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[alsa-devel] [PATCH 001a/003] snd-powermac: enable headphone detection
by Risto Suominen 16 Apr '08

16 Apr '08

From: Risto Suominen <Risto.Suominen(a)gmail.com> Enable port change interrupt while initialising AWACS, Screamer, and Burgundy chipsets. Signed-off-by: Risto Suominen <Risto.Suominen(a)gmail.com> --- Kernel version 2.6.25-rc8 (probably any 2.6) --- sound/ppc/pmac.c.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/pmac.c 2008-04-07 12:18:47.503077312 +0300 @@ -1300,9 +1304,9 @@ int __init snd_pmac_new(struct snd_card snd_pmac_sound_feature(chip, 1); - /* reset */ - if (chip->model == PMAC_AWACS) - out_le32(&chip->awacs->control, 0x11); + /* reset & enable interrupts */ + if (chip->model <= PMAC_BURGUNDY) + out_le32(&chip->awacs->control, chip->control_mask); /* Powerbooks have odd ways of enabling inputs such as an expansion-bay CD or sound from an internal modem

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[alsa-devel] [Fwd: Another unknown bt878]
by John R. 16 Apr '08

16 Apr '08

This card is sold as the PV-981. I tried various digital_rate settings. -------- Original Message -------- Subject: Another unknown bt878 Date: Mon, 14 Apr 2008 23:17:47 -0500 From: John R. <johnr(a)wowway.com> To: alsa-devel(a)alsa-project.org Sound doesn't work on my bt878 based card. Let me know what other information is needed... kernel 2.6.24.2: Apr 14 20:28:38 server kernel: unknown card 0x878-0xaa00:0x1460 Apr 14 20:28:38 server kernel: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> Apr 14 20:28:38 server kernel: ACPI: PCI Interrupt 0000:05:00.1[A] -> GSI 18 (level, low) -> IRQ 18 Apr 14 20:28:38 server kernel: bt87x0: Using board 0, analog, digital (rate 32000 Hz) Apr 14 20:28:38 server kernel: unknown card 0x878-0xaa01:0x1461 Apr 14 20:28:38 server kernel: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> Apr 14 20:28:38 server kernel: ACPI: PCI Interrupt 0000:05:01.1[A] -> GSI 19 (level, low) -> IRQ 19 Apr 14 20:28:38 server kernel: bt87x1: Using board 0, analog, digital (rate 32000 Hz) Apr 14 20:28:38 server kernel: unknown card 0x878-0xaa02:0x1462 Apr 14 20:28:38 server kernel: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> Apr 14 20:28:38 server kernel: ACPI: PCI Interrupt 0000:05:02.1[A] -> GSI 16 (level, low) -> IRQ 16 Apr 14 20:28:38 server kernel: bt87x2: Using board 0, analog, digital (rate 3200 0 Hz) Apr 14 20:28:38 server kernel: unknown card 0x878-0xaa03:0x1463 Apr 14 20:28:38 server kernel: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> Apr 14 20:28:38 server kernel: ACPI: PCI Interrupt 0000:05:03.1[A] -> GSI 17 (level, low) -> IRQ 17 Apr 14 20:28:38 server kernel: bt87x3: Using board 0, analog, digital (rate 32000 Hz) lspci: 05:00.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) 05:00.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) 05:01.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) 05:01.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) 05:02.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) 05:02.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) 05:03.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) 05:03.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) at boot: bttv: driver version 0.9.17 loaded bttv: using 8 buffers with 2080k (520 pages) each for capture bttv: Bt8xx card found (0). ACPI: PCI Interrupt 0000:05:00.0[A] -> GSI 18 (level, low) -> IRQ 18 bttv0: Bt878 (rev 17) at 0000:05:00.0, irq: 18, latency: 64, mmio: 0xdfcfe000 bttv0: detected: Provideo PV150A-1 [card=98], PCI subsystem ID is aa00:1460 bttv0: using: ProVideo PV150 [card=98,autodetected] bttv0: gpio: en=00000000, out=00000000 in=00ffffff [init] bttv0: tuner absent bttv0: registered device video0 bttv0: registered device vbi0 bttv0: PLL: 28636363 => 35468950 .. ok bttv: Bt8xx card found (1). ACPI: PCI Interrupt 0000:05:01.0[A] -> GSI 19 (level, low) -> IRQ 19 bttv1: Bt878 (rev 17) at 0000:05:01.0, irq: 19, latency: 64, mmio: 0xdfcfc000 bttv1: detected: Provideo PV150A-2 [card=98], PCI subsystem ID is aa01:1461 bttv1: using: ProVideo PV150 [card=98,autodetected] bttv1: gpio: en=00000000, out=00000000 in=00ffffff [init] bttv1: tuner absent bttv1: registered device video1 bttv1: registered device vbi1 bttv1: PLL: 28636363 => 35468950 .. ok bttv: Bt8xx card found (2). ACPI: PCI Interrupt 0000:05:02.0[A] -> GSI 16 (level, low) -> IRQ 16 bttv2: Bt878 (rev 17) at 0000:05:02.0, irq: 16, latency: 64, mmio: 0xdfcfa000 bttv2: detected: Provideo PV150A-3 [card=98], PCI subsystem ID is aa02:1462 bttv2: using: ProVideo PV150 [card=98,autodetected] bttv2: gpio: en=00000000, out=00000000 in=00ffffff [init] bttv2: tuner absent bttv2: registered device video2 bttv2: registered device vbi2 bttv2: PLL: 28636363 => 35468950 .. ok bttv: Bt8xx card found (3). ACPI: PCI Interrupt 0000:05:03.0[A] -> GSI 17 (level, low) -> IRQ 17 bttv3: Bt878 (rev 17) at 0000:05:03.0, irq: 17, latency: 64, mmio: 0xdfcf8000 bttv3: detected: Provideo PV150A-4 [card=98], PCI subsystem ID is aa03:1463 bttv3: using: ProVideo PV150 [card=98,autodetected] bttv3: gpio: en=00000000, out=00000000 in=00ffffff [init] bttv3: tuner absent bttv3: registered device video3 bttv3: registered device vbi3 bttv3: PLL: 28636363 => 35468950 .. ok

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[alsa-devel] snd_ctl_elem_write fails -EPERM if snd_pcm is open
by Bouterse, Bob 15 Apr '08

15 Apr '08

I am trying to sent IEC958 status bits on a VT1708A HAD-INTEL codec while audio is being output. snd_ctl_elem_write returns -EPERM from linux-2.6.24.3/sound/core/control.c:769 . Is there some way I can do this? FYI using iecset will also give you an: # iecset copywrite on snd_ctl_elem_write: Operation not permitted I am using alsa-lib-1.0.14a $uname -a Linux vision-vs 2.6.24.3 #1 SMP Fri Mar 28 09:35:45 EDT 2008 i686 unknown $ gcc -v Target: x86_64-linux-gnu Configured with: ../src/configure -v --enable-languages=c,c++,fortran,objc,obj-c++,treelang --prefix=/usr --enable-shared --with-system-zlib --libexecdir=/usr/lib --without-included-gettext --enable-threads=posix --enable-nls --with-gxx-include-dir=/usr/include/c++/4.2 --program-suffix=-4.2 --enable-clocale=gnu --enable-libstdcxx-debug --enable-mpfr --enable-checking=release --build=x86_64-linux-gnu --host=x86_64-linux-gnu --target=x86_64-linux-gnu Thread model: posix gcc version 4.2.3 20071123 (prerelease) (Debian 4.2.2-4) If it helps, here is the code I am using to set the bits. I modeled this after the iecset.c code. Everything opens fine, it is the final call to "snd_ctl_elem_write(this->ctl_handle, cval))" that fails. static int ao_alsa_set_spdif_status(ao_driver_t *this_gen, snd_aes_iec958_t* iec958) { alsa_driver_t *this = (alsa_driver_t *) this_gen; config_values_t *config = this->class->xine->config; char *pcm_device; snd_ctl_card_info_t *hw_info; int err; snd_pcm_info_t *pcm_info; int card; char card_name[20]; snd_ctl_elem_list_t *clist; snd_ctl_elem_id_t *cid; snd_ctl_elem_value_t *cval; unsigned int controls, cidx; const char *spdif_str = SND_CTL_NAME_IEC958("", PLAYBACK, DEFAULT); int opened_audio=0; llprintf(LOG_TRACE_ENTER, "Entered\n"); if(!this->ctl_handle) { if (!this->audio_fd) { pcm_device = config->lookup_entry(config, "audio.device.alsa_passthrough_device")->str_value; llprintf(LOG_MIXER,"Calling local_snd_pcm_open(%s)\n", pcm_device); err=local_snd_pcm_open(&this->audio_fd, pcm_device, SND_PCM_STREAM_PLAYBACK, 1); /* NON-BLOCK mode */ if ( err <0 ) { xine_log (this->class->xine, XINE_LOG_MSG, _("local_snd_pcm_open() failed:%d:%s\n"), err, snd_strerror(err)); return -1; } opened_audio = 1; } snd_pcm_info_alloca(&pcm_info); snd_pcm_info(this->audio_fd, pcm_info); card = snd_pcm_info_get_card(pcm_info); llprintf(LOG_MIXER,"snd_pcm_info_get_card() returned %d\n", card); if(card < 0){ sprintf(card_name, "default"); } else{ sprintf(card_name, "hw:CARD=%i", card); } if (opened_audio && this->audio_fd) { snd_pcm_close(this->audio_fd); this->audio_fd = NULL; opened_audio = 0; } llprintf(LOG_MIXER,"Calling snd_ctl_open(%s)\n", card_name); if ( (err = snd_ctl_open (&this->ctl_handle, card_name, 0)) < 0 ) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "%s: snd_ctl_open(): %s\n", LOG_MODULE, snd_strerror(err)); return -1; } } snd_ctl_card_info_alloca(&hw_info); if ( (err = snd_ctl_card_info (this->ctl_handle, hw_info)) < 0 ) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "%s: snd_ctl_card_info(): %s\n", LOG_MODULE, snd_strerror(err)); snd_ctl_close(this->ctl_handle); this->ctl_handle = 0; return -1; } /* set the spdif status bits */ snd_ctl_elem_list_alloca(&clist); if ((err = snd_ctl_elem_list(this->ctl_handle, clist)) < 0) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "snd_ctl_elem_list(): %s\n", snd_strerror(err)); snd_ctl_close(this->ctl_handle); this->ctl_handle = 0; return -1; } if ((err = snd_ctl_elem_list_alloc_space(clist, snd_ctl_elem_list_get_count(clist))) < 0) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "snd_ctl_elem_list_alloc_space(): %s\n", snd_strerror(err)); snd_ctl_close(this->ctl_handle); return -1; } if ((err = snd_ctl_elem_list(this->ctl_handle, clist)) < 0) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "snd_ctl_elem_list(): %s\n", snd_strerror(err)); snd_ctl_close(this->ctl_handle); this->ctl_handle = 0; return -1; } controls = snd_ctl_elem_list_get_used(clist); for (cidx = 0; cidx < controls; cidx++) { if (!strcmp(snd_ctl_elem_list_get_name(clist, cidx), spdif_str)) break; } if (cidx >= controls) { fprintf(stderr, "control \"%s\" not found\n", spdif_str); snd_ctl_close(this->ctl_handle); this->ctl_handle = 0; return -1; } snd_ctl_elem_id_alloca(&cid); snd_ctl_elem_list_get_id(clist, cidx, cid); snd_ctl_elem_value_alloca(&cval); snd_ctl_elem_value_set_id(cval, cid); llprintf(LOG_MIXER,"Writing AES0=0x%02x,AES1=0x%02x,AES2=0x%02x,AES3=0x%02x\n", iec958->status[0], iec958->status[1], iec958->status[2], iec958->status[3]); /* store the values */ snd_ctl_elem_value_set_iec958(cval, iec958); if ((err = snd_ctl_elem_write(this->ctl_handle, cval)) < 0) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "snd_ctl_elem_write(): %s\n", snd_strerror(err)); snd_ctl_close(this->ctl_handle); this->ctl_handle = 0; } llprintf(LOG_MIXER,"snd_ctl_elem_write() returrned %d\n",err); /*-------------------------------------------------------*/ /*snd_ctl_close (this->ctl_handle);*/ return 0; } ---------------------------------------------------- Bob Bouterse Sr. Software Engineer D&M Holdings Inc.

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[alsa-devel] [PATCH 001/003] snd-powermac: enable headphone detection
by Risto Suominen 15 Apr '08

15 Apr '08

From: Risto Suominen <Risto.Suominen at gmail.com> Enable port change interrupt while initialising AWACS, Screamer, and Burgundy chipsets. Signed-off-by: Risto Suominen <Risto.Suominen at gmail.com> --- Kernel version 2.6.25-rc8 (probably any 2.6) --- sound/ppc/pmac.c.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/pmac.c 2008-04-07 12:18:47.503077312 +0300 @@ -214,7 +214,7 @@ static int snd_pmac_pcm_prepare(struct s int rate_index; long offset; struct pmac_stream *astr; - + rec->dma_size = snd_pcm_lib_buffer_bytes(subs); rec->period_size = snd_pcm_lib_period_bytes(subs); rec->nperiods = rec->dma_size / rec->period_size; @@ -643,7 +643,7 @@ static int snd_pmac_pcm_close(struct snd /* reset constraints */ astr->cur_freqs = chip->freqs_ok; astr->cur_formats = chip->formats_ok; - + return 0; } @@ -1063,6 +1063,10 @@ static int __init snd_pmac_detect(struct else { struct pci_dev *pdev = NULL; +#ifndef for_each_pci_dev +#define for_each_pci_dev(d) \ + while ((d = pci_get_device(PCI_ANY_ID, PCI_ANY_ID, d)) != NULL) +#endif for_each_pci_dev(pdev) { struct device_node *np = pci_device_to_OF_node(pdev); if (np && np == macio->of_node) { @@ -1300,9 +1304,9 @@ int __init snd_pmac_new(struct snd_card snd_pmac_sound_feature(chip, 1); - /* reset */ - if (chip->model == PMAC_AWACS) - out_le32(&chip->awacs->control, 0x11); + /* reset & enable interrupts */ + if (chip->model <= PMAC_BURGUNDY) + out_le32(&chip->awacs->control, chip->control_mask); /* Powerbooks have odd ways of enabling inputs such as an expansion-bay CD or sound from an internal modem

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[alsa-devel] ac97_bus: exports duplicate symbol ac97_bus_type (owned by kernel)
by Vitaly V. Ch 15 Apr '08

15 Apr '08

#modprobe snd_via82xx #dmesg|tail ac97_bus: exports duplicate symbol ac97_bus_type (owned by kernel) snd_ac97_codec: disagrees about version of symbol ac97_bus_type snd_ac97_codec: Unknown symbol ac97_bus_type snd_via82xx: Unknown symbol snd_ac97_resume snd_via82xx: Unknown symbol snd_ac97_set_rate snd_via82xx: Unknown symbol snd_ac97_update_bits snd_via82xx: Unknown symbol snd_ac97_mixer snd_via82xx: Unknown symbol snd_ac97_bus snd_via82xx: Unknown symbol snd_ac97_update_power snd_via82xx: Unknown symbol snd_ac97_suspend snd_via82xx: Unknown symbol snd_ac97_get_short_name snd_via82xx: Unknown symbol snd_ac97_tune_hardware any ideas? \\wbr Vitaly

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[alsa-devel] PXA AC97 updates
by Mark Brown 15 Apr '08

15 Apr '08

This patch series fixes a bug in the currently queued conversion of the PXA AC97 drivers to the clock API and adds support for AC97 on PXA3xx SoCs. Takashi, since these changes build on things currently queued in the ARM tree for 2.6.26, including ARM-specific changes, it's probably best to merge via there if that's OK?

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Re: [alsa-devel] no front speaker sound with ALC262
by Rene Dohmen 15 Apr '08

15 Apr '08

Tobin, Did you receive the unit? I'm wondering if you had some progress already. Greets, Rene Op Mar 31, 2008, om 3:30 PM heeft Tobin Davis het volgende geschreven: > Yes, it would. Hmmm > > Well, I hope they send the unit today. > > Tobin > > On Mon, 2008-03-31 at 13:57 +0200, Rene Dohmen wrote: >> >> with the preinstalled win XP the sound is working OK. >> So that would rule out hardware problems with the soundcard. >> >> >> Rene >> >> >> >> >> >> >> Op Mar 28, 2008, om 7:28 PM heeft Tobin Davis het volgende >> geschreven: >> >>> I'm still waiting for my demo unit from Tangent. Latest update >>> is they think it may be a hardware issue (missing a resistor). I >>> should know more Monday. If this is the case, then the driver >>> should work with very minor modifications (subsystem ID addition). >>> >>> Tobin >>> >>> >>> On Tue, 2008-03-25 at 11:40 +0100, Rene Dohmen wrote: >>>> >>>> Op Mar 22, 2008, om 8:09 PM heeft Tobin Davis het volgende >>>> geschreven: >>>> >>>>> Rene, don't bother bugging Tangent. I have a demo unit on the >>>>> way. My first priority is getting the driver working. >>>>> >>>>> Tangent really won't help much - their a distributor, not the >>>>> OEM. This is an MSI Crystal 945 in a Tangent shell. >>>>> >>>> >>>> >>>> I know that the system is actually a Crystal 945. I got the >>>> touchscreen working via a dev driver from salt (so not tangent) >>>> On the mainboard there is a conenctor: JFP1that goes to the >>>> ms4088 board. The mainboard connects to: >>>> - power led >>>> - hdd led >>>> - speakers >>>> - main power switch >>>> >>>> >>>> I can provide you with some photoos if you like. >>>> >>>> >>>> As stated earlier: the sound is working by with very low volume >>>> levels >>>> >>>> >>>> >>>> >>>>> If you want to try something, see if the bios can be reflashed >>>>> with the Crystal 945 firmware. >>>>> >>>> >>>> >>>> I sound working with the original crystal? I don't think tangent >>>> has flshed their BIOS to another version. >>>> In the crystal 945 specs I found that the're using a Realtek >>>> ALC262 with azalia codec, maybe this helps. >>>> >>>> >>>> Rene >>>> >>>> >>>> >>>>> As to the speaker board, where on the main board is it >>>>> mounted? Is the header labeled? How many pins? It sounds >>>>> like the speakers may be on the front panel header, which would >>>>> make them function like headphones. Or this board is custom >>>>> build for this application and I don't know what I'm talking >>>>> about. >>>>> >>>>> Tobin >>>>> >>>>> >>>>> On Sat, 2008-03-22 at 18:59 +0100, Rene Dohmen wrote: >>>>>> >>>>>> Hi Guys, >>>>>> >>>>>> I'm glad t o read that there are other people with the same >>>>>> problem >>>>>> on the same system. >>>>>> I just opened my Tangent 8000S and it has indeed a MSI mainboard: >>>>>> MS7290. The speakers are connected to a small custom board: >>>>>> MS-4088 >>>>>> There's no extra chip on this customboard and the board is >>>>>> directly >>>>>> connected to the mainboard. So probably the signal leaving the >>>>>> mainboard in amplified already. (or there's a amplifier >>>>>> builtin the >>>>>> speakers) >>>>>> >>>>>> Next week i'll have some time to put tangent under some more >>>>>> pressure >>>>>> for info, so maybe that will speed up things a little bit. >>>>>> >>>>>> Grz Rene >>>>>> >>>>>> _______________________________________________ >>>>>> Alsa-devel mailing list >>>>>> Alsa-devel(a)alsa-project.org >>>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>>> -- Tobin Davis >>>>> >>>>> love, n.: >>>>> When you don't want someone too close--because you're very >>>>> sensitive >>>>> to pleasure. >>>> >>> -- Tobin Davis >>> >>> I can hire one half of the working class to kill the other half. >>> -- Jay Gould >> >> > -- Tobin Davis > > He is no lawyer who cannot take two sides.

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[alsa-devel] dB, dB and more dB....
by Tom Watson 15 Apr '08

15 Apr '08

I find all of this an interesting discussion, and it looks like: 1) People want to hear some output. 2) "Pro" people don't want to blow up speakers. <soapbox> As was said before it is all relative. I wrote an application using ALSA and also needed to allow it to play back CDs at the same level. What I did was take a sample WAV file and burn it to a CD. It had several levels, from full scale, and 3dB (divide by 2) down. It was quite informative for setting mixer levels. I needed it because, while I was using an "add-in" board (M-Audio Revo 7.1), the CD drive was connected to the internal motherboard input place. I used a small jumper plug to connect the output of the motherboard audio out to the Revo7.1 board "in" and setup mixers accordingly. I ended up using a scope to get the levels correct, without overloading (distorting) the audio path. Quite informative. It seems that all of us have differing ideas on "what to expect" when it comes to audio interfaces. Twenty years ago, these problems really didn't exist, since computer audio was in its infancy, but now audio is everywhere, and it muct be accounted for. Alsa's implementation of having audio MUTED on initialization (startup) is a good one, and many distributions have init scripts that take over from there, restoring old values. While not ideal (just set the audio to where I want it!), it is about the best we can do. The "pro" guy who doesn't want to blow up his speakers, I can only offer a suggestion. What do you do when plugging in and out "normal" audio equipment. Most of the time I turn down the level control since my big fat fingers will end up touching the tip/ring of the audio connector and LOTS of hum results. If you don't do this, you may have a PEBKAC problem which I (or any other developer) can solve. I think we are all chasing a problem that really doesn't exist. As for dB, some people refer to 0dB as "full scale". Some refer to it as 0 VU. Some make it out to be 1-v p-p. Go figure. The biggest problem is that crazy people think "louder" is better. It isn't! A small "white paper" on the subject (part of ALSA?) with an agreed upon "introduction to computer audio" might be a good place to start. Hopefully no more than a couple of pages in terms that everyone can understand. </soapbox> -- Tom Watson tsw(a)johana.com ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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[alsa-devel] [PATCH] ASoC: Add support for 19.2 MHz MCLK in TLV320AIC3X
by Jarkko Nikula 15 Apr '08

15 Apr '08

Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/codecs/tlv320aic3x.c | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e6a730b..630684f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -661,42 +661,52 @@ struct aic3x_rate_divs { static const struct aic3x_rate_divs aic3x_divs[] = { /* 8k */ {12000000, 8000, 48000, 0xa, 16, 3840}, + {19200000, 8000, 48000, 0xa, 10, 2400}, {22579200, 8000, 48000, 0xa, 8, 7075}, {33868800, 8000, 48000, 0xa, 5, 8049}, /* 11.025k */ {12000000, 11025, 44100, 0x6, 15, 528}, + {19200000, 11025, 44100, 0x6, 9, 4080}, {22579200, 11025, 44100, 0x6, 8, 0}, {33868800, 11025, 44100, 0x6, 5, 3333}, /* 16k */ {12000000, 16000, 48000, 0x4, 16, 3840}, + {19200000, 16000, 48000, 0x4, 10, 2400}, {22579200, 16000, 48000, 0x4, 8, 7075}, {33868800, 16000, 48000, 0x4, 5, 8049}, /* 22.05k */ {12000000, 22050, 44100, 0x2, 15, 528}, + {19200000, 22050, 44100, 0x2, 9, 4080}, {22579200, 22050, 44100, 0x2, 8, 0}, {33868800, 22050, 44100, 0x2, 5, 3333}, /* 32k */ {12000000, 32000, 48000, 0x1, 16, 3840}, + {19200000, 32000, 48000, 0x1, 10, 2400}, {22579200, 32000, 48000, 0x1, 8, 7075}, {33868800, 32000, 48000, 0x1, 5, 8049}, /* 44.1k */ {12000000, 44100, 44100, 0x0, 15, 528}, + {19200000, 44100, 44100, 0x0, 9, 4080}, {22579200, 44100, 44100, 0x0, 8, 0}, {33868800, 44100, 44100, 0x0, 5, 3333}, /* 48k */ {12000000, 48000, 48000, 0x0, 16, 3840}, + {19200000, 48000, 48000, 0x0, 10, 2400}, {22579200, 48000, 48000, 0x0, 8, 7075}, {33868800, 48000, 48000, 0x0, 5, 8049}, /* 64k */ {12000000, 64000, 96000, 0x1, 16, 3840}, + {19200000, 64000, 96000, 0x1, 10, 2400}, {22579200, 64000, 96000, 0x1, 8, 7075}, {33868800, 64000, 96000, 0x1, 5, 8049}, /* 88.2k */ {12000000, 88200, 88200, 0x0, 15, 528}, + {19200000, 88200, 88200, 0x0, 9, 4080}, {22579200, 88200, 88200, 0x0, 8, 0}, {33868800, 88200, 88200, 0x0, 5, 3333}, /* 96k */ {12000000, 96000, 96000, 0x0, 16, 3840}, + {19200000, 96000, 96000, 0x0, 10, 2400}, {22579200, 96000, 96000, 0x0, 8, 7075}, {33868800, 96000, 96000, 0x0, 5, 8049}, }; @@ -818,6 +828,7 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, switch (freq) { case 12000000: + case 19200000: case 22579200: case 33868800: aic3x->sysclk = freq; -- 1.5.4.4

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[alsa-devel] Patchmix-style application for EMU cards
by Camilo Polymeris 15 Apr '08

15 Apr '08

Hello I am working on a graphical mixing/routing application for the Emu10k series of semi-pro soundcards, but I am having trouble figuring out the routing mechanism of the driver, specially the "Multichannel PCM" part. Is there any technical documentation on this card I could read (I haven't found any), or maybe the driver developer can help me out? Thanks, Camilo.

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