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[alsa-devel] [PATCH] hda: 92HD73xxx distortion fix
by Matthew Ranostay 14 Mar '08

14 Mar '08

Fixed issue on some laptops that if the Master mixer and DAC mixers are turned all the way up that will cause distortion. This is fixed by limiting the max volume with the volume knob nid. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 547e051a88b5 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Mar 12 13:12:15 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Wed Mar 12 21:48:13 2008 -0400 @@ -526,6 +526,25 @@ static struct hda_verb stac92hd73xx_6ch_ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {} +}; + +static struct hda_verb dell_eq_core_init[] = { + /* set master volume to max value without distortion + * and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, + /* setup audio connections */ + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, /* setup import muxs */ { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -3460,17 +3479,19 @@ again: switch (spec->board_config) { case STAC_DELL_M6: - spec->init = dell_m6_core_init; + spec->init = dell_eq_core_init; switch (codec->subsystem_id) { case 0x1028025e: /* Analog Mics */ case 0x1028025f: stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); spec->num_dmics = 0; break; - case 0x10280254: /* Digital Mics */ + case 0x10280271: /* Digital Mics */ + case 0x10280272: + spec->init = dell_m6_core_init; + /* fall-through */ + case 0x10280254: case 0x10280255: - case 0x10280271: - case 0x10280272: stac92xx_set_config_reg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; break;

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[alsa-devel] alsa conf file
by tomars 14 Mar '08

14 Mar '08

I write a asound.conf file in the /etc/ , the detail is as follows. pcm.hardware{ type hw:0 card 0 device 0 } pcm_slave.test { pcm "hardware" format S16_LE rate 44100 channels 2 period_time 20000 buffer_time 80000 } pcm.mix1{ type dmix ipc_key 1235675 ipc_key_add_uid true slave "test" } pcm.mix2{ type dmix ipc_key 1235675 ipc_key_add_uid true slave { pcm "hw:0" format S16_LE rate 44100 channels 2 period_time 20000 buffer_time 80000 } } pcm.!default { type plug slave.pcm mix2 #slave.pcm mix1 } when I use the slave.pcm mix2, the aplay works well. but if I select the slave.pcm mix1, the aplay gets the info: ALSA lib pcm_direct.c:1462:(_snd_pcm_direct_get_slave_ipc_offset) Invalid type 'dmix' for slave PCM I compare the definition of mix1 with the definition of mix2, I feel they have the identical meaning. so why does the mix2 get the error? thank you!

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[alsa-devel] problems howto send simple 'noteon' event with pyalsa...
by ki.ber.kom.uni.st 14 Mar '08

14 Mar '08

e, i was happy user of PySeq [1] and it was easy to send directly 'noteon' event.. more or less in python interactoin session: import pyseq seq = pyseq.PySeq() seq.createOutPort() event = snd_seq_event() event.setSource(0) event.setSubscribers() event.setNoteOn(ch, note, velocity) event.sendAsIs(seq) it seems that pyalsa will go along alsa development so i tried pyalsa.. but.... i tried this: from pyalsa import alsaseq seq = alsaseq.Sequencer() seq.create_simple_port("myPort", alsaseq.SEQ_PORT_TYPE_APPLICATION, alsaseq.SEQ_PORT_CAP_SUBS_READ | alsaseq.SEQ_PORT_CAP_READ | alsaseq.SEQ_PORT_CAP_WRITE | alsaseq.SEQ_PORT_CAP_SUBS_WRITE) event = alsaseq.SeqEvent(alsaseq.SEQ_EVENT_NOTE) event.set_data({'note.note' : 0x40, 'note.velocity' : 127, 'note.duration' : 1, 'note.off_velocity' : 0}) event.dest = (20, 0) seq.output_event(event) seq.drain_output() after drain it says: "SequencerError: Failed to drain output: Invalid argument" i can connect and disconnect ports around and (20,0) is functional port where i send events with no problems from PySeq... any ideas? do i have to use queues to be able to send 'noteon'.. for my purposes it seems to me that that's overkill.... thanx.. [1] http://www.sci.ccny.cuny.edu/~brinkman/software/pyseq/

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[alsa-devel] [PATCH] Fix ALC662 DAC mixer mutes
by Herton Ronaldo Krzesinski 13 Mar '08

13 Mar '08

Currently ALC662 doesn't suport amp mute for AmpOut in nids 0x02, 0x03, 0x04 (see block diagram in ALC662 datasheet page 3, does M correspond to mute?). The result is that currently mute for "Front Playback Switch", "Surround Playback Switch", "Center Playback Switch" and "LFE Playback Switch" mixer items doesn't work (tested on Asus P5GC-MX motherboard with 3stack-6ch model). The solution I found for this is to mute the proper inputs in 0x0c, 0x0d, 0x0e audio mixers. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up linux-2.6.24/sound/pci/hda/patch_realtek.c.orig linux-2.6.24/sound/pci/hda/patch_realtek.c --- linux-2.6.24/sound/pci/hda/patch_realtek.c.orig 2008-03-11 13:45:07.000000000 -0300 +++ linux-2.6.24/sound/pci/hda/patch_realtek.c 2008-03-11 14:01:57.000000000 -0300 @@ -13004,13 +13004,13 @@ static struct hda_channel_mode alc662_5s static struct snd_kcontrol_new alc662_base_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x3, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), /*Input mixer control */ @@ -13027,7 +13027,7 @@ static struct snd_kcontrol_new alc662_ba static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), @@ -13044,13 +13044,13 @@ static struct snd_kcontrol_new alc662_3S static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), -- []'s Herton

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[alsa-devel] Hardware and software period size
by Mathieu Monney 13 Mar '08

13 Mar '08

Hi all, I'm currently developping a layer between ALSA and a VoIP library for an embedded device. The device cannot have a different period size than 1024 for playback and 8 for capture (I don't know why :-( ). So my problem is that the VoIP library ask me to have a period size different of the hardware period size. So I decided to read/write to the sound device using the provided software period size rather than the hardware period size.. Is it a good idea? It seems to work but I have some sound quality issues :-( ! Thank you Mathieu

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Re: [alsa-devel] [Alsa-user] Metallic recording in cs46xx on Thinkpad T22
by Rene Herman 13 Mar '08

13 Mar '08

On 11-03-08 08:24, Gadi Oron wrote: > I am trying to record sound on an old Thinkpad T22 that uses the cs46xx > sound driver. > > Each time you start to record you have a 10% chance of having the > recording completely distorted and having a metallic sound. When you > look at the waveform it looks as though there are small segments with > sharp transitions between them, a little like if these segments were > moved a little from their correct place. No insights, but I confirm the bug with a TerraTec DMX XFire 1024 (CS4624). Rene.

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[alsa-devel] Trigger problems while writing Alsa Shim
by bruce＠azbarclay.com 13 Mar '08

13 Mar '08

I am writing a shim from alsa to our own sound card drivers. I was able to get the init driver working and our sound card recognized. Things seem to be set up correctly. When I do aplay to my device (USB device works find), it calls my copy callback twice and then it calls the trigger and fails. The call that fails is in pcm_write (writei_func) with a -5. I know that my callback returns zero but there must be something else under the covers since that gets to be zero. Any help would be appreciated. T

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Re: [alsa-devel] ALSA utility aplay/arecord error
by Saurav SAHU 13 Mar '08

13 Mar '08

Hi all The platform on which I am trying to perform ALSA playback and record with USB sound device is based on a MIPS CPU running Linux kernel 2.6.14. The kernel uses non-coherent DMA in the configuration: CONFIG_DMA_NONCOHERENT=y CONFIG_DMA_NEED_PCI_MAP_STATE=y CONFIG_CPU_BIG_ENDIAN=y CONFIG_SYS_SUPPORTS_BIG_ENDIAN=y CONFIG_BOOT_ELF32=y CONFIG_MIPS_L1_CACHE_SHIFT=5 I have attached the kernel .config file for further information. So far, I have not succeeded in hearing sound with 'aplay', or capturing sound with 'arecord'. If there have been any similar issues reported on ALSA usage with MIPS CPU based on non-coherent DMA in the past, please let me know. Also, if there have been any patches developed for successful resolution of this issue on similar platform/configuration, it would be very useful for me. Thanks & Regards Saurav ----- Original Message ----- From: Saurav Sahu To: alsa-user(a)lists.sourceforge.net Sent: Friday, March 07, 2008 4:33 PM Subject: Re: ALSA utility aplay/arecord error I have been able to resolve the previously reported errors by setting the environment variable for ALSA configuration file on the target filesystem of the embedded platform. However, I do not hear any audio from USB sound device with speaker-test or aplay: Upon using arecord for recording, I get an error: Is this problem related to a low-level driver issue which may not be working well on target platform? Could there be a DMA related issue as the samples don't seem to be transferred to the USB audio device. ----- Original Message ----- From: Saurav Sahu To: alsa-user(a)lists.sourceforge.net Sent: Thursday, March 06, 2008 4:28 PM Subject: ALSA utility aplay/arecord error Hello I am working towards using ALSA for playback and capture on an embedded platform running Linux. The kernel uses ALSA driver version 1.0.10rc1. I had manually built and installed ALSA library (v1.0.10) and ALSA utilities (v1.0.10) for the target platform filesystem. I have a USB audio device (C-Media) connected to the USB port of the board and it seems to be successfully successfully detected by the driver. However, when I run ALSA utilities like aplay, arecord or speaker-test, I observe errors opening the hardware device which may be related to the configuration on the filesystem. Are these errors from ALSA utility/library due to the detection of hardware, ALSA configuration files or something else related to my target device filesystem? Thanks in advance for any help. Do let me know if you need any more details. Best Regards Saurav

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[alsa-devel] [PATCH] asihpi - new checkpatch = more changes
by Eliot Blennerhassett 13 Mar '08

13 Mar '08

Signed-off-by: Eliot Blennerhasssett <linux(a)audioscience.com> * HPI version 3.09.15 * Updating to latest version of checkpatch resulted in some more warnings to clean up. * A few whitespace cleanups * Removed DBG_TXT and DBG_CHAR macros --- patch attached -- Eliot Blennerhassett www.audioscience.com

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[alsa-devel] Hda-intel: AC_VERB_SET_PIN_WIDGET_CONTROL appears to do nothing for HDMI device
by Ben Firshman 13 Mar '08

13 Mar '08

This is on 1.0.16. Forgive me if I appear clueless, I have only been working on this for the past few hours. In an attempt to get HDMI audio working on the AppleTV, I have added this line to the presets in patch_atihdmi.c: { .id = 0x10951390, .name = "SI1390 HDMI", .patch = patch_atihdmi }, The device is detected fine: ben@mythfrontend:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC882 Analog [ALC882 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC882 Digital [ALC882 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 ben@mythfrontend:~$ and I can send audio to the device, but it doesn't work. I presume this because this is doing nothing: static struct hda_verb atihdmi_basic_init[] = { /* enable digital output on pin widget */ { 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, {} /* terminator */ }; ben@mythfrontend:~$ cat /proc/asound/card0/codec#1 Codec: Generic 1095 SI1390 HDMI Address: 1 Vendor Id: 0x10951390 Subsystem Id: 0xffffffff Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Node 0x03 [Pin Complex] wcaps 0x40738d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0894: OUT Detect R/L Pin Default 0x98460010: [Fixed] SPDIF Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x02 "Pin-ctls: 0x00:" should be reading "Pin-ctls: 0x40: OUT" should it not? What is wrong? Thanks Ben

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