRO/Wireless 3945ABG Network Connection (rev 02) 08:03.0 CardBus bridge: Ricoh Co Ltd RL5c476 II (rev b3) 08:03.1 FireWire (IEEE 1394): Ricoh Co Ltd R5C552 IEEE 1394 Controller (rev 08) 08:03.2 Generic system peripheral [0805]: Ricoh Co Ltd R5C822 SD/SDIO/MMC/MS/MSPro Host Adapter (rev 17) 00:00.0 0600: 8086:27a0 (rev 03) Subsystem: 10cf:1378 Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:02.0 0300: 8086:27a2 (rev 03) (prog-if 00 [VGA]) Subsystem: 10cf:1381 Flags: bus master, fast devsel, latency 0, IRQ 19 Memory at f0300000 (32-bit, non-prefetchable) [size=512K] I/O ports at 1800 [size=8] Memory at e0000000 (32-bit, prefetchable) [size=256M] Memory at f0400000 (32-bit, non-prefetchable) [size=256K] Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [d0] Power Management version 2 00:02.1 0380: 8086:27a6 (rev 03) Subsystem: 10cf:1381 Flags: bus master, fast devsel, latency 0 Memory at f0380000 (32-bit, non-prefetchable) [size=512K] Capabilities: [d0] Power Management version 2 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 10cf:1397 Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at f0640000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) 00:1c.0 0604: 8086:27d0 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 I/O behind bridge: 00002000-00002fff Memory behind bridge: f0000000-f00fffff Prefetchable memory behind bridge: 0000000054000000-00000000540fffff Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 10cf:138b Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1c.1 0604: 8086:27d2 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=03, subordinate=04, sec-latency=0 Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 10cf:138b Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1c.2 0604: 8086:27d4 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=05, subordinate=07, sec-latency=0 Memory behind bridge: 54100000-541fffff Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 10cf:138b Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1d.0 0c03: 8086:27c8 (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 20 I/O ports at 1820 [size=32] 00:1d.1 0c03: 8086:27c9 (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 18 I/O ports at 1840 [size=32] 00:1d.2 0c03: 8086:27ca (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 21 I/O ports at 1860 [size=32] 00:1d.3 0c03: 8086:27cb (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 1880 [size=32] 00:1d.7 0c03: 8086:27cc (rev 02) (prog-if 20 [EHCI]) Subsystem: 10cf:138a Flags: bus master, medium devsel, latency 0, IRQ 20 Memory at f0644000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port 00:1e.0 0604: 8086:2448 (rev e2) (prog-if 01 [Subtractive decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=08, subordinate=0c, sec-latency=32 I/O behind bridge: 00003000-00003fff Memory behind bridge: f0200000-f02fffff Prefetchable memory behind bridge: 0000000050000000-0000000053ffffff Capabilities: [50] Subsystem: 10cf:1383 00:1f.0 0601: 8086:27b9 (rev 02) Subsystem: 10cf:1384 Flags: bus master, medium devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:1f.1 0101: 8086:27df (rev 02) (prog-if 8a [Master SecP PriP]) Subsystem: 10cf:1385 Flags: bus master, medium devsel, latency 0, IRQ 21 I/O ports at 01f0 [size=8] I/O ports at 03f4 [size=1] I/O ports at 0170 [size=8] I/O ports at 0374 [size=1] I/O ports at 1810 [size=16] 00:1f.2 0106: 8086:27c5 (rev 02) (prog-if 01 [AHCI 1.0]) Subsystem: 10cf:1387 Flags: bus master, 66MHz, medium devsel, latency 0, IRQ 220 I/O ports at 18d0 [size=8] I/O ports at 18c4 [size=4] I/O ports at 18c8 [size=8] I/O ports at 18c0 [size=4] I/O ports at 18b0 [size=16] Memory at f0644400 (32-bit, non-prefetchable) [size=1K] Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [70] Power Management version 2 00:1f.3 0c05: 8086:27da (rev 02) Subsystem: 10cf:1388 Flags: medium devsel, IRQ 11 I/O ports at 18e0 [size=32] 02:00.0 0200: 11ab:4363 (rev 12) Subsystem: 10cf:139a Flags: bus master, fast devsel, latency 0, IRQ 219 Memory at f0000000 (64-bit, non-prefetchable) [size=16K] I/O ports at 2000 [size=256] [virtual] Expansion ROM at 54000000 [disabled] [size=128K] Capabilities: [48] Power Management version 3 Capabilities: [50] Vital Product Data Capabilities: [5c] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable+ Capabilities: [e0] Express Legacy Endpoint IRQ 0 Capabilities: [100] Advanced Error Reporting 05:00.0 0280: 8086:4222 (rev 02) Subsystem: 8086:1001 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at 54100000 (32-bit, non-prefetchable) [size=4K] Capabilities: [c8] Power Management version 2 Capabilities: [d0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [e0] Express Legacy Endpoint IRQ 0 Capabilities: [100] Advanced Error Reporting Capabilities: [140] Device Serial Number b5-ec-71-ff-ff-77-1b-00 08:03.0 0607: 1180:0476 (rev b3) Subsystem: 10cf:1254 Flags: bus master, medium devsel, latency 168, IRQ 19 Memory at f0201000 (32-bit, non-prefetchable) [size=4K] Bus: primary=08, secondary=09, subordinate=0c, sec-latency=176 Memory window 0: 50000000-53fff000 (prefetchable) Memory window 1: 58000000-5bfff000 I/O window 0: 00003000-000030ff I/O window 1: 00003400-000034ff 16-bit legacy interface ports at 0001 08:03.1 0c00: 1180:0552 (rev 08) (prog-if 10 [OHCI]) Subsystem: 10cf:1255 Flags: bus master, medium devsel, latency 32, IRQ 22 Memory at f0200000 (32-bit, non-prefetchable) [size=2K] Capabilities: [dc] Power Management version 2 08:03.2 0805: 1180:0822 (rev 17) Subsystem: 10cf:1323 Flags: bus master, medium devsel, latency 32, IRQ 19 Memory at f0200800 (32-bit, non-prefetchable) [size=256] Capabilities: [80] Power Management version 2 --------------------------------------------------------------------------------------------- Veli-Matti Lintu

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[alsa-devel] snd-hda-intel: no sound on ASUS F3Sa
by Luca Tettamanti 11 Oct '07

11 Oct '07

Hi, I can't enable sound on a (new) ASUS F3Sa laptop with this sound device: 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: ASUSTeK Computer Inc. Unknown device 1339 Control: I/O- Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Interrupt: pin A routed to IRQ 22 Region 0: Memory at febf8000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Codec: kronos@nb:~$ cat /proc/asound/card0/codec#0 | head -n1 Codec: Realtek ALC660-VD (full output attached) without arguments I get: ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 22 PCI: Setting latency timer of device 0000:00:1b.0 to 64 codec_mask = 0x3 hda_codec: model '3stack-660' is selected for config 1043:1339 (Asus G1) while with model=auto: ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 22 PCI: Setting latency timer of device 0000:00:1b.0 to 64 codec_mask = 0x3 hda_codec: model 'auto' is selected autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) inputs: mic=0x18, fmic=0x19, line=0x0, fline=0x0, cd=0x0, aux=0x0 I also tried all other possible models without success; I can't get any output on the built-in speakers nor on the headphones. I tried both 2.6.23-git (from a couple of days ago) and ALSA 1.0.15rc3. Luca (CC me, I'm not subscribed) -- "Sei l'unica donna della mia vita". (Adamo)

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[alsa-devel] How to enforce 32bit access?
by Joachim Foerster 11 Oct '07

11 Oct '07

Hi ALSA devs, (based on a Xilinx ML403, PPC405, OPB bus) I have a custom sound controller, which allows 32bit write accesses to its IO memory, only. The controller supports S16_LE, S16_BE, U16_LE, U16_BE with min. and max. 2 channels (stereo), only! => So a frame has to be 32bits wide and one frame has to be written at one point of time (= one 32bit bus access). Now, playing a 16kHz, S16_LE, Mono .wav-file with aplay invokes the conversion framework of the alsa-lib. The following line of code (of pcm_plugin.h) is executed: conv_xx12_xx12: as_u16(dst) = as_u16c(src); This is a 16bit access and thus fails with a Bus error/Kernel Oops. Therefore my question is: In my driver for the controller, how do I tell ALSA, that it is only allowed to write in 32bit quantities? Is this possible at all? Or do I have to use a special .asoundrc file? Thanks, Joachim PS: 16bit, signed/unsigned, LE/BE, stereo (!) .wav files can be played without any problem.

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[alsa-devel] alsa 1.0.15
by paul blakeley 11 Oct '07

11 Oct '07

Can someone please tell me when version 1.0.15 will be released and then become the current stable package?

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[alsa-devel] dont know if bug in /tmp/217usbkeyboardbug.log
by linuxcbon 11 Oct '07

11 Oct '07

hi, I have no idea if this is a bug : tv card is hauppauge wintv model 44346, tuner model is Philips FI1216MF MK2 I found in /tmp/217usbkeyboardbug.log Last 20 lines of dmesg: tveeprom 2-0050: has radio bttv0: Hauppauge eeprom indicates model#44346 bttv0: using tuner=3 bttv0: i2c: checking for MSP34xx @ 0x80... found msp3400 2-0040: MSP3415G-B8 found @ 0x80 (bt878 #0 [sw]) msp3400 2-0040: MSP3415G-B8 supports nicam and radio, mode is autodetect and autoselect bttv0: i2c: checking for TDA9875 @ 0xb0... not found bttv0: i2c: checking for TDA7432 @ 0x8a... not found bttv0: i2c: checking for TDA9887 @ 0x86... not found tuner 2-0061: chip found @ 0xc2 (bt878 #0 [sw]) tuner 2-0061: type set to 3 (Philips (SECAM+PAL_BG) (FI1216MF, FM1216MF, FR1216MF)) tuner 2-0061: type set to 3 (Philips (SECAM+PAL_BG) (FI1216MF, FM1216MF, FR1216MF)) bttv0: registered device video0 bttv0: registered device vbi0 bttv0: registered device radio0 bttv0: PLL: 28636363 => 35468950 .. ok ALSA /mnt/hda3/sources/sources116/alsa-driver-1.0.15rc2/pci/bt87x.c:872: unknown card 0x878-0x70:0x13eb ALSA /mnt/hda3/sources/sources116/alsa-driver-1.0.15rc2/pci/bt87x.c:875: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> ACPI: PCI Interrupt 0000:01:04.1[A] -> Link [LNK1] -> GSI 5 (level, low) -> IRQ 5 ALSA /mnt/hda3/sources/sources116/alsa-driver-1.0.15rc2/pci/bt87x.c:943: bt87x0: Using board 0, analog, digital (rate 32000 Hz) I hope this can help. regards _____________________________________________________________________________ Ne gardez plus qu'une seule adresse mail ! Copiez vos mails vers Yahoo! Mail

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[alsa-devel] vaio_mux's num_items is not equal to the actual items
by zhejiang 11 Oct '07

11 Oct '07

In patch_sigmatel.c, static struct hda_input_mux vaio_mux = { .num_items = 2, .items = { /* { "HP", 0x0 }, */ { "Mic Jack", 0x1 }, { "Internal Mic", 0x2 }, { "PCM", 0x3 }, } }; Why is the num_items not 3 ? The "PCM" won't be selected with num_times = 2.

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[alsa-devel] Creative Labs XFi drivers.
by Tobin Davis 11 Oct '07

11 Oct '07

Creative Labs has released a proprietary driver base for their XFi cards. I downloaded it because it claims to have compatibility with alsa. In fact, it requires alsa include files to be able to compile. I haven't looked at the source yet (this information is all in the Readme.txt in the main download) and probably won't (especially since I don't own a Creative Labs XFi card). What I don't understand, is are they allowed to use our code to build their drivers? They use alsa core modules to interoperate (from what I can tell in the Readme.txt). Also, this line in the License.txt is especially troubling: > In the event that the Software includes Linux code or drivers, then, unless expressly stated to the contrary, such Linux > code and drivers are proprietary to Creative and all rights therein are reserved. I'm deleting all but the big tarball I downloaded. I'll delete it later. -- Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Alsa 1.0.15rc2 and oops with Dell M1210
by zhejiang 10 Oct '07

10 Oct '07

Seems that the spec->input_mux contains no item at all. So system maybe crash in the snd_hda_input_mux_info() when try to copy the items[].label,which is null pointer. May you replace the function stac92xx_mux_enum_info() in alsa-driver-1.0.15rc2/alsa-kernel/pci/hda/patch_sigmatel.c ? static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; printk("spec->input_mux->num_items is %d \n",spec->input_mux->num_items); if(!spec->input_mux->num_items) return 0; return snd_hda_input_mux_info(spec->input_mux, uinfo); } Does it still crash? What's the number of num_items? The pin_config for M1210 didn't include any input pin. I wonder why.

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Re: [alsa-devel] chmod with USB sound device?
by william estrada 10 Oct '07

10 Oct '07

Harald, I revisited /etc/security/console.perms.d/50-default.perms and got it to work. For the list, I changed: In /etc/security/console.perms.d/50-default.perms > # permission definitions . > <console> 0666 <sound> 0666 root And In /etc/udev/rules.d/50-udev.rules > KERNEL=="snd/*", MODE="0666", OWNER="wre00:root" Then I did the udevtrigger. Then i did the udevtrigger, plugged in my USB sound device and got this: > [root@laptop wre00]# ls -lrt /dev/snd > total 0 > crw-rw-rw- 1 wre00 root 116, 2 Oct 10 09:45 timer > crw-rw-rw- 1 wre00 root 116, 3 Oct 10 09:45 seq > crw-rw-rw- 1 wre00 root 116, 4 Oct 10 09:45 pcmC0D0p > crw-rw-rw- 1 wre00 root 116, 5 Oct 10 09:45 pcmC0D0c > crw-rw-rw- 1 wre00 root 116, 6 Oct 10 09:45 controlC0 > crw-rw-rw- 1 wre00 root 116, 7 Oct 10 10:28 pcmC1D0p > crw-rw-rw- 1 wre00 root 116, 8 Oct 10 10:28 pcmC1D0c > crw-rw-rw- 1 wre00 root 116, 9 Oct 10 10:28 controlC1 I guess both changes are required but it now works. Thanks for your time. William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum Harald Hoyer wrote: > william estrada wrote: >> Hi guys, >> >> I'm trying to run 'chmod' on a USB sound card using 'Hot Plug' udev >> support. >> From reading, I think I need to update my >> '/etc/udev/rules.d/50-udev.rules'? >> I have made these changes: >>> # audio devices . >>> KERNEL=="snd/*", MODE="0666" > > You may have a look in pam_console... > /etc/security/console.perms > /etc/security/console.perms.d/* >

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[alsa-devel] snd-hdsp & digiface
by Fernando Lopez-Lezcano 10 Oct '07

10 Oct '07

Hi... We are using both digiface and multiface interfaces here at ccrma and we have a (small) problem of consistency in the channel numbering because of the way they are connected. Both cards are connected digitally to digital mixers, but in the case of the multiface the digital channels (which would be the ones users use) are 9-16 while on the digiface the first eight are 1-8[*]. Is there a way to change the ordering in the multiface so that the digital interface is presented as ports 1-8 instead of 9-16? (even a hack to the driver would be welcome) - and preferably one that does not need the default interface to be changed from hw:0 in jack... -- Fernando [*] accounts are shared so any configuration you do on one machine for 1-8 will be wrong for the other where the real outputs are 9-16.

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Re: [alsa-devel] different user causes 'no such device'
by william estrada 10 Oct '07

10 Oct '07

Thanks, I did a 'chmod -R 666 /dev/snd/' and it worked. Now how do I make the hot plugged USB sound device use these setting?? Thanks for your time. William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum Takashi Iwai wrote: > At Wed, 03 Oct 2007 12:55:27 -0700, > william estrada wrote: >> Hi guys, >> >> I have what seams like a interesting problem. I have written an ALSA app >> to record wav files and it runs while I am logged in an terminal as the >> logged in Gnome user or root. But when I su to another user, it can't >> find the default sound card. >> >> I am running FC6. I tried changing own of /dev/dsp to 'nobody', that did >> not work. >> >> Anyone know of this problem and a fix for it. >> >> Thanks for your time. >> >> -- >> William Estrada >> MrUmunhum(a)popdial.com >> Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) >> Ymessenger: MrUmunhum >> >>> MSGrec - Version 1.1, Mt-Umunhum-Wireless.net >>> WavMSG: Paused: enter 'q' to stop, <ENTER> to start. >>> ALSA lib confmisc.c:848:(snd_func_card_driver) cannot find card '0' >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device >>> ALSA lib confmisc.c:397:(snd_func_concat) error evaluating strings >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_concat returned error: No such device >>> ALSA lib confmisc.c:1248:(snd_func_refer) error evaluating name >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_refer returned error: No such device >>> ALSA lib conf.c:3972:(snd_config_expand) Evaluate error: No such device >>> ALSA lib pcm.c:2145:(snd_pcm_open_noupdate) Unknown PCM default > > ALSA-native apps access only ALSA device files, usually /dev/snd/*, > not /dev/dsp*, those for OSS. In this case, it should be > /dev/snd/controlC* files. Check the premission of these files. > > > Takashi >

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[alsa-devel] Add quirk support for Dell XPS1330 and Inspiron 1420.
by Tim Gardner 10 Oct '07

10 Oct '07

-- Tim Gardner tim.gardner(a)ubuntu.com

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[alsa-devel] [PATCH] stac9200 eapd fixes
by Brian Hinz 10 Oct '07

10 Oct '07

Hi, The following patch addresses alsa bug #0002948 and it's (many) duplicates. Please consider it for inclusion in the next release. Thanks, -brian --- alsa-kernel/pci/hda/hda_codec.c 2007-09-24 21:46:11.000000000 -0400 +++ alsa-kernel/pci/hda/hda_codec.c 2007-09-24 22:00:15.000000000 -0400 @@ -1629,11 +1629,30 @@ power_state); nid = codec->start_nid; + unsigned int pincap; for (i = 0; i < codec->num_nodes; i++, nid++) { - if (get_wcaps(codec, nid) & AC_WCAP_POWER) - snd_hda_codec_write(codec, nid, 0, + if (get_wcaps(codec, nid) & AC_WCAP_POWER) { + pincap = snd_hda_param_read(codec, nid, + AC_PAR_PIN_CAP); + /* + * don't power down the widget if it controls eapd + * and EAPD_BTLENABLE is set. + */ + if (pincap & AC_PINCAP_EAPD) { + int eapd = snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_EAPD_BTLENABLE, 0); + eapd &= 0x02; + if (power_state != AC_PWRST_D3 || !eapd) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_POWER_STATE, + power_state); + } + } else { + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, power_state); + } + } } if (power_state == AC_PWRST_D0) { --- alsa-kernel/pci/hda/patch_sigmatel.c 2007-09-24 21:46:12.000000000 -0400 +++ alsa-kernel/pci/hda/patch_sigmatel.c 2007-09-26 20:57:55.000000000 -0400 @@ -378,6 +378,13 @@ {} }; +static struct hda_verb stac9200_eapd_init[] = { + /* set dac0mux for dac converter */ + { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {} +}; + static struct hda_verb stac925x_core_init[] = { /* set dac0mux for dac converter */ { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -2492,7 +2499,23 @@ spec->num_dmics = 0; spec->num_adcs = 1; - spec->init = stac9200_core_init; + /* + * EAPD needs to be set on resume + */ + switch (codec->subsystem_id) { + case 0x107b0205: /* Gateway S-7110M */ + spec->init = stac9200_eapd_init; + break; + case 0x107b0317: /* Gateway MT3423, MX3412, MX3410, MX3414 */ + spec->init = stac9200_eapd_init; + break; + case 0x107b0318: /* Gateway ML3109, MT3707 */ + spec->init = stac9200_eapd_init; + break; + default: + spec->init = stac9200_core_init; + break; + } spec->mixer = stac9200_mixer; err = stac9200_parse_auto_config(codec);

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[PATCH] UBUNTU: Re-add quirk support for Dell XPS 1330 and Inspiron 1420.
by Tim Gardner 09 Oct '07

09 Oct '07

Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> --- .../alsa-kernel/pci/hda/patch_sigmatel.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/updates/alsa/alsa-driver/alsa-kernel/pci/hda/patch_sigmatel.c b/updates/alsa/alsa-driver/alsa-kernel/pci/hda/patch_sigmatel.c index 27360d2..f714de1 100644 --- a/updates/alsa/alsa-driver/alsa-kernel/pci/hda/patch_sigmatel.c +++ b/updates/alsa/alsa-driver/alsa-kernel/pci/hda/patch_sigmatel.c @@ -1097,11 +1097,13 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_D965_3ST), /* Dell 3 stack systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ed, "Dell ", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f4, "Dell ", STAC_DELL_3ST), /* 965 based 5 stack systems */ + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST), -- 1.5.2.5 --------------080009030006050203010104--

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[alsa-devel] ALSA Support for Maya44 PCI?
by Piotr Makowski 09 Oct '07

09 Oct '07

There is already a programmer working on the driver, but thanks for your interest! You can expect the driver for ESI MAYA44 PCI card soon. There will be also an official announcement on alsa-devel when the programmer will start his work. Regards, Piotr Makowski 2007/10/9, Nicolas Rodriguez <nicolas.rod(a)gmail.com>: > I bought a Maya44 PCI 4-channel soundcard and I'll like to use it in Linux. > I couldn't get it to work with Ubuntu (I don't remember the ALSA version, > sorry). Do you support this card? If you don't, I can create the driver for > it. I've read somewhere that ESI-PRO can give the card specifications to > developers interested in build drivers for their cards. I've sent them an > email asking for the card specifications for I've got no response. What do > you recommend? > > Thanks in advance, > Nico. > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH] [RFC] sis7019: Initial support for the SiS 7019 Audio Accelerator
by Dave Dillow 09 Oct '07

09 Oct '07

Basic audio support for the SiS 7019 Audio Accelerator as found in the SiS 55x SoC. There is currently no synth or power management support at the moment, but the audio playback has been tested with several channel and rate combinations, as well as various period and buffer configurations. Signed-off-by: David Dillow <dave(a)thedillows.org> --- include/sound/sis7019.h | 342 ++++++++++++++ sound/pci/Kconfig | 10 + sound/pci/Makefile | 2 + sound/pci/sis7019.c | 1197 +++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1551 insertions(+), 0 deletions(-) diff --git a/include/sound/sis7019.h b/include/sound/sis7019.h new file mode 100644 index 0000000..d395abb --- /dev/null +++ b/include/sound/sis7019.h @@ -0,0 +1,342 @@ +#ifndef __sis7019_h__ +#define __sis7019_h__ + +/* + * Definitions for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + + +/* General Control Register */ +#define SIS_GCR 0x00 +#define SIS_GCR_MACRO_POWER_DOWN 0x80000000 +#define SIS_GCR_MODEM_ENABLE 0x00010000 +#define SIS_GCR_SOFTWARE_RESET 0x00000001 + +/* General Interrupt Enable Register */ +#define SIS_GIER 0x04 +#define SIS_GIER_MODEM_TIMER_IRQ_ENABLE 0x00100000 +#define SIS_GIER_MODEM_RX_DMA_IRQ_ENABLE 0x00080000 +#define SIS_GIER_MODEM_TX_DMA_IRQ_ENABLE 0x00040000 +#define SIS_GIER_AC97_GPIO1_IRQ_ENABLE 0x00020000 +#define SIS_GIER_AC97_GPIO0_IRQ_ENABLE 0x00010000 +#define SIS_GIER_AC97_SAMPLE_TIMER_IRQ_ENABLE 0x00000010 +#define SIS_GIER_AUDIO_GLOBAL_TIMER_IRQ_ENABLE 0x00000008 +#define SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE 0x00000004 +#define SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE 0x00000002 +#define SIS_GIER_AUDIO_WAVE_ENGINE_IRQ_ENABLE 0x00000001 + +/* General Interrupt Status Register */ +#define SIS_GISR 0x08 +#define SIS_GISR_MODEM_TIMER_IRQ_STATUS 0x00100000 +#define SIS_GISR_MODEM_RX_DMA_IRQ_STATUS 0x00080000 +#define SIS_GISR_MODEM_TX_DMA_IRQ_STATUS 0x00040000 +#define SIS_GISR_AC97_GPIO1_IRQ_STATUS 0x00020000 +#define SIS_GISR_AC97_GPIO0_IRQ_STATUS 0x00010000 +#define SIS_GISR_AC97_SAMPLE_TIMER_IRQ_STATUS 0x00000010 +#define SIS_GISR_AUDIO_GLOBAL_TIMER_IRQ_STATUS 0x00000008 +#define SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS 0x00000004 +#define SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS 0x00000002 +#define SIS_GISR_AUDIO_WAVE_ENGINE_IRQ_STATUS 0x00000001 + +/* DMA Control Register */ +#define SIS_DMA_CSR 0x10 +#define SIS_DMA_CSR_PCI_SETTINGS 0x0000001d +#define SIS_DMA_CSR_CONCURRENT_ENABLE 0x00000200 +#define SIS_DMA_CSR_PIPELINE_ENABLE 0x00000100 +#define SIS_DMA_CSR_RX_DRAIN_ENABLE 0x00000010 +#define SIS_DMA_CSR_RX_FILL_ENABLE 0x00000008 +#define SIS_DMA_CSR_TX_DRAIN_ENABLE 0x00000004 +#define SIS_DMA_CSR_TX_LOWPRI_FILL_ENABLE 0x00000002 +#define SIS_DMA_CSR_TX_HIPRI_FILL_ENABLE 0x00000001 + +/* Playback Channel Start Registers */ +#define SIS_PLAY_START_A_REG 0x14 +#define SIS_PLAY_START_B_REG 0x18 + +/* Playback Channel Stop Registers */ +#define SIS_PLAY_STOP_A_REG 0x1c +#define SIS_PLAY_STOP_B_REG 0x20 + +/* Recording Channel Start Register */ +#define SIS_RECORD_START_REG 0x24 + +/* Recording Channel Stop Register */ +#define SIS_RECORD_STOP_REG 0x28 + +/* Playback Interrupt Status Registers */ +#define SIS_PISR_A 0x2c +#define SIS_PISR_B 0x30 + +/* Recording Interrupt Status Register */ +#define SIS_RISR 0x34 + +/* AC97 AC-link Playback Source Register */ +#define SIS_AC97_PSR 0x40 +#define SIS_AC97_PSR_MODEM_HEADSET_SRC_MIXER 0x0f000000 +#define SIS_AC97_PSR_MODEM_LINE2_SRC_MIXER 0x00f00000 +#define SIS_AC97_PSR_MODEM_LINE1_SRC_MIXER 0x000f0000 +#define SIS_AC97_PSR_PCM_LFR_SRC_MIXER 0x0000f000 +#define SIS_AC97_PSR_PCM_SURROUND_SRC_MIXER 0x00000f00 +#define SIS_AC97_PSR_PCM_CENTER_SRC_MIXER 0x000000f0 +#define SIS_AC97_PSR_PCM_LR_SRC_MIXER 0x0000000f + +/* AC97 AC-link Command Register */ +#define SIS_AC97_CMD 0x50 +#define SIS_AC97_CMD_DATA_MASK 0xffff0000 +#define SIS_AC97_CMD_REG_MASK 0x0000ff00 +#define SIS_AC97_CMD_CODEC3_READ 0x0000000d +#define SIS_AC97_CMD_CODEC3_WRITE 0x0000000c +#define SIS_AC97_CMD_CODEC2_READ 0x0000000b +#define SIS_AC97_CMD_CODEC2_WRITE 0x0000000a +#define SIS_AC97_CMD_CODEC_READ 0x00000009 +#define SIS_AC97_CMD_CODEC_WRITE 0x00000008 +#define SIS_AC97_CMD_CODEC_WARM_RESET 0x00000005 +#define SIS_AC97_CMD_CODEC_COLD_RESET 0x00000004 +#define SIS_AC97_CMD_DONE 0x00000000 + +/* AC97 AC-link Semaphore Register */ +#define SIS_AC97_SEMA 0x54 +#define SIS_AC97_SEMA_BUSY 0x00000001 +#define SIS_AC97_SEMA_RELEASE 0x00000000 + +/* AC97 AC-link Status Register */ +#define SIS_AC97_STATUS 0x58 +#define SIS_AC97_STATUS_AUDIO_D2_INACT_SECS 0x03f00000 +#define SIS_AC97_STATUS_MODEM_ALIVE 0x00002000 +#define SIS_AC97_STATUS_AUDIO_ALIVE 0x00001000 +#define SIS_AC97_STATUS_CODEC3_READY 0x00000400 +#define SIS_AC97_STATUS_CODEC2_READY 0x00000200 +#define SIS_AC97_STATUS_CODEC_READY 0x00000100 +#define SIS_AC97_STATUS_WARM_RESET 0x00000080 +#define SIS_AC97_STATUS_COLD_RESET 0x00000040 +#define SIS_AC97_STATUS_POWERED_DOWN 0x00000020 +#define SIS_AC97_STATUS_NORMAL 0x00000010 +#define SIS_AC97_STATUS_READ_EXPIRED 0x00000004 +#define SIS_AC97_STATUS_SEMAPHORE 0x00000002 +#define SIS_AC97_STATUS_BUSY 0x00000001 + +/* AC97 AC-link Audio Configuration Register */ +#define SIS_AC97_CONF 0x5c +#define SIS_AC97_CONF_AUDIO_ALIVE 0x80000000 +#define SIS_AC97_CONF_WARM_RESET_ENABLE 0x40000000 +#define SIS_AC97_CONF_PR6_ENABLE 0x20000000 +#define SIS_AC97_CONF_PR5_ENABLE 0x10000000 +#define SIS_AC97_CONF_PR4_ENABLE 0x08000000 +#define SIS_AC97_CONF_PR3_ENABLE 0x04000000 +#define SIS_AC97_CONF_PR2_PR7_ENABLE 0x02000000 +#define SIS_AC97_CONF_PR0_PR1_ENABLE 0x01000000 +#define SIS_AC97_CONF_AUTO_PM_ENABLE 0x00800000 +#define SIS_AC97_CONF_PCM_LFE_ENABLE 0x00080000 +#define SIS_AC97_CONF_PCM_SURROUND_ENABLE 0x00040000 +#define SIS_AC97_CONF_PCM_CENTER_ENABLE 0x00020000 +#define SIS_AC97_CONF_PCM_LR_ENABLE 0x00010000 +#define SIS_AC97_CONF_PCM_CAP_MIC_ENABLE 0x00002000 +#define SIS_AC97_CONF_PCM_CAP_LR_ENABLE 0x00001000 +#define SIS_AC97_CONF_PCM_CAP_MIC_FROM_CODEC3 0x00000200 +#define SIS_AC97_CONF_PCM_CAP_LR_FROM_CODEC3 0x00000100 +#define SIS_AC97_CONF_CODEC3_PM_VRM 0x00000080 +#define SIS_AC97_CONF_CODEC_PM_VRM 0x00000040 +#define SIS_AC97_CONF_CODEC3_VRA_ENABLE 0x00000020 +#define SIS_AC97_CONF_CODEC_VRA_ENABLE 0x00000010 +#define SIS_AC97_CONF_CODEC3_PM_EAC 0x00000008 +#define SIS_AC97_CONF_CODEC_PM_EAC 0x00000004 +#define SIS_AC97_CONF_CODEC3_EXISTS 0x00000002 +#define SIS_AC97_CONF_CODEC_EXISTS 0x00000001 + +/* Playback Channel Sync Group registers */ +#define SIS_PLAY_SYNC_GROUP_A 0x80 +#define SIS_PLAY_SYNC_GROUP_B 0x84 +#define SIS_PLAY_SYNC_GROUP_C 0x88 +#define SIS_PLAY_SYNC_GROUP_D 0x8c +#define SIS_MIXER_SYNC_GROUP 0x90 + +/* Wave Engine Config and Control Register */ +#define SIS_WECCR 0xa0 +#define SIS_WECCR_TESTMODE_MASK 0x00300000 +#define SIS_WECCR_TESTMODE_NORMAL 0x00000000 +#define SIS_WECCR_TESTMODE_BYPASS_NSO_ALPHA 0x00100000 +#define SIS_WECCR_TESTMODE_BYPASS_FC 0x00200000 +#define SIS_WECCR_TESTMODE_BYPASS_WOL 0x00300000 +#define SIS_WECCR_RESONANCE_DELAY_MASK 0x00060000 +#define SIS_WECCR_RESONANCE_DELAY_NONE 0x00000000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1F00 0x00020000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1E00 0x00040000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1C00 0x00060000 +#define SIS_WECCR_IGNORE_CHANNEL_PARMS 0x00010000 +#define SIS_WECCR_COMMAND_CHANNEL_ID_MASK 0x0003ff00 +#define SIS_WECCR_COMMAND_MASK 0x00000007 +#define SIS_WECCR_COMMAND_NONE 0x00000000 +#define SIS_WECCR_COMMAND_DONE 0x00000000 +#define SIS_WECCR_COMMAND_PAUSE 0x00000001 +#define SIS_WECCR_COMMAND_TOGGLE_VEG 0x00000002 +#define SIS_WECCR_COMMAND_TOGGLE_MEG 0x00000003 +#define SIS_WECCR_COMMAND_TOGGLE_VEG_MEG 0x00000004 + +/* Wave Engine Volume Control Register */ +#define SIS_WEVCR 0xa4 +#define SIS_WEVCR_LEFT_MUSIC_ATTENUATION_MASK 0xff000000 +#define SIS_WEVCR_RIGHT_MUSIC_ATTENUATION_MASK 0x00ff0000 +#define SIS_WEVCR_LEFT_WAVE_ATTENUATION_MASK 0x0000ff00 +#define SIS_WEVCR_RIGHT_WAVE_ATTENUATION_MASK 0x000000ff + +/* Wave Engine Interrupt Status Registers */ +#define SIS_WEISR_A 0xa8 +#define SIS_WEISR_B 0xac + + +/* Playback DMA parameters (paramter RAM) */ +#define SIS_PLAY_DMA_OFFSET 0x0000 +#define SIS_PLAY_DMA_SIZE 0x10 +#define SIS_PLAY_DMA_ADDR(addr, num) \ + ((num * SIS_PLAY_DMA_SIZE) + (addr) + SIS_PLAY_DMA_OFFSET) + +#define SIS_PLAY_DMA_FORMAT_CSO 0x00 +#define SIS_PLAY_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_PLAY_DMA_FORMAT_8BIT 0x00040000 +#define SIS_PLAY_DMA_FORMAT_MONO 0x00020000 +#define SIS_PLAY_DMA_CSO_MASK 0x0000ffff +#define SIS_PLAY_DMA_BASE 0x04 +#define SIS_PLAY_DMA_CONTROL 0x08 +#define SIS_PLAY_DMA_STOP_AT_SSO 0x04000000 +#define SIS_PLAY_DMA_RELEASE 0x02000000 +#define SIS_PLAY_DMA_LOOP 0x01000000 +#define SIS_PLAY_DMA_INTR_AT_SSO 0x00080000 +#define SIS_PLAY_DMA_INTR_AT_ESO 0x00040000 +#define SIS_PLAY_DMA_INTR_AT_LEO 0x00020000 +#define SIS_PLAY_DMA_INTR_AT_MLP 0x00010000 +#define SIS_PLAY_DMA_LEO_MASK 0x0000ffff +#define SIS_PLAY_DMA_SSO_ESO 0x0c +#define SIS_PLAY_DMA_SSO_MASK 0xffff0000 +#define SIS_PLAY_DMA_ESO_MASK 0x0000ffff + +/* Capture DMA parameters (paramter RAM) */ +#define SIS_CAPTURE_DMA_OFFSET 0x0800 +#define SIS_CAPTURE_DMA_SIZE 0x10 +#define SIS_CAPTURE_DMA_ADDR(addr, num) \ + ((num * SIS_CAPTURE_DMA_SIZE) + (addr) + SIS_CAPTURE_DMA_OFFSET) + +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_0 0 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_1 1 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_2 2 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_3 3 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_4 4 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_5 5 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_6 6 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_7 7 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_8 8 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_9 9 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_10 10 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_11 11 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_12 12 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_13 13 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_14 14 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_15 15 +#define SIS_CAPTURE_CHAN_AC97_PCM_IN 16 +#define SIS_CAPTURE_CHAN_AC97_MIC_IN 17 +#define SIS_CAPTURE_CHAN_AC97_LINE1_IN 18 +#define SIS_CAPTURE_CHAN_AC97_LINE2_IN 19 +#define SIS_CAPTURE_CHAN_AC97_HANDSE_IN 20 + +#define SIS_CAPTURE_DMA_FORMAT_CSO 0x00 +#define SIS_CAPTURE_DMA_MONO_MODE_MASK 0xc0000000 +#define SIS_CAPTURE_DMA_MONO_MODE_AVG 0x00000000 +#define SIS_CAPTURE_DMA_MONO_MODE_LEFT 0x40000000 +#define SIS_CAPTURE_DMA_MONO_MODE_RIGHT 0x80000000 +#define SIS_CAPTURE_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_CAPTURE_DMA_FORMAT_8BIT 0x00040000 +#define SIS_CAPTURE_DMA_FORMAT_MONO 0x00020000 +#define SIS_CAPTURE_DMA_CSO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_BASE 0x04 +#define SIS_CAPTURE_DMA_CONTROL 0x08 +#define SIS_CAPTURE_DMA_STOP_AT_SSO 0x04000000 +#define SIS_CAPTURE_DMA_RELEASE 0x02000000 +#define SIS_CAPTURE_DMA_LOOP 0x01000000 +#define SIS_CAPTURE_DMA_INTR_AT_LEO 0x00020000 +#define SIS_CAPTURE_DMA_INTR_AT_MLP 0x00010000 +#define SIS_CAPTURE_DMA_LEO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_RESERVED 0x0c + + +/* Mixer routing list start pointer (parameter RAM) */ +#define SIS_MIXER_START_OFFSET 0x1000 +#define SIS_MIXER_START_SIZE 0x04 +#define SIS_MIXER_START_ADDR(addr, num) \ + ((num * SIS_MIXER_START_SIZE) + (addr) + SIS_MIXER_START_OFFSET) + +#define SIS_MIXER_START_MASK 0x0000007f + +/* Mixer routing table (parameter RAM) */ +#define SIS_MIXER_OFFSET 0x1400 +#define SIS_MIXER_SIZE 0x04 +#define SIS_MIXER_ADDR(addr, num) \ + ((num * SIS_MIXER_SIZE) + (addr) + SIS_MIXER_OFFSET) + +#define SIS_MIXER_RIGHT_ATTENUTATION_MASK 0xff000000 +#define SIS_MIXER_RIGHT_NO_ATTEN 0xff000000 +#define SIS_MIXER_LEFT_ATTENUTATION_MASK 0xff000000 +#define SIS_MIXER_LEFT_NO_ATTEN 0xff000000 +#define SIS_MIXER_NEXT_ENTRY_MASK 0x00007f00 +#define SIS_MIXER_NEXT_ENTRY_NONE 0x00000000 +#define SIS_MIXER_DEST_MASK 0x0000007f +#define SIS_MIXER_DEST_0 0x00000020 +#define SIS_MIXER_DEST_1 0x00000021 +#define SIS_MIXER_DEST_2 0x00000022 +#define SIS_MIXER_DEST_3 0x00000023 +#define SIS_MIXER_DEST_4 0x00000024 +#define SIS_MIXER_DEST_5 0x00000025 +#define SIS_MIXER_DEST_6 0x00000026 +#define SIS_MIXER_DEST_7 0x00000027 +#define SIS_MIXER_DEST_8 0x00000028 +#define SIS_MIXER_DEST_9 0x00000029 +#define SIS_MIXER_DEST_10 0x0000002a +#define SIS_MIXER_DEST_11 0x0000002b +#define SIS_MIXER_DEST_12 0x0000002c +#define SIS_MIXER_DEST_13 0x0000002d +#define SIS_MIXER_DEST_14 0x0000002e +#define SIS_MIXER_DEST_15 0x0000002f + +/* Wave Engine Control Parameters (parameter RAM) */ +#define SIS_WAVE_OFFSET 0x2000 +#define SIS_WAVE_SIZE 0x40 +#define SIS_WAVE_ADDR(addr, num) \ + ((num * SIS_WAVE_SIZE) + (addr) + SIS_WAVE_OFFSET) + +#define SIS_WAVE_GENERAL 0x00 +#define SIS_WAVE_GENERAL_WAVE_VOLUME 0x80000000 +#define SIS_WAVE_GENERAL_MUSIC_VOLUME 0x00000000 +#define SIS_WAVE_GENERAL_VOLUME_MASK 0x7f000000 +#define SIS_WAVE_GENERAL_ARTICULATION 0x04 +#define SIS_WAVE_GENERAL_ARTICULATION_DELTA_MASK 0x3fff0000 +#define SIS_WAVE_ARTICULATION 0x08 +#define SIS_WAVE_TIMER 0x0c +#define SIS_WAVE_GENERATOR 0x10 +#define SIS_WAVE_CHANNEL_CONTROL 0x14 +#define SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE 0x80000000 +#define SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE 0x40000000 +#define SIS_WAVE_CHANNEL_CONTROL_FILTER_ENABLE 0x20000000 +#define SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE 0x10000000 +#define SIS_WAVE_LFO_EG_CONTROL 0x18 +#define SIS_WAVE_LFO_EG_CONTROL_2 0x1c +#define SIS_WAVE_LFO_EG_CONTROL_3 0x20 +#define SIS_WAVE_LFO_EG_CONTROL_4 0x24 + +#endif /* __sis7019_h__ */ diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index c6b4410..801bb48 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -705,6 +705,16 @@ config SND_RME9652 To compile this driver as a module, choose M here: the module will be called snd-rme9652. +config SND_SIS7019 + tristate "SiS 7019 Audio Accelerator" + depends on SND + select SND_AC97_CODEC + help + Say Y here to include support for the SiS 7019 Audio Accelerator. + + To compile this driver as a module, choose M here: the module + will be called snd-sis7019. + config SND_SONICVIBES tristate "S3 SonicVibes" depends on SND diff --git a/sound/pci/Makefile b/sound/pci/Makefile index cd76e02..1904a11 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -23,6 +23,7 @@ snd-intel8x0m-objs := intel8x0m.o snd-maestro3-objs := maestro3.o snd-rme32-objs := rme32.o snd-rme96-objs := rme96.o +snd-sis7019-objs := sis7019.o snd-sonicvibes-objs := sonicvibes.o snd-via82xx-objs := via82xx.o snd-via82xx-modem-objs := via82xx_modem.o @@ -48,6 +49,7 @@ obj-$(CONFIG_SND_INTEL8X0M) += snd-intel8x0m.o obj-$(CONFIG_SND_MAESTRO3) += snd-maestro3.o obj-$(CONFIG_SND_RME32) += snd-rme32.o obj-$(CONFIG_SND_RME96) += snd-rme96.o +obj-$(CONFIG_SND_SIS7019) += snd-sis7019.o obj-$(CONFIG_SND_SONICVIBES) += snd-sonicvibes.o obj-$(CONFIG_SND_VIA82XX) += snd-via82xx.o obj-$(CONFIG_SND_VIA82XX_MODEM) += snd-via82xx-modem.o diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c new file mode 100644 index 0000000..3a74f64 --- /dev/null +++ b/sound/pci/sis7019.c @@ -0,0 +1,1197 @@ +/* + * Driver for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <linux/time.h> +#include <linux/moduleparam.h> +#include <linux/interrupt.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/sis7019.h> + +MODULE_AUTHOR("David Dillow <dave(a)thedillows.org>"); +MODULE_DESCRIPTION("SiS 7019"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); + +static struct pci_device_id snd_sis7019_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, + { 0, } +}; + +MODULE_DEVICE_TABLE(pci, snd_sis7019_ids); + +/* There are three timing modes for the voices. + * + * For both playback and capture, when the buffer is two periods long, + * we use the hardware's built-in Mid-Loop Interrupt and End-Loop Interrupt + * to let us know when the periods have ended. + * + * When performing playback with more than two periods per buffer, we set + * the "Stop Sample Offset" and tell the hardware to interrupt us when we + * reach it. We then update the offset and continue on until we are + * interrupted for the next period. + * + * Capture channels do not have a SSO, so we allocate a playback channel to + * use as a timer for the capture periods. We use the End-Loop Interrupt + * to clock out the periods, and adjust the Loop End Offset of the timing + * channel to maintain synchronization. This algorithm came from the + * Trident driver. + * + * FIXME: The timing channel is configured to give quite a bit of + * attenuation (~96dB), but it may be possible to gain a bit more, or + * even better, play from a page of silence. + * + * FIXME: It'd be nice to make use of some of the synth features in the + * hardware, but a woeful lack of documentation is a significant roadblock. + */ +struct voice { + u16 flags; +#define VOICE_IN_USE 1 +#define VOICE_CAPTURE 2 +#define VOICE_SSO_TIMING 4 +#define VOICE_SYNC_TIMING 8 + u16 sync_cso; + u16 period_size; + u16 buffer_size; + u32 sso; + struct snd_pcm_substream *substream; + struct voice *timing; + void __iomem *ctrl_base; + void __iomem *wave_base; + void __iomem *sync_base; + int num; +}; + +struct sis7019 { + long unsigned ioport; + void __iomem *ioaddr; + int irq; + int codecs_present; + + struct pci_dev *pci; + struct snd_pcm *pcm; + struct snd_card *card; + struct snd_ac97 *ac97[3]; + + /* hw_lock protects access to the registers and wave engine parameters + * voice_lock protects allocation/freeing of the voice descriptions + */ + spinlock_t hw_lock; + spinlock_t voice_lock; + + struct voice voices[64]; + struct voice capture_voice; +}; + +#define SIS_PRIMARY_CODEC_PRESENT 0x0001 +#define SIS_SECONDARY_CODEC_PRESENT 0x0002 +#define SIS_TERTIARY_CODEC_PRESENT 0x0004 + +static struct snd_pcm_hardware sis_playback_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3ffdc, + .period_bytes_min = 8, + .period_bytes_max = 0x3ffdc, + .periods_min = 1, + .periods_max = 0x1ffe, +}; + +static struct snd_pcm_hardware sis_capture_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3ffdc, + .period_bytes_min = 8, + .period_bytes_max = 0x3ffdc, + .periods_min = 1, + .periods_max = 0x1ffe, +}; + +static void sis_update_end_offsets(struct voice *voice) +{ + void __iomem *base = voice->ctrl_base; + + if (voice->flags & VOICE_SSO_TIMING) { + /* Calculate the new SSO */ + voice->sso += voice->period_size; + if (voice->sso >= voice->buffer_size) + voice->sso -= voice->buffer_size; + + /* The SSO is in the upper 16 bits of the register. */ + writew(voice->sso & 0xffff, base + SIS_PLAY_DMA_SSO_ESO + 2); + } else if (voice->flags & VOICE_SYNC_TIMING) { + u16 cso = readw(voice->sync_base + SIS_CAPTURE_DMA_FORMAT_CSO); + int target = voice->sync_cso; + int sync; + u16 leo; + + /* Calculate our relative offset between the target and + * the actual CSO value. Since we're operating in a loop, + * if the value is more than half way around, we can + * consider ourselves wrapped. + */ + sync = cso; + sync -= target; + if (sync > (voice->buffer_size / 2)) + sync -= voice->buffer_size; + + /* If sync is negative, then we interrupted too early, and + * we'll need to come back in a few samples and try again. + * There's a minimum wait, as it takes some time for the DMA + * engine to startup, etc... + * + * FIXME: the first interrupt seems to come when we're at + * offset 0, so we have a huge early entry. Not a huge issue, + * but it would be nice if it didn't happen. + */ + if (sync < 0) { + leo = -sync; + if (leo < 16) + leo = 16; + writew(leo, base + SIS_PLAY_DMA_CONTROL); + return; + } + + /* Ok, we interrupted right on time, or (hopefully) just + * a bit late. We'll adjst our next waiting period based + * on how close we got. + * + * We need to stay just behind the actual channel to ensure + * it really is past a period when we get our interrupt -- + * otherwise we'll fall into the early code above and have + * minimum wait time, which could cause issues. + * + * But we don't want to be too late, as we don't want to + * cut into the user's time to refresh the buffers too + * much, especially if we're using small periods. + */ + if (sync < 12) + leo = voice->period_size + 2; + else + leo = voice->period_size - 7; + + sync = voice->sync_cso + voice->period_size; + if (sync >= voice->buffer_size) + sync -= voice->buffer_size; + + voice->sync_cso = sync; + writew(leo, base + SIS_PLAY_DMA_CONTROL); + } +} + +static irqreturn_t sis_interrupt(int irq, void *dev) +{ + struct sis7019 *sis = dev; + unsigned long io = sis->ioport; + struct voice *voice; + u32 intr, status, orig; + int bit; + + /* We only use the DMA interrupts, and we don't enable any other + * source of interrupts. But, it is possible to see an interupt + * status that didn't actually interrupt us, so eliminate anything + * we're not expecting to avoid falsely claiming an IRQ, and an + * ensuing endless loop. + */ + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + if (!intr) + return IRQ_NONE; + + spin_lock(&sis->hw_lock); + do { + status = inl(io + SIS_PISR_A); + if (status) { + orig = status; + voice = sis->voices; + while (status) { + bit = __ffs(status); + status >>= bit + 1; + voice += bit; + snd_pcm_period_elapsed(voice->substream); + sis_update_end_offsets(voice); + voice++; + } + outl(orig, io + SIS_PISR_A); + } + + status = inl(io + SIS_PISR_B); + if (status) { + orig = status; + voice = &sis->voices[32]; + while (status) { + bit = __ffs(status); + status >>= bit + 1; + voice += bit; + snd_pcm_period_elapsed(voice->substream); + sis_update_end_offsets(voice); + voice++; + } + outl(orig, io + SIS_PISR_B); + } + + status = inl(io + SIS_RISR); + if (status) { + voice = &sis->capture_voice; + if (!voice->timing) + snd_pcm_period_elapsed(voice->substream); + + outl(status, io + SIS_RISR); + } + + outl(intr, io + SIS_GISR); + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + } while (intr); + spin_unlock(&sis->hw_lock); + + return IRQ_HANDLED; +} + +static u32 sis_rate_to_delta(unsigned int rate) +{ + u32 delta; + + /* This was copied from the trident driver, but it seems its gotten + * around a bit... nevertheless, it works well. + * + * We special case 44100 and 8000 since rounding with the equation + * does not give us an accurate enough value. For 11025 and 22050 + * the equation gives us the best answer. All other frequencies will + * also use the equation. JDW + */ + if (rate == 44100) + delta = 0xeb3; + else if (rate == 8000) + delta = 0x2ab; + else if (rate == 48000) + delta = 0x1000; + else + delta = (((rate << 12) + 24000) / 48000) & 0x0000ffff; + return delta; +} + +static void sis_free_voice(struct sis7019 *sis, struct voice *voice) +{ + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->timing) { + voice->timing->flags = ~(VOICE_IN_USE | VOICE_SSO_TIMING | + VOICE_SYNC_TIMING); + voice->timing = NULL; + } + voice->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | VOICE_SYNC_TIMING); + spin_unlock_irqrestore(&sis->voice_lock, flags); +} + +static struct voice *sis_alloc_playback_voice(struct sis7019 *sis) +{ + struct voice *voice; + unsigned long flags; + int i; + + spin_lock_irqsave(&sis->voice_lock, flags); + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + if (voice->flags & VOICE_IN_USE) + continue; + voice->flags |= VOICE_IN_USE; + goto found_one; + } + voice = NULL; + +found_one: + spin_unlock_irqrestore(&sis->voice_lock, flags); + + return voice; +} + +static int sis_alloc_timing_voice(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + int needed; + + /* If the period size is exactly one half of the buffer, we don't + * need a timing voice, as we can use the middle loop interrupt + * to clock out the periods. + */ + needed = (params_period_size(hw_params) != + (params_buffer_size(hw_params) / 2)); + + if (needed && !voice->timing) { + voice->timing = sis_alloc_playback_voice(sis); + if (!voice->timing) + return -ENOMEM; + voice->timing->substream = substream; + } else if (!needed && voice->timing) { + sis_free_voice(sis, voice); + voice->timing = NULL; + } + + return 0; +} + +static int sis_playback_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice; + + voice = sis_alloc_playback_voice(sis); + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_playback_hw_info; + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_substream_close(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + + sis_free_voice(sis, voice); + return 0; +} + +static int sis_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int sis_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int sis_pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *ctrl_base = voice->ctrl_base; + void __iomem *wave_base = voice->wave_base; + u32 format, dma_addr, control, sso_eso, delta, reg; + u16 leo; + + /* Prepare the parameters for this channel before we take the lock. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format |= SIS_PLAY_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_PLAY_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_PLAY_DMA_FORMAT_MONO; + + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + if (runtime->period_size == (runtime->buffer_size / 2)) { + control = leo | SIS_PLAY_DMA_LOOP; + control |= SIS_PLAY_DMA_INTR_AT_LEO | SIS_PLAY_DMA_INTR_AT_MLP; + sso_eso = leo; + } else { + voice->flags |= VOICE_SSO_TIMING; + voice->sso = runtime->period_size; + voice->period_size = runtime->period_size; + voice->buffer_size = runtime->buffer_size; + + control = leo | SIS_PLAY_DMA_LOOP; + control |= SIS_PLAY_DMA_INTR_AT_SSO | SIS_PLAY_DMA_INTR_AT_LEO; + sso_eso = leo | (runtime->period_size - 1) << 16; + } + + delta = sis_rate_to_delta(runtime->rate); + + + /* Ok, we're ready to go, so take the lock and set up the channel. + */ + spin_lock_irq(&sis->hw_lock); + + writel(format, ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(dma_addr, ctrl_base + SIS_PLAY_DMA_BASE); + writel(control, ctrl_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, ctrl_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + + /* Force PCI writes to post. */ + readl(ctrl_base); + spin_unlock_irq(&sis->hw_lock); + + return 0; +} + +static int sis_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + unsigned long io = sis->ioport; + struct snd_pcm_substream *s; + struct voice *voice; + void *chip; + int starting; + u32 record = 0; + u32 play[2] = { 0, 0 }; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + starting = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + starting = 0; + break; + default: + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + /* Make sure it is for us... */ + chip = snd_pcm_substream_chip(s); + if (chip != sis) + continue; + + voice = s->runtime->private_data; + if (voice->flags & VOICE_CAPTURE) { + record |= 1 << voice->num; + voice = voice->timing; + } + + /* voice could be NULL if this a recording stream, and it + * doesn't have an external timing channel. + */ + if (voice) + play[voice->num / 32] |= 1 << (voice->num & 0x1f); + + snd_pcm_trigger_done(s, substream); + } + + spin_lock(&sis->hw_lock); + if (starting) { + if (record) + outl(record, io + SIS_RECORD_START_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_START_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_START_B_REG); + } else { + if (record) + outl(record, io + SIS_RECORD_STOP_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_STOP_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_STOP_B_REG); + } + spin_unlock(&sis->hw_lock); + return 0; +} + +static snd_pcm_uframes_t sis_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + u32 cso; + + cso = readl(voice->ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + cso &= 0xffff; + return cso; +} + +static int sis_capture_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = &sis->capture_voice; + unsigned long flags; + + /* FIXME: The driver only supports recording from one channel + * at the moment, but it could support more. + */ + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->flags & VOICE_IN_USE) + voice = NULL; + else + voice->flags |= VOICE_IN_USE; + spin_unlock_irqrestore(&sis->voice_lock, flags); + + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_capture_hw_info; + runtime->hw.rates = sis->ac97[0]->rates[AC97_RATES_ADC]; + snd_pcm_limit_hw_rates(runtime); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + int rc; + + rc = snd_ac97_set_rate(sis->ac97[0], AC97_PCM_LR_ADC_RATE, + params_rate(hw_params)); + if (rc) + goto out; + + rc = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (rc < 0) + goto out; + + rc = sis_alloc_timing_voice(substream, hw_params); + +out: + return rc; +} + +static int sis_pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + struct voice *timing = voice->timing; + void __iomem *rec_base = voice->ctrl_base; + void __iomem *play_base = NULL; + void __iomem *wave_base = NULL; + u32 format, dma_addr, rec_ctrl; + u16 leo; + + /* The following variables are only used if there is a timing channel. + */ + u32 uninitialized_var(timing_ctrl); + u32 uninitialized_var(sso_eso); + u32 uninitialized_var(delta); + + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_CAPTURE_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + rec_ctrl = leo | SIS_CAPTURE_DMA_LOOP; + if (timing) { + /* We're using a timing voice, so the capture voice will + * not need to generate interrupts. + */ + timing->flags |= VOICE_SYNC_TIMING; + timing->sync_cso = runtime->period_size; + timing->period_size = runtime->period_size; + timing->buffer_size = runtime->buffer_size; + timing->sync_base = voice->ctrl_base; + + timing_ctrl = leo + 16; + timing_ctrl |= SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_LEO; + sso_eso = leo; + + delta = sis_rate_to_delta(runtime->rate); + + play_base = timing->ctrl_base; + wave_base = timing->wave_base; + } else { + /* We've just got two periods, so we can use the capture + * voice's interrupt generation. + */ + rec_ctrl |= SIS_CAPTURE_DMA_INTR_AT_MLP; + rec_ctrl |= SIS_CAPTURE_DMA_INTR_AT_LEO; + } + + /* We're ready to program the channel(s), so take the lock and go. + */ + spin_lock_irq(&sis->hw_lock); + + writel(format, rec_base + SIS_CAPTURE_DMA_FORMAT_CSO); + writel(dma_addr, rec_base + SIS_CAPTURE_DMA_BASE); + writel(rec_ctrl, rec_base + SIS_CAPTURE_DMA_CONTROL); + + if (play_base) { + u32 reg; + + writel(format, play_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(dma_addr, play_base + SIS_PLAY_DMA_BASE); + writel(timing_ctrl, play_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, play_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + /* We don't want the timing channel to generate any sound, + * so set it to maximum attenuation (31.75 dB), and run it + * through the music amplifier, which has been set to add + * an additional 64 dB of attenuation. That will be 95.76 dB + * total. + * + * FIXME: While this works well in practice, purists may + * wish to investigate using the mixer lists to further + * attenuate the timing channel -- or set aside a page + * of zeros for silence. This may also free up the music + * amplifier for other purposes. + */ + writel(SIS_WAVE_GENERAL_MUSIC_VOLUME | (0x7f << 24), + wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + } + + readl(rec_base + 0); + + spin_unlock_irq(&sis->hw_lock); + return 0; +} + +static struct snd_pcm_ops sis_playback_ops = { + .open = sis_playback_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_playback_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_playback_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static struct snd_pcm_ops sis_capture_ops = { + .open = sis_capture_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_capture_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_capture_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static int __devinit sis_pcm_create(struct sis7019 *sis) +{ + struct snd_pcm *pcm; + int rc; + + rc = snd_pcm_new(sis->card, "SiS 7019", 0, 40, 1, &pcm); + if (rc) + return rc; + + pcm->private_data = sis; + strcpy(pcm->name, "SiS 7019"); + sis->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &sis_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &sis_capture_ops); + + /* Try to preallocate some memory, but it's not the end of the + * world if this fails. + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(sis->pci), 64*1024, 128*1024); + + return 0; +} + +static unsigned short sis_ac97_rw(struct sis7019 *sis, int codec, u32 cmd) +{ + unsigned long io = sis->ioport; + unsigned long flags; + unsigned short val = 0xffff; + u16 status; + u16 rdy; + int count; + const static u16 codec_ready[3] = { + SIS_AC97_STATUS_CODEC_READY, + SIS_AC97_STATUS_CODEC2_READY, + SIS_AC97_STATUS_CODEC3_READY, + }; + + rdy = codec_ready[codec]; + + spin_lock_irqsave(&sis->hw_lock, flags); + + /* Get the AC97 semaphore ... + */ + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + goto timeout; + + /* ... and wait for any outstanding commands to complete ... + */ + count = 0xffff; + do { + status = inw(io + SIS_AC97_STATUS); + if ((status & rdy) && !(status & SIS_AC97_STATUS_BUSY)) + break; + + udelay(1); + } while (--count); + + if (!count) + goto timeout_sema; + + /* ... before sending our command and waiting for it to finish ... + */ + outl(cmd, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* ... and reading the results (if any). + */ + val = inl(io + SIS_AC97_CMD) >> 16; + +timeout_sema: + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); +timeout: + spin_unlock_irqrestore(&sis->hw_lock, flags); + + if (!count) { + printk(KERN_ERR "sis7019: ac97 codec %d timeout cmd 0x%08x\n", + codec, cmd); + } + + return val; +} + +static void sis_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_WRITE, + SIS_AC97_CMD_CODEC2_WRITE, + SIS_AC97_CMD_CODEC3_WRITE, + }; + sis_ac97_rw(ac97->private_data, ac97->num, + (val << 16) | (reg << 8) | cmd[ac97->num]); +} + +static unsigned short sis_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_READ, + SIS_AC97_CMD_CODEC2_READ, + SIS_AC97_CMD_CODEC3_READ, + }; + return sis_ac97_rw(ac97->private_data, ac97->num, + (reg << 8) | cmd[ac97->num]); +} + +static int __devinit sis_mixer_create(struct sis7019 *sis) +{ + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + static struct snd_ac97_bus_ops ops = { + .write = sis_ac97_write, + .read = sis_ac97_read, + }; + int rc; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = sis; + + rc = snd_ac97_bus(sis->card, 0, &ops, NULL, &bus); + if (!rc && sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[0]); + ac97.num = 1; + if (!rc && (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[1]); + ac97.num = 2; + if (!rc && (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[2]); + + /* If we return an error here, then snd_card_free() should + * free up any ac97 codecs that got created, as well as the bus. + */ + return rc; +} + +static int sis_chip_free(struct sis7019 *sis) +{ + /* Reset the chip, and disable all interrputs. + */ + outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); + udelay(10); + outl(0, sis->ioport + SIS_GCR); + outl(0, sis->ioport + SIS_GIER); + + /* Now, free everything we allocated. + */ + if (sis->irq >= 0) + free_irq(sis->irq, sis); + + if (sis->ioaddr) + iounmap(sis->ioaddr); + + pci_release_regions(sis->pci); + pci_disable_device(sis->pci); + + return 0; +} + +static int sis_dev_free(struct snd_device *dev) +{ + struct sis7019 *sis = dev->device_data; + return sis_chip_free(sis); +} + +static int sis_chip_init(struct sis7019 *sis) +{ + unsigned long io = sis->ioport; + void __iomem *ioaddr = sis->ioaddr; + u16 status; + int count; + int i; + + /* Reset the audio controller + */ + outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); + udelay(10); + outl(0, io + SIS_GCR); + + /* Get the AC-link semaphore, and reset the codecs + */ + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + return -EIO; + + outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* Now that we've finished the reset, find out what's attached. + */ + status = inl(io + SIS_AC97_STATUS); + if (status & SIS_AC97_STATUS_CODEC_READY) + sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC2_READY) + sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC3_READY) + sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; + + /* All done, let go of the semaphore, and check for errors + */ + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); + if (!sis->codecs_present || !count) + return -EIO; + + /* Let the hardware know that the audio driver is alive, + * and enable PCM slots on the AC-link for L/R playback (3 & 4) and + * record channels. We're going to want to use Variable Rate Audio + * for recording, to avoid needlessly resampling from 48kHZ. + */ + outl(SIS_AC97_CONF_AUDIO_ALIVE, io + SIS_AC97_CONF); + outl(SIS_AC97_CONF_AUDIO_ALIVE | SIS_AC97_CONF_PCM_LR_ENABLE | + SIS_AC97_CONF_PCM_CAP_MIC_ENABLE | + SIS_AC97_CONF_PCM_CAP_LR_ENABLE | + SIS_AC97_CONF_CODEC_VRA_ENABLE, io + SIS_AC97_CONF); + + /* All AC97 PCM slots should be sourced from sub-mixer 0. + */ + outl(0, io + SIS_AC97_PSR); + + /* There is only one valid DMA setup for a PCI environment. + */ + outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR); + + /* Reset the syncronization groups for all of the channels + * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc. + * we'll need to change how we handle these. Until then, we just + * assign sub-mixer 0 to all playback channels, and avoid any + * attenuation on the audio. + */ + outl(0, io + SIS_PLAY_SYNC_GROUP_A); + outl(0, io + SIS_PLAY_SYNC_GROUP_B); + outl(0, io + SIS_PLAY_SYNC_GROUP_C); + outl(0, io + SIS_PLAY_SYNC_GROUP_D); + outl(0, io + SIS_MIXER_SYNC_GROUP); + + for (i = 0; i < 64; i++) { + writel(i, SIS_MIXER_START_ADDR(ioaddr, i)); + writel(SIS_MIXER_RIGHT_NO_ATTEN | SIS_MIXER_LEFT_NO_ATTEN | + SIS_MIXER_DEST_0, SIS_MIXER_ADDR(ioaddr, i)); + } + + /* Don't attenuate any audio set for the wave amplifier. + * Do maximum attenuation on audio set to use the music amplifier, + * as we'll set any timing channels to use that. + */ + outl(0xffff0000, SIS_WEVCR); + + /* Go ahead and enable the DMA interrupts. They won't go live + * until we start a channel. + */ + outl(SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE | + SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE, io + SIS_GIER); + + return 0; +} + +static int __devinit sis_chip_create(struct snd_card *card, + struct pci_dev *pci) +{ + struct sis7019 *sis = card->private_data; + struct voice *voice; + static struct snd_device_ops ops = { + .dev_free = sis_dev_free, + }; + int rc; + int i; + + rc = pci_enable_device(pci); + if (rc) + goto error_out; + + /* FIXME: Not sure of our DMA limit, use 32 bit mask for now. + */ + if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) { + printk(KERN_ERR "sis7019: architecture does not support " + "32-bit PCI busmaster DMA"); + goto error_out_enabled; + } + + memset(sis, 0, sizeof(*sis)); + spin_lock_init(&sis->hw_lock); + spin_lock_init(&sis->voice_lock); + sis->card = card; + sis->pci = pci; + sis->irq = -1; + sis->ioport = pci_resource_start(pci, 0); + + rc = pci_request_regions(pci, "SiS 7019"); + if (rc) { + printk(KERN_ERR "sis7019: unable request regions\n"); + goto error_out_enabled; + } + + rc = -EIO; + sis->ioaddr = ioremap(pci_resource_start(pci, 1), 0x4000); + if (!sis->ioaddr) { + printk(KERN_ERR "sis7019: unable to remap MMIO, aborting\n"); + goto error_out_cleanup; + } + + rc = sis_chip_init(sis); + if (rc) + goto error_out_cleanup; + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + "SiS 7019", sis)) { + printk(KERN_ERR "unable to allocate irq %d\n", sis->irq); + goto error_out_cleanup; + } + + sis->irq = pci->irq; + pci_set_master(pci); + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + voice->num = i; + voice->ctrl_base = SIS_PLAY_DMA_ADDR(sis->ioaddr, i); + voice->wave_base = SIS_WAVE_ADDR(sis->ioaddr, i); + } + + voice = &sis->capture_voice; + voice->flags = VOICE_CAPTURE; + voice->num = SIS_CAPTURE_CHAN_AC97_PCM_IN; + voice->ctrl_base = SIS_CAPTURE_DMA_ADDR(sis->ioaddr, voice->num); + + rc = snd_device_new(card, SNDRV_DEV_LOWLEVEL, sis, &ops); + if (rc) + goto error_out_cleanup; + + snd_card_set_dev(card, &pci->dev); + + return 0; + +error_out_cleanup: + sis_chip_free(sis); + +error_out_enabled: + pci_disable_device(pci); + +error_out: + return rc; +} + +static int __devinit snd_sis7019_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct snd_card *card; + struct sis7019 *sis; + static int dev; + int rc; + + rc = -ENODEV; + if (dev >= SNDRV_CARDS) + goto error_out; + + rc = -ENOENT; + if (!enable[dev]) { + dev++; + goto error_out; + } + + rc = -ENOMEM; + card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(*sis)); + if (!card) + goto error_out; + + rc = sis_chip_create(card, pci); + if (rc) + goto card_error_out; + + sis = card->private_data; + strcpy(card->driver, "SiS 7019"); + strcpy(card->shortname, "SiS 7019"); + + rc = sis_mixer_create(sis); + if (rc) + goto card_error_out; + + rc = sis_pcm_create(sis); + if (rc) + goto card_error_out; + + snprintf(card->longname, sizeof(card->longname), + "%s Audio Accelerator with %s at 0x%lx, irq %d", + card->shortname, snd_ac97_get_short_name(sis->ac97[0]), + sis->ioport, sis->irq); + + rc = snd_card_register(card); + if (rc) + goto card_error_out; + + pci_set_drvdata(pci, card); + dev++; + return 0; + +card_error_out: + snd_card_free(card); + +error_out: + return rc; +} + +static void __devexit snd_sis7019_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static struct pci_driver sis7019_driver = { + .name = "SiS7019", + .id_table = snd_sis7019_ids, + .probe = snd_sis7019_probe, + .remove = __devexit_p(snd_sis7019_remove), + + /* FIXME: need to support power management */ +}; + +static int __init sis7019_init(void) +{ + return pci_register_driver(&sis7019_driver); +} + +static void __exit sis7019_exit(void) +{ + pci_unregister_driver(&sis7019_driver); +} + +module_init(sis7019_init); +module_exit(sis7019_exit); -- 1.5.0.6

3 5

Re: [alsa-devel] different user causes 'no such device'
by william estrada 09 Oct '07

09 Oct '07

Thanks, I will try this. William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum Takashi Iwai wrote: > At Wed, 03 Oct 2007 12:55:27 -0700, > william estrada wrote: >> Hi guys, >> >> I have what seams like a interesting problem. I have written an ALSA app >> to record wav files and it runs while I am logged in an terminal as the >> logged in Gnome user or root. But when I su to another user, it can't >> find the default sound card. >> >> I am running FC6. I tried changing own of /dev/dsp to 'nobody', that did >> not work. >> >> Anyone know of this problem and a fix for it. >> >> Thanks for your time. >> >> -- >> William Estrada >> MrUmunhum(a)popdial.com >> Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) >> Ymessenger: MrUmunhum >> >>> MSGrec - Version 1.1, Mt-Umunhum-Wireless.net >>> WavMSG: Paused: enter 'q' to stop, <ENTER> to start. >>> ALSA lib confmisc.c:848:(snd_func_card_driver) cannot find card '0' >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device >>> ALSA lib confmisc.c:397:(snd_func_concat) error evaluating strings >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_concat returned error: No such device >>> ALSA lib confmisc.c:1248:(snd_func_refer) error evaluating name >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_refer returned error: No such device >>> ALSA lib conf.c:3972:(snd_config_expand) Evaluate error: No such device >>> ALSA lib pcm.c:2145:(snd_pcm_open_noupdate) Unknown PCM default > > ALSA-native apps access only ALSA device files, usually /dev/snd/*, > not /dev/dsp*, those for OSS. In this case, it should be > /dev/snd/controlC* files. Check the premission of these files. > > > Takashi >

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[alsa-devel] ALSA Support for Maya44 PCI?
by Nicolas Rodriguez 09 Oct '07

09 Oct '07

I bought a Maya44 PCI 4-channel soundcard and I'll like to use it in Linux. I couldn't get it to work with Ubuntu (I don't remember the ALSA version, sorry). Do you support this card? If you don't, I can create the driver for it. I've read somewhere that ESI-PRO can give the card specifications to developers interested in build drivers for their cards. I've sent them an email asking for the card specifications for I've got no response. What do you recommend? Thanks in advance, Nico.

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[alsa-devel] Envy24control s/pdif input bugfix
by Stephan Zaubzer 09 Oct '07

09 Oct '07

Hi! This patch against the stable version of envy24control fixes a bug where the S/Pdif input switch doesn't work on Terratec DMX 6fire cards. This bug first appeared in version 1.0.11rc3. diff -ubr alsa-tools-1.0.14patch/envy24control/hardware.c alsa-tools-1.0.14/envy24control/hardware.c --- alsa-tools-1.0.14patch/envy24control/hardware.c 2007-09-23 15:58:53.000000000 +0200 +++ alsa-tools-1.0.14/envy24control/hardware.c 2007-06-04 13:27:25.000000000 +0200 @@ -775,7 +775,7 @@ snd_ctl_elem_value_set_interface(volume_rate, SND_CTL_ELEM_IFACE_MIXER); snd_ctl_elem_value_set_name(volume_rate, "Multi Track Volume Rate"); - if (card_is_dmx6fire) { + if (! card_is_dmx6fire) { snd_ctl_elem_value_set_interface(spdif_input, SND_CTL_ELEM_IFACE_MIXER); snd_ctl_elem_value_set_name(spdif_input, "Optical Digital Input Switch"); } else { Regards Stephan

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Re: [alsa-devel] [Alsa-user] Asus M2NPV-VM snd_hda_intel problem - also with Alsa 1.0.15rc3
by Ma Begaj 09 Oct '07

09 Oct '07

> Firstly, I don't have the same board. Similar, but different. > > I have an Asus A8N-VM CSM board, but it has the same AD1986A chip on it. > I didn't have the same issue with sound muting, but I did have issues > with surround (it just didn't work). I don't remember the details of how > > > --- patch_analog.c.backup 2007-04-14 11:01:17.209146249 +0100 > +++ patch_analog.c 2007-04-14 13:37:47.005206476 +0100 > @@ -895,14 +895,14 @@ > case AD1986A_3STACK: > spec->num_mixers = 2; > spec->mixers[1] = ad1986a_3st_mixers; > - spec->num_init_verbs = 3; > +/* spec->num_init_verbs = 3; > spec->init_verbs[1] = ad1986a_3st_init_verbs; > - spec->init_verbs[2] = ad1986a_ch2_init; > + spec->init_verbs[2] = ad1986a_ch2_init; */ > spec->channel_mode = ad1986a_modes; > spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes); > spec->need_dac_fix = 1; > - spec->multiout.max_channels = 2; > - spec->multiout.num_dacs = 1; > + spec->multiout.max_channels = 6; > + spec->multiout.num_dacs = 3; > break; > case AD1986A_LAPTOP: > spec->mixers[0] = ad1986a_laptop_mixers; > thanks for your try. it helped partly ... I tried 3 patches (incl. yours) and only yours helped partly at least. I patched Alsa 1.0.15rc3. Sound mutes as soon as I change volume, and I can only rmmod/modprobe snd-hda-intel to get it back. But Surround is working with your patch as I can see: "speaker-test -Dplug:surround51 -c6 -t waw" recognizes correctly all speakears. "speaker-test -c6 -t waw" does not recognize the speakers correctly, but it probably should not, or ?! MythTv is playing stereo files only on the right speakers. I do not now is this a MythTv or Alsa problem. probably MythTv and I should create a .asoundrc which should correct this somehow. I did not try hg, because I thought that Alsa 1.0.15rc3 is very new. Thanks. M.

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[alsa-devel] Capturing of Samsung Q1 Ultra (ALC262) does not work
by Pengcheng Zou 09 Oct '07

09 Oct '07

Hi, capture function of the Samsung Q1 Ultra (using ALC262 codec) does not work. I tried the most recent ALSA build (alsa-driver-1.0.15rc3), the playback works well, however the recording does not work. Samsung Q1 (using AD1986A codec) has been supported already, wonder if there is any patches for Q1 Ultra? please note Samsung released 2 types of UPMC, Q1 and Q1 Ultra. the Q1 ultra is the one I have: http://computers.pricegrabber.com/tablet-pcs/m/37616779/details/ TIA, -- Pengcheng

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[alsa-devel] snd_pcm_drain and lock-ups
by Mike Gorse 08 Oct '07

08 Oct '07

Hi all, I've followed Arve Knudsen over here. I am getting a lock-up when calling snd_pcm_drain(). This happens when using espeak, which is using portaudio. Portaudio calls snd_pcm_drain while trying to recover from a XRUN. snd_pcm_drain is calling snd_pcm_rate_drain in my case. snd_pcm_rate_drain then calls snd_pcm_wait. snd_pcm_wait first checks if snd_pcm_mmap_avail(pcm) >= pcm->avail_min. snd_pcm_map_avail is returning a number less than 300, often between 135 and 140 but not always, where pcm->avail_min == 1024. So it does not check for the XRUN state since it is conditional on snd_pcm_mmap_avail returning >= 1024. This is on my laptop, which is running Linux 2.6.22 and alsa-lib 1.0.15-rc1 and uses the snd-intel8x0 driver. Does this give people enough information to tell what might be going on? I have a small test program that causes the lock-up on my laptop where the rate plugin gets used (it sometimes locks up immediately, sometimes after a few seconds), but it needs espeak and portaudio to be installed. Thanks, -- Mike Gorse / AIM:linvortex / http://mgorse.freeshell.org --

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Re: [alsa-devel] [Alsa-user] How to get set/fixed sample rate of ALSA device?
by Rene Herman 08 Oct '07

08 Oct '07

On 10/08/2007 12:41 AM, Mark Constable wrote: > Any chance that your program could take an arg to investigate > other cards (or perhaps plugins) and also return an errno so > it could be used in scripts ? Only if you promise to never drop CCs again (although I don't know about plugins). > # gcc -W -Wall -o snd_rate snd_rate.c -l asound > snd_rate.c: In function 'main': > snd_rate.c:14: warning: the address of ‘params’ will always evaluate as 'true' That's the ALSA snd_pcm_hw_params_alloca(). No such warning here, but I've made it a regular malloc. > # snd_rate > snd_pcm_open: Device or resource busy > > In fact, why isn't there a single simple "alsainfo" type > program that tells a user some (or a lot) of info about > their hardware and alsa-lib install ? I guess because noone wrote one :-) It seems by the way that dir != 0 _can_ be returned for hw devices, and this just ignores it. Just some rounding issue I guess. Rene. /* gcc -W -Wall -o alsa-rate alsa-rate.c -lasound */ #include <stdio.h> #include <alsa/asoundlib.h> int main(int argc, char *argv[]) { char *device; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int min; unsigned int max; int dir; int err; device = argc > 1 ? argv[1] : "hw:0,0"; err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0); if (err < 0) { fprintf(stderr, "snd_pcm_open: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_malloc(&params); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_alloc: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_any(handle, params); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_any: %s\n", snd_strerror(err)); return err; } err = snd_pcm_close(handle); if (err < 0) { fprintf(stderr, "snd_pcm_close: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_rate_min(params, &min, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_min: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_rate_max(params, &max, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_max: %s\n", snd_strerror(err)); return err; } snd_pcm_hw_params_free(params); printf("%u-%u\n", min, max); return 0; }

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[alsa-devel] Use-before-initialization in alsa-oss
by Alexander E. Patrakov 06 Oct '07

06 Oct '07

dsp->oss_format was used before it was set correctly spotted by running "aoss flite_time 10:00" Signed-Off-By: Alexander E. Patrakov --- alsa-oss-1.0.14.orig/alsa/pcm.c 2007-09-29 16:12:44.000000000 +0600 +++ alsa-oss-1.0.14/alsa/pcm.c 2007-09-29 16:13:06.000000000 +0600 @@ -215,10 +215,10 @@ unsigned int rate, periods_min; if (!pcm) continue; + dsp->format = oss_format_to_alsa(dsp->oss_format); str->frame_bytes = snd_pcm_format_physical_width(dsp->format) * dsp->channels / 8; snd_pcm_hw_params_alloca(&hw); snd_pcm_hw_params_any(pcm, hw); - dsp->format = oss_format_to_alsa(dsp->oss_format); err = snd_pcm_hw_params_set_format(pcm, hw, dsp->format); if (err < 0) -- Alexander E. Patrakov

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[alsa-devel] ASoC v2 on PowerPC
by Jon Smirl 05 Oct '07

05 Oct '07

I'm looking into write a ASoC v2 driver for the Efika. Efika uses a Freescale MPC5200B and a STAC9766. This is a practice driver before writing one for our target hardware (which doesn't work yet, same CPU but different codec). I've brought the changes from wolfson/asoc-v2-dev into my powerpc tree and got a couple of config errors with ARCH=powerpc. sound/soc/pxa/Kconfig:23:warning: 'select' used by config symbol 'SND_PXA2XX_SOC_SSP' refers to undefined symbol 'PXA_SSP' sound/soc/au1x/Kconfig:35:warning: 'select' used by config symbol 'SND_SOC_EXM1200_SSI' refers to undefined symbol 'SND_SOC_CS4251X' Suggestions on the best way to start? Target hardware also has a WM8804 on it. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] Why was the built-in software volume control removed?
by kyle cronan 04 Oct '07

04 Oct '07

I think I may have posted this to the wrong list. There wasn't any response yet on alsa-user, so I'm reposting here. Summary: I'm seeing a lot of glitches ever since the built-in software volume was removed. What was the reason for this change? Softvol doesn't work nearly as well for me. ________________ Hi everyone, I recently updated my media center which is running Fedora devel. It pulled down a new kernel with updated ALSA drivers for my HDA audio chip (Intel 82801H + ALC883). I use IEC958 output to a pair of studio monitors (good sound with minimal fuss). The first thing I noticed about the update was that my software volume control was no longer saving state. It was resetting to something around 80% every time an app opened the audio device, a level which, in my setup, is quite loud! Then I saw that in alsamixer I had no master volume control, not even a PCM control! So I started experimented with my settings, and I found that after I selected "PCM" under "Default Mixer Tracks" in gnome-sound-properties I had a PCM volume control again (still no master), and apps using alsa output would once again remember the system volume. Don't know what that gnome setting actually did, but I've not been able to get it back to the original state, even when I ctrl-click to deselect "PCM" in the list. However, apps that use legacy OSS output still blast me with nearly full volume every time. I've also noticed another glitch: using the volume control applet to adjust my PCM volume, the audio sometimes skips. If I quickly drag the slider back and forth, it always does. I'm pretty sure I would have noticed this if it occurred with my previous kernel (haven't had a chance to test that yet). Anyway, looking into this problem, I found the following note in the changes document at http://www.alsa-project.org/main/index.php/Changes_v1.0.15rc2_v1.0.15rc3 Sound Core ... - Remove the built-in software volume control in favour of softvol. So, my (main) question is: is there a good reason that the built-in software volume was removed? I didn't find anything about it in the archives for this list or -devel. I'm really not in favor of this change, because It used to work flawlessly for me, whereas now I'm running into all these strange problems with my IEC958 output. Thanks, Kyle Cronan <kyle(a)pbx.org>

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[alsa-devel] Tascam US122 on Arm
by ST 04 Oct '07

04 Oct '07

Hi I am a little irritated that there has been no replies to my questions to this list (http://mailman.alsa-project.org/pipermail/alsa-devel/2007-September/003387.…, http://mailman.alsa-project.org/pipermail/alsa-devel/2007-September/003083.…). Am i on the wrong list? Please tell me if so. Probably its also that i didn't make clear in my last post that the tascam driver works on arm with my US122 sound card, for ca. 51 seconds. After that i get the following error: Sequence Error!(hcd_frame=6 ep=8in;wait=65538,frame=2). Most propably some urb of usb-frame 65538 is still missing. Cause could be too long delays in usb-hcd interrupt handling. I also tried insmod ./2.6.18/kernel/sound/usb/usx2y/snd-usb-usx2y.ko nrpacks=X with X=[1,2,4,8]. I thought that if it is really an CPU performance problem than probably a bigger buffer would eliminate the problem. However the error persisted. Even commenting out /* #define USX2Y_NRPACKS_VARIABLE y */ and setting #define USX2Y_NRPACKS 1 didn't help. So my question is now, is the 250Mhz ARM really to slow or is there any chance that there is an error in either the usb stack or the driver exposed only on the ARM architecture? Things speaking for the latter are: *Error allways in the 65536-65540 value. *Error happens most of the times after about 50secs. Since i do not know much about USB any hint is apreciated. ST

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[alsa-devel] about sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?)
by Akio 04 Oct '07

04 Oct '07

Dear ALL: after long time amr-wb codec loopback test. (above 90 hours) I got the following kernel message. it comes from snd_pcm_lib_read1() function. ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) What should I do? My platform is ARM926, with Kernel 2.6.19.2 ALSA version is 1.0.13. Best Regards, Akio

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[alsa-devel] different user causes 'no such device'
by william estrada 03 Oct '07

03 Oct '07

Hi guys, I have what seams like a interesting problem. I have written an ALSA app to record wav files and it runs while I am logged in an terminal as the logged in Gnome user or root. But when I su to another user, it can't find the default sound card. I am running FC6. I tried changing own of /dev/dsp to 'nobody', that did not work. Anyone know of this problem and a fix for it. Thanks for your time. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum > MSGrec - Version 1.1, Mt-Umunhum-Wireless.net > WavMSG: Paused: enter 'q' to stop, <ENTER> to start. > ALSA lib confmisc.c:848:(snd_func_card_driver) cannot find card '0' > ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device > ALSA lib confmisc.c:397:(snd_func_concat) error evaluating strings > ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_concat returned error: No such device > ALSA lib confmisc.c:1248:(snd_func_refer) error evaluating name > ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_refer returned error: No such device > ALSA lib conf.c:3972:(snd_config_expand) Evaluate error: No such device > ALSA lib pcm.c:2145:(snd_pcm_open_noupdate) Unknown PCM default

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