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Re: [alsa-devel] ALSA always writes _whole_ periods, right?
by Takashi Iwai 27 Jun '07

27 Jun '07

At Sun, 24 Jun 2007 11:34:56 +0200, Joachim Förster wrote: > > Hi Takashi, > > thanks for your answer. > > On Wed, 2007-06-20 at 12:45 +0200, Takashi Iwai wrote: > > > Ok, ahm, I think my question was not about what I really wanted to know > > > - sorry. I have to reformulate it: In fact I would like to know, how the > > > ALSA-library mmap code behaves regarding the last chunk of samples, for > > > example: Consider aplay, playing a wav file and the wav file's > > > size/number of samples does _not_ fit to a multiple of a period's size. > > > Then, does the mmap code copy these last samples as is _or_ does it copy > > > one whole last period (which is filled with zeros)? > > > > No, you have to fill silence by yourself before committing the mmap > > chunk. The mmap code itself doesn't care about it. > > By "fill silence by yourself", you mean the application (user-space) > which is using ALSA library (with mmap), right? So this is not in the > responsibility of the sound driver, right? Right. The driver has some functionality to fill silence automatically, but it's better to avoid to play with it :) > Hmmm, so if an application does _not_ transfer sound data in a multiple > of period size, the _last_ "period" will be "incomplete". Then, my > question is: Does ALSA expect to receive a call of snd_period_elapsed() > after these last samples are played, although the it was _not_ a > complete period? I think it might depend on the hardware. Some hardwares are available to stop at the exact position, but most hardware are not. So, it's _safer_ to send always the data in the period size. And, yes, for the last period with less data, the app should fill the data to align to the period size. Takashi

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[alsa-devel] Getting physical address of the DMA buffer
by Pharaoh . 27 Jun '07

27 Jun '07

Hi I have allocated the DMA buffer using snd_pcm_lib_malloc_pages() and it was preallocated as: snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data (GFP_KERNEL), 128 * 1024, 128 * 1024); and then, snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); None of these calls fail. The physical address of the allocated buffer is shown as 0 in my case since according to 'Writing an ALSA driver' the physical address of the DMA buffer will not be accessble from the main CPU. Is that why the physical address is shown as 0? Also, in my pointer function, I need the DMA buffers physical address to calculate the position in buffer etc. How can do that since runtime->dma_addr is 0 for me? Apologies for longish mail but I have to paste the actual paragraphs here to clear my understanding in less number of emails by avoiding further emails. In chapter 5, in DMA buffer information, it is said that, 1. The dma_area holds the buffer pointer (the logical address). You can call memcpy from/to this pointer. Meanwhile, dma_addr holds the physical address of the buffer. This ﬁeld is speciﬁed only when the buffer is a linear buffer. and in descrition of prepare callback it says, 2. The physical address of the allocated buffer is set to runtime->dma_area. Statements 1 and 2 contradict each other or I am mis reading them? How should I get the physical address in pointer callbacks? As runtime->dma_area or runtime->dma_addr which is 0? -pharaoh.

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[alsa-devel] Input/output error while playing a file using aplay
by Pharaoh . 26 Jun '07

26 Jun '07

Hi I am testing my audio driver using aplay. I trying to play an audio file using: #aplay -Dplughw:0,0 --rate=44100 --format=S16 --channels=2 test.wav The log is as follows, snd_card_omap_alsa_open start snd_card_omap_alsa_open end Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo snd_omap_alsa_hw_params start snd_omap_alsa_prepare start snd_omap_alsa_prepare end snd_omap_alsa_trigger start audio_start_dma_chain start audio_start_dma_chain end omap_start_alsa_sound_dma end snd_omap_alsa_trigger end snd_omap_alsa_pointer start audio_get_dma_pos start audio_get_dma_pos end It waits here for sometime and then I get snd_omap_alsa_trigger starte error: Input/output error Then the things are wound up as: audio_stop_dma start omap_stop_alsa_sound_dma start omap_stop_alsa_sound_dma end omap_clear_alsa_sound_dma start omap_clear_alsa_sound_dma end snd_omap_alsa_trigger end snd_omap_alsa_hw_free start snd_omap_alsa_hw_free end snd_card_omap_alsa_close start snd_card_omap_alsa_close end So basically the problem is in trigger where it waits for something and then winds up things. Any idea, what could be the reason. I am thinking it has to do something with DMA interrupts etc.

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[alsa-devel] at91 SoC Modifications
by Paul Kavan 26 Jun '07

26 Jun '07

Hello: Okay, so after probing the various pcm lines, I think I see that the format used for the pcm is a little bit off of what my codec likes to play nice. In my machine/codec driver, I see that I have some lines: /* set cpu DAI configuration */ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); I believe this is where it is selecting the format. In at91-ssc.c, I see that this format is given as: case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: /* * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. * * The SSC transmit and receive clocks are generated from the * MCK divider, and the BCLK signal is output on the SSC TK line. */ rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) | (( 1 << 16) & AT91_SSC_STTDLY) | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) | (( 0 << 16) & AT91_SSC_FSLEN) | (((channels - 1) << 8) & AT91_SSC_DATNB) | (( 1 << 7) & AT91_SSC_MSBF) | (( 0 << 5) & AT91_SSC_LOOP) | (((bits - 1) << 0) & AT91_SSC_DATALEN); tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) | (( 1 << 16) & AT91_SSC_STTDLY) | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) | (( 0 << 23) & AT91_SSC_FSDEN) | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) | (( 0 << 16) & AT91_SSC_FSLEN) | (((channels - 1) << 8) & AT91_SSC_DATNB) | (( 1 << 7) & AT91_SSC_MSBF) | (( 0 << 5) & AT91_SSC_DATDEF) | (((bits - 1) << 0) & AT91_SSC_DATALEN); break; I should have caught this before--I spent enough time dealing with the HW registers. The simple ssc pcm driver I wrote sets the regs a little differently to get the peak performance out of the codec, e.g. In RCMR, I have STTDLY=0, AT91_SSC_CKI_FALLING, AT91_SSC_CKS_PIN. In RFMR everything looks okay except that I have FSOS_NONE as I borrow it from TF0 In TCMR the STTDLY=0 in mine In TFMR I do nothing with the FSEDGE and FSDEN and I have FSLEN=7 and DATADEF=1. So, I left with a decision. Do I add a mode or do I modify DSP A to fit my needs? I am not sure how much coding it would take to add a mode--I *do* know it would take little time to modify DSP A, but I am reluctant to make a change to something that is probably 'standard'. Any thoughts on this? Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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[alsa-devel] Fwd: Fwd: Problems with hda_intel, Santa Rosa, and suspend
by Matt Mullins 25 Jun '07

25 Jun '07

On 6/25/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > Yes, single_cmd mode is used always. > That's bad. I believe we have established that. What can I do about it? *Can* I do anything about it? What functions should I start debugging? Has this happened with other codecs before? What documentation will I need or want other than the Intel HDA and STAC9205 specifications? I'm definitely willing to help work on this, as not many people have this hardware. I'm just new to the sound system, so I'm still trying to figure out what-calls-what. - Matt

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[alsa-devel] Intel Realtek ALC268 don't sound
by antoniorc＠4lineas.net 25 Jun '07

25 Jun '07

Hi, I tried latest HG driver version of Alsa, with patch included for ALC268. Apparently card is detected and mixer show correctly volumes bars, but don't sound. I tried option 3stack and auto in /etc/modprobe.d/alsa-base but nothing difference. I've a HP Pavilion dv6528 with Santa Rosa. Please tell me that I can do? Thanks

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[alsa-devel] problem with audigy2 ZS remote
by Stefano Radaelli 25 Jun '07

25 Jun '07

Hi all! Sorry for my bad english. I have a problem with the livedrive of an audigy2 ZS. After some upgrades (on a gentoo linux system) the eternal panel seems to stop working. Even after the "initialization" (with echo on the midi device), the IR led not blink on remote action. This worked correctly in the past but after some upgrade (i don't know exactly what) it stoipped working. I don't known where i have to look (logs?) or what type of debugging it's necessary. Help me please

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[alsa-devel] ARM in embedded audio
by Alan Horstmann 25 Jun '07

25 Jun '07

Hi, Can anyone help point me to info on using ARM in embedded audio systems? Does anyone use a low-cost chip with I2S interface? Thanks Alan

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Re: [alsa-devel] [alsa-cvslog] alsa-lib: Add struct timeval and timespec definition when _POSIX_C_SOURCE is not defined
by Sergey Vlasov 25 Jun '07

25 Jun '07

On Fri, Oct 06, 2006 at 11:01:44AM +0200, Jaroslav Kysela wrote: > changeset: 2218:603411fd0c77712cab43cdd5afcf0ff4ebe51a7f > tag: tip > user: perex > date: Fri Oct 06 11:01:40 2006 +0200 > files: include/global.h > description: > Add struct timeval and timespec definition when _POSIX_C_SOURCE is not defined > > FIXME: It might cause problems on some platforms when tv_usec is not long type. And now it also breaks compilation of the following code (at least with glibc 2.5): #include <sys/select.h> #include <alsa/global.h> $ gcc -c --std=c99 x.c In file included from x.c:2: /usr/include/alsa/global.h:134: error: redefinition of 'struct timeval' /usr/include/alsa/global.h:139: error: redefinition of 'struct timespec' (Without #include <sys/select.h> only 'struct timespec' is redefined.) IMHO attempts to define structures which should be defined by system headers are too dangerous - if someone is using preprocessor flags which hide required parts of system headers, it is their problem, and the risk of mismatch is too high. BTW, <time.h> does not provide 'struct timeval' at all, even with _GNU_SOURCE - it defines only 'struct timespec'; however, this does not cause problems, because ALSA headers currently use only pointers to snd_timestamp_t, which works even without a complete definition of 'struct timeval'. We can try to include <sys/time.h> to get 'struct timeval', or even <sys/select.h> (which currently seems to provide both 'struct timeval' and 'struct timespec' even with -ansi, and http://www.opengroup.org/onlinepubs/009695399/basedefs/sys/select.h.html tells that it should always provide their definitions). > > > diff -r 7bc021fb4bf1027b23aa5429ceda1e1e35c7d126 -r 603411fd0c77712cab43cdd5afcf0ff4ebe51a7f include/global.h > --- a/include/global.h Fri Oct 06 10:38:39 2006 +0200 > +++ b/include/global.h Fri Oct 06 11:01:40 2006 +0200 > @@ -130,6 +130,18 @@ int snd_shm_area_destroy(struct snd_shm_ > > int snd_user_file(const char *file, char **result); > > +#ifndef _POSIC_C_SOURCE > +struct timeval { > + time_t tv_sec; /* seconds */ > + long tv_usec; /* microseconds */ > +}; > + > +struct timespec { > + time_t tv_sec; /* seconds */ > + long tv_nsec; /* nanoseconds */ > +}; > +#endif > + > /** Timestamp */ > typedef struct timeval snd_timestamp_t; > /** Hi-res timestamp */

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[alsa-devel] SALSA library in ARM platform
by Nobin Mathew 25 Jun '07

25 Jun '07

Hi, I tried compiling SALSA for ARM. I got few errors. CC=arm-none-linux-gnueabi-gcc ./configure --target=arm-linux --host=i386 --prefix=/usr make Then I am getting following errors See the bottom part /bin/bash ../libtool --tag=CC --mode=compile arm-none-linux-gnueabi-gcc -DPACKAGE_NAME=\"\" -DPACKAGE_TARNAME=\"\" -DPACKAGE_VERSION=\"\" -DPACKAGE_STRING=\"\" -DPACKAGE_BUGREPORT=\"\" -DPACKAGE=\"salsa-lib\" -DVERSION=\"0.0.1\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 -DHAVE_DLFCN_H=1 -DSTDC_HEADERS=1 -I. -Wall -g -g -O2 -MT mixer.lo -MD -MP -MF .deps/mixer.Tpo -c -o mixer.lo mixer.c arm-none-linux-gnueabi-gcc -DPACKAGE_NAME=\"\" -DPACKAGE_TARNAME=\"\" -DPACKAGE_VERSION=\"\" -DPACKAGE_STRING=\"\" -DPACKAGE_BUGREPORT=\"\" -DPACKAGE=\"salsa-lib\" -DVERSION=\"0.0.1\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 -DHAVE_DLFCN_H=1 -DSTDC_HEADERS=1 -I. -Wall -g -g -O2 -MT mixer.lo -MD -MP -MF .deps/mixer.Tpo -c mixer.c -o mixer.o In file included from hcontrol.h:36, from mixer.h:25, from mixer.c:30: hctl_macros.h: In function 'snd_hctl_elem_tlv_read': hctl_macros.h:136: warning: 'snd_ctl_elem_tlv_read' is deprecated (declared at ctl_macros.h:1131) hctl_macros.h: In function 'snd_hctl_elem_tlv_write': hctl_macros.h:142: warning: 'snd_ctl_elem_tlv_write' is deprecated (declared at ctl_macros.h:1138) hctl_macros.h: In function 'snd_hctl_elem_tlv_command': hctl_macros.h:148: warning: 'snd_ctl_elem_tlv_command' is deprecated (declared at ctl_macros.h:1145) mv -f .deps/mixer.Tpo .deps/mixer.Plo /bin/bash ../libtool --tag=CC --mode=link arm-none-linux-gnueabi-gcc -Wall -g -g -O2 -version-number 0:0:1 -o libsalsa.la -rpath /usr/lib control.lo cards.lo pcm.lo pcm_params.lo pcm_misc.lo hcontrol.lo mixer.lo mkdir .libs libtool: link: CURRENT `' must be a nonnegative integer libtool: link: `0:0:1' is not valid version information make[1]: *** [libsalsa.la] Error 1 make[1]: Leaving directory `/home/nmathew/Desktop/salsa-lib-0.0.1/src' make: *** [all-recursive] Error 1

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[alsa-devel] [PATCH] hda: support for S/PDIF out on ASUS M2V
by Mike Crash 25 Jun '07

25 Jun '07

Added support for S/PDIF digital output on ASUS M2V motheboard - added new model "3stack-660-digout" and ALC660VD_3ST_DIG Apply to 1.0.14 driver file patch_realtek.c Signed-off-by: Mike Crash <mike(a)mikecrash.com> *** patch_realtek.c 2007-06-04 09:28:54.000000000 +0200 --- patch_realtek2.c 2007-06-16 21:46:14.000000000 +0200 *************** enum { *** 117,118 **** --- 117,119 ---- ALC660VD_3ST, + ALC660VD_3ST_DIG, ALC861VD_3ST, *************** static const char *alc861vd_models[ALC86 *** 9466,9467 **** --- 9467,9469 ---- [ALC660VD_3ST] = "3stack-660", + [ALC660VD_3ST_DIG]= "3stack-660-digout", [ALC861VD_3ST] = "3stack", *************** static struct snd_pci_quirk alc861vd_cfg *** 9477,9479 **** SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), ! SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), --- 9479,9481 ---- SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), ! SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), *************** static struct alc_config_preset alc861vd *** 9501,9502 **** --- 9503,9517 ---- }, + [ALC660VD_3ST_DIG] = { + .mixers = { alc861vd_3st_mixer }, + .init_verbs = { alc861vd_volume_init_verbs, + alc861vd_3stack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), + .dac_nids = alc660vd_dac_nids, + .dig_out_nid = ALC861VD_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), + .adc_nids = alc861vd_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), + .channel_mode = alc861vd_3stack_2ch_modes, + .input_mux = &alc861vd_capture_source, + }, [ALC861VD_3ST] = {

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[alsa-devel] [PATCH] MPU401 lockups
by Serge Nikolaenko 25 Jun '07

25 Jun '07

Hello, The attachement is the fix for mpu to prevent lockups/freezes. Added mutexes with down_trylock() calls, to prevent blocking. Midi in/out not tested - I don't have no midi devices at the moment. My soundcard is ESI Juli@. MB integrated midi is disabled in bios. Last system messages after system freeze when running wine/jackd: kernel: cmd: 0xff failed at 0xac0c (status = 0xb, data = 0x0) ... kernel: BUG: soft lockup detected on CPU#0! kernel: kernel: Call Trace: kernel: <IRQ> [<ffffffff802b0bdb>] softlockup_tick+0xdb/0xf6 kernel: [<ffffffff8028f5d0>] update_process_times+0x42/0x68 kernel: [<ffffffff80271f0c>] smp_local_timer_interrupt+0x34/0x55 kernel: [<ffffffff802725e8>] smp_apic_timer_interrupt+0x51/0x69 kernel: [<ffffffff8025ace6>] apic_timer_interrupt+0x66/0x70 kernel: [<ffffffff889c600d>] :snd_mpu401_uart:mpu401_read_port+0x4/0x8 kernel: [<ffffffff889c62f0>] :snd_mpu401_uart:_snd_mpu401_uart_interrupt+0x4c/0x70 kernel: [<ffffffff889c6391>] :snd_mpu401_uart:snd_mpu401_uart_interrupt+0x13/0x1a kernel: [<ffffffff88a1f34b>] :snd_ice1724:snd_vt1724_interrupt+0x4e/0x13a kernel: [<ffffffff80210b93>] handle_IRQ_event+0x25/0x53 kernel: [<ffffffff802b202a>] handle_fasteoi_irq+0x92/0xd1 kernel: [<ffffffff8026862a>] do_IRQ+0xff/0x16d kernel: [<ffffffff8025a631>] ret_from_intr+0x0/0xa kernel: <EOI> [<ffffffff802f5df7>] proc_info_read+0x0/0xb9 kernel: [<ffffffff802f5df9>] proc_info_read+0x2/0xb9 kernel: [<ffffffff8020b14e>] vfs_read+0xcb/0x173 kernel: [<ffffffff80211847>] sys_read+0x45/0x6e kernel: [<ffffffff8025a11e>] system_call+0x7e/0x83

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Re: [alsa-devel] Fwd: Problems with hda_intel, Santa Rosa, and suspend
by Takashi Iwai 25 Jun '07

25 Jun '07

[Please don't strip Cc to ML!] At Tue, 19 Jun 2007 15:54:42 -0500, Matt Mullins wrote: > > I played around with probe_mask, and found that probe_mask=0x1 causes > the error, but probe_mask=0x2 does not cause the timeout. Codec 1 is > my Sigmatel STAC9205, which is what plays sound on my laptop, and > codec 2 is my Conexant modem. The modem isn't the codec causing > problems; the audio codec is! OK, good to know. Does this problem occur with S2D? What is the very first command that triggers this timeout error? I'm wondering what is the difference between them -- according to your report, reloading the module fixes the problem. But, the resume callback is almost identical with the re-initiailzation. Perhaps it's something to do with the power-state management. What about the patch below? Takashi diff -r c531d63e820b sound/pci/hda/hda_intel.c --- a/sound/pci/hda/hda_intel.c Wed Jun 20 12:03:09 2007 +0200 +++ b/sound/pci/hda/hda_intel.c Wed Jun 20 12:09:40 2007 +0200 @@ -1450,7 +1450,7 @@ static int azx_suspend(struct pci_dev *p pci_disable_msi(chip->pci); pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + /* pci_set_power_state(pci, pci_choose_state(pci, state)); */ return 0; }

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[alsa-devel] hda-intel without headphone output
by Robert Lemmen 25 Jun '07

25 Jun '07

hi everyone, i have a fujitsu siemens amilo pro v3515 notebook, with a via VT8251/ VT8237A chipset. sound works fine through the speakers, but i don't get anything through the headphones, no headphone channel shows up in the mixer, and the headphone jack switch doesn't mute the speakers. i have alsa 1.0.13, and a 2.6.18 kernel. i tried all "model" settings that i could find in ALSA-Configuration in the kernel tree. lspci -nv gives 04:01.0 0403: 1106:3288 (rev 10) Subsystem: 1734:10cb Flags: bus master, fast devsel, latency 0, IRQ 58 Memory at c9100000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 (only the stanza for the sound card, yell if you need more), and /proc/asound/card0/codec#0 gives: Codec: Generic 14f1 ID 5045 Address: 0 Vendor Id: 0x14f15045 Subsystem Id: 0x173410cb Revision Id: 0x100100 Default PCM: rates 0x140, bits 0x0e, types 0x1 Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x10 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x28 0x28] Pincap 0x0810014: OUT EAPD Detect Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 2 0x19* 0x17 Node 0x11 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x0121401f: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Power: 0x0 Connection: 2 0x19* 0x17 Node 0x12 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0xab 0xab] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x01a19020: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x20: IN Power: 0x0 Connection: 2 0x19* 0x17 Node 0x13 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x214571f0: [Jack] SPDIF Out at Sep Rear Conn = Optical, Color = Yellow Pin-ctls: 0x00: Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081124: IN Detect Pin Default 0x97a7012e: [Fixed] Mic at Int Riser Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x15 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Node 0x16 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0b, mute=1 Amp-Out vals: [0x06] Node 0x17 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x14, nsteps=0x2b, stepsize=0x05, mute=1 Amp-In vals: [0x94 0x94] [0x17 0x17] [0x94 0x94] [0x94 0x94] [0x94 0x94] Power: 0x0 Connection: 5 0x19 0x14 0x12 0x11 0x15 Node 0x18 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates 0x040, bits 0x06, types 0x5 Node 0x19 [Audio Output] wcaps 0xc11: Stereo PCM: rates 0x540, bits 0x0e, types 0x1 Power: 0x0 Node 0x1a [Audio Input] wcaps 0x100d0b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x17, stepsize=0x05, mute=1 Amp-In vals: [0x17 0x17] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Power: 0x0 Connection: 5 0x17 0x14 0x12 0x11 0x15* Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono please tell me if there is anything i can do, any additional information you need, or any general tipps. thanks robert -- Robert Lemmen http://www.semistable.com

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[alsa-devel] [1/2] 2.6.22-rc6: known regressions
by Michal Piotrowski 25 Jun '07

25 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc6. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions *STATISTICS* (a.k.a. list of aces) Name Regressions fixed since 21-Jun-2007 Andi Kleen 1 Hugh Dickins 1 Jean Delvare 1 Unclassified Subject : Kernel hang on CMOS_READ References : http://lkml.org/lkml/2007/6/3/138 Submitter : Rodrigo Luiz <troops.of.doom(a)gmail.com> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Status : Unknown Memory management Subject : 2.6.22-rc5-yesterdaygit with VM debug: BUG in mm/rmap.c:66: anon_vma_link ? References : http://lkml.org/lkml/2007/6/25/45 Submitter : Petr Vandrovec <petr(a)vmware.com> Status : unknown SATA/PATA Subject : libata IT821X driver still fails! Hard-freezes system References : http://lkml.org/lkml/2007/6/14/82 Submitter : Rodney Gordon II <meff(a)spherevision.org> Status : unknown Regards, Michal -- LOG http://www.stardust.webpages.pl/log/

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[alsa-devel] alsa-lib: pcm_softvol.c: attenuation curve - it seems to be too quiet at full volume
by John L. Utz III 25 Jun '07

25 Jun '07

Hello All; I am in release testing with a fix to our product to use 4 channels on a CMEDIA 8738-MC6 (previously we only used 2 channels on the CMEDIA and a separate soundcard for the 2 back channels). The 8738 is a pretty funny old duck of a chip and i am trying to work around a subset of it's many idiosyncracies. :-) The most glaring defect is the absence of hard volume on the back channels *and* that the back channel is louder than the front channel. Obviously, this is a great candidate for softvol! Softvol works quite well *except* that the max volume for the rear channel is now much quieter than it is without softvol. :-( Is there anything i can do to alter this behavior? I notice that there is an initial minimal DB in the file: #define PRESET_MIN_DB -51.0 is this where everything starts from? if i bump this up, will i get a higher max volume? tnx! johnu

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[alsa-devel] Query regarding pointer callback
by Pharaoh . 25 Jun '07

25 Jun '07

Hi In Writing an ALSA driver, it is said that pointer callback is called usually from the buffer-update routine in the pcm middle layer, which is invoked when snd_pcm_period_elapsed() is called in the interrupt routine. Then the pcm middle layer updates the position and calculates the available space, and wakes up the sleeping poll threads, etc What are the other places from where pointer callback is called? If it is not called from any other routine then does it mean that interrupts are occuring?

2 2

[alsa-devel] Intel Realtek ALC268 don't sound
by Antonio Rguez. Capita 25 Jun '07

25 Jun '07

Hi, I tried lastest HG driver version of Alsa, with patch included for ALC268. Apparently card is detected and mixer show correctly volumes bars, but don't sound. I tried option 3stack and auto in /etc/modprobe.d/alsa-base but nothing difference. I've a HP Pavilion dv6528 with Santa Rosa. Please tell me that I can do? Thanks

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Re: [alsa-devel] No sound on Medion laptop (intel 82801G)
by albox＠free.fr 25 Jun '07

25 Jun '07

Here is the result : $ sudo dmesg -c $ sudo rmmod snd-hda-intel && sudo modprobe snd-hda-intel probe_mask=1 $ dmesg [ 638.324000] ACPI: PCI interrupt for device 0000:00:1b.0 disabled [ 638.368000] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 22 [ 638.368000] PCI: Setting latency timer of device 0000:00:1b.0 to 64 [ 638.400000] ALSA /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0x3 [ 638.576000] ALSA /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:1820: hda_codec: model 'medion' is selected for config 161f:2054 (Medion laptop) I can hear a quick sound (like a "bip") when I enter the command but after that, the sound doesn't work.

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[alsa-devel] No sound on Medion laptop (intel 82801G)
by albox＠free.fr 25 Jun '07

25 Jun '07

Hi, I reported a bug about medion laptops having an intel 82801G sound card that doesn't work with ALSA 1.0.14rc4 on Kubuntu : https://bugtrack.alsa-project.org/alsa-bug/bug_view_advanced_page.php?bug_i… I didn't get any news yet. As I said, I'm willing to help. Any ideas about what I should check ? Is there any web page that could help me to detect the problem ? Thanks

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[alsa-devel] $1000 bounty for getting audio working under linux on a Osprey 440
by Nathan Allen Stratton 25 Jun '07

25 Jun '07

Need to get audio working on Osprey 440 card. http://www.viewcast.com/pages.asp?page=osprey-440 The card is bt878 based, and ViewCast says it is supported on linux however not something they support. Audio is not detected by default, but does how up when you do a modprobe snd-bt87x load-all(see below). However I have not been able to pull audio from the card (only get high pitched noise). Bid may require patching bt878.c if modprobe options don't work. Only mono support is required. [root@video share]# amixer Simple mixer control 'Bass',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 48 Mono: 30 [62%] Simple mixer control 'Bass Boost',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Treble',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 48 Mono: 24 [50%] Simple mixer control 'PCM',0 Capabilities: pvolume cvolume pswitch pswitch-joined cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 44 Capture 0 - 15 Mono: Capture 15 [100%] [15.00dB] [on] Front Left: Playback 44 [100%] [3.00dB] [on] Front Right: Playback 44 [100%] [3.00dB] [on] Simple mixer control 'PCM Capture Source',0 Capabilities: enum Items: 'PCM' 'Mic' 'Input 2' Item0: 'Input 2' Simple mixer control 'PCM',1 Capabilities: cvolume Capture channels: Mono Limits: Capture 0 - 656 Mono: Capture 0 [0%] [-41.00dB] Simple mixer control 'Mic',0 Capabilities: cvolume Capture channels: Mono Limits: Capture 0 - 15 Mono: Capture 0 [0%] [16.00dB] Simple mixer control 'Auto Gain Control',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] [root@video share]# amixer -c 1 Simple mixer control 'FM',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'Mic/Line',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] Simple mixer control 'Capture',0 Capabilities: cvolume Capture channels: Mono Limits: Capture 0 - 15 Mono: Capture 15 [100%] Simple mixer control 'Capture Boost',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'TV Tuner',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] :::::::::::::: cards :::::::::::::: 0 [INport ]: USB-Audio - INport Xitel INport at usb-0000:00:1d.0-1, full speed 1 [Bt878 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xd8001000, irq 16 2 [Bt878_1 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xd8003000, irq 17 3 [Bt878_2 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xd8005000, irq 18 4 [Bt878_3 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xd8007000, irq 19 :::::::::::::: devices :::::::::::::: 2: : timer 3: : sequencer 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0] : control 7: [ 1- 1]: digital audio capture 8: [ 1- 0]: digital audio capture 9: [ 1] : control 10: [ 2- 1]: digital audio capture 11: [ 2- 0]: digital audio capture 12: [ 2] : control 13: [ 3- 1]: digital audio capture 14: [ 3- 0]: digital audio capture 15: [ 3] : control 16: [ 4- 1]: digital audio capture 17: [ 4- 0]: digital audio capture 18: [ 4] : control :::::::::::::: hwdep :::::::::::::: *** INport: directory *** :::::::::::::: modules :::::::::::::: 0 snd_usb_audio 1 snd_bt87x 2 snd_bt87x 3 snd_bt87x 4 snd_bt87x *** oss: directory *** :::::::::::::: pcm :::::::::::::: 00-00: USB Audio : USB Audio : playback 1 : capture 1 01-01: Bt87x Analog : Bt87x Analog : capture 1 01-00: Bt87x Digital : Bt87x Digital : capture 1 02-01: Bt87x Analog : Bt87x Analog : capture 1 02-00: Bt87x Digital : Bt87x Digital : capture 1 03-01: Bt87x Analog : Bt87x Analog : capture 1 03-00: Bt87x Digital : Bt87x Digital : capture 1 04-01: Bt87x Analog : Bt87x Analog : capture 1 04-00: Bt87x Digital : Bt87x Digital : capture 1 *** seq: directory *** :::::::::::::: timers :::::::::::::: G0: system timer : 1000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE P1-0-1: PCM capture 1-0-1 : SLAVE P1-1-1: PCM capture 1-1-1 : SLAVE P2-0-1: PCM capture 2-0-1 : SLAVE P2-1-1: PCM capture 2-1-1 : SLAVE P3-0-1: PCM capture 3-0-1 : SLAVE P3-1-1: PCM capture 3-1-1 : SLAVE P4-0-1: PCM capture 4-0-1 : SLAVE P4-1-1: PCM capture 4-1-1 : SLAVE :::::::::::::: version :::::::::::::: Advanced Linux Sound Architecture Driver Version 1.0.14rc3 (Wed Mar 14 07:25:50 2007 UTC). unknown card 0x878-0x70:0xff07, using default rate 32000 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)lists.sf.net> ACPI: PCI Interrupt 0000:06:04.1[A] -> GSI 16 (level, low) -> IRQ 16 unknown card 0x878-0x70:0xff07, using default rate 32000 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)lists.sf.net> ACPI: PCI Interrupt 0000:06:05.1[A] -> GSI 17 (level, low) -> IRQ 17 unknown card 0x878-0x70:0xff07, using default rate 32000 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)lists.sf.net> ACPI: PCI Interrupt 0000:06:06.1[A] -> GSI 18 (level, low) -> IRQ 18 unknown card 0x878-0x70:0xff07, using default rate 32000 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)lists.sf.net> ACPI: PCI Interrupt 0000:06:07.1[A] -> GSI 19 (level, low) -> IRQ 19 ><> Nathan Stratton CTO, Voila IP Communications nathan at robotics.net nathan at voilaip.com http://www.robotics.net http://www.voilaip.com

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[alsa-devel] pcm->private_data is NULL in alsa core!!
by Pharaoh . 22 Jun '07

22 Jun '07

Hi I am writing an alsa driver for my omap based board. I am testing it using cat /dev/audio currently, i.e. I have OSS emulation turned on. This is an initial round of testing and I will use alsa-libs and utils later. My problem is: I have set the pcm->private_data = my_chip after registering the pcm ops, after referring various standard drivers in alsa tree. My module is insmoded properly but after I do snd_card_register(card) the pcm->private_data is corrupted !! Before this step, the pcm->private_data is intact. But after looking at the snd_card_register(), I realized that it doesn't even touch the pcm pointer. So, when I cat the device I get huge segfault in snd_pcm_attach_substream(), since pcm->private_data is NULL but surprisingly all the other fields are intact in pcm pointer. Even the OSS code doesn't touch the pcm pointer, what must be wrong? I have checked my source atleast 5 times and couldn't find anything in it. -pharaoh.

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[alsa-devel] AT91 SoC PCM Capture
by Paul Kavan 22 Jun '07

22 Jun '07

I am wondering what piece of code contains the capture functionality for the at91 soc. I am hoping to see how and when the pcm activates the ssc at read time. I just cannot figure out why TD0 would behave at such with this set of code and yet behave correctly with my polling code. Not a criticism....just wanting to figure this out. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

2 1

[alsa-devel] Osprey 440 almost supported!!! Need X9221 i2c support!
by Nathan Allen Stratton 22 Jun '07

22 Jun '07

This card does not use analog bt878a inputs, but rather the digital input. To get audio working I set the mask to 000303, first two bits are for the audio rate, 00 is 32000, 01 is 41000, 10 is 4800. The first two bits in the next byte are for balanced or unbalanced audio input. 0x00 for unbalanced and 0x100 for balanced. So I am using: modprobe bttv card=140 bttv_gpio=1 bttv_verbose=2 bttv_debug=1 gpiomask=0x020303 audioall=0xFFFDFC bttv: Bt8xx card found (0). bttv0: Bt878 (rev 17) at 0000:08:04.0, irq: 24, latency: 132, mmio: 0xc8600000 bttv0: detected: Osprey-440 [card=140], PCI subsystem ID is 0070:ff07 bttv0: using: Osprey 440 [card=140,insmod option] bttv0: gpio config override: mask=0x20303, mux=0xfffdfc,0xfffdfc,0xfffdfc,0xfffdfc bttv0: gpio: en=00000000, out=00000000 in=00feffff [pre-init] bttv0: risc main @ 37f25000 bttv0: gpio: en=00000000, out=00000000 in=00feffff [init] bttv0: BT878A ARESET bttv0: tveeprom i2c attach [addr=0x50,client=tveeprom] bttv0: using tuner=-1 bttv0: i2c: checking for TDA9887 @ 0x86... not found bttv0: registered device video0 bttv0: registered device vbi0 bttv0: gpio: en=00020303, out=00000000 in=00fcfcfc [audio: mute] bttv0: video mux: input=0 mux=2 bttv0: gpio: en=00020303, out=00000000 in=00fcfcfc [audio: mute] bttv0: PLL: 28636363 => 35468950 .<7>bttv0: gpio: en=00020303, out=00000000 in=00fcfcfc [audio: mute] ok So the issue now is that the audio is WAY to high so it is very distorted. The audio is controlled via X9221A chips on the i2c bus at 0x50, there are not drivers for X9221 so I can't lower the audio levels. Anyone able to help me with this last part??? ><> Nathan Stratton CTO, Voila IP Communications nathan at robotics.net nathan at voilaip.com http://www.robotics.net http://www.voilaip.com

2 2

[alsa-devel] Driver for Creamware / SonicCore Pulsar, which information needed?
by Kaspar Schleiser 22 Jun '07

22 Jun '07

Hey, I've got a Creamware Pulsar DSP card. As it actually has NO support for linux, I asked the vendor whether support was planned or if I could get necessary technical information. they answered they'd not have concrete plans on supporting linux and what information I'd need for the development of a driver. As I never wrote a driver, I don't know that, so maybe you could help me giving a useful answer ... What do we need to write a driver? Would anyone be interested in writing a driver? I could lend my card. It's a high quality multichannel card with 3 DSP chips on it. Cheers Kaspar

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[alsa-devel] Steps get aplay and arecord working on an embedded board
by Pharaoh . 22 Jun '07

22 Jun '07

Hi I have to test my audio driver on an omap based board. I have got the lib and utils on my board but dont know the exact steps to follow for using aplay etc. Can anyone point me to relevant documentation? Couldn't find any. Or just brief outline about the steops to follow would be great help. -pharaoh

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[alsa-devel] What is CONFIG_I2C_MODULE?
by Timur Tabi 21 Jun '07

21 Jun '07

In a lot of ALSA drivers, I see this: #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) What is CONFIG_I2C_MODULE? I've searched all the Kconfigs, and I cannot see this configuration option. -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] HDA - Change default configuration for Fujitsu Si3515.
by Tobin Davis 21 Jun '07

21 Jun '07

This patch changes the default configuration for the Fujitsu Siemens Si3515 Laptop. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Help with SoC alsa.conf
by Paul Kavan 21 Jun '07

21 Jun '07

I am trying to determine an appropriate conf file for my AT91SAM9260ek running linux-2.6.22-rc1 and the following alsa drivers: <6>Advanced Linux Sound Architecture Driver Version 1.0.14rc4 (Wed May 09 09:51:39 2007 UTC). <6>ASoC version 0.13.0 <4>AT91SAM9260EK W6811 Driver <1> <6>w6811: W6811 PCM SoC Audio 0.1 <6>asoc: W6811 <-> at91-ssc0 mapping ok <6>grh_ek_w6811: W6811 PCM in Slave Mode <6>ALSA device list: <6> #0: GRH_EK_W6811 (W6811) I have installed alsa-lib-1.0.14a and alsa-utils-1.0.14 on the at91sam9260. The default conf file produces sound using speaker-test and aplay, but will not record using arecord. Frank Mandarino has been kind enough to lend me a barebones conf that worked for him: pcm.!default { type hw card 0 } pcm.hw { type hw card 0 device 0 subdevice 0 } ctl.!default { type hw card 0 } ctl.hw { type hw card 0 } but produces the following for me: /usr/share/alsa $ aplay -l **** List of PLAYBACK Hardware Devices **** ALSA lib conf.c:3949:(snd_config_expand) Unknown parameters 0 ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL hw:0 aplay: device_list:213: control open (0): Invalid argument ALSA is *very* new to me and I am unsure how to interpret any of this. I am using a dumb W6811 pcm codec. I am fairly confident the machine/codec driver is working as expected, so I need to determine what I can do to get alsa to play nice with it as well. When I use the default alsa.conf produced from making alsa-libs, I get the following: /usr/share/alsa $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: GRHEKW6811 [GRH_EK_W6811], device 0: W6811 W6811-PCM-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 record and playing do not seem to work. When I try: /usr/share/alsa $ arecord -d5 test.wav Recording WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono Aborted by signal Interrupt... It does not stop at 5 seconds....I have to interrupt it manually. Furthermore, it does not appear to record anything as when I: /usr/share/alsa $ aplay test.wav Playing WAVE ' test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono The speaker pops once, but no audio. Finally, when I use speaker-test, I get some weird results. Doing: /usr/share/alsa $ speaker-test -r8000 -c1 -f1000 -tsine -l1 speaker-test 1.0.14 Playback device is default Stream parameters are 8000Hz, S16_LE, 1 channels Sine wave rate is 1000.0000Hz Rate set to 8000Hz (requested 8000Hz) Buffer size range from 64 to 32768 Period size range from 32 to 8192 Using max buffer size 32768 Periods = 4 was set period_size = 8192 was set buffer_size = 32768 0 - Front Left Time per period = 5.086305 Gives me a wave composed of both the 1k sine and likely a 2k sine. If I go much below 1k, I get a periodic wave, but it looks more like a heart beat. My codec expects u-law, so that may be the problem with speaker-test. However, the fact the aplay and arecord are not working give me pause. I would be greatly appreciative of any insights, pointers, etc. If there is more information that would be helpful, please ask. Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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Re: [alsa-devel] OSS/4Front
by Ash Willis 21 Jun '07

21 Jun '07

Hi Clemens, > You could compare <http://manuals.opensound.com/devlists/Linux.html> > with ALSA's sound card matrix, but the latter tends to be out of date. > Yeah, that thing is oooooold. Hopefully the new wiki on't be too bad. > > With the recent open sourcing of OSS, it would be nice to port code > > to ALSA for devices that are only currently covered by OSS. > > AFAICS all devices supported by the open-source OSS are also supported > by ALSA. > Ah, looks can be deceiving. Although both systems have 'geode' support, it seems that the OSS driver supports the CS5530 VSA2 and the SC1200. Also, there's an open bug for the Neomagic 256 driver, which may have a solution hiding away in OSS somewhere (someone with the hardware is hopefully going to help me look into that). There's a couple of things right off the bat and I don't know ALSA very well at all compared to most of you. And yeah, I know this stuff isn't in great demand, but it'd be nice if people weren't able to say that OSS can do XYZ and ALSA can't. Ash -- Get a Free E-mail Account at Mail.com! Choose From 100+ Personalized Domains Visit http://www.mail.com today

1 0

[alsa-devel] OSS/4Front
by Ash Willis 21 Jun '07

21 Jun '07

Hi, Does anyone know if there's any kind of list showing a comparison of hardware support between ALSA and OSS/4Front? With the recent open sourcing of OSS, it would be nice to port code to ALSA for devices that are only currently covered by OSS. There has to be at least a few devices that we can grab support for, right? It may at least provide us with insight into a few bugfixes. Ash -- Get a Free E-mail Account at Mail.com! Choose From 100+ Personalized Domains Visit http://www.mail.com today

4 3

[alsa-devel] Yamaha Cards
by derek＠darlingsong.net 21 Jun '07

21 Jun '07

Hi all, first post for me. I have a Yamaha DS2416 mixer card and Yamaha SW1000XG sound card that I would like to use onLinux (Fedora 6). I have used them on the "other" OS, along with ProTools, etc. Does anyone know if Linux drivers are available? I have looked around, and it seems not to be the case. any help would bve appreciated. Derekd Vancouver, BC

2 1

[alsa-devel] No time for cs4624
by chris＠idlelion.net 21 Jun '07

21 Jun '07

I really wanted to work on the cs4624 suspend resume bug (#305), but there is no time, especially given the lack of technical information. I'm wondering if I were to lend or give a laptop that exhibits the problem to one of you alsa developers, if that would be useful and provide some impetus to fix the bug, or if it would just be a big pain? Opinions? Chris

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[alsa-devel] [1/2] 2.6.22-rc5: known regressions
by Michal Piotrowski 21 Jun '07

21 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc5. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Unclassified Subject : Kernel hang on CMOS_READ References : http://lkml.org/lkml/2007/6/3/138 Submitter : Rodrigo Luiz <troops.of.doom(a)gmail.com> Status : Unknown Subject : Device hang when offlining a CPU due to IRQ misrouting References : http://lkml.org/lkml/2007/5/31/419 Submitter : Darrick J. Wong <djwong(a)us.ibm.com> Handled-By : Siddha, Suresh B <suresh.b.siddha(a)intel.com> Status : problem is being debugged ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Status : Unknown FS Subject : 2.6.22-rc4-git5 reiserfs: null ptr deref. References : http://lkml.org/lkml/2007/6/13/322 Submitter : Randy Dunlap <randy.dunlap(a)oracle.com> Handled-By : Vladimir V. Saveliev <vs(a)namesys.com> Status : problem is being debugged PCI Subject : PCI setup hangs on Asus Notebook (nolapic helps) References : http://lkml.org/lkml/2007/6/14/161 Submitter : Malte Cornils <malte(a)cornils.net> Status : unknown Regards, Michal -- LOG http://www.stardust.webpages.pl/log/

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Re: [alsa-devel] Using aplay and arecord on an ARM
by Paul Kavan 20 Jun '07

20 Jun '07

On 6/20/07, Paul Kavan <pkavan(a)gmail.com> wrote: > > Okay: > > Here is where I ran into troubles yesterday. I am working on configuring > alsa-utils. When I do: > > CC=arm-linux-gcc ./configure --target=arm-linux --with-versioned=no > --prefix=/usr --host=i686-pc-linux-gnu > > I get the error: > > grhuser@pcL7:$ CC=arm-linux-gcc ./configure --target=arm-linux > --with-versioned=no --prefix=/usr --host=i686-pc-linux-gnu > configure: WARNING: If you wanted to set the --build type, don't use > --host. > If a cross compiler is detected then cross compile mode will be used. > checking for a BSD-compatible install... /usr/bin/install -c > checking whether build environment is sane... yes > checking for gawk... no > checking for mawk... mawk > checking whether make sets $(MAKE)... yes > checking for i686-pc-linux-gnu-strip... no > checking for strip... strip > checking whether NLS is requested... yes > checking for msgfmt... /usr/bin/msgfmt > checking for gmsgfmt... /usr/bin/msgfmt > checking for xgettext... /usr/bin/xgettext > checking for msgmerge... /usr/bin/msgmerge > checking for style of include used by make... GNU > checking for i686-pc-linux-gnu-gcc... arm-linux-gcc > checking for C compiler default output file name... a.out > checking whether the C compiler works... yes > checking whether we are cross compiling... yes > checking for suffix of executables... > checking for suffix of object files... o > checking whether we are using the GNU C compiler... yes > checking whether arm-linux-gcc accepts -g... yes > checking for arm-linux-gcc option to accept ISO C89... none needed > checking dependency style of arm-linux-gcc... gcc3 > checking build system type... i686-pc-linux-gnu > checking host system type... i686-pc-linux-gnu > checking for ld used by GCC... > /home/grhuser/Linux_Builds/buildroot/build_arm/staging_dir/arm-linux-uclibc/bin/ld > checking if the linker > (/home/grhuser/Linux_Builds/buildroot/build_arm/staging_dir/arm-linux-uclibc/bin/ld) > is GNU ld... yes > checking for shared library run path origin... done > checking for CFPreferencesCopyAppValue... no > checking for CFLocaleCopyCurrent... no > checking for GNU gettext in libc... no > checking for iconv... no, consider installing GNU libiconv > checking for GNU gettext in libintl... no > checking whether to use NLS... no > checking for i686-pc-linux-gnu-gcc... (cached) arm-linux-gcc > checking whether we are using the GNU C compiler... (cached) yes > checking whether arm-linux-gcc accepts -g... (cached) yes > checking for arm-linux-gcc option to accept ISO C89... (cached) none > needed > checking dependency style of arm-linux-gcc... (cached) gcc3 > checking for a BSD-compatible install... /usr/bin/install -c > checking whether ln -s works... yes > checking for ALSA CFLAGS... > checking for ALSA LDFLAGS... -lasound -lm -ldl -lpthread > checking for libasound headers version >= 1.0.12... not present. > configure: error: Sufficiently new version of libasound not found. > > I know there has to be a flag or something here I need to set. Does anyone > know what that might be? > > Paul > I think what occurs here is that the configure script is using my cross-compilers and associated libs in $HOME/Linux_Builds/buildroot/build_arm/staging_dir/ while the libs are actually given in $HOME/Linux_Builds/buildroot/build_arm/root/usr/lib/ I am looking at the README and INSTALL and do not see a solution to this. Does anyone have a solution to this? Thanks. Paul

4 11

[alsa-devel] Kernel 2.6.21 M-Audio Delta 1010L
by Suso Comesaña 20 Jun '07

20 Jun '07

Hello... My name is Suso Comesaña...I have 2 computers with a M-Audio card Delta 1010L...the problem is the following one: With kernel 2.6.17-2-multimedia-486... I can record sounds perfectly... but, kernel 2.6.21... I cannot record nothing... You have some idea?... You need that she sends some information to you? Tanks... Suso Comesaña. Peatón. susobaco(a)yahoo.es Usuario de Musix, GNU-Linux, basada en Debian/Knoppix. 100% Software Libre. ¡Descubre una nueva forma de obtener respuestas a tus preguntas! Entra en Yahoo! Respuestas. LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com ____________________________________________________________________________________ ¡Descubre una nueva forma de obtener respuestas a tus preguntas! Entra en Yahoo! Respuestas. http://es.answers.yahoo.com/info/welcome

2 1

[alsa-devel] Output levels on SBLive! 5.1 on ALSA vs WIndows
by ChernYue Kwok 20 Jun '07

20 Jun '07

Hello, I am using an SBLive! 5.1 sound card (emu10k1 chipset) with 6 channel output. Due to the nature of my work, I need to be able to play sound through all 6 channels at the same output level to an amplifier. Using Linux and the emu10k1 ALSA driver, I measured output voltage levels on an oscilloscope while playing a sine wave on all 6 channels, and I got three different output voltage amplitudes for the front, rear and center/subwoofer channels. Repeating the same test with the same card on WIndows using the Creative SBLive drivers from their website, I am able to get the same output voltage amplitude on all 6 channels. For both tests, all output levels in the relevant mixers were set to their maximum levels. I would like to know if this output normalizing 'feature' of the SBLive Windows driver can be found in the emu10k1 ALSA driver, or in any other ALSA driver. Failing that, is there any other solution to set output levels in the driver, or am I doomed to twiddle in alsamixer to get the output levels that I want? Thanks in advance.

2 3

[alsa-devel] Using aplay and arecord on an ARM
by Paul Kavan 20 Jun '07

20 Jun '07

Liam: I have managed to get the alsa-libs and aplay, arecord, and speaker-test on my board, but I am unsure how to get them to play. I have tried making the device nodes and am using a simple alsa.conf, but still no joy. I am sure there is something simple I am missing. Trying to find some documentation on this. I am assuming there is some disconnect between the interface and my conf file. I have my alsa.conf file in /usr/share/alsa/ and it is: pcm.!default { type hw card 0 } pcm.hw { type hw card 0 device 0 subdevice 0 } ctl.!default { type hw card 0 } ctl.hw { type hw card 0 } ctl.w6811 { type hw card 0 } pcm.w6811 { type hw card 0 device 0 subdevice 0 } ctl.!default = ctl.w6811 pcm.!default = pcm.w6811 ctl.hw { @args[ CARD ] @args.CARD { type string default "0" } type hw card $CARD } pcm.hw { @args[ CARD DEV SUBDEV ] @args.CARD { type string default "0" } @args.DEV { type integer default 0 } @args.SUBDEV { type integer default 0 } type hw card $CARD device $DEV subdevice $SUBDEV } I have some /proc/ entries: /proc/asound $ ls GRHEKW6811 cards pcm version card0 devices timers /proc/asound $ cat cards 0 [GRHEKW6811 ]: W6811 - GRH_EK_W6811 GRH_EK_W6811 (W6811) /proc/asound $ cat devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer /proc/asound $ cat pcm 00-00: W6811 W6811-PCM-0 : : playback 1 : capture 1 /proc/asound $ cat timers G0: system timer : 10000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE /proc/asound $ cat version Advanced Linux Sound Architecture Driver Version 1.0.14rc4 (Wed May 09 09:51:39. Now, into GRHEKW6811 (which takes me to card0) I get a pcm0c and pcm0p directory. Using cat info, I get: /proc/asound/card0/pcm0c $ cat info card: 0 device: 0 subdevice: 0 stream: CAPTURE id: W6811 W6811-PCM-0 name: subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 /proc/asound/card0/pcm0c $ cd ../pcm0p/ /proc/asound/card0/pcm0p $ cat info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: W6811 W6811-PCM-0 name: subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 I then did: mknod -m 666 /dev/snd/pcm0p c 116 0 mknod -m 666 /dev/snd/pcm0c c 116 0 However, when I try speaker-test, I get: /usr/lib/bin $ ./speaker-test -D pcm0p speaker-test 1.0.14 Playback device is pcm0p Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM pcm0p Playback open error: -2,No such file or directory Same for default: /usr/lib/bin $ ./speaker-test -D default speaker-test 1.0.14 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory When I try aplay -l and arecord -l, I get: /usr/lib/bin $ ./aplay -l aplay: device_list:204: no soundcards found... /usr/lib/bin $ ./arecord -l arecord: device_list:204: no soundcards found... I am at a loss here. Anyone have any advice, pointers, tips, etc...... Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

4 9

[alsa-devel] ALSA always writes _whole_ periods, right?
by Joachim Förster 20 Jun '07

20 Jun '07

Hi ALSA devs, I have a quick question regarding the filling of the usual ring buffer in RAM (used by sound controllers via DMA), just to be really sure: Is it true, that ALSA always writes (via mmap'ing) _whole_ periods? I took snd-dummy and modified it to use "ack()" method and tried to confirm, if the above is true _without_ memory mapping (classical read/write). It seem's to be true, ALSA is "filling up" the last (not always complete) period - e.g. if the played wav file is not a multiple of periodsize in length. But well, with mmap, I cannot see the copying action (done by ALSA library, right?) ... so, thus my question ;-) . Thanks, Joachim

2 3

[alsa-devel] Surround problems with cs46xx
by Vasilis Manwlopoulos 20 Jun '07

20 Jun '07

Hi, Sorry for bothering the list again, but I have major problems with the surround 5.1 sound. I have manually compiled/installed th latest alsa-1.0.14 drivers but nothing improved. I am using Ubuntu 7.04 and my soundcard is the Hercules Fortissimo III. This is, as far as I know by reading on Internet, supported by the module snd-cs46xx in ALSA. I have enabled the surround sound by creating the following .asoundrc file. pcm.!default { type plug slave.pcm "surround51" slave.channels 6 route_policy duplicate } The problem is that sound pauses whenever I try to do something "heavy" such as opening a new firefox window or generally another app. Also, there are delays and echo between the front and the rear speakers from time to time. I think that the problem is that, this alsa setup(meaning surround5.1 and the specific soundcard) places a heavy load on the cpu. There are 99% cpu usage times when the only thing active is a player playing a "normal bitrate" mp3. When I delete the .asoundrc file and I switch to default stereo sound using only the 2 front speakers the problems disappear. Can anyone suggest anything about this? Thanks in Advance V.Manolopoulos

3 3

[alsa-devel] Kernel 2.6.21 M-Audio Delta 1010L
by Suso Comesaña 20 Jun '07

20 Jun '07

Hello... My name is Suso Comesaña...I have 2 computers with a M-Audio card Delta 1010L...the problem is the following one: With kernel 2.6.17-2-multimedia-486... I can record sounds perfectly... but, kernel 2.6.21... I cannot record nothing... You have some idea?... You need that she sends some information to you? Tanks... Suso Comesaña. Peatón. susobaco(a)yahoo.es Usuario de Musix, GNU-Linux, basada en Debian/Knoppix. 100% Software Libre. ¡Descubre una nueva forma de obtener respuestas a tus preguntas! Entra en Yahoo! Respuestas. LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com ____________________________________________________________________________________ ¡Descubre una nueva forma de obtener respuestas a tus preguntas! Entra en Yahoo! Respuestas. http://es.answers.yahoo.com/info/welcome

1 0

[alsa-devel] [PATCH] Output MFG information for HDA devices
by Jonathan Phénix 19 Jun '07

19 Jun '07

Hi, this patch output information about the MFG (Modem Function Group) to the /proc/asound/card?/codec#? file. This information is important to identify if a HDA device has a modem or not. Regards, Jonathan Phénix Technical specialist / Linuxant www.linuxant.com jonathan(a)linuxant.com

2 3

Re: [alsa-devel] Writing an AT91SAM9260-EK Dumb Codec Driver
by Paul Kavan 19 Jun '07

19 Jun '07

On 6/19/07, Liam Girdwood <lg(a)opensource.wolfsonmicro.com> wrote: > > On Mon, 2007-06-18 at 14:30 -0500, Paul Kavan wrote: > > > > > Am I at a point now where I can try to get alsa-lib and alsa-util in > > place? If so, what version? I found some tars at: > > > > http://www.alsa-project.org/alsa/ftp/lib/ > > > > Is that the best place to get them? I am using buildroot, but not sure > > if ALSA stuff is ready for primetime there. I guess I should check on > > this asap. > > > > Afaik, there is an issue with buildroot atm (please someone correct me > if it's now fixed) when it comes to building alsa-lib for ARM. (please > search archives for details). No, I think you are right. I have tried a few recommended patches, but they do not work. Fwiw, I use openembedded for all my embedded builds without any problems > atm. http://www.openembedded.org At some point when I have some time, I am going to give it a try. I have managed to build the libs and have those in my rfs using: CC=arm-linux-gcc ./configure --host=arm-linux --prefix=$HOME/buildroot/build_arm/root/usr --with-versioned=no ...and then... make install With this, I get the /usr/ entries in my rfs. However, I have had troubles building alsa-utils. When I try: CC=arm-linux-gcc ./configure --host=arm-linux --prefix=$HOME/buildroot/build_arm/root/usr --with-versioned=no I get: grhuser@pcL7:$ CC=arm-linux-gcc ./configure --host=arm-linux --prefix=$HOME/buildroot/build_arm/root/usr --with-versioned=no configure: WARNING: If you wanted to set the --build type, don't use --host. If a cross compiler is detected then cross compile mode will be used. checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for gawk... no checking for mawk... mawk checking whether make sets $(MAKE)... yes checking for arm-linux-strip... arm-linux-strip checking whether NLS is requested... yes checking for msgfmt... /usr/bin/msgfmt checking for gmsgfmt... /usr/bin/msgfmt checking for xgettext... /usr/bin/xgettext checking for msgmerge... /usr/bin/msgmerge checking for style of include used by make... GNU checking for arm-linux-gcc... arm-linux-gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... yes checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether arm-linux-gcc accepts -g... yes checking for arm-linux-gcc option to accept ISO C89... none needed checking dependency style of arm-linux-gcc... gcc3 checking build system type... i686-pc-linux-gnu checking host system type... arm-unknown-linux-gnu checking for ld used by GCC... /home/grhuser/buildroot/build_arm/staging_dir/arm-linux-uclibc/bin/ld checking if the linker (/home/grhuser/buildroot/build_arm/staging_dir/arm-linux-uclibc/bin/ld) is GNU ld... yes checking for shared library run path origin... done checking for CFPreferencesCopyAppValue... no checking for CFLocaleCopyCurrent... no checking for GNU gettext in libc... no checking for iconv... no, consider installing GNU libiconv checking for GNU gettext in libintl... no checking whether to use NLS... no checking for arm-linux-gcc... (cached) arm-linux-gcc checking whether we are using the GNU C compiler... (cached) yes checking whether arm-linux-gcc accepts -g... (cached) yes checking for arm-linux-gcc option to accept ISO C89... (cached) none needed checking dependency style of arm-linux-gcc... (cached) gcc3 checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for ALSA CFLAGS... checking for ALSA LDFLAGS... -lasound -lm -ldl -lpthread checking for libasound headers version >= 1.0.12... not present. configure: error: Sufficiently new version of libasound not found. Not sure what is the problem here. I have file:///home/grhuser/buildroot/build_arm/root/usr/lib/libasound.so.2.0.0 in my rfs. Any suggestions? Paul Liam > > -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

2 1

[alsa-devel] [PATCH] hda: add eapd support to additional idt codecs
by Matthew Ranostay 19 Jun '07

19 Jun '07

Added support for EAPD on the 927x, and 9227-8 IDT HDA codecs. Enabling EAPD powers the internal speaker amp otherwise there is no sound on systems with an internal amp. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 96495fe7cc13 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Jun 14 12:58:23 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Mon Jun 18 17:10:38 2007 -0400 @@ -832,6 +832,18 @@ static void stac92xx_set_config_regs(str } } +static void stac92xx_enable_eapd(struct hda_codec *codec) +{ + /* Configure GPIO0 as output */ + snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_DIRECTION, 0x00000001); + /* Configure GPIO0 as CMOS */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); + /* Assert GPIO0 high */ + snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_DATA, 0x00000001); + /* Enable GPIO0 */ + snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_MASK, 0x00000001); +} + /* * Analog playback callbacks */ @@ -2193,7 +2205,8 @@ static int patch_stac927x(struct hda_cod } spec->multiout.dac_nids = spec->dac_nids; - + stac92xx_enable_eapd(codec); + err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { if (spec->board_config < 0) { @@ -2254,18 +2267,7 @@ static int patch_stac9205(struct hda_cod spec->mixer = stac9205_mixer; spec->multiout.dac_nids = spec->dac_nids; - - /* Configure GPIO0 as EAPD output */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, 0x00000001); - /* Configure GPIO0 as CMOS */ - snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); - /* Assert GPIO0 high */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, 0x00000001); - /* Enable GPIO0 */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_MASK, 0x00000001); + stac92xx_enable_eapd(codec); err = stac92xx_parse_auto_config(codec, 0x1f, 0x20); if (!err) {

2 1

[alsa-devel] Is snd_magic_kcalloc() deprecated?
by Pharaoh . 19 Jun '07

19 Jun '07

2 1

[alsa-devel] Fwd: Problems with hda_intel, Santa Rosa, and suspend
by Matt Mullins 19 Jun '07

19 Jun '07

I posted this to the LKML, I thought it should be reproduced on this list. > > On Sunday, 17 June 2007 03:36, Matt Mullins wrote: > > > I just received a Dell Latitude D630, with the new Intel Santa Rosa > > > platform. Currently, the only major driver issue I have is sound. It > > > worked fine in Ubuntu Feisty's 2.6.20 kernel, but now I am using Gutsy > > > so I can have graphics drivers. Gutsy's 2.6.22-rc3-based kernel no > > > longer recognized my soundcard, a Sigmatel STAC9205, which uses the > > > hda_intel driver. I have since fixed that problem by compiling the > > > hda_intel driver into the kernel (I used 2.6.22-rc4 vanilla sources), > > > instead of a module, but it is still a bug that it does not work as a > > > module. > > > > > > All the following debug information was obtained from kernel 2.6.22-rc4. > > > > > > However, the bug that is currently affecting me is that sound no > > > longer works after using ACPI suspend-to-RAM or swsusp. I compiled > > > ALSA debugging in, and as soon as I resume, I get tons of messages > > > like: > > > [ 8.474830] hda-intel: send_cmd timeout: IRS=0x1, val=0xd0970500 > > > [ 8.474954] hda-intel: send_cmd timeout: IRS=0x1, val=0xd0af0009 > > > [ 8.475078] hda-intel: send_cmd timeout: IRS=0x1, val=0xd0a70500 > > > [ 8.475207] hda-intel: send_cmd timeout: IRS=0x1, val=0xd0bf0009 > > > (etc) > > > > > > There may be more before that, but that is more than the kernel > > > message buffer can hold, so I can't ever see it. I get similar > > > timeout messages each time a PCM gets set up: > > > [ 8.972859] hda_codec_setup_stream: NID=0x10, stream=0x5, channel=0, format=0 > > > x11 > > > [ 8.972987] hda-intel: send_cmd timeout: IRS=0x1, val=0x1070650 > > > [ 9.006071] hda-intel: send_cmd timeout: IRS=0x1, val=0x1020011 > > > [ 9.006081] hda_codec_setup_stream: NID=0x11, stream=0x5, channel=0, format=0 > > > x11 > > > [ 9.006207] hda-intel: send_cmd timeout: IRS=0x1, val=0x1170650 > > > [ 9.016117] hda-intel: send_cmd timeout: IRS=0x1, val=0x1120011 > > > > > > I read some of sound/pci/hda/hda_intel.c, specifically the code that > > > output those messages. After octupling my kernel log buffer (to 1MB), > > > I noticed that for the first set of timeouts (immediately upon > > > resume), all the "codec" nibbles are odd, the last word of the val > > > ("verb" and "para") are all either 0x0500 or 0x0009. For the second > > > set (when something tries to use the card) that last word is 0x0650 or > > > 0x0011. > > > > > > I have yet to go back and re-test compiling snd-hda-intel as a module. On 6/17/07, Matt Mullins <mokomull(a)gmail.com> wrote: > I've been reading up on the Intel HD Audio specifications, and I found > that the problem is most likely that the driver has to resort to PI/O > (single_cmd) mode. I really haven't had enough time or experience to > figure out what is actually causing this. This is the first time I've > done any kernel development; thankfully, Intel has good documentation. > I'm going to keep hacking to see what information I can figure out. > > I just tried 2.6.22-rc5, and no behavior has changed.

3 3

[alsa-devel] Bad digital out of ASUS M2V
by Mike Crash 19 Jun '07

19 Jun '07

In ALSA 1.0.14 HDA Intel with ASUS M2V motherboard - detected the chip ALC660VD, but digital out is missing. I added workaround - added new model name 3stack-660-digout with digital output, diff file follows: ---------------------------------------------- 117a118 > ALC660VD_3ST_DIG, 9466a9468 > [ALC660VD_3ST_DIG]= "3stack-660-digout", 9478c9480 < SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST), --- > SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), 9501a9504,9516 > [ALC660VD_3ST_DIG] = { > .mixers = { alc861vd_3st_mixer }, > .init_verbs = { alc861vd_volume_init_verbs, > alc861vd_3stack_init_verbs }, > .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), > .dac_nids = alc660vd_dac_nids, > .dig_out_nid = ALC861VD_DIGOUT_NID, > .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), > .adc_nids = alc861vd_adc_nids, > .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), > .channel_mode = alc861vd_3stack_2ch_modes, > .input_mux = &alc861vd_capture_source, > },

2 1

[alsa-devel] question about bluez using alsa and patches
by Walt Shekrota 19 Jun '07

19 Jun '07

I was told that when using the btsco(bluez) and/or plugs bluetooth headset software a patch is always required. In the Gentoo community and perhaps everywhere it is suggested to use alsa compiled into your kernel. As I go to new kernels frequently these folks (sco headset) seem to have no such forward support ie they have not worked on the patches yet for the newer kernels. Am I wrong or at some point in time should this app not be weaned from using patches? Is it the fact they use sco that they have to patch? Is there not an alsa library they can program to instead. I guess I don't understand the overall design so I can't see why this goes on forever. I'm somewhat apprehensive about trusting others to continually modify my kernel. Am I out in left field here? Can anyone explain why they must choose this path? Thanks very much. -Walt

3 3

[alsa-devel] Difference between substream->dma_buffer and substream->runtime->dma_buffer_p?
by Timur Tabi 19 Jun '07

19 Jun '07

The snd_pcm_substream structure has a snd_dma_buffer structure. snd_dma_buffer looks like this: struct snd_dma_buffer { struct snd_dma_device dev; /* device type */ unsigned char *area; /* virtual pointer */ dma_addr_t addr; /* physical address */ size_t bytes; /* buffer size in bytes */ void *private_data; /* private for allocator; don't touch */ }; snd_pcm_substream also has a snd_pcm_runtime structure, which looks like this: struct snd_pcm_runtime { ... /* -- DMA -- */ unsigned char *dma_area; /* DMA area */ dma_addr_t dma_addr; /* physical bus address (not accessible from main CPU) */ size_t dma_bytes; /* size of DMA area */ struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */ Why the redundancy? Not only does snd_pcm_runtime have dma_area, dma_addr, and dma_bytes, but it also has a pointer to a dns_dma_buffer structure, which also has this information! So given a snd_pcm_substream structure, you can obtain the physical address of the DMA buffer three ways: substream->dma_buffer.addr substream->runtime->dma_addr substream->runtime->dma_buffer_p->addr When will these three variable contain different values? If the answer is never, then why do they all exist? -- Timur Tabi Linux Kernel Developer @ Freescale

3 2

[alsa-devel] [PATCH] ASoC S3C24xx machine drivers - lm4857-h missing patch
by Graeme Gregory 19 Jun '07

19 Jun '07

Somewhere when dealing with the s3c24xx patches and requested changes I managed to miss this file on re-submission. Its a header file required for compilation of neo1973-wm8573.c and should be a no brainer. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Graeme Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

2 1

[alsa-devel] Changing the sample rate/channels of an already opened PCM stream
by Ed Schouten 19 Jun '07

19 Jun '07

Hello, I am the author of the command line music player Herrie (http://herrie.info/). For a long time folks have asked me to add support for OSS to Herrie, so this morning I decided to install Linux on a spare disk of mine (Ubuntu 7.04, FWIW) to hack on ALSA support for Herrie. After (a real lot of) hours of reading code, reading the Doxygen stuff, etc. I now have a functional ALSA output module for Herrie. It was quite hard to get it working, so I guess I owe myself a beer! ;-) Anyway, there still is one disadvantage to my module; each time I detect that a new track is going to be started that needs a different amount of channels or sample rate, I reopen the ALSA device, something that is just impossible. The reason why it's impossible: Herrie has a nifty chroot() functionality that allows users to trap the application in a certain directory. I have been trying to get the device to reset so I can change the sample rate, but for some reason, it just gets ignored. Can someone explain me how it's possible to reset the sample rate after preparing the handle? For your information, here's a link to the latest version of the source code of my ALSA module: http://g-rave.nl/gitweb?p=herrie;a=blob;f=herrie/src/audio_output_alsa.c Yours, -- Ed Schouten <ed(a)fxq.nl> WWW: http://g-rave.nl/

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[alsa-devel] Writing an AT91SAM9260-EK Dumb Codec Driver
by Paul Kavan 18 Jun '07

18 Jun '07

Hello all: I am working on a driver for a dumb codec to work with the SSC as a PCM on the AT91SAM9260. I have, with the gracious help of Frank Mandarino, have the glue code written without too many problems. However, I need some guidance on getting a few things configured correctly for my codec and then getting it to compile and load correctly. For configuration, I am using the Winbond W6811. For this, I am going to use a 256kHz BCLK and an 8kHz frame. Each sample consists of 8-bit u-law PCM, MSB first. When I execute a read from the codec to the processor, I need to make the reads on the falling edge of the BCLK at each sync pulse. When executing a write from the processor to the codec I need to write on the rising edge at each sync pulse. I think I the timing correct in my driver with: cmr_div = 187; /* PCM_CLK = ~96MHz/(2*187) = 256kHz */ period = 15; /* PCM_SYNC = PCM_CLK/(2*(15+1)) = 8000Hz */ However, I am not sure how to get the following correct for my needs: #define W6811_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE) Now to making the driver once the code is done. I am relatively new to writing linux drivers. In the 6 or so that I have written, I usually cross-compile them as modules and then insmod them. I have tried that here and am getting a strange warning that I am unsure of. I have a pretty simple Makefile and when I execute it, I get: make ARCH=arm CROSS_COMPILE=arm-linux- make -C ~/Linux_Builds/linux-2.6.21.5M=/home/grhuser/Door_Access_System/Remotes/Driver/sound modules make[1]: Entering directory `/home/grhuser/Linux_Builds/linux- 2.6.21.5' CC [M] /home/grhuser/Door_Access_System/Remotes/Driver/sound/grh_ek_w6811.o Building modules, stage 2. MODPOST 1 modules WARNING: "at91_ssc_dai" [/home/grhuser/Door_Access_System/Remotes/Driver/sound/grh_ek_w6811.ko] undefined! CC /home/grhuser/Door_Access_System/Remotes/Driver/sound/grh_ek_w6811.mod.o LD [M] /home/grhuser/Door_Access_System/Remotes/Driver/sound/grh_ek_w6811.ko make[1]: Leaving directory `/home/grhuser/Linux_Builds/linux- 2.6.21.5' As you can see, I am using linux-2.6.21.5 I have ALSA checked and AT91-SoC checked in my .config file. For this attempt to build the driver, I have a separate folder than contains: at91-ssc.h and at91-pcm.h. If there is a more correct way to do this and be able to debug it, please let me know. I have attached copies of my driver. I would appreciate any and all feedback and assistance on getting this running.

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[alsa-devel] .asoundrc
by Nobin Mathew 18 Jun '07

18 Jun '07

Hi, I am trying to write one .asoundrc file What i am currently doing is rate conversion My .asounrc file contains pcm.sirf3a { type rate slave { pcm "hw:0,0" rate 8000 } } I am using following command aplay -M -D sirf3a /usr/share/sounds/alsa/Side_Right.wav But it is giving me some errors aplay: set_params:909: Channels count non available Any idea why this? where I am going wrong?

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[alsa-devel] [ASoC] Missing shutdown() vectors in ASoC
by Nobin Mathew 18 Jun '07

18 Jun '07

Liam, Shutdown() vectors is missing in ASoC, we need this to ensure proper shutdown of the system. * codec device */ struct snd_soc_codec_device { int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); int (*suspend)(struct platform_device *pdev, pm_message_t state); int (*resume)(struct platform_device *pdev); }; /* SoC CPU DAI */ struct snd_soc_cpu_dai { /* DAI description */ char *name; unsigned int id; unsigned char type; /* DAI callbacks */ int (*probe)(struct platform_device *pdev); void (*remove)(struct platform_device *pdev); int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); unsigned int (*config_sysclk)(struct snd_soc_cpu_dai *cpu_dai, struct snd_soc_clock_info *info, unsigned int clk); /* DAI capabilities */ struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; struct snd_soc_dai_cap caps; /* SoC platform interface */ struct snd_soc_platform { char *name; int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); /* pcm creation and destruction */ /* SoC machine */ struct snd_soc_machine { char *name; int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); /* the pre and post PM functions are used to do any PM work before and * after the codec and DAI's do any PM work. */ int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); int (*suspend_post)(struct platform_device *pdev, pm_message_t state); int (*resume_pre)(struct platform_device *pdev); int (*resume_post)(struct platform_device *pdev); Nobin Mathew

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[alsa-devel] Query regarding pcm->private_data
by Pharaoh . 18 Jun '07

18 Jun '07

Hi I am reading 'Wring an alsa driver'. It says following about the private_data: You can allocate a record for the substream and store it in runtime->private_data. Usually, this is done in the open callback. Don't mix this with pcm->private_data. The pcm->private_data usually points the chip instance assigned statically at the creation of PCM, while the runtime->private_data points a dynamic data created at the PCM open callback. What does static and dynamic mean here? Apologies for this question if it sounds very novice, since my understanding is it doesn't mean static and dynamic in the usual way like stack and heap etc. Why I am being so paranoid about this has reason: In my ALSA driver I am assigning pcm->private_data = mychip, but it is NULL in the snd_pcm_attach_substream() function. I have checked my code many times and I tried with another driver which was working when tested using ALSA utils. Currently I am testing using the OSS emulation i.e. simply cat /dev/audio.

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[alsa-devel] Alsa mixer interface limitations?
by Ctirad Fertr 17 Jun '07

17 Jun '07

Hello! I'm currently in phase of adding support for higher sample rates into e-mu 1212m/1820m driver. These cards have a lot of physical inputs and outputs and also a bunch of internal buses used for capture, playback and internal routing between physical inputs and outputs. Both physical I/O and internal buses are shown in alsamixer, so one can set routing easily. It's a lot of switches, anyway. The card has three modes. The standard (or "single speed") mode is the only mode currently supported in emu10k driver. It is capable of 44.1/48kHz sample rate only. If we need higher sample rates, we have to switch the card into either "double speed" (for 88.2/96kHz) or "quadruple speed" (for 176.4/192kHz) mode. The problem is that the number of inputs, outputs and especially internal buses significantly changes at higher speed modes. The reason is that internal buses are 44.1/48kHz only and thus we must use two (or four) of them per channel. Also some physical I/O ports disappear at double and quadruple speeds (e.g. ADAT has 4 channles at 96k and 2 at 192k, S/PDIF is completely disabled at 192k). The question is how to deal with that in alsamixer? IMHO the best way would be to change the number of switches in alsamixer on the fly. Unfortunatelly, I'm afraid it's impossible. Another way would be to make a particular swiches inactive, though I don't know if it is a good idea and it wold require a lot of code anyway. Any ideas? Please help! Ctirad

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[alsa-devel] Got microphone working with SigmaTel STAC 9221 A1 on a Dell XPS M1210
by Eran Guendelman 17 Jun '07

17 Jun '07

I installed SuSE 10.2 on my new XPS M1210 laptop and found that the microphone was not working (headphones and internal speakers worked fine). I eventually got it working, and wanted to write an email describing how I did this. The two main important steps (which I will describe in more detail below) were: - Changing the default pin configurations ref922x_pin_configs in patch_sigmatel.c to values that were based on the windows driver's INI file but with one important tweak. - Running the driver using the "model=ref" parameter so that the above modified pin configurations are used. Note that it could be that my solution is overkill and there's actually a way to get this working with the unmodified alsa drivers... But I did not find any such solutions (e.g. I've tried setting "model=ref", "model=3stack", "model=5stack", but none of those fixed the microphone). In general, it seems that the solutions proposed on the internet to getting the SigmaTel codec working are very dependent on the exact type of system you've got... So this will likely only work for a Dell XPS M1210. Here is some more system info: "lspci -v" gives this as my audio card: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01) Subsystem: Dell Unknown device 01d7 Flags: bus master, fast devsel, latency 0, IRQ 233 Memory at efffc000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) My card uses the snd_hda_intel module, and the codec type (given by "cat /proc/asound/card0/codec#0") gives this information at the top: Codec: SigmaTel STAC9221 A1 Address: 0 Vendor Id: 0x83847680 Subsystem Id: 0x102801d7 Revision Id: 0x103601 So here's what I did (taking many days and many wrong steps before arriving at this solution): - I have a dual-boot system and installed the official DELL drivers for the sound card in Windows XP. My goal was to find which INI file corresponded to my installed sound card, so I did the following: - Using Windows device manager, I looked at the properties for the SigmaTel codec driver and found that one of the property items had a string that included the substring VEN_8384&DEV_7680&SUBSYS_102801D7 - The self-extracting DELL driver extracted files into c:\dell\drivers\r122161, and in the wdm subdirectory I searched for the above string and found that the files 92xxM.INI, Sthda.ini, and sthda64.ini all contained the above string and also had the suffix =92xxM8-2.ini appended to it. This led me to deduce that the relevant file was 92xxm8-2.ini - This is an excerpt from 92xxm8-2.ini: [HKR\Settings\pin\0A] CfgDflt = dword: 0x0221121E AltCfg = hex: 20,12,11,02 Action = hex: 80,00,87,00,81,0D [HKR\Settings\pin\0B] CfgDflt = dword: 0x408103FF [HKR\Settings\pin\0C] CfgDflt = dword: 0x0281123E AltCfg = hex: 22,12,11,02,30,12,A1,02 Bias = hex: FF,04,02 Action: hex: 0A, 0C [HKR\Settings\pin\0D] CfgDflt = dword: 0x90100310 [HKR\Settings\pin\0E] CfgDflt = dword: 0x408003F1 [HKR\Settings\pin\0F] CfgDflt = dword: 0x0221122F AltCfg = hex: 21,12,11,02 Action = hex: 80,00,87,00,81,0D [HKR\Settings\pin\10] CfgDflt = dword: 0x03451340 [HKR\Settings\pin\11] CfgDflt = dword: 0x40C003F2 [HKR\Settings\pin\15] CfgDflt = dword: 0x50A003F3 [HKR\Settings\pin\1B] CfgDflt = dword: 0x405003F4 - This will be useful later... Now back to linux. - I downloaded latest alsa-driver package (version 1.0.14). - I edited alsa-kernel/pci/hda/patch_sigmatel.c, and replaced the contents of the ref922x_pin_configs array with the default pin configurations found in the INI file above WITH ONE IMPORTANT TWEAK. Specifically, this is my new array: static unsigned int ref922x_pin_configs[10] = { 0x0221121E, 0x408103FF, 0x02A1123E, 0x90100310, 0x408003F1, 0x0221122F, 0x03451340, 0x40C003F2, 0x50A003F3, 0x405003F4 }; - If you compare these numbers to the CfgDflt values from the Windows INI file above, you'll see that they match except that for the third value, instead of 0x281123E I use 0x2A1123E Here's why: I figured out that pin complex 0x0c was my microphone jack (the right-most of the three jacks in the front of my laptop). The 0x281123E configuration indicates a "Line In at Ext Front" jack... Indeed, this jack can serve as a line in or mic in or even I believe as output to rear speakers. But designating it as Line In by default seems to confuse the patch_sigmatel.c routines that set up the codec... They assume it's just a line in jack and do not mark it as potential microphone input. Changing the 8 to an A in the value designates it as "Mic at Ext Front", so the codec recognizes it as the microphone input for the card. What ends up happening is that as a microphone input, the Pin-ctls for the pin gets assigned to 0x24, whereas when it just thinks it's a Line In the Pin-ctls gets set to 0x20. Both designate an input, but the 4 in 0x24 seems to also give it a Voltage Reference Signal which I guess is important for microphone inputs. (I don't fully understand most of this stuff... I've just been reading up about the HD Audio specs over the last few days). - Build and install drivers using "./configure --with-cards=hda-intel --with-sequencer=yes; make; make install" - Make sure the snd_hda_intel gets loaded with model=ref option. You need to specify the ref model so that the pin configurations set up above are used. Of course, the right way to do this would be to make a whole new model, but I didn't bother doing that in my quick hack. Anyway, in SuSE 10.2 you can set the model through YaST, or you can manually edit /etc/modprobe.d/sound and make sure model=ref is listed as one of the options in the "options snd-hda-intel . . ." line - Reset sound drivers. (Best to just reset computer, but on SuSE 10.2 it usually works if I do "/etc/init.d/alsasound reset") - And that's basically it... In the mixer (e.g. alsamixer) it's important to make sure the Capture channel has "CAPTUR" on and that it's at a high value, and that the Mux is at its max too. It does not allow me to change "Input Source", but that's fine because the above essentially hard codes it to be Front Mic which is what I wanted. I've attached the contents of /proc/asound/card0/codec#0 in its good (mic working) state. The final path is 0x0c (mic) -> 0x12 (audio selector, stereo amp-out; Mux in the mixer) -> 0x17 (audio selector, stereo amp-in; Capture in the mixer) -> 0x06 (audio input) After searching on the internet, reading about HD Audio, and poking around the driver code, I do have a number of lingering questions I would be happy to have answered: - Why does patch_sigmatel.c "fix" mux capture levels to 2? The Mux (0x12) ends up in the path from the mic in jack to the audio in, so is it supposed to act as a Mic Boost or something? - I noticed my headphone jacks (0x0a, 0x0f) have Pin-ctls set at 0xc0 which is OUT HP (i.e. uses a low impedance amplifier for headphone output). Would it damage my sound card or speakers if I connected these jacks to powered speakers or to an audio receiver (i.e. more traditionally connected to a Line Out)? - Does the snd_hda_intel driver currently (or will it eventually) support dynamic switching so I can switch from the Mic In to Line In or to Rear Speakers Out, etc, just like in the Windows driver? - Some solutions on the internet talked about creating an .asoundrc file to get the codec working... I don't understand what that file is for and whether I need to tweak it at all... But since I have a working solution without the file I guess I'm fine? Thanks, Eran. P.S. Not subscribed, so please respond by email... Codec: SigmaTel STAC9221 A1 Address: 0 Vendor Id: 0x83847680 Subsystem Id: 0x102801d7 Revision Id: 0x103601 Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x0e, stepsize=0x05, mute=1 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x6b 0x6b] Power: 0x0 Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Power: 0x0 Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Power: 0x0 Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x66 0x6d] Power: 0x0 Node 0x06 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x17 Node 0x07 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x18 Node 0x08 [Audio Output] wcaps 0x40211: Stereo Digital PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x09 [Audio Input] wcaps 0x140311: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 1 0x11 Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173f: IN OUT HP Detect Pin Default 0x0221121e: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 1 0x02 Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x408103ff: [N/A] Line In at Ext N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x20: IN Connection: 1 0x04 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x02a1123e: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Connection: 1 0x03 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173f: IN OUT HP Detect Pin Default 0x90100310: [Fixed] Speaker at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x0e [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0824: IN Detect Pin Default 0x408003f1: [N/A] Line In at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x20: IN Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Pin Default 0x0221122f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 1 0x05 Node 0x10 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x03451340: [Jack] SPDIF Out at Ext Left Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 3 0x08* 0x17 0x19 Node 0x11 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x0810024: IN EAPD Detect Pin Default 0x40c003f2: [N/A] SPDIF In at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x02 0x02] Connection: 7 0x0e 0x15 0x0f 0x0b 0x0c* 0x0d 0x0a Node 0x13 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 7 0x0e 0x15 0x0f 0x0b 0x0c* 0x0d 0x0a Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x15 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x50a003f3: [N/A] Mic at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Node 0x16 [Volume Knob Widget] wcaps 0x600000: Mono Node 0x17 [Audio Selector] wcaps 0x300903: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x0e 0x0e] Connection: 1 0x12 Node 0x18 [Audio Selector] wcaps 0x300903: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x80 0x80] Connection: 1 0x13 Node 0x19 [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital Node 0x1a [Audio Output] wcaps 0x30201: Stereo Digital Node 0x1b [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x405003f4: [N/A] Digital Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x1a

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[alsa-devel] Sound problem in Toshiba
by Furkan Cayci 17 Jun '07

17 Jun '07

Hi, I am using Ubuntu 7.04. After installing Ubuntu, I could't hear any sound from my Toshiba Satellite A205-S4577, so I did the things related with this article; http://pastebin.ca/522731 After restarting laptop, I didn't take any errors, but there was no sound again. Then I realized the output of the dmesg. $ dmesg|grep alsa [ 18.948000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 18.948000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 18.948000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 18.948000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 [ 19.672000] ALSA /home/leonetix/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [ 19.680000] ALSA /home/leonetix/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) [ 33.572000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:1810: hda_codec: model 'auto' is selected [ 33.620000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 33.620000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 33.620000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 33.620000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 [ 34.360000] ALSA /home/leonetix/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [ 34.368000] ALSA /home/leonetix/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) ------------------------- So, there is something wrong, but I didn't understand and I was wondering if you could solve or show a way to do it. Also, here is the output of lcpci -v "00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Subsystem: Toshiba America Info Systems Unknown device ff10 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at f0b40000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0" Thanks for your help, Furkan

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[alsa-devel] Software mixing according to priority of streams
by Nobin Mathew 15 Jun '07

15 Jun '07

Hi, Is there any sound mixer which does mixing according to priority of streams, i.e. i will have streams with different priority like audio stream , bluettoth sco stream, gps navigation sound stream etc. Since my hardware will not support hardware mixing i will have to do this mixing in software. If bluetooth sco stream is opened while audio stream is playing then the application (sound server) should stop the playback of audio stream and should start playback of bletooth sco stream. priority order is bluettoth sco > gps navigation sound > audio Can dmix and Jack server the purpose. Nobin Mathew

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[alsa-devel] [RFC][ASoC] New device/driver arch
by Liam Girdwood 15 Jun '07

15 Jun '07

I'd like to gather some feedback on some changes I've been making to ASoC recently. The changes were made to fit in better with the PPC device tree and to help with easier porting to new platforms. They were also introduced to make dealing with device probe errors (e.g. I2C probe failure) a little easier. The largest change is the introduction of an ASoC bus device. All component drivers (that is, codec, codec DAI, CPU DAI, and DMA PCM drivers) register with this bus. This should simplify device registration and hopefully better fit in with the PPC model. A component driver now typically consists of the standard driver ops and a set of ASoC/ALSA ops to configure audio. The new soc.h shows this here :- http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… Example component drivers are here :- I2S http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… DMA http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… WM8753 Codec http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… Another major change is that there is no more snd_dai_link (structure that connected codec <--> CPU). This has been replaced by an ASoC PCM device that describes each PCM device in the machine. This new ASoC PCM device has it's own driver and registers with the bus (like the others). This allows an ASoC PCM driver to be defined like :- static struct snd_soc_device_driver hifi_pcm_drv = { .type = SND_SOC_BUS_TYPE_PCM, .driver = { .name = "mainstone-hifi-pcm", .owner = THIS_MODULE, .probe = hifi_pcm_probe, .remove = hifi_pcm_remove, }, .components = { .cpu_dai = pxa2xx_i2s, .codec = wm8753_codec, .codec_dai = wm8753_hifi_dai, .platform = pxa2xx_pcm, }, }; This now shows each component within the driver structure. The snd_soc_machine driver has changed and can now contain numerous ASoC PCM devices. It is now a platform driver. Example machine driver for Mainstone with WM8753 is here :- http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… The machine driver now also does the I2C/SPI probing (instead of the codec) and also does any codec IO (to better support multiple controllers). Fwiw, I've not tested any of this code atm. I plan to start this tomorrow on pxa2xx and i.MX31 and would like to get some feedback before porting remaining platforms and codecs over. Liam

4 6

[alsa-devel] Documentation for params_xxx() macros?
by Timur Tabi 14 Jun '07

14 Jun '07

Is there documentation that describes each of the params_xxx() macros? If not, then which macro tells me how many periods there are? Is it params_periods()? -- Timur Tabi Linux Kernel Developer @ Freescale

1 0

[alsa-devel] aplay hangs with execve("/build/bin/aplay", ["aplay"], [/* 8 vars */
by Pharaoh . 14 Jun '07

14 Jun '07

Hi I am using alsa lib and alsa utils 1.0.14 on my omap based board. When I try to test my my alsa driver using aplay it just hangs at the very first step. The strace output shows nothing else except the following line, execve("/build/bin/aplay", ["aplay"], [/* 8 vars */ What I am missing?

1 0

[alsa-devel] Drivers for M-Audio Firewire 410
by Cédric 14 Jun '07

14 Jun '07

Hello, Im a french user of Linux (Ubuntu Studio 7.04) and cant install my new hardware (M-Audio Firewire 410). Ive send a mail to the M-Audios technical support, but they have no solution. They recommended me to contact yourself. Could you help me for this problem. Do you need more informations about my computer hardware ? Syncerly. Cédric

2 1

[alsa-devel] [PATCH]is_power_of_2-sound/pcm_params.h
by vignesh babu 14 Jun '07

14 Jun '07

Replacing (n & (n-1)) in the context of power of 2 checks with is_power_of_2 Signed-off-by: vignesh babu <vignesh.babu(a)wipro.com> --- diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 85cf1cf..dee298e 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -21,6 +21,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ +#include <linux/log2.h> int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, @@ -170,7 +171,7 @@ static inline int snd_mask_single(const struct snd_mask *mask) for (i = 0; i < SNDRV_MASK_SIZE; i++) { if (! mask->bits[i]) continue; - if (mask->bits[i] & (mask->bits[i] - 1)) + if (!is_power_of_2(mask->bits[i])) return 0; if (c) return 0; -- Vignesh Babu BM _____________________________________________________________ "Why is it that every time I'm with you, makes me believe in magic?" The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[alsa-devel] [PATCH] Fix build error with strict mm checking
by Martin Habets 13 Jun '07

13 Jun '07

Turning on STRICT_MM_TYPECHECKS in include/asm/page.h causes a build error for sparc32. This patch against 2.6.21 fixes this. Martin Signed-off-by: Martin Habets <errandir_news(a)mph.eclipse.co.uk> --- --- linux/sound/core/sgbuf.c.orig 2007-06-06 23:53:34.000000000 +0100 +++ linux/sound/core/sgbuf.c 2007-06-06 23:54:45.000000000 +0100 @@ -96,7 +96,7 @@ void *snd_malloc_sgbuf_pages(struct devi } sgbuf->size = size; - dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL); + dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, __pgprot(PAGE_KERNEL)); if (! dmab->area) goto _failed; return dmab->area;

2 1

[alsa-devel] Trouble understanding ALSA's DMA buffers
by Timur Tabi 12 Jun '07

12 Jun '07

I'm writing an ALSA SOC driver for an I2S-based device, and I'm having a really hard time understanding how ALSA uses the DMA buffers. And yes, I've read the documentation and studied some sample source code. I used to write audio drivers for a living, but that was many years ago, and it wasn't for Linux. Perhaps the concepts in my head are outdated, but I just don't see enough explanation as to how DMA buffers are supposed to work. Back then, audio drivers used "ping pong" DMA buffers. A single DMA buffer is allocated, and the audio hardware is programmed to read from that buffer in a loop. That is, it would automatically restart reading from the beginning of the buffer without any reprogramming. The hardware would also be programmed to issue an interrupt when it got to the end of the buffer, and when it got to the half-way point. To start playback, you first filled the whole buffer with audio data, and then told the hardware to start playing. After the hardware got to the half-way point, it would issue an interrupt. You would then tell the OS you need more data, and you'd get it. You then copy that data into the first half of the buffer *while* the hardware was playing the second half. Later, the hardware would interrupt you when it got to the end of the buffer. You'd then copy more data to the 2nd half while the hardware is playing the first half. And so - hardware plays one half while you copy data to the other half. Hence, "ping pong". So how do I implement this in ALSA? The "Writing an ALSA Driver" document doesn't even contain the words "ping" or "pong". -- Timur Tabi Linux Kernel Developer @ Freescale

4 6

[alsa-devel] [ASoC] bug in soc_pcm_open
by Matthieu CASTET 12 Jun '07

12 Jun '07

Hi, it seems there is a bug in soc_pcm_open. In some case [1] we are in a error path, but we don't set the ret return value. This cause bug/crash from alsa core. Thanks Matthieu [1]http://hg-mirror.alsa-project.org/alsa-kernel/file/38926de0e79c/soc/soc-c… if (!runtime->hw.rates) { 218 printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", 219 codec_dai->name, cpu_dai->name); 220 goto machine_err; 221 } 222 if (!runtime->hw.formats) { 223 printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", 224 codec_dai->name, cpu_dai->name); 225 goto machine_err; 226 } 227 if (!runtime->hw.channels_min || !runtime->hw.channels_max) { 228 printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", 229 codec_dai->name, cpu_dai->name); 230 goto machine_err; 231 }

4 4

[alsa-devel] Realtek 268
by Steven Zakulec 12 Jun '07

12 Jun '07

I have tried the current (from about 6 hours ago) drivers+lib from the HG repository. My card is now labelled as an ALC 268, and shows me what appears to be a correct number of options in the mixer. Unfortunately, I still get no sound. I've compiled the driver with --debug=full, and the lib with the debug. The following is from dmesg when I removed the old snd-hda-intel module, did a make install, then did modprobe snd-hda-intel: (I cleared out dmesg before I did the modprobe) [ 1473.892000] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 21 [ 1473.892000] PCI: Setting latency timer of device 0000:00:1b.0 to 64 [ 1474.092000] hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... [ 1474.140000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 1474.140000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 1474.140000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 1474.140000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 [ 1474.864000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [ 1474.872000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) My /proc/asound/card0/codeco#0 is as follows: cat /proc/asound/card0/codec#0 Codec: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1179ff10 Revision Id: 0x100003 Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x02211020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x02a11830: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: Pincap 0x083724: IN Detect Pin Default 0x99a30940: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x24: IN Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4015812d: [N/A] Speaker at Ext N/A Conn = Optical, Color = Purple Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x81 0x81] Connection: 7 0x18 0x19 0x1a 0x1c 0x14 0x15* 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x81 0x81] Connection: 7 0x18 0x19 0x1a 0x1c 0x14* 0x15 0x13 Output for the audio device (lspci -vvv) below: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Subsystem: Toshiba America Info Systems Unknown device ff10 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 21 Region 0: Memory at f0b40000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0

2 1

[alsa-devel] alsa on Atmel at91
by Michel Benoit 12 Jun '07

12 Jun '07

Hello I am trying to run alsa on an ARM based target AT91SAM9260 based board. The cpu is connected over i2s (atmel's ssc) to a PCM1725 chip. I modifed the AT73C213 driver available in the latest experimental patches from www.at91.com to behave correctly for the PCM1725 chip. Basically all I have done is remove the spi communication with the AT73C213 that regulates the sound output volume. The PCM chip has no such volume control and I use the i2s interface setup as is to send sample data. I cross compile alsa lib and run the program test/pcm_min which generates as expected random noise on the speaker output. However when I try to run test/pcm I get a divide by zero floating point exception because the library returns a period_size of 0. This despite the fact that the period time is set to 100000 and buffer time is set to 500000. I enabled some debug info in the library and I see the following data dumped to the console. ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: (20354 20355) PERIOD_SIZE: 977 PERIOD_BYTES: 3908 PERIODS: (4 5) BUFFER_TIME: (81479 81480) BUFFER_SIZE: 3911 BUFFER_BYTES: 15644 TICK_TIME: 10000 How is it possible that the following code from test/pcm.c err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, &dir); if (err < 0) { printf("Unable to set period time %i for playback: %s\n", period_time, snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_period_size(params, &size, &dir); if (err < 0) { printf("Unable to get period size for playback: %s\n", snd_strerror(err)); return err; } returns a period_size of 0 when the debug output shows that the period_size has been calculated to be 977? Can it be that the driver has not initialised the alsa structures properly (the driver file is attached to this email)? Have I mucked something up by removing the spi comm. with the at73c213? Could this be due to a problem in the cross compilation of alsa-lib? I use the following configure command line: CC=arm-linux-gcc ./configure \ --host=arm-linux\ --prefix=$HOME/bsp/rootfs/buildroot/build_arm/root/usr\ --with-softfloat (note: I have also tried it without --with-softfloat) I have tried digging through the alsa lib code but it is not very clear to me where the returned value for period_size comes from. I appreciate any pointers, hints or tips that can help me to solve and/or understand this problem. Thanks in advance! Michel

5 20

[alsa-devel] Source of timing for MIDI and/or audio synchronization
by Mark Rivera 12 Jun '07

12 Jun '07

Hello (new to list), and thank you for the work you have been doing. As an end-user of the ALSA project's development, I am very grateful for being able to use softwares which provide audio functionality to Linux. I have been reading up on the docs and list archives, but I still have an open question for which I hope you can lead me to an answer. I designed and built a MIDI sequencer based around a Microchip PIC18 MCU, and used the built-in hardware timers and interrupt levels to drive the 'scheduler' for initiating MIDI output (EUSART) routines. In this same vein, I would like to write a software sequencer with as 'tight' timing as possible. So, in a similar concept for Linux - from where does the timing of audio and MIDI events come from? It occurred to me that it would be nice to have a loadable kernel module (LKM) to which userland programs could subscribe for receiving 'soft interrupts' based on some timer (cpu ticks?) so that various software could all reference the same notion of the "current time" - a sort of "master clock" with start/stop/continue as well. Does ALSA provide this functionality? Can someone point me in the right direction for learning more about this? Many thanks. Mark

3 2

[alsa-devel] snd_pcm_lib_malloc_pages()
by Timur Tabi 12 Jun '07

12 Jun '07

(resending since I haven't gotten a reply yet) Liam Why do none of the ASoC drivers use snd_pcm_lib_malloc_pages()? They all allocate their DMA buffers by hand. -- Timur Tabi Linux Kernel Developer @ Freescale

2 1

[alsa-devel] [PATCH] snd-usb-caiaq: Fix compatibility with kernels < 2.6.19
by Sergey Vlasov 12 Jun '07

12 Jun '07

Kernels before 2.6.19 used a different signature for the urb->complete callback (an additional "struct pt_regs *regs" parameter was present), therefore snd-usb-caiaq needs some compatibility patches in alsa-driver, like snd-usb-audio. Signed-off-by: Sergey Vlasov <vsu(a)altlinux.ru> --- The snd_usb_caiaq_midi_output_done() part is somewhat hackish - the problem is that snd_usb_caiaq_midi_output_done() is declared in caiaq-midi.h, and the alsa-driver build system does not support patching of *.h files. The rest is basically copied from usbaudio.patch. Without the patch compiling for 2.6.18 produces warnings: .../usb/caiaq/caiaq-device.c: In function 'init_card': .../usb/caiaq/caiaq-device.c:316: warning: passing argument 6 of 'usb_fill_bulk_urb' from incompatible pointer type .../usb/caiaq/caiaq-device.c:321: warning: passing argument 6 of 'usb_fill_bulk_urb' from incompatible pointer type .../usb/caiaq/caiaq-audio.c: In function 'alloc_urbs': .../usb/caiaq/caiaq-audio.c:583: warning: assignment from incompatible pointer type --- alsa-driver-1.0.14/usb/caiaq/Makefile.alt-caiaq-api 2007-06-11 16:29:28 +0400 +++ alsa-driver-1.0.14/usb/caiaq/Makefile 2007-06-11 19:50:39 +0400 @@ -5,6 +5,13 @@ endif include $(SND_TOPDIR)/toplevel.config include $(SND_TOPDIR)/Makefile.conf +clean-files := caiaq-audio.c caiaq-device.c + include $(SND_TOPDIR)/alsa-kernel/usb/caiaq/Makefile +EXTRA_CFLAGS += -I$(SND_TOPDIR)/alsa-kernel/usb/caiaq + include $(SND_TOPDIR)/Rules.make + +caiaq-audio.c: caiaq-audio.patch $(SND_TOPDIR)/alsa-kernel/usb/caiaq/caiaq-audio.c +caiaq-device.c: caiaq-device.patch $(SND_TOPDIR)/alsa-kernel/usb/caiaq/caiaq-device.c --- alsa-driver-1.0.14/usb/caiaq/caiaq-audio.patch.alt-caiaq-api 2007-06-11 19:18:40 +0400 +++ alsa-driver-1.0.14/usb/caiaq/caiaq-audio.patch 2007-06-11 19:18:17 +0400 @@ -0,0 +1,26 @@ +--- ../../alsa-kernel/usb/caiaq/caiaq-audio.c 2007-06-11 16:29:28 +0400 ++++ caiaq-audio.c 2007-06-11 19:17:38 +0400 +@@ -464,7 +464,11 @@ static void fill_out_urb(struct snd_usb_ + check_for_elapsed_periods(dev, dev->sub_playback); + } + ++#if !defined(OLD_USB) && !defined(CONFIG_SND_NEW_IRQ_HANDLER) ++static void read_completed(struct urb *urb, struct pt_regs *regs) ++#else + static void read_completed(struct urb *urb) ++#endif + { + struct snd_usb_caiaq_cb_info *info = urb->context; + struct snd_usb_caiaqdev *dev; +@@ -518,7 +522,11 @@ static void read_completed(struct urb *u + usb_submit_urb(urb, GFP_ATOMIC); + } + ++#if !defined(OLD_USB) && !defined(CONFIG_SND_NEW_IRQ_HANDLER) ++static void write_completed(struct urb *urb, struct pt_regs *regs) ++#else + static void write_completed(struct urb *urb) ++#endif + { + struct snd_usb_caiaq_cb_info *info = urb->context; + struct snd_usb_caiaqdev *dev = info->dev; --- alsa-driver-1.0.14/usb/caiaq/caiaq-device.patch.alt-caiaq-api 2007-06-11 19:55:17 +0400 +++ alsa-driver-1.0.14/usb/caiaq/caiaq-device.patch 2007-06-11 19:54:49 +0400 @@ -0,0 +1,31 @@ +--- ../../alsa-kernel/usb/caiaq/caiaq-device.c 2007-06-11 16:29:28 +0400 ++++ caiaq-device.c 2007-06-11 19:54:46 +0400 +@@ -100,7 +100,11 @@ static struct usb_device_id snd_usb_id_t + { /* terminator */ } + }; + ++#if !defined(OLD_USB) && !defined(CONFIG_SND_NEW_IRQ_HANDLER) ++static void usb_ep1_command_reply_dispatch (struct urb* urb, struct pt_regs *regs) ++#else + static void usb_ep1_command_reply_dispatch (struct urb* urb) ++#endif + { + int ret; + struct snd_usb_caiaqdev *dev = urb->context; +@@ -295,6 +299,16 @@ static struct snd_card* create_card(stru + return card; + } + ++#if !defined(OLD_USB) && !defined(CONFIG_SND_NEW_IRQ_HANDLER) ++static void snd_usb_caiaq_midi_output_done_wrapper(struct urb* urb, ++ struct pt_regs *regs) ++{ ++ snd_usb_caiaq_midi_output_done(urb); ++} ++#undef snd_usb_caiaq_midi_output_done ++#define snd_usb_caiaq_midi_output_done snd_usb_caiaq_midi_output_done_wrapper ++#endif ++ + static int init_card(struct snd_usb_caiaqdev *dev) + { + char *c;

2 1

[alsa-devel] [PATCH] asihpi: Fix compatibility with 64-bit resource_size_t
by Sergey Vlasov 12 Jun '07

12 Jun '07

In recent kernels the resource_size_t type can be either 32-bit or 64-bit depending on config options. The proper way to printk such values is to cast them to unsigned long long. Signed-off-by: Sergey Vlasov <vsu(a)altlinux.ru> --- Without the patch compilation on 2.6.18 produces warnings: .../pci/asihpi/hpimod.c: In function 'adapter_probe': .../pci/asihpi/hpimod.c:460: warning: format '%x' expects type 'unsigned int', but argument 4 has type 'resource_size_t' .../pci/asihpi/hpimod.c:460: warning: format '%x' expects type 'unsigned int', but argument 5 has type 'resource_size_t' --- alsa-driver-1.0.14/pci/asihpi/hpimod.c.alt-asihpi-warnings 2007-06-11 16:29:28 +0400 +++ alsa-driver-1.0.14/pci/asihpi/hpimod.c 2007-06-11 19:58:30 +0400 @@ -457,10 +457,10 @@ static int __devinit adapter_probe(struc // fill in HPI_PCI information from kernel provided information for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) { - HPI_DEBUG_LOG4(DEBUG, "Resource %d %s %x-%x\n", idx, + HPI_DEBUG_LOG4(DEBUG, "Resource %d %s %llx-%llx\n", idx, pci_dev->resource[idx].name, - pci_resource_start(pci_dev, idx), - pci_resource_end(pci_dev, idx)); + (unsigned long long)pci_resource_start(pci_dev, idx), + (unsigned long long)pci_resource_end(pci_dev, idx)); memlen = pci_resource_len(pci_dev, idx); if (memlen) {

2 1

[alsa-devel] Shutdown() vector in ALSA
by Nobin Mathew 12 Jun '07

12 Jun '07

why no ALSA driver has shutdown() vector implemented? Is this vector needed to ensure the proper shutdown of the system? What all needs to be taken care if i want to implement shutdown(), except hardware shutdown. struct snd_shutdown_f_ops { struct file_operations f_ops; struct snd_shutdown_f_ops *next; }; what this operations means? defined in sound/core/init.c Is there any need to call these functions? snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); for (i = 0; i < chip->pcm_devs; i++) snd_pcm_suspend_all(chip->pcm[i]);

3 8

Re: [alsa-devel] Alsa 1.0.13 (and others) does not work on ALC880
by Andrew Callan 12 Jun '07

12 Jun '07

I have been without sound using Fedora Core ever since the 2.6.16 or 2.6.17 kernel update (2.6.16 crashed during initialization, so I could not tell whether sound worked or not). It had worked well through the 2.6.15 Fedora kernels. I finally sat down tonight to work the problem as hard as I could and happened upon Eric and Tobin's timely exchange. First, simply adding probe_mask=1 to the snd-hda-intel options worked for me as well; I did not end up having to specify a specific model. Second, I too was seeing a LONG hang when trying to do: cat /proc/asound/card0/codec#0 but now with the probe_mask=1, I do not see this. If it helps, the alsa-info.sh output for my system is at http://pastebin.ca/496293 If it makes any difference, I'm not sure which model I had specified on the modprobe line at the time I captured this output. One thing I do notice is that after adding probe_mask=1, aplay -l does not show the Si3054 Modem which shows up in the alsa-info output. Thanks for the help you gave in the thread with Eric. Andrew

5 16

[alsa-devel] hda-codec - Add HP Pavillion quirk to Realtek code.
by Tobin Davis 12 Jun '07

12 Jun '07

This patch adds the HP Pavillion a1589L desktop system to the ALC883 configuration table. Signed off by Tobin Davis <tdavis(a)dsl-only.net> Resent from the correct email address. Ignore previous email.

2 1

[alsa-devel] snd_pcm_lib_malloc_pages()
by Timur Tabi 12 Jun '07

12 Jun '07

Liam Why do none of the ASoC drivers use snd_pcm_lib_malloc_pages()? They all allocate their DMA buffers by hand. -- Timur Tabi Linux Kernel Developer @ Freescale

2 1

[alsa-devel] [ASoC] machine driver and channel configuration
by Matthieu CASTET 12 Jun '07

12 Jun '07

Hi, If the platform/cpu_dai driver and codec support capture and playback channels. But on the board only the playback link is connected. In this configuration, the codec and platform/cpu_dai driver shouldn't try to use the capture path (and the resource associated to this). Is there a way to do something like that ? Thanks Matthieu

3 3

[alsa-devel] Questions on usb audio development
by LCID Fire 12 Jun '07

12 Jun '07

Hi. I know there is quite some documentation how to develop a pci driver for alsa - but since I'll try to get a usb device working I was wondering how to write a usb driver? Is it just the same? How to interface the various aspects of the device with regards to descriptor reading, ...

4 11

[alsa-devel] hda-codec - Add HP Pavillion quirk to Realtek code.
by Tobin Davis 11 Jun '07

11 Jun '07

This patch adds the HP Pavillion a1589L desktop system to the ALC883 configuration table. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

1 0

[alsa-devel] Patch - Revo7.1 PCM Playback Switch
by dustin＠seznam.cz 11 Jun '07

11 Jun '07

Hello Takashi, I have attached a patch to support mute for front channels of M-Audio Revolution 7.1 (the DAC AK4381 features a mute bit). Signed-off-by: Pavel Hofman <dustin at seznam.cz> Please could I ask you for your opinion on the following issue (so far no response): I am working together with Ctirad on supporting more capture channels in E-MU1212. So far we have managed to increase the number of channels from the default of 8 to 16 (A_FXWC2 = 0xffffffff, added new snd_emu10k1_audigy_dsp_convert_32_to_2x16 calls in emufx.c). There is still a number of issues to solve but the 16 channel capture already works fine. I would like to get your opinion on the form of adding this support to the module. The options are either hard-coding the 16 capture channels into the code, assuming that users would prefer as many channels as possible and do the routing in jack, or introducing a module parameter emu1010-capture-channels (range e.g. 1 to 16 - untested yet) and leaving the default number at the existing 8. Furthermore, the PCMs other than efx seem to be of no use for the EMU-1x1x models, probably fairly confusing for new users. Would you find it OK, if they were removed for this specific card (if (emu->card_capabilities->emu1010)), keeping only the playback and capture PCM id 2? Thanks a lot for your suggestions. Based on them, I will prepare the patch (with comments on what we found out about the card - it is a pitty that the driver has almost no big-picture comments on the supported cards' specifics, making new developers spend a number of long nights going through the nontrivial code and guessing). Best regards, Pavel.

4 7

Re: [alsa-devel] [2/3] 2.6.22-rc4: known regressions v2
by Michal Piotrowski 09 Jun '07

09 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc4. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Status : Unknown IDE Subject : libata and legacy ide pcmcia failure References : http://lkml.org/lkml/2007/5/17/305 Submitter : Robert de Rooy <robert.de.rooy(a)gmail.com> Handled-By : Mark Lord <mlord(a)pobox.com> Status : problem is being debugged SATA/PATA Subject : 22-rc3 broke the CDROM in Dell notebook References : http://lkml.org/lkml/2007/5/27/63 Submitter : Gregor Jasny <gjasny(a)googlemail.com> Handled-By : Tejun Heo <htejun(a)gmail.com> Jeff Garzik <jeff(a)garzik.org> Caused-By : Tejun Heo <htejun(a)gmail.com> commit d4b2bab4f26345ea1803feb23ea92fbe3f6b77bc Status : problem is being debugged Sparc64 Subject : 2.6.22-rc broke X on Ultra5 References : http://lkml.org/lkml/2007/5/22/78 Submitter : Mikael Pettersson <mikpe(a)it.uu.se> Handled-By : David Miller <davem(a)davemloft.net> Status : problem is being debugged Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

2 1

[alsa-devel] probing state of high definition audio card in windows / SigmaTel STAC9221 microphone problems
by Eran Guendelman 09 Jun '07

09 Jun '07

Hi, I am struggling to get the (external) microphone working on a SigmaTel STAC9221 A1 device on my Dell XPS M1210 laptop. I saw other emails indicating that this was still problematic, and the latest drivers from alsa 1.0.14 did not seem to fix the problem. So now that I've started reading and trying to understand the basics behind the hda codec in an effort to fix this problem, I was wondering if it was possible for me to boot into windows (I've got a dual-boot setup), get into a state where the microphone is working, and somehow probe the codec's state at that time (all of the pin configurations, etc.)... then theoretically, if I reproduce that configuration on the linux side, it should work (right?). So the question is, does anyone have a ready-made utility for Windows to do this probe of the high definition audio codec? I could probably try writing one myself, but thought I'd check if it already exists. Thanks. Please send responses by email too (not subscribed).

3 3

[alsa-devel] Sound Devices USBPre
by Brian Phelps 08 Jun '07

08 Jun '07

Hey guys, I contacted Sound Devices at sounddevices.com about writing a Linux driver (I was going to head the project but would take any help I can get) for the USBPre, a 2ch-in 2ch-out USB preamp with highquality balanced XLR Mic inputs. They said that they would love to have a Linux Driver and will provide specs for me, I am still waiting on the info though. I have not asked if I can view their firmware sourcecode yet, and if it requires an NDA, I will have to work around that. What advice do you have for writing a driver for alsa? I have written some small kernel drivers for embedded Linux devices, a few audio programs in C, and I come from a DSP background, but I am not sure exactly where to start, besides the specs and firmware. Thanks for your help -Brian Phelps

1 0

[alsa-devel] Callback trouble
by Juan Manuel Lopez Baio 08 Jun '07

08 Jun '07

Hello. I'm having some problems with the asynchronous usage of ALSA API, and I'm not sure if this usage is correct. What I have done so far is the following: I have an class that encapsulates the ALSA API, working in a multithreaded environment. I use alsa to capture audio and want to do so in an asynchronous and non-blocing way, so I defined a static method in my class that functions as callback, which receives a pointer to the instance of my class and calls on a member function, like this: void alsaDevice::myCallback(snd_async_handler_t *handler) //STATIC { alsaDevice *dev = (alsaDevice*)snd_async_handler_get_callback_private(handler); dev->triggerCondition(); } void alsaDevice::triggerCondition() { LogError(0,"Trigger en placa %d device %d",cn,dn); m_condMutex.lock(); m_noDataCondition.trigger(); m_condMutex.unlock(); } To init ALSA, I do this: int mode = SND_PCM_ASYNC | SND_PCM_NONBLOCK; if ((err=snd_pcm_open(&pcm_cap_handle, pcm_name.str().c_str(), SND_PCM_STREAM_CAPTURE, mode) < 0)) { LogError(0,"Opening ALSA PCM device %s: %s",pcm_name.str().c_str(),snd_strerror(err)); pcm_cap_handle=NULL; return false; } then set all the pertinent parameters, and finally if((asyncaddresult=snd_async_add_pcm_handler(&ahandler,pcm_cap_handle,alsaDevice::myCallback,this))<0) { LogError(0,"No se pudo registrar la callback; PCM Error: %i",asyncaddresult); } if ((asyncaddresult=snd_pcm_start(pcm_cap_handle))<0) LogError(0,"No se pudo start pcm; PCM Error: %i",asyncaddresult); Afterwards, the first time that a call to snd_pcm_readn returns EAGAIN I call wait() on a condition object with a 100 ms timeout, which is triggered by the callback, as seen before. The problem is that the callback never seems to get called (so shows the log), so I'm not really sure if the previous steps were right. Any light you can shed on this would be much appreciated. Thanks! juan

1 0

[alsa-devel] snd-usb-audio: Add another QuickCam ID
by Ritesh Raj Sarraf 08 Jun '07

08 Jun '07

Hi, This patch adds another Logitech QuickCam USB ID for the device which is shipped with Dell's XPS M1210 Notebooks. Signed-off-by: Ritesh Raj Sarraf <rrs(a)researchut.com> diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 374fbf6..c69a56f 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -70,6 +70,15 @@ .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, + .idProduct = 0x08c6, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, PS: Request you to CC me if there are any comments. I'm not subscribed to the list. Thanks, Ritesh -- Ritesh Raj Sarraf RESEARCHUT - http://www.researchut.com "Necessity is the mother of invention."

2 2

[alsa-devel] [PATCH] soc/sh: let SND_SOC_PCM_SH7760 depend on SH_DMABRG
by Manuel Lauss 08 Jun '07

08 Jun '07

Let SND_SOC_PCM_SH7760 depend on SH_DMABRG rather than select'ing it. Gets rid of the Kbuild warning about SH_DMABRG being undefined on non-sh builds. Signed-off-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> --- linux-2.6.22-rc4-mm2/sound/soc/sh/Kconfig~ 2007-06-07 11:09:30.000000000 +0200 +++ linux-2.6.22-rc4-mm2/sound/soc/sh/Kconfig 2007-06-07 11:09:30.000000000 +0200 @@ -2,8 +2,7 @@ menu "SoC Audio support for SuperH" config SND_SOC_PCM_SH7760 tristate "SoC Audio support for Renesas SH7760" - depends on CPU_SUBTYPE_SH7760 && SND_SOC - select SH_DMABRG + depends on CPU_SUBTYPE_SH7760 && SND_SOC && SH_DMABRG help Enable this option for SH7760 AC97/I2S audio support.

2 1

[alsa-devel] [PATCH] Add quirk for another MSI laptop to alc883_cfg_tbl[]
by Daniel T. Chen 08 Jun '07

08 Jun '07

Hi, Attached please find an hg changeset against current alsa-kernel tip adding the SSID of another MSI laptop to the quirk list. Thanks, Daniel Chen

2 1

[alsa-devel] [ASoC]one Small Suggestion
by Nobin Mathew 08 Jun '07

08 Jun '07

Liam, Thank you for your support, my work is almost complete now, driver is working perfectly now. I have one suggestion, I modified the wm9713.c codec driver in probe function , so that if(socdev->machine->dai_link->cpu_dai->private_data) codec->ac97->bus->private_data = socdev->machine->dai_link->cpu_dai->private_data; Here i am assigning the controller private data. so that my following ac97 read and write to access codec registers become good. static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct platform_controller *controller = ac97->bus->private_data; struct aaci *aaci = (struct aaci *)controller->bus_controller; static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct platform_controller *controller = ac97->bus->private_data; struct aaci *aaci = (struct aaci *)controller->bus_controller; u32 v; Is there a way to accomodate this in existing ASoC design? Or a better way to do this?

3 5

[alsa-devel] [RFC] [PATCH 1/2] ASoC sound support for SMDK2440 boards: UDA1341 driver
by Zoltan Devai 08 Jun '07

08 Jun '07

ASoC Support for the UDA1341 codec. Please note that until there is a final L3-bus driver, this patch won't work. --- --- /dev/null +++ /sound/soc/codecs/uda1341.c @@ -0,0 +1,519 @@ +/* + * uda1341.c -- UDA1341 ALSA SoC Codec driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie + * + * 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/l3.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "../s3c24xx/s3c24xx-i2s.h" +#include "uda1341.h" + +#define UDA1341_DEBUG 0 + +#ifdef UDA1341_DEBUG +#define DBG(format, arg...) \ + printk(KERN_DEBUG "UDA1341: %s:" format "\n" ,__FUNCTION__, ## arg) +#else +#define DBG(format, arg...) do {} while (0) +#endif + +#define UDA1341_RATES SNDRV_PCM_RATE_8000_48000 +#define UDA1341_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE ) + +struct uda1341_priv { + int sysclk; + int dai_fmt; +}; + +/* In-data addresses are hard-coded into the reg-cache values */ +static const char uda1341_reg[UDA1341_REGS_NUM] = { + /* Extended address registers */ + 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, + /* Status, data regs */ + 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, +}; + +/* + * The codec has no support for reading its registers except for peak level... + */ +static inline unsigned int uda1341_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + DBG("reg: %d", reg); + + if (reg >= UDA1341_REGS_NUM) + return -1; + return cache[reg]; +} + +/* + * Write the register cache + */ +static inline void uda1341_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, unsigned int value) +{ + u8 *cache = codec->reg_cache; + + DBG("reg: %02X", reg); + + if (reg >= UDA1341_REGS_NUM) + return; + cache[reg] = value; +} + +/* + * Write to the uda1341 registers + * + */ +static int uda1341_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + int ret; + int addr; + u8 data = value; + + DBG("reg: %02X, value:%02X", reg, value); + + if (reg >= UDA1341_REGS_NUM) { + DBG("Unkown register: reg: %d", reg); + return -EINVAL; + } + + uda1341_write_reg_cache(codec, reg, value); + + switch (reg) { + case UDA1341_STATUS0: + case UDA1341_STATUS1: + addr = UDA1341_STATUS_ADDR; + break; + case UDA1341_DATA000: + case UDA1341_DATA001: + case UDA1341_DATA010: + addr = UDA1341_DATA0_ADDR; + break; + case UDA1341_DATA1: + addr = UDA1341_DATA1_ADDR; + break; + default: + /* It's an extended address register */ + addr = (reg | UDA1341_EXTADDR_PREFIX); + + ret = codec->hw_write_l3(codec->control_data, + UDA1341_DATA0_ADDR, (char*)&addr, 1); + if (ret != 1) + return -EIO; + + addr = UDA1341_DATA0_ADDR; + data = (value | UDA1341_EXTDATA_PREFIX); + break; + } + + ret = codec->hw_write_l3(codec->control_data, addr, (char*)&data, 1); + if (ret != 1) + return -EIO; + + return 0; +} + +static inline void uda1341_reset(struct snd_soc_codec *codec) +{ + u8 reset_reg = uda1341_read_reg_cache(codec, UDA1341_STATUS0); + uda1341_write(codec, UDA1341_STATUS0, reset_reg | (1<<6)); +} + +static int uda1341_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = uda1341_read_reg_cache(codec, UDA1341_DATA010); + + DBG("mute: %d", mute); + + if (mute) + mute_reg |= (1<<2); + else + mute_reg &= ~(1<<2); + + uda1341_write(codec, UDA1341_DATA010, mute_reg & ~(1<<2)); + + return 0; +} + +static int uda1341_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + u8 hw_params = uda1341_read_reg_cache(codec, UDA1341_STATUS0); + hw_params &= STATUS0_SYSCLK_MASK; + hw_params &= STATUS0_DAIFMT_MASK; + + DBG("sysclk: %d, rate:%d", uda1341->sysclk, params_rate(params)); + + /* set SYSCLK / fs ratio */ + switch (uda1341->sysclk / params_rate(params)) { + case 512: + break; + case 384: + hw_params |= (1<<4); + break; + case 256: + hw_params |= (1<<5); + break; + default: + return -EINVAL; + break; + } + + DBG("dai_fmt: %d, params_format:%d", uda1341->dai_fmt, + params_format(params)); + + + /* set DAI format and word length */ + switch (uda1341->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch(params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + hw_params |= (1<<1); + break; + case SNDRV_PCM_FORMAT_S18_3LE: + hw_params |= (1<<2); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + hw_params |= ((1<<2) | (1<<1)); + break; + default: + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_LEFT_J: + hw_params |= (1<<3); + break; + default: + return -EINVAL; + } + + uda1341_write(codec, UDA1341_STATUS0, hw_params); + + return 0; +} + +static int uda1341_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + DBG("clk_id: %d, freq: %d, dir: %d", clk_id, freq, dir); + + /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable + we'll error out on set_hw_params if it's not OK */ + if ( (freq >= (256 * 8000)) && (freq <= (512 * 48000)) ) { + uda1341->sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int uda1341_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + DBG("fmt: %08X", fmt); + + /* codec supports only full slave mode */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + + /* no support for clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + + /* We can't setup DAI format here as it depends on the word bit num */ + /* so let's just store the value for later */ + uda1341->dai_fmt = fmt; + + return 0; +} + +static int uda1341_dapm_event(struct snd_soc_codec *codec, int event) +{ + u8 reg; + + DBG("event: %08X", event); + + switch (event) { + case SNDRV_CTL_POWER_D0: /* full On */ + /* ADC, DAC on */ + reg = uda1341_read_reg_cache(codec, UDA1341_STATUS1); + uda1341_write(codec, UDA1341_STATUS1, reg | 0x03); + break; + case SNDRV_CTL_POWER_D1: /* partial On */ + case SNDRV_CTL_POWER_D2: /* partial On */ + case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + break; + case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + /* mute, ADC, DAC power off */ + reg = codec->read(codec, UDA1341_STATUS1); + uda1341_write(codec, UDA1341_STATUS1, reg & ~(0x03)); + break; + } + codec->dapm_state = event; + return 0; +} + +static const char *uda1341_dsp_setting[] = {"Flat", "Minimum1", "Minimum2", "Maximum"}; +static const char *uda1341_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *uda1341_mixmode[] = {"DD", "Input 2", "Input 2", "Digital mixer"}; + +static const struct soc_enum uda1341_mixer_enum[] = { +SOC_ENUM_SINGLE(UDA1341_DATA010, 0, 0x03, uda1341_dsp_setting), +SOC_ENUM_SINGLE(UDA1341_DATA010, 3, 0x03, uda1341_deemph), +SOC_ENUM_SINGLE(UDA1341_EA010, 0, 0x03, uda1341_mixmode), +}; + +static const struct snd_kcontrol_new uda1341_snd_controls[] = { +SOC_SINGLE("Playback Volume", UDA1341_DATA000, 0, 0x3F, 1), +SOC_SINGLE("Mic gain", UDA1341_EA010, 2, 0x07, 0), +SOC_SINGLE("Channel 1 mixer gain", UDA1341_EA000, 0, 0x1F, 1), +SOC_SINGLE("Channgel 2 mixer gain", UDA1341_EA001, 0, 0x1F, 1), +SOC_SINGLE("Input channel 2 amp gain", UDA1341_EA101, 0, 0x1F, 0), + +SOC_SINGLE("Bass boost", UDA1341_DATA001, 2, 0xF, 0), +SOC_SINGLE("Treble", UDA1341_DATA001, 0, 3, 0), + +SOC_ENUM("DSP setting", uda1341_mixer_enum[0]), +SOC_ENUM("Playback De-emphasis", uda1341_mixer_enum[1]), +SOC_ENUM("Mixer mode", uda1341_mixer_enum[2]), + +/* This should be an ext control with own handler, if one wants + to set the values in 0.5dB steps instead of 3dB */ +SOC_SINGLE("AGC output level", UDA1341_EA110, 0, 0x03, 1), +SOC_SINGLE("AGC time const", UDA1341_EA110, 2, 0x07, 0), + +SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0), +SOC_SINGLE("ADC Polarity switch", UDA1341_STATUS1, 4, 1, 0), +SOC_SINGLE("DAC Polarity switch", UDA1341_STATUS1, 3, 1, 0), +SOC_SINGLE("Double speed playback switch", UDA1341_STATUS1, 2, 1, 0), +}; + +static int uda1341_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(uda1341_snd_controls); i++) { + if ((err = snd_ctl_add(codec->card, + snd_soc_cnew(&uda1341_snd_controls[i],codec, NULL))) < 0) + return err; + } + + return 0; +} + +struct snd_soc_codec_dai uda1341_dai = { + .name = "UDA1341", + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1341_RATES, + .formats = UDA1341_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1341_RATES, + .formats = UDA1341_FORMATS, + }, + /* pcm operations */ + .ops = { + .hw_params = uda1341_hw_params, + }, + /* DAI operations */ + .dai_ops = { + .digital_mute = uda1341_mute, + .set_sysclk = uda1341_set_dai_sysclk, + .set_fmt = uda1341_set_dai_fmt, + } +}; +EXPORT_SYMBOL(uda1341_dai); + + +static int uda1341_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct uda1341_priv *uda1341; + void * codec_setup_data = socdev->codec_data; + int ret = -ENOMEM; + + printk(KERN_INFO "UDA1341 SoC Audio Codec\n"); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return ret; + + codec = socdev->codec; + + uda1341 = kzalloc(sizeof(struct uda1341_priv), GFP_KERNEL); + if (uda1341 == NULL) + goto priv_err; + + codec->private_data = uda1341; + + codec->reg_cache = kmemdup(uda1341_reg, sizeof(uda1341_reg), \ + GFP_KERNEL); + + if (codec->reg_cache == NULL) + goto reg_err; + + mutex_init(&codec->mutex); + + codec->reg_cache_size = sizeof(uda1341_reg); + codec->reg_cache_step = 1; + + codec->name = "UDA1341"; + codec->owner = THIS_MODULE; + codec->dai = &uda1341_dai; + codec->num_dai = 1; + codec->read = uda1341_read_reg_cache; + codec->write = uda1341_write; + codec->dapm_event = uda1341_dapm_event; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + if (!codec_setup_data) { + printk(KERN_ERR "UDA1341 SoC codec: \ + Unable to find an L3 bus adapter\n"); + ret = -ENODEV; + goto pcm_err; + } + + codec->control_data = codec_setup_data; + codec->hw_write_l3 = (hw_write_l3_t)l3_write; + + uda1341_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register pcms\n"); + goto pcm_err; + } + + ret = uda1341_add_controls(codec); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register controls\n"); + goto pcm_err; + } + + //uda1341_add_widgets(codec); + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); +reg_err: + kfree(codec->private_data); +priv_err: + kfree(codec); + return ret; +} + +/* power down chip */ +static int uda1341_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} + +#if defined(CONFIG_PM) +static int uda1341_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + return 0; +} + +static int uda1341_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda1341_reg); i++) { + codec->write(codec, i, *cache++); + } + uda1341_dapm_event(codec, codec->suspend_dapm_state); + return 0; +} +#endif /* CONFIG_PM */ + +struct snd_soc_codec_device soc_codec_dev_uda1341 = { + .probe = uda1341_soc_probe, + .remove = uda1341_soc_remove, +#if defined(CONFIG_PM) + .suspend = uda1341_soc_suspend, + .resume = uda1341_soc_resume, +#endif /* CONFIG_PM */ +}; + +EXPORT_SYMBOL(soc_codec_dev_uda1341); + +MODULE_DESCRIPTION("UDA1341 ALSA soc codec driver"); +MODULE_AUTHOR("Zoltan Devai"); +MODULE_LICENSE("GPL"); --- /dev/null +++ /sound/soc/codecs/uda1341.h @@ -0,0 +1,47 @@ +/* + * uda1341.h -- UDA1341 ALSA SoC Codec driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * 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. + */ + +#ifndef _UDA1341_H +#define _UDA1341_H + +#define UDA1341_L3ADDR 5 +#define UDA1341_DATA0_ADDR ((UDA1341_L3ADDR << 2) | 0) +#define UDA1341_DATA1_ADDR ((UDA1341_L3ADDR << 2) | 1) +#define UDA1341_STATUS_ADDR ((UDA1341_L3ADDR << 2) | 2) + +#define UDA1341_EXTADDR_PREFIX 0xC0 +#define UDA1341_EXTDATA_PREFIX 0xE0 + +/* UDA1341 registers */ +#define UDA1341_EA000 0 +#define UDA1341_EA001 1 +#define UDA1341_EA010 2 +#define UDA1341_EA011 3 +#define UDA1341_EA100 4 +#define UDA1341_EA101 5 +#define UDA1341_EA110 6 +#define UDA1341_EA111 7 +#define UDA1341_STATUS0 8 +#define UDA1341_STATUS1 9 +#define UDA1341_DATA000 10 +#define UDA1341_DATA001 11 +#define UDA1341_DATA010 12 +#define UDA1341_DATA1 13 + +#define UDA1341_REGS_NUM 14 + +#define STATUS0_DAIFMT_MASK (~(7<<1)) +#define STATUS0_SYSCLK_MASK (~(3<<4)) + +extern struct snd_soc_codec_dai uda1341_dai; +extern struct snd_soc_codec_device soc_codec_dev_uda1341; + +#endif /* _UDA1341_H */

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[alsa-devel] [PATCH] Add SSID for Dell XPS400 to stac922x_cfg_tbl[]
by Daniel T. Chen 08 Jun '07

08 Jun '07

Hi, Attached please find an hg changeset against current alsa-kernel tip adding a quirk entry for a Dell XPS400. Thanks, Daniel Chen

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[alsa-devel] Testing/forcing XRUN
by Eliot Blennerhassett 08 Jun '07

08 Jun '07

Hi, I want to test my driver behaviour when an xrun happens. I have tried using aplay/arecord, and suspending (ctrl-Z) them while playing/recording but this doesn't have the desired effect. (I don't see my driver code detecting an xrun). Instead I see a trigger-stop command to the driver. The apps do report an overrun when they are restarted though. Can anyone answer these: 1) Is this a valid way to simulate an xrun due to application not keeping up? 1.1) If not, what is? Do I have to write an app that deliberately starves the streams? 2) Where is the xrun being detected if my driver doesnt see it? Where does the trigger-stop command come from when I suspend the app? thanks Eliot Blennerhassett AudioScience Inc.

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[alsa-devel] Audio driver for omap 24xx.
by Pharaoh . 07 Jun '07

07 Jun '07

Hi I am referring the EAC driver for omap 24xx which is used by n800 board. The EAC 24xx driver doesn't look complete since it has only hardware init sequences and very basic ALSA related functionalities i.e. register the sound card etc. Is omap 24xx eac driver still in intermediate stage?Or it is complete but not yet integrated in to omap git tree? In later case, where can I find the complete eac driver? Also, omap 24xx driver doesn't seem to follow the framework/structure of other omap alsa drivers in sound/arm/omap/ , any particular reason? Please enlighten. I know this might be little off topic here, but there are only two places where I can ask this question, so I am trying both. -pharaoh.

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[alsa-devel] Surround sound problem with CS46xx
by Vasilis Manwlopoulos 06 Jun '07

06 Jun '07

Hi to everybody, I am kind of new in linux and to alsa and I am facing a problem with the setup for 5.1 surround sound with my sound card Hercules Fortissimo III, recognized having the chip cs46xx. I have managed to enable surround sound by creating a .asounrc file with the followings pcm.!default { type plug slave.pcm "surround51" slave.channels 6 route_policy duplicate } The problem is that I am facing weird cpu load spikes (sometimes isn't just spikes, but just 99% usage for a long time!) even when just playing a normal mp3 file at 128Kbps. This is happening when playing with various players (totem,rhythbox,vlc). Also the sound has cracklings or there is a delay between the front and the rear speakers. i have send this email to alsa-users too, but since there wasn't anyone to propose a workaround I was adviced to also send it to alsa-devel. I am using the distro ubuntu 7.04 with the default alsa libraries version 1.0.13ubuntu5. So, if anyone has any idea plz give me a hint, cause I am stuck with it about two weeks now, without having any clue to proceed with!!! PS. If any other information or command output is necessary to help me, tell me so.

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[alsa-devel] [ASoC]Changing the order of trigger in soc_pcm_trigger
by Nobin Mathew 06 Jun '07

06 Jun '07

Liam, I have one suggestion. Can we use this if (rtd->codec_dai->ops.trigger) { ret = rtd->codec_dai->ops.trigger(substream, cmd); if (ret < 0) return ret; } if (rtd->cpu_dai->ops.trigger) { ret = rtd->cpu_dai->ops.trigger(substream, cmd); if (ret < 0) return ret; } if (platform->pcm_ops->trigger) { ret = platform->pcm_ops->trigger(substream, cmd); if (ret < 0) return ret; } Instead of if (rtd->codec_dai->ops.trigger) { ret = rtd->codec_dai->ops.trigger(substream, cmd); if (ret < 0) return ret; } if (platform->pcm_ops->trigger) { ret = platform->pcm_ops->trigger(substream, cmd); if (ret < 0) return ret; } if (rtd->cpu_dai->ops.trigger) { ret = rtd->cpu_dai->ops.trigger(substream, cmd); if (ret < 0) return ret; } That means trigger the Cpu before pcm. Because most of the controllers needs to be initialized and triggered before external DMA is triggered. Correct me if i am going wrong. This was causing so many problems in my hardware. Nobin Mathew

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[alsa-devel] Realtek 268
by Steven Zakulec 06 Jun '07

06 Jun '07

Hello ALSA developers, I have one of the Intel HDA cards that is not supported currently: Realtek 268. It was previously mentioned in May (http://mailman.alsa-project.org/pipermail/alsa-devel/2007-May/000828.html). The post seemed to imply it was not a terribly difficult task to add support for the card, but was very vague on details. If it is in fact just a matter of copying an existing preset and changing the values, I would probably try to do it myself. If it is any more difficult than that, I'd like to formally ask the developers if they could add any support for the card. The lspci and /proc/asound/codec info is identical to that in the posting I mentioned. If you need anymore information, I would be happy to provide it. Thank you!

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[alsa-devel] Connecting two USB-Soundcard to JACK
by Carsten Beth 06 Jun '07

06 Jun '07

Hello! We want to connect two USB soundcards to one multichannel device at the alsa layer and to use that virtual device with JACK. But we don't find a way to do so. The USB soundcard is self-made. It woks as a high speed USB device with sample rates of up to 96 KHz, has four capture and two playback cannels. The clocks of two cards can be synchronized at hardware level. Using one card by its own works perfectly both, at the alsa and at the JACK level, also as a duplex device. All is done with alsalib 1.0.13, jackd 0.101.1 and qjackctl 0.2.21. Now I'm trying to build a virtual device with 8 capture and 4 playback channels at the alsa level. But, as it can be read at several locations, this only works for capture only or playback. It has been reported, that duplex operation is possible after patching pcm_multi.c. I have applyed the patch, but this has not changed anything. Now I want to ask you for support, to get the device running. I made some experiments with different configurations of alsa/jackd and had some strange results. You can read about the results and used the asoundrc below. Questions, that I can't answer are - Why do I see just the channels of one card when using the multi plugin in try 2), 5) and 7)? - Why is the processor load so high with playback only in try 2) and 3) but not in duplex operation? - Why are there lots non-adible xruns with playback only in try 2) and 3)? 1) playback only with p_multi ALSA: mmap-based access is not possible for the playback stream of this audio interface 2) playback only with p_multi_plug Playback of just the 1st cards two channel. Lot of processor load: Both cores of the Core Duo 2GHz system are at about 90% load. Lot of XRUNs: 08:01:35.514 XRUN callback (48501 skipped). 08:01:37.517 XRUN callback (80633 skipped). 08:01:39.519 XRUN callback (70612 skipped). but XRUNS can't be heard. Sometimes jackd terminates after a while: ALSA: could not complete playback of 256 frames: error = -32 cycle execution failure, exiting DRIVER NT: could not run driver cycle jack main caught signal 12 [...] zombified - calling shutdown handler [...] 08:01:40.926 Post-shutdown script terminated with exit status=256. 3) playback only with p_multi_route Number of channels as expected (4 channels, two from each card). Lot of processor load, like 2) Lot of non-adible XRUNS, like 2). Sometimes jackd terminates after a while with the same message as in 2). 4) capture only with c_multi ALSA: mmap-based access is not possible for the capture stream of this audio interface 5) capture only with c_multi_plug Captures just two channel from the 1st card Low processor load (near 0%) No XRUNS. 6) capture only with c_multi_route Number of channels as expected (8 channels, four from each card). Low load and no XRUNS, like 5) 7) duplex with p_multi_plug and c_multi_plug Playback and capture works just for the 1st card (two playback and 4 capture channels). Low processor load, like 2) Lot of XRUNs, but less the 2) (and now the XRUNS can be heard). Here is a snip: **** alsa_pcm: xrun of at least 0.010 msecs 11:34:00.013 XRUN callback (9 skipped). **** alsa_pcm: xrun of at least 0.014 msecs 11:34:02.037 XRUN callback (11 skipped). **** alsa_pcm: xrun of at least 0.018 msecs **** alsa_pcm: xrun of at least 0.016 msecs 11:34:04.069 XRUN callback (11 skipped). **** alsa_pcm: xrun of at least 0.016 msecs 11:34:06.099 XRUN callback (11 skipped). **** alsa_pcm: xrun of at least 0.018 msecs **** alsa_pcm: xrun of at least 0.013 msecs 11:34:08.116 XRUN callback (12 skipped). Sometimes jackd terminates after a while with the same message as in 2). 8) duplex with p_multi_route and c_multi_route Exactly the same as in 7) expect for the number of channels. Now all channels are available (four playback and four capture channels) Here the used asoundrc: ##################################### # 1st USB card ##################################### pcm.usb0 { type hw card 1 } ctl.usb0 { type hw card 1 } ##################################### # 2nd USB card ##################################### pcm.usb1 { type hw card 2 } ctl.usb1 { type hw card 2 } pcm.p_multi { type multi; slaves.a.pcm "usb0"; slaves.a.channels 2; slaves.b.pcm "usb1"; slaves.b.channels 2; bindings.0.slave a; bindings.0.channel 0; bindings.1.slave a; bindings.1.channel 1; bindings.2.slave b; bindings.2.channel 0; bindings.3.slave b; bindings.3.channel 1; } ctl.p_multi { type hw; card 1; } pcm.c_multi { type multi; slaves.a.pcm "usb0"; slaves.a.channels 4; slaves.b.pcm "usb1"; slaves.b.channels 4; bindings.0.slave a; bindings.0.channel 0; bindings.1.slave a; bindings.1.channel 1; bindings.2.slave a; bindings.2.channel 2; bindings.3.slave a; bindings.3.channel 3; bindings.4.slave b; bindings.4.channel 0; bindings.5.slave b; bindings.5.channel 1; bindings.6.slave b; bindings.6.channel 2; bindings.7.slave b; bindings.7.channel 3; } ctl.c_multi { type hw; card 1; } pcm.p_multi_plug { type plug; slave.pcm "p_multi"; ttable.0.0 1; ttable.1.1 1; ttable.2.2 1; ttable.3.3 1; } ctl.p_multi_plug { type hw; card 1; } pcm.c_multi_plug { type plug; slave.pcm "c_multi"; ttable.0.0 1; ttable.1.1 1; ttable.2.2 1; ttable.3.3 1; ttable.4.4 1; ttable.5.5 1; ttable.6.6 1; ttable.7.7 1; } ctl.c_multi_plug { type hw; card 1; } pcm.p_multi_route { type route; slave.pcm "p_multi"; ttable.0.0 1; ttable.1.1 1; ttable.2.2 1; ttable.3.3 1; } ctl.p_multi_route { type hw; card 1; } pcm.c_multi_route { type route; slave.pcm "c_multi"; ttable.0.0 1; ttable.1.1 1; ttable.2.2 1; ttable.3.3 1; ttable.4.4 1; ttable.5.5 1; ttable.6.6 1; ttable.7.7 1; } ctl.c_multi_route { type hw; card 1; } Thank you Carsten Beth -- Dipl.-Inform. Carsten Beth OFFIS - Institute for Information technology R&D Division Embedded Hardware-/Software-Systems Escherweg 2 - 26121 Oldenburg - Germany - http://www.offis.de/ Phone: +49-441-9722-241 - Fax: +49-441-9722-282

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Re: [alsa-devel] Power management state flow in ALSA
by Nobin Mathew 06 Jun '07

06 Jun '07

Liam, In soc_resume() we should do this snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0); instead of snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot); Because snd_pcm_resume in pcm_native.c expects the card to be in SNDRV_CTL_POWER_D0 state. Other ioctl will wait forever. static int snd_pcm_resume(struct snd_pcm_substream *substream) { struct snd_card *card = substream->pcm->card; int res; snd_power_lock(card); if ((res = snd_power_wait(card, SNDRV_CTL_POWER_D0)) >= 0) res = snd_pcm_action_lock_irq(&snd_pcm_action_resume, substream, 0); snd_power_unlock(card); return res; } One more thing why we are using snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3cold); every other driver is doing snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); SNDRV_CTL_POWER_D3cold is true for deep sleep where power to the controller is switched off . In sleep, only clock to the controller is switched off. Nobin Mathew On 6/4/07, Nobin Mathew <nobin.mathew(a)gmail.com> wrote: > Liam > > I will get back, i need to set up the board > > On 6/1/07, Liam Girdwood <lg(a)opensource.wolfsonmicro.com> wrote: > > Nobin, > > > > Can you try the following patch :- > > > > http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… > > > > This will notify pcm's of their suspend state. > > > > Liam > > > > On Wed, 2007-05-30 at 12:58 +0200, Takashi Iwai wrote: > > > At Wed, 30 May 2007 16:06:54 +0530, > > > Nobin Mathew wrote: > > > > > > > > I am writing an ASoC driver, where i can place these calls > > > > > > > > snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); > > > > snd_pcm_suspend_all(chip->pcm[i]); > > > > > > > > and snd_power_change_state(card, SNDRV_CTL_POWER_D0); > > > > > > > > > > > > In soc-core.c ? > > > > > > I suppose so. > > > In soc_suspend() and soc_resume(), you can get the card instance from > > > codec->card, at least. > > > > > > But currently SoC doesn't keep the PCM instances, so it cannot be > > > implemented as it is... Liam, any plan or known issue regarding this? > > > > > > > > > Takashi > > > > >

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[alsa-devel] Problem restarting alsa: can't set buffer size
by Juan Manuel Lopez Baio 06 Jun '07

06 Jun '07

Hello. I'm developing an application that records input from ALSA and process it. To do so I user the API starting with snd_pcm_open and follow with the parameters settings and so on, and once the app is done and exits, it closes with snd_pcm_hw_params_free and snd_pcm_close. The problem is that afterwards, if I restart the application, I can't set up ALSA, becausea continuous calls to snd_pcm_hw_params_set_buffer_size always fail (I start with a buffer size of 8192 and keep dividing it by 2). I finally have to restart the system in order to get it to work again. I'm on a Mandriva box with gcc 4.1.2 ¿Any thoughts on this? Thanks in advance! juan

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[alsa-devel] using 'type softvol' in a card conf file in /usr/share/alsa/cards, am i doing this right?
by John L. Utz III 05 Jun '07

05 Jun '07

Hello; I am working with an CMEDIA 8738-MC6 and i am trying to gain control of the second DAC by way of softvolume. but i seem to not be succeeding because my application cant find my 'Rear' control, am i doing this correctly? the original /usr/share/alsa/cards/CMI8738-MC6.conf CMI8738-MC6.pcm.rear.0 { @args [ CARD ] @args.CARD { type string } type hw card $CARD device 1 } i have modified it to look thusly: notice that the type is now softvol and the original hardware description is encapsulated in slave.pcm # 2nd DAC CMI8738-MC6.pcm.rear.0 { @args [ CARD ] @args.CARD { type string } type softvol slave.pcm { type hw card $CARD device 1 } control { name "Rear Playback Volume" card $CARD } } Shouldnt i now be able to see 'Rear Playback Volume' in the output of amixer controls ?? as it stands right now, i can not. what am i doing incorrectly? tnx! johnu

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Re: [alsa-devel] [2/5] 2.6.22-rc4: known regressions
by Michal Piotrowski 05 Jun '07

05 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc4. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions ACPI Subject : ACPI_CPUFREQ fails to load - HP ML110 G4 Xeon 3040 References : http://bugzilla.kernel.org/show_bug.cgi?id=8570 Submitter : Marc Tuerschmann <marc(a)tuerschmann.de> Status : Unknown (possible hardware issues) Subject : unable to shutdown on kernel 2.6.22-rc2 References : http://bugzilla.kernel.org/show_bug.cgi?id=8516 Submitter : Thierry Volpiatto <tvolpiatt(a)neuf.fr> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Status : Unknown File systems Subject : OOPS (NULL pointer dereference) in v2.6.22-rc3 References : http://lkml.org/lkml/2007/6/1/389 Submitter : Alex Riesen <raa.lkml(a)gmail.com> Status : Unknown Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

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[alsa-devel] Alsa-OSS Duplex bug (revisited)
by Alan Horstmann 05 Jun '07

05 Jun '07

About a year ago I posted an explaination of the cause of non-functioning with Alsa-OSS in duplex, together with a workaround patch. After some discussion the thread ended. However since then there has been a steady trickle of downloads of the patches from my host (it can also be found elsewhere) and some reports of great success,eg http://audacityteam.org/forum/thread/1388 (2nd page especially) In attempting to tidy up some loose ends, I hope you can bear with me revisiting this and making the case for it's acceptance. I am aware the Alsa-OSS is not of much interest to developers, and it might be said 'use native Alsa'; however that is not possible or desirable in all applications, and anyway, why provide an OSS emulation that cannot be made to work properly? To summarise the problem first: OSS provides a single SNDCTL_DSP_CHANNELS ioctl; where a device is duplex the number of capture and playback channels cannot be set separately. With original OSS this was not a problem AFAIK because separate devices are created for capture and playback, ie they are not used duplex. However Alsa-OSS in most cases creates a single combined device, which only operates correctly when an equal number of capture and playback channels are used. The Workaround: The workaround I proposed and now use constantly is like a secret trapdoor. Where the existing interface is adequate (ie captue channels == playback channels, or non-duplex) nothing changes, and SNDCTL_DSP_CHANNELS is set with a number between 0 and 128. It is fully compatible with apps that don't know of the secret door -the front door is used! However, if an app needs to set capture channels != playback channels, in duplex, at present the stream will malfunction every time. But with this patch applied, if the app knows the workaround, then it can simply set SNDCTL_DSP_CHANNELS as playback + 256*capture channels, and this will be interpreted correctly in the patched Alsa-OSS and the stream functions correctly. As I say, there is therefore no loss of compatibility in either direction. Providing the app only uses the special formula when necessary rather than always, it is only providing a way to fix something that is broken. It would seem to me of value to add the patch to Alsa so that the workaround can be effected by changes to the app only, without also having to patch and compile a replacement Alsa. Whilst it is by no means ideal, it does overcome the underlying problem successfully. Details of the workaround could be added to the OSS-Emulation.html document. Patch for 1.0.14rc4 is attached for consideration, together with example portaudio patch. Regards Alan

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[alsa-devel] ALSA 1.0.14 release
by Jaroslav Kysela 04 Jun '07

04 Jun '07

Hi all, finally, ALSA 1.0.14 is out. For list of changes, please, check: http://www.alsa-project.org/changes/v1-0-13--v1-0-14.txt http://www.alsa-project.org/changes/v1-0-14rc4--v1-0-14.txt Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] Alsa guru needed for short-term contracting gig
by Darren Gibbs 04 Jun '07

04 Jun '07

Are there any SF Bay Area local Alsa experts out there that could provide assistance bringing up Alsa with driver for an I2S codec on an ARM embedded system?

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[alsa-devel] [PATCH] ALSA: Fix prototype for snd_soc_dai_link.init() function pointer
by Timur Tabi 04 Jun '07

04 Jun '07

Change the snd_soc_dai_link.init() function to take a pointer to the corresponding snd_soc_dai_link structure, instead of a pointer to a snd_soc_codec structure. This allows the initialization function to initialize its own structure. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This is a fix for ALSA bug 3135 (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3135). Since I don't have an ARM cross-compiler, and currently all ASoC drivers are for ARM platforms only, I'm not sure if this fix is correct. include/sound/soc.h | 2 +- sound/soc/at91/eti_b1_wm8731.c | 3 ++- sound/soc/pxa/corgi.c | 3 ++- sound/soc/pxa/poodle.c | 3 ++- sound/soc/pxa/spitz.c | 3 ++- sound/soc/pxa/tosa.c | 3 ++- sound/soc/soc-core.c | 2 +- 7 files changed, 12 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index db6edba..86e1b1d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -410,7 +410,7 @@ struct snd_soc_dai_link { struct snd_soc_ops *ops; /* codec/machine specific init - e.g. add machine controls */ - int (*init)(struct snd_soc_codec *codec); + int (*init)(struct snd_soc_dai_link *codec); }; /* SoC machine */ diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 820a676..a4f5a99 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -216,9 +216,10 @@ static const char *intercon[][3] = { /* * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. */ -static int eti_b1_wm8731_init(struct snd_soc_codec *codec) +static int eti_b1_wm8731_init(struct snd_soc_dai_link *dai) { int i; + struct snd_soc_codec *codec = dai->codec_dai->codec; DBG("eti_b1_wm8731_init() called\n"); diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 5ee51a9..43237f9 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -286,9 +286,10 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = { /* * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device */ -static int corgi_wm8731_init(struct snd_soc_codec *codec) +static int corgi_wm8731_init(struct snd_soc_dai_link *dai) { int i, err; + struct snd_soc_codec *codec = dai->codec_dai->codec; snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 0915cf7..5e8cb76 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -246,9 +246,10 @@ static const snd_kcontrol_new_t wm8731_poodle_controls[] = { /* * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device */ -static int poodle_wm8731_init(struct snd_soc_codec *codec) +static int poodle_wm8731_init(struct snd_soc_dai_link *dai) { int i, err; + struct snd_soc_codec *codec = dai->codec_dai->codec; snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 80e8210..95dcaa4 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -292,9 +292,10 @@ static const struct snd_kcontrol_new wm8750_spitz_controls[] = { /* * Logic for a wm8750 as connected on a Sharp SL-Cxx00 Device */ -static int spitz_wm8750_init(struct snd_soc_codec *codec) +static int spitz_wm8750_init(struct snd_soc_dai_link *dai) { int i, err; + struct snd_soc_codec *codec = dai->codec_dai->codec; /* NC codec pins */ snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0); diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 5504e30..b438349 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -192,9 +192,10 @@ static const struct snd_kcontrol_new tosa_controls[] = { tosa_set_spk), }; -static int tosa_ac97_init(struct snd_soc_codec *codec) +static int tosa_ac97_init(struct snd_soc_dai_link *dai) { int i, err; + struct snd_soc_codec *codec = dai->codec_dai->codec; snd_soc_dapm_set_endpoint(codec, "OUT3", 0); snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 92d5d91..654a517 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1093,7 +1093,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) mutex_lock(&codec->mutex); for(i = 0; i < machine->num_links; i++) { if (socdev->machine->dai_link[i].init) { - err = socdev->machine->dai_link[i].init(codec); + err = socdev->machine->dai_link[i].init(&machine->dai_link[i]); if (err < 0) { printk(KERN_ERR "asoc: failed to init %s\n", socdev->machine->dai_link[i].stream_name); -- 1.5.0.2.260.g2eb065

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[alsa-devel] SALSA-Lib version 0.0.1 release
by Takashi Iwai 04 Jun '07

04 Jun '07

Hi all, I created a small replacement of ALSA-library, mainly for embedded systems with limited resources. The library is named as "SALSA-Lib" (too easy, eh?) It has no plugin layer but only hw layer, thus no sexy features like dmix and format conversions are available. It supports no shiny and complex configuration files. And it supports no sequencer. The only goal of this library is the reduction of the size of library and running binaries by inlining functions with direct syscalls. The library is designed to be source-level API compatible with the normal alsa-lib 1.0.14. Note that it is _not_ designed to be binary-compatible at all, and it will be never so. The first released version 0.0.1 is available at: ftp://ftp.suse.com/pub/people/tiwai/salsa-lib/salsa-lib-0.0.1.tar.bz2 The package is released under LGPL as well as the original alsa-lib. I've tested it only on i386, and tests on other architectures would be greatly appreciated. (And, no, I didn't make it on HG tree yet.) Takashi

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[alsa-devel] Datasheet Needed
by chris＠idlelion.net 04 Jun '07

04 Jun '07

(working on bug #305) I'm looking for a proper datasheet on the Cirrus Logic / Crystal Semiconductor CS4624 (or CS4622) sound chip. It's used in a number of popular laptops made around 1999-2001. I don't need the 7- or 8-page feature list, or the 30 (or so) page reference design document. I need register lists and theory of operation and a discussion of power modes. The chip was end-of-lifed in 2004. Many thanks, Chris

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[alsa-devel] v1.0.14 - test packages #2
by Jaroslav Kysela 04 Jun '07

04 Jun '07

Hi, I recreated driver and tools packages in /pub/testing to fix reported problems. Please, test... Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] Problem with ncurses and alsamixer (cannot open terminal: vt100 or xterm)
by David Topper 04 Jun '07

04 Jun '07

I finally managed to get the alsa-1.0.4 utils to compile on an ARM based SBC. When I launch alsamixer, however, I get an error: Error opening terminal: xterm When I ssh into the board. Or: Error opening terminal: vt100 When connected via serial cable. I suspect this has something to do with the linking (or lack thereof) of alsamixer for ncurses. The ncurrses library, by the way, I cross compiled with the following command: CC=/opt/crosstool/arm-unknown-linux-gnu/gcc-3.3.4-glibc-2.3.2/bin/arm-unknown-linux-gnu-gcc ./configure --host=arm-linux --prefix=/home/topper/ts-packages/ --exec-prefix=/home/topper/ts-packages/ But for the alsa utils, I simple ran ./configure, and the script detected the ncurses library, ran through and compiled. Yet, something seems amuck because: root@ts7000:root# ldd /bin/alsamixer libasound.so.2 => /lib/libasound.so.2 (0x2aac8000) libm.so.6 => /lib/libm.so.6 (0x2ab6f000) libdl.so.2 => /lib/libdl.so.2 (0x2abe9000) libpthread.so.0 => /lib/libpthread.so.0 (0x2abf3000) libc.so.6 => /lib/libc.so.6 (0x2ac4b000) /lib/ld-linux.so.2 => /lib/ld-linux.so.2 (0x2aaab000) Whereas on my Linux Intel laptop: awn:~$ ldd /usr/bin/alsamixer libncurses.so.5 => /lib/libncurses.so.5 (0x4002c000) libasound.so.2 => /usr/lib/libasound.so.2 (0x40069000) libm.so.6 => /lib/libm.so.6 (0x4010d000) libdl.so.2 => /lib/libdl.so.2 (0x40130000) libpthread.so.0 => /lib/libpthread.so.0 (0x40133000) libc.so.6 => /lib/libc.so.6 (0x40184000) /lib/ld-linux.so.2 => /lib/ld-linux.so.2 (0x40000000) So I have two questions: 1. How might ensure proper linking of the ncurses library. 2. Is there another way to tell alsa to activate various cards, set volume, etc.. so I can test to see if the alsa driver is even working on this board. Thanks, Dave Topper -- Technical Director - Virginia Center for Computer Music http://www.virginia.edu/music/VCCM/

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[alsa-devel] 1.0.14 test packages
by Jaroslav Kysela 04 Jun '07

04 Jun '07

Hello all, I put 1.0.14-final test packages to ftp://ftp.alsa-project.org/pub/testing Note that these packages are not an official release. They are indended for compilation tests. Anyway, if you find a compilation (not functionality) issue, please, let me know ASAP through e-mail. I'm going to release 1.0.14 packages on next Monday. Note for alsa developers: Please, no more HG repository updates until packages are released. Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] snd_hda_intel: Conexant CX20551 (Waikiki)
by Edwin Smulders 03 Jun '07

03 Jun '07

Hi, I believe there's still a problem with the HDA Intel chip Conexant CX20551 (Waikiki) not working. However, im not entirely sure the problem the problem is with alsa, because it starts working when i boot with acpi=off (a fix suggested on the ubuntu forums, but not quite useful on a laptop [plus, it breaks my ethernet]) The reason I'm mailing is, can you confirm this is something alsa can/should/will fix or are we all looking in the wrong direction and this is really an acpi problem? If you can fix this or know of something that might work, I would be happy to hear it. I've attached my lspci -nv and cat /proc/asound/card0/codec\#* just in case. I would be happy to give you any more info you might need. ~~ Edwin

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[alsa-devel] [PATCH] Add support for another X-Fi Extreme Audio (really SB Live 24bit)
by Daniel T. Chen 03 Jun '07

03 Jun '07

Hi, Attached please find an hg changeset against current alsa-kernel tip. Thanks, Daniel Chen

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[alsa-devel] Next steps
by chris＠idlelion.net 03 Jun '07

03 Jun '07

Hey gang, I'm trying to work on issue 305, suspend/resume for cs46xx. I'm running CentOS 5, so I'd like to work on what's current for that distribution first, and patch to latest if needed after that. I think I have the correct source package, but the only mention of cs46xx I could find in the source is a configuration file. Running make doesn't make the cs46xx module, AFAICT. Which source (package)(s) should I have? How do I track down the source file(s) I need to examine? Thanks, Chris

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Re: [alsa-devel] ASoC and a codec that can't be controlled
by Timur Tabi 02 Jun '07

02 Jun '07

Liam Girdwood wrote: >> I guess I just don't understand why the codec driver is acting like the "master" driver of >> ASOC. IMHO, the codec driver should be doing two things: > > I guess this is due to an ordering issue we had during early > development. We had to make sure that the I2C probe of the codec > succeeded before registering the card, pcms, etc. Fwiw, I do intend to > address this in future versions (I should probably add a road map to the > wiki now). In that case, I think we can solve this problem as well as the PowerPC device tree problem in one shot, because they're basically the same problem. Anyway, let me see if I get this right: 1. The first function called is eti_b1_init(). 2. eti_b1_init() calls platform_device_add() to add an soc-audio class device and register the eti_b1_snd_devdata structure. 3. The registration causes a number of things to happen, one of which is a call to soc_probe(). 4. soc_probe() sees that the snd_soc_machine_eti_b1.probe is NULL, so that part is skipped. 3. soc_probe() sees that snd_soc_machine_eti_b1.dai_link->cpu_dai->probe is also NULL, so that part is skipped. (eti_b1_snd_devdata.machine == snd_soc_machine_eti_b1) 4. soc_probe() sees that soc_codec_dev_wm8731.probe is not NULL, so it calls wm8731_probe(). 5. wm8731_probe() registers an I2C driver. 6. The I2C class driver calls wm8731_i2c_driver.attach_adapter, which is wm8731_i2c_attach(). 7. wm8731_i2c_attach() calls i2c_probe(), which in turn calls wm8731_codec_probe() 8. wm8731_codec_probe() allocates a snd_soc_codec structure. 8. wm8731_codec_probe() calls wm8731_init(). This means that if there is no I2C support, wm8731_init() will never be called. 9. wm8731_init() initializes the snd_soc_codec structure. Question: why doesn't wm8731_probe() initialize the non-I2C parts of the snd_soc_codec structure? For example, snd_soc_codec.dai, snd_soc_codec.name, and snd_soc_codec.owner? That way, the snd_soc_codec structure will be properly initialized even when there's no I2C. 10. wm8731_init() calls snd_soc_new_pcms. 11. wm8731_init() exits, wm8731_codec_probe() exits, wm8731_i2c_attach() exits, and then wm8731_probe() exits. Control returns to soc_probe(). 12. soc_probe() notices that eti_b1_snd_devdata.platform.probe is NULL, so it skips that. Question: why don't you define a function for at91_soc_platform.probe, and in that function you can call snd_soc_new_pcms()? That way, you guarantee the codec driver's I2C interface is initialized before snd_soc_new_pcms() is called, and you avoid have PCM calls in the codec driver. >> ASOC and the machine driver should then work in tandem to decide which driver will do what >> and which capabilities are *actually* supported. *Something* needs to look at the entire >> system and say to each device, "Well, yes, I know about this little feature of yours, but >> we're just not going to support that today." >> > > The machine driver pretty much already does this. It can override valid > hardware configurations and return -EINVAl if required. Do you have an example of that? Would that be eti_b1_hw_params()? -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] [1/3] 2.6.22-rc3: known regressions v2
by Michal Piotrowski 01 Jun '07

01 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc3. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Unclassified Subject : build failed in function `vic_sys_interrupt' References : http://lkml.org/lkml/2007/5/31/227 Submitter : Toralf Förster <toralf.foerster(a)gmx.de> Status : Unknown Subject : long freezes on thinkpad t60 References : http://lkml.org/lkml/2007/5/24/100 Submitter : Miklos Szeredi <miklos(a)szeredi.hu> Handled-By : Ingo Molnar <mingo(a)elte.hu> Status : problem is being debugged ACPI Subject : unable to shutdown on kernel 2.6.22-rc2 References : http://bugzilla.kernel.org/show_bug.cgi?id=8516 Submitter : Thierry Volpiatto <tvolpiatt(a)neuf.fr> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Status : Unknown Networking Subject : b44: regression in 2.6.22 References : http://lkml.org/lkml/2007/5/27/108 Submitter : Maximilian Engelhardt <maxi(a)daemonizer.de> Handled-By : Michael Buesch <mb(a)bu3sch.de> Status : problem is being debugged Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

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[alsa-devel] Configure problems for alsa-utils-1.0.4
by David Topper 01 Jun '07

01 Jun '07

So I've managed to compile alsa-drivers-1.0.4 for ARM. I can even load the modules. I've also successfully compiled alsa-libs-1.0.4 ... but I had to do it on the board. Cross compiling simply didn't seem to work. I've been using --prefix=/ and --exec-prefix=/ for configures. But when I do ./configure --prefix=/ --exec-prefix=/ for alsa-utils-1.0.4 it fails with: checking for libasound headers version >= 0.9.0... not present. configure: error: Sufficiently new version of libasound not found. But they're there, for sure. Perhaps it doesn't like looking in /include/alsa? Thanks, DT

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[alsa-devel] snd_soc_dai_link.init()
by Timur Tabi 01 Jun '07

01 Jun '07

Liam, Why does the init() function pointer in the snd_soc_dai_link structure take a snd_soc_codec pointer? Wouldn't it make more sense for it to take a snd_soc_dai_link pointer? That way, I could use the init() function to initialize the structure. I'm trying to write a function that will initialize the codec_dai pointer. I thought I could use snd_soc_dai_link.init(), but I don't know how to get the pointer to the snd_soc_dai_link structure from a snd_soc_codec structure. /* SoC machine DAI configuration, glues a codec and cpu DAI together */ struct snd_soc_dai_link { char *name; /* Codec name */ char *stream_name; /* Stream name */ /* DAI */ struct snd_soc_codec_dai *codec_dai; struct snd_soc_cpu_dai *cpu_dai; /* machine stream operations */ struct snd_soc_ops *ops; /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_codec *codec); }; -- Timur Tabi Linux Kernel Developer @ Freescale

2 7

[alsa-devel] Power management state flow in ALSA
by Nobin Mathew 01 Jun '07

01 Jun '07

I am having a doubt regarding ALSA power management. My understanding of APM suspend() is like this. Freeze the ALSA apps Call ALSA driver suspend () in the ALSA suspend() function it saves the current state of substream and changes the state of substream to SUSPENDED. My understanding of APM resume() is like this Call ALSA driver resume () Activate the ALSA apps In ALSA resume function it restores the saved state of substream. So my question is when ALSA app will get to know the SUSPENDED state of substream.??? I am using linux kernel 2.6.17 ALSA driver version 1.0.11rc4 ASoC patch version 0.12.4

3 9

[alsa-devel] ASoC and a codec that can't be controlled
by Timur Tabi 31 May '07

31 May '07

I'm working on some ASoC drivers for a new board, and we're using a CS4270. The CS4270 is interesting in that if it is connected in stand-alone mode, there is no way to configure it. The board wirings determine all the parameters. Therefore, my CS4270 codec driver will probably be very skimpy. My question is: does this mean that my cs4270.c file will *never* call these functions: snd_ctl_add snd_soc_cnew snd_soc_dapm_new_control snd_soc_dapm_connect_input snd_soc_dapm_new_widgets These are the functions used to add new controls and widgets. So how does ALSA know that it needs to call my I2S driver *instead* of my codec driver to do stuff like change volume? -- Timur Tabi Linux Kernel Developer @ Freescale

2 15

[alsa-devel] Will the real alsa wiki please stand up?
by j t 31 May '07

31 May '07

There something at: http://bugtrack.alsa-project.org/main/index.php/Main_Page which thinks it's the alsa wiki, and there something at: http://alsa.opensrc.org/Main_Page which alsa thinks it's the alsa wiki. Well, devs, which is the imposter? And would it be a good idea to reduce the number of distinct alsa wikis to something smaller, perhaps "1"? Jaime :-) PS: apologies to "To Tell the Truth".

5 7

[alsa-devel] Native Instruments - Audio Kontrol 1 - snd_usb_caiaq module
by Alex Pintilie 31 May '07

31 May '07

Hallo! I just registered to this mailing list because a developer kindly advised me to do so. I own the external sound card Audio Kontrol 1 from Native Instruments and I was happy to find that drivers for it are already in the making. I downloaded alsa-driver-1.0.14rc4 and compiled successfully with ./hgcompile, then I installed the module with: # install -m 644 snd-usb-caiaq.ko /lib/modules/`uname -r`/kernel/drivers/snd-usb-caiaq.ko # /sbin/depmod -a After the command: # modprobe snd-usb-caiaq I get the following error: FATAL: Error inserting snd_usb_caiaq (/lib/modules/2.6.20-15-lowlatency/kernel/drivers/snd-usb-caiaq.ko): Unknown symbol in module, or unknown parameter (see dmesg) dmesg shows the following: [10095.330000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_new [10095.330000] snd_usb_caiaq: Unknown symbol snd_pcm_new [10095.330000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_limit_hw_rates [10095.330000] snd_usb_caiaq: Unknown symbol snd_pcm_limit_hw_rates [10095.331000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_lib_preallocate_pages_for_all [10095.331000] snd_usb_caiaq: Unknown symbol snd_pcm_lib_preallocate_pages_for_all [10095.331000] snd_usb_caiaq: Unknown symbol snd_hidden_kfree [10095.331000] snd_usb_caiaq: Unknown symbol snd_verbose_printk [10095.331000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_lib_malloc_pages [10095.332000] snd_usb_caiaq: Unknown symbol snd_pcm_lib_malloc_pages [10095.332000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_lib_ioctl [10095.332000] snd_usb_caiaq: Unknown symbol snd_pcm_lib_ioctl [10095.332000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_lib_free_pages [10095.332000] snd_usb_caiaq: Unknown symbol snd_pcm_lib_free_pages [10095.332000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_set_ops [10095.332000] snd_usb_caiaq: Unknown symbol snd_pcm_set_ops [10095.333000] snd_usb_caiaq: Unknown symbol snd_hidden_kmalloc [10095.333000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_period_elapsed [10095.333000] snd_usb_caiaq: Unknown symbol snd_pcm_period_elapsed This may not look like much but it's the effort from a couple of days searching, reading and testing. I hope you can help me. -- Alex

4 5

[alsa-devel] Hardware-accelerated OpenAL support with ALSA?
by Chris Rankin 31 May '07

31 May '07

Hi, I have been looking at getting OpenAL working on my machine, since OpenAL is the only way that games under Windows Vista can now provide hardware-accelerated sound effects. This suggests that a lot of PC games could be adding OpenAL support soon, which means that it would be handy if Linux had an efficient native OpenAL implementation that Wine could wrap. Fedora Core does provide an OpenAL library from www.openal.org, but it looks as if this library's ALSA backend does not provide any hardware acceleration. (I have an "Audigy 2 ZS" sound card, which can at least support hardware mixing.) Various alternative ALSA-based solutions do seem to exist on the Web, e.g. http://dino.e4a.it/openal-alsa/ and http://www.lost.org.uk/openal.html However, it seems to me that OpenAL support would be better served as an actual ALSA package that distros would be able to ship. Are there any plans to provide official OpenAL support in ALSA, please? Thanks for any information, Chris P.S. My Audigy 2 also seems to have EAX support. Does the ALSA driver support that functionality too? I can't say that "World of Warcraft" seems to use it, although Wine's DirectSound3D support still has lots of gaps. _________________________________________________________ Inbox cluttering up with junk? Clean up with Yahoo! Mail. http://uk.docs.yahoo.com/mail/isp_targeting.html

1 0

[alsa-devel] Userspace Device Driver
by Klaus Zimmermann 31 May '07

31 May '07

Greetings List, on various places on the alsa site I read about the possibility of writing a userspace device driver. I am interested thatin, since I have a nice Edirol FA-66 ieee1394 interface which works nicely under jack with the ffado (http://ffado.org , former freebob) driver. >From what I read it seems I could either write an external plugin to the alsa-lib, which would be truly userspace and fine except I it would always have to be configured from .asoundrc and would probably not be found by legacy programs like ekiga. Or I could get really messy and have a userspace daemon relay audio data between the ffado library, i.e. librawieee1394, and a yet to be written alsa-driver module. This would at least yield the advantage, that the driver was indistinguishable from a true hardware driver. So: Is it possible to write a userspace driver that is indistinguishable from a hardware driver? TIA Klaus Zimmermann P.S.: I am aware of the jackplug plugin. However even with very conservative buffer settings I get underruns and x-runs quickly leading to a failing jack :(

1 0

Re: [alsa-devel] [RFC] [PATCH 0/2] ASoC sound support forSMDK2440boards
by Jose Henrique Spahn Torres 30 May '07

30 May '07

-----Original Message----- From: Zoltan Devai [mailto:zdevai@gmail.com] Sent: sexta-feira, 25 de Maio de 2007 14:40 To: Jose Henrique Spahn Torres Subject: Re: [alsa-devel] [RFC] [PATCH 0/2] ASoC sound support forSMDK2440boards 2007/5/25, Jose Henrique Spahn Torres <jose.torres(a)eldorado.org.br>: > > 2007/5/24, Jose Henrique Spahn Torres <jose.torres(a)eldorado.org.br>: > usbcore: registered new interface driver hci_usb Advanced Linux Sound > Architecture Driver Version 1.0.14rc4 (Wed May 16 > 09:45:46 2007 UTC). > ASoC version 0.13.1 > SMDK2440 ALSA SoC Audio driver > UDA1341 SoC Audio Codec > L3 write > L3 transfer > L3 sendbyte: 16 > L3 sendbyte: 40 > asoc: UDA1341 <-> s3c24xx-i2s mapping ok ALSA device list: > #0: SMDK2440 (UDA1341) > TCP cubic registered > ... Fine. > However, when triyng to play something I get this: > > asoc: machine hw_params failed > > This is in sound/soc/soc-core.c, line 494. > I couldn't go any futher then this. > Any hint? This combination of processor, crystal and codec only supports the following rates: 11.025 Khz, 22.050Khz and 44.100 Khz Please try to play a file with these sample rates. (If you have alsa-tools, you could try 'speaker-test -r 44.100'). BTW, could you tell me the freq of the crystal on the processor board ? (Is it 16.934 Mhz?) Zoltan Hi, I used speaker-test as indicated above and the result is: pendrive $ ./speaker-test -r 44100 speaker-test 1.0.11 Playback device is default Stream parameters are 44100Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory I have also tried: /pendrive $ ./aplay -f cd 01-jack_johnson-times_like_these-rns.wav ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:547: audio open error: No such file or directory /pendrive $ so I don't know what is happening here. Does anyone have any idea? Thanks, Henrique

3 3

[alsa-devel] Fw: hardware not asking for more data using asyn call back
by stan 30 May '07

30 May '07

Begin forwarded message: Date: Tue, 29 May 2007 15:19:12 -0400 From: "Ashlesha Shintre" <ashlesha.shintre(a)gmail.com> To: stanl(a)cox.net, alsa-devel(a)alsa-project.org Subject: Re: hardware not asking for more data using asyn call back Hi, Thanks Stan, for your response - Instead of copying pcm data from a wav file, i decided to copy all the data to a master buffer first and then see if the circular buffer implementation works - so now instead of copying from the wave file, i m copying data from the master buffer to the circular buffer.. however, as per your suggestion, if i copy the data in the while loop in main, then, it might not always be copied between 2 consecutive callbacks, but maybe more, as the callbacks are asynchronous. however, the hardware still does not ask for more data after executing the callback function about twice -- is there a way to flush the hardware buffer before beginning playback? I have pasted my code below Regards, Ashlesha. <snip> I've been doing some thinking about what you are trying to accomplish and have a suggestion. I think you would be better off letting jack handle the back end/sound portion. Run a daemon/service listening on an arbitrary tcp/ip port. On startup it initializes alsa and jack. When it receives a request, it forks a process which opens a new channel on jack and creates a buffer for it. It then plays the sound it receives on that link via jack, silence when the buffer is empty. You would have to write client software for the user on the network to contact your daemon/service. You might even find something like this already out there.

1 0

[alsa-devel] How the dma interrupt handlers have been registered in ALSA?
by Pharaoh . 30 May '07

30 May '07

Hi I am referring omap-alsa-dma.c and the related codec files. We usually register the IRQ handler by calling request_irq() function but in this mentioned files the IRQ handler etc is defined but it is not registered anywhere i.e. request_irq is not used anywhere. Is there any other way of registering irq handlers? The irq handlers address is passed to omap_request_dma, but there also it is not registered. I am missing the entire thing?

2 3

[alsa-devel] [1/4] 2.6.22-rc3: known regressions
by Michal Piotrowski 30 May '07

30 May '07

Hi all, Here is a list of some known regressions in 2.6.22-rc3. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Unclassified Subject : long freezes on thinkpad t60 References : http://lkml.org/lkml/2007/5/24/100 Submitter : Miklos Szeredi <miklos(a)szeredi.hu> Handled-By : Ingo Molnar <mingo(a)elte.hu> Status : problem is being debugged ACPI Subject : unable to shutdown on kernel 2.6.22-rc2 References : http://bugzilla.kernel.org/show_bug.cgi?id=8516 Submitter : Thierry Volpiatto <tvolpiatt(a)neuf.fr> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Status : Unknown File systems Subject : Oops in dentry_iput with 2.6.22-rc2 on AMD64 References : http://lkml.org/lkml/2007/5/22/4 Submitter : Florin Iucha <florin(a)iucha.net> Status : Unknown Kbuild Subject : make M=$PWD modules_install does nothing References : http://lkml.org/lkml/2007/5/27/190 Submitter : Andrey Borzenkov <arvidjaar(a)mail.ru> Status : Unknown Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

4 3

Re: [alsa-devel] hardware not asking for more data using asyn call back
by stan 30 May '07

30 May '07

Somehow I replied to the wrong message and it went to a personal email address. Putting it on the list so searches can find it (for what that is worth :-) ) Begin forwarded message: Date: Tue, 29 May 2007 18:18:22 -0700 From: stan <stanl(a)cox.net> To: "Ashlesha Shintre" <ashlesha.shintre(a)gmail.com> Subject: Re: hardware not asking for more data using asyn call back On Tue, 29 May 2007 15:19:12 -0400 "Ashlesha Shintre" <ashlesha.shintre(a)gmail.com> wrote: <snip> I didn't understand what you were asking before. It sounds to me like you want to have a thread running that accepts input from arbitrary processes and puts it in the main buffer. Then when the alsa callback occurs you want to take any new sound samples and send them to alsa. If that is the case, it makes more sense for the main buffer to be the circular buffer and the alsa buffer to be just a regular buffer. That's just personal preference though. As far as the handle error, I don't know what it means beyond what it says. You would have to look in the sources for the alsa library to see what causes that error to be generated. I don't see anything wrong in the logic of the code below, but the output indicates that you are not keeping up with the play rate. This is astonishing. At each play you can see that the available room in the buffer grows larger until larger than 1000, the buffer you have allocated. I'm not sure what infinite loop means and how your program is recovering from it. I presume that is where the time is being eaten up that isn't allowing you to keep up with the requests for data. I don't think you need to check if there is room when you get a callback. You should be able to just write the data to alsa using the writei call. i.e. eliminate the snd_pcm_avail_update call. I'm not familiar enough with alsa to help you further. Perhaps someone else can give you a hint or point you to the right place to look. > > /* Reading wav file into a buffer and then trying to play it */ > > > > #include <stdio.h> > > #include </usr/include/alsa/asoundlib.h> > > #include </usr/include/alsa/pcm.h> > > #include <sys/types.h> > > #include <unistd.h> > > #include <fcntl.h> > > > > #define SIZE 128 > > //mpos is the position in the master buffer to write from > > short int *buffer, *mbuf, *mpos; > > int ind=10, fd1, length; > > static snd_pcm_t *handle; > > static char *device = "default"; /* playback > > device */ > > snd_output_t *output = NULL; > > static short int *wto, *rfrom, buflen; > > > > void MyCallback(snd_async_handler_t *pcm_callback); > > > > int main() > > { > > int i,j,n,count,a,err,k=0; > > char c; > > snd_pcm_sframes_t frames; > > snd_async_handler_t *pcm_callback; > > > > > > > > if((fd1=open("SA2.WAV",O_RDONLY,0))==-1) > > printf("error opening wav file\n"); > > > > count=0; > > while(count++<40) > > a=read(fd1,&c,sizeof(char)); > > > > a=read(fd1,&length,sizeof(int)); > > > > n=length/SIZE; > > > > printf("length = %d\n",length); > > mbuf = (short int *) malloc (length * sizeof(short int)); > > mpos = &mbuf[0]; > > > > buflen = 4*SIZE; > > buffer = (short int *) malloc (buflen*sizeof(short int)); > > > > wto = NULL; > > rfrom = &buffer[0]; > > > > /* reading to master buffer */ > > count=0; > > a=1; > > while(count<length && a>0) > > a=read(fd1,&mbuf[count++],sizeof(short int)); > > if(a<0){ > > printf("error in reading from wav file\n"); > > exit(1); > > } > > close (fd1); > > > > for(count =0; count < buflen; count++) > > { > > buffer[count]=mbuf[count]; > > } > > /******************************/ > > > > /*ALSA DEV INIT*/ > > if((err = snd_pcm_open(&handle, device, > > SND_PCM_STREAM_PLAYBACK, 0)) < 0) { > > printf("error opening device: %s\n",snd_strerror(err)); > > return -1; > > } > > > > if((err = snd_pcm_set_params(handle, > > SND_PCM_FORMAT_S16_LE, > > SND_PCM_ACCESS_RW_INTERLEAVED, > > 1, > > 16000, > > 1, > > 1000)) < 0) { /* 2 sec */ > > printf("Playback open error: %s\n", snd_strerror(err)); > > return -2; > > } > > /*****************************/ > > > > printf("n=%d\n",n); > > for (i=0; i<50; i++) > > { > > printf("i=%d\n",i); > > frames = snd_pcm_writei(handle, rfrom, SIZE); > > if(frames < 0) // underrun > > { > > printf("underrun recovery\n"); > > frames = snd_pcm_recover(handle, frames,0); > > if(frames < 0){ > > printf("error in recovery\n"); > > return -3; > > } > > } > > if(frames >0 && frames < SIZE) > > printf("expected to write %d, wrote %d\n",SIZE,frames); > > > > printf("distance between ptrs before reinit is %d\n",rfrom > > - wto); if(rfrom < &buffer[buflen-1]) > > { > > wto = rfrom; > > rfrom = rfrom + SIZE; > > } > > if(rfrom >= &buffer[buflen-1]) > > { > > printf("rfm at the end, rfrom -buffer[buflen-1] > > =%d\n",(*rfrom - buffer[buflen-1])); > > //wto = rfrom; > > rfrom = &buffer[0]; > > } > > > > if((a = writetobuf())<0) > > printf("error in reading from wav file k = %d\n"); > > > > printf("buffer[i*SIZE] - rfrom = %d\n",(buffer[(i+1)*SIZE] - > > *rfrom)); > > > > } > > > > /* Async Handler */ > > > > > > err = > > snd_async_add_pcm_handler(&pcm_callback,handle,MyCallback,NULL); > > if(err<0) printf("add pcm handler error = > > %d\n%s\n",err,snd_strerror(err)); > > > > err = snd_pcm_start(handle); > > if(err<0) > > printf("error in starting snd pcm start err > > :%s\n",snd_strerror(err)); > > /******************************/ > > > > while (1) { > > > > if(wto == NULL) > > break; // redundancy -- already checking > > for this condition in MyCallback > > printf("infinite loop\n"); > > sleep(1); > > } > > > > err = snd_pcm_close(handle); > > if(err<0) > > printf("error in closing pcm device: > > %s\n",snd_strerror(err)); > > > > return 0; > > } > > > > void MyCallback(snd_async_handler_t *pcm_callback) > > { > > > > snd_pcm_t *pcm_handle = snd_async_handler_get_pcm(pcm_callback); > > snd_pcm_sframes_t avail; > > int count,a; > > snd_pcm_uframes_t period_size = 64; > > snd_pcm_sframes_t frames; > > > > if((avail=snd_pcm_avail_update(pcm_handle)) >= period_size) ^ From the alsa lib api docs. | | Using of this function is useless for the standard read/write operations. Use it only for mmap access. See to snd_pcm_delay. Obtain delay for a running PCM handle. Parameters: pcm PCM handle delayp Returned delay in frames Returns: 0 on success otherwise a negative error code Delay is distance between current application frame position and sound frame position. It's positive and less than buffer size in normal situation, negative on playback underrun and greater than buffer size on capture overrun. > > { > > printf("available frames = %d\n",avail); > > frames = snd_pcm_writei(pcm_handle, rfrom, SIZE); Using rfrom is going to be problematic if you get to a > > if(frames < 0) // underrun > > { > > printf("underrun recovery\n"); > > frames = snd_pcm_prepare(pcm_handle); > > printf("error from snd_pcm_prepare is: > > %d\n",frames); if(frames < 0){ > > printf("error in recovery\n"); > > > > } > > } > > > > if(frames >0 && frames < SIZE) > > printf("expected to write %d, wrote > > %d\n",SIZE,frames); > > > > > > if(rfrom < &buffer[buflen-1]) > > { > > wto = rfrom; > > rfrom = rfrom + SIZE; > > } > > if(rfrom >= &buffer[buflen-1]) > > { > > // wto = rfrom; > > rfrom = &buffer[0]; > > } //location of where to start > > writing from next > > > > if(writetobuf()<0){ > > wto=NULL; > > exit(0); // returns 0 to the operating system -- so > > it ends the program! > > } > > ++ind; > > } > > > > printf("going out of callback, ind = %d\n",ind); > > } > > > > /*Function to copy data from master buffer (mbuf) to circular buffer > > (buffer)*/ > > int writetobuf() > > { > > int count=-1; > > > > while((++count < SIZE) && (mpos < &mbuf[length])) > > *(mpos++) = *(wto+count); > > > > if(mpos >= &mbuf[length-1]) > > return -1; > > else > > return 0; > > } > > i=49 error in starting snd pcm start err :File descriptor in bad state infinite loop inside callback, available frames = 364 going out of callback, ind = 1 infinite loop inside callback, available frames = 577 going out of callback, ind = 2 infinite loop inside callback, available frames = 790 underrun recovery error from snd_pcm_prepare is: 0 going out of callback, ind = 3 infinite loop infinite loop infinite loop

1 0

Re: [alsa-devel] hardware not asking for more data using asyn call back
by Ashlesha Shintre 30 May '07

30 May '07

Hi, Thanks Stan, for your response - Instead of copying pcm data from a wav file, i decided to copy all the data to a master buffer first and then see if the circular buffer implementation works - so now instead of copying from the wave file, i m copying data from the master buffer to the circular buffer.. however, as per your suggestion, if i copy the data in the while loop in main, then, it might not always be copied between 2 consecutive callbacks, but maybe more, as the callbacks are asynchronous. however, the hardware still does not ask for more data after executing the callback function about twice -- is there a way to flush the hardware buffer before beginning playback? I have pasted my code below Regards, Ashlesha. /* Reading wav file into a buffer and then trying to play it */ > > #include <stdio.h> > #include </usr/include/alsa/asoundlib.h> > #include </usr/include/alsa/pcm.h> > #include <sys/types.h> > #include <unistd.h> > #include <fcntl.h> > > #define SIZE 128 > //mpos is the position in the master buffer to write from > short int *buffer, *mbuf, *mpos; > int ind=10, fd1, length; > static snd_pcm_t *handle; > static char *device = "default"; /* playback device > */ > snd_output_t *output = NULL; > static short int *wto, *rfrom, buflen; > > void MyCallback(snd_async_handler_t *pcm_callback); > > int main() > { > int i,j,n,count,a,err,k=0; > char c; > snd_pcm_sframes_t frames; > snd_async_handler_t *pcm_callback; > > > > if((fd1=open("SA2.WAV",O_RDONLY,0))==-1) > printf("error opening wav file\n"); > > count=0; > while(count++<40) > a=read(fd1,&c,sizeof(char)); > > a=read(fd1,&length,sizeof(int)); > > n=length/SIZE; > > printf("length = %d\n",length); > mbuf = (short int *) malloc (length * sizeof(short int)); > mpos = &mbuf[0]; > > buflen = 4*SIZE; > buffer = (short int *) malloc (buflen*sizeof(short int)); > > wto = NULL; > rfrom = &buffer[0]; > > /* reading to master buffer */ > count=0; > a=1; > while(count<length && a>0) > a=read(fd1,&mbuf[count++],sizeof(short int)); > if(a<0){ > printf("error in reading from wav file\n"); > exit(1); > } > close (fd1); > > for(count =0; count < buflen; count++) > { > buffer[count]=mbuf[count]; > } > /******************************/ > > /*ALSA DEV INIT*/ > if((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < > 0) { > printf("error opening device: %s\n",snd_strerror(err)); > return -1; > } > > if((err = snd_pcm_set_params(handle, > SND_PCM_FORMAT_S16_LE, > SND_PCM_ACCESS_RW_INTERLEAVED, > 1, > 16000, > 1, > 1000)) < 0) { /* 2 sec */ > printf("Playback open error: %s\n", snd_strerror(err)); > return -2; > } > /*****************************/ > > printf("n=%d\n",n); > for (i=0; i<50; i++) > { > printf("i=%d\n",i); > frames = snd_pcm_writei(handle, rfrom, SIZE); > if(frames < 0) // underrun > { > printf("underrun recovery\n"); > frames = snd_pcm_recover(handle, frames,0); > if(frames < 0){ > printf("error in recovery\n"); > return -3; > } > } > if(frames >0 && frames < SIZE) > printf("expected to write %d, wrote %d\n",SIZE,frames); > > printf("distance between ptrs before reinit is %d\n",rfrom - wto); > if(rfrom < &buffer[buflen-1]) > { > wto = rfrom; > rfrom = rfrom + SIZE; > } > if(rfrom >= &buffer[buflen-1]) > { > printf("rfm at the end, rfrom -buffer[buflen-1] > =%d\n",(*rfrom - buffer[buflen-1])); > //wto = rfrom; > rfrom = &buffer[0]; > } > > if((a = writetobuf())<0) > printf("error in reading from wav file k = %d\n"); > > printf("buffer[i*SIZE] - rfrom = %d\n",(buffer[(i+1)*SIZE] - > *rfrom)); > > } > > /* Async Handler */ > > > err = snd_async_add_pcm_handler(&pcm_callback,handle,MyCallback,NULL); > if(err<0) > printf("add pcm handler error = %d\n%s\n",err,snd_strerror(err)); > > err = snd_pcm_start(handle); > if(err<0) > printf("error in starting snd pcm start err > :%s\n",snd_strerror(err)); > /******************************/ > > while (1) { > > if(wto == NULL) > break; // redundancy -- already checking for this > condition in MyCallback > printf("infinite loop\n"); > sleep(1); > } > > err = snd_pcm_close(handle); > if(err<0) > printf("error in closing pcm device: %s\n",snd_strerror(err)); > > return 0; > } > > void MyCallback(snd_async_handler_t *pcm_callback) > { > > snd_pcm_t *pcm_handle = snd_async_handler_get_pcm(pcm_callback); > snd_pcm_sframes_t avail; > int count,a; > snd_pcm_uframes_t period_size = 64; > snd_pcm_sframes_t frames; > > if((avail=snd_pcm_avail_update(pcm_handle)) >= period_size) > { > printf("available frames = %d\n",avail); > frames = snd_pcm_writei(pcm_handle, rfrom, SIZE); > if(frames < 0) // underrun > { > printf("underrun recovery\n"); > frames = snd_pcm_prepare(pcm_handle); > printf("error from snd_pcm_prepare is: %d\n",frames); > if(frames < 0){ > printf("error in recovery\n"); > > } > } > > if(frames >0 && frames < SIZE) > printf("expected to write %d, wrote %d\n",SIZE,frames); > > > if(rfrom < &buffer[buflen-1]) > { > wto = rfrom; > rfrom = rfrom + SIZE; > } > if(rfrom >= &buffer[buflen-1]) > { > // wto = rfrom; > rfrom = &buffer[0]; > } //location of where to start writing > from next > > if(writetobuf()<0){ > wto=NULL; > exit(0); // returns 0 to the operating system -- so it > ends the program! > } > ++ind; > } > > printf("going out of callback, ind = %d\n",ind); > } > > /*Function to copy data from master buffer (mbuf) to circular buffer > (buffer)*/ > int writetobuf() > { > int count=-1; > > while((++count < SIZE) && (mpos < &mbuf[length])) > *(mpos++) = *(wto+count); > > if(mpos >= &mbuf[length-1]) > return -1; > else > return 0; > } >

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[alsa-devel] CMI8788............
by Khalid Paden 29 May '07

29 May '07

Does anyone have a driver for the CMI8788 chipset? AKA The X-Meridian? -- Khalid Paden _______________________________________________________________________ So that's it?!?!?! Not quite. Right now he is taking the safety's off all his weapons. He won't make the same mistake twice. -Hunt for Red October

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[alsa-devel] Reasonable sound hardware parameters?
by Lorenz Kolb 29 May '07

29 May '07

> Message: 4 > Date: Mon, 28 May 2007 10:52:18 +0200 (CEST) > From: Jaroslav Kysela <perex(a)suse.cz> > Subject: Re: [alsa-devel] Reasonable sound hardware parameters? > To: Joachim F?rster <mls.JOFT(a)gmx.de> > Cc: alsa-devel(a)alsa-project.org > Message-ID: <Pine.LNX.4.61.0705281045310.7894(a)tm8103.perex-int.cz> > Content-Type: text/plain; charset="iso8859-2" > > On Sat, 26 May 2007, Joachim F?rster wrote: > > > > > First question: To implement mmap on such kind of device IO memory, > > I have to use snd_pcm_lib_mmap_iomem() as mmap() callback - like it > > is used in the rme32 driver, right? > > Yes. rme32 is quite similar. > > > Second question: Do the following parameters make sense regarding > > the size of the HW ring buffer and period sizes: > > > > .buffer_bytes_max=16*1024; > > .period_bytes_max=8*1024; > > .period_bytes_min=?? resonable value ?? > > 32 or 64 > > > .periods_min=2; > > .periods_max=?? resonable value ?? > > 16 > > > Furthermore, the sound controller will fire an interrupt after one > > period is played. Are there any arguments against these values? Any > > comments? > > Yes, I think that the access through a PCI I/O window to a PCI device > is much slower than PCI DMA transfers with bursts. I would really > consider to design hardware with real DMA. Of course, if it's only > experimental stuff, then don't worry. > > Jaroslav > > ----- > Jaroslav Kysela <perex(a)suse.cz> > Linux Kernel Sound Maintainer > ALSA Project, SUSE Labs Hi Jaroslav, First of all: thanks for Your answer. Second: We do not have a PCI Bus. It is an embedded system with an OCP-Bus (aka OPB). The ML403 holds a SoC with PPC and FPGA. So the only difference between DMA with bursts and a dedicated memory (as far as we do see it) is the amount of time the bus gets occupied. With dedicated memory the system will (hopefully ;-)) do a burst write each period. So what happens in DMA case (except the more complex hardware as the DMA controller has to be written (in VHDL) as well and even more memory is located at an more complex bus (PLB - processor local bus): the bus will be occupied twice: (1) CPU writes data to memory. --> start DMA --> (2) Controller fetches data from memory. While (2) is in progress the poor CPU is "disconnected" from the bus. In an embedded system with a not too powerful CPU (PPC 405 @300MHz) this might be quite inefficient. Or did we miss something? Regards, Lorenz -- Lorenz Kolb <alsa-dev(a)lkmail.de> Embedded specialist ESIC-Solutions

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[alsa-devel] SigmaTel STAC9221 - Mux Amp-Out > 0x02 == microphone not working
by Ivan N. Zlatev 29 May '07

29 May '07

Hello, I am on an iMac ("*imac-20" model/pinconfig) with a Intel HD card with the SigmaTel STAC9221 codec. I have attached the inital codec info after modprobe. I am using Alsa HG. Setting the mux gain to a value > 20 (0x02 Amp-Out value) results a non-working microphone. Steps and codec diffs as follows: 1) initial - microphone working 2) Mux Gain = 10 - microphone working Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 - Amp-Out vals: [0x00 0x00] + Amp-Out vals: [0x01 0x01] Connection: 7 0x0e 0x15 0x0f 0x0b* 0x0c 0x0d 0x0a 3) Mux Gain = 20 - microphone working Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 - Amp-Out vals: [0x01 0x01] + Amp-Out vals: [0x02 0x02] Connection: 7 0x0e 0x15 0x0f 0x0b* 0x0c 0x0d 0x0a 4) Mux Gain = 30 - microphone not working Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 - Amp-Out vals: [0x02 0x02] + Amp-Out vals: [0x03 0x03] Connection: 7 0x0e 0x15 0x0f 0x0b* 0x0c 0x0d 0x0a 4) Mux Gain = 40 - microphone not working Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 - Amp-Out vals: [0x03 0x03] + Amp-Out vals: [0x04 0x04] Connection: 7 0x0e 0x15 0x0f 0x0b* 0x0c 0x0d 0x0a Regards -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

2 6

[alsa-devel] Intel HD SigmaTel STAC9221 - Wrong input after output change
by Ivan N. Zlatev 29 May '07

29 May '07

Hello, I am on an iMac ("*imac-20" model/pinconfig) with a Intel HD card with the SigmaTel STAC9221 codec. I have attached the inital codec info after modprobe. When I plug in and then unplug my speakers the microphone stops working. Reproduction steps with codec dump diffs as follows 1) Initial codec info - attached - Microphone working 2) Plug in speakers - Microphone working Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x9017e110: [Fixed] Speaker at Int N/A Conn = Analog, Color = White - Pin-ctls: 0x00: + Pin-ctls: 0x40: OUT Connection: 1 0x03 3) Unplug speakers - Microphone not working Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x90a7012e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown - Pin-ctls: 0x44: OUT + Pin-ctls: 0x04: Connection: 1 0x04 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x9017e110: [Fixed] Speaker at Int N/A Conn = Analog, Color = White - Pin-ctls: 0x40: OUT + Pin-ctls: 0x00: Connection: 1 0x03 Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Pin Default 0x0181e020: [Jack] Line In at Ext Rear Conn = 1/8, Color = White - Pin-ctls: 0x40: OUT + Pin-ctls: 0x00: Connection: 1 0x05 4) Plug in speakers again - Microphone not working Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x90a7012e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown - Pin-ctls: 0x04: + Pin-ctls: 0x44: OUT Connection: 1 0x04 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x9017e110: [Fixed] Speaker at Int N/A Conn = Analog, Color = White - Pin-ctls: 0x00: + Pin-ctls: 0x40: OUT Connection: 1 0x03 Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Pin Default 0x0181e020: [Jack] Line In at Ext Rear Conn = 1/8, Color = White - Pin-ctls: 0x00: + Pin-ctls: 0x40: OUT Connection: 1 0x05 -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

2 4

[alsa-devel] Question on snd_soc_dai_link
by Timur Tabi 29 May '07

29 May '07

I have a question about these DAI structs for ASoC. Let's take eti_b1_wm8731.c as an example. It contains this structure: static struct snd_soc_dai_link eti_b1_dai = { .name = "WM8731", .stream_name = "WM8731", .cpu_dai = &at91_i2s_dai[1], .codec_dai = &wm8731_dai, .init = eti_b1_wm8731_init, .ops = &eti_b1_ops, }; Notice that the .cpu_dai field is hard-coded to use the 2nd I2S device on the AT91. The problem I'm having with this approach is that everything is hard-coded. On PowerPC, this approach doesn't work well, because we have a "device tree" that dictates what devices are present on the SOC. For a Freescale SOC that has two I2S devices (for example), both devices would be listed in the device tree, with the base address of the memory-mapped registers. This means that a Freescale I2S driver would have its 'probe' function called (much like PCI does) each time the kernel processes an I2S node in the device tree. This means that when the machine driver (which has the snd_soc_dai_link structure) loads, there's no guarantee that the I2S driver has also loaded and the probe functions have been called. It gets even more complicated if only one I2S device has a codec connected to it, because then I need to kmalloc the snd_soc_cpu_dai structures and somehow tell the snd_soc_dai_link structure which one to use. Has any thought been given to making these structures more dynamic and handling interdependencies among the drivers? As it stands now, it looks like I'm going to have to hard-code everything in the drivers and break the device tree paradigm. -- Timur Tabi Linux Kernel Developer @ Freescale

2 5

[alsa-devel] [PATCH] Sigmatel STAC9221 - Add Configs for all Apple Intel Macs
by Ivan N. Zlatev 29 May '07

29 May '07

This patch: * adds the pinconfigs for all 5 Apple boards and 14 Subsystem IDs (support for possibly all iMac, Mac, MacMini etc etc) * adds "intel-mac-v1" to v5 models which replace the current * reflects changes in Alsa-Configuration.txt Please review. Regards. -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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Re: [alsa-devel] [PATCH] Disable debugging output for the ALS300 driver
by Ash Willis 29 May '07

29 May '07

> ----- Original Message ----- > From: "Takashi Iwai" <tiwai(a)suse.de> > To: "Ash Willis" <ashwillis(a)programmer.net> > Subject: Re: [alsa-devel] [PATCH] Disable debugging output for the ALS300 driver > Date: Tue, 29 May 2007 15:08:30 +0200 > > > At Sat, 26 May 2007 13:26:35 +0000, > Ash Willis wrote: > > > > Disables debugging output in the ALS300 driver. > > Also contains a whitespace cleanup and a fix for a potential bug. > > > > Signed-off-by: Ash Willis <ashwillis(a)programmer.net> > > Thanks for the patch. It seems that your MUA breaks the embedded > patch, so I had to apply it manually. Could you check whether the > latest HG is OK for you? I'm away from home right now. I'll test it as soon as I have a chance but it looks fine. > > Also I changed "if ((err = ...))" to two individual lines, which is a > newer standard kernel coding style. > Yeah, I left that for now. I'll go through the tree and tidy up remaining occurences of this style when I have some time. It seems that when people add drivers to the tree, they generally copy already existing drivers. It doesn't make much sense to be copying code that doesn't fit CodingStyle. Ash = Sports Memorabilia License Plates Signs All license plates $6 all signs $10 all frames $8 - same day shipping - fun company and selection. http://a8-asy.a8ww.net/a8-ads/adftrclick?redirectid=a44f49e42c5a283d05f0c96…

2 1

[alsa-devel] [PATCH] Disable debugging output for the ALS300 driver
by Ash Willis 29 May '07

29 May '07

Disables debugging output in the ALS300 driver. Also contains a whitespace cleanup and a fix for a potential bug. Signed-off-by: Ash Willis <ashwillis(a)programmer.net> ---------------------------------------------------- diff -r 337962e19de6 -r 8ea4fa014563 pci/als300.c --- a/pci/als300.c Fri May 25 12:18:49 2007 +0200 +++ b/pci/als300.c Sat May 26 14:04:14 2007 +0000 @@ -88,8 +88,8 @@ #define PLAYBACK_BLOCK_COUNTER 0x9A #define RECORD_BLOCK_COUNTER 0x9B -#define DEBUG_CALLS 1 -#define DEBUG_PLAY_REC 1 +#define DEBUG_CALLS 0 +#define DEBUG_PLAY_REC 0 #if DEBUG_CALLS #define snd_als300_dbgcalls(format, args...) printk(format, ##args) @@ -730,10 +730,9 @@ static int __devinit snd_als300_create(s } chip->irq = pci->irq; - snd_als300_init(chip); - if (snd_als300_ac97(chip) < 0) { + if ((err = snd_als300_ac97(chip)) < 0) { snd_printk(KERN_WARNING "Could not create ac97\n"); snd_als300_free(chip); return err;

2 1

[alsa-devel] sound on smdk2410
by Jose Henrique Spahn Torres 29 May '07

29 May '07

Hi, I am trying recent Zoltan Devai's patchs for kernel 2.6.22-rc2 on a smdk2410 board. I reached the point where the kernel recognized the uda1341 sound chip, however, when I try to play something, I get the output below. Can anyone, please, show me the path to follow to get sound working? Thanks, Henrique /home $ l drwxr-xr-x 3 root root 0 Jan 1 00:00 . drwxr-xr-x 18 root root 1024 May 18 2007 .. -r--r--r-- 1 root root 3419671 Jan 1 1970 jj.mp3 /home $ mpg123 jj.mp3 High Performance MPEG 1.0/2.0/2.5 Audio Player for Layers 1, 2 anasoc: UDA1341 <-> s3c24xx-i2s info: asoc: rate mask 0xfe asoc: min ch 2 max ch 2 asoc: min rate 8000 max rate 48000 version 0.65; written and copyright by Michael Hipp and others free software (LGPL/GPL) without any warranty but with best wishes asoc: machine hw_params failed ... (lots of asoc: messages) ... asoc: machine hw_params failed pop wq checking: Playback status: inactive waiting: yes pop wq D3 UDA1341 Playback Playing MPEG stream 1 of 1: jj.mp3 ... Title: Times Like These Artist: Jack Johnson Comment: Album: On And On Year: 2003 Genre: Acoustic MPEG 1.0 layer III, 192 kbits/s, 44100 Hz stereo [audio.c:264] error: No supported rate found! /home $

3 3

[alsa-devel] HDA Codec: Add support for MSI K9N Ultra
by Tobin Davis 29 May '07

29 May '07

This patch adds the MSI K9N Ultra system to the realtek patch. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] ICE1724 alsa driver
by Rossco 29 May '07

29 May '07

Hi, I hope someone can help me. (tried the user list but no response) I have a hitpoint 7.1ch Envy24ht-s based card with the usual vt1616 multichannel codec and the wolfson WM8728. The card is recognised as a chaintech av710 with the only difference being that the card has spdif in as well as out. It appears to be physically similar to the maddog entertainer (would be good to know if this is also recognised as an AV710) By default 2 channel stereo in played through the "front" channel of the card through the vt1616 codec and I appear to be able to set spdif out through the mixer (but dont have another device to test at the moment). I have been able to change default output to go through the wolfson dac through an asound.state file posted on the web. I have to admit I do not quite understand what the file has done as it only appears to have removed some of the controls rather than change or add anything. Only two devices are detected for the card the vt1616 and IEC958, neither are shown as having any subdevices, therefore I am unsure how the asound.state file was able to address the wolfson chip which requires i2s input rather ac-link. This seems to indicate i2s and ac-link modes of the envy24ht-s chip are active. Most AV710 users seem happy enough to be able to route to toslink out and/or the higher quality wolfson chip, therefore there is little discussion of addressing the other channels of the card. I really want to be able to address each of the channels of the card so that I can use the convolution programme Brutefir with the card to create a multiway crossover for my loudspeakers. Is there anyway I can define each of the cards outputs so that I can select to send audio to any one of them? I am also keen to get the spdif in working and particularly the syncing to spdif in of the clock that the ice1724 chip supports. Can these features be added to the driver? Please advise of any required info, I have all the datasheets and can trace chip pins also. Thanks in advance. Ross

2 5

[alsa-devel] [PATCH] Add quirk for Supermicro PDSBA to alc883_cfg_tbl[]
by Daniel T. Chen 29 May '07

29 May '07

Attached is a diff against current alsa-kernel hg tip. Thanks, Daniel Chen

1 1

[alsa-devel] hardware not asking for more data using asyn call back
by Ashlesha Shintre 29 May '07

29 May '07

Hi, In my asynchronous playback program, the hardware stops asking for any data and the program gets stuck after one call to the callback function. I have pasted the program below -- please let me know if something is wrong - Thanks, Ashlesha. /* Reading wav file into a buffer and then trying to play it */ > > #include <stdio.h> > #include </usr/include/alsa/asoundlib.h> > #include </usr/include/alsa/pcm.h> > #include <sys/types.h> > #include <unistd.h> > #include <fcntl.h> > > #define SIZE 128 > > short int *buffer; > int ind=10, fd1; > static snd_pcm_t *handle; > static char *device = "default"; /* playback device > */ > snd_output_t *output = NULL; > static short int *wto, *rfrom, buflen; > > void MyCallback(snd_async_handler_t *pcm_callback); > > int main() > { > int i,j,n,length,count,a,err,k=0; > char c; > snd_pcm_sframes_t frames; > snd_async_handler_t *pcm_callback; > > > > if((fd1=open("SA2.WAV",O_RDONLY,0))==-1) > printf("error opening wav file\n"); > > count=0; > while(count++<40) > a=read(fd1,&c,sizeof(char)); > > a=read(fd1,&length,sizeof(int)); > > n=length/SIZE; > > printf("length = %d\n",length); > > buflen = 4*SIZE; > buffer = (short int *) malloc (buflen*sizeof(short int)); > > wto = NULL; > rfrom = &buffer[0]; > > count=0; > a=1; > while(count<buflen && a>0) > a=read(fd1,&buffer[count++],sizeof(short int)); > if(a<0) > printf("error in reading from wav file\n"); > > if((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < > 0) { > printf("error opening device: %s\n",snd_strerror(err)); > return -1; > } > > if((err = snd_pcm_set_params(handle, > SND_PCM_FORMAT_S16_LE, > SND_PCM_ACCESS_RW_INTERLEAVED, > 1, > 16000, > 1, > 1000)) < 0) { /* 2 sec */ > printf("Playback open error: %s\n", snd_strerror(err)); > return -2; > } > printf("n=%d\n",n); > for (i=0; i<50; i++) > { > printf("i=%d\n",i); > frames = snd_pcm_writei(handle, rfrom, SIZE); > if(frames < 0) // underrun > { > printf("underrun recovery\n"); > frames = snd_pcm_recover(handle, frames,0); > if(frames < 0){ > printf("error in recovery\n"); > return -3; > } > } > if(frames >0 && frames < SIZE) > printf("expected to write %d, wrote %d\n",SIZE,frames); > printf("distance between ptrs before reinit is %d\n",rfrom - wto); > if(rfrom < &buffer[buflen-1]) > { > wto = rfrom; > rfrom = rfrom + SIZE; > } > else > { > printf("rfm at the end, rfrom -buffer[buflen-1] > =%d\n",(*rfrom - buffer[buflen-1])); > wto = rfrom; > rfrom = &buffer[0]; > } > for(k=0; k<SIZE; k++) > { > a=read(fd1,(wto+k),sizeof(short int)); > if(a<0) > printf("error in reading from wav file k = %d\n"); > } > printf("buffer[i*SIZE] - rfrom = %d\n",(buffer[(i+1)*SIZE] - > *rfrom)); > > } > > /* Async Handler */ > > // err = > snd_async_add_pcm_handler(&pcm_callback,handle,MyCallback,buffer); > err = snd_async_add_pcm_handler(&pcm_callback,handle,MyCallback,NULL); > if(err<0) > printf("add pcm handler error = %d\n%s\n",err,snd_strerror(err)); > > err = snd_pcm_start(handle); > if(err<0) > printf("error in starting snd pcm start err > :%s\n",snd_strerror(err)); > > while (1) { > if(wto == NULL) > break; > sleep(1); > } > > err = snd_pcm_close(handle); > if(err<0) > printf("error in closing pcm device: %s\n",snd_strerror(err)); > close(fd1); > return 0; > } > > void MyCallback(snd_async_handler_t *pcm_callback) > { > > snd_pcm_t *pcm_handle = snd_async_handler_get_pcm(pcm_callback); > snd_pcm_sframes_t avail; > int count,a; > snd_pcm_uframes_t period_size = 64; > snd_pcm_sframes_t frames; > //short int *bufferin = > snd_async_handler_get_callback_private(pcm_callback); > > while((avail=snd_pcm_avail_update(pcm_handle)) >= period_size) > { > printf("available frames = %d\n",avail); > frames = snd_pcm_writei(pcm_handle, rfrom, SIZE); > if(frames < 0) // underrun > { > printf("underrun recovery\n"); > frames = snd_pcm_prepare(pcm_handle); > printf("error from snd_pcm_prepare is: %d\n",frames); > if(frames < 0){ > printf("error in recovery\n"); > // break; > } > } > > if(frames >0 && frames < SIZE) > printf("expected to write %d, wrote %d\n",SIZE,frames); > count = 0; > a=1; > while(count<SIZE && a>0){ > a=read(fd1,(wto+count),sizeof(short int)); //write to > the buffer from file > ++count; > } > printf("count = %d\n",count); > if(a <0){ //EOF > wto = NULL; > printf("end of file reached\n"); > break; > } > else{ > if(rfrom < &buffer[buflen-1]) > { > wto = wto+count; > rfrom = rfrom + SIZE; > } > else > { > wto = wto + count; > rfrom = &buffer[0]; > } //location of where to start > writing from next > } > ++ind; > } > > printf("going out of callback, ind = %d\n",ind); > } > > >

2 1

[alsa-devel] async playback works despite error in snd_pcm_start.
by Ashlesha Shintre 28 May '07

28 May '07

Hi, I managed to write code to playback sound asynchronously -- however, this code works (i.e data is read from .wav file and output to the playback device) despite the snd_pcm_start giving a "file descriptor in bad state" error.. can someone explain this please? Thanks, Ashlesha.

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Re: [alsa-devel] another ALSA question
by Jaroslav Kysela 28 May '07

28 May '07

I'm CCing alsa-devel because this discussion might be interesting for other developers, too. On Mon, 28 May 2007, Maarten Lankhorst wrote: > Jaroslav Kysela schreef: > > On Mon, 28 May 2007, Maarten Lankhorst wrote: > > > >> After playing around a little more with alsa I tried to do manipulation > >> of pointer using the instructions, I wrote some simple code either does > >> rewind or forward to get to the right point, taking in respect the > >> amount of delay and where to go. > >> > >> However it doesn't seem to work: Looking at the source code it seems > >> rewinding isn't implemented, using the #if 0'd code there gives problems > >> when doing snd_pcm_forward later: It then ends up at the pointer where > >> the sound is currently being played at. > >> > > > > Not all alsa-lib plugins are capable or have implemented this fine > > position control (check error value). > > > For a plugin like dmix, I would expect it to be implemented, especially > since most cards sold are integrated cards that don't have multiple > sound buffers. Go ahead and send us patches. We're doing open source :-) Anyway any rewind operation mens "undo" - it will eat extra CPU ticks for nothing. > >> As workaround I used the following in my own code: > >> typeof(snd_pcm_forward) snd_pcm_mmap_appl_forward; > >> #define snd_pcm_forward snd_pcm_mmap_appl_forward > >> > >> typeof(snd_pcm_rewind) snd_pcm_mmap_appl_backward; > >> #define snd_pcm_rewind snd_pcm_mmap_appl_backward > >> > >> This works, but I wonder if there is a cleaner way to do it. > >> > > > > It won't work with all plugins and mentioned functions are not part of > > public API. > > > As I work in wine on it's DirectSound implementation, I can tell it > relies heavily on the fact that it can lock a buffer at an arbitrary > position, and that it works with absolute pointers, rather then > relative. So if snd_pcm_rewind/snd_pcm_forward don't work as expected, > an alsa driver for it would become nearly useless, I don't expect it to > always work, but I would expect it to work for a common plugin like dmix > at least. > >> Also, I noticed some audio glitches when using this in combination with > >> alsa-lib's resampling, is that because of the way it is done or is > >> something else wrong? > >> > > > > You should not use mentioned functions. > > > > You can try to use the direct hardware access (hw:CARD_NAME) devices > > (you'll lose all stream conversions from the ALSA side) or use an > > intermediate buffer for fine position control and configure ALSA to use > > smallest ring buffer as possible (to minimalize latency). > > > > Jarosla > I wish that would work, directsound expects to have access to a fixed > size buffer, and if I go for direct hardware access, I lose the ability > to work properly with most cheap cards on the market, since people want > to be able to listen to music on background while gaming. For now it > seems I have to use the internal calls in case of dmix, or modify > already committed data, which is what winealsa's current driver does. I > don't know what is the least of those 2 evils, or if there is a third > option. I'm open for advice. I'm afraid, that you're trying to exactly copy DirectSound API. Playing with the buffer position in this way is always dangerous, because it's really hard to predict the final result in the multitasking environment. Think about these proposals: 1) what sample (ring buffer) rewritting is supposed to do - maybe you can change only not yet commited samples with satisfactory results 2) use a small ALSA ring buffer and use own "big" ring buffer for changes before you commit samples to ALSA's ring buffer - you'll increase a bit sample flow latency, but you can do anything you want (you just add extra buffering to avoid rewind / forward for ALSA API) 3) fix (or better implement) dmix plugin rewind function Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] [ALSA] ALSA Power Management, Drivers behaving unexpectedly after suspend/resume cycle
by Nobin Mathew 28 May '07

28 May '07

I am implementing power management in my ALSA sound drivers. ALSA drivers are not working properly after resume. All registers values are proper and I am able to read back the codec register contents. If I do a playback after resume then I am getting lots of "underrun". After a reboot playback is working fine. Is there any known issues with ALSA power management? Why Ubuntu and Redhat does ALSA modules removal before suspend and insertion after resume?

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[alsa-devel] Reasonable sound hardware parameters?
by Joachim Förster 28 May '07

28 May '07

Hi ALSA devs, as I already described in one of my last posts on the list, I'm going to write a ALSA driver for a sound controller which will also be "written" at the same time (VHDL for FPGA) - on top of an embedded system (Xilinx ML403 EvalBoard, PowerPC 405). We will do playback feature only. The point is, that we will have _no_ real DMA and we don't want to use the copy/silence or ack/workqueue (with intermediate buffer). So we decided to move the ring buffer, which usually exists in RAM into the sound controller and handle it as some kind of device IO memory, which should be mmap'able (kernel function io_remap_pfn_range()). So, the important goal is, to have mmap feature in the ALSA driver. First question: To implement mmap on such kind of device IO memory, I have to use snd_pcm_lib_mmap_iomem() as mmap() callback - like it is used in the rme32 driver, right? Second question: Do the following parameters make sense regarding the size of the HW ring buffer and period sizes: .buffer_bytes_max=16*1024; .period_bytes_max=8*1024; .period_bytes_min=?? resonable value ?? .periods_min=2; .periods_max=?? resonable value ?? Furthermore, the sound controller will fire an interrupt after one period is played. Are there any arguments against these values? Any comments? Thanks, Joachim

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[alsa-devel] alsa playback channel interleaving question
by Matthias Nyffenegger 27 May '07

27 May '07

Hi, Earlier this year I wrote to this list: > I am trying to write an alsa driver for my audiowerk8 soundcard. > The aw8 uses a Philips saa7146 chipset. There are 2 capture and > 8 playback channels mapped on the saa7146's DMA as follows: > > DMA1: 2xcapture > DMA2: 4xplayback > DMA3: 4xplayback > > Sample format is 16bit. > > Frame format is as follows: > (A number corresponds to a channel, a stereo channel-pair is embraced > by [] and 1 digit corresponds to one sample byte) > > capture channels [0,1] on DMA1: 00--11-- > playback channels [2,3],[4,5] on DMA2: 22443355 > playback channels [6,7],[8,9] on DMA3: 66887799 I tried to find a way to handle this in the card, but this special interleaving for the playback channels is by hardware restriction and can't be worked around in the soundcard itself. AFAIK alsa provides the samples interleaved like 22334455 (66778899). To avoid channel switching I guess some special processing in the alsa driver is required. Where is this done best? Do I have to move around the samples in the interrupt handler or does alsa provide some (possibly better) means to achive the correct (for the soundcard) interleaving? Regards, Matthias

1 0

[alsa-devel] Suggestion for getting alsa to use Conexant
by Victor Schneider 26 May '07

26 May '07

How about being able to parameterize Alsa modules so that the intel 80xm module finds any hda modem that also acts as an Intel hda chip? I am referring to the Conexant combination modem and hda chip that gets identified as "Conexant Waikiki" by alsamixer, but that 80xm can't find as a modem. Giving the 80xm module the device:identifier value to override its detection of the Intel version of the Conexant hda modem might let slmodemd work.

3 5

Re: [alsa-devel] [RFC] [PATCH 0/2] ASoC sound support for SMDK2440boards
by Zoltan Devai 25 May '07

25 May '07

2007/5/24, Jose Henrique Spahn Torres <jose.torres(a)eldorado.org.br>: > I have a SMDK2410 and I am trying to run kernel 2.6.20 on it. The kernel > is running Ok, but the sound is dead. I gonna try your patch and I would > like to know which files from the RK's patch have you used? The attached two patches should enable the sound support fully. One of them adds the l3-bus, the other one the necessary Kconfig and Makefile changes, along with an ASoC hack. Add this patch also: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-May/001105.html Everything should apply cleanly to 2.6.22-rc2, but probably also to 2.6.21. Zoltan

2 3

[alsa-devel] Sound on MSI S420
by Baruch Even 25 May '07

25 May '07

Hello, I've succeeded by guessing to get my MSI S420 laptop sound to work. Attached is a one-liner patch to do that. The patch is against kernel 2.6.22-rc2. Since the patch is based on guesswork it might not be optimal but it does work for me. Baruch

2 1

Re: [alsa-devel] [PATCH] Add support for Cyrix/NatSemi Geode CS5530 (VSA1)
by Ash Willis 25 May '07

25 May '07

> Thanks. Now committed to HG tree. > > One more possible fix is about the build of snd-sb16-dsp and dependent > modules. We don't need snd-sb16 itself. > > Could you try the patch below (both for alsa-driver and alsa-kernel)? > It introduces the intermediate configs for snd-sb16-dsp, snd-sb8-dsp > and snd-sb-common modules. > Yeah, that makes more sense...and it works fine. Thanks, Ash =

2 1

[alsa-devel] Testing ALSA and OSS drivers
by Pharaoh . 24 May '07

24 May '07

Hi I am planning to write an ALSA driver for an OMAP based board, I already have an OSS driver written for my board, but it is a 2.4 kernel. Question: I am trying to bring up Linux for my board (the board is supported), how can I test whether the oss drivers are working fine or not? I just have to check whether operations like playback and recording etc work as expected. Any pointers will be helpful. Regards, Pharaoh.

2 1

[alsa-devel] [RFC] [PATCH 2/2] ASoC sound support for SMDK2440 boards: board driver
by Zoltan Devai 24 May '07

24 May '07

--- /dev/null +++ /sound/soc/s3c24xx/smdk2440.c @@ -0,0 +1,162 @@ +/* + * smdk2440.c -- SMDK2440 ALSA SoC Audio board driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * 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/l3.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <asm/arch/regs-iis.h> + +#include "../codecs/uda1341.h" +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" + +static int smdk2440_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + unsigned int div = 0; + int ret = 0; + + switch (params_rate(params)) { + case 11025: + div = 16; + clk = 4233600; + break; + case 22050: + div = 8; + clk = 8467200; + break; + case 44100: + div = 4; + clk = 16934400; + break; + default: + return -EINVAL; + } + + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK , clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_384FS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div,div)); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static const char* smdk2440_l3_link_name = "s3c24xx-l3"; + +static struct snd_soc_ops smdk2440_ops = { + .hw_params = smdk2440_hw_params, +}; + +static struct snd_soc_dai_link smdk2440_dai_link = { + .name = "UDA1341", + .stream_name = "UDA1341", + .codec_dai = &uda1341_dai, + .cpu_dai = &s3c24xx_i2s_dai, + .ops = &smdk2440_ops, +}; + +static struct snd_soc_machine snd_soc_machine_smdk2440 = { + .name = "SMDK2440", + .dai_link = &smdk2440_dai_link, + .num_links = 1, +}; + +static struct snd_soc_device smdk2440_snd_devdata = { + .machine = &snd_soc_machine_smdk2440, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_uda1341, + .codec_data = &smdk2440_l3_link_name, +}; + +static struct platform_device *smdk2440_snd_device; + +static int __init smdk2440_uda1341_init(void) +{ + int ret; + + printk(KERN_INFO "SMDK2440 ALSA SoC Audio driver\n"); + + if (!(machine_is_s3c2440())) + return -ENODEV; + + smdk2440_snd_devdata.codec_data = l3_get_adapter("s3c24xx-l3"); + if (smdk2440_snd_devdata.codec_data == NULL) { + printk(KERN_WARNING "SMDK2440 SoC Audio: \ + Unable to find L3 bus adapter\n"); + return -ENODEV; + } + + smdk2440_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk2440_snd_device) { + printk(KERN_ERR "SMDK2440 SoC Audio: Unable to register\n"); + return -ENOMEM; + } + + platform_set_drvdata(smdk2440_snd_device, &smdk2440_snd_devdata); + smdk2440_snd_devdata.dev = &smdk2440_snd_device->dev; + ret = platform_device_add(smdk2440_snd_device); + + if (ret) + platform_device_put(smdk2440_snd_device); + + return ret; +} + +static void __exit smdk2440_uda1341_exit(void) +{ + platform_device_unregister(smdk2440_snd_device); +} + +module_init(smdk2440_uda1341_init); +module_exit(smdk2440_uda1341_exit); + +MODULE_AUTHOR("Zoltan Devai"); +MODULE_DESCRIPTION("SMDK2440 ALSA SoC audio driver"); +MODULE_LICENSE("GPL");

2 1

[alsa-devel] Default Sample rate Conversion
by Prabhu Sivaraja 24 May '07

24 May '07

Does ALSA automatically convert to my devices sample rate? My alsa.conf file is below pcm.mix { type dmix ipc_key 1024 slave { pcm "hw:0,0" rate 48000 } } pcm.!default { type plug slave.pcm "mix" } When I play a 32KHz file with aplay does it get up sampled to my device's 48KHz sample rate? I am assuming it does as it plays just fine but I am not sure where the upsampling happens? Thanks.

2 1

Re: [alsa-devel] [PATCH] Add support for Cyrix/NatSemi Geode CS5530 (VSA1)
by Ash Willis 24 May '07

24 May '07

> > Takashi, here are my modified patches. > > One small one for alsa-driver. > > I took your advice and merged the rest into alsa-kernel. > > Everything should be in order this time. > > > > Add support for Cyrix/NatSemi Geode SC5530 (VSA1) > > > > Signed-off-by Ash Willis <ashwillis(a)programmer.net> > > Thanks. One thing I overlooked at the previous review is the kconfig > dependency below: Ok, hopefully we're finished now :) Here are the two patches. Ash --- diff -r fb391138d29c -r 994136cf989e pci/cs5530.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/cs5530.c Thu May 24 20:45:25 2007 +0100 @@ -0,0 +1,2 @@ +#include "../alsa-kernel/pci/cs5530.c" +EXPORT_NO_SYMBOLS; --- diff -r 45179b325c8e -r 9ca89d69569c Documentation/ALSA-Configuration.txt --- a/Documentation/ALSA-Configuration.txt Wed May 23 16:27:32 2007 +0200 +++ b/Documentation/ALSA-Configuration.txt Thu May 24 23:16:59 2007 +0100 @@ -467,7 +467,12 @@ Prior to version 0.9.0rc4 options had a above explicitly. The power-management is supported. - + + Module snd-cs5530 + _________________ + + Module for Cyrix/NatSemi Geode 5530 chip. + Module snd-cs5535audio ---------------------- diff -r 45179b325c8e -r 9ca89d69569c include/sb.h --- a/include/sb.h Wed May 23 16:27:32 2007 +0200 +++ b/include/sb.h Thu May 24 23:16:59 2007 +0100 @@ -38,6 +38,7 @@ enum sb_hw_type { SB_HW_ALS100, /* Avance Logic ALS100 chip */ SB_HW_ALS4000, /* Avance Logic ALS4000 chip */ SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */ + SB_HW_CS5530, /* Cyrix/NatSemi 5530 VSA1 */ }; #define SB_OPEN_PCM 0x01 diff -r 45179b325c8e -r 9ca89d69569c isa/sb/sb16_main.c --- a/isa/sb/sb16_main.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb16_main.c Thu May 24 23:16:59 2007 +0100 @@ -563,6 +563,11 @@ static int snd_sb16_playback_open(struct __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->playback_substream = substream; @@ -633,6 +638,11 @@ static int snd_sb16_capture_open(struct __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->capture_substream = substream; diff -r 45179b325c8e -r 9ca89d69569c isa/sb/sb_common.c --- a/isa/sb/sb_common.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb_common.c Thu May 24 23:16:59 2007 +0100 @@ -128,7 +128,7 @@ static int snd_sbdsp_probe(struct snd_sb minor = version & 0xff; snd_printdd("SB [0x%lx]: DSP chip found, version = %i.%i\n", chip->port, major, minor); - + switch (chip->hardware) { case SB_HW_AUTO: switch (major) { @@ -167,6 +167,9 @@ static int snd_sbdsp_probe(struct snd_sb break; case SB_HW_DT019X: str = "(DT019X/ALS007)"; + break; + case SB_HW_CS5530: + str = "16 (CS5530)"; break; default: return -ENODEV; diff -r 45179b325c8e -r 9ca89d69569c isa/sb/sb_mixer.c --- a/isa/sb/sb_mixer.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb_mixer.c Thu May 24 23:16:59 2007 +0100 @@ -821,6 +821,7 @@ int snd_sbmixer_new(struct snd_sb *chip) break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: if ((err = snd_sbmixer_init(chip, snd_sb16_controls, ARRAY_SIZE(snd_sb16_controls), @@ -950,6 +951,7 @@ void snd_sbmixer_suspend(struct snd_sb * break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: save_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: @@ -975,6 +977,7 @@ void snd_sbmixer_resume(struct snd_sb *c break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: restore_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: diff -r 45179b325c8e -r 9ca89d69569c pci/Kconfig --- a/pci/Kconfig Wed May 23 16:27:32 2007 +0200 +++ b/pci/Kconfig Thu May 24 23:16:59 2007 +0100 @@ -214,6 +214,16 @@ config SND_CS46XX_NEW_DSP Say Y here to use a new DSP image for SPDIF and dual codecs. This works better than the old code, so say Y. + +config SND_CS5530 + tristate "CS5530 Audio" + depends on SND && ISA_DMA_API + select SND_SB16 + help + Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-cs5530. config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" diff -r 45179b325c8e -r 9ca89d69569c pci/Makefile --- a/pci/Makefile Wed May 23 16:27:32 2007 +0200 +++ b/pci/Makefile Thu May 24 23:16:59 2007 +0100 @@ -12,6 +12,7 @@ snd-bt87x-objs := bt87x.o snd-bt87x-objs := bt87x.o snd-cmipci-objs := cmipci.o snd-cs4281-objs := cs4281.o +snd-cs5530-objs := cs5530.o snd-ens1370-objs := ens1370.o snd-ens1371-objs := ens1371.o snd-es1938-objs := es1938.o @@ -36,6 +37,7 @@ obj-$(CONFIG_SND_BT87X) += snd-bt87x.o obj-$(CONFIG_SND_BT87X) += snd-bt87x.o obj-$(CONFIG_SND_CMIPCI) += snd-cmipci.o obj-$(CONFIG_SND_CS4281) += snd-cs4281.o +obj-$(CONFIG_SND_CS5530) += snd-cs5530.o obj-$(CONFIG_SND_ENS1370) += snd-ens1370.o obj-$(CONFIG_SND_ENS1371) += snd-ens1371.o obj-$(CONFIG_SND_ES1938) += snd-es1938.o diff -r 45179b325c8e -r 9ca89d69569c pci/cs5530.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/cs5530.c Thu May 24 23:16:59 2007 +0100 @@ -0,0 +1,306 @@ +/* + * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio + * + * (C) Copyright 2007 Ash Willis <ashwillis(a)programmer.net> + * (C) Copyright 2003 Red Hat Inc <alan(a)redhat.com> + * + * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did + * mess with it a bit. The chip seems to have to have trouble with full duplex + * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to + * simultaneously play back audio at 16bit 44100kHz, the device actually plays + * back in the same format in which it is capturing. By forcing the chip to + * always play/capture in 16/44100, we can let alsa-lib convert the samples and + * that way we can hack up some full duplex audio. + * + * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. + * The older version (VSA1) provides fairly good soundblaster emulation + * although there are a couple of bugs: large DMA buffers break record, + * and the MPU event handling seems suspect. VSA2 allows the native driver + * to control the AC97 audio engine directly and requires a different driver. + * + * Thanks to National Semiconductor for providing the needed information + * on the XpressAudio(tm) internals. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2, or (at your option) any + * later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * TO DO: + * Investigate whether we can portably support Cognac (5520) in the + * same manner. + */ + +#include <sound/driver.h> +#include <linux/delay.h> +#include <linux/moduleparam.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Ash Willis"); +MODULE_DESCRIPTION("CS5530 Audio"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +struct snd_cs5530 { + struct snd_card *card; + struct pci_dev *pci; + struct snd_sb *sb; + unsigned long pci_base; +}; + +static struct pci_device_id snd_cs5530_ids[] = { + {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, + PCI_ANY_ID, 0, 0}, + {0,} +}; + +MODULE_DEVICE_TABLE(pci, snd_cs5530_ids); + +static int snd_cs5530_free(struct snd_cs5530 *chip) +{ + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return 0; +} + +static int snd_cs5530_dev_free(struct snd_device *device) +{ + struct snd_cs5530 *chip = device->device_data; + return snd_cs5530_free(chip); +} + +static void __devexit snd_cs5530_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg) +{ + outb(reg, io + 4); + udelay(20); + reg = inb(io + 5); + udelay(20); + return reg; +} + +static int __devinit snd_cs5530_create(struct snd_card *card, + struct pci_dev *pci, + struct snd_cs5530 **rchip) +{ + struct snd_cs5530 *chip; + unsigned long sb_base; + u8 irq, dma8, dma16 = 0; + u16 map; + void __iomem *mem; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_cs5530_dev_free, + }; + *rchip = NULL; + + err = pci_enable_device(pci); + if (err < 0) + return err; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + chip->pci = pci; + + err = pci_request_regions(pci, "CS5530"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->pci_base = pci_resource_start(pci, 0); + + mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + if (mem == NULL) { + kfree(chip); + pci_disable_device(pci); + return -EBUSY; + } + + map = readw(mem + 0x18); + iounmap(mem); + + /* Map bits + 0:1 * 0x20 + 0x200 = sb base + 2 sb enable + 3 adlib enable + 5 MPU enable 0x330 + 6 MPU enable 0x300 + + The other bits may be used internally so must be masked */ + + sb_base = 0x220 + 0x20 * (map & 3); + + if (map & (1<<2)) + printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base); + else { + printk(KERN_ERR "Could not find XpressAudio!\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (map & (1<<5)) + printk(KERN_INFO "CS5530: MPU at 0x300\n"); + else if (map & (1<<6)) + printk(KERN_INFO "CS5530: MPU at 0x330\n"); + + irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F; + dma8 = snd_cs5530_mixer_read(sb_base, 0x81); + + if (dma8 & 0x20) + dma16 = 5; + else if (dma8 & 0x40) + dma16 = 6; + else if (dma8 & 0x80) + dma16 = 7; + else { + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (dma8 & 0x01) + dma8 = 0; + else if (dma8 & 02) + dma8 = 1; + else if (dma8 & 0x08) + dma8 = 3; + else { + printk(KERN_ERR "CS5530: No 8bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (irq & 1) + irq = 9; + else if (irq & 2) + irq = 5; + else if (irq & 4) + irq = 7; + else if (irq & 8) + irq = 10; + else { + printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, + dma16); + + err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, dma8, + dma16, SB_HW_CS5530, &chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create SoundBlaster\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create PCM\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sbmixer_new(chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create Mixer\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_cs5530_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + return 0; +} + +static int __devinit snd_cs5530_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct snd_cs5530 *chip = NULL; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + + if (card == NULL) + return -ENOMEM; + + err = snd_cs5530_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "CS5530"); + strcpy(card->shortname, "CS5530 Audio"); + sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + pci_set_drvdata(pci, card); + dev++; + return 0; +} + +static struct pci_driver driver = { + .name = "CS5530_Audio", + .id_table = snd_cs5530_ids, + .probe = snd_cs5530_probe, + .remove = __devexit_p(snd_cs5530_remove), +}; + +static int __init alsa_card_cs5530_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit alsa_card_cs5530_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_cs5530_init) +module_exit(alsa_card_cs5530_exit) + =

2 1

[alsa-devel] [RFC] [PATCH 0/2] ASoC sound support for SMDK2440 boards
by Zoltan Devai 24 May '07

24 May '07

Hi, The following patches are the first working version of drivers for the UDA1341 codec and SMDK2440 boards. Any comments for improvement are welcome. I also would like to ask for help in the following topics: - The codec requires an L3-bus driver. I'm using one made by RMK, but it's not mainline, and probably needs adoption to the driver model. (http://lkml.org/lkml/2005/10/28/194) It could also make more sense to integrate the driver into the i2c or spi subsystems, or just make it a S3C24xx specific gpio hack. Any volunteers ? - I only have access to a SMDK2440 board. Could someone verify if this would work on other SMDK's as well ? Regards, Zoltan

1 0

[alsa-devel] [PATCH] Add support for Cyrix/NatSemi Geode CS5530 (VSA1)
by Ash Willis 24 May '07

24 May '07

Takashi, here are my modified patches. One small one for alsa-driver. I took your advice and merged the rest into alsa-kernel. Everything should be in order this time. Add support for Cyrix/NatSemi Geode SC5530 (VSA1) Signed-off-by Ash Willis <ashwillis(a)programmer.net> --- diff -r fb391138d29c -r 994136cf989e pci/cs5530.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/cs5530.c Thu May 24 20:45:25 2007 +0100 @@ -0,0 +1,2 @@ +#include "../alsa-kernel/pci/cs5530.c" +EXPORT_NO_SYMBOLS; --- diff -r 45179b325c8e -r 0b0c03c6c6ca Documentation/ALSA-Configuration.txt --- a/Documentation/ALSA-Configuration.txt Wed May 23 16:27:32 2007 +0200 +++ b/Documentation/ALSA-Configuration.txt Thu May 24 20:58:27 2007 +0100 @@ -467,7 +467,12 @@ Prior to version 0.9.0rc4 options had a above explicitly. The power-management is supported. - + + Module snd-cs5530 + _________________ + + Module for Cyrix/NatSemi Geode 5530 chip. + Module snd-cs5535audio ---------------------- diff -r 45179b325c8e -r 0b0c03c6c6ca include/sb.h --- a/include/sb.h Wed May 23 16:27:32 2007 +0200 +++ b/include/sb.h Thu May 24 20:58:27 2007 +0100 @@ -38,6 +38,7 @@ enum sb_hw_type { SB_HW_ALS100, /* Avance Logic ALS100 chip */ SB_HW_ALS4000, /* Avance Logic ALS4000 chip */ SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */ + SB_HW_CS5530, /* Cyrix/NatSemi 5530 VSA1 */ }; #define SB_OPEN_PCM 0x01 diff -r 45179b325c8e -r 0b0c03c6c6ca isa/sb/sb16_main.c --- a/isa/sb/sb16_main.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb16_main.c Thu May 24 20:58:27 2007 +0100 @@ -563,6 +563,11 @@ static int snd_sb16_playback_open(struct __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->playback_substream = substream; @@ -633,6 +638,11 @@ static int snd_sb16_capture_open(struct __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->capture_substream = substream; diff -r 45179b325c8e -r 0b0c03c6c6ca isa/sb/sb_common.c --- a/isa/sb/sb_common.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb_common.c Thu May 24 20:58:27 2007 +0100 @@ -128,7 +128,7 @@ static int snd_sbdsp_probe(struct snd_sb minor = version & 0xff; snd_printdd("SB [0x%lx]: DSP chip found, version = %i.%i\n", chip->port, major, minor); - + switch (chip->hardware) { case SB_HW_AUTO: switch (major) { @@ -167,6 +167,9 @@ static int snd_sbdsp_probe(struct snd_sb break; case SB_HW_DT019X: str = "(DT019X/ALS007)"; + break; + case SB_HW_CS5530: + str = "16 (CS5530)"; break; default: return -ENODEV; diff -r 45179b325c8e -r 0b0c03c6c6ca isa/sb/sb_mixer.c --- a/isa/sb/sb_mixer.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb_mixer.c Thu May 24 20:58:27 2007 +0100 @@ -821,6 +821,7 @@ int snd_sbmixer_new(struct snd_sb *chip) break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: if ((err = snd_sbmixer_init(chip, snd_sb16_controls, ARRAY_SIZE(snd_sb16_controls), @@ -950,6 +951,7 @@ void snd_sbmixer_suspend(struct snd_sb * break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: save_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: @@ -975,6 +977,7 @@ void snd_sbmixer_resume(struct snd_sb *c break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: restore_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: diff -r 45179b325c8e -r 0b0c03c6c6ca pci/Kconfig --- a/pci/Kconfig Wed May 23 16:27:32 2007 +0200 +++ b/pci/Kconfig Thu May 24 20:58:27 2007 +0100 @@ -214,6 +214,16 @@ config SND_CS46XX_NEW_DSP Say Y here to use a new DSP image for SPDIF and dual codecs. This works better than the old code, so say Y. + +config SND_CS5530 + tristate "CS5530 Audio" + depends on SND + select SND_SB16 + help + Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-cs5530. config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" diff -r 45179b325c8e -r 0b0c03c6c6ca pci/Makefile --- a/pci/Makefile Wed May 23 16:27:32 2007 +0200 +++ b/pci/Makefile Thu May 24 20:58:27 2007 +0100 @@ -12,6 +12,7 @@ snd-bt87x-objs := bt87x.o snd-bt87x-objs := bt87x.o snd-cmipci-objs := cmipci.o snd-cs4281-objs := cs4281.o +snd-cs5530-objs := cs5530.o snd-ens1370-objs := ens1370.o snd-ens1371-objs := ens1371.o snd-es1938-objs := es1938.o @@ -36,6 +37,7 @@ obj-$(CONFIG_SND_BT87X) += snd-bt87x.o obj-$(CONFIG_SND_BT87X) += snd-bt87x.o obj-$(CONFIG_SND_CMIPCI) += snd-cmipci.o obj-$(CONFIG_SND_CS4281) += snd-cs4281.o +obj-$(CONFIG_SND_CS5530) += snd-cs5530.o obj-$(CONFIG_SND_ENS1370) += snd-ens1370.o obj-$(CONFIG_SND_ENS1371) += snd-ens1371.o obj-$(CONFIG_SND_ES1938) += snd-es1938.o diff -r 45179b325c8e -r 0b0c03c6c6ca pci/cs5530.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/cs5530.c Thu May 24 20:58:27 2007 +0100 @@ -0,0 +1,306 @@ +/* + * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio + * + * (C) Copyright 2007 Ash Willis <ashwillis(a)programmer.net> + * (C) Copyright 2003 Red Hat Inc <alan(a)redhat.com> + * + * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did + * mess with it a bit. The chip seems to have to have trouble with full duplex + * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to + * simultaneously play back audio at 16bit 44100kHz, the device actually plays + * back in the same format in which it is capturing. By forcing the chip to + * always play/capture in 16/44100, we can let alsa-lib convert the samples and + * that way we can hack up some full duplex audio. + * + * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. + * The older version (VSA1) provides fairly good soundblaster emulation + * although there are a couple of bugs: large DMA buffers break record, + * and the MPU event handling seems suspect. VSA2 allows the native driver + * to control the AC97 audio engine directly and requires a different driver. + * + * Thanks to National Semiconductor for providing the needed information + * on the XpressAudio(tm) internals. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2, or (at your option) any + * later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * TO DO: + * Investigate whether we can portably support Cognac (5520) in the + * same manner. + */ + +#include <sound/driver.h> +#include <linux/delay.h> +#include <linux/moduleparam.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Ash Willis"); +MODULE_DESCRIPTION("CS5530 Audio"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +struct snd_cs5530 { + struct snd_card *card; + struct pci_dev *pci; + struct snd_sb *sb; + unsigned long pci_base; +}; + +static struct pci_device_id snd_cs5530_ids[] = { + {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, + PCI_ANY_ID, 0, 0}, + {0,} +}; + +MODULE_DEVICE_TABLE(pci, snd_cs5530_ids); + +static int snd_cs5530_free(struct snd_cs5530 *chip) +{ + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return 0; +} + +static int snd_cs5530_dev_free(struct snd_device *device) +{ + struct snd_cs5530 *chip = device->device_data; + return snd_cs5530_free(chip); +} + +static void __devexit snd_cs5530_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg) +{ + outb(reg, io + 4); + udelay(20); + reg = inb(io + 5); + udelay(20); + return reg; +} + +static int __devinit snd_cs5530_create(struct snd_card *card, + struct pci_dev *pci, + struct snd_cs5530 **rchip) +{ + struct snd_cs5530 *chip; + unsigned long sb_base; + u8 irq, dma8, dma16 = 0; + u16 map; + void __iomem *mem; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_cs5530_dev_free, + }; + *rchip = NULL; + + err = pci_enable_device(pci); + if (err < 0) + return err; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + chip->pci = pci; + + err = pci_request_regions(pci, "CS5530"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->pci_base = pci_resource_start(pci, 0); + + mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + if (mem == NULL) { + kfree(chip); + pci_disable_device(pci); + return -EBUSY; + } + + map = readw(mem + 0x18); + iounmap(mem); + + /* Map bits + 0:1 * 0x20 + 0x200 = sb base + 2 sb enable + 3 adlib enable + 5 MPU enable 0x330 + 6 MPU enable 0x300 + + The other bits may be used internally so must be masked */ + + sb_base = 0x220 + 0x20 * (map & 3); + + if (map & (1<<2)) + printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base); + else { + printk(KERN_ERR "Could not find XpressAudio!\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (map & (1<<5)) + printk(KERN_INFO "CS5530: MPU at 0x300\n"); + else if (map & (1<<6)) + printk(KERN_INFO "CS5530: MPU at 0x330\n"); + + irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F; + dma8 = snd_cs5530_mixer_read(sb_base, 0x81); + + if (dma8 & 0x20) + dma16 = 5; + else if (dma8 & 0x40) + dma16 = 6; + else if (dma8 & 0x80) + dma16 = 7; + else { + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (dma8 & 0x01) + dma8 = 0; + else if (dma8 & 02) + dma8 = 1; + else if (dma8 & 0x08) + dma8 = 3; + else { + printk(KERN_ERR "CS5530: No 8bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (irq & 1) + irq = 9; + else if (irq & 2) + irq = 5; + else if (irq & 4) + irq = 7; + else if (irq & 8) + irq = 10; + else { + printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, + dma16); + + err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, dma8, + dma16, SB_HW_CS5530, &chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create SoundBlaster\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create PCM\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sbmixer_new(chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create Mixer\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_cs5530_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + return 0; +} + +static int __devinit snd_cs5530_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct snd_cs5530 *chip = NULL; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + + if (card == NULL) + return -ENOMEM; + + err = snd_cs5530_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "CS5530"); + strcpy(card->shortname, "CS5530 Audio"); + sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + pci_set_drvdata(pci, card); + dev++; + return 0; +} + +static struct pci_driver driver = { + .name = "CS5530_Audio", + .id_table = snd_cs5530_ids, + .probe = snd_cs5530_probe, + .remove = __devexit_p(snd_cs5530_remove), +}; + +static int __init alsa_card_cs5530_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit alsa_card_cs5530_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_cs5530_init) +module_exit(alsa_card_cs5530_exit) + =

2 1

Re: [alsa-devel] [PATCH] Add support for Cyrix/NatSemi Geode CS5530 (VSA1)
by Ash Willis 24 May '07

24 May '07

I'm sorry. I forgot to mention that these patches replace the two previous patches :) Ash > ----- Original Message ----- > From: "Ash Willis" <ashwillis(a)programmer.net> > To: "Takashi Iwai" <tiwai(a)suse.de> > Subject: [alsa-devel] [PATCH] Add support for Cyrix/NatSemi Geode CS5530 (VSA1) > Date: Thu, 24 May 2007 09:09:29 -0500 > > > Takashi, here are my modified patches. > One small one for alsa-driver. > I took your advice and merged the rest into alsa-kernel. > Everything should be in order this time. > > Add support for Cyrix/NatSemi Geode SC5530 (VSA1) > > Signed-off-by Ash Willis <ashwillis(a)programmer.net> > --- > > diff -r fb391138d29c -r 994136cf989e pci/cs5530.c > --- /dev/null Thu Jan 01 00:00:00 1970 +0000 > +++ b/pci/cs5530.c Thu May 24 20:45:25 2007 +0100 > @@ -0,0 +1,2 @@ > +#include "../alsa-kernel/pci/cs5530.c" > +EXPORT_NO_SYMBOLS; > > > --- > > diff -r 45179b325c8e -r 0b0c03c6c6ca Documentation/ALSA-Configuration.txt > --- a/Documentation/ALSA-Configuration.txt Wed May 23 16:27:32 2007 +0200 > +++ b/Documentation/ALSA-Configuration.txt Thu May 24 20:58:27 2007 +0100 > @@ -467,7 +467,12 @@ Prior to version 0.9.0rc4 options had a > above explicitly. > > The power-management is supported. > - > + > + Module snd-cs5530 > + _________________ > + > + Module for Cyrix/NatSemi Geode 5530 chip. > + > Module snd-cs5535audio > ---------------------- > > diff -r 45179b325c8e -r 0b0c03c6c6ca include/sb.h > --- a/include/sb.h Wed May 23 16:27:32 2007 +0200 > +++ b/include/sb.h Thu May 24 20:58:27 2007 +0100 > @@ -38,6 +38,7 @@ enum sb_hw_type { > SB_HW_ALS100, /* Avance Logic ALS100 chip */ > SB_HW_ALS4000, /* Avance Logic ALS4000 chip */ > SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */ > + SB_HW_CS5530, /* Cyrix/NatSemi 5530 VSA1 */ > }; > > #define SB_OPEN_PCM 0x01 > diff -r 45179b325c8e -r 0b0c03c6c6ca isa/sb/sb16_main.c > --- a/isa/sb/sb16_main.c Wed May 23 16:27:32 2007 +0200 > +++ b/isa/sb/sb16_main.c Thu May 24 20:58:27 2007 +0100 > @@ -563,6 +563,11 @@ static int snd_sb16_playback_open(struct > __open_ok: > if (chip->hardware == SB_HW_ALS100) > runtime->hw.rate_max = 48000; > + if (chip->hardware == SB_HW_CS5530) { > + runtime->hw.buffer_bytes_max = 32 * 1024; > + runtime->hw.periods_min = 2; > + runtime->hw.rate_min = 44100; > + } > if (chip->mode & SB_RATE_LOCK) > runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; > chip->playback_substream = substream; > @@ -633,6 +638,11 @@ static int snd_sb16_capture_open(struct > __open_ok: > if (chip->hardware == SB_HW_ALS100) > runtime->hw.rate_max = 48000; > + if (chip->hardware == SB_HW_CS5530) { > + runtime->hw.buffer_bytes_max = 32 * 1024; > + runtime->hw.periods_min = 2; > + runtime->hw.rate_min = 44100; > + } > if (chip->mode & SB_RATE_LOCK) > runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; > chip->capture_substream = substream; > diff -r 45179b325c8e -r 0b0c03c6c6ca isa/sb/sb_common.c > --- a/isa/sb/sb_common.c Wed May 23 16:27:32 2007 +0200 > +++ b/isa/sb/sb_common.c Thu May 24 20:58:27 2007 +0100 > @@ -128,7 +128,7 @@ static int snd_sbdsp_probe(struct snd_sb > minor = version & 0xff; > snd_printdd("SB [0x%lx]: DSP chip found, version = %i.%i\n", > chip->port, major, minor); > - > + > switch (chip->hardware) { > case SB_HW_AUTO: > switch (major) { > @@ -167,6 +167,9 @@ static int snd_sbdsp_probe(struct snd_sb > break; > case SB_HW_DT019X: > str = "(DT019X/ALS007)"; > + break; > + case SB_HW_CS5530: > + str = "16 (CS5530)"; > break; > default: > return -ENODEV; > diff -r 45179b325c8e -r 0b0c03c6c6ca isa/sb/sb_mixer.c > --- a/isa/sb/sb_mixer.c Wed May 23 16:27:32 2007 +0200 > +++ b/isa/sb/sb_mixer.c Thu May 24 20:58:27 2007 +0100 > @@ -821,6 +821,7 @@ int snd_sbmixer_new(struct snd_sb *chip) > break; > case SB_HW_16: > case SB_HW_ALS100: > + case SB_HW_CS5530: > if ((err = snd_sbmixer_init(chip, > snd_sb16_controls, > ARRAY_SIZE(snd_sb16_controls), > @@ -950,6 +951,7 @@ void snd_sbmixer_suspend(struct snd_sb * > break; > case SB_HW_16: > case SB_HW_ALS100: > + case SB_HW_CS5530: > save_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); > break; > case SB_HW_ALS4000: > @@ -975,6 +977,7 @@ void snd_sbmixer_resume(struct snd_sb *c > break; > case SB_HW_16: > case SB_HW_ALS100: > + case SB_HW_CS5530: > restore_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); > break; > case SB_HW_ALS4000: > diff -r 45179b325c8e -r 0b0c03c6c6ca pci/Kconfig > --- a/pci/Kconfig Wed May 23 16:27:32 2007 +0200 > +++ b/pci/Kconfig Thu May 24 20:58:27 2007 +0100 > @@ -214,6 +214,16 @@ config SND_CS46XX_NEW_DSP > Say Y here to use a new DSP image for SPDIF and dual codecs. > > This works better than the old code, so say Y. > + > +config SND_CS5530 > + tristate "CS5530 Audio" > + depends on SND > + select SND_SB16 > + help > + Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. > + > + To compile this driver as a module, choose M here: the module > + will be called snd-cs5530. > > config SND_CS5535AUDIO > tristate "CS5535/CS5536 Audio" > diff -r 45179b325c8e -r 0b0c03c6c6ca pci/Makefile > --- a/pci/Makefile Wed May 23 16:27:32 2007 +0200 > +++ b/pci/Makefile Thu May 24 20:58:27 2007 +0100 > @@ -12,6 +12,7 @@ snd-bt87x-objs := bt87x.o > snd-bt87x-objs := bt87x.o > snd-cmipci-objs := cmipci.o > snd-cs4281-objs := cs4281.o > +snd-cs5530-objs := cs5530.o > snd-ens1370-objs := ens1370.o > snd-ens1371-objs := ens1371.o > snd-es1938-objs := es1938.o > @@ -36,6 +37,7 @@ obj-$(CONFIG_SND_BT87X) += snd-bt87x.o > obj-$(CONFIG_SND_BT87X) += snd-bt87x.o > obj-$(CONFIG_SND_CMIPCI) += snd-cmipci.o > obj-$(CONFIG_SND_CS4281) += snd-cs4281.o > +obj-$(CONFIG_SND_CS5530) += snd-cs5530.o > obj-$(CONFIG_SND_ENS1370) += snd-ens1370.o > obj-$(CONFIG_SND_ENS1371) += snd-ens1371.o > obj-$(CONFIG_SND_ES1938) += snd-es1938.o > diff -r 45179b325c8e -r 0b0c03c6c6ca pci/cs5530.c > --- /dev/null Thu Jan 01 00:00:00 1970 +0000 > +++ b/pci/cs5530.c Thu May 24 20:58:27 2007 +0100 > @@ -0,0 +1,306 @@ > +/* > + * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio > + * > + * (C) Copyright 2007 Ash Willis <ashwillis(a)programmer.net> > + * (C) Copyright 2003 Red Hat Inc <alan(a)redhat.com> > + * > + * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did > + * mess with it a bit. The chip seems to have to have trouble with > full duplex > + * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to > + * simultaneously play back audio at 16bit 44100kHz, the device > actually plays > + * back in the same format in which it is capturing. By forcing the chip to > + * always play/capture in 16/44100, we can let alsa-lib convert > the samples and > + * that way we can hack up some full duplex audio. > + * > + * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. > + * The older version (VSA1) provides fairly good soundblaster emulation > + * although there are a couple of bugs: large DMA buffers break record, > + * and the MPU event handling seems suspect. VSA2 allows the native driver > + * to control the AC97 audio engine directly and requires a different driver. > + * > + * Thanks to National Semiconductor for providing the needed information > + * on the XpressAudio(tm) internals. > + * > + * This program is free software; you can redistribute it and/or modify it > + * under the terms of the GNU General Public License as published by the > + * Free Software Foundation; either version 2, or (at your option) any > + * later version. > + * > + * This program is distributed in the hope that it will be useful, but > + * WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * General Public License for more details. > + * > + * TO DO: > + * Investigate whether we can portably support Cognac (5520) in the > + * same manner. > + */ > + > +#include <sound/driver.h> > +#include <linux/delay.h> > +#include <linux/moduleparam.h> > +#include <linux/pci.h> > +#include <sound/core.h> > +#include <sound/sb.h> > +#include <sound/initval.h> > + > +MODULE_AUTHOR("Ash Willis"); > +MODULE_DESCRIPTION("CS5530 Audio"); > +MODULE_LICENSE("GPL"); > + > +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > + > +struct snd_cs5530 { > + struct snd_card *card; > + struct pci_dev *pci; > + struct snd_sb *sb; > + unsigned long pci_base; > +}; > + > +static struct pci_device_id snd_cs5530_ids[] = { > + {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, > + PCI_ANY_ID, 0, 0}, > + {0,} > +}; > + > +MODULE_DEVICE_TABLE(pci, snd_cs5530_ids); > + > +static int snd_cs5530_free(struct snd_cs5530 *chip) > +{ > + pci_release_regions(chip->pci); > + pci_disable_device(chip->pci); > + kfree(chip); > + return 0; > +} > + > +static int snd_cs5530_dev_free(struct snd_device *device) > +{ > + struct snd_cs5530 *chip = device->device_data; > + return snd_cs5530_free(chip); > +} > + > +static void __devexit snd_cs5530_remove(struct pci_dev *pci) > +{ > + snd_card_free(pci_get_drvdata(pci)); > + pci_set_drvdata(pci, NULL); > +} > + > +static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg) > +{ > + outb(reg, io + 4); > + udelay(20); > + reg = inb(io + 5); > + udelay(20); > + return reg; > +} > + > +static int __devinit snd_cs5530_create(struct snd_card *card, > + struct pci_dev *pci, > + struct snd_cs5530 **rchip) > +{ > + struct snd_cs5530 *chip; > + unsigned long sb_base; > + u8 irq, dma8, dma16 = 0; > + u16 map; > + void __iomem *mem; > + int err; > + > + static struct snd_device_ops ops = { > + .dev_free = snd_cs5530_dev_free, > + }; > + *rchip = NULL; > + > + err = pci_enable_device(pci); > + if (err < 0) > + return err; > + > + chip = kzalloc(sizeof(*chip), GFP_KERNEL); > + if (chip == NULL) { > + pci_disable_device(pci); > + return -ENOMEM; > + } > + > + chip->card = card; > + chip->pci = pci; > + > + err = pci_request_regions(pci, "CS5530"); > + if (err < 0) { > + kfree(chip); > + pci_disable_device(pci); > + return err; > + } > + chip->pci_base = pci_resource_start(pci, 0); > + > + mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); > + if (mem == NULL) { > + kfree(chip); > + pci_disable_device(pci); > + return -EBUSY; > + } > + > + map = readw(mem + 0x18); > + iounmap(mem); > + > + /* Map bits > + 0:1 * 0x20 + 0x200 = sb base > + 2 sb enable > + 3 adlib enable > + 5 MPU enable 0x330 > + 6 MPU enable 0x300 > + > + The other bits may be used internally so must be masked */ > + > + sb_base = 0x220 + 0x20 * (map & 3); > + > + if (map & (1<<2)) > + printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base); > + else { > + printk(KERN_ERR "Could not find XpressAudio!\n"); > + snd_cs5530_free(chip); > + return -ENODEV; > + } > + > + if (map & (1<<5)) > + printk(KERN_INFO "CS5530: MPU at 0x300\n"); > + else if (map & (1<<6)) > + printk(KERN_INFO "CS5530: MPU at 0x330\n"); > + > + irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F; > + dma8 = snd_cs5530_mixer_read(sb_base, 0x81); > + > + if (dma8 & 0x20) > + dma16 = 5; > + else if (dma8 & 0x40) > + dma16 = 6; > + else if (dma8 & 0x80) > + dma16 = 7; > + else { > + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); > + snd_cs5530_free(chip); > + return -ENODEV; > + } > + > + if (dma8 & 0x01) > + dma8 = 0; > + else if (dma8 & 02) > + dma8 = 1; > + else if (dma8 & 0x08) > + dma8 = 3; > + else { > + printk(KERN_ERR "CS5530: No 8bit DMA enabled\n"); > + snd_cs5530_free(chip); > + return -ENODEV; > + } > + > + if (irq & 1) > + irq = 9; > + else if (irq & 2) > + irq = 5; > + else if (irq & 4) > + irq = 7; > + else if (irq & 8) > + irq = 10; > + else { > + printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n"); > + snd_cs5530_free(chip); > + return -ENODEV; > + } > + > + printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, > + dma16); > + > + err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, dma8, > + dma16, SB_HW_CS5530, &chip->sb); > + if (err < 0) { > + printk(KERN_ERR "CS5530: Could not create SoundBlaster\n"); > + snd_cs5530_free(chip); > + return err; > + } > + > + err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm); > + if (err < 0) { > + printk(KERN_ERR "CS5530: Could not create PCM\n"); > + snd_cs5530_free(chip); > + return err; > + } > + > + err = snd_sbmixer_new(chip->sb); > + if (err < 0) { > + printk(KERN_ERR "CS5530: Could not create Mixer\n"); > + snd_cs5530_free(chip); > + return err; > + } > + > + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); > + if (err < 0) { > + snd_cs5530_free(chip); > + return err; > + } > + > + snd_card_set_dev(card, &pci->dev); > + *rchip = chip; > + return 0; > +} > + > +static int __devinit snd_cs5530_probe(struct pci_dev *pci, > + const struct pci_device_id *pci_id) > +{ > + static int dev; > + struct snd_card *card; > + struct snd_cs5530 *chip = NULL; > + int err; > + > + if (dev >= SNDRV_CARDS) > + return -ENODEV; > + if (!enable[dev]) { > + dev++; > + return -ENOENT; > + } > + > + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); > + > + if (card == NULL) > + return -ENOMEM; > + > + err = snd_cs5530_create(card, pci, &chip); > + if (err < 0) { > + snd_card_free(card); > + return err; > + } > + > + strcpy(card->driver, "CS5530"); > + strcpy(card->shortname, "CS5530 Audio"); > + sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base); > + > + err = snd_card_register(card); > + if (err < 0) { > + snd_card_free(card); > + return err; > + } > + pci_set_drvdata(pci, card); > + dev++; > + return 0; > +} > + > +static struct pci_driver driver = { > + .name = "CS5530_Audio", > + .id_table = snd_cs5530_ids, > + .probe = snd_cs5530_probe, > + .remove = __devexit_p(snd_cs5530_remove), > +}; > + > +static int __init alsa_card_cs5530_init(void) > +{ > + return pci_register_driver(&driver); > +} > + > +static void __exit alsa_card_cs5530_exit(void) > +{ > + pci_unregister_driver(&driver); > +} > + > +module_init(alsa_card_cs5530_init) > +module_exit(alsa_card_cs5530_exit) > + > > > > = > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > =

1 0

[alsa-devel] [1/3] 2.6.22-rc2: known regressions v2
by Michal Piotrowski 24 May '07

24 May '07

Hi all, Here is a list of some known regressions in 2.6.22-rc2. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Unclassified Subject : lag problems while browsing sites with heavy JS and or flash usage References : http://lkml.org/lkml/2007/5/19/189 Submitter : Anant Nitya <kernel(a)prachanda.info> Handled-By : Ingo Molnar <mingo(a)elte.hu> David Miller <davem(a)davemloft.net> Status : problem is being debugged ACPI Subject : unable to shutdown on kernel 2.6.22-rc2 References : http://bugzilla.kernel.org/show_bug.cgi?id=8516 Submitter : Thierry Volpiatto <tvolpiatt(a)neuf.fr> Status : Unknown Subject : BAT0 no longer seen, ACPI errors References : http://bugzilla.kernel.org/show_bug.cgi?id=8493 Submitter : Ben Collins <bcollins(a)ubuntu.com> Workaround : BIOS upgrade Status : Unknown Subject : nx6125 has lost fan control References : http://lkml.org/lkml/2007/5/16/249 Submitter : Ray Lee <ray-lk(a)madrabbit.org> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Status : Unknown Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

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Re: [alsa-devel] [PATCH 2/2] Add support for Cyrix/NatSemi Geode SC5530 (VSA1)
by Ash Willis 24 May '07

24 May '07

> Avoid the style "if ((err = xx) < 0)" in new codes as much as > possible. Just write: > err = pci_enable(pci); > if (err < 0) > return err; > Now the kernel coding style recommends this. Sorry, I wasn't aware this had changed :) > > + if (dma8 & 0x20) > > + dma16 = 5; > > + else if (dma8 & 0x40) > > + dma16 = 6; > > + else if (dma8 & 0x80) > > + dma16 = 7; > > + else > > + { > > + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); > > + snd_cs5530_free(chip); > > + return -ENODEV; > > + } > > Ditto. > (BTW, is dma16 really necessary to be zero-initialized at beginning?) > It is, I think. It supresses a compiler warning, due the to fact that the final else path does not set dma16 and so if this path is taken the variable is uninitialied...not that we care, but we certainly don't need a compiler warning. > > Is the driver confirmed to work with the recent kernel? > If yes, we can put cs5530.c to alsa-kernel so that it'll be merged to > the next linux kernel. alsa-driver tree still needs cs5530.c just > including alsa-kernel code, though. > Also, don't forget to add a description to ALSA-Configuration.txt. > > Could you fix these issues and resend the patch? It was devleoped on 2.6.21.1. OK, I'll make all the necessary changes. Thanks, Ash =

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[alsa-devel] [PATCH 2/2] Add support for Cyrix/NatSemi Geode SC5530 (VSA1)
by Ash Willis 24 May '07

24 May '07

Add support for the Cyrix/NatSemi Geode SC5530 chips. This driver is ported from oss/kahlua.c and is made up of two patches. Patch 1/2 adds some stuff to the SB layer to allow for some hardware quirks and is to be applied to alsa-kernel. Patch 2/2 adds the actual driver code to be applied to alsa-driver. Patch 2 depends on patch 1. Signed off by: Ash Willis <ashwillis(a)programmer.net> --- summary: Add Cyrix/NatSemi Geode CS5530 driver diff -r 5c25d0e8819c -r c9be7b8611e7 pci/Kconfig --- a/pci/Kconfig Fri May 18 17:56:53 2007 +0200 +++ b/pci/Kconfig Thu May 24 02:24:57 2007 +0100 @@ -20,5 +20,15 @@ config SND_ASIHPI help Say 'Y' or 'M' to include support for AudioScience ASIxxxx soundcards. +config SND_CS5530 + tristate "CS5530 Audio" + depends on SND + select SND_SB16 + help + Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-cs5530. + endmenu diff -r 5c25d0e8819c -r c9be7b8611e7 pci/Makefile --- a/pci/Makefile Fri May 18 17:56:53 2007 +0200 +++ b/pci/Makefile Thu May 24 02:24:57 2007 +0100 @@ -13,6 +13,9 @@ clean-files := ad1889.c bt87x.c intel8x0 obj-$(CONFIG_SND) += pdplus/ asihpi/ +snd-cs5530-objs := cs5530.o +obj-$(CONFIG_SND_CS5530) += snd-cs5530.o + # # # diff -r 5c25d0e8819c -r c9be7b8611e7 pci/cs5530.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/cs5530.c Thu May 24 02:24:57 2007 +0100 @@ -0,0 +1,296 @@ +/* + * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio + * + * (C) Copyright 2007 Ash Willis <ashwillis(a)programmer.net> + * (C) Copyright 2003 Red Hat Inc <alan(a)redhat.com> + * + * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did + * mess with it a bit. The chip seems to have to have trouble with full duplex + * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to + * simultaneously play back audio at 16bit 44100kHz, the device actually plays + * back in the same format in which it is capturing. By forcing the chip to + * always play/capture in 16/44100, we can let alsa-lib convert the samples and + * that way we can hack up some full duplex audio. + * + * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. + * The older version (VSA1) provides fairly good soundblaster emulation + * although there are a couple of bugs: large DMA buffers break record, + * and the MPU event handling seems suspect. VSA2 allows the native driver + * to control the AC97 audio engine directly and requires a different driver. + * + * Thanks to National Semiconductor for providing the needed information + * on the XpressAudio(tm) internals. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2, or (at your option) any + * later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * TO DO: + * Investigate whether we can portably support Cognac (5520) in the + * same manner. + */ + +#include <sound/driver.h> +#include <linux/delay.h> +#include <linux/moduleparam.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Ash Willis"); +MODULE_DESCRIPTION("CS5530 Audio"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +struct snd_cs5530 { + struct snd_card *card; + struct pci_dev *pci; + struct snd_sb *sb; + unsigned long pci_base; +}; + +static struct pci_device_id snd_cs5530_ids[] = { + {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, + PCI_ANY_ID, 0, 0}, + {0,} +}; + +MODULE_DEVICE_TABLE(pci, snd_cs5530_ids); + +static int snd_cs5530_free(struct snd_cs5530 *chip) +{ + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return 0; +} + +static int snd_cs5530_dev_free(struct snd_device *device) +{ + struct snd_cs5530 *chip = device->device_data; + return snd_cs5530_free(chip); +} + +static void __devexit snd_cs5530_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg) +{ + outb(reg, io + 4); + udelay(20); + reg = inb(io + 5); + udelay(20); + return reg; +} + +static int __devinit snd_cs5530_create(struct snd_card *card, + struct pci_dev *pci, + struct snd_cs5530 **rchip) +{ + struct snd_cs5530 *chip; + unsigned long sb_base; + u8 irq, dma8, dma16 = 0; + u16 map; + void __iomem *mem; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_cs5530_dev_free, + }; + *rchip = NULL; + + if ((err = pci_enable_device(pci)) < 0) + return err; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + chip->pci = pci; + + if ((err = pci_request_regions(pci, "CS5530")) < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->pci_base = pci_resource_start(pci, 0); + + mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + map = readw(mem + 0x18); + iounmap(mem); + + /* Map bits + 0:1 * 0x20 + 0x200 = sb base + 2 sb enable + 3 adlib enable + 5 MPU enable 0x330 + 6 MPU enable 0x300 + + The other bits may be used internally so must be masked */ + + sb_base = 0x220 + 0x20 * (map & 3); + + if (map & (1<<2)) + printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base); + else + { + printk(KERN_ERR "Could not find XpressAudio!\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (map & (1<<5)) + printk(KERN_INFO "CS5530: MPU at 0x300\n"); + else if (map & (1<<6)) + printk(KERN_INFO "CS5530: MPU at 0x330\n"); + + irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F; + dma8 = snd_cs5530_mixer_read(sb_base, 0x81); + + if (dma8 & 0x20) + dma16 = 5; + else if (dma8 & 0x40) + dma16 = 6; + else if (dma8 & 0x80) + dma16 = 7; + else + { + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (dma8 & 0x01) + dma8 = 0; + else if (dma8 & 02) + dma8 = 1; + else if (dma8 & 0x08) + dma8 = 3; + else + { + printk(KERN_ERR "CS5530: No 8bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (irq & 1) + irq = 9; + else if (irq & 2) + irq = 5; + else if (irq & 4) + irq = 7; + else if (irq & 8) + irq = 10; + else + { + printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, + dma16); + + if ((err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, + dma8, dma16, SB_HW_CS5530, &chip->sb)) < 0) { + printk(KERN_ERR "CS5530: Could not create SoundBlaster\n"); + snd_cs5530_free(chip); + return err; + } + + if ((err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm)) < 0) { + printk(KERN_ERR "CS5530: Could not create PCM\n"); + snd_cs5530_free(chip); + return err; + } + + if ((err = snd_sbmixer_new(chip->sb)) < 0) { + printk(KERN_ERR "CS5530: Could not create Mixer\n"); + snd_cs5530_free(chip); + return err; + } + + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { + snd_cs5530_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + return 0; +} + +static int __devinit snd_cs5530_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct snd_cs5530 *chip = NULL; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + + if (card == NULL) + return -ENOMEM; + + if ((err = snd_cs5530_create(card, pci, &chip)) < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "CS5530"); + strcpy(card->shortname, "CS5530 Audio"); + sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base); + + if ((err = snd_card_register(card)) < 0) { + snd_card_free(card); + return err; + } + pci_set_drvdata(pci, card); + dev++; + return 0; +} + +static struct pci_driver driver = { + .name = "CS5530_Audio", + .id_table = snd_cs5530_ids, + .probe = snd_cs5530_probe, + .remove = __devexit_p(snd_cs5530_remove), +}; + +static int __init alsa_card_cs5530_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit alsa_card_cs5530_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_cs5530_init) +module_exit(alsa_card_cs5530_exit) + = Work From Home & Learn How To Make Money Learn How To Make Money Online With Help & Guidance From The Pro's. http://a8-asy.a8ww.net/a8-ads/adftrclick?redirectid=db055100ca96c7d9527d2fa…

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[alsa-devel] [patch] bugfix and clean driver/pci/asihpi/hpi6205.[ch]
by Eliot Blennerhassett 24 May '07

24 May '07

Attached patch Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Patch makes power of 2 buffering unconditional. and fixes incorrect buffer allocation when requested size was rounded up to a power of 2. -- Eliot

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[alsa-devel] Any standard ALSA application for simultaneous capture and playabck ?????
by Nobin Mathew 24 May '07

24 May '07

Hi, Is there any standard ALSA appplication which does simultaneous capture and playback?? What are the complications in implementing simultaneous playback and capture in ALSA??? What are the plugin (.asoundrc) configurations needed for simultaneous playback and capture???? Thanks Nobin Mathew

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Re: [alsa-devel] support for Medion laptops
by albox＠free.fr 24 May '07

24 May '07

I haven't get any news since my last email 2 weeks ago. Can anybody help me to find where the problem comes from ? Quoting albox(a)free.fr: > I have just installed 1.0.14rc4. Here are the logs : > > [ 20.132000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: > codec_mask = 0x3 > [ 20.308000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:1820: > hda_codec: model 'medion' is selected for config 161f:2054 (Medion laptop) > [ 21.032000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:245: > si3054: cannot initialize. EXT MID = 0000 > [ 21.040000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:257: > Link Frame Detect(FDT) is not ready (line status: 0000) > [ 71.352000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > [ 71.352000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 > [ 71.360000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 > [ 71.368000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 > [ 71.380000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 > [ 71.480000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 > [ 71.488000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 > [ 71.496000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 > [ 71.504000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 > [ 71.524000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > [ 71.524000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 > [ 71.532000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 > [ 71.540000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 > [ 71.548000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 > [ 71.560000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 > [ 71.568000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 > [ 71.576000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 > [ 71.584000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 > [ 71.752000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > [ 71.752000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 > [ 71.760000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 > [ 71.768000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 > [ 71.776000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 > > If you want more information, let me know...

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[alsa-devel] [PATCH 1/2] Add SB_HW_CS5530 to the SB code
by Ash Willis 23 May '07

23 May '07

Add support for the Cyrix/NatSemi Geode SC5530 chips. This driver is ported from oss/kahlua.c and is made up of two patches. Patch 1/2 adds some stuff to the SB layer to allow for some hardware quirks and is to be applied to alsa-kernel. Patch 2/2 adds the actual driver code to be applied to alsa-driver. Patch 2 depends on patch 1. Signed off by: Ash Willis <ashwillis(a)programmer.net> --- summary: Add SB_HW_CS5530 to the SB code diff -r 45179b325c8e -r b8441e56e853 include/sb.h --- a/include/sb.h Wed May 23 16:27:32 2007 +0200 +++ b/include/sb.h Thu May 24 03:09:49 2007 +0100 @@ -38,6 +38,7 @@ enum sb_hw_type { SB_HW_ALS100, /* Avance Logic ALS100 chip */ SB_HW_ALS4000, /* Avance Logic ALS4000 chip */ SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */ + SB_HW_CS5530, /* Cyrix/NatSemi 5530 VSA1 */ }; #define SB_OPEN_PCM 0x01 diff -r 45179b325c8e -r b8441e56e853 isa/sb/sb16_main.c --- a/isa/sb/sb16_main.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb16_main.c Thu May 24 03:09:49 2007 +0100 @@ -563,6 +563,11 @@ static int snd_sb16_playback_open(struct __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->playback_substream = substream; @@ -633,6 +638,11 @@ static int snd_sb16_capture_open(struct __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->capture_substream = substream; diff -r 45179b325c8e -r b8441e56e853 isa/sb/sb_common.c --- a/isa/sb/sb_common.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb_common.c Thu May 24 03:09:49 2007 +0100 @@ -128,7 +128,7 @@ static int snd_sbdsp_probe(struct snd_sb minor = version & 0xff; snd_printdd("SB [0x%lx]: DSP chip found, version = %i.%i\n", chip->port, major, minor); - + switch (chip->hardware) { case SB_HW_AUTO: switch (major) { @@ -167,6 +167,9 @@ static int snd_sbdsp_probe(struct snd_sb break; case SB_HW_DT019X: str = "(DT019X/ALS007)"; + break; + case SB_HW_CS5530: + str = "16 (CS5530)"; break; default: return -ENODEV; diff -r 45179b325c8e -r b8441e56e853 isa/sb/sb_mixer.c --- a/isa/sb/sb_mixer.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb_mixer.c Thu May 24 03:09:49 2007 +0100 @@ -821,6 +821,7 @@ int snd_sbmixer_new(struct snd_sb *chip) break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: if ((err = snd_sbmixer_init(chip, snd_sb16_controls, ARRAY_SIZE(snd_sb16_controls), @@ -950,6 +951,7 @@ void snd_sbmixer_suspend(struct snd_sb * break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: save_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: @@ -975,6 +977,7 @@ void snd_sbmixer_resume(struct snd_sb *c break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: restore_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: =

1 0

Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 23 May '07

23 May '07

EPPROM ON: OK, I formated that dos partition and simply sys a: c:, then I installed the driver (copying them to the partition from the linux install), in theinstall I selected the defaults settings (220 irq5 dma1...) and rebooted, now in dos again I used diagnose to test the audio and hwset /mr /ms to reset and save the epprom. now I rebooted and loged to linux: sudo modprobe snd-sb8 port=0x220 irq=5 dma8=1 FATAL: Error inserting snd_sb8 (/lib/modules/2.6.20-15-generic/kernel/sound/isa/sb/snd-sb8.ko): No such device dmesg last lines ( [ 319.827327] Sound Blaster soundcard not found or device busy -reboot- EPPROM OFF (SOFTWARE ON): in dos: diagnose works, reboot and back to lin: sudo modprobe snd-sb8 port=0x220 irq=5 dma8=1 no error, no mention in dmesg, but xmms gives the usual error... Heres lsmod just for kicks: Module Size Used by snd_sb8 9216 0 snd_sb8_dsp 12288 1 snd_sb8 snd_sb_common 17792 2 snd_sb8,snd_sb8_dsp snd_opl3_lib 11520 1 snd_sb8 snd_hwdep 9988 1 snd_opl3_lib snd_pcm_oss 44544 0 snd_pcm 79876 3 snd_sb8,snd_sb8_dsp,snd_pcm_oss snd_mixer_oss 17408 1 snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 32896 0 snd_seq_midi 9600 0 snd_rawmidi 25472 2 snd_sb8_dsp,snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 52592 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 23684 3 snd_opl3_lib,snd_pcm,snd_seq snd_seq_device 9100 6 snd_opl3_lib,snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54020 13 snd_sb8,snd_sb8_dsp,snd_sb_common,snd_opl3_lib,snd_hwdep,snd_pcm_oss,snd_pcm,snd_mixer_oss,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 8672 1 snd snd_page_alloc 10888 1 snd_pcm nfs 240876 0 nfsd 218992 17 exportfs 6912 1 nfsd lockd 64904 3 nfs,nfsd sunrpc 161340 12 nfs,nfsd,lockd apm 22752 2 ppdev 10116 0 cpufreq_stats 7360 0 cpufreq_ondemand 9228 0 cpufreq_powersave 2688 0 freq_table 5792 2 cpufreq_stats,cpufreq_ondemand cpufreq_conservative 8200 0 cpufreq_userspace 5408 0 nls_iso8859_1 5120 1 nls_cp437 6784 1 vfat 14208 1 fat 53916 1 vfat ext2 66824 1 lp 12452 0 parport_pc 36388 1 parport 36936 3 ppdev,lp,parport_pc pcspkr 4224 0 ipv6 268704 8 tsdev 8768 0 evdev 11008 1 ext3 133128 1 jbd 59816 1 ext3 mbcache 9604 2 ext2,ext3 ide_disk 17024 5 8139too 27648 0 ide_cd 32672 0 cdrom 37664 1 ide_cd floppy 59524 0 ehci_hcd 34188 0 8139cp 25088 0 mii 6528 2 8139too,8139cp uhci_hcd 25360 0 usbcore 134280 3 ehci_hcd,uhci_hcd piix 10756 0 [permanent] generic 5124 0 [permanent] ata_generic 9092 0 libata 125720 1 ata_generic scsi_mod 142348 1 libata fbcon 42656 0 tileblit 3584 1 fbcon font 9216 1 fbcon bitblit 6912 1 fbcon softcursor 3200 1 bitblit vesafb 9220 0 capability 5896 0 commoncap 8192 1 capability So now atleast we know its not the jumpers ;-) Is their the slight chance xmms is the problem ? gxine crushs too... Is their something like "beep" only for the sound card ? Maybe droping some data to /dev/### to produce a beep ? Ram Kromberg ramkromberg(a)mail.com

4 13

[alsa-devel] No state is present for card CMI8738
by Tomas Pospisek's Mailing Lists 23 May '07

23 May '07

Since alsa-user is rather ... errr ... silent, I dare to repost the same question here. I've seen that Alsa likes the Terratec cards, however since they're also based on the CMI chipset, and at least /my/ onboard CMI chipset here sucks badly I wonder what low cost card I can use? I hope somebody can give me a clue, *t PS: Mic input is barely loud enough with my onboard sound. Mic boost has no effect. -- ----------------------------------------------------------- Tomas Pospisek http://sourcepole.com - Linux & Open Source Solutions ----------------------------------------------------------- ---------- Forwarded message ---------- Date: Mon, 14 May 2007 14:39:36 +0200 (CEST) From: Tomas Pospisek's Mailing Lists <tpo2(a)sourcepole.ch> To: alsa-user(a)lists.sourceforge.net Subject: [Alsa-user] No state is present for card CMI8738 The following is my problem: $ alsactl -f .alsavoip restore No state is present for card CMI8738 I see this "sometimes". Sometimes the above works and sometimes it returns with that error message. It needs to be noted that I am suspending to ram. Nevertheless, the behaveour is not consistent in between "s2ram"s and I don't know whether that has anything to do with it either. It also needs to be noted that alsamixer *never* has any problems accessing the card. But even after changing the card settings with alsamixer, alsactl will keep on repeating that error message. * $ lspci 00:0d.0 Multimedia audio controller: C-Media Electronics Inc CM8738 (rev 10) * alsa userspace is at 1.0.13 * kernel is Debian's 2.6.18-4-686 Further, minor problems: * After waking up from suspend the card looses its microphone settings - that is - "Mic Capture" is allway reset to zero. * Further on the choice to hide the microphone setting "Mic Capture" in alsamixer behind F5 irritates me a bit - there are *a lot* of very obscure settings (such as "IEC958 5V" - huh?), that are displayed by default. What's up with that error above? What does it mean? Is there any way I can fix or circumvent it? Anybody a hint? *t -- ----------------------------------------------------------- Tomas Pospisek http://sourcepole.com - Linux & Open Source Solutions ----------------------------------------------------------- ------------------------------------------------------------------------- This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ _______________________________________________ Alsa-user mailing list Alsa-user(a)lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user

2 4

[alsa-devel] A52 plugin segfault and patch
by Fabian van der Werf 23 May '07

23 May '07

Hi All, I bumped into segfault when trying to use the a52 plugin. I linked the plugin with the latest svn revision of ffmpeg (r9102). Starting program: /usr/bin/aplay -Dsurround test.wav Failed to read a valid object file image from memory. [Thread debugging using libthread_db enabled] [New Thread -1211078432 (LWP 9835)] Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1211078432 (LWP 9835)] av_crc (ctx=0x0, crc=0, buffer=0x8072c14 , length=1116) at crc.c:77 77 if(!ctx[256]) (gdb) bt #0 av_crc (ctx=0x0, crc=0, buffer=0x8072c14 , length=1116) at crc.c:77 #1 0xb78ce8a1 in output_frame_end (s=0x0) at ac3enc.c:1135 #2 0xb78d1010 in AC3_encode_frame (avctx=0x806d3c0, frame=0x8072c10 "\vw", buf_size=6136, data=0x806e400) at ac3enc.c:1261 #3 0xb7730f68 in avcodec_encode_audio (avctx=0x806d3c0, buf=0x8072c10 "\vw", buf_size=6136, samples=0x0) at utils.c:863 #4 0xb772f66c in convert_data (rec=0x805c868) at pcm_a52.c:57 #5 0xb7730101 in a52_transfer (io=0x805c868, areas=0x80551f0, offset=0, size=1536) at pcm_a52.c:197 #6 0xb7efcf11 in ioplug_priv_transfer_areas (pcm=0x806c138, areas=0x80551f0, offset=0, size=0) at pcm_ioplug.c:521 #7 0xb7efcfd6 in snd_pcm_ioplug_mmap_commit (pcm=0x806c138, offset=0, size=1536) at pcm_ioplug.c:590 #8 0xb7ebf7b7 in snd_pcm_mmap_commit (pcm=0x0, offset=0, frames=1536) at pcm.c:6305 #9 0xb7ecc7e4 in snd_pcm_plugin_write_areas (pcm=0x8054f48, areas=0xbfc8ea10, offset=0, size=1536) at pcm_plugin.c:282 #10 0xb7ebfc42 in snd_pcm_write_areas (pcm=0x8054f48, areas=0xbfc8ea10, offset=0, size=1536, func=0xb7ecc740 <snd_pcm_plugin_write_areas>) at pcm.c:6492 #11 0xb7ecbdff in snd_pcm_plugin_writei (pcm=0x8054f48, buffer=0x8074410, size=1536) at pcm_plugin.c:353 #12 0xb7eb7da5 in snd_pcm_writei (pcm=0x8068488, buffer=0x8074410, size=0) at pcm_local.h:401 #13 0x0804d05a in pcm_write (data=0x8074410 "", count=1536) at aplay.c:1260 #14 0x0804d66f in playback_go (fd=8, loaded=0, count=48457728, rtype=2, name=0xbfc8fa1d "test.wav") at aplay.c:1986 #15 0x0804e9b2 in playback (name=0xbfc8fa1d "test.wav") at aplay.c:2044 #16 0x08050ea9 in main (argc=3, argv=0xbfc8ed64) at aplay.c:601 The patch below fixes this segmentation fault by adding a call to avcodec_init() diff -Naur alsa-plugins-1.0.13/a52/pcm_a52.c alsa-plugins-1.0.13-patched/a52/pcm_a52.c --- alsa-plugins-1.0.13/a52/pcm_a52.c 2006-09-29 13:51:37.000000000 +0200 +++ alsa-plugins-1.0.13-patched/a52/pcm_a52.c 2007-05-22 18:49:56.000000000 +0200 @@ -676,6 +676,7 @@ rec->channels = channels; rec->format = format; + avcodec_init(); avcodec_register_all(); rec->codec = avcodec_find_encoder(CODEC_ID_AC3); if (! rec->codec) { Regards, Fabian

2 1

Re: [alsa-devel] bios update alters hda-intel and alsamixer
by Tobin Davis 23 May '07

23 May '07

On Wed, 2007-05-23 at 01:21 -0500, Chris Pemberton wrote: > On 5/23/07, Tobin Davis <tdavis(a)dsl-only.net> wrote: > > Could you post the output from this script: > > http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh > > > > It was written by a user on the #alsa irc channel, and has been very > > helpful in debugging sound driver issues, at least for me. > > > > Tobin > > > > On Tue, 2007-05-22 at 23:55 -0500, Chris Pemberton wrote: > > > > > After updating the bios on a Lenovo notebook, my hda-intel sound "changed" > > > > > > Before the bios update I used kernel 2.6.20.3, external > > > alsa-driver-1.0.14rc3 and the attached patch_realtek.c (in-kernel alsa > > > wouldn't work) > > > > > > After the bios update I use just the alsa contained in kernel 2.6.20.3. > > > > > > Before bios update, my mixer contained many items: including master, > > > front, mic, pcm, etc. > > > > > > Now it only contains master and pcm. > > > > > > So, I have a few questions: > > > > > > How can I determine what items are "supposed" to show up in alsamixer? > > > > > > Where can I submit my laptop soundcard information to help finalize > > > support of this model/card? > > > > > > Thank you > > > _______________________________________________ > > > Alsa-devel mailing list > > > Alsa-devel(a)alsa-project.org > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > -- > > Tobin Davis <tdavis(a)dsl-only.net> > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > It is located here: > > http://pastebin.ca/504675 Ok, according to what I can tell, you don't have the most recent version of the driver (your original post didn't have an attachement). I do know that the Lenovo support for the ALC861VD was added after 1.0.14rc4 snapshot was made. Try installing the most recent HG daily snapshot from ftp://ftp.suse.com/pub/projects/alsa/snapshot/driver . If that doesn't fix it, rerun the alsa-info.sh script and repost the new data. -- Tobin Davis <tdavis(a)dsl-only.net>

1 0

[alsa-devel] bios update alters hda-intel and alsamixer
by Chris Pemberton 23 May '07

23 May '07

After updating the bios on a Lenovo notebook, my hda-intel sound "changed" Before the bios update I used kernel 2.6.20.3, external alsa-driver-1.0.14rc3 and the attached patch_realtek.c (in-kernel alsa wouldn't work) After the bios update I use just the alsa contained in kernel 2.6.20.3. Before bios update, my mixer contained many items: including master, front, mic, pcm, etc. Now it only contains master and pcm. So, I have a few questions: How can I determine what items are "supposed" to show up in alsamixer? Where can I submit my laptop soundcard information to help finalize support of this model/card? Thank you

2 1

Re: [alsa-devel] I don't understand snd_pcm_ops
by Ash Willis 22 May '07

22 May '07

> First off, I thought PCM stood for Pulse Code Modulation, but that > just describes the > format of the digital data. I don't understand what a PCM operation could be. Exactly, it described the format of the digital data used i audio processing. PCM does stand for pulse code modulation, but here it is a reference to a device that processes sample based digital audio i.e not midi etc. The ALSA middle layer (which includes PCM) takes care of all the common stuff that all drivers need to do. In sound drivers, we have multiple types of devices: PCM, midi, mixers etc. The PCM layer is responsible for doing all the digital audio work. I.e preparing the card for playback, initiaing transfer to and from the device etc. In short, if you want to playback or capture sound, you're going to need a PCM :) snd_pcm_ops is a structure of callbacks that relate to different events regarding the PCM interface. Use this to run your own code at specific events. I.e. it's most probable that you'll want to prepare the device in some manner before audio transfer starts, so you would put the code to do this into your prepare callback. > > I've seen documentation where PCM is considered an alternative to > I2S or AC97. But if you > look at eti_b1_wm8731.c, you'll see this: > > #include "at91-pcm.h" > #include "at91-i2s.h" > > So is the AT91 a PCM device or an I2S device??? None of these are alternatives to each other. They're all different things. The anser to your question would be that this is an i2c device that has a PCM. Have you read "Writing an ALSA driver"? It generally describes ALSA drivers as they relate to PCI, and how to write them, but it does a good job at explaining the general ALSA driver architecture. Ash =

2 1

[alsa-devel] [PATCH] Fix ASoC s3c24xx-pcm spinlock bug
by Zoltan Devai 22 May '07

22 May '07

Hi, This should fix a spinlock lockup bug on the s3c24xx arch. Regards, Zoltan --- --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -337,6 +337,8 @@ static int s3c24xx_pcm_open(struct snd_p if (prtd == NULL) return -ENOMEM; + spin_lock_init(&prtd->lock); + runtime->private_data = prtd; return 0; }

2 1

[alsa-devel] Broken JACK plugin in alsa-plugins and a way to fix it.
by Jasiu 22 May '07

22 May '07

Hi. I want to note that JACK plugin in alsa-plugins in broken. (At least on my two systems.) It just plays for a few seconds and then stops hanging the application that was sending data to it. When it hangs jackd spits out this: subgraph starting at alsa-jack.jackP.3635.0 timed out (subgraph_wait_fd=7, status = 0, state = Running) What I want to note is that there is a fix to that and I would like it to be included in the tree. I'm not sure if it doesn't break anything else, but it works for me ok. Open pcm_jack.c and on the line 146 you'll find: write(jack->fd, buf, 1); /* for polling */ Deleting this seems to get rid of the problem. So please if you can patch this out. (I didn't make a patch myself since this is so tiny fix that it isn't really worth.) Cheers.

2 3

[alsa-devel] [PATCH] remove email addresses from MODULE_AUTHOR tag
by Rene Herman 22 May '07

22 May '07

Hi Takashi. Following up the recent MODULE_MAINTAINER discussion on LKML: http://lkml.org/lkml/2007/4/4/170 that concluded with MODULE_MAINTAINER not being a good idea, here's a patch that just deletes the email addresses from the MODULE_AUTHOR tags for ALSA. The email address is the problem I was trying to fix; with multiple current and non-current authors and maintainers who might not even be authors the address(es) available from the tag confuse the issue of whom to contact. It's moreover also information that easily outdated. A bit more than half of the tags in the kernel don't include an email address already (the core contributors typically don't include any) and I'll submit patches removing more. (this was generated against current -linus). aoa/codecs/snd-aoa-codec-onyx.c | 2 +- aoa/codecs/snd-aoa-codec-tas.c | 2 +- aoa/codecs/snd-aoa-codec-toonie.c | 2 +- aoa/core/snd-aoa-core.c | 2 +- aoa/fabrics/snd-aoa-fabric-layout.c | 2 +- aoa/soundbus/core.c | 2 +- aoa/soundbus/i2sbus/i2sbus-core.c | 2 +- arm/sa11xx-uda1341.c | 2 +- core/hwdep.c | 2 +- core/memalloc.c | 2 +- core/oss/mixer_oss.c | 2 +- core/oss/pcm_oss.c | 2 +- core/pcm.c | 2 +- core/rawmidi.c | 2 +- core/seq/instr/ainstr_fm.c | 2 +- core/seq/instr/ainstr_gf1.c | 2 +- core/seq/instr/ainstr_iw.c | 2 +- core/seq/instr/ainstr_simple.c | 2 +- core/seq/oss/seq_oss.c | 2 +- core/seq/seq.c | 2 +- core/seq/seq_device.c | 2 +- core/seq/seq_dummy.c | 2 +- core/seq/seq_instr.c | 2 +- core/seq/seq_midi.c | 2 +- core/seq/seq_midi_emul.c | 2 +- core/seq/seq_midi_event.c | 2 +- core/seq/seq_virmidi.c | 2 +- core/sound.c | 2 +- core/timer.c | 2 +- drivers/dummy.c | 2 +- drivers/mpu401/mpu401.c | 2 +- drivers/mpu401/mpu401_uart.c | 2 +- drivers/mts64.c | 2 +- drivers/opl3/opl3_lib.c | 2 +- drivers/opl3/opl3_seq.c | 2 +- drivers/opl4/opl4_lib.c | 2 +- drivers/opl4/opl4_seq.c | 2 +- drivers/virmidi.c | 2 +- drivers/vx/vx_core.c | 2 +- i2c/cs8427.c | 2 +- i2c/i2c.c | 2 +- i2c/l3/uda1341.c | 2 +- i2c/other/ak4114.c | 2 +- i2c/other/ak4117.c | 2 +- i2c/other/ak4xxx-adda.c | 2 +- i2c/other/pt2258.c | 2 +- i2c/other/tea575x-tuner.c | 2 +- i2c/tea6330t.c | 2 +- isa/ad1816a/ad1816a.c | 2 +- isa/ad1848/ad1848.c | 2 +- isa/ad1848/ad1848_lib.c | 2 +- isa/als100.c | 2 +- isa/azt2320.c | 2 +- isa/cmi8330.c | 2 +- isa/cs423x/cs4231.c | 2 +- isa/cs423x/cs4231_lib.c | 2 +- isa/cs423x/cs4236.c | 2 +- isa/cs423x/cs4236_lib.c | 2 +- isa/dt019x.c | 2 +- isa/es1688/es1688.c | 2 +- isa/es1688/es1688_lib.c | 2 +- isa/es18xx.c | 2 +- isa/gus/gus_main.c | 2 +- isa/gus/gus_synth.c | 2 +- isa/gus/gusclassic.c | 2 +- isa/gus/gusextreme.c | 2 +- isa/gus/gusmax.c | 2 +- isa/gus/interwave.c | 2 +- isa/opl3sa2.c | 2 +- isa/opti9xx/miro.c | 2 +- isa/opti9xx/opti92x-ad1848.c | 2 +- isa/sb/es968.c | 2 +- isa/sb/sb16.c | 2 +- isa/sb/sb16_csp.c | 2 +- isa/sb/sb16_main.c | 2 +- isa/sb/sb8.c | 2 +- isa/sb/sb8_main.c | 2 +- isa/sb/sb_common.c | 2 +- isa/sgalaxy.c | 2 +- isa/wavefront/wavefront.c | 2 +- mips/au1x00.c | 2 +- parisc/harmony.c | 2 +- pci/ac97/ac97_codec.c | 2 +- pci/ac97/ak4531_codec.c | 2 +- pci/ad1889.c | 2 +- pci/ali5451/ali5451.c | 2 +- pci/als300.c | 2 +- pci/als4000.c | 2 +- pci/atiixp.c | 2 +- pci/atiixp_modem.c | 2 +- pci/azt3328.c | 2 +- pci/bt87x.c | 2 +- pci/ca0106/ca0106_main.c | 2 +- pci/cmipci.c | 2 +- pci/cs4281.c | 2 +- pci/cs46xx/cs46xx.c | 2 +- pci/echoaudio/echoaudio.c | 2 +- pci/emu10k1/emu10k1.c | 2 +- pci/emu10k1/emu10k1x.c | 2 +- pci/ens1370.c | 2 +- pci/es1938.c | 2 +- pci/fm801.c | 2 +- pci/hda/hda_codec.c | 2 +- pci/ice1712/ak4xxx.c | 2 +- pci/ice1712/ice1712.c | 2 +- pci/ice1712/ice1724.c | 2 +- pci/intel8x0.c | 2 +- pci/intel8x0m.c | 2 +- pci/korg1212/korg1212.c | 2 +- pci/maestro3.c | 2 +- pci/mixart/mixart.c | 2 +- pci/nm256/nm256.c | 2 +- pci/pcxhr/pcxhr.c | 2 +- pci/riptide/riptide.c | 2 +- pci/rme32.c | 2 +- pci/rme96.c | 2 +- pci/rme9652/hdsp.c | 2 +- pci/rme9652/rme9652.c | 2 +- pci/sonicvibes.c | 2 +- pci/trident/trident.c | 2 +- pci/trident/trident_synth.c | 2 +- pci/via82xx.c | 2 +- pci/via82xx_modem.c | 2 +- pci/vx222/vx222.c | 2 +- pci/ymfpci/ymfpci.c | 2 +- pcmcia/pdaudiocf/pdaudiocf.c | 2 +- pcmcia/vx/vxpocket.c | 2 +- soc/at91/at91-i2s.c | 2 +- soc/at91/at91-pcm.c | 2 +- soc/at91/eti_b1_wm8731.c | 2 +- soc/pxa/pxa2xx-i2s.c | 2 +- soc/soc-core.c | 2 +- soc/soc-dapm.c | 2 +- usb/usbaudio.c | 2 +- usb/usbmidi.c | 2 +- usb/usx2y/usbusx2y.c | 2 +- 136 files changed, 136 insertions(+), 136 deletions(-) Signed-off-by: Rene Herman <rene.herman(a)gmail.com> Rene. commit 7c179369bc673d2bac7f67fb94aee1d39b4c3f4d Author: Rene Herman <rene.herman(a)gmail.com> Date: Fri May 11 03:03:00 2007 +0200 module_author: alsa Signed-off-by: Rene Herman <rene.herman(a)gmail.com> diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.c b/sound/aoa/codecs/snd-aoa-codec-onyx.c index e91f9f6..f9b8159 100644 --- a/sound/aoa/codecs/snd-aoa-codec-onyx.c +++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c @@ -33,7 +33,7 @@ */ #include <linux/delay.h> #include <linux/module.h> -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa"); diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/snd-aoa-codec-tas.c index 041fe52..b1d2287 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas.c +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c @@ -67,7 +67,7 @@ #include <linux/module.h> #include <linux/mutex.h> -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("tas codec driver for snd-aoa"); diff --git a/sound/aoa/codecs/snd-aoa-codec-toonie.c b/sound/aoa/codecs/snd-aoa-codec-toonie.c index 3c7d1d8..0ccdaf9 100644 --- a/sound/aoa/codecs/snd-aoa-codec-toonie.c +++ b/sound/aoa/codecs/snd-aoa-codec-toonie.c @@ -11,7 +11,7 @@ */ #include <linux/delay.h> #include <linux/module.h> -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("toonie codec driver for snd-aoa"); diff --git a/sound/aoa/core/snd-aoa-core.c b/sound/aoa/core/snd-aoa-core.c index 19fdae4..23372e3 100644 --- a/sound/aoa/core/snd-aoa-core.c +++ b/sound/aoa/core/snd-aoa-core.c @@ -13,7 +13,7 @@ #include "snd-aoa-alsa.h" MODULE_DESCRIPTION("Apple Onboard Audio Sound Driver"); -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_LICENSE("GPL"); /* We allow only one fabric. This simplifies things, diff --git a/sound/aoa/fabrics/snd-aoa-fabric-layout.c b/sound/aoa/fabrics/snd-aoa-fabric-layout.c index 9880628..f421ae7 100644 --- a/sound/aoa/fabrics/snd-aoa-fabric-layout.c +++ b/sound/aoa/fabrics/snd-aoa-fabric-layout.c @@ -17,7 +17,7 @@ #include "../aoa.h" #include "../soundbus/soundbus.h" -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Layout-ID fabric for snd-aoa"); diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index 8b2e9b9..d5f6851 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -9,7 +9,7 @@ #include <linux/module.h> #include "soundbus.h" -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Apple Soundbus"); diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c index 0fccdbf..a99e26c 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c +++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c @@ -21,7 +21,7 @@ #include "i2sbus.h" MODULE_LICENSE("GPL"); -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_DESCRIPTION("Apple Soundbus: I2S support"); /* for auto-loading, declare that we handle this weird * string that macio puts into the relevant device */ diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index c7e1b26..0eaf1d0 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -106,7 +106,7 @@ /* {{{ Type definitions */ -MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek(a)seznam.cz>"); +MODULE_AUTHOR("Tomas Kasparek"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA"); MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}"); diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 51ad95b..2198ecb 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -31,7 +31,7 @@ #include <sound/hwdep.h> #include <sound/info.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Hardware dependent layer"); MODULE_LICENSE("GPL"); diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index f057430..0f79a51 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -37,7 +37,7 @@ #endif -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>, Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Takashi Iwai, Jaroslav Kysela"); MODULE_DESCRIPTION("Memory allocator for ALSA system."); MODULE_LICENSE("GPL"); diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index fccad8f..36d417a 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -33,7 +33,7 @@ #define OSS_ALSAEMULVER _SIOR ('M', 249, int) -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Mixer OSS emulation for ALSA."); MODULE_LICENSE("GPL"); MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_MIXER); diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index fc11572..97dcf58 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -48,7 +48,7 @@ static int dsp_map[SNDRV_CARDS]; static int adsp_map[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1}; static int nonblock_open = 1; -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Abramo Bagnara <abramo(a)alsa-project.org>"); +MODULE_AUTHOR("Jaroslav Kysela, Abramo Bagnara"); MODULE_DESCRIPTION("PCM OSS emulation for ALSA."); MODULE_LICENSE("GPL"); module_param_array(dsp_map, int, NULL, 0444); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 2743414..8be04b2 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -30,7 +30,7 @@ #include <sound/control.h> #include <sound/info.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Abramo Bagnara <abramo(a)alsa-project.org>"); +MODULE_AUTHOR("Jaroslav Kysela, Abramo Bagnara"); MODULE_DESCRIPTION("Midlevel PCM code for ALSA."); MODULE_LICENSE("GPL"); diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index e470c3c..1419c4a 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -37,7 +37,7 @@ #include <sound/minors.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Midlevel RawMidi code for ALSA."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/instr/ainstr_fm.c b/sound/core/seq/instr/ainstr_fm.c index f80fab8..e9d680d 100644 --- a/sound/core/seq/instr/ainstr_fm.c +++ b/sound/core/seq/instr/ainstr_fm.c @@ -25,7 +25,7 @@ #include <sound/initval.h> #include <asm/uaccess.h> -MODULE_AUTHOR("Uros Bizjak <uros(a)kss-loka.si>"); +MODULE_AUTHOR("Uros Bizjak"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture FM Instrument support."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/instr/ainstr_gf1.c b/sound/core/seq/instr/ainstr_gf1.c index c640e1c..3d4a411 100644 --- a/sound/core/seq/instr/ainstr_gf1.c +++ b/sound/core/seq/instr/ainstr_gf1.c @@ -26,7 +26,7 @@ #include <sound/initval.h> #include <asm/uaccess.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture GF1 (GUS) Patch support."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/instr/ainstr_iw.c b/sound/core/seq/instr/ainstr_iw.c index 5367bae..d2d42c9 100644 --- a/sound/core/seq/instr/ainstr_iw.c +++ b/sound/core/seq/instr/ainstr_iw.c @@ -26,7 +26,7 @@ #include <sound/initval.h> #include <asm/uaccess.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture IWFFFF support."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/instr/ainstr_simple.c b/sound/core/seq/instr/ainstr_simple.c index ac717be..12aeb66 100644 --- a/sound/core/seq/instr/ainstr_simple.c +++ b/sound/core/seq/instr/ainstr_simple.c @@ -26,7 +26,7 @@ #include <sound/initval.h> #include <asm/uaccess.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture Simple Instrument support."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index bc09923..fcb85f0 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -33,7 +33,7 @@ /* * module option */ -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("OSS-compatible sequencer module"); MODULE_LICENSE("GPL"); /* Takashi says this is really only for sound-service-0-, but this is OK. */ diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 2f0d877..c5f98e0 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -53,7 +53,7 @@ int seq_default_timer_device = int seq_default_timer_subdevice = 0; int seq_default_timer_resolution = 0; /* Hz */ -MODULE_AUTHOR("Frank van de Pol <fvdpol(a)coil.demon.nl>, Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Frank van de Pol, Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 37852cd..c05f067 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -47,7 +47,7 @@ #include <linux/slab.h> #include <linux/mutex.h> -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("ALSA sequencer device management"); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index e55488d..bc84eb0 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -60,7 +60,7 @@ */ -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("ALSA sequencer MIDI-through client"); MODULE_LICENSE("GPL"); MODULE_ALIAS("snd-seq-client-" __stringify(SNDRV_SEQ_CLIENT_DUMMY)); diff --git a/sound/core/seq/seq_instr.c b/sound/core/seq/seq_instr.c index f30d171..1332de0 100644 --- a/sound/core/seq/seq_instr.c +++ b/sound/core/seq/seq_instr.c @@ -26,7 +26,7 @@ #include <sound/seq_instr.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer instrument library."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 1daa5b0..07e8738 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -40,7 +40,7 @@ Possible options for midisynth module: #include <sound/seq_midi_event.h> #include <sound/initval.h> -MODULE_AUTHOR("Frank van de Pol <fvdpol(a)coil.demon.nl>, Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Frank van de Pol, Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer MIDI synth."); MODULE_LICENSE("GPL"); static int output_buffer_size = PAGE_SIZE; diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index d7c4fb8..bf23c5c 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -39,7 +39,7 @@ #include <sound/initval.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Takashi Iwai / Steve Ratcliffe"); +MODULE_AUTHOR("Takashi Iwai, Steve Ratcliffe"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer MIDI emulation."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index 5ff80b7..e0e7557 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -28,7 +28,7 @@ #include <sound/seq_midi_event.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>, Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Takashi Iwai, Jaroslav Kysela"); MODULE_DESCRIPTION("MIDI byte <-> sequencer event coder"); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 061a7c6..52ce7c9 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -48,7 +48,7 @@ #include <sound/seq_midi_event.h> #include <sound/seq_virmidi.h> -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("Virtual Raw MIDI client on Sequencer"); MODULE_LICENSE("GPL"); diff --git a/sound/core/sound.c b/sound/core/sound.c index 70600df..fac94ea 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -42,7 +42,7 @@ EXPORT_SYMBOL(snd_major); static int cards_limit = 1; -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture driver for soundcards."); MODULE_LICENSE("GPL"); module_param(major, int, 0444); diff --git a/sound/core/timer.c b/sound/core/timer.c index 67520b3..e7c0552 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -44,7 +44,7 @@ #endif static int timer_limit = DEFAULT_TIMER_LIMIT; -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Jaroslav Kysela, Takashi Iwai"); MODULE_DESCRIPTION("ALSA timer interface"); MODULE_LICENSE("GPL"); module_param(timer_limit, int, 0444); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index a0f28f5..5a9e5cc 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -34,7 +34,7 @@ #include <sound/rawmidi.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Dummy soundcard (/dev/null)"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ALSA,Dummy soundcard}}"); diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 2de181a..6f7af4c 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -30,7 +30,7 @@ #include <sound/mpu401.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("MPU-401 UART"); MODULE_LICENSE("GPL"); diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 3daa9fa..9aa30a7 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -39,7 +39,7 @@ #include <sound/core.h> #include <sound/mpu401.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of MPU-401 in UART mode"); MODULE_LICENSE("GPL"); diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 6c9f4c9..cd4e604 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -47,7 +47,7 @@ MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); module_param_array(enable, bool, NULL, S_IRUGO); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); -MODULE_AUTHOR("Matthias Koenig <mk(a)phasorlab.de>"); +MODULE_AUTHOR("Matthias Koenig"); MODULE_DESCRIPTION("ESI Miditerminal 4140"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ESI,Miditerminal 4140}}"); diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 87fe376..f041d4e 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -31,7 +31,7 @@ #include <linux/ioport.h> #include <sound/minors.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Hannu Savolainen 1993-1996, Rob Hooft"); +MODULE_AUTHOR("Jaroslav Kysela, Hannu Savolainen, Rob Hooft"); MODULE_DESCRIPTION("Routines for control of AdLib FM cards (OPL2/OPL3/OPL4 chips)"); MODULE_LICENSE("GPL"); diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index 96762c9..d5771d9 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -27,7 +27,7 @@ #include <linux/moduleparam.h> #include <sound/initval.h> -MODULE_AUTHOR("Uros Bizjak <uros(a)kss-loka.si>"); +MODULE_AUTHOR("Uros Bizjak"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("ALSA driver for OPL3 FM synth"); diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index 01997f2..8611a8e 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -23,7 +23,7 @@ #include <linux/init.h> #include <asm/io.h> -MODULE_AUTHOR("Clemens Ladisch <clemens(a)ladisch.de>"); +MODULE_AUTHOR("Clemens Ladisch"); MODULE_DESCRIPTION("OPL4 driver"); MODULE_LICENSE("GPL"); diff --git a/sound/drivers/opl4/opl4_seq.c b/sound/drivers/opl4/opl4_seq.c index 43d8a2b..96f5b50 100644 --- a/sound/drivers/opl4/opl4_seq.c +++ b/sound/drivers/opl4/opl4_seq.c @@ -36,7 +36,7 @@ #include <linux/moduleparam.h> #include <sound/initval.h> -MODULE_AUTHOR("Clemens Ladisch <clemens(a)ladisch.de>"); +MODULE_AUTHOR("Clemens Ladisch"); MODULE_DESCRIPTION("OPL4 wavetable synth driver"); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 46f3d34..fb324c2 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -55,7 +55,7 @@ /* hack: OSS defines midi_devs, so undefine it (versioned symbols) */ #undef midi_devs -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("Dummy soundcard for virtual rawmidi devices"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ALSA,Virtual rawmidi device}}"); diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index ed19bc1..594f577 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -35,7 +35,7 @@ #include <sound/vx_core.h> #include "vx_cmd.h" -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("Common routines for Digigram VX drivers"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 64388cb..73586e7 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -32,7 +32,7 @@ static void snd_cs8427_reset(struct snd_i2c_device *cs8427); -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("IEC958 (S/PDIF) receiver & transmitter by Cirrus Logic"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c index b60fb18..1218b9b 100644 --- a/sound/i2c/i2c.c +++ b/sound/i2c/i2c.c @@ -28,7 +28,7 @@ #include <sound/core.h> #include <sound/i2c.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Generic i2c interface for ALSA"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c index b074fdd..7d18dd6 100644 --- a/sound/i2c/l3/uda1341.c +++ b/sound/i2c/l3/uda1341.c @@ -921,7 +921,7 @@ static void __exit uda1341_exit(void) module_init(uda1341_init); module_exit(uda1341_exit); -MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek(a)seznam.cz>"); +MODULE_AUTHOR("Tomas Kasparek"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Philips UDA1341 CODEC driver for ALSA"); MODULE_SUPPORTED_DEVICE("{{UDA1341,UDA1341TS}}"); diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index adbfd58..c32b717 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -29,7 +29,7 @@ #include <sound/ak4114.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("AK4114 IEC958 (S/PDIF) receiver by Asahi Kasei"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index c022f29..0391f31 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -29,7 +29,7 @@ #include <sound/ak4117.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("AK4117 IEC958 (S/PDIF) receiver by Asahi Kasei"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 8805110..a38d2a2 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -31,7 +31,7 @@ #include <sound/tlv.h> #include <sound/ak4xxx-adda.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Jaroslav Kysela, Takashi Iwai"); MODULE_DESCRIPTION("Routines for control of AK452x / AK43xx AD/DA converters"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/other/pt2258.c b/sound/i2c/other/pt2258.c index e91cc3b..c8499ac 100644 --- a/sound/i2c/other/pt2258.c +++ b/sound/i2c/other/pt2258.c @@ -26,7 +26,7 @@ #include <sound/i2c.h> #include <sound/pt2258.h> -MODULE_AUTHOR("Jochen Voss <voss(a)seehuhn.de>"); +MODULE_AUTHOR("Jochen Voss"); MODULE_DESCRIPTION("PT2258 volume controller (Princeton Technology Corp.)"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 4c2fd14..6ef2981 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -28,7 +28,7 @@ #include <sound/core.h> #include <sound/tea575x-tuner.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c index ae5b1e3..91e9ab5 100644 --- a/sound/i2c/tea6330t.c +++ b/sound/i2c/tea6330t.c @@ -27,7 +27,7 @@ #include <sound/control.h> #include <sound/tea6330t.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of the TEA6330T circuit via i2c bus"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 5903450..74adb20 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -32,7 +32,7 @@ #define PFX "ad1816a: " -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_AUTHOR("Massimo Piccioni"); MODULE_DESCRIPTION("AD1816A, AD1815"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Highscreen,Sound-Boostar 16 3D}," diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index 74e501d..49f1ef3 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -32,7 +32,7 @@ #include <sound/ad1848.h> #include <sound/initval.h> -MODULE_AUTHOR("Tugrul Galatali <galatalt(a)stuy.edu>, Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Tugrul Galatali, Jaroslav Kysela"); MODULE_DESCRIPTION("AD1848/AD1847/CS4248"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Analog Devices,AD1848}," diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index 8094282..edad9ce 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -35,7 +35,7 @@ #include <asm/io.h> #include <asm/dma.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of AD1848/AD1847/CS4248"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/als100.c b/sound/isa/als100.c index f2bcfb2..8939193 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -34,7 +34,7 @@ #define PFX "als100: " -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_AUTHOR("Massimo Piccioni"); MODULE_DESCRIPTION("Avance Logic ALS1X0"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS100 - PRO16PNP}," diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c index b615538..e974387 100644 --- a/sound/isa/azt2320.c +++ b/sound/isa/azt2320.c @@ -45,7 +45,7 @@ #define PFX "azt2320: " -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_AUTHOR("Massimo Piccioni"); MODULE_DESCRIPTION("Aztech Systems AZT2320"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Aztech Systems,PRO16V}," diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index c09a800..3d4b2f3 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -62,7 +62,7 @@ /* */ -MODULE_AUTHOR("George Talusan <gstalusan(a)uwaterloo.ca>"); +MODULE_AUTHOR("George Talusan"); MODULE_DESCRIPTION("C-Media CMI8330"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8330,isapnp:{CMI0001,@@@0001,@X@0001}}}"); diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index 696a5c8..4a0e35d 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -32,7 +32,7 @@ #include <sound/mpu401.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Generic CS4231"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4231}}"); diff --git a/sound/isa/cs423x/cs4231_lib.c b/sound/isa/cs423x/cs4231_lib.c index 75c7c5f..77b7af7 100644 --- a/sound/isa/cs423x/cs4231_lib.c +++ b/sound/isa/cs423x/cs4231_lib.c @@ -39,7 +39,7 @@ #include <asm/dma.h> #include <asm/irq.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of CS4231(A)/CS4232/InterWave & compatible chips"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 07ffd5c..9dbcd2f 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -32,7 +32,7 @@ #include <sound/opl3.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_LICENSE("GPL"); #ifdef CS4232 MODULE_DESCRIPTION("Cirrus Logic CS4232"); diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 7a5a6c7..2cd2055 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -89,7 +89,7 @@ #include <sound/cs4231.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of CS4235/4236B/4237B/4238B/4239 chips"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/dt019x.c b/sound/isa/dt019x.c index ce57d52..7955a1e 100644 --- a/sound/isa/dt019x.c +++ b/sound/isa/dt019x.c @@ -34,7 +34,7 @@ #define PFX "dt019x: " -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_AUTHOR("Massimo Piccioni"); MODULE_DESCRIPTION("Diamond Technologies DT-019X / Avance Logic ALS-007"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X}," diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 65f97ff..f496783 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -35,7 +35,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("ESS ESx688 AudioDrive"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ESS,ES688 PnP AudioDrive,pnp:ESS0100}," diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index a2ab99f..2e407e8 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -32,7 +32,7 @@ #include <asm/io.h> #include <asm/dma.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("ESS ESx688 lowlevel module"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 725c115..e951f05 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1983,7 +1983,7 @@ static int __devinit snd_es18xx_mixer(struct snd_es18xx *chip) /* Card level */ -MODULE_AUTHOR("Christian Fischbach <fishbach(a)pool.informatik.rwth-aachen.de>, Abramo Bagnara <abramo(a)alsa-project.org>"); +MODULE_AUTHOR("Christian Fischbach, Abramo Bagnara"); MODULE_DESCRIPTION("ESS ES18xx AudioDrive"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ESS,ES1868 PnP AudioDrive}," diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index 8ced5e8..45ca31c 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -31,7 +31,7 @@ #include <asm/dma.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for Gravis UltraSound soundcards"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/gus/gus_synth.c b/sound/isa/gus/gus_synth.c index 3e4d4d6..4ebd902 100644 --- a/sound/isa/gus/gus_synth.c +++ b/sound/isa/gus/gus_synth.c @@ -26,7 +26,7 @@ #include <sound/gus.h> #include <sound/seq_device.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for Gravis UltraSound soundcards - Synthesizer"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index 0395e2e..ef6edeb 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -33,7 +33,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Gravis UltraSound Classic"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Classic}}"); diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 4f55fc3..76c04f5 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -37,7 +37,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Gravis UltraSound Extreme"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Extreme}}"); diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index d1ad90c..9ec9486 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -34,7 +34,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Gravis UltraSound MAX"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound MAX}}"); diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 4ec2d79..08f1099 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -41,7 +41,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_LICENSE("GPL"); #ifndef SNDRV_STB MODULE_DESCRIPTION("AMD InterWave"); diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index f3db686..34f84d0 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -37,7 +37,7 @@ #include <asm/io.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Yamaha OPL3SA2+"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Yamaha,YMF719E-S}," diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 1dd9837..758c511 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -43,7 +43,7 @@ #include <sound/initval.h> #include "miro.h" -MODULE_AUTHOR("Martin Langer <martin-langer(a)gmx.de>"); +MODULE_AUTHOR("Martin Langer"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Miro miroSOUND PCM1 pro, PCM12, PCM20 Radio"); MODULE_SUPPORTED_DEVICE("{{Miro,miroSOUND PCM1 pro}, " diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index df22737..bf172e9 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -53,7 +53,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_AUTHOR("Massimo Piccioni"); MODULE_LICENSE("GPL"); #ifdef OPTi93X MODULE_DESCRIPTION("OPTi93X"); diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c index d4b2187..2d8b942 100644 --- a/sound/isa/sb/es968.c +++ b/sound/isa/sb/es968.c @@ -31,7 +31,7 @@ #define PFX "es968: " -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_AUTHOR("Massimo Piccioni"); MODULE_DESCRIPTION("ESS AudioDrive ES968"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ESS,AudioDrive ES968}}"); diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index d64e67f..ac84169 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -44,7 +44,7 @@ #define PFX "sb16: " #endif -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_LICENSE("GPL"); #ifndef SNDRV_SBAWE MODULE_DESCRIPTION("Sound Blaster 16"); diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 3d9d7e0..fd27bac 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -33,7 +33,7 @@ #include <sound/sb16_csp.h> #include <sound/initval.h> -MODULE_AUTHOR("Uros Bizjak <uros(a)kss-loka.si>"); +MODULE_AUTHOR("Uros Bizjak"); MODULE_DESCRIPTION("ALSA driver for SB16 Creative Signal Processor"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 383911b..1c290a8 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -45,7 +45,7 @@ #include <sound/control.h> #include <sound/info.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of 16-bit SoundBlaster cards and clones"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index be1e83e..f11e672 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -31,7 +31,7 @@ #include <sound/opl3.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Sound Blaster 1.0/2.0/Pro"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB 1.0/SB 2.0/SB Pro}}"); diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index aea9e5e..8bdca08 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -38,7 +38,7 @@ #include <sound/core.h> #include <sound/sb.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Uros Bizjak <uros(a)kss-loka.si>"); +MODULE_AUTHOR("Jaroslav Kysela, Uros Bizjak"); MODULE_DESCRIPTION("Routines for control of 8-bit SoundBlaster cards and clones"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 3094f38..658b571 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -33,7 +33,7 @@ #include <asm/io.h> #include <asm/dma.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("ALSA lowlevel driver for Sound Blaster cards"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/sgalaxy.c b/sound/isa/sgalaxy.c index 4fcd0f4..76d7cbf 100644 --- a/sound/isa/sgalaxy.c +++ b/sound/isa/sgalaxy.c @@ -38,7 +38,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Christopher Butler <chrisb(a)sandy.force9.co.uk>"); +MODULE_AUTHOR("Christopher Butler"); MODULE_DESCRIPTION("Aztech Sound Galaxy"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Aztech Systems,Sound Galaxy}}"); diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index e2fdd5f..826d6d8 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -32,7 +32,7 @@ #include <sound/opl3.h> #include <sound/snd_wavefront.h> -MODULE_AUTHOR("Paul Barton-Davis <pbd(a)op.net>"); +MODULE_AUTHOR("Paul Barton-Davis"); MODULE_DESCRIPTION("Turtle Beach Wavefront"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Turtle Beach,Maui/Tropez/Tropez+}}"); diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 8a61a11..3d4986a 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -48,7 +48,7 @@ #include <asm/mach-au1x00/au1000.h> #include <asm/mach-au1x00/au1000_dma.h> -MODULE_AUTHOR("Charles Eidsness <charles(a)cooper-street.com>"); +MODULE_AUTHOR("Charles Eidsness"); MODULE_DESCRIPTION("Au1000 AC'97 ALSA Driver"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{AMD,Au1000 AC'97}}"); diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index ff705c6..d549168 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -1036,7 +1036,7 @@ alsa_harmony_fini(void) } MODULE_LICENSE("GPL"); -MODULE_AUTHOR("Kyle McMartin <kyle(a)parisc-linux.org>"); +MODULE_AUTHOR("Kyle McMartin"); MODULE_DESCRIPTION("Harmony sound driver"); module_init(alsa_harmony_init); diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 3bfb210..fe48c1b 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -39,7 +39,7 @@ #include "ac97_id.h" #include "ac97_patch.h" -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Universal interface for Audio Codec '97"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ac97/ak4531_codec.c index dc26820..6b43fd4 100644 --- a/sound/pci/ac97/ak4531_codec.c +++ b/sound/pci/ac97/ak4531_codec.c @@ -29,7 +29,7 @@ #include <sound/ak4531_codec.h> #include <sound/tlv.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Universal routines for AK4531 codec"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 98970d4..eb1ae50 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -53,7 +53,7 @@ #define AD1889_DRVVER "Version: 1.7" -MODULE_AUTHOR("Kyle McMartin <kyle(a)parisc-linux.org>, Thibaut Varene <t-bone(a)parisc-linux.org>"); +MODULE_AUTHOR("Kyle McMartin, Thibaut Varene"); MODULE_DESCRIPTION("Analog Devices AD1889 ALSA sound driver"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Analog Devices,AD1889}}"); diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index ba7fa22..14b5537 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -41,7 +41,7 @@ #include <sound/mpu401.h> #include <sound/initval.h> -MODULE_AUTHOR("Matt Wu <Matt_Wu(a)acersoftech.com.cn>"); +MODULE_AUTHOR("Matt Wu"); MODULE_DESCRIPTION("ALI M5451"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ALI,M5451,pci},{ALI,M5451}}"); diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 8afcb98..b0c922f 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -109,7 +109,7 @@ enum {DEVICE_ALS300, DEVICE_ALS300_PLUS}; -MODULE_AUTHOR("Ash Willis <ashwillis(a)programmer.net>"); +MODULE_AUTHOR("Ash Willis"); MODULE_DESCRIPTION("Avance Logic ALS300"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS300},{Avance Logic,ALS300+}}"); diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 8fb55d3..af56e9e 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -79,7 +79,7 @@ #include <sound/sb.h> #include <sound/initval.h> -MODULE_AUTHOR("Bart Hartgers <bart(a)etpmod.phys.tue.nl>"); +MODULE_AUTHOR("Bart Hartgers"); MODULE_DESCRIPTION("Avance Logic ALS4000"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS4000}}"); diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 7d8053b..3ce34a3 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -35,7 +35,7 @@ #include <sound/ac97_codec.h> #include <sound/initval.h> -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("ATI IXP AC97 controller"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ATI,IXP150/200/250/300/400}}"); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 904023f..346912a 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -35,7 +35,7 @@ #include <sound/ac97_codec.h> #include <sound/initval.h> -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("ATI IXP MC97 controller"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ATI,IXP150/200/250}}"); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 43edd28..8321f55 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -126,7 +126,7 @@ #include <sound/initval.h> #include "azt3328.h" -MODULE_AUTHOR("Andreas Mohr <andi AT lisas.de>"); +MODULE_AUTHOR("Andreas Mohr"); MODULE_DESCRIPTION("Aztech AZF3328 (PCI168)"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index e9b029e..b8a2130 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -35,7 +35,7 @@ #include <sound/control.h> #include <sound/initval.h> -MODULE_AUTHOR("Clemens Ladisch <clemens(a)ladisch.de>"); +MODULE_AUTHOR("Clemens Ladisch"); MODULE_DESCRIPTION("Brooktree Bt87x audio driver"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Brooktree,Bt878}," diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index ea6712b..416567f 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -145,7 +145,7 @@ #include <sound/ac97_codec.h> #include <sound/info.h> -MODULE_AUTHOR("James Courtier-Dutton <James(a)superbug.demon.co.uk>"); +MODULE_AUTHOR("James Courtier-Dutton"); MODULE_DESCRIPTION("CA0106"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Creative,SB CA0106 chip}}"); diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 7d3c5ee..ed43681 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -41,7 +41,7 @@ #include <sound/asoundef.h> #include <sound/initval.h> -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("C-Media CMI8x38 PCI"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8738}," diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 44cf546..9bec625 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -38,7 +38,7 @@ #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Cirrus Logic CS4281"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,CS4281}}"); diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 8b6cd14..6e0926a 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -34,7 +34,7 @@ #include <sound/cs46xx.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Cirrus Logic Sound Fusion CS46XX"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,Sound Fusion (CS4280)}," diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index e413da0..2f7c3f6 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -16,7 +16,7 @@ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */ -MODULE_AUTHOR("Giuliano Pochini <pochini(a)shiny.it>"); +MODULE_AUTHOR("Giuliano Pochini"); MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Echoaudio " ECHOCARD_NAME " soundcards driver"); MODULE_SUPPORTED_DEVICE("{{Echoaudio," ECHOCARD_NAME "}}"); diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 55caf34..34b9c35 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -32,7 +32,7 @@ #include <sound/emu10k1.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("EMU10K1"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB Live!/PCI512/E-mu APS}," diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index bb0fec7..f299cf3 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -43,7 +43,7 @@ #include <sound/info.h> #include <sound/rawmidi.h> -MODULE_AUTHOR("Francisco Moraes <fmoraes(a)nc.rr.com>"); +MODULE_AUTHOR("Francisco Moraes"); MODULE_DESCRIPTION("EMU10K1X"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Dell Creative Labs,SB Live!}"); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 425b167..48ff186 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -61,7 +61,7 @@ #endif -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Thomas Sailer <sailer(a)ife.ee.ethz.ch>"); +MODULE_AUTHOR("Jaroslav Kysela, Thomas Sailer"); MODULE_LICENSE("GPL"); #ifdef CHIP1370 MODULE_DESCRIPTION("Ensoniq AudioPCI ES1370"); diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index fec29a1..0e084d3 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -66,7 +66,7 @@ #include <asm/io.h> -MODULE_AUTHOR("Jaromir Koutek <miri(a)punknet.cz>"); +MODULE_AUTHOR("Jaromir Koutek"); MODULE_DESCRIPTION("ESS Solo-1"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ESS,ES1938}," diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 6dc578b..02256f2 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -42,7 +42,7 @@ #define TEA575X_RADIO 1 #endif -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("ForteMedia FM801"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ForteMedia,FM801}," diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8f34fb4..48d05e6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -34,7 +34,7 @@ #include "hda_local.h" -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("Universal interface for High Definition Audio Codec"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/ice1712/ak4xxx.c b/sound/pci/ice1712/ak4xxx.c index ab00cce..fab0813 100644 --- a/sound/pci/ice1712/ak4xxx.c +++ b/sound/pci/ice1712/ak4xxx.c @@ -30,7 +30,7 @@ #include <sound/initval.h> #include "ice1712.h" -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("ICEnsemble ICE17xx <-> AK4xxx AD/DA chip interface"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 830a1bb..2d62c5e 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -73,7 +73,7 @@ #include "ews.h" #include "hoontech.h" -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("ICEnsemble ICE1712 (Envy24)"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{" diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 1127ebd..aeff346 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -52,7 +52,7 @@ #include "phase.h" #include "wtm.h" -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("VIA ICEnsemble ICE1724/1720 (Envy24HT/PT)"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{" diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 7cf2dcb..ba95dc2 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -43,7 +43,7 @@ #include <asm/pgtable.h> #include <asm/cacheflush.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH}," diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index c155e1f..8736c02 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -37,7 +37,7 @@ #include <sound/info.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; " "SiS 7013; NVidia MCP/2/2S/3 modems"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 21d0899..9b98090 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -429,7 +429,7 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for Korg 1212 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Korg 1212 soundcard."); -MODULE_AUTHOR("Haroldo Gamal <gamal(a)alternex.com.br>"); +MODULE_AUTHOR("Haroldo Gamal"); static struct pci_device_id snd_korg1212_ids[] = { { diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 4526904..0f9f218 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -51,7 +51,7 @@ #include <sound/initval.h> #include <asm/byteorder.h> -MODULE_AUTHOR("Zach Brown <zab(a)zabbo.net>, Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Zach Brown, Takashi Iwai"); MODULE_DESCRIPTION("ESS Maestro3 PCI"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ESS,Maestro3 PCI}," diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index ac007ce..c2ebdd9 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -42,7 +42,7 @@ #define CARD_NAME "miXart" -MODULE_AUTHOR("Digigram <alsa(a)digigram.com>"); +MODULE_AUTHOR("Digigram"); MODULE_DESCRIPTION("Digigram " CARD_NAME); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Digigram," CARD_NAME "}}"); diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 03b3a47..a47a5e8 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -44,7 +44,7 @@ #define CARD_NAME "NeoMagic 256AV/ZX" #define DRIVER_NAME "NM256" -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("NeoMagic NM256AV/ZX"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{NeoMagic,NM256AV}," diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index d974134..cd29197 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -44,7 +44,7 @@ #define DRIVER_NAME "pcxhr" -MODULE_AUTHOR("Markus Bollinger <bollinger(a)digigram.com>"); +MODULE_AUTHOR("Markus Bollinger"); MODULE_DESCRIPTION("Digigram " DRIVER_NAME " " PCXHR_DRIVER_VERSION_STRING); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Digigram," DRIVER_NAME "}}"); diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 952625d..6854b62 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -113,7 +113,7 @@ #define SUPPORT_JOYSTICK 1 #endif -MODULE_AUTHOR("Peter Gruber <nokos(a)gmx.net>"); +MODULE_AUTHOR("Peter Gruber"); MODULE_DESCRIPTION("riptide"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Conexant,Riptide}}"); diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 6bb7ac6..19fb89a 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -101,7 +101,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable RME Digi32 soundcard."); module_param_array(fullduplex, bool, NULL, 0444); MODULE_PARM_DESC(fullduplex, "Support full-duplex mode."); -MODULE_AUTHOR("Martin Langer <martin-langer(a)gmx.de>, Pilo Chambert <pilo.c(a)wanadoo.fr>"); +MODULE_AUTHOR("Martin Langer, Pilo Chambert"); MODULE_DESCRIPTION("RME Digi32, Digi32/8, Digi32 PRO"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{RME,Digi32}," "{RME,Digi32/8}," "{RME,Digi32 PRO}}"); diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index e3304b7..aed6e67 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -43,7 +43,7 @@ /* note, two last pcis should be equal, it is not a bug */ -MODULE_AUTHOR("Anders Torger <torger(a)ludd.luth.se>"); +MODULE_AUTHOR("Anders Torger"); MODULE_DESCRIPTION("RME Digi96, Digi96/8, Digi96/8 PRO, Digi96/8 PST, " "Digi96/8 PAD"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 89b3c7f..70ce246 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -54,7 +54,7 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for RME Hammerfall DSP interface."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable/disable specific Hammerfall DSP soundcards."); -MODULE_AUTHOR("Paul Davis <paul(a)linuxaudiosystems.com>, Marcus Andersson, Thomas Charbonnel <thomas(a)undata.org>"); +MODULE_AUTHOR("Paul Davis, Marcus Andersson, Thomas Charbonnel"); MODULE_DESCRIPTION("RME Hammerfall DSP"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{RME Hammerfall-DSP}," diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index cc3bdec..8235870 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -51,7 +51,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable/disable specific RME96{52,36} soundcards."); module_param_array(precise_ptr, bool, NULL, 0444); MODULE_PARM_DESC(precise_ptr, "Enable precise pointer (doesn't work reliably)."); -MODULE_AUTHOR("Paul Davis <pbd(a)op.net>, Winfried Ritsch"); +MODULE_AUTHOR("Paul Davis, Winfried Ritsch"); MODULE_DESCRIPTION("RME Digi9652/Digi9636"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{RME,Hammerfall}," diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 9f25d93..1fff81b 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -42,7 +42,7 @@ #include <asm/io.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("S3 SonicVibes PCI"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{S3,SonicVibes PCI}}"); diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 9145f7c..f6019a4 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -30,7 +30,7 @@ #include <sound/trident.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, <audio(a)tridentmicro.com>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Trident 4D-WaveDX/NX & SiS SI7018"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Trident,4DWave DX}," diff --git a/sound/pci/trident/trident_synth.c b/sound/pci/trident/trident_synth.c index 9b7dee8..e29553e 100644 --- a/sound/pci/trident/trident_synth.c +++ b/sound/pci/trident/trident_synth.c @@ -28,7 +28,7 @@ #include <sound/trident.h> #include <sound/seq_device.h> -MODULE_AUTHOR("Scott McNab <jedi(a)tartarus.uwa.edu.au>"); +MODULE_AUTHOR("Scott McNab"); MODULE_DESCRIPTION("Routines for Trident 4DWave NX/DX soundcards - Synthesizer"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index a289922..e905bd3 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -68,7 +68,7 @@ #define POINTER_DEBUG #endif -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("VIA VT82xx audio"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{VIA,VT82C686A/B/C,pci},{VIA,VT8233A/C,8235}}"); diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index b338e15..17a2cb0 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -50,7 +50,7 @@ #define POINTER_DEBUG #endif -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("VIA VT82xx modem"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{VIA,VT82C686A/B/C modem,pci}}"); diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index 474eac9..7aedb2f 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -31,7 +31,7 @@ #define CARD_NAME "VX222" -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("Digigram VX222 V2/Mic"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Digigram," CARD_NAME "}}"); diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index fd9b7b8..73e204f 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -30,7 +30,7 @@ #include <sound/opl3.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Yamaha DS-1 PCI"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Yamaha,YMF724}," diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 2d40cc7..3176554 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -33,7 +33,7 @@ #define CARD_NAME "PDAudio-CF" -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Sound Core " CARD_NAME); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Sound Core," CARD_NAME "}}"); diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 363bcb5..7d58d16 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -32,7 +32,7 @@ /* */ -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("Digigram VXPocket"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Digigram,VXPocket},{Digigram,VXPocket440}}"); diff --git a/sound/soc/at91/at91-i2s.c b/sound/soc/at91/at91-i2s.c index 9fc0c03..3b33274 100644 --- a/sound/soc/at91/at91-i2s.c +++ b/sound/soc/at91/at91-i2s.c @@ -716,6 +716,6 @@ struct snd_soc_cpu_dai at91_i2s_dai[NUM_SSC_DEVICES] = { EXPORT_SYMBOL_GPL(at91_i2s_dai); /* Module information */ -MODULE_AUTHOR("Frank Mandarino, fmandarino(a)endrelia.com, www.endrelia.com"); +MODULE_AUTHOR("Frank Mandarino"); MODULE_DESCRIPTION("AT91 I2S ASoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c index b39b95a..f548391 100644 --- a/sound/soc/at91/at91-pcm.c +++ b/sound/soc/at91/at91-pcm.c @@ -427,6 +427,6 @@ struct snd_soc_platform at91_soc_platform = { EXPORT_SYMBOL_GPL(at91_soc_platform); -MODULE_AUTHOR("Frank Mandarino <fmandarino(a)endrelia.com>"); +MODULE_AUTHOR("Frank Mandarino"); MODULE_DESCRIPTION("Atmel AT91 PCM module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 8179df3..58bcbb8 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -370,6 +370,6 @@ module_init(eti_b1_init); module_exit(eti_b1_exit); /* Module information */ -MODULE_AUTHOR("Frank Mandarino <fmandarino(a)endrelia.com>"); +MODULE_AUTHOR("Frank Mandarino"); MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 50c5c83..3da33b6 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -313,6 +313,6 @@ struct snd_soc_cpu_dai pxa_i2s_dai = { EXPORT_SYMBOL_GPL(pxa_i2s_dai); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood"); MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 36519ae..49be095 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1582,6 +1582,6 @@ module_init(snd_soc_init); module_exit(snd_soc_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood"); MODULE_DESCRIPTION("ALSA SoC Core"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7caf8c7..303846a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1318,6 +1318,6 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) EXPORT_SYMBOL_GPL(snd_soc_dapm_free); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); MODULE_LICENSE("GPL"); diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index b6d8863..21ec11c 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -57,7 +57,7 @@ #include "usbaudio.h" -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("USB Audio"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Generic,USB Audio}}"); diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 24f5a26..b0d304e 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -64,7 +64,7 @@ #define ERROR_DELAY_JIFFIES (HZ / 10) -MODULE_AUTHOR("Clemens Ladisch <clemens(a)ladisch.de>"); +MODULE_AUTHOR("Clemens Ladisch"); MODULE_DESCRIPTION("USB Audio/MIDI helper module"); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index e011fca..926d652 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -147,7 +147,7 @@ -MODULE_AUTHOR("Karsten Wiese <annabellesgarden(a)yahoo.de>"); +MODULE_AUTHOR("Karsten Wiese"); MODULE_DESCRIPTION("TASCAM "NAME_ALLCAPS" Version 0.8.7.2"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{TASCAM(0x1604), "NAME_ALLCAPS"(0x8001)(0x8005)(0x8007) }}");

3 6

Re: [alsa-devel] Any standard ALSA application for simultaneous capture and playabck ?????
by Ciaccia 22 May '07

22 May '07

It looks like your application uses the following pattern: for (;;) { ReadFromAlsa WriteToAlsa } The ReadFormAlsa function waits for the capture channel to become available (i.e., there is some data to be read), while WriteToAlsa waits till the output buffer becomes available (i.e., it's possible to send data to the card without blocking). Ho do you know that this is the correct order? Could it be that you should before write data to the output buffer and then read data from the capture stream? I think your design is not completely correct... Look at my code, it works fine, but maybe there is a better approach... Andrea ----- Original Message ---- From: Nobin Mathew <nobin.mathew(a)gmail.com> To: Ciaccia <ciacciax(a)yahoo.com> Cc: alsa-devel(a)alsa-project.org Sent: Tuesday, May 22, 2007 10:53:54 AM Subject: Re: [alsa-devel] Any standard ALSA application for simultaneous capture and playabck ????? I have a single threaded ALSA application which does simultaneous capture and playback. But i am getting overflow error XRUN from capture side. I am attaching the code to this mail. Is there any problem with single threaded approach? Is there any global configuration for ALSA lib? When i run this app, ALSA state is getting corrupted, even aplay and arecord is not working properly after that. Any comments on this issue? Thanks in advance Nobin Mathew On 5/22/07, Ciaccia <ciacciax(a)yahoo.com> wrote: > > What are the complications in implementing simultaneous playback and > > capture in ALSA??? > > > I would like to add something to this post: what is the "best" way to > program a full duplex application? I would like to have an application > that does full-duplex, with the lowest possible latency and using less > CPU resource as possible. I tried different approaches, but I was not > completely satisfied with none of them.... > > > Here is what I tried: > -having two threads, one for playback and one for capture > -defining 2 callbacks with async_handles and do all the processing in the callback functions > (the last attempt) > -using poll() to wait for one of the streams to be ready (see attached file fullduplex.c) > > Could some of you look at the attached code (it is not generic and contains lots of asserts, but it's supposed to work only on one single machine) and tell me if the last approach using poll is well structured? > > The main loop looks as following: > > > while (1) { > > err = poll(ufds, 2, -1); > > > > for (i = 0; i < 2; i++) > > { > > if (snd_pcm_poll_descriptors_revents(handles[i], &ufds[i], 1, &revents) < 0) > > { > > printf("Error getting revents for %s\n", descriptors[i].name); > > exit(1); > > } > > > > if (revents & descriptors[i].poll_flag) > > { > > if (transfer_loop(handles[i], &first[i], descriptors[i].func) < 0) > > { > > printf("transfer_loop error for %s\n", descriptors[i].name); > > // TODO exit?!? > > } > > } > > } > > } > > and the transfer_loop is a modified version of the loop found in the ALSA examples (pcm.c and latency.c). > Is this strategy optimal in respect to low latency and low CPU usage? > > Every comment is greatly appreciated > Andrea > > > > > > ____________________________________________________________________________________Luggage? GPS? Comic books? > Check out fitting gifts for grads at Yahoo! Search > http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz > ____________________________________________________________________________________ Need Mail bonding? Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. http://answers.yahoo.com/dir/?link=list&sid=396546091

1 0

Re: [alsa-devel] Any standard ALSA application for simultaneous capture and playabck ?????
by Ciaccia 22 May '07

22 May '07

> What are the complications in implementing simultaneous playback and > capture in ALSA??? I would like to add something to this post: what is the "best" way to program a full duplex application? I would like to have an application that does full-duplex, with the lowest possible latency and using less CPU resource as possible. I tried different approaches, but I was not completely satisfied with none of them.... Here is what I tried: -having two threads, one for playback and one for capture -defining 2 callbacks with async_handles and do all the processing in the callback functions (the last attempt) -using poll() to wait for one of the streams to be ready (see attached file fullduplex.c) Could some of you look at the attached code (it is not generic and contains lots of asserts, but it's supposed to work only on one single machine) and tell me if the last approach using poll is well structured? The main loop looks as following: > while (1) { > err = poll(ufds, 2, -1); > > for (i = 0; i < 2; i++) > { > if (snd_pcm_poll_descriptors_revents(handles[i], &ufds[i], 1, &revents) < 0) > { > printf("Error getting revents for %s\n", descriptors[i].name); > exit(1); > } > > if (revents & descriptors[i].poll_flag) > { > if (transfer_loop(handles[i], &first[i], descriptors[i].func) < 0) > { > printf("transfer_loop error for %s\n", descriptors[i].name); > // TODO exit?!? > } > } > } > } and the transfer_loop is a modified version of the loop found in the ALSA examples (pcm.c and latency.c). Is this strategy optimal in respect to low latency and low CPU usage? Every comment is greatly appreciated Andrea ____________________________________________________________________________________Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz

2 1

[alsa-devel] Firmware to add - Dreamcast AICA
by Adrian McMenamin 22 May '07

22 May '07

Please add this to the firmware repo

2 4

[alsa-devel] Question on buffer size setting
by Allyn, Mark A 21 May '07

21 May '07

Hello: I am attempting to do some debugging of a new audio device, and I need to change the number of buffer descriptor list entries from the default of 16 to something much smaller, like 3. I am a bit confused while looking at the code. There seems to be a value labeled SNDRV_PCM_HW_PARAM_BUFFER_SIZE, but I can't figure how it is set. Is this a parameter that is passed as part of the configure prior to doing a make for the driver? Is this a parameter passed to modprobe when loading the driver? Thanks for your help. Mark

2 1

[alsa-devel] [PATCH] hda-intel: fix ASUS M2V detection
by Daniel Drake 21 May '07

21 May '07

Commit f32610edab47f36946d23b883aeae91e15986121 added ALC660VD support, but this caused a 2.6.21 regression for some users. The ASUS M2V device is now detected as ALC660VD rather than ALC660/861 but the PCI quirk was not carried over. This patch allows affected users to use audio again. http://bugzilla.kernel.org/show_bug.cgi?id=8273 https://bugs.gentoo.org/show_bug.cgi?id=178243 Signed-off-by: Daniel Drake <dsd(a)gentoo.org> Index: linux/sound/pci/hda/patch_realtek.c =================================================================== --- linux.orig/sound/pci/hda/patch_realtek.c +++ linux/sound/pci/hda/patch_realtek.c @@ -8384,7 +8384,6 @@ static struct snd_pci_quirk alc861_cfg_t SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS), - SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660_3ST), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31), @@ -8990,6 +8989,7 @@ static const char *alc861vd_models[ALC86 static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), + SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),

2 1

Re: [alsa-devel] porter wanted: AMD Geode SCx200 driver OSS to ALSA
by Ash Willis 20 May '07

20 May '07

Sorry for this really late reply, I just wanted to add a note: > > (Would now also be a good time to ask for snd-sc5535audio to be > > renamed snd-sc5535 for consistence with other driver names?) > > I agree with you, I think it should be renamed. I've just ported the SND_KAHLUA driver from OSS (patch not released yet). It covers the cs5530 (VSA1 only). I've named it snd-cs5530. This audio suffix seems a bit wacky to me. Surely the snd prefix is adequate to distinguish it from an IDE driver, for example :) Ash = Nose Surgery Specialist in Manhattan, NY Rhinoplasty surgeon Geoffrey W. Tobias has pioneered the "un-nose job," the nose job that doesn't look like one. Chosen by "New York Magazine" as a nose expert. Video clips. http://a8-asy.a8ww.net/a8-ads/adftrclick?redirectid=1b57edb21974e0b43470c0c…

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[alsa-devel] Strange findings about the microphone in SigmaTel STAC9221 A1
by Ivan N. Zlatev 20 May '07

20 May '07

My hda-intel card is with the following codec: Codec: SigmaTel STAC9221 A1 Address: 0 Vendor Id: 0x83847680 Subsystem Id: 0x106b1700 Revision Id: 0x103401 I am using alsa 1.0.14rc4. I have a built-in microphone, which doesn't work, but today I found out something interesting. If I set the capture to minimum and place Mux at specific positions (Works with middle-ish and top, not bottom-ish) suddenly the mic started recording, but the quality is extremly bad. I have attached amixer and co. Does anyone have a clue what the issue is and how could possibly this little bit of random foundings to be used to possibly fix the mic? Regards. -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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[alsa-devel] ICE1712/1724 HW mixing?
by Rene Herman 19 May '07

19 May '07

Good day. I've been updating the TerraTec vendor page in the new wiki matrix: http://bugtrack.alsa-project.org/main/index.php/Matrix:Vendor-Terratec The problem with information that looks maintained though is that when something's _not_ mentioned, it might be interpreted to mean it's not actually working.... I have the older TerraTec cards (both ISA and PCI) but not any of the ICE1712/ICE1724 ones. Could someone verify/confirm that the ICE1712 (Envy24?) does not support 24-bit samples? And that ICE1724 supports only 32-bit in hardware? Does either chip/driver support hardware mixing? Filling out some of the Digital I/O tags for any Aureon card that you know would be nice as well. Rene.

3 5

[alsa-devel] I don't understand snd_pcm_ops
by Timur Tabi 19 May '07

19 May '07

I'm new to ALSA driver development, and I'm writing an SOC driver for new hardware. Can someone explain to me what the snd_pcm_ops structure is for? And it sounds like a stupid question, but after studying documentation and sample code, I'm still confused. First off, I thought PCM stood for Pulse Code Modulation, but that just describes the format of the digital data. I don't understand what a PCM operation could be. I've seen documentation where PCM is considered an alternative to I2S or AC97. But if you look at eti_b1_wm8731.c, you'll see this: #include "at91-pcm.h" #include "at91-i2s.h" So is the AT91 a PCM device or an I2S device??? Some background: I'm writing driver for a new SOC that has an I2S interface, and it's talking to a CS4270 codec, but the CS4270 codec is wired up in "stand-alone" mode, so there's no software configuration. -- Timur Tabi Linux Kernel Developer @ Freescale

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Re: [alsa-devel] Alsa 1.0.13 (and others) does not work on ALC880
by Eric De Vito 19 May '07

19 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear Tobin thank you for your so helpful recommendations ! In fact the key was probe_mask=1. But I did not see this option in the hda-intel module description :-( ... Now I added probe_mask)=1 in /etc/modprobe.d/alsa-base and it works either with 1.0.13 or 14rc4 :-). In fact when alsa is compiled with debug=detect this option is not necessary any more and sound works even without this option ... Does this option (probe_mask=1) limitates something in the sound system ? Is it better to not use it or is it painless for audio possibilities ? Thank you again for your help !!! Now to be more precise : 1) If I compile 14rc4 with debug=detect it can be compiled, modules can be inserted : - --> when inserted with # modprobe -v snd-hda-intel model=clevo I get in /var/log/messages : May 18 22:32:05 gondor kernel: [ 6012.711362] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 17 (level, low) -> IRQ 20 May 18 22:32:05 gondor kernel: [ 6012.774627] ALSA /usr/src/modules/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0x3 May 18 22:32:06 gondor kernel: [ 6013.404197] ALSA /usr/src/modules/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:1795: hda_codec: model 'clevo' is selected May 18 22:32:08 gondor kernel: [ 6015.083303] ALSA /usr/src/modules/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:558: hda_intel: azx_get_response timeout, switching to polling mode... - -> when inserted with # modprobe -v snd-hda-intel model=clevo probe_mask=1 I get in /var/log/messages : May 18 22:32:50 gondor kernel: [ 6057.162063] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 17 (level, low) -> IRQ 20 May 18 22:32:50 gondor kernel: [ 6057.243574] ALSA /usr/src/modules/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0x3 May 18 22:32:50 gondor kernel: [ 6057.873098] ALSA /usr/src/modules/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:1795: hda_codec: model 'clevo' is selected 2) If I compile 14rc4 with debug=none it can be compiled, modules can be inserted : - --> when inserted with # modprobe -v snd-hda-intel model=clevo I get in /var/log/messages : May 18 19:39:46 gondor kernel: [ 2366.086099] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 17 (level, low) -> IRQ 20 May 18 19:39:49 gondor kernel: [ 2368.491630] ALSA /usr/src/modules/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:558: hda_intel: azx_get_response timeout, switching to polling mode... and aplay gives : eric@gondor:/usr/src/modules/alsa-driver-1.0.14rc4$ aplay ALSA lib pcm_direct.c:867:(snd_pcm_direct_initialize_slave) snd_pcm_hw_params_any failed ALSA lib pcm_dmix.c:876:(snd_pcm_dmix_open) unable to initialize slave aplay: main:550: Erreur d'ouverture audio: Argument invalide eric@gondor:/usr/src/modules/alsa-driver-1.0.14rc4$ - --> when inserted with # modprobe -v snd-hda-intel model=clevo probe_mask=1 May 18 19:41:45 gondor kernel: [ 2484.283312] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 17 (level, low) -> IRQ 20 And it works !!!! Eric Tobin Davis a écrit : > Ok, you have 14rc4 (according to your email). Try loading the driver > with "model=clevo", as it appears that is the system type you have. > Make sure the drivers are built with "--with-debug=detect" so that we > can get good dmesg output. You shouldn't need any other parameters. > The clevo model was added not too long ago. The other model to try is > "model=test". This will give you control over all nodes and functions. > > You may also need to add "probe_mask=1" (or something like that) to > limit the number of codecs being detected. I don't know why > cat /proc/asound/card0/codec#0 would hang. > > Tobin > > On Mon, 2007-05-14 at 14:56 +0200, Eric wrote: > >> I installed Ubuntu Feisty on a D900t laptop : it contains a ALC880 audio >> chipset >> >> eric@gondor:~$ lspci |grep Audio >> 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 >> Family) High Definition Audio Controller (rev 03) >> >> When modprobing the modules I get : >> eric@gondor:~$ sudo modprobe -v snd-hda-intel model=3stack-digout >> insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-codec.ko >> insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-intel.ko model=3stack-digout >> >> while /var/log/messages says : >> May 14 14:42:45 gondor kernel: [ 3699.070904] hda_intel: >> azx_get_response timeout, switching to polling mode... >> >> >> Feisty dist. is provided with alsa-1.0.13. I get no sound, here are the >> results of aplay : >> >> eric@gondor:~$ aplay >> ALSA lib pcm_direct.c:867:(snd_pcm_direct_initialize_slave) >> snd_pcm_hw_params_any failed >> ALSA lib pcm_dmix.c:876:(snd_pcm_dmix_open) unable to initialize slave >> aplay: main:550: Erreur d'ouverture audio: Argument invalide >> eric@gondor:~$ >> >> Here are some other infos : >> >> eric@gondor:~$ tail -2 /proc/asound/oss/sndstat >> 0: Realtek ALC880 >> 1: SAA7134 Mixer >> >> amixer shows this : >> eric@gondor:~$ amixer >> Simple mixer control 'Headphone',0 >> Capabilities: pswitch >> Playback channels: Front Left - Front Right >> Mono: >> Front Left: Playback [on] >> Front Right: Playback [on] >> Simple mixer control 'PCM',0 >> Capabilities: pvolume >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 255 >> Mono: >> Front Left: Playback 228 [89%] >> Front Right: Playback 228 [89%] >> Simple mixer control 'Front',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 60 [47%] [-2144.00dB] [on] >> Front Right: Playback 60 [47%] [-2144.00dB] [on] >> Simple mixer control 'Front Mic',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 55 [43%] [-2304.00dB] [on] >> Front Right: Playback 55 [43%] [-2304.00dB] [on] >> Simple mixer control 'Surround',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 64 [50%] [-2016.00dB] [on] >> Front Right: Playback 64 [50%] [-2016.00dB] [on] >> Simple mixer control 'Center',0 >> Capabilities: pvolume pvolume-joined pswitch pswitch-joined >> Playback channels: Mono >> Limits: Playback 0 - 127 >> Mono: Playback 56 [44%] [-2272.00dB] [on] >> Simple mixer control 'LFE',0 >> Capabilities: pvolume pvolume-joined pswitch pswitch-joined >> Playback channels: Mono >> Limits: Playback 0 - 127 >> Mono: Playback 58 [46%] [-2208.00dB] [on] >> Simple mixer control 'Line',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 60 [47%] [-2144.00dB] [on] >> Front Right: Playback 60 [47%] [-2144.00dB] [on] >> Simple mixer control 'CD',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 58 [46%] [-2208.00dB] [on] >> Front Right: Playback 58 [46%] [-2208.00dB] [on] >> Simple mixer control 'Mic',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 57 [45%] [-2240.00dB] [on] >> Front Right: Playback 57 [45%] [-2240.00dB] [on] >> Simple mixer control 'IEC958',0 >> Capabilities: pswitch pswitch-joined >> Playback channels: Mono >> Mono: Playback [off] >> Simple mixer control 'PC Speaker',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 53 [42%] [-2368.00dB] [on] >> Front Right: Playback 53 [42%] [-2368.00dB] [on] >> Simple mixer control 'Capture',0 >> Capabilities: cvolume cswitch >> Capture channels: Front Left - Front Right >> Limits: Capture 0 - 127 >> Front Left: Capture 31 [24%] [-3072.00dB] [on] >> Front Right: Capture 31 [24%] [-3072.00dB] [on] >> Simple mixer control 'Capture',1 >> Capabilities: cvolume cswitch >> Capture channels: Front Left - Front Right >> Limits: Capture 0 - 127 >> Front Left: Capture 27 [21%] [-3200.00dB] [on] >> Front Right: Capture 27 [21%] [-3200.00dB] [on] >> Simple mixer control 'Capture',2 >> Capabilities: cvolume cswitch >> Capture channels: Front Left - Front Right >> Limits: Capture 0 - 127 >> Front Left: Capture 24 [19%] [-3296.00dB] [on] >> Front Right: Capture 24 [19%] [-3296.00dB] [on] >> Simple mixer control 'Caller ID',0 >> Capabilities: pswitch pswitch-joined >> Playback channels: Mono >> Mono: Playback [on] >> Simple mixer control 'Channel Mode',0 >> Capabilities: enum >> Items: '2ch' '6ch' >> Item0: '6ch' >> Simple mixer control 'Input Source',0 >> Capabilities: enum >> Items: 'Mic' 'Front Mic' 'Line' 'CD' >> Item0: 'Mic' >> Simple mixer control 'Input Source',1 >> Capabilities: enum >> Items: 'Mic' 'Front Mic' 'Line' 'CD' >> Item0: 'Mic' >> Simple mixer control 'Input Source',2 >> Capabilities: enum >> Items: 'Mic' 'Front Mic' 'Line' 'CD' >> Item0: 'Mic' >> Simple mixer control 'Off-hook',0 >> Capabilities: pswitch pswitch-joined >> Playback channels: Mono >> Mono: Playback [on] >> >> aplay -l seems to work : >> >> eric@gondor:~$ aplay -l >> **** Liste des PLAYBACK périphériques **** >> carte 0: Intel [HDA Intel], périphérique 0 : ALC880 Analog [ALC880 >> Analog] >> Sous-périphériques: 1/1 >> Sous-périphérique: #0: subdevice #0 >> carte 0: Intel [HDA Intel], périphérique 1 : ALC880 Digital [ALC880 >> Digital] >> Sous-périphériques: 1/1 >> Sous-périphérique: #0: subdevice #0 >> carte 0: Intel [HDA Intel], périphérique 6 : Si3054 Modem [Si3054 >> Modem] >> Sous-périphériques: 1/1 >> Sous-périphérique: #0: subdevice #0 >> >> >> eric@gondor:~$ cat /proc/asound/cards >> 0 [Intel ]: HDA-Intel - HDA Intel >> HDA Intel at 0xb0000000 irq 17 >> 1 [SAA7134 ]: SAA7134 - SAA7134 >> saa7133[0] at 0xb3006000 irq 22 >> >> if I try to do a cat /proc/asound/card0/codec#0 it hangs >> while /var/log/messages gives a lot of >> >> May 14 14:44:52 gondor kernel: [ 3825.457412] hda_codec: invalid >> dep_range_val 0:7fff >> May 14 14:44:52 gondor kernel: [ 3825.457696] hda_codec: invalid >> dep_range_val 0:7fff >> ........................ >> >> eric@gondor:~$ sudo lspci -nv >> 00:00.0 0600: 8086:2580 (rev 04) >> Subsystem: 1558:0900 >> Flags: bus master, fast devsel, latency 0 >> Capabilities: [e0] Vendor Specific Information >> >> 00:01.0 0604: 8086:2581 (rev 04) (prog-if 00 [Normal decode]) >> Flags: bus master, fast devsel, latency 0 >> Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 >> Memory behind bridge: b1000000-b2ffffff >> Prefetchable memory behind bridge: c0000000-cfffffff >> Capabilities: [88] Subsystem: 8086:2582 >> Capabilities: [80] Power Management version 2 >> Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- >> Queue=0/0 Enable- >> Capabilities: [a0] Express Root Port (Slot+) IRQ 0 >> >> 00:1b.0 0403: 8086:2668 (rev 03) >> Subsystem: 1558:0900 >> Flags: bus master, fast devsel, latency 0, IRQ 20 >> Memory at b0000000 (64-bit, non-prefetchable) [size=16K] >> Capabilities: [50] Power Management version 2 >> Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ >> Queue=0/0 Enable- >> Capabilities: [70] Express Unknown type IRQ 0 >> >> 00:1d.0 0c03: 8086:2658 (rev 03) (prog-if 00 [UHCI]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0, IRQ 18 >> I/O ports at 1800 [size=32] >> >> 00:1d.1 0c03: 8086:2659 (rev 03) (prog-if 00 [UHCI]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0, IRQ 19 >> I/O ports at 3000 [size=32] >> >> 00:1d.2 0c03: 8086:265a (rev 03) (prog-if 00 [UHCI]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0, IRQ 17 >> I/O ports at 3020 [size=32] >> >> 00:1d.3 0c03: 8086:265b (rev 03) (prog-if 00 [UHCI]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0, IRQ 20 >> I/O ports at 3040 [size=32] >> >> 00:1d.7 0c03: 8086:265c (rev 03) (prog-if 20 [EHCI]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0, IRQ 18 >> Memory at b0004000 (32-bit, non-prefetchable) [size=1K] >> Capabilities: [50] Power Management version 2 >> Capabilities: [58] Debug port >> >> 00:1e.0 0604: 8086:244e (rev d3) (prog-if 01 [Subtractive decode]) >> Flags: bus master, fast devsel, latency 0 >> Bus: primary=00, secondary=0a, subordinate=0e, sec-latency=32 >> I/O behind bridge: 00004000-00004fff >> Memory behind bridge: b3000000-b30fffff >> Prefetchable memory behind bridge: >> 0000000050000000-0000000055ffffff >> Capabilities: [50] Subsystem: 1558:0900 >> >> 00:1f.0 0601: 8086:2640 (rev 03) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0 >> >> 00:1f.1 0101: 8086:266f (rev 03) (prog-if 8a [Master SecP PriP]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0, IRQ 17 >> I/O ports at 01f0 [size=8] >> I/O ports at 03f4 [size=1] >> I/O ports at 0170 [size=8] >> I/O ports at 0374 [size=1] >> I/O ports at 3080 [size=16] >> >> 00:1f.3 0c05: 8086:266a (rev 03) >> Subsystem: 1558:0900 >> Flags: medium devsel, IRQ 10 >> I/O ports at 3060 [size=32] >> >> 01:00.0 0300: 10de:00c8 (rev a2) (prog-if 00 [VGA]) >> Subsystem: 1558:0900 >> Flags: bus master, fast devsel, latency 0, IRQ 11 >> Memory at b2000000 (32-bit, non-prefetchable) [size=16M] >> Memory at c0000000 (64-bit, prefetchable) [size=256M] >> Memory at b1000000 (64-bit, non-prefetchable) [size=16M] >> Capabilities: [60] Power Management version 2 >> Capabilities: [68] Message Signalled Interrupts: Mask- 64bit+ >> Queue=0/0 Enable- >> Capabilities: [78] Express Endpoint IRQ 0 >> >> 0a:00.0 0607: 104c:ac50 (rev 02) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 168, IRQ 17 >> Memory at b3007000 (32-bit, non-prefetchable) [size=4K] >> Bus: primary=0a, secondary=0b, subordinate=0e, sec-latency=176 >> Memory window 0: 50000000-53fff000 (prefetchable) >> Memory window 1: 58000000-5bfff000 >> I/O window 0: 00004800-000048ff >> I/O window 1: 00004c00-00004cff >> 16-bit legacy interface ports at 0001 >> >> 0a:01.0 0c00: 104c:8023 (prog-if 10 [OHCI]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 64, IRQ 21 >> Memory at b3004000 (32-bit, non-prefetchable) [size=2K] >> Memory at b3000000 (32-bit, non-prefetchable) [size=16K] >> Capabilities: [44] Power Management version 2 >> >> 0a:02.0 0104: 105a:3373 (rev 02) >> Subsystem: 1558:0900 >> Flags: bus master, 66MHz, medium devsel, latency 96, IRQ 19 >> I/O ports at 4480 [size=64] >> I/O ports at 44c0 [size=16] >> I/O ports at 4400 [size=128] >> Memory at b3005000 (32-bit, non-prefetchable) [size=4K] >> Memory at b3020000 (32-bit, non-prefetchable) [size=128K] >> Capabilities: [60] Power Management version 2 >> >> 0a:03.0 0200: 10ec:8169 (rev 10) >> Subsystem: 1558:0900 >> Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 22 >> I/O ports at 4000 [size=256] >> Memory at b3004800 (32-bit, non-prefetchable) [size=256] >> [virtual] Expansion ROM at 54000000 [disabled] [size=128K] >> Capabilities: [dc] Power Management version 2 >> >> 0a:04.0 0480: 1131:7133 (rev f0) >> Subsystem: 1461:f31e >> Flags: bus master, medium devsel, latency 64, IRQ 23 >> Memory at b3006000 (32-bit, non-prefetchable) [size=2K] >> Capabilities: [40] Power Management version 2 >> >> 0a:05.0 0280: 1814:0201 (rev 01) >> Subsystem: 1462:6833 >> Flags: bus master, slow devsel, latency 64, IRQ 20 >> Memory at b3008000 (32-bit, non-prefetchable) [size=8K] >> Capabilities: [40] Power Management version 2 >> >> asoundconf output : >> eric@gondor:~$ asoundconf list >> Names of available sound cards: >> Intel >> SAA7134 >> >> I tried a lot of workarounds but none worked : boot options (irqpoll, >> pci=noacpi), single_cmd=1, position_fix=1, several models of hda-intel >> driver, I tried 1.0.13 and 1.0.14rc4 driver, with debug on... >> >> I must add that I already had this problem with previous distributions >> of Ubuntu. It worked with an old one (Breezy) then in Dapper and Edgy, >> it did worked ONLY when I self-compiled the driver with debug option >> enabled I never understood why. And now with Feisty/alsa 1.0.13/ubuntu >> kernel 2.6.20, or an ubuntu kernel with self-compiled alsa 1.0.14rc4, it >> does not work. I cannot have one sound :-(. >> >> Please help me .... >> >> >> Eric >> >> >> >> >> >> >> >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGThYExI+RZbxK37ARAl6lAKDOecNscFCyIk5LSG2J1Tu8thsxiACgvSTX zk0O2gqEl6t/bFmi76jztqw= =LN4P -----END PGP SIGNATURE-----

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[alsa-devel] Linux 2.6.21 regression with hda-intel (Conexant 5045)
by Gunter Ohrner 18 May '07

18 May '07

Hi! First of all: Sorry if this question is dumb or a fix should be obvious, but as a non-ALSA_expert I'm a bit a a loss at the moment... I searched the bugtracker and this list but could not find anything appropriate - most probably I missed a relevant post or didn't recognize it as being relevant... Any pointers are greatly appreciated! The situation is as follows: Using Linux 2.6.20.4, my laptop's soundcard works just fine. Using 2.6.21.1, it also works, but the headphone socket is dead. Plugging a headphone into it correctly mustes the internal speakers, but no sound is output via the headphones. Details: - Fujitsu Simens Amilo Si1520 - Codec: Conexant ID 5045 (as reported by Linux 2.6.20.4) Alsa device details between both Linux kernels differ completely, with Linux 2.6.20.4 only reporting a single a single analog PCM device and Linux 2.6.21.1 reporting a Conexant Analog and a Conexant Digital device. Should I still post the complete information from /proc/asound/* for both kernels? I tried applying conexant-test22.patch by Tobin Davis, but it does not seem to match the in-kernel ALSA code. It also does not seem to be related to the 5045 chip... Any ideas? Greetings, Gunter

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Re: [alsa-devel] [PATCH] SM56 Modem in Asus A8J Notebook Series
by Christian Rothländer 18 May '07

18 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 > > It's a process to merge to the upstream kernel. We need to give a > > kind of approval from the patch authors and contributors. > > It consists of a line ok .. something new for me :-) Here it goes: Signed-off-by: Christian Rothlaender <christian(a)rothlaender.net> ================================================================ hda-codec - patch_si3045.c This patch adds support for the ASUS A8J Series 56k Modem (Motorola SM56) Signed-off-by: Christian Rothlaender <christian(a)rothlaender.net> THE PATCH STARTS BELOW ================================================================ > Just a small issue is left. With the alsa driver buildin into 2.6.19 > slmodem is working, with any newer one (either > standalone or buildin driver) we have a "hda-intel: invalid > format_val, rate=9600, ch=1, format=2" error in dmesg. Seems > something has been broken on that step (probably 1.0.13 to > 1.0.14?). Sadly I'm not able to tell right now which alsa was > included into 2.6.19. I'm trying to compare the files to get an idea > what's wrong (#2799 and #2820 are reporting similar > problems). > > > > This should have been fixed in the recent version. > > The driver dropped the support of 9600HZ accidentally. That's really good to hear. Can you close the Bug entry #2732 then? (#2799 and #2820 can be closed too, if the 9600 Hz problem is solved) So thanks a lot ... Christian -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.4 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGTbwUYMBp7Qf+VpkRApyKAJ4xfBkrLiG1D7whZIV1CteUTRhFagCfd/LU 2jeMsy1MQ/WxX5A7P/iThjM= =A9s6 -----END PGP SIGNATURE-----

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Re: [alsa-devel] [PATCH] SM56 Modem in Asus A8J Notebook Series
by Christian Rothländer 18 May '07

18 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there It's support for the SI3054 Chip build into the Asus A8J Notebook Series. In Windows it's called "Motorola SM56". What do you mean with "sign-off"? I'm pretty happy when it walks into upstream, so that the requests on my homepage about how to patch alsa to run that modem are ceasing down. Just a small issue is left. With the alsa driver buildin into 2.6.19 slmodem is working, with any newer one (either standalone or buildin driver) we have a "hda-intel: invalid format_val, rate=9600, ch=1, format=2" error in dmesg. Seems something has been broken on that step (probably 1.0.13 to 1.0.14?). Sadly I'm not able to tell right now which alsa was included into 2.6.19. I'm trying to compare the files to get an idea what's wrong (#2799 and #2820 are reporting similar problems). Christian Takashi Iwai wrote: > Thanks, the patch looks good to me. > Could you give a brief description of the patch and your sign-off to > merge to the upstream? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.4 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGTbafYMBp7Qf+VpkRAgl1AJ4uWR1vVVE6APAMB2SQjcPmhByJpQCfdweP vJZuXa7/0njBJ0h8vFWdUH0= =Dbm9 -----END PGP SIGNATURE-----

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[alsa-devel] [PATCH] SM56 Modem in Asus A8J Notebook Series
by Christian Rothländer 18 May '07

18 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there after I submitted the solution 5 month ago in the Bug system, I am trying it now directly. Attached to this posting is the necessary patch to get this modem working with slmodemd. The Patch is against alsa-driver-1.0.14rc4/alsa-kernel/pci/hda/patch_si3054.c if included it's Bug #0002732, assigned to tiwai. Greetings Christian -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.4 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGTadQYMBp7Qf+VpkRAp8MAJ9lii0FA/lBZPRabFxWIiVVJnAh7wCdGEkk fH0mFw4iOP3S9BHcTor3qVo= =suhj -----END PGP SIGNATURE----- --- patch_si3054.c 2007-05-18 15:06:23.000000000 +0200 +++ patch_si3054.c 2007-05-18 15:06:56.000000000 +0200 @@ -305,6 +305,7 @@ { .id = 0x10573055, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x10573057, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x10573155, .name = "Si3054", .patch = patch_si3054 }, + { .id = 0x15433155, .name = "Si3054", .patch = patch_si3054 }, /*Asus A8J Modem (SM56)*/ {} };

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[alsa-devel] [PATCH] Add support for gain in softvol plugin
by Steve Longerbeam 18 May '07

18 May '07

Hi all, What do you think of this patch? I tested this using a stac9205 HDA codec with all sample formats supported by softvol (s32, s16, s24, big and little endian). The potential drawback I see is that softvol now has to use a intermediate long long multiply and add when computing amplified 32-bit samples. This wasn't an issue with the system I tested on, but possibly could be on slower machines. Also, are their any portability issues with using a long long datatype in alsa-lib? What is the list of embedded architectures that alsa-lib is targeted to build on? I see long long used in many other places in alsa-lib, so I'm hoping this isn't an issue. Steve

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[alsa-devel] submitting a patch against the ALSA repos
by Adrian McMenamin 18 May '07

18 May '07

I have submitted my patch against the 2.6.21 kernel tree but I dare say I need to actually submit it against the alsa repos. I have a few questions: 1. Which set of sources do I patch - I assume alsa-driver - is that right? 2. How do I patch the firmware repo? As it is a binary I assume it is by some means other than patch(1) 3. This is a new architecture - there are no other SuperH drivers in alsa (afaik) - how do I add a new architecture? My memory is that when I tried this a year ago it was a bit of a nightmare.

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Re: [alsa-devel] [PATCH] asihpi 3.07.04
by Takashi Iwai 18 May '07

18 May '07

At Fri, 18 May 2007 13:16:44 +1200, Eliot Blennerhassett wrote: > > Greetings, > > Attached patch for alsa-driver/pci/asihpi > > Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> > The accompanying patch also requires updated firmware > http://audioscience.com/internet/download/firmware/asihpi_firmware-3.07.04.… > > Based on > http://audioscience.com/internet/download/hpklinux-3.07.04.tar.bz2 > Release notes > http://audioscience.com/internet/download/drvnotes.txt > http://audioscience.com/internet/download/drvnotes_rc.txt > > Main points > The bulk of this patch is addition of doxygen documentation to hpi.h and > hpifunc.c. (hopefully coming to the end of this soon) > > Add several new adapter types ASI6618, ASI6518. ASI8900 > Bug fixes to ASI6416 DSP code > Filtered out some more conditionally not compiled for ALSA. > > asihpi.c: experimental MMAP mode (disabled) > hpidebug.h: simplify debug macros > hpios_linux_kernel.h: remove a layer of macros wrapping spinlocks > hpicmn.c: pull more common code from hpi backends > hpifunc.c: remove more obsolete functions > hpimsgx.c: remove dead code > hpios.h pOsReference now struct pci_dev* not void* > hpi6205.c: use power of 2 buffers (REQUIRES matching firmware) Thanks. Many chunks look like a good clean up. Now committed to ALSA HG tree (both driver and firmware). Takashi

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[alsa-devel] HDA: Fix headphone mute issue on non-eapd Conexant systems.
by Tobin Davis 18 May '07

18 May '07

This patch fixes an automute code issue for systems that do not rely on eapd support. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] [PATCH] asihpi 3.07.04
by Eliot Blennerhassett 18 May '07

18 May '07

Greetings, There is a new patch for alsa-driver/pci/asihpi here http://audioscience.com/internet/download/alsa_driver3282_asihpi30704.patch Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> The accompanying patch also requires updated firmware http://audioscience.com/internet/download/firmware/asihpi_firmware-3.07.04.… Based on http://audioscience.com/internet/download/hpklinux-3.07.04.tar.bz2 Release notes http://audioscience.com/internet/download/drvnotes.txt http://audioscience.com/internet/download/drvnotes_rc.txt Main points The bulk of this patch is addition of doxygen documentation to hpi.h and hpifunc.c. (hopefully coming to the end of this soon) Add several new adapter types ASI6618, ASI6518. ASI8900 Bug fixes to ASI6416 DSP code Filtered out some more conditionally not compiled for ALSA. asihpi.c: experimental MMAP mode (disabled) hpidebug.h: simplify debug macros hpios_linux_kernel.h: remove a layer of macros wrapping spinlocks hpicmn.c: pull more common code from hpi backends hpifunc.c: remove more obsolete functions hpimsgx.c: remove dead code hpios.h pOsReference now struct pci_dev* not void* hpi6205.c: use power of 2 buffers (REQUIRES matching firmware) -- Eliot

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[alsa-devel] Newbie questions regarding ALSA drivers
by Pharaoh . 17 May '07

17 May '07

Hi, I am very new to ALSA and just trying to get familier with the things, I have got few questions. 1. Is it true that all the OSS drivers are included only in directory linux/sound/oss and no where else? 2. There are arch specific folders like ARM, MIPS etc, so all the drivers in these folders are ALSA or OSS? 3. If I write a new driver for an OMAP based board, it should go in to linux/sound/arm/omap/ right? Or only ALSA drivers for this board should go there? 4. I have some old OSS drivers which were written for my board, do I need to make them ALSA drivers to be in line with the latest trend i.e. phasing out OSS drivers? -Pharaoh.

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[alsa-devel] Audio driver for OMAP 730 based board - hpw to migrate from OSS to ALSA
by Pharaoh . 17 May '07

17 May '07

Hi I am new here and have slight idea about the Linux audio drivers. I am working on OMAP 730 based p2-sample board. AFAIK, there is no audio driver support for this board. I have found 2.4 drivers for the same board which I have ported to 2.6, but it is an OSS driver. So, the questions: 1. Has anyone written audio drivers for the above mentioned board? Any plans to make them open source? Any pointers will be useful regarding the same. 2. How should I go about migrating the OSS driver to ALSA, if something like this is necessary? 3. Any resources which will be helpful for writing ALSA drivers? Thanks & regards, Pharaoh.

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 17 May '07

17 May '07

Yes, its like you said Rene, sb8 works although the sound is distorted and the module can't be removed... Later I can modprobe sgalaxy too, and the sound works without a hitch. Also since now I can compile some code - your "eeprom.c" compiles and works - I'm available for further testing... I also noticed that the binary you sent was 506kb while my compilation is 9kb, maybe you sent an early/later version with some additional includes for verbose ? I compiled it with "gcc -W -Wall -o eeprom eeprom.c" like you said (on my xubuntu 7.04 with build-essentials deb package installed). See attachment... Ram Kromberg ramkromberg(a)mail.com

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 17 May '07

17 May '07

You did it Rene !!! >sudo ./eeprom >sudo modprobe snd-sgalaxy sbport=0x220 wssport=0x530 It's working now !! I downloaded the binary (i didn't compile the code), chmod it, sudo it and the sgalaxy worked !!! the sb8 didn't work but I don't really care for that anyhow :-) For the record, I tested it on xubuntu 7.04 with xmms. Are you now planning to implement the code to the driver itself ? If you want me to test anything, let me know. i'm also considering installing slackware 11 since I need the preformance... THANKS !!! Ram Kromberg ramkromberg(a)mail.com

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[alsa-devel] [1/3] 2.6.22-rc1: known regressions v2
by Michal Piotrowski 16 May '07

16 May '07

Hi all, Here is a list of some known regressions in 2.6.22-rc1. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Unclassified Subject : AMD k6-III/450 won't boot w/2.6.22-rc1 References : http://lkml.org/lkml/2007/5/16/192 Submitter : Bob Tracy <rct(a)gherkin.frus.com> Handled-By : Dave Jones <davej(a)codemonkey.org.uk> Status : problem is being debugged ACPI Subject : nx6125 has lost fan control References : http://lkml.org/lkml/2007/5/16/249 Submitter : Ray Lee <ray-lk(a)madrabbit.org> Status : Unknown ALSA Subject : lost snd_4236 device References : http://lkml.org/lkml/2007/5/13/87 Submitter : Pete Clements <clem(a)clem.clem-digital.net> Handled-By : Rene Herman <rene.herman(a)gmail.com> Patch : http://lkml.org/lkml/2007/5/15/23 Status : patch available Block devices Subject : loop devices limited to one single device References : http://lkml.org/lkml/2007/5/16/229 Submitter : Uwe Bugla <uwe.bugla(a)gmx.de> Status : Unknown Cryptography Subject : cryptomgr oops References : http://lkml.org/lkml/2007/5/14/283 Submitter : Luca Tettamanti <kronos.it(a)gmail.com> Handled-By : Herbert Xu <herbert(a)gondor.apana.org.au> Status : problem is being debugged Regards, Michal -- Michal K. K. Piotrowski Kernel Monkeys (http://kernel.wikidot.com/start)

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[alsa-devel] HDA: Add more systems to Sigmatel codec.
by Tobin Davis 16 May '07

16 May '07

This patch adds more Dell systems and a Panasonic laptop with Stac9200 codecs. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] HDA: Add support for Gateway NX860
by Tobin Davis 16 May '07

16 May '07

This patch adds support for the Gateway NX860 system. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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