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

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[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

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