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);

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[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

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[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

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[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

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[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?

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[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.

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