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

[alsa-devel] [PATCH] hda-codec - Add missing #defines (and 1 rename) in hda_codec.h.
by Andrew Paprocki 13 Dec '07

13 Dec '07

hda-codec - Add missing #defines (and 1 rename) in hda_codec.h. Added AC_VERB_GET_DIGI_CONVERT_2 and renamed AC_VERB_GET_DIGI_CONVERT to AC_VERB_GET_DIGI_CONVERT_1 to stay consistent with the SET variants. Added AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, and AC_PINCAP_LR_SWAP. The missing fields were listed in the ALC883 datasheet rev 1.3. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r eb09731e73d7928c70ba776c1719149d112ee96a -r da002fb740a42b4fd6dcbd45693f9bee3c828812 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/hda_codec.c Mon Dec 03 22:52:58 2007 -0500 @@ -1434,8 +1434,9 @@ int snd_hda_create_spdif_out_ctls(struct if (err < 0) return err; } - codec->spdif_ctls = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + codec->spdif_ctls = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls); return 0; } @@ -1482,7 +1483,7 @@ static int snd_hda_spdif_in_status_get(s unsigned short val; unsigned int sbits; - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); sbits = convert_to_spdif_status(val); ucontrol->value.iec958.status[0] = sbits; ucontrol->value.iec958.status[1] = sbits >> 8; @@ -1533,7 +1534,8 @@ int snd_hda_create_spdif_in_ctls(struct return err; } codec->spdif_in_enable = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0) & + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0) & AC_DIG1_ENABLE; return 0; } diff -r eb09731e73d7928c70ba776c1719149d112ee96a -r da002fb740a42b4fd6dcbd45693f9bee3c828812 pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/hda_codec.h Mon Dec 03 22:52:58 2007 -0500 @@ -77,12 +77,14 @@ enum { #define AC_VERB_GET_PIN_SENSE 0x0f09 #define AC_VERB_GET_BEEP_CONTROL 0x0f0a #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c -#define AC_VERB_GET_DIGI_CONVERT 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 #define AC_VERB_GET_GPIO_MASK 0x0f16 #define AC_VERB_GET_GPIO_DIRECTION 0x0f17 +#deifne AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 @@ -110,6 +112,7 @@ enum { #define AC_VERB_SET_GPIO_DATA 0x715 #define AC_VERB_SET_GPIO_MASK 0x716 #define AC_VERB_SET_GPIO_DIRECTION 0x717 +#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e @@ -190,6 +193,7 @@ enum { #define AC_PINCAP_OUT (1<<4) /* output capable */ #define AC_PINCAP_IN (1<<5) /* input capable */ #define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */ +#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */ #define AC_PINCAP_VREF (0x37<<8) #define AC_PINCAP_VREF_SHIFT 8 #define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ diff -r eb09731e73d7928c70ba776c1719149d112ee96a -r da002fb740a42b4fd6dcbd45693f9bee3c828812 pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/patch_realtek.c Mon Dec 03 22:52:58 2007 -0500 @@ -564,7 +564,7 @@ static int alc_spdif_ctrl_get(struct snd unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long *valp = ucontrol->value.integer.value; unsigned int val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, 0x00); + AC_VERB_GET_DIGI_CONVERT_1, 0x00); *valp = (val & mask) != 0; return 0; @@ -578,7 +578,7 @@ static int alc_spdif_ctrl_put(struct snd unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long val = *ucontrol->value.integer.value; unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, + AC_VERB_GET_DIGI_CONVERT_1, 0x00); /* Set/unset the masked control bit(s) as needed */

2 2

[alsa-devel] M-audio Delta Audiophile 192 drivers
by Manel Clos 13 Dec '07

13 Dec '07

Hi all! We purchased a M-audio Delta Audiophile 192, which uses the ICE1724 chipset. Using Ubuntu gutsy 7.10 and alsa 1.0.14. PCM is working, but as soon as we start Rosegarden the complete system hangs (SysRq doesn't even work). Also, the input of the card is not shown in gnome-mixer. Is there anything we can do to help the development of the driver? We are able to compile kernel + patches, do C programming, test PCM, Midi... Thanks in advance!

3 2

[alsa-devel] Question about pcm_formats[] array in sound/core/pcm_misc.c
by Timur Tabi 13 Dec '07

13 Dec '07

I'm confused about the entries for unsigned 24-bit sound in the pcm_formats[] array: [SNDRV_PCM_FORMAT_U24_BE] = { .width = 24, .phys = 32, .le = 0, .signd = 0, .silence = { 0x80, 0x00, 0x00 }, }, and [SNDRV_PCM_FORMAT_U24_3BE] = { .width = 24, .phys = 24, .le = 0, .signd = 0, .silence = { 0x80, 0x00, 0x00 }, }, SNDRV_PCM_FORMAT_U24_BE means a 24-bit sample packed into a 32-bit word. So I presume that if the 24-bit sample is: 0x123456 then when that sample is written to memory, it looks like this: X X + 1 X + 2 X + 3 0x00 0x12 0x34 0x56 (where X is some address in memory). If that's so, shouldn't the entry for SNDRV_PCM_FORMAT_U24_BE look like this: [SNDRV_PCM_FORMAT_U24_BE] = { .width = 24, .phys = 32, .le = 0, .signd = 0, .silence = { 0x00, 0x80, 0x00, 0x00 }, }, ^^^^^^ -- Timur Tabi Linux kernel developer at Freescale

2 1

[alsa-devel] Sound problems in PCCHIPS A13G+
by alsa-devel.20.pris＠spamgourmet.com 11 Dec '07

11 Dec '07

My motherboard: PCCHIPS A13G+ http://www.pcchips.com.tw/PCCWebSite...uID=44&LanID=0 I'm using Ubuntu 7.10 - the Gutsy Gibbon, I was using Feisty before and had the same problems. The interesting thing: the system sounds all work. Some example .ogg files that came with Ubuntu also work. There is one example file however "fables_01_01_aesop.spx" which only gives a very low sound, almost inaudible. The real strange thing happens with ".avi" and mp3 files. They sound distorted and very low, almost inaudible. Some have better/louder sound, but still distorted, but I can understand what is being spoken, others not. Thanks for any help! If you need more information just ask. Technical details follow(including the Alsa Audio Debug): The sound hardware of my Mobo is in the NVIDIA MCP61S Single chipset. This chipset consists of: GeForce(R) 6100 GPU and MCP nForce(R) 405 codec(less /proc/asound/card0/codec#0): Codec: Realtek ALC660-VD Address: 0 Vendor Id: 0x10ec0660 Subsystem Id: 0x1019c601 Revision Id: 0x100001 The aplay command lists another codec: ALC861VD aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC861VD Analog [ALC861VD Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 Here is the Alsa Audio Debug: ALSA Audio Debug v0.1.0 - Mon Nov 12 00:00:12 BRST 2007 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/gpl.txt Kernel ---------------------------------------------------- Linux 2.6.22-14-generic #1 SMP Sun Oct 14 23:05:12 GMT 2007 i686 GNU/Linux Loaded Modules -------------------------------------------- snd_hda_intel 263712 4 snd_pcm_oss 44672 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm 80388 3 snd_hda_intel,snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 33152 0 snd_seq_midi 9600 0 snd_rawmidi 25728 1 snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 53232 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24324 3 snd_pcm,snd_seq snd_seq_device 9228 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54660 15 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device snd_page_alloc 11400 2 snd_hda_intel,snd_pcm Modprobe Conf --------------------------------------------- Warning: module config file does not exist This means any kernel modules will not be auto loaded See your linux distro docs on how to create this file Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 1.0.14 (Thu May 31 09:03:25 2007 UTC). 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfe028000 irq 16 2: : timer 3: : sequencer 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0] : control cat: /proc/asound/hwdep: No such file or directory 00-00: ALC861VD Analog : ALC861VD Analog : playback 1 : capture 2 Client info cur clients : 3 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Connecting To: 15:0 Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 15 : "OSS sequencer" [Kernel] Port 0 : "Receiver" (-we-) Connected From: 0:1 Dev Snd --------------------------------------------------- controlC0 pcmC0D0c pcmC0D0p seq timer CPU ------------------------------------------------------- model name : AMD Processor model unknown cpu MHz : 1900.000 RAM ------------------------------------------------------- MemTotal: 969524 kB SwapTotal: 1951888 kB Hardware -------------------------------------------------- 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control

2 2

[alsa-devel] Steinberg Midex 8 driver
by Jason Woodward 11 Dec '07

11 Dec '07

I am wondering if there is still work being done on a driver for the Steinberg Midex 8. I found a message from May 2007 by Norman Schleicher that midi out is working. I had given up hope on this long ago, and am excited at the possiblity of using my Midex in linux. I would love to have even a partially working driver to play with. Thanks, Jason Woodward

1 0

[alsa-devel] Capturing from Line and Mic simultaneously
by Christophe Osuna 11 Dec '07

11 Dec '07

Hi, I would like to capture sound from the Line and Mic inputs simultaneously. The sound chip seems capable to do so: inside alsamixer, in the "Capture" tab I can select up to three subdevices, and from the "Playback" tab I can set the input source one to "Mic" and the input source two to "Line" (and three to whatever). When trying to use several output ports at the same time I use four, six or eight channels. I tried the same thing for input (i.e. use four channels) but that failed. I guess I have to use subdevices but I have no idea on how to start working with them. Could anyone: * tell me if ALSA is limited to two capture channels, and in that case what should be done; * point me to some documentation showing what to do, or give me some general hints Thanks.

2 3

[alsa-devel] cmipci: have only surround40 on 6 ch card
by SeNS 11 Dec '07

11 Dec '07

Hello, All! Sorry for repost; probably, it's a driver problem. I have 6 channel CMI8738-based card (it works in 6 channel mode under win32; and works before under Linux with ALSA). =============================================== lspci output: "0000:00:0a.0 Multimedia audio controller: C-Media Electronics Inc CM8738 (rev 10)" =============================================== my kernel version 2.4.34 alsa version 1.0.15 =============================================== cat /proc/asound/cards 0 [CMI8738 ]: CMI8738 - C-Media CMI8738 C-Media CMI8738 (model 37) at 0xe000, irq 10 =============================================== cat /proc/asound/card0/cmipci C-Media PCI CMI8738 (model 37) at 0xe000, irq 10 00: 02 00 01 00 04: 10 1c 00 00 08: 03 00 00 01 0c: 00 00 01 00 10: d4 00 00 40 14: 00 00 00 00 18: 00 80 80 08 1c: fa 4c 04 33 20: 10 01 00 00 24: 20 01 ff 00 28: 00 00 00 00 2c: 00 00 00 00 30: ff ff ff ff 34: 00 00 00 00 38: 00 00 00 00 3c: 00 00 00 00 =============================================== WBR, SeNS

2 5

Re: [alsa-devel] [alsa-cvslog] alsa-kernel: cmipci: disable "Modem" control on version 39 or newer chips
by Thierry Vignaud 10 Dec '07

10 Dec '07

Clemens Ladisch <clemens(a)ladisch.de> writes: > changeset: 5582:5082de4abb26 > tag: tip > user: cladisch > date: Mon Dec 10 12:28:52 2007 +0100 > files: pci/cmipci.c > description: > cmipci: disable "Modem" control on version 39 or newer chips > > On version 39 or newer chips, we better remove the "Modem" control > because this register bit now mutes the front channels of the > multichannel stream. shouldn't that deserve a comment in the code? > diff -r eb09731e73d7 -r 5082de4abb26 pci/cmipci.c > --- a/pci/cmipci.c Mon Dec 03 17:08:40 2007 +0100 > +++ b/pci/cmipci.c Mon Dec 10 12:28:52 2007 +0100 > @@ -2651,10 +2651,8 @@ static struct snd_kcontrol_new snd_cmipc > }; > > /* card control switches */ > -static struct snd_kcontrol_new snd_cmipci_control_switches[] __devinitdata = { > - // DEFINE_CARD_SWITCH("Joystick", joystick), /* now module option */ > - DEFINE_CARD_SWITCH("Modem", modem), > -}; > +static struct snd_kcontrol_new snd_cmipci_modem_switch __devinitdata = > +DEFINE_CARD_SWITCH("Modem", modem); > > > static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_device) > @@ -2735,9 +2733,9 @@ static int __devinit snd_cmipci_mixer_ne > } > > /* card switches */ > - sw = snd_cmipci_control_switches; > - for (idx = 0; idx < ARRAY_SIZE(snd_cmipci_control_switches); idx++, sw++) { > - err = snd_ctl_add(cm->card, snd_ctl_new1(sw, cm)); > + if (cm->chip_version < 39) { > + err = snd_ctl_add(cm->card, > + snd_ctl_new1(&snd_cmipci_modem_switch, cm)); > if (err < 0) > return err; > } > > > ------------------------------------------------------------------------- > SF.Net email is sponsored by: > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > Alsa-cvslog mailing list > Alsa-cvslog(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-cvslog

2 2

[alsa-devel] Segfault in mix_areas3_smp with Edirol UA-25 in advance-mode
by Lasse Kärkkäinen 10 Dec '07

10 Dec '07

No asound.conf, just the default dmix configuration. Trying to play audio via this card randomly causes crashes. Might be a race condition between closing the PCM at the same time as opening a new one or something like that, but I didn't debug it further. I get this with both Xine and GStreamer, using ALSA as audio output. Using other sound cards or using UA-25 without advance mode¹ there are no segfaults. Advanced Linux Sound Architecture Driver Version 1.0.14 (Fri Jul 20 09:12:58 2007 UTC). alsa-lib version: 1.0.14a GStreamer: #0 0x00002b54714d58dc in mix_areas3_smp () from /usr/lib/libasound.so.2 #1 0x00002b54714d6477 in snd_pcm_dmix_sync_area () from /usr/lib/libasound.so.2 #2 0x00002b54714d6a4e in snd_pcm_dmix_start () from /usr/lib/libasound.so.2 #3 0x00002b54714bff60 in snd_pcm_rate_start () from /usr/lib/libasound.so.2 #4 0x00002b54714ac357 in snd_pcm_write_areas () from /usr/lib/libasound.so.2 #5 0x00002b54714b4ca2 in snd_pcm_mmap_writei () from /usr/lib/libasound.so.2 #6 0x00002aaaaf89dc1c in gst_alsasink_write (asink=0x1bf1620, data=0x1a51a40, length=3760) at gstalsasink.c:786 #7 0x00002aaaadf46ca0 in audioringbuffer_thread_func (buf=<value optimized out>) at gstaudiosink.c:226 #8 0x00002b5472bf72b4 in g_thread_create_proxy () from /usr/lib/libglib-2.0.so.0 #9 0x00002b547202bfe7 in start_thread () from /lib/libpthread.so.0 Xine: #0 0x00002b9c574d88dc in mix_areas3_smp () from /usr/lib/libasound.so.2 #1 0x00002b9c574d9477 in snd_pcm_dmix_sync_area () from /usr/lib/libasound.so.2 #2 0x00002b9c574d98fa in snd_pcm_dmix_mmap_commit () from /usr/lib/libasound.so.2 #3 0x00002b9c574b96f0 in snd_pcm_plugin_write_areas () from /usr/lib/libasound.so.2 #4 0x00002b9c574af29a in snd_pcm_write_areas () from /usr/lib/libasound.so.2 #5 0x00002b9c574b9052 in snd_pcm_plugin_writei () from /usr/lib/libasound.so.2 #6 0x00002aaaaefa8078 in ao_alsa_write () from /usr/lib64/xine/plugins/1.1.8/xineplug_ao_out_alsa.so #7 0x00002b9c542397a7 in ao_loop () from /usr/lib/libxine.so.1 #8 0x00002b9c5802efe7 in start_thread () from /lib/libpthread.so.0 ¹) There is a switch on this USB sound card for switching between modes. In advance mode the device is not standard USB Audio device, but it supports 24 bit packed samples and sample rates higher than 44.1 kHz. Otherwise it is a standard USB audio device, but only supports 16 bit 44.1 kHz.

2 2

[alsa-devel] Query regarding external alsa plugins
by Pharaoh . 10 Dec '07

10 Dec '07

Hi I am writing an external IO type plugin for my board since my audio driver is going to be a user space driver i.e. no alsa kernel driver because of some technical reasons. My pluging works okay and now I want to provide a mixer control interface like volume control and playback path selection etc. Is there anything like mixer plugin etc which can be intergrated with my existing plugin? Can I get a test code somewhere? Regards, Sandeep.

1 1