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

[alsa-devel] Correct i2s implementation for s3c2416
by Heiko Stübner 04 Apr '11

04 Apr '11

Hi, I'm trying to make sound work on a s3c2416 based device and am quite confused about the underlying I2S stuff as it seems there are 3 different I2S implementations for samsung SoCs. The i2s.c and .h seem to be made for the s3c64xx and up (according to the initial commit message). This leaves the s3c-i2s-v2+s3c2412_i2s combination and s3c24xx-iis. Documentation on this seems a bit sparse and I'm still taking first steps in kernel-development at all - so I would be really thankfull for a pointer in the right direction. Thanks Heiko

2 2

Re: [alsa-devel] linux-next: Tree for April 4 (sound/soc)
by Randy Dunlap 04 Apr '11

04 Apr '11

On Mon, 4 Apr 2011 14:35:46 +1000 Stephen Rothwell wrote: > Hi all, > > Changes since 20110401: When CONFIG_SPI=n, CONFIG_I2C=n: sound/soc/soc-cache.c:418: error: implicit declaration of function 'i2c_master_send' sound/soc/soc-cache.c:421: error: implicit declaration of function 'do_spi_write' --- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

3 3

[alsa-devel] [PATCH] ALSA: usb-audio: define another USB ID for a buggy USB MIDI cable
by Tarek Soliman 04 Apr '11

04 Apr '11

There are many USB MIDI cables out there that have buggy firmware that reports it can do more than 4 bytes in a packet when they can only properly handle 4 This patch adds the ID of yet another one of those cables Signed-off-by: Tarek Soliman <tarek(a)bashasoliman.com> --- sound/usb/midi.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b4b39c0..bdf5d9d 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1299,6 +1299,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, case USB_ID(0x0a92, 0x1020): /* ESI M4U */ case USB_ID(0x1430, 0x474b): /* RedOctane GH MIDI INTERFACE */ case USB_ID(0x15ca, 0x0101): /* Textech USB Midi Cable */ + case USB_ID(0xfc08, 0x0101): /* Unknown vendor Cable */ case USB_ID(0x15ca, 0x1806): /* Textech USB Midi Cable */ case USB_ID(0x1a86, 0x752d): /* QinHeng CH345 "USB2.0-MIDI" */ ep->max_transfer = 4; -- 1.7.4.1 -- Tarek

2 2

[alsa-devel] alsa-tools HDSP mixer broken
by Klaus Schulz 04 Apr '11

04 Apr '11

Hi folks. The official version of alsa-tools 1.0.24.1 seems to be broken. HDSP mixer won't run. I made it work though with the git version. Would be nice if you could release a tools 1.0.24.2 soon. I consider this a critical bug. Cheers Klaus

2 1

[alsa-devel] "default" device. Is there a way to detect if it's a plugin, and if so which one?
by Colin Guthrie 04 Apr '11

04 Apr '11

Hi, If the user opens the "default" device in alsa, is there a way to find out if that "device" is actually a plugin? i.e. is it possible to tell if you have opened the pulseaudio plugin via alsa? There are various hacky ways (e.g. checking hints etc.) but that relies on downstream packaging which certainly isn't fool proof. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

3 3

[alsa-devel] [PATCH 3/3 v5] alsa-lib: fixed coverity reported issues under "USE_AFTER_FREE" checker.
by sudarshan.bisht＠nokia.com 04 Apr '11

04 Apr '11

From: Sudarshan Bisht <sudarshan.bisht(a)nokia.com> Coverity Static Analysis helps developers find hard-to-spot, yet potentially crash-causing defects early in the development phase, reducing the cost,time, and risk of software errors. This patch has a fix for conditions where 1) variable is used evern after getting freed thus causing junk assignment and 2) double free of memory. --- modules/mixer/simple/python.c | 1 + src/control/control_hw.c | 1 + src/mixer/mixer.c | 1 - 3 files changed, 2 insertions(+), 1 deletions(-) diff --git a/modules/mixer/simple/python.c b/modules/mixer/simple/python.c index c822c52..784d9a0 100644 --- a/modules/mixer/simple/python.c +++ b/modules/mixer/simple/python.c @@ -663,6 +663,7 @@ pymixer_attach_hctl(struct pymixer *pymixer, PyObject *args) return NULL; err = snd_mixer_attach_hctl(pymixer->mixer, hctl); if (err < 0) { + snd_hctl_close(hctl); PyErr_Format(PyExc_RuntimeError, "Cannot attach hctl: %s", snd_strerror(err)); return NULL; } diff --git a/src/control/control_hw.c b/src/control/control_hw.c index cf258b4..16c4987 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -414,6 +414,7 @@ int snd_ctl_hw_open(snd_ctl_t **handle, const char *name, int card, int mode) if (err < 0) { close(fd); free(hw); + return err; } ctl->ops = &snd_ctl_hw_ops; ctl->private_data = hw; diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 85d843f..5e5789a 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -228,7 +228,6 @@ int snd_mixer_attach_hctl(snd_mixer_t *mixer, snd_hctl_t *hctl) return -ENOMEM; err = snd_hctl_nonblock(hctl, 1); if (err < 0) { - snd_hctl_close(hctl); free(slave); return err; } -- 1.7.0.4

1 0

[alsa-devel] snd_pcm_hw_params_set_buffer_time_near: invalid argument
by Jean-Yves Avenard 04 Apr '11

04 Apr '11

Hi there. I'm a bit puzzled as to what values are considered valids and which one aren't when it comes to setting the buffer time with snd_pcm_hw_params_set_buffer_time_near. For example setting a value of 295ms : all ok. From 300ms to 450ms -> "Invalid argument". but 500ms for example is okay. What's the logic behind it ? It's obviously related to the sampling rate, the audio format and the number of channels ; as the values giving me "Invalid argument" change then. Any comments would be greatly appreciated. Thanks! JY

4 11

[alsa-devel] Media players report 96khz, proc reports 44khz
by TS Lura 04 Apr '11

04 Apr '11

Hey, I am trying to play a 96kHz/24-bit pcm file: {{{ [2309][root@ubuntu:~]# file test.wav test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 32 bit, stereo 96000 Hz }}} Which plays with full 96kHz resolution under aplay. But under any other player as VLC, mplayer, totem it plays at 44kHz reported by /proc/asound/card1/stream0, even if the player is reporting to be playing at 96kHz. Others have reported the same problem: http://www.head-fi.org/forum/thread/243027/how-to-make-m-audio-transit-worn… Some people at mplayer @ irc.freenode.net encouraged me to file a bug report on this issue. Hope it is useful. Cheers, TSLura I have also filed a ticked with pulseAudio -- Ticket URL: <http://pulseaudio.org/ticket/930> PulseAudio <http://pulseaudio.org/> The PulseAudio Sound Server {{{ [2336][torstefan@ubuntu:/proc/asound/card1]$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: STAC92xx Digital [STAC92xx Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: USB [Transit USB], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0 card 1: USB [Transit USB], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 }}} {{{ [2329][root@ubuntu:~]# aplay -D plughw:1,0 test.wav Playing WAVE 'test.wav' : Signed 32 bit Little Endian, Rate 96000 Hz, Stereo Every 0.5s: cat stream0 Sun Apr 3 23:29:28 2011 M-Audio Transit USB at usb-0000:00:1d.7-1.2.3, full speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 URBs = 3 [ 8 8 8 ] Packet Size = 576 Momentary freq = 96000 Hz (0x60.0000) Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 48001 - 96000 (continuous) Interface 1 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (NONE) }}} {{{ [2329][root@ubuntu:~]# mplayer test.wav MPlayer 1.0rc4-4.4.5 (C) 2000-2010 MPlayer Team mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing test.wav. Audio only file format detected. ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder AUDIO: 96000 Hz, 2 ch, s32le, 6144.0 kbit/100.00% (ratio: 768000->768000) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== AO: [pulse] 96000Hz 2ch s32le (4 bytes per sample) Video: no video Starting playback... }}} {{{ Every 0.5s: cat stream0 Sun Apr 3 23:30:40 2011 M-Audio Transit USB at usb-0000:00:1d.7-1.2.3, full speed : USB Audio Playback: Status: Running Interface = 1 Altset = 3 URBs = 6 [ 4 4 4 4 4 4 ] Packet Size = 196 Momentary freq = 44100 Hz (0x2c.199a) Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 48001 - 96000 (continuous) Interface 1 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (NONE) }}} {{{ [2330][torstefan@ubuntu:~]$ vlc test.wav VLC media player 1.1.4 The Luggage (revision exported) Blocked: call to unsetenv("DBUS_ACTIVATION_ADDRESS") Blocked: call to unsetenv("DBUS_ACTIVATION_BUS_TYPE") [0x9894914] main libvlc: Running vlc with the default interface. Use 'cvlc' to use vlc without interface. Blocked: call to setlocale(6, "") Blocked: call to sigaction(17, 0xb48ff0d4, 0xb48ff048) Warning: call to signal(13, 0x1) Warning: call to signal(13, 0x1) Warning: call to srand(1302210069) Warning: call to rand() Blocked: call to setlocale(6, "") (process:11503): Gtk-WARNING **: Locale not supported by C library. Using the fallback 'C' locale. Warning: call to signal(13, 0x1) Warning: call to signal(13, 0x1) }}} {{{ Every 0.5s: cat stream0 Sun Apr 3 23:31:17 2011 M-Audio Transit USB at usb-0000:00:1d.7-1.2.3, full speed : USB Audio Playback: Status: Running Interface = 1 Altset = 3 URBs = 6 [ 4 4 4 4 4 4 ] Packet Size = 196 Momentary freq = 44100 Hz (0x2c.199a) Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 48001 - 96000 (continuous) Interface 1 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (NONE) Rates: 8000 - 48000 (continuous) Interface 1 Altset 3 Format: S16_LE Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Capture: Status: Stop Interface 2 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 5 IN (ASYNC) Rates: 48001 - 96000 (continuous) Interface 2 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 5 IN (ASYNC) Rates: 8000 - 48000 (continuous) Interface 2 Altset 3 Format: S16_LE Channels: 2 Endpoint: 5 IN (ASYNC) Rates: 8000 - 48000 (continuous) }}} {{{ [2332][torstefan@ubuntu:~]$ totem test.wav Every 0.5s: cat stream0 Sun Apr 3 23:32:33 2011 M-Audio Transit USB at usb-0000:00:1d.7-1.2.3, full speed : USB Audio Playback: Status: Running Interface = 1 Altset = 3 URBs = 6 [ 4 4 4 4 4 4 ] Packet Size = 196 Momentary freq = 44100 Hz (0x2c.199a) Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 48001 - 96000 (continuous) Interface 1 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (NONE) Rates: 8000 - 48000 (continuous) Interface 1 Altset 3 Format: S16_LE Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Capture: Status: Stop Interface 2 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 5 IN (ASYNC) Rates: 48001 - 96000 (continuous) Interface 2 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 5 IN (ASYNC) Rates: 8000 - 48000 (continuous) Interface 2 Altset 3 Format: S16_LE Channels: 2 Endpoint: 5 IN (ASYNC) Rates: 8000 - 48000 (continuous) }}} System dmsg http://pastebin.com/SmWamhja -- Ticket URL: <http://pulseaudio.org/ticket/930> PulseAudio <http://pulseaudio.org/> The PulseAudio Sound Server

2 1

[alsa-devel] [PATCH] alsactl init: Mute CD Playback volume by default
by David Henningsson 04 Apr '11

04 Apr '11

The CD analog playback line is seldom used these days, as all modern players rip the CD and use the DAC on the sound card. In addition, it causes background hum on some machines. Therefore keep it muted by default. BugLink: http://bugs.launchpad.net/bugs/747184 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[PATCH] alsactl init: Mute CD Playback volume by default
by David Henningsson 04 Apr '11

04 Apr '11

The CD analog playback line is seldom used these days, as all modern players rip the CD and use the DAC on the sound card. In addition, it causes background hum on some machines. Therefore keep it muted by default. BugLink: http://bugs.launchpad.net/bugs/747184 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- alsactl/init/default | 13 +++---------- 1 files changed, 3 insertions(+), 10 deletions(-) diff --git a/alsactl/init/default b/alsactl/init/default index 7f8ec4c..b7cb941 100644 --- a/alsactl/init/default +++ b/alsactl/init/default @@ -144,17 +144,10 @@ CTL{name}="Music Playback Switch",CTL{do_search}=="1", \ CTL{values}="on" CTL{reset}="mixer" -CTL{name}="CD Playback Volume",PROGRAM!="__ctl_search", GOTO="" -# if master volume control is present, turn CD volume to max -ENV{has_pmaster_vol}=="true",CTL{write}=="0dB",GOTO="" -ENV{has_pmaster_vol}=="true",CTL{write}=="100%",GOTO="" -# exception - some HDA codecs have shifted dB range -CTL{dBmin}=="-34.50dB",CTL{dBmax}=="12.00dB",CTL{write}=="0dB",GOTO="" -CTL{dBmin}=="-30.00dB",CTL{dBmax}=="0dB",CTL{write}=="0dB",GOTO="" -CTL{write}!="$env{pvolume}",CTL{values}="$env{ppercent}" -LABEL="" +CTL{name}="CD Playback Volume",CTL{do_search}=="1", \ + CTL{values}="0%" CTL{name}="CD Playback Switch",CTL{do_search}=="1", \ - CTL{values}="on" + CTL{values}="off" CTL{reset}="mixer" CTL{name}="Mono Playback Volume",CTL{do_search}=="1", \ -- 1.7.4.1 --------------010109070108060801050503--

1 0

[alsa-devel] [PATCH] hda-emu: Run without optimizations
by David Henningsson 04 Apr '11

04 Apr '11

Hi, This one has been lying in my local tree for a while. Feel free to apply if you think it makes sense. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH 1/4 v2] alsa-lib: fixed coverity reported issues under "FORWARD_NULL" checker.
by sudarshan.bisht＠nokia.com 04 Apr '11

04 Apr '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> --- modules/mixer/simple/sbasedl.c | 2 +- src/conf.c | 1 + src/hwdep/hwdep.c | 1 + src/pcm/pcm_hooks.c | 1 + src/pcm/pcm_simple.c | 2 +- src/rawmidi/rawmidi.c | 1 + src/rawmidi/rawmidi_virt.c | 8 +++++--- 7 files changed, 11 insertions(+), 5 deletions(-) diff --git a/modules/mixer/simple/sbasedl.c b/modules/mixer/simple/sbasedl.c index 0137586..494802f 100644 --- a/modules/mixer/simple/sbasedl.c +++ b/modules/mixer/simple/sbasedl.c @@ -99,7 +99,7 @@ int mixer_simple_basic_dlopen(snd_mixer_class_t *class, __error: if (initflag) free(priv); - if (h == NULL) + if (h) snd_dlclose(h); free(xlib); return -ENXIO; diff --git a/src/conf.c b/src/conf.c index 8939d62..ddefff6 100644 --- a/src/conf.c +++ b/src/conf.c @@ -3268,6 +3268,7 @@ static int snd_config_hooks_call(snd_config_t *root, snd_config_t *config, snd_c snd_config_iterator_t i, next; if (snd_config_get_type(func_conf) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for func %s definition", str); + err = -EINVAL; goto _err; } snd_config_for_each(i, next, func_conf) { diff --git a/src/hwdep/hwdep.c b/src/hwdep/hwdep.c index b882b35..5dc791c 100644 --- a/src/hwdep/hwdep.c +++ b/src/hwdep/hwdep.c @@ -78,6 +78,7 @@ static int snd_hwdep_open_conf(snd_hwdep_t **hwdep, if (err >= 0) { if (snd_config_get_type(type_conf) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for HWDEP type %s definition", str); + err = -EINVAL; goto _err; } snd_config_for_each(i, next, type_conf) { diff --git a/src/pcm/pcm_hooks.c b/src/pcm/pcm_hooks.c index 3a99d55..404d51e 100644 --- a/src/pcm/pcm_hooks.c +++ b/src/pcm/pcm_hooks.c @@ -385,6 +385,7 @@ static int snd_pcm_hook_add_conf(snd_pcm_t *pcm, snd_config_t *root, snd_config_ if (err >= 0) { if (snd_config_get_type(type) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for PCM type %s definition", str); + err = -EINVAL; goto _err; } snd_config_for_each(i, next, type) { diff --git a/src/pcm/pcm_simple.c b/src/pcm/pcm_simple.c index 975f699..f943ec0 100644 --- a/src/pcm/pcm_simple.c +++ b/src/pcm/pcm_simple.c @@ -89,7 +89,7 @@ static int set_hw_params(snd_pcm_t *pcm, return err; if (periods == 1) return -EINVAL; - if (*period_time == 0) { + if (period_time) { err = INTERNAL(snd_pcm_hw_params_get_period_time)(hw_params, period_time, NULL); if (err < 0) return err; diff --git a/src/rawmidi/rawmidi.c b/src/rawmidi/rawmidi.c index b28488a..0bd6b96 100644 --- a/src/rawmidi/rawmidi.c +++ b/src/rawmidi/rawmidi.c @@ -201,6 +201,7 @@ static int snd_rawmidi_open_conf(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp if (err >= 0) { if (snd_config_get_type(type_conf) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for RAWMIDI type %s definition", str); + err = -EINVAL; goto _err; } snd_config_for_each(i, next, type_conf) { diff --git a/src/rawmidi/rawmidi_virt.c b/src/rawmidi/rawmidi_virt.c index 52b8984..e5b17e4 100644 --- a/src/rawmidi/rawmidi_virt.c +++ b/src/rawmidi/rawmidi_virt.c @@ -383,9 +383,11 @@ int snd_rawmidi_virtual_open(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp, _err: if (seq_handle) snd_seq_close(seq_handle); - if (virt->midi_event) - snd_midi_event_free(virt->midi_event); - free(virt); + if (virt) { + if (virt->midi_event) + snd_midi_event_free(virt->midi_event); + free(virt); + } if (inputp) free(*inputp); if (outputp) -- 1.7.0.4

2 4

[alsa-devel] [PATCH 1/2] ALSA: ASoc: TLV320AIC3X: ad SPI and clock on GPIO2 or BCLK
by Prchal Jiří 04 Apr '11

04 Apr '11

Hi, this patch ads SPI communication for codecs TLV320AIC3X and clock input on GPIO2 or BCLK. Tested on TLV320AIC3106 and AT91SAM9260. TODO: Set the model in SPI probe the right way. I don't know how. This codec communicates on SPI in 1st byte: 7 MSB is register address, 1 LSB is read/write, 2nd byte data. For this reason there is new functions in soc-cache.c snd_soc_7_8_*. Kernel version: 2.6.38 Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/codecs/Kconfig /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/codecs/Kconfig --- linux-2.6.38-vanilla/sound/soc/codecs/Kconfig 2011-03-15 02:20:32.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/codecs/Kconfig 2011-03-17 08:37:14.997500191 +0100 @@ -37,7 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER - select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_TLV320AIC3X if SND_SOC_I2C_AND_SPI select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE @@ -118,7 +118,7 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate - + config SND_SOC_ADS117X tristate diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/codecs/tlv320aic3x.c /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/codecs/tlv320aic3x.c --- linux-2.6.38-vanilla/sound/soc/codecs/tlv320aic3x.c 2011-03-15 02:20:32.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/codecs/tlv320aic3x.c 2011-03-24 09:34:58.769370117 +0100 @@ -2,7 +2,9 @@ * ALSA SoC TLV320AIC3X codec driver * * Author: Vladimir Barinov, <vbarinov(a)embeddedalley.com> + * Jiri Prchal, <jiri.prchal(a)aksignal.cz * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> + * (C) 2011 AK signal Brno * * Based on sound/soc/codecs/wm8753.c by Liam Girdwood * @@ -42,6 +44,7 @@ #include <linux/regulator/consumer.h> #include <linux/platform_device.h> #include <linux/slab.h> +#include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -984,6 +987,13 @@ static int aic3x_set_dai_sysclk(struct s { struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + u8 data; + + /* set external clock on GPIO2 or BCLK */ + data = snd_soc_read(codec, AIC3X_CLKGEN_CTRL_REG); + data &= 0x0f; + data |= ((clk_id << PLLCLK_IN_SHIFT) | (clk_id << CLKDIV_IN_SHIFT)); + snd_soc_write(codec, AIC3X_CLKGEN_CTRL_REG, data); aic3x->sysclk = freq; return 0; @@ -1371,9 +1381,12 @@ static int aic3x_probe(struct snd_soc_co aic3x->codec = codec; codec->dapm.idle_bias_off = 1; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type); + if (aic3x->control_type == SND_SOC_I2C) + ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type); + else + ret = snd_soc_codec_set_cache_io(codec, 7, 8, aic3x->control_type); if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; } @@ -1472,6 +1485,54 @@ static struct snd_soc_codec_driver soc_c .resume = aic3x_resume, }; +#if defined(CONFIG_SPI_MASTER) +static int aic3x_spi_probe(struct spi_device *spi) +{ + struct aic3x_pdata *pdata = spi->dev.platform_data; + struct aic3x_priv *aic3x; + int ret; + + aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); + if (aic3x == NULL) { + dev_err(&spi->dev, "failed to create private data\n"); + return -ENOMEM; + } + + aic3x->control_type = SND_SOC_SPI; + spi_set_drvdata(spi, aic3x); + + if (pdata) { + aic3x->gpio_reset = pdata->gpio_reset; + aic3x->setup = pdata->setup; + } else { + aic3x->gpio_reset = -1; + } + + aic3x->model = AIC3X_MODEL_3X; + + ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_aic3x, &aic3x_dai, 1); + if (ret < 0) + kfree(aic3x); + return ret; +} + +static int __devexit aic3x_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + kfree(spi_get_drvdata(spi)); + return 0; +} + +static struct spi_driver aic3x_spi_driver = { + .driver = { + .name = "tlv320aic3x-codec", + .owner = THIS_MODULE, + }, + .probe = aic3x_spi_probe, + .remove = __devexit_p(aic3x_spi_remove), +}; +#endif /* CONFIG_SPI_MASTER */ + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * AIC3X 2 wire address can be up to 4 devices with device addresses @@ -1557,6 +1618,13 @@ static int __init aic3x_modinit(void) ret); } #endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&aic3x_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register AIC3X SPI driver: %d\n", + ret); + } +#endif return ret; } module_init(aic3x_modinit); @@ -1566,6 +1634,9 @@ static void __exit aic3x_exit(void) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&aic3x_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&aic3x_spi_driver); +#endif } module_exit(aic3x_exit); diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/codecs/tlv320aic3x.h /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/codecs/tlv320aic3x.h --- linux-2.6.38-vanilla/sound/soc/codecs/tlv320aic3x.h 2011-03-15 02:20:32.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/codecs/tlv320aic3x.h 2011-03-22 13:37:54.751566089 +0100 @@ -178,6 +178,13 @@ #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 +#define PLLCLK_IN_SHIFT 4 +#define CLKDIV_IN_SHIFT 6 +/* clock in source */ +#define CLKIN_MCLK 0 +#define CLKIN_GPIO2 1 +#define CLKIN_BCLK 2 + /* Software reset register bits */ #define SOFT_RESET 0x80 diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/soc-cache.c /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/soc-cache.c --- linux-2.6.38-vanilla/sound/soc/soc-cache.c 2011-03-15 02:20:32.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/soc-cache.c 2011-03-24 09:23:00.984373772 +0100 @@ -99,6 +99,58 @@ static int snd_soc_4_12_spi_write(void * #define snd_soc_4_12_spi_write NULL #endif +/* special functions for codecs with 7 bit register address and LSB read/write (like TLV320AIC3X) */ +static unsigned int snd_soc_7_8_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + int ret; + unsigned int val; + + if (reg >= codec->driver->reg_cache_size || + snd_soc_codec_volatile_register(codec, reg)) { + if (codec->cache_only) + return -1; + + BUG_ON(!codec->hw_read); + return codec->hw_read(codec, ((reg << 1) | 1)); + } + + ret = snd_soc_cache_read(codec, reg, &val); + if (ret < 0) + return -1; + return val; +} + +static int snd_soc_7_8_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + int ret; + + data[0] = (reg << 1); + data[1] = value; + + if (!snd_soc_codec_volatile_register(codec, reg) && + reg < codec->driver->reg_cache_size) { + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return -1; + } + + if (codec->cache_only) { + codec->cache_sync = 1; + return 0; + } + + ret = codec->hw_write(codec->control_data, data, 2); + if (ret == 2) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -661,6 +713,11 @@ static struct { .spi_write = snd_soc_4_12_spi_write, }, { + .addr_bits = 7, .data_bits = 8, + .write = snd_soc_7_8_write, .read = snd_soc_7_8_read, + .spi_write = snd_soc_8_8_spi_write, + }, + { .addr_bits = 7, .data_bits = 9, .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, .spi_write = snd_soc_7_9_spi_write,

2 5

[PATCH] hda-emu: Run without optimizations
by David Henningsson 04 Apr '11

04 Apr '11

hda-emu does not need the extra performance, better have all more reliable debugging. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- configure.ac | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/configure.ac b/configure.ac index c964761..a8c3308 100644 --- a/configure.ac +++ b/configure.ac @@ -13,7 +13,7 @@ WFLAGS="-Wall -Wundef -Wstrict-prototypes -Wno-trigraphs -fno-strict-aliasing -f WFLAGS="$WFLAGS -Wdeclaration-after-statement" WFLAGS="$WFLAGS -Wno-pointer-sign" WFLAGS="$WFLAGS -fwrapv" -OPTFLAGS="-O2 -g" +OPTFLAGS="-O0 -g" CFLAGS="$CFLAGS $OPTFLAGS $WFLAGS" AC_ARG_ENABLE(old-workq, -- 1.7.4.1 --------------020500080700030001070405--

1 0

[alsa-devel] [PATCH 1/2] ASoC: Name jack GPIOs based on jack not codec
by Stephen Warren 03 Apr '11

03 Apr '11

snd_soc_jack_gpio has a name field. Use that name when registering the IRQ, since this is far more informative than the codec driver name. This shows up in /proc/interrupts. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-jack.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fcab80b..6203a72 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -325,7 +325,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, gpio_handler, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, - jack->codec->dev->driver->name, + gpios[i].name, &gpios[i]); if (ret) goto err; -- 1.7.1

2 2

[alsa-devel] [PATCH] ASoC: PXA: Fix oops in __pxa2xx_pcm_prepare
by Vasily Khoruzhick 03 Apr '11

03 Apr '11

pxa2xx_pcm_hw_free frees dma channel and sets prtd->dma_ch to -1, but does not set prtd->params to NULL, so if pxa2xx_pcm_hw_params will be called immediately, it leaves prtd->dma_ch initialized with -1, and it results in oops in __pxa2xx_pcm_prepare. This bug is triggered via SDL. This patch adds check for prtd->dma_ch to __pxa2xx_pcm_prepare and cleans prtd->params, so now it works properly. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/arm/pxa2xx-pcm-lib.c | 3 +++ sound/soc/pxa/pxa2xx-pcm.c | 1 + 2 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 8808b82..52b1e17 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -140,6 +140,9 @@ int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) if (!prtd || !prtd->params) return 0; + if (prtd->dma_ch == -1) + return -EIO; + DCSR(prtd->dma_ch) &= ~DCSR_RUN; DCSR(prtd->dma_ch) = 0; DCMD(prtd->dma_ch) = 0; diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 02fb664..2ce0b2d 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -65,6 +65,7 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) if (prtd->dma_ch >= 0) { pxa_free_dma(prtd->dma_ch); prtd->dma_ch = -1; + prtd->params = NULL; } return 0; -- 1.7.4.1

2 2

[alsa-devel] [PATCH] ASoC: Remove special casing for registerless widgets
by Mark Brown 02 Apr '11

02 Apr '11

Since we recently explicitly set the register for registerless widgets to no register there is no longer any need to special case power updates for them, we can allow them to be handled with the register compression code as other widgets are. As this is the only remaining user of dapm_generic_apply_power() and dapm_update_bits() also remove those functions. Noticed-by: Lu Guanqun <guanqun.lu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Compile tested only currently, sending it out now as part of discussion with Lu Guanqun. sound/soc/soc-dapm.c | 100 -------------------------------------------------- 1 files changed, 0 insertions(+), 100 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b11d011..c9196d6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -322,45 +322,6 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm, return -ENODEV; } -/* update dapm codec register bits */ -static int dapm_update_bits(struct snd_soc_dapm_widget *widget) -{ - int change, power; - unsigned int old, new; - struct snd_soc_codec *codec = widget->codec; - struct snd_soc_dapm_context *dapm = widget->dapm; - struct snd_soc_card *card = dapm->card; - - /* check for valid widgets */ - if (widget->reg < 0 || widget->id == snd_soc_dapm_input || - widget->id == snd_soc_dapm_output || - widget->id == snd_soc_dapm_hp || - widget->id == snd_soc_dapm_mic || - widget->id == snd_soc_dapm_line || - widget->id == snd_soc_dapm_spk) - return 0; - - power = widget->power; - if (widget->invert) - power = (power ? 0:1); - - old = snd_soc_read(codec, widget->reg); - new = (old & ~(0x1 << widget->shift)) | (power << widget->shift); - - change = old != new; - if (change) { - pop_dbg(dapm->dev, card->pop_time, - "pop test %s : %s in %d ms\n", - widget->name, widget->power ? "on" : "off", - card->pop_time); - pop_wait(card->pop_time); - snd_soc_write(codec, widget->reg, new); - } - dev_dbg(dapm->dev, "reg %x old %x new %x change %d\n", widget->reg, - old, new, change); - return change; -} - /* create new dapm mixer control */ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *w) @@ -644,57 +605,6 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_reg_event); -/* Standard power change method, used to apply power changes to most - * widgets. - */ -static int dapm_generic_apply_power(struct snd_soc_dapm_widget *w) -{ - int ret; - - /* call any power change event handlers */ - if (w->event) - dev_dbg(w->dapm->dev, "power %s event for %s flags %x\n", - w->power ? "on" : "off", - w->name, w->event_flags); - - /* power up pre event */ - if (w->power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } - - /* power down pre event */ - if (!w->power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - - dapm_update_bits(w); - - /* power up post event */ - if (w->power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMU)) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } - - /* power down post event */ - if (!w->power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMD)) { - ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } - - return 0; -} - /* Generic check to see if a widget should be powered. */ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) @@ -981,16 +891,6 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, NULL, SND_SOC_DAPM_POST_PMD); break; - case snd_soc_dapm_input: - case snd_soc_dapm_output: - case snd_soc_dapm_hp: - case snd_soc_dapm_mic: - case snd_soc_dapm_line: - case snd_soc_dapm_spk: - /* No register support currently */ - ret = dapm_generic_apply_power(w); - break; - default: /* Queue it up for application */ cur_sort = sort[w->id]; -- 1.7.4.1

2 4

[alsa-devel] [PATCH 1/2] ASoC: remove dapm_update_bits()
by Lu Guanqun 02 Apr '11

02 Apr '11

dapm_update_bits() is only called in dapm_generic_apply_power(). And in turn, dapm_generic_apply_power() is only called in dapm_seq_run() when w->id is snd_soc_dapm_{input|output|hp|line|mic}. Under these cases, dapm_update_bits() is rather useless, because the below tests are always true: /* check for valid widgets */ if (widget->reg < 0 || widget->id == snd_soc_dapm_input || widget->id == snd_soc_dapm_output || widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line || widget->id == snd_soc_dapm_spk) return 0; Therefore, we better omit this function for simplicity. Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- sound/soc/soc-dapm.c | 41 ----------------------------------------- 1 files changed, 0 insertions(+), 41 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 81c4052..d511bbe 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -322,45 +322,6 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm, return -ENODEV; } -/* update dapm codec register bits */ -static int dapm_update_bits(struct snd_soc_dapm_widget *widget) -{ - int change, power; - unsigned int old, new; - struct snd_soc_codec *codec = widget->codec; - struct snd_soc_dapm_context *dapm = widget->dapm; - struct snd_soc_card *card = dapm->card; - - /* check for valid widgets */ - if (widget->reg < 0 || widget->id == snd_soc_dapm_input || - widget->id == snd_soc_dapm_output || - widget->id == snd_soc_dapm_hp || - widget->id == snd_soc_dapm_mic || - widget->id == snd_soc_dapm_line || - widget->id == snd_soc_dapm_spk) - return 0; - - power = widget->power; - if (widget->invert) - power = (power ? 0:1); - - old = snd_soc_read(codec, widget->reg); - new = (old & ~(0x1 << widget->shift)) | (power << widget->shift); - - change = old != new; - if (change) { - pop_dbg(dapm->dev, card->pop_time, - "pop test %s : %s in %d ms\n", - widget->name, widget->power ? "on" : "off", - card->pop_time); - pop_wait(card->pop_time); - snd_soc_write(codec, widget->reg, new); - } - dev_dbg(dapm->dev, "reg %x old %x new %x change %d\n", widget->reg, - old, new, change); - return change; -} - /* create new dapm mixer control */ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *w) @@ -673,8 +634,6 @@ static int dapm_generic_apply_power(struct snd_soc_dapm_widget *w) return ret; } - dapm_update_bits(w); - /* power up post event */ if (w->power && w->event && (w->event_flags & SND_SOC_DAPM_POST_PMU)) { -- 1.7.2.3

2 4

[alsa-devel] Multiple sound problems with ALC888 (1025:013f) on Acer Aspire 5930
by Elisavet Vorgia 02 Apr '11

02 Apr '11

Hi All, I have sound issues with my laptop, I cannot get 5.1 sound and jack sensing does not work (but works on Windows 7; dual-boot). I have an Acer Aspire 5930 (chipset: ALC888) running Ubuntu 10.10 (64-bit). I have tested with Alsa 1.0.23 (which comes with Ubuntu 10.10) and also compiled Alsa 1.0.24. Here is the Alsa-info with Alsa 1.0.24 (latest), and "model=auto", http://www.alsa-project.org/db/?f=2cbab ... 31dc6a7d70 <http://www.alsa-project.org/db/?f=2cbab995516f4b56a7faecf537f61f31dc6a7d70> Here is alsa-info with "model=acer-aspire-6530g", http://www.alsa-project.org/db/?f=40992 ... 17160cde64 <http://www.alsa-project.org/db/?f=409923a3d8f042804e61f21c7b8ae917160cde64> However, on Linux, 1. the sound volume is too low compared to Windows 7. For example, 50% on the volume bar in Ubuntu is roughly equivalent to 10% in Windows 7. 2. the subwoofer control bar in Sound Preferences Settings seems to control the master volume bar. 3. it seems that only 2 speakers actually work (Front Left and Front Right) as tested with speaker-test -c6 -twav 4. when I plug in the headphones jack the internal speakers don't get muted, instead I have combined sound output. I tried to change the model from auto to "acer" (tested all the acer and the 6stack-dig models and settled down to acer-aspire-6530g as a popular suggestion) but it didn't change much. (You'll find the alsa-info.txt as an attachment to this message.) Is there any possible solution or suggestion for this problem? I am happy to try out patches. Thanks, Elisavet

2 1

[alsa-devel] export pcm_debug_name?
by Eliot Blennerhassett 01 Apr '11

01 Apr '11

Currently defined locally in pcm_lib.c, this function would be useful for debug output from drivers too? Any objection to exporting this function? > static void pcm_debug_name(struct snd_pcm_substream *substream, > char *name, size_t len) > { > snprintf(name, len, "pcmC%dD%d%c:%d", > substream->pcm->card->number, > substream->pcm->device, > substream->stream ? 'c' : 'p', > substream->number); > } -- Eliot Blennerhassett AudioScience Inc.

2 3

[alsa-devel] Resource temporarily unavailable
by Nikhil 01 Apr '11

01 Apr '11

Hi all, when I use the function " r = snd_pcm_readi(alsa.icard, buf+readpos , left);" I got error "Resource temporarily unavailable". Please tel me how to solve the issue.My sound card not busy.. Thanks Nikhil

1 0

Re: [alsa-devel] Linux 2.6.38 freeze because of sound/core/pcm_lib.c commit 59ff878ffb26bc0be812ca8295799164f413ae88
by Clemens Ladisch 01 Apr '11

01 Apr '11

Christopher K. wrote: > Kernel 2.6.38 has been freezing ever since rc1 on my asus p7p55d. > After booting up, the kernel seems to use a lot of cpu-time and after > 20-30 seconds > the system comes to a complete halt. > > I just finished bisecting 2.6.37 - 2.6.38-rc1 and found this commit to > be responsible: > > 59ff878ffb26bc0be812ca8295799164f413ae88 Please try this patch: --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -425,6 +425,8 @@ static int snd_pcm_hw_params(struct snd_ runtime->no_period_wakeup = (params->info & SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) && (params->flags & SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP); + runtime->hw_ptr_buffer_jiffies = + max(1uL, (runtime->buffer_size * HZ) / runtime->rate); bits = snd_pcm_format_physical_width(runtime->format); runtime->sample_bits = bits; @@ -866,8 +868,6 @@ static void snd_pcm_post_start(struct sn struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); runtime->hw_ptr_jiffies = jiffies; - runtime->hw_ptr_buffer_jiffies = (runtime->buffer_size * HZ) / - runtime->rate; runtime->status->state = state; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0)

3 6

Re: [alsa-devel] Question about your DSP topic
by Patrick Lai 01 Apr '11

01 Apr '11

Hi Liam/Mark, I ran some tests on top of soc-dsp framework pulled from Liam's topic/dsp-upstream(http://git.kernel.org/?p=linux/kernel/git/lrg/asoc- 2.6.git;a=summary). I found there is a scenario that soc-dsp framework erroneously start pcm playabck/capture. Here is the scenario: In the platform driver, I have this route table defined FE1 Playback -> Mixer 1 -> BE1 BE2 -> FE1 Capture FE = Front-end, BE = Back-end While PCM playback is going from FE1 playback to BE1, I switch off FE1 playback to Mixer 1. This caused soc_dsp_runtime_update called. Framework correctly close BE1 as it is no longer needed. Eventually, framework finds BE2 is connected to FE1 capture. Framework, without checking if FE1 capture is activated by user-space application, simply goes ahead activate BE2. Since FE1 capture is never activated, runtime structure is not allocated. This inherently results NULL pointer dereference exception. For now, in soc-dsp.c be_connect function(), I have a check to make sure fe->dsp[stream].runtime is not NULL. I don't know if it's appropriate fix. Can you please take a look? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

3 7

[alsa-devel] Query - ASoC headphone detection stereo vs. mono
by Patrick Lai 01 Apr '11

01 Apr '11

I would like to get a better understanding on how to propagate channel mode of inserted headphone through soc-jack framework? I looked at jack.h and I don't see any definition of channel mode. I would assume that user-space needs this information in order to set up the path in the audio CODEC properly such as mixing stereo content into mono stream inside CODEC. How is use case handled on other platform? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

3 2

[alsa-devel] git trees
by Steve Calfee 01 Apr '11

01 Apr '11

Hi Mark, I am starting to look at a new codec driver. I have been looking at the alsa-project wiki describing the various git trees. I am confused because they seem to be lagging the linux releases (ie 2.6.38 is out, not in the two trees I have tried. The wiki proposed occasional developers use 3 trees - not clear what is in each. ie: git clone http://git.alsa-project.org/http/alsa-driver.git alsa-driver git clone http://git.alsa-project.org/http/alsa-kmirror.git alsa-kmirror if you like do only changes in code in alsa-kmirror, please, use alsa-kernel repository git clone git://git.alsa-project.org/alsa-kernel.git alsa-kernel I really don't understand the sentence: "if you like do only changes in code in alsa-kmirror, please, use alsa-kernel repository" When I built alsa-kernel, I kernel version 2.6.38-rc8+ - which built and boots on my target. I also cloned the kernel.org broonie/sound-2.6.git tree But when I look at branches I see: asoc$ git branch -r origin/ac97-runtime-pm origin/ads117x origin/asoc-regulator origin/asoc-v2-dev origin/bias-off origin/dapm origin/dapm-log origin/dev origin/ep93xx origin/for-2.6.29 origin/for-2.6.30 origin/for-2.6.31 origin/for-2.6.32 origin/for-2.6.33 origin/for-2.6.34 origin/for-2.6.35 origin/for-2.6.36 origin/for-2.6.37 origin/for-2.6.38 origin/for-2.6.39 origin/for-2.6.40 origin/for-next origin/init-card origin/new-maintainers origin/pending origin/pxa-ssp origin/refactoring origin/reg-cache origin/tegra-arch origin/twl4030-mfd origin/untested So I checked out origin/for-2.6.39 when i built that I got a kernel version 2.6.38-rc1 or something. (and it built but didn't boot) What I need to know is which tree and which branch or tag I should use to develop on and to submit alsa patches from. Thanks for the help. Regards, Steve

2 3

[alsa-devel] config file documentation?
by Chris Furlough 01 Apr '11

01 Apr '11

Greetings fellow developers! I'm working on configuring a USB sound device, and running into the occasional difficulty. During these times of duress, I turn to the web to look for documentation on the specific facet that I'm struggling with at the moment. Without fail, I'll come across some feature of the config files that I didn't previously know of. My question is this. Does there exist a comprehensive compendium of config file settings, and options that I can peruse? My current config is a hacked together menagerie of bit and pieces gleaned from many internet sources. I'd like to do it *RIGHT*. Thanks!

1 0

[alsa-devel] ALSA hw sample rate conversion support
by Aisheng Dong 01 Apr '11

01 Apr '11

Hi ALL, Does ALSA core support hw sample rate conversion now? Our soc has a HW sample rate converter module inside, I wonder if current ALSA core provide interface to support hw sample rate conversion? Thanks Regards Aisheng

5 5

[alsa-devel] [PATCH] ALSA: HDA: Add dock mic quirk for Lenovo Thinkpad X220
by David Henningsson 31 Mar '11

31 Mar '11

This quirk is needed for the docking station mic of Lenovo Thinkpad X220 to function correctly. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] Capture Source/Input Souce of Alsa sound cards and pulseaudio source
by Raymond Yau 31 Mar '11

31 Mar '11

There are 4 Capture Volume/Switch (Front, Surround, Synth and Wave) and Capture source of ac97 codec contain 8 input sources (Mic, CD, Video , Aux, Line, Stereo Mix, Mono Mix and Phone) on SB Live! Platinum How can ALSA application know whether all 12 sources can be selected for device 0 ? card 1: Live [SB Live! Platinum [CT4760P]], device 0: emu10k1 [ADC Capture/Standard PCM Playback] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Live [SB Live! Platinum [CT4760P]], device 1: emu10k1 mic [Mic Capture] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Live [SB Live! Platinum [CT4760P]], device 2: emu10k1 efx [Multichannel Capture/PT Playback] Subdevices: 1/1 Subdevice #0: subdevice #0

1 0

[PATCH] ALSA: HDA: Add dock mic quirk for Lenovo Thinkpad X220
by David Henningsson 31 Mar '11

31 Mar '11

This quirk is needed for the docking station mic of Lenovo Thinkpad X220 to function correctly. BugLink: http://bugs.launchpad.net/bugs/746259 Cc: stable(a)kernel.org Tested-by: James Ferguson <james.ferguson(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d08cf31..69e3386 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3034,6 +3034,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} -- 1.7.1 --------------090408050708050906070307--

1 0

Re: [alsa-devel] 2.6.38+ hard lockup in pcm_lib.c
by Clemens Ladisch 31 Mar '11

31 Mar '11

Kelly Anderson wrote: > I had a consistent hard-lockup on one of my machines, an Amd X6, just > after Kdm loaded the desktop (most certainly when it decides to start > talking to the sound card). Everything was dead, even ssh access to the > computer. I git bisected it until I found the commit that was causing > it. After reverting the patch everything is fine. > > commit 59ff878ffb26bc0be812ca8295799164f413ae88 > Author: Clemens Ladisch > Date: Thu Nov 18 09:43:52 2010 +0100 > > ALSA: pcm: detect xruns in no-period-wakeup mode This looks as if you get an infinite loop, probably because hw_ptr_buffer_jiffies is zero. Please try this patch: --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -425,6 +425,8 @@ static int snd_pcm_hw_params(struct snd_ runtime->no_period_wakeup = (params->info & SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) && (params->flags & SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP); + runtime->hw_ptr_buffer_jiffies = + max(1uL, (runtime->buffer_size * HZ) / runtime->rate); bits = snd_pcm_format_physical_width(runtime->format); runtime->sample_bits = bits; @@ -866,8 +868,6 @@ static void snd_pcm_post_start(struct sn struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); runtime->hw_ptr_jiffies = jiffies; - runtime->hw_ptr_buffer_jiffies = (runtime->buffer_size * HZ) / - runtime->rate; runtime->status->state = state; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0)

1 0

[alsa-devel] [Fwd: Re: Problems with snd_hda_intel in Linux kernel 2.6.38]
by Svante Signell 31 Mar '11

31 Mar '11

I am subscribed. What's the problem? (now without attachment). How can I send the attachment to the developers? > Your mail to 'Alsa-devel' with the subject > > Re: [alsa-devel] Problems with snd_hda_intel in Linux kernel > 2.6.38 > > Is being held until the list moderator can review it for approval. > > The reason it is being held: > > Message body is too big: 488063 bytes with a limit of 60 KB > > Either the message will get posted to the list, or you will receive > notification of the moderator's decision. If you would like to cancel > this posting, please visit the following URL: > > > http://mailman.alsa-project.org/mailman/confirm/alsa-devel/9dc4fe44b477fdc3… > -------- Forwarded Message -------- From: Svante Signell <srs(a)kth.se> To: Clemens Ladisch <clemens(a)ladisch.de> Cc: Takashi Iwai <tiwai(a)suse.de>, 619034(a)bugs.debian.org, alsa-devel(a)alsa-project.org, 613979(a)bugs.debian.org Subject: Re: [alsa-devel] Problems with snd_hda_intel in Linux kernel 2.6.38 Date: Thu, 31 Mar 2011 00:42:46 +0200 On Wed, 2011-03-30 at 15:13 +0200, Clemens Ladisch wrote: > Svante Signell wrote: > > Code: f4 01 00 00 ef 31 f6 48 89 df e8 15 dd ff ff 85 c0 > > 0f 88 2b 03 00 00 48 89 ef e8 ee 11 b9 e0 8b 7b 40 e8 9f 25 a7 e0 48 8b > > 43 38 <66> 8b 10 66 89 14 24 8b 43 14 83 e8 03 83 f8 01 77 32 31 d2 be > > 5: 31 f6 xor %esi,%esi > 7: 48 89 df mov %rbx,%rdi > a: e8 15 dd ff ff callq 0xffffffffffffdd24 > f: 85 c0 test %eax,%eax > 11: 0f 88 2b 03 00 00 js 0x342 > 17: 48 89 ef mov %rbp,%rdi > 1a: e8 ee 11 b9 e0 callq 0xffffffffe0b9120d > 1f: 8b 7b 40 mov 0x40(%rbx),%edi > 22: e8 9f 25 a7 e0 callq 0xffffffffe0a725c6 > 27: 48 8b 43 38 mov 0x38(%rbx),%rax > 2b: 66 8b 10 mov (%rax),%dx <-- crash here > 2e: 66 89 14 24 mov %dx,(%rsp) > 32: 8b 43 14 mov 0x14(%rbx),%eax > 35: 83 e8 03 sub $0x3,%eax > 38: 83 f8 01 cmp $0x1,%eax > 3b: 77 32 ja 0x6f > 3d: 31 d2 xor %edx,%edx > > This is the azx_readw(chip, GCAP) in azx_create(); chip->remap_addr is > 0xffffc90011c08000 which does look like a valid pointer, but isn't. Thank you Clemens! Maybe your input is sufficient to solve this problem. I have now installed the debug version of the kernel, the objdump output is attached (please let me know if you are missing something).sorry, I don't know where to find the relevant information in this file, but that is all I have (still very large). (Does not include the error messages on stderr, maybe something is still missing.) Thanks!

1 1

[alsa-devel] [PATCH 1/2] ASoC: soc-cache: Fix indentation issues
by Dimitris Papastamos 31 Mar '11

31 Mar '11

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 18 +++++++++--------- 1 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 1210a6f..22c0b95 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -93,13 +93,13 @@ static unsigned int do_hw_read(struct snd_soc_codec *codec, unsigned int reg) } static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, - unsigned int reg) + unsigned int reg) { return do_hw_read(codec, reg); } static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) + unsigned int value) { u8 data[2]; @@ -111,7 +111,7 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, #if defined(CONFIG_SPI_MASTER) static int snd_soc_4_12_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[2]; @@ -209,7 +209,7 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, #if defined(CONFIG_SPI_MASTER) static int snd_soc_8_16_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[3]; @@ -257,7 +257,7 @@ static unsigned int do_i2c_read(struct snd_soc_codec *codec, #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) + unsigned int r) { u8 reg = r; u8 data; @@ -307,13 +307,13 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, #endif static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, - unsigned int reg) + unsigned int reg) { return do_hw_read(codec, reg); } static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) + unsigned int value) { u8 data[3]; @@ -327,7 +327,7 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, #if defined(CONFIG_SPI_MASTER) static int snd_soc_16_8_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[3]; @@ -379,7 +379,7 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, #if defined(CONFIG_SPI_MASTER) static int snd_soc_16_16_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[4]; -- 1.7.4.1

2 2

[alsa-devel] [PATCH 01/11] ASoC: Use data based init for ak4535 DAPM
by Lu Guanqun 31 Mar '11

31 Mar '11

CC: Liam Girdwood <lrg(a)slimlogic.co.uk> CC: Mark Brown <broonie(a)opensource.wolfsonmicro.com> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.de> CC: Jarkko Nikula <jhnikula(a)gmail.com> CC: Timur Tabi <timur(a)freescale.com> CC: Sascha Hauer <s.hauer(a)pengutronix.de> Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- sound/soc/codecs/ak4535.c | 19 +++++-------------- 1 files changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 8b38739..e1a214e 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -230,7 +230,7 @@ static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = { SND_SOC_DAPM_INPUT("AIN"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route ak4535_audio_map[] = { /*stereo mixer */ {"Stereo Mixer", "Playback Switch", "DAC"}, {"Stereo Mixer", "Mic Sidetone Switch", "Mic"}, @@ -287,17 +287,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Input Mixer", "Aux Capture Switch", "Aux In"}, }; -static int ak4535_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, ak4535_dapm_widgets, - ARRAY_SIZE(ak4535_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { @@ -457,8 +446,6 @@ static int ak4535_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, ak4535_snd_controls, ARRAY_SIZE(ak4535_snd_controls)); - ak4535_add_widgets(codec); - return 0; } @@ -480,6 +467,10 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .reg_cache_size = ARRAY_SIZE(ak4535_reg), .reg_word_size = sizeof(u8), .reg_cache_default = ak4535_reg, + .dapm_widgets = ak4535_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4535_dapm_widgets), + .dapm_routes = ak4535_audio_map, + .num_dapm_routes = ARRAY_SIZE(ak4535_audio_map), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.2.3

3 12

Re: [alsa-devel] [PATCH 1/2] ASoC: soc-cache: Fix indentation issues
by Liam Girdwood 30 Mar '11

30 Mar '11

On 28/03/11 11:39, dp(a)opensource.wolfsonmicro.com wrote: > From: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> > > Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> All Acked-by: Liam Girdwood <lrg(a)ti.com>

1 0

[alsa-devel] ASoC update for 2.6.39
by Mark Brown 30 Mar '11

30 Mar '11

The following changes since commit d454f39f3ff3ee5a818c271026081a48a537bd41: Merge branch 'for-2.6.39' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc (2011-03-28 13:02:29 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Mark Brown (1): ASoC: Fix CODEC device name for Corgi sound/soc/pxa/corgi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

2 1

[alsa-devel] [PATCH] sound/soc/pxa/zylonite.c: set .codec_dai_name in initializer
by Antonio Ospite 30 Mar '11

30 Mar '11

Fix the initialization of .codec_dai_name in zylonite_dai initializer, do not mix it with the initialization of .codec_name which is set already a few lines above. Signed-off-by: Antonio Ospite <ospite(a)studenti.unina.it> --- Hi, I didn't test that because I don't have the hardware, I just noticed it by reading the code from 2.6.38 and thought this may be the right fix. Is there any mechanism to spot out when the same field in a struct initializer is set more than once? Regards, Antonio Ospite http://ao2.it sound/soc/pxa/zylonite.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 25bba10..5e96f7e 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -167,7 +167,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97", - .codec_name = "wm9713-hifi", + .codec_dai_name = "wm9713-hifi", .init = zylonite_wm9713_init, }, { @@ -176,7 +176,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97-aux", - .codec_name = "wm9713-aux", + .codec_dai_name = "wm9713-aux", }, { .name = "WM9713 Voice", @@ -184,7 +184,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa-ssp-dai.2", - .codec_name = "wm9713-voice", + .codec_dai_name = "wm9713-voice", .ops = &zylonite_voice_ops, }, }; -- 1.7.4.1

3 2

[alsa-devel] [PATCH 1/2] ASoC: SSM2602: drop rate and sample bits constraints
by Mike Frysinger 30 Mar '11

30 Mar '11

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/ssm2602.c | 53 -------------------------------------------- 1 files changed, 0 insertions(+), 53 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 2727bef..70a33414 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -49,8 +49,6 @@ struct ssm2602_priv { unsigned int sysclk; enum snd_soc_control_type control_type; void *control_data; - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; }; /* @@ -278,15 +276,9 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c = codec->control_data; u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; int i = get_coeff(ssm2602->sysclk, params_rate(params)); - if (substream == ssm2602->slave_substream) { - dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); - return 0; - } - /*no match is found*/ if (i == ARRAY_SIZE(coeff_div)) return -EINVAL; @@ -316,44 +308,6 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ssm2602_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c = codec->control_data; - struct snd_pcm_runtime *master_runtime; - - /* The DAI has shared clocks so if we already have a playback or - * capture going then constrain this substream to match it. - * TODO: the ssm2602 allows pairs of non-matching PB/REC rates - */ - if (ssm2602->master_substream) { - master_runtime = ssm2602->master_substream->runtime; - dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", - master_runtime->sample_bits, - master_runtime->rate); - - if (master_runtime->rate != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - if (master_runtime->sample_bits != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); - - ssm2602->slave_substream = substream; - } else - ssm2602->master_substream = substream; - - return 0; -} - static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -370,16 +324,10 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); /* deactivate */ if (!codec->active) ssm2602_write(codec, SSM2602_ACTIVE, 0); - - if (ssm2602->master_substream == substream) - ssm2602->master_substream = ssm2602->slave_substream; - - ssm2602->slave_substream = NULL; } static int ssm2602_mute(struct snd_soc_dai *dai, int mute) @@ -505,7 +453,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops ssm2602_dai_ops = { - .startup = ssm2602_startup, .prepare = ssm2602_pcm_prepare, .hw_params = ssm2602_hw_params, .shutdown = ssm2602_shutdown, -- 1.7.4.1

3 18

[alsa-devel] [PATCH] ASoC: Blackfin: add ad193x sysclk configuration
by Mike Frysinger 30 Mar '11

30 Mar '11

From: Scott Jiang <scott.jiang(a)analog.com> Signed-off-by: Scott Jiang <scott.jiang(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-ad193x.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index d3ccb92..3367882 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -60,8 +60,16 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int clk = 0; unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7}; int ret = 0; + + switch (params_rate(params)) { + case 48000: + clk = 12288000; + break; + } + /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); @@ -74,6 +82,12 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + /* set codec DAI slots, 8 channels, all channels are enabled */ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32); if (ret < 0) -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: SSM2602: fix codec name
by Mike Frysinger 30 Mar '11

30 Mar '11

The codec name should not have a "-codec" suffix since this is not part of a MFD. This was incorrectly changed during the multi-component updated. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/ssm2602.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 2727bef..5a5a6b1 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -651,7 +651,7 @@ MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); /* corgi i2c codec control layer */ static struct i2c_driver ssm2602_i2c_driver = { .driver = { - .name = "ssm2602-codec", + .name = "ssm2602", .owner = THIS_MODULE, }, .probe = ssm2602_i2c_probe, -- 1.7.4.1

3 6

[alsa-devel] [PATCH] ASoC: fix sorting order of codecs in kconfig
by Mike Frysinger 30 Mar '11

30 Mar '11

Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f310043..b814ed0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,8 +16,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS - select SND_SOC_ADS117X select SND_SOC_AD73311 + select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C -- 1.7.4.1

3 2

[alsa-devel] [PATCH] ASoC: ad74111: new codec driver
by Mike Frysinger 30 Mar '11

30 Mar '11

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Scott Jiang <scott.jiang(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/Kconfig | 8 +++- sound/soc/codecs/Makefile | 4 ++- sound/soc/codecs/ad74111.c | 76 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad74111.h | 63 ++++++++++++++++++++++++++++++++++++ 4 files changed, 148 insertions(+), 3 deletions(-) create mode 100644 sound/soc/codecs/ad74111.c create mode 100644 sound/soc/codecs/ad74111.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index dc6d253..ee65b7c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,8 +16,9 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD183X if SPI_MASTER select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS - select SND_SOC_ADS117X select SND_SOC_AD73311 if I2C + select SND_SOC_AD74111 + select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C @@ -126,7 +127,10 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate - + +config SND_SOC_AD74111 + tristate + config SND_SOC_ADS117X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 44d3278..b834755 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-ad183x-objs := ad183x.o snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o +snd-soc-ad74111-objs := ad74111.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o @@ -90,7 +91,8 @@ obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD183X) += snd-soc-ad183x.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o -obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_AD74111) += snd-soc-ad74111.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o diff --git a/sound/soc/codecs/ad74111.c b/sound/soc/codecs/ad74111.c new file mode 100644 index 0000000..9d247ae --- /dev/null +++ b/sound/soc/codecs/ad74111.c @@ -0,0 +1,76 @@ +/* + * ALSA Soc AD74111 codec support + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "ad74111.h" + +static struct snd_soc_dai_driver ad74111_dai = { + .name = "ad74111-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_ad74111; + +static int ad74111_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_ad74111, &ad74111_dai, 1); +} + +static int __devexit ad74111_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver ad74111_codec_driver = { + .driver = { + .name = "ad74111-codec", + .owner = THIS_MODULE, + }, + + .probe = ad74111_probe, + .remove = __devexit_p(ad74111_remove), +}; + +static int __init ad74111_init(void) +{ + return platform_driver_register(&ad74111_codec_driver); +} +module_init(ad74111_init); + +static void __exit ad74111_exit(void) +{ + platform_driver_unregister(&ad74111_codec_driver); +} +module_exit(ad74111_exit); + +MODULE_DESCRIPTION("ASoC AD74111 driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad74111.h b/sound/soc/codecs/ad74111.h new file mode 100644 index 0000000..19a8183 --- /dev/null +++ b/sound/soc/codecs/ad74111.h @@ -0,0 +1,63 @@ +/* + * definitions for AD74111 registers + * + * Copyright 2006-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef __AD74111_H__ +#define __AD74111_H__ + +#define AD_READ 0x0000 +#define AD_WRITE 0x8000 + +/* Control register A */ +#define CTRL_REG_A (0 << 11) + +#define REGA_REFAMP (1 << 2) +#define REGA_REF (1 << 3) +#define REGA_DAC (1 << 4) +#define REGA_ADC (1 << 5) +#define REGA_ADC_INPAMP (1 << 6) + +/* Control register B */ +#define CTRL_REG_B (1 << 11) + +#define REGB_FCLKDIV(x) ((x) & 0x3) +#define REGB_SCLKDIV(x) (((x) & 0x3) << 2) +#define REGB_TCLKDIV(x) (((x) & 0x3) << 4) + +/* Control register C */ +#define CTRL_REG_C (2 << 11) + +#define REGC_ADC_HP (1 << 0) +#define REGC_DAC_DEEMPH(x) (((x) & 0x3) << 1) +#define REGC_LG_DELAY (1 << 3) +#define REGC_WORD_WIDTH(x) (((x) & 0x3) << 4) + +/* Control register D */ +#define CTRL_REG_D (3 << 11) + +#define REGD_MASTER (1 << 0) +#define REGD_FDCLK (1 << 1) +#define REGD_DSP_MODE (1 << 2) +#define REGD_MIX_MODE (1 << 3) +#define REGD_MFS (1 << 9) + +/* Control register E */ +#define CTRL_REG_E (4 << 11) + +#define REGE_DAC_MUTE (1 << 0) +#define REGE_ADC_MUTE (1 << 1) +#define REGE_ADC_GAIN(x) (((x) & 0x7) << 2) +#define REGE_ADC_PEAKEN (1 << 5) + +/* Control register F */ +#define CTRL_REG_F (5 << 11) +#define REGF_DAC_VOL(x) ((x) & 0x3F) + +/* Control register G */ +#define CTRL_REG_G (6 << 11) + +#endif -- 1.7.4.1

2 7

[alsa-devel] [PATCH 1/4] ASoC: Fix CODEC device name for Corgi
by Mark Brown 29 Mar '11

29 Mar '11

Got typoed in the multi-component changes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/corgi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 784cff5..9027da4 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -310,7 +310,7 @@ static struct snd_soc_dai_link corgi_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec-0.001b", + .codec_name = "wm8731-codec.0-001b", .init = corgi_wm8731_init, .ops = &corgi_ops, }; -- 1.7.4.1

2 4

[alsa-devel] radio-maestro broken (conflicts with snd-es1968)
by Ondrej Zary 29 Mar '11

29 Mar '11

Hello, the radio-maestro driver is badly broken. It's intended to drive the radio on MediaForte ESS Maestro-based sound cards with integrated radio (like SF64-PCE2-04). But it conflicts with snd_es1968, ALSA driver for the sound chip itself. If one driver is loaded, the other one does not work - because a driver is already registered for the PCI device (there is only one). This was probably broken by conversion of PCI probing in 2006: ttp://lkml.org/lkml/2005/12/31/93 How to fix it properly? Include radio functionality in snd-es1968 and delete radio-maestro? -- Ondrej Zary

4 22

[alsa-devel] emu10k1, midi on audigy platinum ex
by Goncalo Pinheira 29 Mar '11

29 Mar '11

Hello, I've noticed midi in (DIN connector) on an audigy platinum ex doesn't work. There's a patch in a 2009 forum thread http://ubuntuforums.org/showthread.php?t=1354637&page=3 The author seems to have contacted with the alsa project, but the latest kernel shows no sign of the code. Shouldn't this be commited?

2 2

[alsa-devel] [PATCH alsa-kernel] snd-hda-codec-realtek: Add quirk for Jetway NF92
by Forest Bond 29 Mar '11

29 Mar '11

The Jetway NF92 mainboard works with model=3stack-6ch-dig. Add a quirk to automatically configure the codec accordingly. Signed-off-by: Forest Bond <forest.bond(a)rapidrollout.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f1a03f2..6cc8378 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18777,6 +18777,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E), + SND_PCI_QUIRK(0x16f3, 0xa625, "Jetway NF92", ALC662_3ST_6ch_DIG), {} }; -- 1.7.0.4

3 5

Re: [alsa-devel] Loud pops from Intel HDA onboard sound, may damage speakers (fwd)
by Takashi Iwai 29 Mar '11

29 Mar '11

At Sun, 27 Mar 2011 03:27:28 +0100 (BST), Chris Wilson wrote: > > Hi Takashi, > > On Sat, 26 Mar 2011, Takashi Iwai wrote: > > > I attached below. This is no final version and we are working on it. But > > it'd be helpful if you can test it whether it solves the issue. > > Thanks for sending the patch :) I think it makes some progress towards > solving the issue. On suspend I now get only one pop instead of two, and > none on resume. The pop that I still get on suspend is the one that causes > my amplifier to trip out (which was the second one previously). Good to hear of improvements. I cleaned up the patch a bit. Could you try the patch below instead? I removed the second msleep() in the resume callback, supposedly it's irrelevant. If you get a noise again in the resume path, try to add msleep(150) after codec->patch_ops.init() in alc_resume(). The remaining noise in the suspend path is still unclear. We have one place that we do NOT turn down the power to D3 -- in hda_set_power_state() in hda_codec.c, D3-transition is skipped on a widget with EAPD control. It's interesting whether any widgets hit this. Otherwise, we may need to check which call actually gives the noise. You can put a printk() and a long timeout at each line, and watch the kernel message.... thanks, Takashi --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 12c6f45..29eb397 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -375,6 +375,7 @@ struct alc_spec { #ifdef CONFIG_SND_HDA_POWER_SAVE void (*power_hook)(struct hda_codec *codec); #endif + void (*shutup)(struct hda_codec *codec); /* for pin sensing */ unsigned int sense_updated: 1; @@ -1236,6 +1237,15 @@ static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on) on ? 2 : 0); } +static void alc_eapd_shutup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + set_eapd(codec, 0x14, 0); + set_eapd(codec, 0x15, 0); + msleep(200); +} + static void alc_auto_init_amp(struct hda_codec *codec, int type) { unsigned int tmp; @@ -4193,6 +4203,10 @@ static int alc_build_pcms(struct hda_codec *codec) static inline void alc_shutup(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; + + if (spec && spec->shutup) + spec->shutup(codec); snd_hda_shutup_pins(codec); } @@ -4263,6 +4277,7 @@ static int alc_suspend(struct hda_codec *codec, pm_message_t state) #ifdef SND_HDA_NEEDS_RESUME static int alc_resume(struct hda_codec *codec) { + msleep(150); /* to avoid pop noise */ codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); @@ -13018,6 +13033,7 @@ static int patch_alc262(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC262_AUTO) spec->init_hook = alc262_auto_init; + spec->shutup = alc_eapd_shutup; alc_init_jacks(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -14092,6 +14108,7 @@ static int patch_alc268(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC268_AUTO) spec->init_hook = alc268_auto_init; + spec->shutup = alc_eapd_shutup; alc_init_jacks(codec); @@ -17410,6 +17418,8 @@ static int patch_alc861vd(struct hda_codec *codec) if (board_config == ALC861VD_AUTO) spec->init_hook = alc861vd_auto_init; + spec->shutup = alc_eapd_shutup; + #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc861vd_loopbacks; @@ -19611,6 +19621,7 @@ static int patch_alc662(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC662_AUTO) spec->init_hook = alc662_auto_init; + spec->shutup = alc_eapd_shutup; alc_init_jacks(codec);

1 0

[alsa-devel] snd-hda-intel + pc-speaker regression since kernel 2.6.35
by Halim Sahin 29 Mar '11

29 Mar '11

Hello all, Sorry for bringing up this topic again but in my case a working speaker is needed for accessiblity puposes. The internal pc speaker of my samsung n140 netbook worked since kernel version 2.6.34. Tested with 2.6.35/36 no pc speaker works and I see no beep control in alsamixer Today I've installed a fresh 2.6.38 kernel with same results. It would be really nice if some of you could fix this! alsamixer shows the following: Card: HDA Intel F1: Help Chip: Realtek ALC269 F2: System information Attached you find the output of alsa-info. My snd-hda-intel is build into the kernel. There no output of loaded alsa modules. BR. Halim

1 0

[alsa-devel] [PATCH 0/2] Adding support for ASoC drivers on SPEAr13XX platform
by Rajeev Kumar 29 Mar '11

29 Mar '11

This patchset contains two patches which add support for ASoC sound drivers on ST's SPEAr13XX platform. Details of the SPEAr13XX platforms can be seen here: http://www.st.com/internet/mcu/product/250658.jsp The ARCH and PLATFORM specific code for SPEAr13XX are already under review in Russell King's ARM mailing list. Please review the same and consider for mainline inclusion. Rajeev Kumar (2): sound: asoc: Adding support for SPEAr13XX ASoC driver sound: asoc: Adding support for STA529 Audio Codec sound/soc/Kconfig | 20 +-- sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta529.c | 366 ++++++++++++++++++++++++ sound/soc/codecs/sta529.h | 61 ++++ sound/soc/spear/Kconfig | 19 ++ sound/soc/spear/Makefile | 6 + sound/soc/spear/evb_sta529.c | 135 +++++++++ sound/soc/spear/spear13xx-i2s.c | 598 +++++++++++++++++++++++++++++++++++++++ sound/soc/spear/spear13xx-i2s.h | 22 ++ sound/soc/spear/spear13xx-pcm.c | 496 ++++++++++++++++++++++++++++++++ sound/soc/spear/spear13xx-pcm.h | 54 ++++ 13 files changed, 1767 insertions(+), 18 deletions(-) create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/codecs/sta529.h create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/spear/evb_sta529.c create mode 100644 sound/soc/spear/spear13xx-i2s.c create mode 100644 sound/soc/spear/spear13xx-i2s.h create mode 100644 sound/soc/spear/spear13xx-pcm.c create mode 100644 sound/soc/spear/spear13xx-pcm.h

3 15

[alsa-devel] [PATCH alsa-python] alsamixer: Fix bad PyArg_ParseTuple format strings
by Forest Bond 29 Mar '11

29 Mar '11

A number of calls to PyArg_ParseTuple use "I" for ints and "L" for long ints. Fix these to be "i" and "l" respectively. Signed-off-by: Forest Bond <forest.bond(a)rapidrollout.com> --- pyalsa/alsamixer.c | 46 +++++++++++++++++++++++----------------------- 1 files changed, 23 insertions(+), 23 deletions(-) diff --git a/pyalsa/alsamixer.c b/pyalsa/alsamixer.c index 81719a7..d686f16 100644 --- a/pyalsa/alsamixer.c +++ b/pyalsa/alsamixer.c @@ -372,7 +372,7 @@ pyalsamixerelement_ismono(struct pyalsamixerelement *pyelem, PyObject *args) { int res, dir = 0; - if (!PyArg_ParseTuple(args, "|I", &dir)) + if (!PyArg_ParseTuple(args, "|i", &dir)) return NULL; if (dir == 0) @@ -393,7 +393,7 @@ pyalsamixerelement_haschannel(struct pyalsamixerelement *pyelem, PyObject *args) { int res, dir = 0, chn = SND_MIXER_SCHN_MONO; - if (!PyArg_ParseTuple(args, "|II", &chn, &dir)) + if (!PyArg_ParseTuple(args, "|ii", &chn, &dir)) return NULL; if (dir == 0) @@ -415,7 +415,7 @@ pyalsamixerelement_hasvolume(struct pyalsamixerelement *pyelem, PyObject *args) { int res, dir = 0; - if (!PyArg_ParseTuple(args, "|I", &dir)) + if (!PyArg_ParseTuple(args, "|i", &dir)) return NULL; if (dir == 0) @@ -444,7 +444,7 @@ pyalsamixerelement_hasswitch(struct pyalsamixerelement *pyelem, PyObject *args) { int res, dir = 0; - if (!PyArg_ParseTuple(args, "|I", &dir)) + if (!PyArg_ParseTuple(args, "|i", &dir)) return NULL; if (dir == 0) @@ -473,7 +473,7 @@ pyalsamixerelement_askvoldB(struct pyalsamixerelement *pyelem, PyObject *args) int res, dir = 0; long volume, val; - if (!PyArg_ParseTuple(args, "|LI", &volume, &dir)) + if (!PyArg_ParseTuple(args, "|li", &volume, &dir)) return NULL; if (dir == 0) @@ -496,7 +496,7 @@ pyalsamixerelement_askdBvol(struct pyalsamixerelement *pyelem, PyObject *args) int res, xdir = -1, dir = 0; long dBvolume, val; - if (!PyArg_ParseTuple(args, "|LII", &dBvolume, &xdir, &dir)) + if (!PyArg_ParseTuple(args, "|lii", &dBvolume, &xdir, &dir)) return NULL; if (dir == 0) @@ -519,7 +519,7 @@ pyalsamixerelement_getvolume(struct pyalsamixerelement *pyelem, PyObject *args) int res, dir = 0, chn = SND_MIXER_SCHN_MONO; long val; - if (!PyArg_ParseTuple(args, "|II", &chn, &dir)) + if (!PyArg_ParseTuple(args, "|ii", &chn, &dir)) return NULL; if (dir == 0) @@ -543,7 +543,7 @@ pyalsamixerelement_getvolumetuple(struct pyalsamixerelement *pyelem, PyObject *a long val; PyObject *t; - if (!PyArg_ParseTuple(args, "|I", &dir)) + if (!PyArg_ParseTuple(args, "|i", &dir)) return NULL; if (dir == 0) { @@ -617,7 +617,7 @@ pyalsamixerelement_getvolumearray(struct pyalsamixerelement *pyelem, PyObject *a long val; PyObject *t, *l; - if (!PyArg_ParseTuple(args, "|I", &dir)) + if (!PyArg_ParseTuple(args, "|i", &dir)) return NULL; if (dir == 0) { @@ -694,7 +694,7 @@ pyalsamixerelement_setvolume(struct pyalsamixerelement *pyelem, PyObject *args) int res, dir = 0, chn = SND_MIXER_SCHN_MONO; long val; - if (!PyArg_ParseTuple(args, "L|II", &val, &chn, &dir)) + if (!PyArg_ParseTuple(args, "l|ii", &val, &chn, &dir)) return NULL; if (dir == 0) res = snd_mixer_selem_set_playback_volume(pyelem->elem, chn, val); @@ -718,7 +718,7 @@ pyalsamixerelement_setvolumetuple(struct pyalsamixerelement *pyelem, PyObject *a int i, res, dir = 0; long val; - if (!PyArg_ParseTuple(args, "O|I", &t, &dir)) + if (!PyArg_ParseTuple(args, "O|i", &t, &dir)) return NULL; if (!PyTuple_Check(t) && !PyList_Check(t)) return PyErr_Format(PyExc_RuntimeError, "Volume values in tuple are expected!"); @@ -770,7 +770,7 @@ pyalsamixerelement_setvolumeall(struct pyalsamixerelement *pyelem, PyObject *arg int res, dir = 0; long val; - if (!PyArg_ParseTuple(args, "L|I", &val, &dir)) + if (!PyArg_ParseTuple(args, "l|i", &val, &dir)) return NULL; if (dir == 0) res = snd_mixer_selem_set_playback_volume_all(pyelem->elem, val); @@ -791,7 +791,7 @@ pyalsamixerelement_getrange(struct pyalsamixerelement *pyelem, PyObject *args) long min, max; PyObject *t; - if (!PyArg_ParseTuple(args, "|I", &dir)) + if (!PyArg_ParseTuple(args, "|i", &dir)) return NULL; if (dir == 0) res = snd_mixer_selem_get_playback_volume_range(pyelem->elem, &min, &max); @@ -816,7 +816,7 @@ pyalsamixerelement_setrange(struct pyalsamixerelement *pyelem, PyObject *args) int dir = 0, res; long min, max; - if (!PyArg_ParseTuple(args, "LL|I", &min, &max, &dir)) + if (!PyArg_ParseTuple(args, "ll|i", &min, &max, &dir)) return NULL; if (dir == 0) res = snd_mixer_selem_set_playback_volume_range(pyelem->elem, min, max); @@ -835,7 +835,7 @@ pyalsamixerelement_getswitch(struct pyalsamixerelement *pyelem, PyObject *args) { int res, dir = 0, chn = SND_MIXER_SCHN_MONO, val; - if (!PyArg_ParseTuple(args, "|II", &chn, &dir)) + if (!PyArg_ParseTuple(args, "|ii", &chn, &dir)) return NULL; if (dir == 0) @@ -862,7 +862,7 @@ pyalsamixerelement_getswitchtuple(struct pyalsamixerelement *pyelem, PyObject *a int res, dir = 0, i, last, val; PyObject *t; - if (!PyArg_ParseTuple(args, "|I", &dir)) + if (!PyArg_ParseTuple(args, "|i", &dir)) return NULL; if (dir == 0) { @@ -934,7 +934,7 @@ pyalsamixerelement_setswitch(struct pyalsamixerelement *pyelem, PyObject *args) { int res, dir = 0, chn = SND_MIXER_SCHN_MONO, val; - if (!PyArg_ParseTuple(args, "I|II", &val, &chn, &dir)) + if (!PyArg_ParseTuple(args, "i|ii", &val, &chn, &dir)) return NULL; if (dir == 0) res = snd_mixer_selem_set_playback_switch(pyelem->elem, chn, val); @@ -954,7 +954,7 @@ pyalsamixerelement_setswitchtuple(struct pyalsamixerelement *pyelem, PyObject *a PyObject *t, *o; int i, res, dir = 0, val; - if (!PyArg_ParseTuple(args, "O|I", &t, &dir)) + if (!PyArg_ParseTuple(args, "O|i", &t, &dir)) return NULL; if (!PyTuple_Check(t)) return PyErr_Format(PyExc_RuntimeError, "Switch state values in tuple are expected!"); @@ -982,7 +982,7 @@ pyalsamixerelement_setswitchall(struct pyalsamixerelement *pyelem, PyObject *arg { int res, dir = 0, val; - if (!PyArg_ParseTuple(args, "I|I", &val, &dir)) + if (!PyArg_ParseTuple(args, "i|i", &val, &dir)) return NULL; if (dir == 0) res = snd_mixer_selem_set_playback_switch_all(pyelem->elem, val); @@ -1002,7 +1002,7 @@ pyalsamixerelement_getvolumedb(struct pyalsamixerelement *pyelem, PyObject *args int res, dir = 0, chn = SND_MIXER_SCHN_MONO; long val; - if (!PyArg_ParseTuple(args, "|II", &chn, &dir)) + if (!PyArg_ParseTuple(args, "|ii", &chn, &dir)) return NULL; if (dir == 0) @@ -1025,7 +1025,7 @@ pyalsamixerelement_setvolumedb(struct pyalsamixerelement *pyelem, PyObject *args int res, dir = 0, dir1 = 0, chn = SND_MIXER_SCHN_MONO; long val; - if (!PyArg_ParseTuple(args, "L|III", &val, &chn, &dir, &dir1)) + if (!PyArg_ParseTuple(args, "l|iii", &val, &chn, &dir, &dir1)) return NULL; if (dir == 0) res = snd_mixer_selem_set_playback_dB(pyelem->elem, chn, val, dir1); @@ -1045,7 +1045,7 @@ pyalsamixerelement_setvolumealldb(struct pyalsamixerelement *pyelem, PyObject *a int res, dir = 0, dir1 = 0; long val; - if (!PyArg_ParseTuple(args, "L|II", &val, &dir, &dir1)) + if (!PyArg_ParseTuple(args, "l|ii", &val, &dir, &dir1)) return NULL; if (dir == 0) res = snd_mixer_selem_set_playback_dB_all(pyelem->elem, val, dir1); @@ -1066,7 +1066,7 @@ pyalsamixerelement_getrangedb(struct pyalsamixerelement *pyelem, PyObject *args) long min, max; PyObject *t; - if (!PyArg_ParseTuple(args, "|I", &dir)) + if (!PyArg_ParseTuple(args, "|i", &dir)) return NULL; if (dir == 0) res = snd_mixer_selem_get_playback_dB_range(pyelem->elem, &min, &max); -- 1.7.0.4

2 1

[alsa-devel] capture file size is 44 bytes only from "arecord audio.wav"
by hong zhang 29 Mar '11

29 Mar '11

List, I create a HDMI interface for TI davinic SoC. But capture file, my.wav, size is only 44 bytes after "arecord my.wav". 1. "arecord -l" display interface as: root@c6a816x-evm:~# arecord -l **** List of CAPTURE Hardware Devices **** card 0: hdmi [TI81XX hdmi], device 0: HDMI McASP Receive hdmi_sii9135-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 2. root@c6a816x-evm:/dev/snd# ls controlC0 pcmC0D0c timer I check sound/soc/davinci and see snd_pcm_lib_ioctl() is useless. Could anyone please give me hint where to start debug? Thanks. ---henry

1 0

[alsa-devel] Problems with snd_hda_intel in Linux kernel 2.6.38
by Svante Signell 29 Mar '11

29 Mar '11

Trying again to get this message to the alsa-devel list, as recommended by the Debian maintainer of the kernel, without being subscribed. Does alsa not have a bug report page? Booting kernel 2.6.38 hangs at: During boot of kernel 2.6.38 (and 2.6.37) udev bugs out: Waiting for /dev to be fully populated BUG: Unable to handle kernel paging request at ffffc90013cd8000 axz_probe+ ... [snd_hda_intel] ...lots of output lost... udevadm timeout 180 sec ... udevd[390]: worker [439] failed while handling '/devices/pci0000:80/0000:80:01.0' No sound card is recognized after the timeout of the boot as shown above. According to the /proc/asound/cards there are no sound cards See bug Debian bug #619034 for more information. http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=619034 This problem was also reported as Debian bug #613979, as kernel ticket https://bugzilla.kernel.org/show_bug.cgi?id=30552 and sent to this mailing list in February 2011. snd_hda_intel works properly with 2.6.32-5-amd64 (and earlier kernels).

1 0

[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 28 Mar '11

28 Mar '11

The following changes since commit 333802e90d3f0366c4a1cb767e2783d2e1df73a8: ASoC: Support !REGULATOR build for sgtl5000 (2011-03-22 18:26:30 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Mark Brown (1): ASoC: Explicitly say registerless widgets have no register Wolfram Sang (2): ASoC: imx: set watermarks for mx2-dma ASoC: imx: fix burstsize for DMA include/sound/soc-dapm.h | 16 ++++++++-------- sound/soc/imx/imx-pcm-dma-mx2.c | 9 +++++++-- sound/soc/imx/imx-ssi.h | 3 --- 3 files changed, 15 insertions(+), 13 deletions(-)

2 1

[alsa-devel] [PATCH 1/2] ASoC: soc-cache: Fix indentation issues
by Dimitris Papastamos 28 Mar '11

28 Mar '11

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 18 +++++++++--------- 1 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 1210a6f..22c0b95 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -93,13 +93,13 @@ static unsigned int do_hw_read(struct snd_soc_codec *codec, unsigned int reg) } static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, - unsigned int reg) + unsigned int reg) { return do_hw_read(codec, reg); } static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) + unsigned int value) { u8 data[2]; @@ -111,7 +111,7 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, #if defined(CONFIG_SPI_MASTER) static int snd_soc_4_12_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[2]; @@ -209,7 +209,7 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, #if defined(CONFIG_SPI_MASTER) static int snd_soc_8_16_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[3]; @@ -257,7 +257,7 @@ static unsigned int do_i2c_read(struct snd_soc_codec *codec, #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) + unsigned int r) { u8 reg = r; u8 data; @@ -307,13 +307,13 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, #endif static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, - unsigned int reg) + unsigned int reg) { return do_hw_read(codec, reg); } static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) + unsigned int value) { u8 data[3]; @@ -327,7 +327,7 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, #if defined(CONFIG_SPI_MASTER) static int snd_soc_16_8_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[3]; @@ -379,7 +379,7 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, #if defined(CONFIG_SPI_MASTER) static int snd_soc_16_16_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[4]; -- 1.7.4.1

1 1

[alsa-devel] getting dynamic state from ac97 bus ops
by Mike Frysinger 28 Mar '11

28 Mar '11

is there an obvious way that i'm missing to get at the bus state from the funcs in the global soc_ac97_ops structure ? the ac97 bus drivers that i see in the tree simply have a global variable for state and that sort of defeats the purpose of having multi-instance support in the first place ... -mike

2 3

[alsa-devel] [PATCH] hdspconf: more string constant fixes
by Tim Blechmann 28 Mar '11

28 Mar '11

Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- hdspconf/pixmaps/alsalogo.xpm | 2 +- hdspconf/pixmaps/lad_banner.xpm | 2 +- hdspconf/pixmaps/rme.xpm | 2 +- hdspconf/src/HC_PrefSyncRef.h | 6 +++--- hdspconf/src/HC_XpmRenderer.cxx | 6 +++--- hdspconf/src/HC_XpmRenderer.h | 8 ++++---- hdspconf/src/pixmaps.h | 10 +++++----- 7 files changed, 18 insertions(+), 18 deletions(-) diff --git a/hdspconf/pixmaps/alsalogo.xpm b/hdspconf/pixmaps/alsalogo.xpm index b32c32d..cd22dd2 100644 --- a/hdspconf/pixmaps/alsalogo.xpm +++ b/hdspconf/pixmaps/alsalogo.xpm @@ -1,5 +1,5 @@ /* XPM */ -char * alsalogo_xpm[] = { +const char * alsalogo_xpm[] = { "50 50 398 2", " c None", ". c #C0C0C0", diff --git a/hdspconf/pixmaps/lad_banner.xpm b/hdspconf/pixmaps/lad_banner.xpm index 8bb8545..11d4357 100644 --- a/hdspconf/pixmaps/lad_banner.xpm +++ b/hdspconf/pixmaps/lad_banner.xpm @@ -1,5 +1,5 @@ /* XPM */ -char * lad_banner_xpm[] = { +const char * lad_banner_xpm[] = { "113 39 1833 2", " c None", ". c #C0C0C0", diff --git a/hdspconf/pixmaps/rme.xpm b/hdspconf/pixmaps/rme.xpm index a892603..b94fa7e 100644 --- a/hdspconf/pixmaps/rme.xpm +++ b/hdspconf/pixmaps/rme.xpm @@ -1,5 +1,5 @@ /* XPM */ -char * rme_xpm[] = { +const char * rme_xpm[] = { "113 35 2300 2", " c None", ". c #030505", diff --git a/hdspconf/src/HC_PrefSyncRef.h b/hdspconf/src/HC_PrefSyncRef.h index 5ef8875..8c3a71f 100644 --- a/hdspconf/src/HC_PrefSyncRef.h +++ b/hdspconf/src/HC_PrefSyncRef.h @@ -1,8 +1,8 @@ /* * HDSPConf - * + * * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) - * + * * 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 of the License, or @@ -42,7 +42,7 @@ public: Fl_Round_Button *adat_sync; void setRef(int r); private: - char *adat_name; + const char *adat_name; }; #endif diff --git a/hdspconf/src/HC_XpmRenderer.cxx b/hdspconf/src/HC_XpmRenderer.cxx index 72742f8..ed66d4d 100644 --- a/hdspconf/src/HC_XpmRenderer.cxx +++ b/hdspconf/src/HC_XpmRenderer.cxx @@ -1,8 +1,8 @@ /* * HDSPConf - * + * * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) - * + * * 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 of the License, or @@ -21,7 +21,7 @@ #pragma implementation #include "HC_XpmRenderer.h" -HC_XpmRenderer::HC_XpmRenderer(int x, int y, int w, int h, char * const *xpm):Fl_Widget(x, y, w, h) +HC_XpmRenderer::HC_XpmRenderer(int x, int y, int w, int h, const char * const *xpm):Fl_Widget(x, y, w, h) { pixmap = xpm; } diff --git a/hdspconf/src/HC_XpmRenderer.h b/hdspconf/src/HC_XpmRenderer.h index 3c6f768..1e18541 100644 --- a/hdspconf/src/HC_XpmRenderer.h +++ b/hdspconf/src/HC_XpmRenderer.h @@ -1,8 +1,8 @@ /* * HDSPConf - * + * * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) - * + * * 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 of the License, or @@ -30,10 +30,10 @@ class HC_XpmRenderer:public Fl_Widget { public: - HC_XpmRenderer(int x, int y, int w, int h, char * const *xpm); + HC_XpmRenderer(int x, int y, int w, int h, const char * const *xpm); void draw(); private: - char * const *pixmap; + const char * const *pixmap; }; #endif diff --git a/hdspconf/src/pixmaps.h b/hdspconf/src/pixmaps.h index c193ed2..4c5aa65 100644 --- a/hdspconf/src/pixmaps.h +++ b/hdspconf/src/pixmaps.h @@ -1,8 +1,8 @@ /* * HDSPConf - * + * * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) - * + * * 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 of the License, or @@ -21,9 +21,9 @@ #ifndef pixmaps_H #define pixmaps_H -extern char * rme_xpm[]; -extern char * alsalogo_xpm[]; -extern char * lad_banner_xpm[]; +extern const char * rme_xpm[]; +extern const char * alsalogo_xpm[]; +extern const char * lad_banner_xpm[]; #endif -- 1.7.1

2 1

[alsa-devel] [PATCH 1/2] hdspconf: fix some string constant conversions
by Tim Blechmann 28 Mar '11

28 Mar '11

Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- hdspconf/src/HC_BreakoutCable.cxx | 8 ++++---- hdspconf/src/HC_CardPane.cxx | 20 ++++++++++---------- hdspconf/src/HC_ClockSource.cxx | 14 +++++++------- hdspconf/src/HC_SpdifFreq.cxx | 6 +++--- hdspconf/src/HC_SyncCheck.cxx | 8 ++++---- hdspconf/src/HC_SystemClock.cxx | 6 +++--- hdspconf/src/labels.cxx | 12 ++++++------ hdspconf/src/labels.h | 12 ++++++------ 8 files changed, 43 insertions(+), 43 deletions(-) diff --git a/hdspconf/src/HC_BreakoutCable.cxx b/hdspconf/src/HC_BreakoutCable.cxx index 89d3ec8..0d7edcb 100644 --- a/hdspconf/src/HC_BreakoutCable.cxx +++ b/hdspconf/src/HC_BreakoutCable.cxx @@ -1,8 +1,8 @@ /* * HDSPConf - * + * * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) - * + * * 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 of the License, or @@ -21,7 +21,7 @@ #pragma implementation #include "HC_BreakoutCable.h" -static void setXlrStatus(char *ctl_name, int val, int card_index) +static void setXlrStatus(const char *ctl_name, int val, int card_index) { int err; char card_name[6]; @@ -67,7 +67,7 @@ HC_BreakoutCable::HC_BreakoutCable(int x, int y, int w, int h):Fl_Group(x, y, w, xlr = new Fl_Check_Button(x+15, y, w-30, 20, "XLR"); xlr->labelsize(10); xlr->callback(xlr_cb, (void *)parent()); - end(); + end(); } void HC_BreakoutCable::setXlr(unsigned char val) diff --git a/hdspconf/src/HC_CardPane.cxx b/hdspconf/src/HC_CardPane.cxx index 8371dbc..1e346b7 100644 --- a/hdspconf/src/HC_CardPane.cxx +++ b/hdspconf/src/HC_CardPane.cxx @@ -1,8 +1,8 @@ /* * HDSPConf - * + * * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) - * + * * 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 of the License, or @@ -21,7 +21,7 @@ #pragma implementation #include "HC_CardPane.h" -extern char *card_names[5]; +extern const char *card_names[5]; HC_CardPane::HC_CardPane(int alsa_idx, int idx, HDSP_IO_Type t):Fl_Group(PANE_X, PANE_Y, PANE_W, PANE_H) { @@ -39,20 +39,20 @@ HC_CardPane::HC_CardPane(int alsa_idx, int idx, HDSP_IO_Type t):Fl_Group(PANE_X, spdif_in = new HC_SpdifIn(x()+166, y()+20, 148, V_STEP*3); spdif_out = new HC_SpdifOut(x()+166, y()+40+V_STEP*3, 148, V_STEP*4); spdif_freq = new HC_SpdifFreq(x()+166, y()+60+V_STEP*7, 148, V_STEP); - + sync_ref = new HC_PrefSyncRef(x()+323, y()+20, 148, V_STEP*4); autosync_ref = new HC_AutoSyncRef(x()+323, y()+40+V_STEP*4, 148, V_STEP*2); system_clock = new HC_SystemClock(x()+323, y()+60+V_STEP*6, 148, V_STEP*2); } else if (type == Digiface) { - + clock_source = new HC_ClockSource(x()+9, y()+20, 148, V_STEP*7); sync_check = new HC_SyncCheck(x()+9, y()+40+V_STEP*7, 148, V_STEP*6); spdif_in = new HC_SpdifIn(x()+166, y()+20, 148, V_STEP*3); spdif_out = new HC_SpdifOut(x()+166, y()+40+V_STEP*3, 148, V_STEP*4); spdif_freq = new HC_SpdifFreq(x()+166, y()+60+V_STEP*7, 148, V_STEP); - + sync_ref = new HC_PrefSyncRef(x()+323, y()+20, 148, V_STEP*6); autosync_ref = new HC_AutoSyncRef(x()+323, y()+40+V_STEP*6, 148, V_STEP*2); system_clock = new HC_SystemClock(x()+323, y()+60+V_STEP*8, 148, V_STEP*2); @@ -65,7 +65,7 @@ HC_CardPane::HC_CardPane(int alsa_idx, int idx, HDSP_IO_Type t):Fl_Group(PANE_X, spdif_in = new HC_SpdifIn(x()+166, y()+20, 148, V_STEP*3); spdif_out = new HC_SpdifOut(x()+166, y()+40+V_STEP*3, 148, V_STEP*4); spdif_freq = new HC_SpdifFreq(x()+166, y()+60+V_STEP*7, 148, V_STEP); - + aeb = new HC_Aeb(x()+323, y()+20, 148, V_STEP); sync_ref = new HC_PrefSyncRef(x()+323, y()+40+V_STEP, 148, V_STEP*6); autosync_ref = new HC_AutoSyncRef(x()+323, y()+60+V_STEP*7, 148, V_STEP*2); @@ -78,17 +78,17 @@ HC_CardPane::HC_CardPane(int alsa_idx, int idx, HDSP_IO_Type t):Fl_Group(PANE_X, spdif_in = new HC_SpdifIn(x()+126, y()+20, 110, V_STEP*4); spdif_out = new HC_SpdifOut(x()+126, y()+40+V_STEP*4, 110, V_STEP*4); spdif_freq = new HC_SpdifFreq(x()+126, y()+60+V_STEP*8, 110, V_STEP); - + aeb = new HC_Aeb(x()+244, y()+20, 110, V_STEP); sync_ref = new HC_PrefSyncRef(x()+244, y()+40+V_STEP, 110, V_STEP*3); autosync_ref = new HC_AutoSyncRef(x()+244, y()+60+V_STEP*4, 110, V_STEP*2); system_clock = new HC_SystemClock(x()+244, y()+80+V_STEP*6, 110, V_STEP*2); - + breakout_cable = new HC_BreakoutCable(x()+362, y()+20, 110, V_STEP); input_level = new HC_InputLevel(x()+362, y()+40+V_STEP, 110, V_STEP*3); output_level = new HC_OutputLevel(x()+362, y()+60+V_STEP*4, 110, V_STEP*3); phones = new HC_Phones(x()+362, y()+80+V_STEP*7, 110, V_STEP*3); - + } end(); } diff --git a/hdspconf/src/HC_ClockSource.cxx b/hdspconf/src/HC_ClockSource.cxx index 4bb978a..d7b2a07 100644 --- a/hdspconf/src/HC_ClockSource.cxx +++ b/hdspconf/src/HC_ClockSource.cxx @@ -1,8 +1,8 @@ /* * HDSPConf - * + * * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) - * + * * 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 of the License, or @@ -21,7 +21,7 @@ #pragma implementation #include "HC_ClockSource.h" -extern char *freqs[10]; +extern const char *freqs[10]; void clock_source_cb(Fl_Widget *w, void *arg) { @@ -55,7 +55,7 @@ void clock_source_cb(Fl_Widget *w, void *arg) } else if (source == cs->khz192) { src = 9; } - + snprintf(card_name, 6, "hw:%i", pane->alsa_index); snd_ctl_elem_value_alloca(&ctl); snd_ctl_elem_id_alloca(&id); @@ -67,7 +67,7 @@ void clock_source_cb(Fl_Widget *w, void *arg) if ((err = snd_ctl_open(&handle, card_name, SND_CTL_NONBLOCK)) < 0) { fprintf(stderr, "Error opening ctl interface on card %s\n", card_name); return; - } + } if ((err = snd_ctl_elem_write(handle, ctl)) < 0) { snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_PCM); snd_ctl_elem_value_set_id(ctl, id); @@ -111,7 +111,7 @@ HC_ClockSource::HC_ClockSource(int x, int y, int w, int h):Fl_Group(x, y, w, h, khz176_4->labelsize(10); khz176_4->type(FL_RADIO_BUTTON); khz192 = new Fl_Round_Button(x+10, y+V_STEP*i++, w-20, V_STEP, freqs[9]); - khz192->callback(clock_source_cb, (void *)this); + khz192->callback(clock_source_cb, (void *)this); khz192->labelsize(10); khz192->type(FL_RADIO_BUTTON); } @@ -129,7 +129,7 @@ HC_ClockSource::HC_ClockSource(int x, int y, int w, int h):Fl_Group(x, y, w, h, khz88_2->type(FL_RADIO_BUTTON); khz96->labelsize(10); khz96->type(FL_RADIO_BUTTON); - end(); + end(); } void HC_ClockSource::setSource(unsigned char s) diff --git a/hdspconf/src/HC_SpdifFreq.cxx b/hdspconf/src/HC_SpdifFreq.cxx index 0fed38f..b607d65 100644 --- a/hdspconf/src/HC_SpdifFreq.cxx +++ b/hdspconf/src/HC_SpdifFreq.cxx @@ -1,8 +1,8 @@ /* * HDSPConf - * + * * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) - * + * * 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 of the License, or @@ -21,7 +21,7 @@ #pragma implementation #include "HC_SpdifFreq.h" -extern char *freqs[10]; +extern const char *freqs[10]; HC_SpdifFreq::HC_SpdifFreq(int x, int y, int w, int h):Fl_Widget(x, y, w, h, "SPDIF Freq.") { diff --git a/hdspconf/src/HC_SyncCheck.cxx b/hdspconf/src/HC_SyncCheck.cxx index 4a088ba..111c9bc 100644 --- a/hdspconf/src/HC_SyncCheck.cxx +++ b/hdspconf/src/HC_SyncCheck.cxx @@ -1,8 +1,8 @@ /* * HDSPConf - * + * * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) - * + * * 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 of the License, or @@ -21,7 +21,7 @@ #pragma implementation #include "HC_SyncCheck.h" -extern char *lock_status[3]; +extern const char *lock_status[3]; HC_SyncCheck::HC_SyncCheck(int x, int y, int w, int h):Fl_Widget(x, y, w, h, "SyncCheck") { @@ -35,7 +35,7 @@ HC_SyncCheck::HC_SyncCheck(int x, int y, int w, int h):Fl_Widget(x, y, w, h, "Sy adat_name = "ADAT In"; } else { adat_name = "ADAT1 In"; - } + } h_step = (int)(w/2.0f); draw_box = Fl::get_boxtype(FL_ENGRAVED_FRAME); label("SyncCheck"); diff --git a/hdspconf/src/HC_SystemClock.cxx b/hdspconf/src/HC_SystemClock.cxx index f1f88bb..d549a6f 100644 --- a/hdspconf/src/HC_SystemClock.cxx +++ b/hdspconf/src/HC_SystemClock.cxx @@ -1,8 +1,8 @@ /* * HDSPConf - * + * * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) - * + * * 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 of the License, or @@ -21,7 +21,7 @@ #pragma implementation #include "HC_SystemClock.h" -extern char *freqs[10]; +extern const char *freqs[10]; HC_SystemClock::HC_SystemClock(int x, int y, int w, int h):Fl_Widget(x, y, w, h, "System Clock") { diff --git a/hdspconf/src/labels.cxx b/hdspconf/src/labels.cxx index 59880b6..f8900be 100644 --- a/hdspconf/src/labels.cxx +++ b/hdspconf/src/labels.cxx @@ -1,8 +1,8 @@ /* * HDSPConf - * + * * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) - * + * * 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 of the License, or @@ -20,7 +20,7 @@ #include "labels.h" -char *card_names[5] = { +const char *card_names[5] = { "Digiface", "Multiface", "HDSP9652", @@ -28,7 +28,7 @@ char *card_names[5] = { "Undefined", }; -char *freqs[10] = { +const char *freqs[10] = { "32.0 kHz", "44.1 kHz", "48.0 kHz", @@ -41,7 +41,7 @@ char *freqs[10] = { "192.0 kHz", }; -char *ref[7] = { +const char *ref[7] = { "Word Clock", "ADAT Sync", "SPDIF", @@ -51,7 +51,7 @@ char *ref[7] = { "ADAT3" }; -char *lock_status[3] = { +const char *lock_status[3] = { "No Lock", "Lock", "Sync" diff --git a/hdspconf/src/labels.h b/hdspconf/src/labels.h index 42964c0..da996f9 100644 --- a/hdspconf/src/labels.h +++ b/hdspconf/src/labels.h @@ -1,8 +1,8 @@ /* * HDSPConf - * + * * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) - * + * * 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 of the License, or @@ -21,10 +21,10 @@ #ifndef LABELS_H #define LABELS_H -extern char *card_names[5]; -extern char *freqs[10]; -extern char *ref[7]; -extern char *lock_status[3]; +extern const char *card_names[5]; +extern const char *freqs[10]; +extern const char *ref[7]; +extern const char *lock_status[3]; #endif -- 1.7.1

2 2

[alsa-devel] [GIT PULL] ASoC fixes for 2.6.39
by Liam Girdwood 28 Mar '11

28 Mar '11

Hi Takashi, Some TI codec fixes for 2.6.39. Thanks Liam --- The following changes since commit fa4d1f595a89234647b34a7f507436be873091bc: ASoC: omap: rx51: Enable McBSP2 sidetone (2011-03-09 10:52:28 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.39 Axel Lin (1): ASoC: twl6040: Return -ENOMEM if create_singlethread_workqueue fails Peter Ujfalusi (2): ASoC: tlv320dac33: Fix inconsistent spinlock usage ASoC: tlv320dac33: Restore L/R DAC power control register sound/soc/codecs/tlv320dac33.c | 32 ++++++++++++++++++++------------ sound/soc/codecs/twl6040.c | 4 +++- 2 files changed, 23 insertions(+), 13 deletions(-)

2 1

[alsa-devel] [PATCH 1/3] ASoC: Add default snd_soc_default_writable_register() callback
by Dimitris Papastamos 28 Mar '11

28 Mar '11

By using struct snd_soc_reg_access for the read/write/vol attributes of the registers, we provide callbacks that automatically determine whether a given register is readable/writable or volatile. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 4 ++++ sound/soc/soc-cache.c | 14 ++++++++++++++ sound/soc/soc-core.c | 3 +++ 3 files changed, 21 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index bfa4836..982e63e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -292,6 +292,8 @@ int snd_soc_default_volatile_register(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_default_readable_register(struct snd_soc_codec *codec, unsigned int reg); +int snd_soc_default_writable_register(struct snd_soc_codec *codec, + unsigned int reg); /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); @@ -523,6 +525,7 @@ struct snd_soc_codec { size_t reg_size; /* reg_cache_size * reg_word_size */ int (*volatile_register)(struct snd_soc_codec *, unsigned int); int (*readable_register)(struct snd_soc_codec *, unsigned int); + int (*writable_register)(struct snd_soc_codec *, unsigned int); /* runtime */ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ @@ -587,6 +590,7 @@ struct snd_soc_codec_driver { size_t, unsigned int); int (*volatile_register)(struct snd_soc_codec *, unsigned int); int (*readable_register)(struct snd_soc_codec *, unsigned int); + int (*writable_register)(struct snd_soc_codec *, unsigned int); short reg_cache_size; short reg_cache_step; short reg_word_size; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 5d76da4..657889d 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1677,3 +1677,17 @@ int snd_soc_default_readable_register(struct snd_soc_codec *codec, return codec->driver->reg_access_default[index].read; } EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); + +int snd_soc_default_writable_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].write; +} +EXPORT_SYMBOL_GPL(snd_soc_default_writable_register); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4dda589..1cb1446 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3666,6 +3666,7 @@ int snd_soc_register_codec(struct device *dev, codec->read = codec_drv->read; codec->volatile_register = codec_drv->volatile_register; codec->readable_register = codec_drv->readable_register; + codec->writable_register = codec_drv->writable_register; codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.dev = dev; codec->dapm.codec = codec; @@ -3700,6 +3701,8 @@ int snd_soc_register_codec(struct device *dev, codec->volatile_register = snd_soc_default_volatile_register; if (!codec->readable_register) codec->readable_register = snd_soc_default_readable_register; + if (!codec->writable_register) + codec->writable_register = snd_soc_default_writable_register; } for (i = 0; i < num_dai; i++) { -- 1.7.4.1

3 6

[alsa-devel] [PATCH] alsa-lib: Conf: Fix channels of ICE1712 to 10/12
by David Henningsson 28 Mar '11

28 Mar '11

I know this issue has been brought up before, without resolution. Basically PulseAudio cannot open cards such as M-Audio 2496 and M-Audio Delta 1010LT because it does not try to open "front" with 10 channels, only with 2. I think it is reasonable to assume that a front:x stream should be able to open with 2 channels only, therefore I'm proposing this fix, and I'm applying it to Ubuntu as well, unless you have a better idea. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

3 2

[PATCH] Conf: Fix channels of ICE1712 to 10/12
by David Henningsson 28 Mar '11

28 Mar '11

The ICE1712 kernel driver accepts 10 playback channels and 12 recording channels only. Therefore fix this channel count in alsa-lib. BugLink: http://bugs.launchpad.net/bugs/178442 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/conf/cards/ICE1712.conf | 28 ++++++++++++++++++++++------ 1 files changed, 22 insertions(+), 6 deletions(-) diff --git a/src/conf/cards/ICE1712.conf b/src/conf/cards/ICE1712.conf index 01e50d2..f5b8add 100644 --- a/src/conf/cards/ICE1712.conf +++ b/src/conf/cards/ICE1712.conf @@ -32,12 +32,26 @@ ICE1712.pcm.front.0 { @args.CARD { type string } - type route - ttable.0.0 1 - ttable.1.1 1 - slave.pcm { - type hw - card $CARD + type asym + playback.pcm { + type route + ttable.0.0 1 + ttable.1.1 1 + slave.pcm { + type hw + card $CARD + } + slave.channels 10 + } + capture.pcm { + type route + ttable.0.0 1 + ttable.1.1 1 + slave.pcm { + type hw + card $CARD + } + slave.channels 12 } } @@ -57,6 +71,7 @@ ICE1712.pcm.surround40.0 { type hw card $CARD } + slave.channels 10 } <confdir:pcm/surround41.conf> @@ -79,6 +94,7 @@ ICE1712.pcm.surround51.0 { type hw card $CARD } + slave.channels 10 } <confdir:pcm/iec958.conf> -- 1.7.4.1 --------------000103060007030806090305--

1 0

[alsa-devel] [PATCH] ASoC: add SSM2604 codec driver
by Mike Frysinger 27 Mar '11

27 Mar '11

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2604.c | 643 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ssm2604.h | 92 +++++++ 4 files changed, 741 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ssm2604.c create mode 100644 sound/soc/codecs/ssm2604.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5da30eb..5db82d2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -28,6 +28,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C + select SND_SOC_SSM2604 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -150,6 +151,9 @@ config SND_SOC_SPDIF config SND_SOC_SSM2602 tristate +config SND_SOC_SSM2604 + tristate + config SND_SOC_STAC9766 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 91429ea..7593eeb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -16,6 +16,7 @@ snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-ssm2604-objs := ssm2604.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -81,6 +82,7 @@ obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_SSM2604) += snd-soc-ssm2604.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o diff --git a/sound/soc/codecs/ssm2604.c b/sound/soc/codecs/ssm2604.c new file mode 100644 index 0000000..7522586 --- /dev/null +++ b/sound/soc/codecs/ssm2604.c @@ -0,0 +1,643 @@ +/* + * Driver for ssm2604 sound codec + * + * Copyright 2010 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ssm2604.h" + +#define SSM2604_VERSION "0.1" + +struct snd_soc_codec_device soc_codec_dev_ssm2604; +static struct snd_soc_codec *ssm2604_codec; +/* codec private data */ +struct ssm2604_priv { + unsigned int sysclk; + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; + struct snd_soc_codec codec; +}; + +/* + * ssm2604 register cache + * We can't read the ssm2604 register space when we are + * using 2 wire for device control, so we cache them instead. + * There is no point in caching the reset register + */ +static const u16 ssm2604_reg[SSM2604_CACHEREGNUM] = { + 0x0017, 0x0017, 0x0000, 0x0000, + 0x0000, 0x000a, 0x0000, 0x0000 +}; + +#define ssm2604_reset(c) snd_soc_write(c, SSM2604_RESET, 0) + +static const char *ssm2604_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; + +static const struct soc_enum ssm2604_enum[] = { + SOC_ENUM_SINGLE(SSM2604_APDIGI, 1, 4, ssm2604_deemph), +}; + +static const struct snd_kcontrol_new ssm2604_snd_controls[] = { + +SOC_DOUBLE_R("Capture Volume", SSM2604_LINVOL, SSM2604_RINVOL, 0, 31, 0), +SOC_DOUBLE_R("Capture Switch", SSM2604_LINVOL, SSM2604_RINVOL, 7, 1, 1), + +SOC_SINGLE("ADC High Pass Filter Switch", SSM2604_APDIGI, 0, 1, 1), +SOC_SINGLE("Store DC Offset Switch", SSM2604_APDIGI, 4, 1, 0), + +SOC_ENUM("Capture Source", ssm2604_enum[0]), + +SOC_ENUM("Playback De-emphasis", ssm2604_enum[1]), +}; + +/* Output Mixer */ +static const struct snd_kcontrol_new ssm2604_output_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", SSM2604_APANA, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget ssm2604_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Output Mixer", SSM2604_PWR, 4, 1, + &ssm2604_output_mixer_controls[0], + ARRAY_SIZE(ssm2604_output_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2604_PWR, 3, 1), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2604_PWR, 2, 1), +SND_SOC_DAPM_PGA("Line Input", SSM2604_PWR, 0, 1, NULL, 0), +SND_SOC_DAPM_INPUT("RLINEIN"), +SND_SOC_DAPM_INPUT("LLINEIN"), +}; + +static const struct snd_soc_dapm_route audio_conn[] = { + /* output mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "HiFi Playback Switch", "DAC"}, + + /* outputs */ + {"ROUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + + /* input mux */ + {"Input Mux", "Line", "Line Input"}, + {"ADC", NULL, "Input Mux"}, + + /* inputs */ + {"Line Input", NULL, "LLINEIN"}, + {"Line Input", NULL, "RLINEIN"}, +}; + +static int ssm2604_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ssm2604_dapm_widgets, + ARRAY_SIZE(ssm2604_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn)); + + return 0; +} + +struct _coeff_div { + u32 mclk; + u32 rate; + u16 fs; + u8 sr:4; + u8 bosr:1; + u8 usb:1; +}; + +/* codec mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 48k */ + {12288000, 48000, 256, 0x0, 0x0, 0x0}, + {18432000, 48000, 384, 0x0, 0x1, 0x0}, + {12000000, 48000, 250, 0x0, 0x0, 0x1}, + + /* 32k */ + {12288000, 32000, 384, 0x6, 0x0, 0x0}, + {18432000, 32000, 576, 0x6, 0x1, 0x0}, + {12000000, 32000, 375, 0x6, 0x0, 0x1}, + + /* 8k */ + {12288000, 8000, 1536, 0x3, 0x0, 0x0}, + {18432000, 8000, 2304, 0x3, 0x1, 0x0}, + {11289600, 8000, 1408, 0xb, 0x0, 0x0}, + {16934400, 8000, 2112, 0xb, 0x1, 0x0}, + {12000000, 8000, 1500, 0x3, 0x0, 0x1}, + + /* 96k */ + {12288000, 96000, 128, 0x7, 0x0, 0x0}, + {18432000, 96000, 192, 0x7, 0x1, 0x0}, + {12000000, 96000, 125, 0x7, 0x0, 0x1}, + + /* 44.1k */ + {11289600, 44100, 256, 0x8, 0x0, 0x0}, + {16934400, 44100, 384, 0x8, 0x1, 0x0}, + {12000000, 44100, 272, 0x8, 0x1, 0x1}, + + /* 88.2k */ + {11289600, 88200, 128, 0xf, 0x0, 0x0}, + {16934400, 88200, 192, 0xf, 0x1, 0x0}, + {12000000, 88200, 136, 0xf, 0x1, 0x1}, +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return i; +} + +static int ssm2604_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + u16 srate; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct ssm2604_priv *ssm2604 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *i2c = codec->control_data; + u16 iface = snd_soc_read(codec, SSM2604_IFACE) & 0xfff3; + int i = get_coeff(ssm2604->sysclk, params_rate(params)); + + if (substream == ssm2604->slave_substream) { + dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); + return 0; + } + + /*no match is found*/ + if (i == ARRAY_SIZE(coeff_div)) + return -EINVAL; + + srate = (coeff_div[i].sr << 2) | + (coeff_div[i].bosr << 1) | coeff_div[i].usb; + + snd_soc_write(codec, SSM2604_ACTIVE, 0); + snd_soc_write(codec, SSM2604_SRATE, srate); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x000c; + break; + } + snd_soc_write(codec, SSM2604_IFACE, iface); + snd_soc_write(codec, SSM2604_ACTIVE, ACTIVE_ACTIVATE_CODEC); + return 0; +} + +static int ssm2604_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct ssm2604_priv *ssm2604 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *i2c = codec->control_data; + struct snd_pcm_runtime *master_runtime; + + /* The DAI has shared clocks so if we already have a playback or + * capture going then constrain this substream to match it. + * TODO: the ssm2604 allows pairs of non-matching PB/REC rates + */ + if (ssm2604->master_substream) { + master_runtime = ssm2604->master_substream->runtime; + dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", + master_runtime->sample_bits, + master_runtime->rate); + + if (master_runtime->rate != 0) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + if (master_runtime->sample_bits != 0) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + ssm2604->slave_substream = substream; + } else + ssm2604->master_substream = substream; + + return 0; +} + +static int ssm2604_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + /* set active */ + snd_soc_write(codec, SSM2604_ACTIVE, ACTIVE_ACTIVATE_CODEC); + + return 0; +} + +static void ssm2604_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct ssm2604_priv *ssm2604 = snd_soc_codec_get_drvdata(codec); + + /* deactivate */ + if (!codec->active) + snd_soc_write(codec, SSM2604_ACTIVE, 0); + + if (ssm2604->master_substream == substream) + ssm2604->master_substream = ssm2604->slave_substream; + + ssm2604->slave_substream = NULL; +} + +static int ssm2604_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = snd_soc_read(codec, SSM2604_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE; + if (mute) + snd_soc_write(codec, SSM2604_APDIGI, + mute_reg | APDIGI_ENABLE_DAC_MUTE); + else + snd_soc_write(codec, SSM2604_APDIGI, mute_reg); + return 0; +} + +static int ssm2604_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ssm2604_priv *ssm2604 = snd_soc_codec_get_drvdata(codec); + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + ssm2604->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int ssm2604_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface |= 0x0040; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0013; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0003; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + /* set iface */ + snd_soc_write(codec, SSM2604_IFACE, iface); + return 0; +} + +static int ssm2604_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = snd_soc_read(codec, SSM2604_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + /* vref/mid, osc on, dac unmute */ + snd_soc_write(codec, SSM2604_PWR, reg); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + snd_soc_write(codec, SSM2604_PWR, reg | PWR_CLK_OUT_PDN); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + snd_soc_write(codec, SSM2604_ACTIVE, 0); + snd_soc_write(codec, SSM2604_PWR, 0xffff); + break; + + } + codec->bias_level = level; + return 0; +} + +#define SSM2604_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define SSM2604_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops ssm2604_dai_ops = { + .startup = ssm2604_startup, + .prepare = ssm2604_pcm_prepare, + .hw_params = ssm2604_hw_params, + .shutdown = ssm2604_shutdown, + .digital_mute = ssm2604_mute, + .set_sysclk = ssm2604_set_dai_sysclk, + .set_fmt = ssm2604_set_dai_fmt, +}; + +struct snd_soc_dai ssm2604_dai = { + .name = "SSM2604", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SSM2604_RATES, + .formats = SSM2604_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SSM2604_RATES, + .formats = SSM2604_FORMATS,}, + .ops = &ssm2604_dai_ops, +}; +EXPORT_SYMBOL_GPL(ssm2604_dai); + +static int ssm2604_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + ssm2604_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ssm2604_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(ssm2604_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + ssm2604_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + ssm2604_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +static int ssm2604_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int reg, ret = 0; + + socdev->card->codec = ssm2604_codec; + codec = ssm2604_codec; + + ssm2604_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "ssm2604: failed to create pcms\n"); + goto pcm_err; + } + /*power on device*/ + snd_soc_write(codec, SSM2604_ACTIVE, 0); + /* set the update bits */ + reg = snd_soc_read(codec, SSM2604_LINVOL); + snd_soc_write(codec, SSM2604_LINVOL, reg | LINVOL_LRIN_BOTH); + reg = snd_soc_read(codec, SSM2604_RINVOL); + snd_soc_write(codec, SSM2604_RINVOL, reg | RINVOL_RLIN_BOTH); + + snd_soc_write(codec, SSM2604_APANA, APANA_SELECT_DAC); + snd_soc_write(codec, SSM2604_PWR, 0); + + snd_soc_add_controls(codec, ssm2604_snd_controls, + ARRAY_SIZE(ssm2604_snd_controls)); + ssm2604_add_widgets(codec); + + return ret; + +pcm_err: + return ret; +} + +/* remove everything here */ +static int ssm2604_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ssm2604 = { + .probe = ssm2604_probe, + .remove = ssm2604_remove, + .suspend = ssm2604_suspend, + .resume = ssm2604_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2604); + +static int ssm2604_register(struct ssm2604_priv *ssm2604, enum snd_soc_control_type control) +{ + struct snd_soc_codec *codec = &ssm2604->codec; + int ret = 0; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->name = "SSM2604"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = ssm2604_set_bias_level; + codec->dai = &ssm2604_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(ssm2604_reg); + codec->reg_cache = kmemdup(ssm2604_reg, sizeof(ssm2604_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&ssm2604_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return ret; +} + +static void ssm2604_unregister(struct ssm2604_priv *ssm2604) +{ + struct snd_soc_codec *codec = &ssm2604->codec; + + ssm2604_set_bias_level(&ssm2604->codec, SND_SOC_BIAS_OFF); + kfree(codec->reg_cache); + snd_soc_unregister_dai(&ssm2604_dai); + snd_soc_unregister_codec(&ssm2604->codec); + kfree(ssm2604); + ssm2604_codec = NULL; +} + +static int ssm2604_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ssm2604_priv *ssm2604; + struct snd_soc_codec *codec; + + ssm2604 = kzalloc(sizeof(struct ssm2604_priv), GFP_KERNEL); + if (ssm2604 == NULL) + return -ENOMEM; + codec = &ssm2604->codec; + snd_soc_codec_set_drvdata(codec, ssm2604); + + i2c_set_clientdata(i2c, ssm2604); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + ssm2604_codec = codec; + + return ssm2604_register(ssm2604, SND_SOC_I2C); +} + +static __devexit int ssm2604_i2c_remove(struct i2c_client *client) +{ + struct ssm2604_priv *ssm2604 = i2c_get_clientdata(client); + ssm2604_unregister(ssm2604); + return 0; +} + +static const struct i2c_device_id ssm2604_i2c_id[] = { + { "ssm2604", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2604_i2c_id); + +/* corgi i2c codec control layer */ +static struct i2c_driver ssm2604_i2c_driver = { + .driver = { + .name = "ssm2604", + .owner = THIS_MODULE, + }, + .probe = ssm2604_i2c_probe, + .remove = __devexit_p(ssm2604_i2c_remove), + .id_table = ssm2604_i2c_id, +}; + +static int __init ssm2604_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&ssm2604_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register ssm2604 I2C driver: %d\n", + ret); + } + + return ret; +} +module_init(ssm2604_modinit); + +static void __exit ssm2604_exit(void) +{ + i2c_del_driver(&ssm2604_i2c_driver); +} +module_exit(ssm2604_exit); + +MODULE_DESCRIPTION("ASoC SSM2604 driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2604.h b/sound/soc/codecs/ssm2604.h new file mode 100644 index 0000000..3b24a08 --- /dev/null +++ b/sound/soc/codecs/ssm2604.h @@ -0,0 +1,92 @@ +/* + * Header for ssm2604 sound codec + * + * Copyright 2010 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef _SSM2604_H +#define _SSM2604_H + +/* SSM2604 Codec Register definitions */ + +#define SSM2604_LINVOL 0x00 +#define SSM2604_RINVOL 0x01 +#define SSM2604_APANA 0x04 +#define SSM2604_APDIGI 0x05 +#define SSM2604_PWR 0x06 +#define SSM2604_IFACE 0x07 +#define SSM2604_SRATE 0x08 +#define SSM2604_ACTIVE 0x09 +#define SSM2604_RESET 0x0f + +/* SSM2604 Codec Register Field definitions + * (Mask value to extract the corresponding Register field) + */ + +/* Left ADC Volume Control (SSM2604_REG_LEFT_ADC_VOL) */ +#define LINVOL_LIN_VOL 0x01F /* Left Channel PGA Volume control */ +#define LINVOL_LIN_ENABLE_MUTE 0x080 /* Left Channel Input Mute */ +#define LINVOL_LRIN_BOTH 0x100 /* Left Channel Line Input Volume update */ + +/* Right ADC Volume Control (SSM2604_REG_RIGHT_ADC_VOL) */ +#define RINVOL_RIN_VOL 0x01F /* Right Channel PGA Volume control */ +#define RINVOL_RIN_ENABLE_MUTE 0x080 /* Right Channel Input Mute */ +#define RINVOL_RLIN_BOTH 0x100 /* Right Channel Line Input Volume update */ + + +/* Analogue Audio Path Control (SSM2604_REG_ANALOGUE_PATH) */ +#define APANA_ENABLE_BYPASS 0x008 /* Line input bypass to line output */ +#define APANA_SELECT_DAC 0x010 /* Select DAC (1=Select DAC, 0=Don't Select DAC) */ + +/* Digital Audio Path Control (SSM2604_REG_DIGITAL_PATH) */ +#define APDIGI_ENABLE_ADC_HPF 0x001 /* Enable/Disable ADC Highpass Filter */ +#define APDIGI_DE_EMPHASIS 0x006 /* De-Emphasis Control */ +#define APDIGI_ENABLE_DAC_MUTE 0x008 /* DAC Mute Control */ +#define APDIGI_STORE_OFFSET 0x010 /* Store/Clear DC offset when HPF is disabled */ + +/* Power Down Control (SSM2604_REG_POWER) + * (1=Enable PowerDown, 0=Disable PowerDown) + */ +#define PWR_LINE_IN_PDN 0x001 /* Line Input Power Down */ +#define PWR_ADC_PDN 0x004 /* ADC Power Down */ +#define PWR_DAC_PDN 0x008 /* DAC Power Down */ +#define PWR_OSC_PDN 0x020 /* Oscillator Power Down */ +#define PWR_CLK_OUT_PDN 0x040 /* CLKOUT Power Down */ +#define PWR_POWER_OFF 0x080 /* POWEROFF Mode */ + +/* Digital Audio Interface Format (SSM2604_REG_DIGITAL_IFACE) */ +#define IFACE_IFACE_FORMAT 0x003 /* Digital Audio input format control */ +#define IFACE_AUDIO_DATA_LEN 0x00C /* Audio Data word length control */ +#define IFACE_DAC_LR_POLARITY 0x010 /* Polarity Control for clocks in RJ,LJ and I2S modes */ +#define IFACE_DAC_LR_SWAP 0x020 /* Swap DAC data control */ +#define IFACE_ENABLE_MASTER 0x040 /* Enable/Disable Master Mode */ +#define IFACE_BCLK_INVERT 0x080 /* Bit Clock Inversion control */ + +/* Sampling Control (SSM2604_REG_SAMPLING_CTRL) */ +#define SRATE_ENABLE_USB_MODE 0x001 /* Enable/Disable USB Mode */ +#define SRATE_BOS_RATE 0x002 /* Base Over-Sampling rate */ +#define SRATE_SAMPLE_RATE 0x03C /* Clock setting condition (Sampling rate control) */ +#define SRATE_CORECLK_DIV2 0x040 /* Core Clock divider select */ +#define SRATE_CLKOUT_DIV2 0x080 /* Clock Out divider select */ + +/* Active Control (SSM2604_REG_ACTIVE_CTRL) */ +#define ACTIVE_ACTIVATE_CODEC 0x001 /* Activate Codec Digital Audio Interface */ + +/*********************************************************************/ + +#define SSM2604_CACHEREGNUM 9 + +#define SSM2604_SYSCLK 0 +#define SSM2604_DAI 0 + +struct ssm2604_setup_data { + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai ssm2604_dai; +extern struct snd_soc_codec_device soc_codec_dev_ssm2604; + +#endif -- 1.7.2

3 6

[alsa-devel] [PATCH] ASoC: Blackfin I2S: add 8-bit sample support
by Mike Frysinger 26 Mar '11

26 Mar '11

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-i2s.c | 11 +++++++++-- 1 files changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index d453b1e..37d9d3c 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -140,6 +140,10 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, bf5xx_i2s.tcr2 &= ~0x1f; bf5xx_i2s.rcr2 &= ~0x1f; switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bf5xx_i2s->tcr2 |= 7; + bf5xx_i2s->rcr2 |= 7; + sport_handle->wdsize = 1; case SNDRV_PCM_FORMAT_S16_LE: bf5xx_i2s.tcr2 |= 15; bf5xx_i2s.rcr2 |= 15; @@ -266,8 +270,11 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ SNDRV_PCM_RATE_96000) -#define BF5XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) +#define BF5XX_I2S_FORMATS \ + (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { .shutdown = bf5xx_i2s_shutdown, -- 1.7.4.1

3 2

[alsa-devel] [PATCH 0/3] Fix breakages due to recent updates
by Wolfram Sang 26 Mar '11

26 Mar '11

I wanted to get sound out of my mx27-based PCM038 and it failed. I found two flaws which were introduced by recent update-works. While being here, I found some redundant code. Wolfram Sang (3): ASoC: imx: set watermarks for mx2-dma ASoC: imx: fix burstsize for DMA ASoC: imx: remove superfluous code in imx-ssi.c sound/soc/imx/imx-pcm-dma-mx2.c | 9 +++++++-- sound/soc/imx/imx-ssi.c | 6 ------ sound/soc/imx/imx-ssi.h | 3 --- 3 files changed, 7 insertions(+), 11 deletions(-) -- 1.7.2.5

3 9

[alsa-devel] [PATCH] ASoC: soc-cache: Return -ENOSYS instead of -EINVAL
by Dimitris Papastamos 26 Mar '11

26 Mar '11

These functions fail with -EINVAL if the corresponding callbacks are not implemented. Change them to return -ENOSYS as it is more appropriate for unimplemented callbacks. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 86beaad..0e2871e 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1295,7 +1295,7 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) codec->cache_ops->name, codec->name); return codec->cache_ops->init(codec); } - return -EINVAL; + return -ENOSYS; } /* @@ -1310,7 +1310,7 @@ int snd_soc_cache_exit(struct snd_soc_codec *codec) codec->cache_ops->name, codec->name); return codec->cache_ops->exit(codec); } - return -EINVAL; + return -ENOSYS; } /** @@ -1334,7 +1334,7 @@ int snd_soc_cache_read(struct snd_soc_codec *codec, } mutex_unlock(&codec->cache_rw_mutex); - return -EINVAL; + return -ENOSYS; } EXPORT_SYMBOL_GPL(snd_soc_cache_read); @@ -1359,7 +1359,7 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, } mutex_unlock(&codec->cache_rw_mutex); - return -EINVAL; + return -ENOSYS; } EXPORT_SYMBOL_GPL(snd_soc_cache_write); @@ -1382,7 +1382,7 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) } if (!codec->cache_ops || !codec->cache_ops->sync) - return -EINVAL; + return -ENOSYS; if (codec->cache_ops->name) name = codec->cache_ops->name; -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: ad193x: tweak style to match other codecs
by Mike Frysinger 26 Mar '11

26 Mar '11

From: Scott Jiang <scott.jiang(a)analog.com> Rename the snd_soc_control_type field from "bus_type" to "control_type", and drop the now unused "control_data" field. Signed-off-by: Scott Jiang <scott.jiang(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/ad193x.c | 19 +++++++------------ 1 files changed, 7 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index da46479..824529d 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,8 +23,7 @@ /* codec private data */ struct ad193x_priv { - enum snd_soc_control_type bus_type; - void *control_data; + enum snd_soc_control_type control_type; int sysclk; }; @@ -354,14 +353,12 @@ static int ad193x_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - codec->control_data = ad193x->control_data; - if (ad193x->bus_type == SND_SOC_I2C) - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ad193x->bus_type); + if (ad193x->control_type == SND_SOC_I2C) + ret = snd_soc_codec_set_cache_io(codec, 8, 8, ad193x->control_type); else - ret = snd_soc_codec_set_cache_io(codec, 16, 8, ad193x->bus_type); + ret = snd_soc_codec_set_cache_io(codec, 16, 8, ad193x->control_type); if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", - ret); + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; } @@ -408,8 +405,7 @@ static int __devinit ad193x_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, ad193x); - ad193x->control_data = spi; - ad193x->bus_type = SND_SOC_SPI; + ad193x->control_type = SND_SOC_SPI; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_ad193x, &ad193x_dai, 1); @@ -454,8 +450,7 @@ static int __devinit ad193x_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, ad193x); - ad193x->control_data = client; - ad193x->bus_type = SND_SOC_I2C; + ad193x->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_ad193x, &ad193x_dai, 1); -- 1.7.4.1

3 2

[alsa-devel] [PATCH] ASoC: ad73311: drop I2C requirement
by Mike Frysinger 26 Mar '11

26 Mar '11

From: Scott Jiang <scott.jiang(a)analog.com> The AD73311 codec does not use I2C, so don't require it in Kconfig. Signed-off-by: Scott Jiang <scott.jiang(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d6976ed..74a4629 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,7 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD183X if SPI_MASTER select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS - select SND_SOC_AD73311 if I2C + select SND_SOC_AD73311 select SND_SOC_AD74111 select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER -- 1.7.4.1

3 2

[alsa-devel] [PATCH] ASoC: twl6040: Return -ENOMEM if create_singlethread_workqueue fails
by Axel Lin 26 Mar '11

26 Mar '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/twl6040.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 482fcdb..255901c 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1629,8 +1629,10 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->naudint = naudint; priv->workqueue = create_singlethread_workqueue("twl6040-codec"); - if (!priv->workqueue) + if (!priv->workqueue) { + ret = -ENOMEM; goto work_err; + } INIT_DELAYED_WORK(&priv->delayed_work, twl6040_accessory_work); -- 1.7.1

3 2

[alsa-devel] [PATCH] ASoC: tlv320dac33: Restore L/R DAC power control register
by Peter Ujfalusi 26 Mar '11

26 Mar '11

Register 0x40, 0x41 need to be restored after power up, since it contains gain related fields, which affects playback volume. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 93f38b1..6c3735d 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -324,6 +324,10 @@ static void dac33_init_chip(struct snd_soc_codec *codec) dac33_write(codec, DAC33_OUT_AMP_CTRL, dac33_read_reg_cache(codec, DAC33_OUT_AMP_CTRL)); + dac33_write(codec, DAC33_LDAC_PWR_CTRL, + dac33_read_reg_cache(codec, DAC33_LDAC_PWR_CTRL)); + dac33_write(codec, DAC33_RDAC_PWR_CTRL, + dac33_read_reg_cache(codec, DAC33_RDAC_PWR_CTRL)); } static inline int dac33_read_id(struct snd_soc_codec *codec) -- 1.7.4.1

3 2

[alsa-devel] [PATCH] ASoC: tlv320dac33: Move codec power up to DAPM
by Peter Ujfalusi 26 Mar '11

26 Mar '11

Move the codec power on (in reg 0x01, bit 4) from set_bias_level:SND_SOC_BIAS_ON to a DAPM supply. In this way we can be sure, that all the things within the codec is powered before the external amp is going to be enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 9 ++++++--- 1 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 6c3735d..c109f6d 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -587,6 +587,9 @@ static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Right DAC Power", DAC33_RDAC_PWR_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Codec Power", + DAC33_PWR_CTRL, 4, 0, NULL, 0), + SND_SOC_DAPM_PRE("Pre Playback", dac33_playback_event), SND_SOC_DAPM_POST("Post Playback", dac33_playback_event), }; @@ -619,6 +622,9 @@ static const struct snd_soc_dapm_route audio_map[] = { /* output */ {"LEFT_LO", NULL, "Output Left Amplifier"}, {"RIGHT_LO", NULL, "Output Right Amplifier"}, + + {"LEFT_LO", NULL, "Codec Power"}, + {"RIGHT_LO", NULL, "Codec Power"}, }; static int dac33_add_widgets(struct snd_soc_codec *codec) @@ -636,13 +642,10 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { case SND_SOC_BIAS_ON: - if (!dac33->substream) - dac33_soft_power(codec, 1); break; case SND_SOC_BIAS_PREPARE: break; -- 1.7.4.1

3 2

[alsa-devel] [PATCH] ASoC: ad183x: add support for more multiple parts
by Mike Frysinger 26 Mar '11

26 Mar '11

From: Barry Song <barry.song(a)analog.com> This extends the AD1836 codec driver to also support the parts: - AD1835A - AD1837A - AD1838A - AD1839A Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Scott Jiang <scott.jiang(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/ad1836.c | 338 -------------------------------- sound/soc/codecs/ad1836.h | 63 ------ sound/soc/codecs/ad183x.c | 474 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad183x.h | 57 ++++++ 6 files changed, 535 insertions(+), 405 deletions(-) delete mode 100644 sound/soc/codecs/ad1836.c delete mode 100644 sound/soc/codecs/ad1836.h create mode 100644 sound/soc/codecs/ad183x.c create mode 100644 sound/soc/codecs/ad183x.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6943e24..dc6d253 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -13,7 +13,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS - select SND_SOC_AD1836 if SPI_MASTER + select SND_SOC_AD183X if SPI_MASTER select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_ADS117X @@ -115,7 +115,7 @@ config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC -config SND_SOC_AD1836 +config SND_SOC_AD183X tristate config SND_SOC_AD193X diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 379bc55..44d3278 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,6 +1,6 @@ snd-soc-88pm860x-objs := 88pm860x-codec.o snd-soc-ac97-objs := ac97.o -snd-soc-ad1836-objs := ad1836.o +snd-soc-ad183x-objs := ad183x.o snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o @@ -87,7 +87,7 @@ snd-soc-wm9090-objs := wm9090.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o -obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o +obj-$(CONFIG_SND_SOC_AD183X) += snd-soc-ad183x.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c deleted file mode 100644 index ab63d52..0000000 --- a/sound/soc/codecs/ad1836.c +++ /dev/null @@ -1,338 +0,0 @@ -/* - * File: sound/soc/codecs/ad1836.c - * Author: Barry Song <Barry.Song(a)analog.com> - * - * Created: Aug 04 2009 - * Description: Driver for AD1836 sound chip - * - * Modified: - * Copyright 2009 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * 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 of the License, or - * (at your option) any later version. - */ - -#include <linux/init.h> -#include <linux/slab.h> -#include <linux/module.h> -#include <linux/kernel.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> -#include <sound/tlv.h> -#include <linux/spi/spi.h> -#include "ad1836.h" - -/* codec private data */ -struct ad1836_priv { - enum snd_soc_control_type control_type; - void *control_data; -}; - -/* - * AD1836 volume/mute/de-emphasis etc. controls - */ -static const char *ad1836_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"}; - -static const struct soc_enum ad1836_deemp_enum = - SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp); - -static const struct snd_kcontrol_new ad1836_snd_controls[] = { - /* DAC volume control */ - SOC_DOUBLE_R("DAC1 Volume", AD1836_DAC_L1_VOL, - AD1836_DAC_R1_VOL, 0, 0x3FF, 0), - SOC_DOUBLE_R("DAC2 Volume", AD1836_DAC_L2_VOL, - AD1836_DAC_R2_VOL, 0, 0x3FF, 0), - SOC_DOUBLE_R("DAC3 Volume", AD1836_DAC_L3_VOL, - AD1836_DAC_R3_VOL, 0, 0x3FF, 0), - - /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD1836_ADC_CTRL2, AD1836_ADCL1_MUTE, - AD1836_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD1836_ADC_CTRL2, AD1836_ADCL2_MUTE, - AD1836_ADCR2_MUTE, 1, 1), - - /* DAC switch control */ - SOC_DOUBLE("DAC1 Switch", AD1836_DAC_CTRL2, AD1836_DACL1_MUTE, - AD1836_DACR1_MUTE, 1, 1), - SOC_DOUBLE("DAC2 Switch", AD1836_DAC_CTRL2, AD1836_DACL2_MUTE, - AD1836_DACR2_MUTE, 1, 1), - SOC_DOUBLE("DAC3 Switch", AD1836_DAC_CTRL2, AD1836_DACL3_MUTE, - AD1836_DACR3_MUTE, 1, 1), - - /* ADC high-pass filter */ - SOC_SINGLE("ADC High Pass Filter Switch", AD1836_ADC_CTRL1, - AD1836_ADC_HIGHPASS_FILTER, 1, 0), - - /* DAC de-emphasis */ - SOC_ENUM("Playback Deemphasis", ad1836_deemp_enum), -}; - -static const struct snd_soc_dapm_widget ad1836_dapm_widgets[] = { - SND_SOC_DAPM_DAC("DAC", "Playback", AD1836_DAC_CTRL1, - AD1836_DAC_POWERDOWN, 1), - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1836_ADC_CTRL1, - AD1836_ADC_POWERDOWN, 1, NULL, 0), - SND_SOC_DAPM_OUTPUT("DAC1OUT"), - SND_SOC_DAPM_OUTPUT("DAC2OUT"), - SND_SOC_DAPM_OUTPUT("DAC3OUT"), - SND_SOC_DAPM_INPUT("ADC1IN"), - SND_SOC_DAPM_INPUT("ADC2IN"), -}; - -static const struct snd_soc_dapm_route audio_paths[] = { - { "DAC", NULL, "ADC_PWR" }, - { "ADC", NULL, "ADC_PWR" }, - { "DAC1OUT", "DAC1 Switch", "DAC" }, - { "DAC2OUT", "DAC2 Switch", "DAC" }, - { "DAC3OUT", "DAC3 Switch", "DAC" }, - { "ADC", "ADC1 Switch", "ADC1IN" }, - { "ADC", "ADC2 Switch", "ADC2IN" }, -}; - -/* - * DAI ops entries - */ - -static int ad1836_set_dai_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) -{ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - /* at present, we support adc aux mode to interface with - * blackfin sport tdm mode - */ - case SND_SOC_DAIFMT_DSP_A: - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_IB_IF: - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - /* ALCLK,ABCLK are both output, AD1836 can only be master */ - case SND_SOC_DAIFMT_CBM_CFM: - break; - default: - return -EINVAL; - } - - return 0; -} - -static int ad1836_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - int word_len = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - - /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - word_len = 3; - break; - case SNDRV_PCM_FORMAT_S20_3LE: - word_len = 1; - break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: - word_len = 0; - break; - } - - snd_soc_update_bits(codec, AD1836_DAC_CTRL1, - AD1836_DAC_WORD_LEN_MASK, word_len); - - snd_soc_update_bits(codec, AD1836_ADC_CTRL2, - AD1836_ADC_WORD_LEN_MASK, word_len); - - return 0; -} - -#ifdef CONFIG_PM -static int ad1836_soc_suspend(struct snd_soc_codec *codec, - pm_message_t state) -{ - /* reset clock control mode */ - u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); - adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; - - return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); -} - -static int ad1836_soc_resume(struct snd_soc_codec *codec) -{ - /* restore clock control mode */ - u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); - adc_ctrl2 |= AD1836_ADC_AUX; - - return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); -} -#else -#define ad1836_soc_suspend NULL -#define ad1836_soc_resume NULL -#endif - -static struct snd_soc_dai_ops ad1836_dai_ops = { - .hw_params = ad1836_hw_params, - .set_fmt = ad1836_set_dai_fmt, -}; - -/* codec DAI instance */ -static struct snd_soc_dai_driver ad1836_dai = { - .name = "ad1836-hifi", - .playback = { - .stream_name = "Playback", - .channels_min = 2, - .channels_max = 6, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .capture = { - .stream_name = "Capture", - .channels_min = 2, - .channels_max = 4, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .ops = &ad1836_dai_ops, -}; - -static int ad1836_probe(struct snd_soc_codec *codec) -{ - struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret = 0; - - codec->control_data = ad1836->control_data; - ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", - ret); - return ret; - } - - /* default setting for ad1836 */ - /* de-emphasis: 48kHz, power-on dac */ - snd_soc_write(codec, AD1836_DAC_CTRL1, 0x300); - /* unmute dac channels */ - snd_soc_write(codec, AD1836_DAC_CTRL2, 0x0); - /* high-pass filter enable, power-on adc */ - snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100); - /* unmute adc channles, adc aux mode */ - snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180); - /* left/right diff:PGA/MUX */ - snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A); - /* volume */ - snd_soc_write(codec, AD1836_DAC_L1_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_R1_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_L2_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_R2_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_L3_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_R3_VOL, 0x3FF); - - snd_soc_add_controls(codec, ad1836_snd_controls, - ARRAY_SIZE(ad1836_snd_controls)); - snd_soc_dapm_new_controls(dapm, ad1836_dapm_widgets, - ARRAY_SIZE(ad1836_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); - - return ret; -} - -/* power down chip */ -static int ad1836_remove(struct snd_soc_codec *codec) -{ - /* reset clock control mode */ - u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); - adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; - - return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); -} - -static struct snd_soc_codec_driver soc_codec_dev_ad1836 = { - .probe = ad1836_probe, - .remove = ad1836_remove, - .suspend = ad1836_soc_suspend, - .resume = ad1836_soc_resume, - .reg_cache_size = AD1836_NUM_REGS, - .reg_word_size = sizeof(u16), -}; - -static int __devinit ad1836_spi_probe(struct spi_device *spi) -{ - struct ad1836_priv *ad1836; - int ret; - - ad1836 = kzalloc(sizeof(struct ad1836_priv), GFP_KERNEL); - if (ad1836 == NULL) - return -ENOMEM; - - spi_set_drvdata(spi, ad1836); - ad1836->control_data = spi; - ad1836->control_type = SND_SOC_SPI; - - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_ad1836, &ad1836_dai, 1); - if (ret < 0) - kfree(ad1836); - return ret; -} - -static int __devexit ad1836_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); - return 0; -} - -static struct spi_driver ad1836_spi_driver = { - .driver = { - .name = "ad1836-codec", - .owner = THIS_MODULE, - }, - .probe = ad1836_spi_probe, - .remove = __devexit_p(ad1836_spi_remove), -}; - -static int __init ad1836_init(void) -{ - int ret; - - ret = spi_register_driver(&ad1836_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register ad1836 SPI driver: %d\n", - ret); - } - - return ret; -} -module_init(ad1836_init); - -static void __exit ad1836_exit(void) -{ - spi_unregister_driver(&ad1836_spi_driver); -} -module_exit(ad1836_exit); - -MODULE_DESCRIPTION("ASoC ad1836 driver"); -MODULE_AUTHOR("Barry Song <21cnbao(a)gmail.com>"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h deleted file mode 100644 index 8455967..0000000 --- a/sound/soc/codecs/ad1836.h +++ /dev/null @@ -1,63 +0,0 @@ -/* - * File: sound/soc/codecs/ad1836.h - * Based on: - * Author: Barry Song <Barry.Song(a)analog.com> - * - * Created: Aug 04, 2009 - * Description: definitions for AD1836 registers - * - * Modified: - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * 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 of the License, or - * (at your option) any later version. - */ - -#ifndef __AD1836_H__ -#define __AD1836_H__ - -#define AD1836_DAC_CTRL1 0 -#define AD1836_DAC_POWERDOWN 2 -#define AD1836_DAC_SERFMT_MASK 0xE0 -#define AD1836_DAC_SERFMT_PCK256 (0x4 << 5) -#define AD1836_DAC_SERFMT_PCK128 (0x5 << 5) -#define AD1836_DAC_WORD_LEN_MASK 0x18 - -#define AD1836_DAC_CTRL2 1 -#define AD1836_DACL1_MUTE 0 -#define AD1836_DACR1_MUTE 1 -#define AD1836_DACL2_MUTE 2 -#define AD1836_DACR2_MUTE 3 -#define AD1836_DACL3_MUTE 4 -#define AD1836_DACR3_MUTE 5 - -#define AD1836_DAC_L1_VOL 2 -#define AD1836_DAC_R1_VOL 3 -#define AD1836_DAC_L2_VOL 4 -#define AD1836_DAC_R2_VOL 5 -#define AD1836_DAC_L3_VOL 6 -#define AD1836_DAC_R3_VOL 7 - -#define AD1836_ADC_CTRL1 12 -#define AD1836_ADC_POWERDOWN 7 -#define AD1836_ADC_HIGHPASS_FILTER 8 - -#define AD1836_ADC_CTRL2 13 -#define AD1836_ADCL1_MUTE 0 -#define AD1836_ADCR1_MUTE 1 -#define AD1836_ADCL2_MUTE 2 -#define AD1836_ADCR2_MUTE 3 -#define AD1836_ADC_WORD_LEN_MASK 0x30 -#define AD1836_ADC_SERFMT_MASK (7 << 6) -#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) -#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) -#define AD1836_ADC_AUX (0x6 << 6) - -#define AD1836_ADC_CTRL3 14 - -#define AD1836_NUM_REGS 16 - -#endif diff --git a/sound/soc/codecs/ad183x.c b/sound/soc/codecs/ad183x.c new file mode 100644 index 0000000..2dd548a --- /dev/null +++ b/sound/soc/codecs/ad183x.c @@ -0,0 +1,474 @@ +/* + * AD183X Audio Codec driver supporting AD1835A, AD1836, AD1837A, AD1838A, AD1839A + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/spi/spi.h> +#include "ad183x.h" + +/* codec private data */ +struct ad183x_chl_ctrls { + struct snd_kcontrol_new *snd_ctrls; + struct snd_soc_dapm_widget *dapm_widgets; + struct snd_soc_dapm_route *audio_paths; + int ctrl_num; + int widget_num; + int path_num; +}; + +struct ad183x_priv { + enum snd_soc_control_type control_type; + struct ad183x_chl_ctrls chl_ctrl; +}; + +/* + * AD183X volume/mute/de-emphasis etc. controls + */ +static const char *ad183x_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"}; + +static const struct soc_enum ad183x_deemp_enum = +SOC_ENUM_SINGLE(AD183X_DAC_CTRL1, 8, 4, ad183x_deemp); + +/* AD1835A/AD1837A: 4 stereo DAC, 1 stereo ADC; */ +static const struct snd_kcontrol_new ad1835a_ad1837a_snd_controls[] = { + /* DAC volume control */ + SOC_DOUBLE_R("DAC1 Volume", AD183X_DAC_L1_VOL, + AD183X_DAC_R1_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC2 Volume", AD183X_DAC_L2_VOL, + AD183X_DAC_R2_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC3 Volume", AD183X_DAC_L3_VOL, + AD183X_DAC_R3_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC4 Volume", AD183X_DAC_L4_VOL, + AD183X_DAC_R4_VOL, 0, 0x3FF, 0), + + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD183X_ADC_CTRL2, AD183X_ADCL1_MUTE, + AD183X_ADCR1_MUTE, 1, 1), + + /* DAC switch control */ + SOC_DOUBLE("DAC1 Switch", AD183X_DAC_CTRL2, AD183X_DACL1_MUTE, + AD183X_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD183X_DAC_CTRL2, AD183X_DACL2_MUTE, + AD183X_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD183X_DAC_CTRL2, AD183X_DACL3_MUTE, + AD183X_DACR3_MUTE, 1, 1), + SOC_DOUBLE("DAC4 Switch", AD183X_DAC_CTRL2, AD183X_DACL4_MUTE, + AD183X_DACR4_MUTE, 1, 1), + + /* ADC high-pass filter */ + SOC_SINGLE("ADC High Pass Filter Switch", AD183X_ADC_CTRL1, + AD183X_ADC_HIGHPASS_FILTER, 1, 0), + + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad183x_deemp_enum), +}; + +static const struct snd_soc_dapm_widget ad1835a_ad1837a_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", AD183X_DAC_CTRL1, + AD183X_DAC_POWERDOWN, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD183X_ADC_CTRL1, + AD183X_ADC_POWERDOWN, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_OUTPUT("DAC4OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), +}; + +static const struct snd_soc_dapm_route ad1835a_ad1837a_audio_paths[] = { + { "DAC", NULL, "ADC_PWR" }, + { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "DAC3OUT", "DAC4 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, +}; + +/* AD1836: 3 stereo DAC, 2 stereo ADC; */ +static const struct snd_kcontrol_new ad1836_snd_controls[] = { + /* DAC volume control */ + SOC_DOUBLE_R("DAC1 Volume", AD183X_DAC_L1_VOL, + AD183X_DAC_R1_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC2 Volume", AD183X_DAC_L2_VOL, + AD183X_DAC_R2_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC3 Volume", AD183X_DAC_L3_VOL, + AD183X_DAC_R3_VOL, 0, 0x3FF, 0), + + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD183X_ADC_CTRL2, AD183X_ADCL1_MUTE, + AD183X_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD183X_ADC_CTRL2, AD183X_ADCL2_MUTE, + AD183X_ADCR2_MUTE, 1, 1), + + /* DAC switch control */ + SOC_DOUBLE("DAC1 Switch", AD183X_DAC_CTRL2, AD183X_DACL1_MUTE, + AD183X_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD183X_DAC_CTRL2, AD183X_DACL2_MUTE, + AD183X_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD183X_DAC_CTRL2, AD183X_DACL3_MUTE, + AD183X_DACR3_MUTE, 1, 1), + + /* ADC high-pass filter */ + SOC_SINGLE("ADC High Pass Filter Switch", AD183X_ADC_CTRL1, + AD183X_ADC_HIGHPASS_FILTER, 1, 0), + + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad183x_deemp_enum), +}; + +static const struct snd_soc_dapm_widget ad1836_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", AD183X_DAC_CTRL1, + AD183X_DAC_POWERDOWN, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD183X_ADC_CTRL1, + AD183X_ADC_POWERDOWN, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), + SND_SOC_DAPM_INPUT("ADC2IN"), +}; + +static const struct snd_soc_dapm_route ad1836_audio_paths[] = { + { "DAC", NULL, "ADC_PWR" }, + { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, + { "ADC", "ADC2 Switch", "ADC2IN" }, +}; + +/* AD1838A/AD1939A: 3 stereo DAC, 1 stereo ADC; */ +static const struct snd_kcontrol_new ad1838a_ad1839a_snd_controls[] = { + /* DAC volume control */ + SOC_DOUBLE_R("DAC1 Volume", AD183X_DAC_L1_VOL, + AD183X_DAC_R1_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC2 Volume", AD183X_DAC_L2_VOL, + AD183X_DAC_R2_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC3 Volume", AD183X_DAC_L3_VOL, + AD183X_DAC_R3_VOL, 0, 0x3FF, 0), + + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD183X_ADC_CTRL2, AD183X_ADCL1_MUTE, + AD183X_ADCR1_MUTE, 1, 1), + + /* DAC switch control */ + SOC_DOUBLE("DAC1 Switch", AD183X_DAC_CTRL2, AD183X_DACL1_MUTE, + AD183X_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD183X_DAC_CTRL2, AD183X_DACL2_MUTE, + AD183X_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD183X_DAC_CTRL2, AD183X_DACL3_MUTE, + AD183X_DACR3_MUTE, 1, 1), + + /* ADC high-pass filter */ + SOC_SINGLE("ADC High Pass Filter Switch", AD183X_ADC_CTRL1, + AD183X_ADC_HIGHPASS_FILTER, 1, 0), + + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad183x_deemp_enum), +}; + +static const struct snd_soc_dapm_widget ad1838a_ad1839a_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", AD183X_DAC_CTRL1, + AD183X_DAC_POWERDOWN, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD183X_ADC_CTRL1, + AD183X_ADC_POWERDOWN, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), +}; + +static const struct snd_soc_dapm_route ad1838a_ad1839a_audio_paths[] = { + { "DAC", NULL, "ADC_PWR" }, + { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, +}; + +/* + * DAI ops entries + */ + +static int ad183x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + /* at present, we support adc aux mode to interface with + * blackfin sport tdm mode + */ + case SND_SOC_DAIFMT_DSP_A: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + /* ALCLK,ABCLK are both output, AD183X can only be master */ + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ad183x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int word_len = 0; + + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + word_len = 3; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + word_len = 1; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + word_len = 0; + break; + } + + snd_soc_update_bits(codec, AD183X_DAC_CTRL1, + AD183X_DAC_WORD_LEN_MASK, word_len); + + snd_soc_update_bits(codec, AD183X_ADC_CTRL2, + AD183X_ADC_WORD_LEN_MASK, word_len); + return 0; +} + +static struct snd_soc_dai_ops ad183x_dai_ops = { + .hw_params = ad183x_hw_params, + .set_fmt = ad183x_set_dai_fmt, +}; + +/* codec DAI instance */ +static struct snd_soc_dai_driver ad183x_dai = { + .name = "ad183x-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &ad183x_dai_ops, +}; + +static int ad183x_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + /* reset clock control mode */ + u16 adc_ctrl2 = snd_soc_read(codec, AD183X_ADC_CTRL2); + adc_ctrl2 &= ~AD183X_ADC_SERFMT_MASK; + + return snd_soc_write(codec, AD183X_ADC_CTRL2, adc_ctrl2); +} + +static int ad183x_resume(struct snd_soc_codec *codec) +{ + /* restore clock control mode */ + u16 adc_ctrl2 = snd_soc_read(codec, AD183X_ADC_CTRL2); + adc_ctrl2 |= AD183X_ADC_AUX; + + return snd_soc_write(codec, AD183X_ADC_CTRL2, adc_ctrl2); +} + +static int ad183x_probe(struct snd_soc_codec *codec) +{ + struct ad183x_priv *ad183x = snd_soc_codec_get_drvdata(codec); + struct ad183x_chl_ctrls *chl_ctrl; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 4, 12, ad183x->control_type); + if (ret < 0) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); + return ret; + } + + /* default setting for ad183x */ + /* de-emphasis: 48kHz, power-on dac */ + snd_soc_write(codec, AD183X_DAC_CTRL1, 0x300); + /* unmute dac channels */ + snd_soc_write(codec, AD183X_DAC_CTRL2, 0x0); + /* high-pass filter enable, power-on adc */ + snd_soc_write(codec, AD183X_ADC_CTRL1, 0x100); + /* unmute adc channles, adc aux mode */ + snd_soc_write(codec, AD183X_ADC_CTRL2, 0x180); + /* left/right diff:PGA/MUX */ + snd_soc_write(codec, AD183X_ADC_CTRL3, 0x3A); + /* volume */ + snd_soc_write(codec, AD183X_DAC_L1_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_R1_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_L2_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_R2_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_L3_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_R3_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_L4_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_R4_VOL, 0x3FF); + + chl_ctrl = &ad183x->chl_ctrl; + + snd_soc_add_controls(codec, chl_ctrl->snd_ctrls, chl_ctrl->ctrl_num); + snd_soc_dapm_new_controls(dapm, chl_ctrl->dapm_widgets, chl_ctrl->widget_num); + snd_soc_dapm_add_routes(dapm, chl_ctrl->audio_paths, chl_ctrl->path_num); + + return 0; +} + +/* power down chip */ +static int ad183x_remove(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ad183x = { + .probe = ad183x_probe, + .remove = ad183x_remove, + .suspend = ad183x_suspend, + .resume = ad183x_resume, + .reg_cache_size = AD183X_NUM_REGS, + .reg_word_size = sizeof(u16), +}; + +static int __devinit ad183x_spi_probe(struct spi_device *spi) +{ + struct ad183x_priv *ad183x; + int ret; + char *chip_name = spi->dev.platform_data; + + if (!chip_name) + return -ENODEV; + + ad183x = kzalloc(sizeof(struct ad183x_priv), GFP_KERNEL); + if (ad183x == NULL) + return -ENOMEM; + + if ((strcmp(chip_name, "ad1835a") == 0) || (strcmp(chip_name, "ad1837a") == 0)) { + ad183x->chl_ctrl.snd_ctrls = (struct snd_kcontrol_new *)ad1835a_ad1837a_snd_controls; + ad183x->chl_ctrl.dapm_widgets = (struct snd_soc_dapm_widget *)ad1835a_ad1837a_dapm_widgets; + ad183x->chl_ctrl.audio_paths = (struct snd_soc_dapm_route *)ad1835a_ad1837a_audio_paths; + ad183x->chl_ctrl.ctrl_num = ARRAY_SIZE(ad1835a_ad1837a_snd_controls); + ad183x->chl_ctrl.widget_num = ARRAY_SIZE(ad1835a_ad1837a_dapm_widgets); + ad183x->chl_ctrl.path_num = ARRAY_SIZE(ad1835a_ad1837a_audio_paths); + ad183x_dai.playback.channels_max = 8; + ad183x_dai.capture.channels_max = 2; + } else if ((strcmp(chip_name, "ad1838a") == 0) || (strcmp(chip_name, "ad1839a") == 0)) { + ad183x->chl_ctrl.snd_ctrls = (struct snd_kcontrol_new *)ad1838a_ad1839a_snd_controls; + ad183x->chl_ctrl.dapm_widgets = (struct snd_soc_dapm_widget *)ad1838a_ad1839a_dapm_widgets; + ad183x->chl_ctrl.audio_paths = (struct snd_soc_dapm_route *)ad1838a_ad1839a_audio_paths; + ad183x->chl_ctrl.ctrl_num = ARRAY_SIZE(ad1838a_ad1839a_snd_controls); + ad183x->chl_ctrl.widget_num = ARRAY_SIZE(ad1838a_ad1839a_dapm_widgets); + ad183x->chl_ctrl.path_num = ARRAY_SIZE(ad1838a_ad1839a_audio_paths); + ad183x_dai.playback.channels_max = 6; + ad183x_dai.capture.channels_max = 2; + } else if (strcmp(chip_name, "ad1836") == 0) { + ad183x->chl_ctrl.snd_ctrls = (struct snd_kcontrol_new *)ad1836_snd_controls; + ad183x->chl_ctrl.dapm_widgets = (struct snd_soc_dapm_widget *)ad1836_dapm_widgets; + ad183x->chl_ctrl.audio_paths = (struct snd_soc_dapm_route *)ad1836_audio_paths; + ad183x->chl_ctrl.ctrl_num = ARRAY_SIZE(ad1836_snd_controls); + ad183x->chl_ctrl.widget_num = ARRAY_SIZE(ad1836_dapm_widgets); + ad183x->chl_ctrl.path_num = ARRAY_SIZE(ad1836_audio_paths); + ad183x_dai.playback.channels_max = 6; + ad183x_dai.capture.channels_max = 4; + } else { + dev_err(&spi->dev, "not supported chip type\n"); + return -EINVAL; + } + + spi_set_drvdata(spi, ad183x); + ad183x->control_type = SND_SOC_SPI; + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_ad183x, &ad183x_dai, 1); + + if (ret < 0) + kfree(ad183x); + return ret; +} + +static int __devexit ad183x_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + kfree(spi_get_drvdata(spi)); + + return 0; +} + +static struct spi_driver ad183x_spi_driver = { + .driver = { + .name = "ad183x-codec", + .owner = THIS_MODULE, + }, + .probe = ad183x_spi_probe, + .remove = __devexit_p(ad183x_spi_remove), +}; + +static int __init ad183x_init(void) +{ + int ret = 0; +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&ad183x_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register ad183x SPI driver: %d\n", + ret); + } +#endif + return ret; +} +module_init(ad183x_init); + +static void __exit ad183x_exit(void) +{ +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&ad183x_spi_driver); +#endif +} +module_exit(ad183x_exit); + +MODULE_DESCRIPTION("ASoC ad183x driver"); +MODULE_AUTHOR("Barry Song <21cnbao(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad183x.h b/sound/soc/codecs/ad183x.h new file mode 100644 index 0000000..f7af3db --- /dev/null +++ b/sound/soc/codecs/ad183x.h @@ -0,0 +1,57 @@ +/* + * AD183X Audio Codec driver supporting AD1835A, AD1836, AD1837A, AD1838A, AD1839A + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef __AD183X_H__ +#define __AD183X_H__ + +#define AD183X_DAC_CTRL1 0 +#define AD183X_DAC_POWERDOWN 2 +#define AD183X_DAC_SERFMT_MASK 0xE0 +#define AD183X_DAC_SERFMT_PCK256 (0x4 << 5) +#define AD183X_DAC_SERFMT_PCK128 (0x5 << 5) +#define AD183X_DAC_WORD_LEN_MASK 0x18 + +#define AD183X_DAC_CTRL2 1 +#define AD183X_DACL1_MUTE 0 +#define AD183X_DACR1_MUTE 1 +#define AD183X_DACL2_MUTE 2 +#define AD183X_DACR2_MUTE 3 +#define AD183X_DACL3_MUTE 4 +#define AD183X_DACR3_MUTE 5 +#define AD183X_DACL4_MUTE 6 +#define AD183X_DACR4_MUTE 7 + +#define AD183X_DAC_L1_VOL 2 +#define AD183X_DAC_R1_VOL 3 +#define AD183X_DAC_L2_VOL 4 +#define AD183X_DAC_R2_VOL 5 +#define AD183X_DAC_L3_VOL 6 +#define AD183X_DAC_R3_VOL 7 +#define AD183X_DAC_L4_VOL 8 +#define AD183X_DAC_R4_VOL 9 + +#define AD183X_ADC_CTRL1 12 +#define AD183X_ADC_POWERDOWN 7 +#define AD183X_ADC_HIGHPASS_FILTER 8 + +#define AD183X_ADC_CTRL2 13 +#define AD183X_ADCL1_MUTE 0 +#define AD183X_ADCR1_MUTE 1 +#define AD183X_ADCL2_MUTE 2 +#define AD183X_ADCR2_MUTE 3 +#define AD183X_ADC_WORD_LEN_MASK 0x30 +#define AD183X_ADC_SERFMT_MASK (7 << 6) +#define AD183X_ADC_SERFMT_PCK256 (0x4 << 6) +#define AD183X_ADC_SERFMT_PCK128 (0x5 << 6) +#define AD183X_ADC_AUX (0x6 << 6) + +#define AD183X_ADC_CTRL3 14 + +#define AD183X_NUM_REGS 16 + +#endif -- 1.7.4.1

2 1

Re: [alsa-devel] Loud pops from Intel HDA onboard sound, may damage speakers (fwd)
by Wu Fengguang 26 Mar '11

26 Mar '11

Hi Chris, Would you run http://www.alsa-project.org/alsa-info.sh and email the results? Thanks, Fengguang On Thu, Mar 24, 2011 at 01:35:54AM +0000, Chris Wilson wrote: > ---------- Forwarded message ---------- > Date: Thu, 24 Mar 2011 00:49:55 +0000 (GMT) > From: Chris Wilson <chris(a)qwirx.com> > To: Takashi Iwai <tiwai(a)suse.de>, PeiSen Hou <pshou(a)realtek.com.tw>, > matt.jared(a)intel.com, andy.kopp(a)intel.com, dan.d.kogan(a)intel.com > Subject: Loud pops from Intel HDA onboard sound, may damage speakers > > Dear sirs, > > I'm writing to you as the named contacts in hda_intel.c in the Linux > kernel. Please let me know if I should contact someone else instead. > > I refer to this downstream bug on Ubuntu Launchpad, which I am > hereby taking upstream: > https://bugs.launchpad.net/ubuntu/+source/linux/+bug/184632 > > I have a desktop system (Fujitsu P9900) with the onboard sound > connected to my hi-fi amplifier, driven by snd_hda_intel. The audio > chipset is: > > 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset > High Definition Audio (rev 05) > > The sound card outputs loud popping sounds whenever I suspend and > resume it. One of them is loud or severe enough to cause my hi-fi > amplifier to go into automatic shutdown mode, and I'm worried that > it may damage my speakers. > > If I enable power_save=1, as suggested by powertop, then this > happens about 2 seconds after any audio finishes playing as well, so > I suspect it's related to putting the sound card in or out of > suspend mode. > > The pops seem to be happening to others who have this chipset as > well, as seen on the bug report, but they seem mostly to be using > laptops or otherwise not bothered by potential damage to their > speakers. > > I would really appreciate it if you could help me to find a way to > avoid these pops, for example by muting the audio output before > shutting down the DAC? I think it would be great if this could make > it into the standard kernel driver. > > Many thanks in advance, > > Chris Wilson. > -- > _____ __ _ > \ __/ / ,__(_)_ | Chris Wilson <chris+sig(a)qwirx.com> Cambs UK | > / (_/ ,\/ _/ /_ \ | Security/C/C++/Java/Ruby/Perl/SQL Developer | > \__/_/_/_//_/___/ | We are GNU : free your mind & your software |

4 17

[alsa-devel] [PATCH 1/3] Support single-rate no-SRC cards.
by linux＠audioscience.com 25 Mar '11

25 Mar '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Cards without settable local samplerate and without SRC still must have a valid samplerate. This fixed rate is determined by reading the current rate for the card. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- pci/asihpi/asihpi.c | 16 +++++++--------- 1 files changed, 7 insertions(+), 9 deletions(-) diff --git a/pci/asihpi/asihpi.c b/pci/asihpi/asihpi.c index 0ac1f98..22606e3 100644 --- a/pci/asihpi/asihpi.c +++ b/pci/asihpi/asihpi.c @@ -381,13 +381,13 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi, "No local sampleclock, err %d\n", err); } - for (idx = 0; idx < 100; idx++) { - if (hpi_sample_clock_query_local_rate( - h_control, idx, &sample_rate)) { - if (!idx) - snd_printk(KERN_ERR - "Local rate query failed\n"); - + for (idx = -1; idx < 100; idx++) { + if (idx == -1) { + if (hpi_sample_clock_get_sample_rate(h_control, + &sample_rate)) + continue; + } else if (hpi_sample_clock_query_local_rate(h_control, + idx, &sample_rate)) { break; } @@ -440,8 +440,6 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi, } } - /* printk(KERN_INFO "Supported rates %X %d %d\n", - rates, rate_min, rate_max); */ pcmhw->rates = rates; pcmhw->rate_min = rate_min; pcmhw->rate_max = rate_max; -- 1.7.0.4

2 3

[alsa-devel] snd_soc_dapm_force_enable_pin question
by Koul, Vinod 25 Mar '11

25 Mar '11

Hi, I have a Vibra controller on the codec which can be programmed using alsa controls. Now after configuration I added an enable control which when turned On, forces the Vibra Pin to be ON. The Pin is on, but not the path, I need the DAC and amplifier connected to be ON as well? Is this not what the force_enable_pin() is supposed to do? If not how should I ensure this path and codec BIAS is on when Vibra is activated by user? -- ~Vinod

2 4

[alsa-devel] QasMixer 0.10.1 available
by Sebastian H. 25 Mar '11

25 Mar '11

Hello Alsa and Qt users QasMixer version 0.10.1 is now available. This release fixes some issues with session mangement and the spanish translation is updated. Changes: * Improved single instance check * Fix for not shutting down on a session shutdown * Spanish translation updated (Jose L.) Homepage with more informations and screenshots http://xwmw.org/qasmixer Files are hosted at sourceforge http://sourceforge.net/projects/qasmixer/ Happy volume changing! -- Sebastian Holtermann

1 0

[alsa-devel] no external mic
by leon 25 Mar '11

25 Mar '11

Hello, i've Laptop Dell Inspiron 1520, the internal mic works perfectly, but external mic doesn't work, i've followed this link: http://ubuntuforums.org/showthread.php?p=10352564 but it didn't help me, here's the output of ALSA Information Script: http://www.alsa-project.org/db/?f=3aa8a44412081f533b2de3e32aae76ff4370eae9 btw, i've been using Ubuntu, and Debian now, both have same problem. so i haven't tried other Distributions. any idea? thanks. ----------------------------------------------------------------------------------------------------------------------- Send big files for free. Simple steps. No registration. Visit now http://www.nawelny.com

3 2

[alsa-devel] DAPM widgets beyond 2.6.38
by Peter Hsiang 25 Mar '11

25 Mar '11

1) In 2.6.38, after these 2 operations: snd_soc_dapm_new_controls(dapm, dapm_widgets, ARRAY_SIZE(dapm_widgets)); snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); Is it *always* necessary to call snd_soc_dapm_new_widgets(dapm)? I have seen one case where it was not called. 2) In 2.6.39, to do the same, is there any function to call other than the table entries? Thanks, Peter

2 1

[alsa-devel] [PATCH] ALSA: HDA: New AD1984A model for Dell Precision R5500
by David Henningsson 25 Mar '11

25 Mar '11

For codec AD1984A, add a new model to support Dell Precision R5500 or the microphone jack won't work correctly. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

Re: [alsa-devel] Asus E35M1-M Pro and Realtek AL887-VD - no love from ALSA for Sourround
by Raymond Yau 25 Mar '11

25 Mar '11

>> I recently purchased *Asus* *e35M1*-*M* *Pro* and I would like to get souround >> sound going fro*m* *m*y 5.1 syste*m*. I have it connected accordingly to the *>> m*anual (line in - rear speakers, line out - front speaker, *m*ic in - >> bass/center) but anywy I try that is using guides like >> http://www.gentoo-wiki.info/HOWTO_Surround_Sound <http://www.gentoo-wiki.info/HOWTO_Surround_Sound> I a*m* *no*t able to get >> sound fro*m* any other speaker then front left and right one. *Alsa**m*ixer >> does *no*t show any devices as shared to be available to switch to the >> output fro*m* input >> speaker-test plays every file in the left and right front speaker at the >> sa*m*e ti*m*e >> Here is *m*y syste*m* infor*m*ation using 2.6.38-rc8-git3 >> http://www.*alsa*-*pro*ject.org/db/?f=0dd55b4640a8afff1552c4d0be0105e416ad1… <http://www.alsa-project.org/db/?f=0dd55b4640a8afff1552c4d0be0105e416ad1cf3> >> and 2.6.38-rc8-next-20110314 >> http://www.*alsa*-*pro*ject.org/db/?f=a8733bf0083929540948f51b5f33d9c8a62d9… <http://www.alsa-project.org/db/?f=a8733bf0083929540948f51b5f33d9c8a62d9994> *>> M*y question is what should I try next? Any desktop motherboard with 3 jacks in rear panel should use "3stack-6ch" to support analog surround51 or "3stack-6ch-dig" to analog surround51 and digital out

1 0

[alsa-devel] driver silence callback
by Eliot Blennerhassett 24 Mar '11

24 Mar '11

Greetings, Is the driver's silence callback meant to be able to *overwrite* some part of the audio buffer that has already been written e.g. via the playback *copy* callback? In the case of my (older) hardware, once audio has been written via the copy callback, it is gone (into a fifo), there is no way to overwrite with silence. The asihpi silence call back will _append_ a period of silence to the already written audio queue. I'm wondering if this is wrong? Perhaps I should not have a silence callback at all? I'm looking at the case in pcm_lib.c snd_pcm_playback_silence() where "when runtime->silence_size >= runtime->boundary - fill processed area with silence immediately" My reference for silence callback. http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ch11s02.html regards Eliot

2 2

[alsa-devel] [PATCH] TerraTec DMX 6Fire USB: Don't leak firmware when usb6fire_fw_ihex_init() fails.
by Jesper Juhl 24 Mar '11

24 Mar '11

We need to release_firmware() what we request_firmware()'ed when usb6fire_fw_ihex_init() fails or we'll leak memory. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- firmware.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 9081a54..2761392 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -254,6 +254,7 @@ static int usb6fire_fw_ezusb_upload( ret = usb6fire_fw_ihex_init(fw, rec); if (ret < 0) { kfree(rec); + release_firmware(fw); snd_printk(KERN_ERR PREFIX "error validating ezusb " "firmware %s.\n", fwname); return ret; -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

1 0

[alsa-devel] [PATCH] ASoC: Add max98095 CODEC driver
by Peter Hsiang 24 Mar '11

24 Mar '11

This patch adds the MAX98095 CODEC driver. Signed-off-by: Peter Hsiang <peter.hsiang(a)maxim-ic.com> Please see the attached patch file 0001-ASoC-Add-max98095-CODEC-driver.patch Thanks, Peter

3 2

[alsa-devel] Coverity tool reported issues related patches
by sudarshan.bisht＠nokia.com 24 Mar '11

24 Mar '11

Hi, Few days back I sent some patches for review and you reviewed them, after incorporating review comments I sent updated patches too. So I just wanted to know that the way I sent them was right ? And when will those patches be incorporated to alsa-lib upstream ? If something has to be done from my side then please let me know. Br, Sudarshan Bisht

2 1

Re: [alsa-devel] 答复: [PATCH 7/16] ALSA: VIA HDA: Modify notify_aa_path_ctls() function
by Takashi Iwai 24 Mar '11

24 Mar '11

At Thu, 24 Mar 2011 15:32:51 +0800, <LydiaWang(a)viatech.com.cn> wrote: > > > > > -----邮件原件----- > > 发件人: Takashi Iwai [mailto:tiwai@suse.de] > > 发送时间: 2011年3月21日 19:08 > > 收件人: Jaroslav Kysela > > 抄送: Lydia Wang; alsa-devel(a)alsa-project.org; Harald Welte > > 主题: Re: [alsa-devel] [PATCH 7/16] ALSA: VIA HDA: Modify notify_aa_path_ctls() > > function > > > > At Mon, 21 Mar 2011 08:32:43 +0100 (CET), > > Jaroslav Kysela wrote: > > > > > > On Mon, 21 Mar 2011, Lydia Wang wrote: > > > > > > > From: Lydia Wang <lydiawang(a)viatech.com.cn> > > > > Subject: ALSA: VIA HDA: Modify notify_aa_path_ctls() function. > > > > > > > > Modify notify_aa_path_ctls() function to get volume control items name > > > > dynamically instead of by giving static item name. > > > > > > This code looks ugly to me. If you like to have a dynamic set of notified > > > controls, create one using snd_array in via_spec. > > > > Agreed. The current code is too fragile. > > > > > > thanks, > > > > Takashi > > I've created such control type of snd_array. When I want to update a-a path volume and mute control, I must find out the controls among all such ones. But in new alsa driver, I found the mic's a-a path name will be mic, or rear mic according to its pin position. In old driver, the a-a path name is static, but now it's dynamic. So I write this function to find out the actual name of controls. I'm not very clear why it's ugly. Please tell me. Thank you! First off, looking over all controls of the card instance isn't considered as a valid task for the local codec driver. If such a feature is inevitably necessary, we should implement a helper function in ALSA core. Secondly, the controls you look at might not belong to your codec. HD-audio bus can contain multiple codecs, and it might come from some others. HD-audio codec has already an array of mixer elements. codec->mixers is an array for struct hda_nid_item. You can use this instead for look-up. If more performance optimization is needed, create another local array to point only the needed kctls. thanks, Takashi

1 0

[alsa-devel] Bad file descriptor : -EBADFD
by Irfan Shaikh 24 Mar '11

24 Mar '11

Hello In the below logic, snd_pcm_write always returns -EBADFD . snd_pcm_writei not at all recovers from this state infinitly. How can i solve this problem. while(1) { /*Read from capture device*/ snd_pcm_writei(() if(error) xrun_recovery(pAlsaConfigP->playback_handle, retVal); } **** ERROR values returned from snd_pcm_writei: Only forst time : -11 Follwed by -EBADFD always static int xrun_recovery(snd_pcm_t *handle, int err) { if (err == -EPIPE) { /* under-run */ err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(err)); return 0; } else if (err == -ESTRPIPE) { while ((err = snd_pcm_resume(handle)) == -EAGAIN) sleep(1); /* wait until the suspend flag is released */ if (err < 0) { err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from suspend, prepare failed: %s\n", snd_strerror(err)); } return 0; } return err; } SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

2 5

[alsa-devel] [PATCH 0/6] ALSA: VIA HDA: patch series
by Lydia Wang 24 Mar '11

24 Mar '11

Following patch series mainly includes: -fix side channel mute invalid issue of VT2002P, VT1812 and VT1802 -add VT1802 check in via_speaker_automute function -update VT1708 initial verbs -update VT1718S initial verbs -update VT2002P initial verbs -update unsolicited event function to process more events

2 1

[alsa-devel] [PATCH 6/6] ALSA: VIA HDA: Update unsolicited event function
by Lydia Wang 24 Mar '11

24 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Update unsolicited event function. Update unsolicited event process function via_unsol_event() to make it can process more unsolicited events. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -234,12 +234,12 @@ return codec_type; }; +#define VIA_JACK_EVENT 0x20 #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 -#define VIA_JACK_EVENT 0x04 -#define VIA_MONO_EVENT 0x08 -#define VIA_SPEAKER_EVENT 0x10 -#define VIA_BIND_HP_EVENT 0x20 +#define VIA_MONO_EVENT 0x03 +#define VIA_SPEAKER_EVENT 0x04 +#define VIA_BIND_HP_EVENT 0x05 enum { VIA_CTL_WIDGET_VOL, @@ -1746,17 +1746,21 @@ unsigned int res) { res >>= 26; - if (res & VIA_HP_EVENT) - via_hp_automute(codec); - if (res & VIA_GPIO_EVENT) - via_gpio_control(codec); + if (res & VIA_JACK_EVENT) set_widgets_power_state(codec); - if (res & VIA_MONO_EVENT) + + res &= ~VIA_JACK_EVENT; + + if (res == VIA_HP_EVENT) + via_hp_automute(codec); + else if (res == VIA_GPIO_EVENT) + via_gpio_control(codec); + else if (res == VIA_MONO_EVENT) via_mono_automute(codec); - if (res & VIA_SPEAKER_EVENT) + else if (res == VIA_SPEAKER_EVENT) via_speaker_automute(codec); - if (res & VIA_BIND_HP_EVENT) + else if (res == VIA_BIND_HP_EVENT) via_hp_bind_automute(codec); }

1 0

[alsa-devel] [PATCH 5/6] ALSA: VIA HDA: Update VT2002P initial verbs
by Lydia Wang 24 Mar '11

24 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Update VT2002P initial verbs. Add some hardware related verbs in VT2002P initial verbs. These verbs are used to fix Class-D speaker no sound issue. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 4 ++++ 1 file changed, 4 insertions(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -5267,6 +5267,10 @@ }; static struct hda_verb vt2002P_volume_init_verbs[] = { + /* Class-D speaker related verbs */ + {0x1, 0xfe0, 0x4}, + {0x1, 0xfe9, 0x80}, + {0x1, 0xfe2, 0x22}, /* * Unmute ADC0-1 and set the default input to mic-in */

1 0

[alsa-devel] [PATCH 4/6] ALSA: VIA HDA: Update VT1718S initial verbs
by Lydia Wang 24 Mar '11

24 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Update VT1718S initial verbs. Add a verb to enable control amplifier of stereo mixer in VT1718S initial verbs. Set stereo mixer default amplifier value as un-mute. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -4255,7 +4255,8 @@ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - + /* Enable MW0 adjust Gain 5 */ + {0x1, 0xfb2, 0x10}, /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback * mixer widget */ @@ -4264,7 +4265,7 @@ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, /* Setup default input of Front HP to MW9 */ {0x28, AC_VERB_SET_CONNECT_SEL, 0x1},

1 0

[alsa-devel] [PATCH 3/6] ALSA: VIA HDA: Update VT1708 initial verbs
by Lydia Wang 24 Mar '11

24 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Update VT1708 initial verbs. Add a verb of power down jack detect in VT1708 initial verbs. This verb is used to avoid noise caused by hardware issue. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 2 ++ 1 file changed, 2 insertions(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1200,6 +1200,8 @@ {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* power down jack detect function */ + {0x1, 0xf81, 0x1}, { } };

1 0

[alsa-devel] [PATCH 2/6] ALSA: VIA HDA: Add VT1802 check in via_speaker_automute function
by Lydia Wang 24 Mar '11

24 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add VT1802 check in via_speaker_automute function. Add VT1802 check in via_speaker_automute() function. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1685,7 +1685,7 @@ unsigned int hp_present; struct via_spec *spec = codec->spec; - if (spec->codec_type != VT2002P && spec->codec_type != VT1812) + if (!VT2002P_COMPATIBLE(spec)) return; hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);

1 0

[alsa-devel] [PATCH 1/6] ALSA: VIA HDA: Fix side channel mute invalid issue
by Lydia Wang 24 Mar '11

24 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix side channel mute invalid issue. Modify side_mute_channel() and update_side_mute_status() functions to fix invalid side channel mute issue of VT2002P, VT1812 and VT1802 codecs. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -708,6 +708,9 @@ case VT1709_10CH: return 0x29; case VT1708B_8CH: /* fall thru */ case VT1708S: return 0x27; + case VT2002P: return 0x19; + case VT1802: return 0x15; + case VT1812: return 0x15; default: return 0; } } @@ -716,13 +719,22 @@ { /* mute side channel */ struct via_spec *spec = codec->spec; - unsigned int parm = spec->hp_independent_mode - ? AMP_OUT_MUTE : AMP_OUT_UNMUTE; + unsigned int parm; hda_nid_t sw3 = side_mute_channel(spec); - if (sw3) - snd_hda_codec_write(codec, sw3, 0, AC_VERB_SET_AMP_GAIN_MUTE, - parm); + if (sw3) { + if (VT2002P_COMPATIBLE(spec)) + parm = spec->hp_independent_mode ? + AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1); + else + parm = spec->hp_independent_mode ? + AMP_OUT_MUTE : AMP_OUT_UNMUTE; + snd_hda_codec_write(codec, sw3, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm); + if (spec->codec_type == VT1812) + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm); + } return 0; }

1 0

[alsa-devel] [PATCH] ASoC: Explicitly say registerless widgets have no register
by Mark Brown 24 Mar '11

24 Mar '11

This stops code that handles widgets generically from attempting to access registers for these widgets. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 979ed84..ddc2b3d 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -45,25 +45,25 @@ /* platform domain */ #define SND_SOC_DAPM_INPUT(wname) \ { .id = snd_soc_dapm_input, .name = wname, .kcontrols = NULL, \ - .num_kcontrols = 0} + .num_kcontrols = 0, .reg = SND_SOC_NOPM } #define SND_SOC_DAPM_OUTPUT(wname) \ { .id = snd_soc_dapm_output, .name = wname, .kcontrols = NULL, \ - .num_kcontrols = 0} + .num_kcontrols = 0, .reg = SND_SOC_NOPM } #define SND_SOC_DAPM_MIC(wname, wevent) \ { .id = snd_soc_dapm_mic, .name = wname, .kcontrols = NULL, \ - .num_kcontrols = 0, .event = wevent, \ + .num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD} #define SND_SOC_DAPM_HP(wname, wevent) \ { .id = snd_soc_dapm_hp, .name = wname, .kcontrols = NULL, \ - .num_kcontrols = 0, .event = wevent, \ + .num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD} #define SND_SOC_DAPM_SPK(wname, wevent) \ { .id = snd_soc_dapm_spk, .name = wname, .kcontrols = NULL, \ - .num_kcontrols = 0, .event = wevent, \ + .num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD} #define SND_SOC_DAPM_LINE(wname, wevent) \ { .id = snd_soc_dapm_line, .name = wname, .kcontrols = NULL, \ - .num_kcontrols = 0, .event = wevent, \ + .num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD} /* path domain */ @@ -189,11 +189,11 @@ /* events that are pre and post DAPM */ #define SND_SOC_DAPM_PRE(wname, wevent) \ { .id = snd_soc_dapm_pre, .name = wname, .kcontrols = NULL, \ - .num_kcontrols = 0, .event = wevent, \ + .num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD} #define SND_SOC_DAPM_POST(wname, wevent) \ { .id = snd_soc_dapm_post, .name = wname, .kcontrols = NULL, \ - .num_kcontrols = 0, .event = wevent, \ + .num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD} /* stream domain */ -- 1.7.4.1

2 1

[alsa-devel] [PATCH 1/1] Support for Boss JS-8 Jam Station
by Keith A. Milner 24 Mar '11

24 Mar '11

From: Keith A. Milner <maillist(a)superlative.org> Signed-off-by: Keith A. Milner <maillist(a)superlative.org> diff --git a/usb/quirks-table.h b/usb/quirks-table.h index 196c753..1cefb4a 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -1581,6 +1581,33 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Boss JS-8 Jam Station */ + USB_DEVICE(0x0582, 0x0109), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "JS-8", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, +{ /* has ID 0x0110 when not in Advanced Driver mode */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x010f), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- 1.7.1

2 1

[alsa-devel] [PATCH] sound/oss/opl3: validate voice and channel indexes
by Dan Rosenberg 24 Mar '11

24 Mar '11

User-controllable indexes for voice and channel values may cause reading and writing beyond the bounds of their respective arrays, leading to potentially exploitable memory corruption. Validate these indexes. Signed-off-by: Dan Rosenberg <drosenberg(a)vsecurity.com> Cc: stable(a)kernel.org --- sound/oss/opl3.c | 15 +++++++++++++-- 1 files changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index 938c48c..e9d443e 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -849,6 +849,10 @@ static int opl3_load_patch(int dev, int format, const char __user *addr, static void opl3_panning(int dev, int voice, int value) { + + if (voice < 0 || voice >= devc->nr_voice) + return; + devc->voc[voice].panning = value; } @@ -1066,8 +1070,15 @@ static int opl3_alloc_voice(int dev, int chn, int note, struct voice_alloc_info static void opl3_setup_voice(int dev, int voice, int chn) { - struct channel_info *info = - &synth_devs[dev]->chn_info[chn]; + struct channel_info *info; + + if (voice < 0 || voice >= devc->nr_voice) + return; + + if (chn < 0 || chn > 15) + return; + + info = &synth_devs[dev]->chn_info[chn]; opl3_set_instr(dev, voice, info->pgm_num);

2 1

[alsa-devel] [PATCH v3] sound/oss: remove offset from load_patch callbacks
by Dan Rosenberg 24 Mar '11

24 Mar '11

Was: [PATCH] sound/oss/midi_synth: prevent underflow, use of uninitialized value, and signedness issue The offset passed to midi_synth_load_patch() can be essentially arbitrary. If it's greater than the header length, this will result in a copy_from_user(dst, src, negative_val). While this will just return -EFAULT on x86, on other architectures this may cause memory corruption. Additionally, the length field of the sysex_info structure may not be initialized prior to its use. Finally, a signed comparison may result in an unintentionally large loop. On suggestion by Takashi Iwai, version two removes the offset argument from the load_patch callbacks entirely, which also resolves similar issues in opl3. Compile tested only. v3 adjusts comments and hopefully gets copy offsets right. Signed-off-by: Dan Rosenberg <drosenberg(a)vsecurity.com> --- sound/oss/dev_table.h | 2 +- sound/oss/midi_synth.c | 30 +++++++++++++----------------- sound/oss/midi_synth.h | 2 +- sound/oss/opl3.c | 8 ++------ sound/oss/sequencer.c | 2 +- 5 files changed, 18 insertions(+), 26 deletions(-) diff --git a/sound/oss/dev_table.h b/sound/oss/dev_table.h index b7617be..0199a31 100644 --- a/sound/oss/dev_table.h +++ b/sound/oss/dev_table.h @@ -271,7 +271,7 @@ struct synth_operations void (*reset) (int dev); void (*hw_control) (int dev, unsigned char *event); int (*load_patch) (int dev, int format, const char __user *addr, - int offs, int count, int pmgr_flag); + int count, int pmgr_flag); void (*aftertouch) (int dev, int voice, int pressure); void (*controller) (int dev, int voice, int ctrl_num, int value); void (*panning) (int dev, int voice, int value); diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c index 3c09374..2292c23 100644 --- a/sound/oss/midi_synth.c +++ b/sound/oss/midi_synth.c @@ -476,7 +476,7 @@ EXPORT_SYMBOL(midi_synth_hw_control); int midi_synth_load_patch(int dev, int format, const char __user *addr, - int offs, int count, int pmgr_flag) + int count, int pmgr_flag) { int orig_dev = synth_devs[dev]->midi_dev; @@ -491,33 +491,29 @@ midi_synth_load_patch(int dev, int format, const char __user *addr, if (!prefix_cmd(orig_dev, 0xf0)) return 0; + /* Invalid patch format */ if (format != SYSEX_PATCH) - { -/* printk("MIDI Error: Invalid patch format (key) 0x%x\n", format);*/ return -EINVAL; - } + + /* Patch header too short */ if (count < hdr_size) - { -/* printk("MIDI Error: Patch header too short\n");*/ return -EINVAL; - } + count -= hdr_size; /* - * Copy the header from user space but ignore the first bytes which have - * been transferred already. + * Copy the header from user space */ - if(copy_from_user(&((char *) &sysex)[offs], &(addr)[offs], hdr_size - offs)) + if (copy_from_user(&sysex, addr, hdr_size)) return -EFAULT; - - if (count < sysex.len) - { -/* printk(KERN_WARNING "MIDI Warning: Sysex record too short (%d<%d)\n", count, (int) sysex.len);*/ + + /* Sysex record too short */ + if ((unsigned)count < (unsigned)sysex.len) sysex.len = count; - } - left = sysex.len; - src_offs = 0; + + left = sysex.len; + src_offs = 0; for (i = 0; i < left && !signal_pending(current); i++) { diff --git a/sound/oss/midi_synth.h b/sound/oss/midi_synth.h index 6bc9d00..b64ddd6 100644 --- a/sound/oss/midi_synth.h +++ b/sound/oss/midi_synth.h @@ -8,7 +8,7 @@ int midi_synth_open (int dev, int mode); void midi_synth_close (int dev); void midi_synth_hw_control (int dev, unsigned char *event); int midi_synth_load_patch (int dev, int format, const char __user * addr, - int offs, int count, int pmgr_flag); + int count, int pmgr_flag); void midi_synth_panning (int dev, int channel, int pressure); void midi_synth_aftertouch (int dev, int channel, int pressure); void midi_synth_controller (int dev, int channel, int ctrl_num, int value); diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index 938c48c..cbf9574 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -820,7 +820,7 @@ static void opl3_hw_control(int dev, unsigned char *event) } static int opl3_load_patch(int dev, int format, const char __user *addr, - int offs, int count, int pmgr_flag) + int count, int pmgr_flag) { struct sbi_instrument ins; @@ -830,11 +830,7 @@ static int opl3_load_patch(int dev, int format, const char __user *addr, return -EINVAL; } - /* - * What the fuck is going on here? We leave junk in the beginning - * of ins and then check the field pretty close to that beginning? - */ - if(copy_from_user(&((char *) &ins)[offs], addr + offs, sizeof(ins) - offs)) + if (copy_from_user(&ins, addr, sizeof(ins))) return -EFAULT; if (ins.channel < 0 || ins.channel >= SBFM_MAXINSTR) diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index 5ea1098..30bcfe4 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -241,7 +241,7 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun return -ENXIO; fmt = (*(short *) &event_rec[0]) & 0xffff; - err = synth_devs[dev]->load_patch(dev, fmt, buf, p + 4, c, 0); + err = synth_devs[dev]->load_patch(dev, fmt, buf + p, c, 0); if (err < 0) return err;

2 1

[alsa-devel] Problem and dirty Fix for hda-intel on ALC892
by Jan Losinski 23 Mar '11

23 Mar '11

Hello, Since last week I compiled 2.6.38 on my Desktop. After booting I noticed a regression at the sound stuff. The surround speaker where no longer controlled by the master volume control. I can control it separately, but not with the master control that controls the front, center and LFE volume. After some debugging I found some differencies at module-loading output in the kernel log and in the output from alsactl store command. The dmesg and the diff of the alsactl output can be found at the end of this message. After that I tried to understand the hda-codec code and debugged there with a lot of printks and reboots. It takes another couple of hours to find out that the index of the surround volume and switch was 0 at pre 2.6.38 and is now 1. So I wrote a dirty hack to fix this (please ignore the debugging printks): diff --git a/pci/hda/hda_codec.c b/pci/hda/hda_codec.c index ae5c5d5..d3fd6d2 100644 --- a/pci/hda/hda_codec.c +++ b/pci/hda/hda_codec.c @@ -3819,7 +3819,16 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) kctl->id.device = addr; if (idx > 0) kctl->id.index = idx; + + snd_printk(KERN_INFO "PRE snd_hda_ctl_add - addr = %d, idx = %d, kctl->id.name = %s, kctl->id.index = %d, kctl->id.device = %d, kctl->count = %d", + addr, idx, kctl->id.name, kctl->id.index, kctl->id.device, kctl->count); err = snd_hda_ctl_add(codec, 0, kctl); + snd_printk(KERN_INFO "PASSED snd_hda_ctl_add - err = %d, addr = %d, idx = %d, kctl->id.name = %s, kctl->id.index = %d, kctl->id.device = %d, kctl->count = %d", + err, addr, idx, kctl->id.name, kctl->id.index, kctl->id.device, kctl->count); + if (strncmp("Surround Playback Switch", kctl->id.name, strlen("Surround Playback Switch"))) + kctl->id.index = 0; + if (strncmp("Surround Playback Volume", kctl->id.name, strlen("Surround Playback Volume"))) + kctl->id.index = 0; if (!err) break; /* try first with another device index corresponding to At all I dont understand what I've exactly done. I also know that this is not a real solution, because it will break the code for many other plattforms. So I hope someone can convert this to a real fix cor can guide me to make such one. The platform is a ASUS M4A89GTD-PRO/USB3 Board. If any further information needed, please tell me. Thanks, Jan Losinski dmesg of 2.6.37 HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 hda_intel: position_fix set to 1 for device 1043:8410 ALSA sound/pci/hda/hda_intel.c:2532: chipset global capabilities = 0x4401 ALSA sound/pci/hda/hda_intel.c:913: codec_mask = 0x1 ALSA sound/pci/hda/hda_intel.c:1351: codec #0 probed OK ALSA sound/pci/hda/patch_realtek.c:1524: SKU: Nid=0x1d sku_cfg=0x4005e601 ALSA sound/pci/hda/patch_realtek.c:1526: SKU: port_connectivity=0x1 ALSA sound/pci/hda/patch_realtek.c:1527: SKU: enable_pcbeep=0x0 ALSA sound/pci/hda/patch_realtek.c:1528: SKU: check_sum=0x00000005 ALSA sound/pci/hda/patch_realtek.c:1529: SKU: customization=0x000000e6 ALSA sound/pci/hda/patch_realtek.c:1530: SKU: external_amp=0x0 ALSA sound/pci/hda/patch_realtek.c:1531: SKU: platform_type=0x0 ALSA sound/pci/hda/patch_realtek.c:1532: SKU: swap=0x0 ALSA sound/pci/hda/patch_realtek.c:1533: SKU: override=0x1 hda_codec: ALC892: BIOS auto-probing. ALSA sound/pci/hda/hda_codec.c:4633: autoconfig: line_outs=4 (0x14/0x15/0x16/0x17/0x0) ALSA sound/pci/hda/hda_codec.c:4637: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:4641: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:4642: mono: mono_out=0x0 ALSA sound/pci/hda/hda_codec.c:4645: dig-out=0x11/0x1e ALSA sound/pci/hda/hda_codec.c:4646: inputs: ALSA sound/pci/hda/hda_codec.c:4650: Rear Mic=0x18 ALSA sound/pci/hda/hda_codec.c:4650: Front Mic=0x19 ALSA sound/pci/hda/hda_codec.c:4650: Line=0x1a ALSA sound/pci/hda/hda_codec.c:4652: ALSA sound/pci/hda/patch_realtek.c:1581: realtek: No valid SSID, checking pincfg 0x4005e601 for NID 0x1d ALSA sound/pci/hda/patch_realtek.c:1597: realtek: Enabling init ASM_ID=0xe601 CODEC_ID=10ec0892 ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Side Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Speaker Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Mono Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Line-Out Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave PCM Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Side Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Speaker Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Mono Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Line-Out Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave PCM Playback Switch, skipped dmesg of 2.6.38 HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 hda_intel: position_fix set to 1 for device 1043:8410 ALSA sound/pci/hda/hda_intel.c:2535: chipset global capabilities = 0x4401 ALSA sound/pci/hda/hda_intel.c:913: codec_mask = 0x1 ALSA sound/pci/hda/hda_intel.c:1353: codec #0 probed OK ALSA sound/pci/hda/patch_realtek.c:1527: SKU: Nid=0x1d sku_cfg=0x4005e601 ALSA sound/pci/hda/patch_realtek.c:1529: SKU: port_connectivity=0x1 ALSA sound/pci/hda/patch_realtek.c:1530: SKU: enable_pcbeep=0x0 ALSA sound/pci/hda/patch_realtek.c:1531: SKU: check_sum=0x00000005 ALSA sound/pci/hda/patch_realtek.c:1532: SKU: customization=0x000000e6 ALSA sound/pci/hda/patch_realtek.c:1533: SKU: external_amp=0x0 ALSA sound/pci/hda/patch_realtek.c:1534: SKU: platform_type=0x0 ALSA sound/pci/hda/patch_realtek.c:1535: SKU: swap=0x0 ALSA sound/pci/hda/patch_realtek.c:1536: SKU: override=0x1 hda_codec: ALC892: BIOS auto-probing. ALSA sound/pci/hda/hda_codec.c:4633: autoconfig: line_outs=4 (0x14/0x15/0x16/0x17/0x0) ALSA sound/pci/hda/hda_codec.c:4637: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:4641: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:4642: mono: mono_out=0x0 ALSA sound/pci/hda/hda_codec.c:4645: dig-out=0x11/0x1e ALSA sound/pci/hda/hda_codec.c:4646: inputs: ALSA sound/pci/hda/hda_codec.c:4650: Rear Mic=0x18 ALSA sound/pci/hda/hda_codec.c:4650: Front Mic=0x19 ALSA sound/pci/hda/hda_codec.c:4650: Line=0x1a ALSA sound/pci/hda/hda_codec.c:4652: ALSA sound/pci/hda/patch_realtek.c:1584: realtek: No valid SSID, checking pincfg 0x4005e601 for NID 0x1d ALSA sound/pci/hda/patch_realtek.c:1600: realtek: Enabling init ASM_ID=0xe601 CODEC_ID=10ec0892 ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Surround Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Side Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Speaker Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Mono Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Line-Out Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave PCM Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Surround Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Side Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Speaker Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Mono Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave Line-Out Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:2164: Cannot find slave PCM Playback Switch, skipped Diff of alsactl store output: --- store.27 2011-03-17 15:17:19.850768983 +0100 +++ store.28 2011-03-17 15:19:46.996710838 +0100 @@ -29,8 +29,9 @@ comment.dbmax 0 iface MIXER name 'Surround Playback Volume' - value.0 64 - value.1 64 + index 1 + value.0 38 + value.1 38 } control.4 { comment.access 'read write' @@ -38,6 +39,7 @@ comment.count 2 iface MIXER name 'Surround Playback Switch' + index 1 value.0 true value.1 true } @@ -130,7 +132,6 @@ comment.dbmax 1200 iface MIXER name 'Front Mic Playback Volume' - index 1 value.0 0 value.1 0 } @@ -140,7 +141,6 @@ comment.count 2 iface MIXER name 'Front Mic Playback Switch' - index 1 value.0 false value.1 false } @@ -173,7 +173,7 @@ comment.dbmin 0 comment.dbmax 3000 iface MIXER - name 'Rear Mic Boost' + name 'Rear Mic Boost Volume' value.0 0 value.1 0 } @@ -185,8 +185,7 @@ comment.dbmin 0 comment.dbmax 3000 iface MIXER - name 'Front Mic Boost' - index 1 + name 'Front Mic Boost Volume' value.0 0 value.1 0 } @@ -314,7 +313,7 @@ comment.count 1 iface MIXER name 'Master Playback Switch' - value true + value false } control.32 { comment.access 'read write user'

3 5

[alsa-devel] [PATCH v2] sound/oss: remove offset from load_patch callbacks
by Dan Rosenberg 23 Mar '11

23 Mar '11

Was: [PATCH] sound/oss/midi_synth: prevent underflow, use of uninitialized value, and signedness issue The offset passed to midi_synth_load_patch() can be essentially arbitrary. If it's greater than the header length, this will result in a copy_from_user(dst, src, negative_val). While this will just return -EFAULT on x86, on other architectures this may cause memory corruption. Additionally, the length field of the sysex_info structure may not be initialized prior to its use. Finally, a signed comparison may result in an unintentionally large loop. On suggestion by Takashi Iwai, version two removes the offset argument from the load_patch callbacks entirely, which also resolves similar issues in opl3. Compile tested only. Signed-off-by: Dan Rosenberg <drosenberg(a)vsecurity.com> Cc: stable(a)kernel.org --- sound/oss/dev_table.h | 2 +- sound/oss/midi_synth.c | 27 ++++++++++++--------------- sound/oss/midi_synth.h | 2 +- sound/oss/opl3.c | 4 ++-- sound/oss/sequencer.c | 2 +- 5 files changed, 17 insertions(+), 20 deletions(-) diff --git a/sound/oss/dev_table.h b/sound/oss/dev_table.h index b7617be..0199a31 100644 --- a/sound/oss/dev_table.h +++ b/sound/oss/dev_table.h @@ -271,7 +271,7 @@ struct synth_operations void (*reset) (int dev); void (*hw_control) (int dev, unsigned char *event); int (*load_patch) (int dev, int format, const char __user *addr, - int offs, int count, int pmgr_flag); + int count, int pmgr_flag); void (*aftertouch) (int dev, int voice, int pressure); void (*controller) (int dev, int voice, int ctrl_num, int value); void (*panning) (int dev, int voice, int value); diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c index 3c09374..8d3611c 100644 --- a/sound/oss/midi_synth.c +++ b/sound/oss/midi_synth.c @@ -476,7 +476,7 @@ EXPORT_SYMBOL(midi_synth_hw_control); int midi_synth_load_patch(int dev, int format, const char __user *addr, - int offs, int count, int pmgr_flag) + int count, int pmgr_flag) { int orig_dev = synth_devs[dev]->midi_dev; @@ -491,16 +491,14 @@ midi_synth_load_patch(int dev, int format, const char __user *addr, if (!prefix_cmd(orig_dev, 0xf0)) return 0; + /* Invalid patch format */ if (format != SYSEX_PATCH) - { -/* printk("MIDI Error: Invalid patch format (key) 0x%x\n", format);*/ return -EINVAL; - } + + /* Patch header too short */ if (count < hdr_size) - { -/* printk("MIDI Error: Patch header too short\n");*/ return -EINVAL; - } + count -= hdr_size; /* @@ -508,16 +506,15 @@ midi_synth_load_patch(int dev, int format, const char __user *addr, * been transferred already. */ - if(copy_from_user(&((char *) &sysex)[offs], &(addr)[offs], hdr_size - offs)) + if (copy_from_user(&sysex, addr, hdr_size)) return -EFAULT; - - if (count < sysex.len) - { -/* printk(KERN_WARNING "MIDI Warning: Sysex record too short (%d<%d)\n", count, (int) sysex.len);*/ + + /* Sysex record too short */ + if ((unsigned)count < (unsigned)sysex.len) sysex.len = count; - } - left = sysex.len; - src_offs = 0; + + left = sysex.len; + src_offs = 0; for (i = 0; i < left && !signal_pending(current); i++) { diff --git a/sound/oss/midi_synth.h b/sound/oss/midi_synth.h index 6bc9d00..b64ddd6 100644 --- a/sound/oss/midi_synth.h +++ b/sound/oss/midi_synth.h @@ -8,7 +8,7 @@ int midi_synth_open (int dev, int mode); void midi_synth_close (int dev); void midi_synth_hw_control (int dev, unsigned char *event); int midi_synth_load_patch (int dev, int format, const char __user * addr, - int offs, int count, int pmgr_flag); + int count, int pmgr_flag); void midi_synth_panning (int dev, int channel, int pressure); void midi_synth_aftertouch (int dev, int channel, int pressure); void midi_synth_controller (int dev, int channel, int ctrl_num, int value); diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index 938c48c..a3bec38 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -820,7 +820,7 @@ static void opl3_hw_control(int dev, unsigned char *event) } static int opl3_load_patch(int dev, int format, const char __user *addr, - int offs, int count, int pmgr_flag) + int count, int pmgr_flag) { struct sbi_instrument ins; @@ -834,7 +834,7 @@ static int opl3_load_patch(int dev, int format, const char __user *addr, * What the fuck is going on here? We leave junk in the beginning * of ins and then check the field pretty close to that beginning? */ - if(copy_from_user(&((char *) &ins)[offs], addr + offs, sizeof(ins) - offs)) + if (copy_from_user(&ins, addr, sizeof(ins))) return -EFAULT; if (ins.channel < 0 || ins.channel >= SBFM_MAXINSTR) diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index 5ea1098..2e842cb 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -241,7 +241,7 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun return -ENXIO; fmt = (*(short *) &event_rec[0]) & 0xffff; - err = synth_devs[dev]->load_patch(dev, fmt, buf, p + 4, c, 0); + err = synth_devs[dev]->load_patch(dev, fmt, buf, c, 0); if (err < 0) return err;

2 3

[alsa-devel] PROBLEM: Asoc driver in 2.6.37.3 for AT91SAM9260 / TLV320AIC3X is broken
by Prchal Jiří 23 Mar '11

23 Mar '11

Hallo, at first I would like to apologize of sending this report to too many mails, but I can't find the right person / list. [2.] Full description of the problem/report: I try for almost 2 month to make kernel with alsa for my hardware. I first try version 2.6.37, now trying 2.6.37.3. My hw i similar to board afeb-9260, just with difference my codec is TLV320AIC3106 connected as slave with clock on BCLK. So I change file snd-soc-afeb9260.c, diff is attached, and Kconfig and Makefile too. In version 2.6.35.11 it works good, but MMC/SD card doesn't work. (Something is broken it that driver just in this version). I need to work sound and SD card. I thing the problem is somewhere between "tlv320aic3x-codec 0-001a: probe" and "i2c-core: driver [tlv320aic3x-codec] registered". In version 2.6.35.11 there is I2C communication, in 2.6.37 and higher nothing on I2C. I also try to put newer alsa 1.0.24 to kernel 2.6.37.3 without any improvement. Trying to put older alsa (but same version?) from kernel 2.6.35.11 to 2.6.37.3 produce soundcard, but when try to play kernel oops, "cannot dereference null pointer" and freezes. I try many versions of /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link afeb9260_dai = { .name = "TLV320AIC3X", //AIC3X .stream_name = "AIC3X", .cpu_dai_name = "atmel-ssc-dai.0", //atmel-ssc-dai .codec_dai_name = "tlv320aic3x-hifi", //aic3x, aic3x-hifi, tlv320aic3x .platform_name = "atmel-pcm-audio", //atmel_pcm-audio .codec_name = "tlv320aic3x-codec.0-001a", //tlv320aic3x-codec.0-0x1a .init = afeb9260_aic3x_init, .ops = &afeb9260_ops, }; but without any improvement. I really don't know what should I try at next, I thing I've tryed all. [3.] Keywords (i.e., modules, networking, kernel): ALSA, Asoc, soc, audio, sound, codec [4.] Kernel information [4.1.] Kernel version (from /proc/version): Linux version 2.6.37.3-cdu (prchal@prchal) (gcc version 4.4.5 (GCC) ) #26 PREEMPT Thu Mar 10 16:41:10 CET 2011 [4.2.] Kernel .config file: # # Automatically generated make config: don't edit # Linux/arm 2.6.37.3 Kernel Configuration # Thu Mar 10 16:40:55 2011 # CONFIG_ARM=y CONFIG_SYS_SUPPORTS_APM_EMULATION=y CONFIG_GENERIC_GPIO=y # CONFIG_ARCH_USES_GETTIMEOFFSET is not set CONFIG_GENERIC_CLOCKEVENTS=y CONFIG_HAVE_PROC_CPU=y CONFIG_GENERIC_HARDIRQS=y CONFIG_STACKTRACE_SUPPORT=y CONFIG_HAVE_LATENCYTOP_SUPPORT=y CONFIG_LOCKDEP_SUPPORT=y CONFIG_TRACE_IRQFLAGS_SUPPORT=y CONFIG_HARDIRQS_SW_RESEND=y CONFIG_GENERIC_IRQ_PROBE=y CONFIG_RWSEM_GENERIC_SPINLOCK=y CONFIG_ARCH_HAS_CPU_IDLE_WAIT=y CONFIG_GENERIC_HWEIGHT=y CONFIG_GENERIC_CALIBRATE_DELAY=y CONFIG_NEED_DMA_MAP_STATE=y CONFIG_GENERIC_HARDIRQS_NO__DO_IRQ=y CONFIG_VECTORS_BASE=0xffff0000 CONFIG_DEFCONFIG_LIST="/lib/modules/$UNAME_RELEASE/.config" CONFIG_CONSTRUCTORS=y CONFIG_HAVE_IRQ_WORK=y # # General setup # CONFIG_EXPERIMENTAL=y CONFIG_BROKEN_ON_SMP=y CONFIG_LOCK_KERNEL=y CONFIG_INIT_ENV_ARG_LIMIT=32 CONFIG_CROSS_COMPILE="" CONFIG_LOCALVERSION="-cdu" CONFIG_LOCALVERSION_AUTO=y CONFIG_HAVE_KERNEL_GZIP=y CONFIG_HAVE_KERNEL_LZMA=y CONFIG_HAVE_KERNEL_LZO=y CONFIG_KERNEL_GZIP=y # CONFIG_KERNEL_LZMA is not set # CONFIG_KERNEL_LZO is not set # CONFIG_SWAP is not set CONFIG_SYSVIPC=y CONFIG_SYSVIPC_SYSCTL=y CONFIG_POSIX_MQUEUE=y CONFIG_POSIX_MQUEUE_SYSCTL=y # CONFIG_BSD_PROCESS_ACCT is not set # CONFIG_TASKSTATS is not set # CONFIG_AUDIT is not set # CONFIG_HAVE_GENERIC_HARDIRQS is not set # CONFIG_SPARSE_IRQ is not set # # RCU Subsystem # CONFIG_TREE_PREEMPT_RCU=y # CONFIG_TINY_RCU is not set # CONFIG_TINY_PREEMPT_RCU is not set CONFIG_PREEMPT_RCU=y # CONFIG_RCU_TRACE is not set CONFIG_RCU_FANOUT=32 # CONFIG_RCU_FANOUT_EXACT is not set # CONFIG_TREE_RCU_TRACE is not set # CONFIG_IKCONFIG is not set CONFIG_LOG_BUF_SHIFT=14 # CONFIG_CGROUPS is not set CONFIG_NAMESPACES=y CONFIG_UTS_NS=y CONFIG_IPC_NS=y CONFIG_USER_NS=y CONFIG_PID_NS=y CONFIG_NET_NS=y # CONFIG_SYSFS_DEPRECATED is not set # CONFIG_RELAY is not set # CONFIG_BLK_DEV_INITRD is not set # CONFIG_CC_OPTIMIZE_FOR_SIZE is not set CONFIG_SYSCTL=y CONFIG_ANON_INODES=y # CONFIG_EMBEDDED is not set CONFIG_UID16=y CONFIG_SYSCTL_SYSCALL=y CONFIG_KALLSYMS=y # CONFIG_KALLSYMS_ALL is not set # CONFIG_KALLSYMS_EXTRA_PASS is not set CONFIG_HOTPLUG=y CONFIG_PRINTK=y CONFIG_BUG=y CONFIG_ELF_CORE=y CONFIG_BASE_FULL=y CONFIG_FUTEX=y CONFIG_EPOLL=y CONFIG_SIGNALFD=y CONFIG_TIMERFD=y CONFIG_EVENTFD=y CONFIG_SHMEM=y CONFIG_AIO=y CONFIG_HAVE_PERF_EVENTS=y CONFIG_PERF_USE_VMALLOC=y # # Kernel Performance Events And Counters # # CONFIG_PERF_EVENTS is not set # CONFIG_PERF_COUNTERS is not set CONFIG_VM_EVENT_COUNTERS=y CONFIG_COMPAT_BRK=y CONFIG_SLAB=y # CONFIG_SLUB is not set # CONFIG_PROFILING is not set CONFIG_HAVE_OPROFILE=y # CONFIG_KPROBES is not set CONFIG_HAVE_KPROBES=y CONFIG_HAVE_KRETPROBES=y CONFIG_HAVE_REGS_AND_STACK_ACCESS_API=y CONFIG_HAVE_CLK=y # # GCOV-based kernel profiling # # CONFIG_GCOV_KERNEL is not set CONFIG_HAVE_GENERIC_DMA_COHERENT=y CONFIG_SLABINFO=y CONFIG_RT_MUTEXES=y CONFIG_BASE_SMALL=0 CONFIG_MODULES=y # CONFIG_MODULE_FORCE_LOAD is not set CONFIG_MODULE_UNLOAD=y # CONFIG_MODULE_FORCE_UNLOAD is not set # CONFIG_MODVERSIONS is not set # CONFIG_MODULE_SRCVERSION_ALL is not set CONFIG_BLOCK=y CONFIG_LBDAF=y # CONFIG_BLK_DEV_BSG is not set # CONFIG_BLK_DEV_INTEGRITY is not set # # IO Schedulers # CONFIG_IOSCHED_NOOP=y # CONFIG_IOSCHED_DEADLINE is not set # CONFIG_IOSCHED_CFQ is not set CONFIG_DEFAULT_NOOP=y CONFIG_DEFAULT_IOSCHED="noop" # CONFIG_INLINE_SPIN_TRYLOCK is not set # CONFIG_INLINE_SPIN_TRYLOCK_BH is not set # CONFIG_INLINE_SPIN_LOCK is not set # CONFIG_INLINE_SPIN_LOCK_BH is not set # CONFIG_INLINE_SPIN_LOCK_IRQ is not set # CONFIG_INLINE_SPIN_LOCK_IRQSAVE is not set # CONFIG_INLINE_SPIN_UNLOCK is not set # CONFIG_INLINE_SPIN_UNLOCK_BH is not set # CONFIG_INLINE_SPIN_UNLOCK_IRQ is not set # CONFIG_INLINE_SPIN_UNLOCK_IRQRESTORE is not set # CONFIG_INLINE_READ_TRYLOCK is not set # CONFIG_INLINE_READ_LOCK is not set # CONFIG_INLINE_READ_LOCK_BH is not set # CONFIG_INLINE_READ_LOCK_IRQ is not set # CONFIG_INLINE_READ_LOCK_IRQSAVE is not set # CONFIG_INLINE_READ_UNLOCK is not set # CONFIG_INLINE_READ_UNLOCK_BH is not set # CONFIG_INLINE_READ_UNLOCK_IRQ is not set # CONFIG_INLINE_READ_UNLOCK_IRQRESTORE is not set # CONFIG_INLINE_WRITE_TRYLOCK is not set # CONFIG_INLINE_WRITE_LOCK is not set # CONFIG_INLINE_WRITE_LOCK_BH is not set # CONFIG_INLINE_WRITE_LOCK_IRQ is not set # CONFIG_INLINE_WRITE_LOCK_IRQSAVE is not set # CONFIG_INLINE_WRITE_UNLOCK is not set # CONFIG_INLINE_WRITE_UNLOCK_BH is not set # CONFIG_INLINE_WRITE_UNLOCK_IRQ is not set # CONFIG_INLINE_WRITE_UNLOCK_IRQRESTORE is not set # CONFIG_MUTEX_SPIN_ON_OWNER is not set # CONFIG_FREEZER is not set # # System Type # CONFIG_MMU=y # CONFIG_ARCH_AAEC2000 is not set # CONFIG_ARCH_INTEGRATOR is not set # CONFIG_ARCH_REALVIEW is not set # CONFIG_ARCH_VERSATILE is not set # CONFIG_ARCH_VEXPRESS is not set CONFIG_ARCH_AT91=y # CONFIG_ARCH_BCMRING is not set # CONFIG_ARCH_CLPS711X is not set # CONFIG_ARCH_CNS3XXX is not set # CONFIG_ARCH_GEMINI is not set # CONFIG_ARCH_EBSA110 is not set # CONFIG_ARCH_EP93XX is not set # CONFIG_ARCH_FOOTBRIDGE is not set # CONFIG_ARCH_MXC is not set # CONFIG_ARCH_STMP3XXX is not set # CONFIG_ARCH_NETX is not set # CONFIG_ARCH_H720X is not set # CONFIG_ARCH_IOP13XX is not set # CONFIG_ARCH_IOP32X is not set # CONFIG_ARCH_IOP33X is not set # CONFIG_ARCH_IXP23XX is not set # CONFIG_ARCH_IXP2000 is not set # CONFIG_ARCH_IXP4XX is not set # CONFIG_ARCH_DOVE is not set # CONFIG_ARCH_KIRKWOOD is not set # CONFIG_ARCH_LOKI is not set # CONFIG_ARCH_LPC32XX is not set # CONFIG_ARCH_MV78XX0 is not set # CONFIG_ARCH_ORION5X is not set # CONFIG_ARCH_MMP is not set # CONFIG_ARCH_KS8695 is not set # CONFIG_ARCH_NS9XXX is not set # CONFIG_ARCH_W90X900 is not set # CONFIG_ARCH_NUC93X is not set # CONFIG_ARCH_TEGRA is not set # CONFIG_ARCH_PNX4008 is not set # CONFIG_ARCH_PXA is not set # CONFIG_ARCH_MSM is not set # CONFIG_ARCH_SHMOBILE is not set # CONFIG_ARCH_RPC is not set # CONFIG_ARCH_SA1100 is not set # CONFIG_ARCH_S3C2410 is not set # CONFIG_ARCH_S3C64XX is not set # CONFIG_ARCH_S5P64X0 is not set # CONFIG_ARCH_S5P6442 is not set # CONFIG_ARCH_S5PC100 is not set # CONFIG_ARCH_S5PV210 is not set # CONFIG_ARCH_S5PV310 is not set # CONFIG_ARCH_SHARK is not set # CONFIG_ARCH_TCC_926 is not set # CONFIG_ARCH_LH7A40X is not set # CONFIG_ARCH_U300 is not set # CONFIG_ARCH_U8500 is not set # CONFIG_ARCH_NOMADIK is not set # CONFIG_ARCH_DAVINCI is not set # CONFIG_ARCH_OMAP is not set # CONFIG_PLAT_SPEAR is not set CONFIG_HAVE_AT91_USART3=y CONFIG_HAVE_AT91_USART4=y CONFIG_HAVE_AT91_USART5=y # # Atmel AT91 System-on-Chip # # CONFIG_ARCH_AT91RM9200 is not set CONFIG_ARCH_AT91SAM9260=y # CONFIG_ARCH_AT91SAM9261 is not set # CONFIG_ARCH_AT91SAM9G10 is not set # CONFIG_ARCH_AT91SAM9263 is not set # CONFIG_ARCH_AT91SAM9RL is not set # CONFIG_ARCH_AT91SAM9G20 is not set # CONFIG_ARCH_AT91SAM9G45 is not set # CONFIG_ARCH_AT91CAP9 is not set # CONFIG_ARCH_AT572D940HF is not set # CONFIG_ARCH_AT91X40 is not set CONFIG_AT91_PMC_UNIT=y # # AT91SAM9260 Variants # # CONFIG_ARCH_AT91SAM9260_SAM9XE is not set # # AT91SAM9260 / AT91SAM9XE Board Type # # CONFIG_MACH_AT91SAM9260EK is not set CONFIG_MACH_CDU=y # CONFIG_MACH_CAM60 is not set # CONFIG_MACH_SAM9_L9260 is not set # CONFIG_MACH_AFEB9260 is not set # CONFIG_MACH_USB_A9260 is not set # CONFIG_MACH_QIL_A9260 is not set # CONFIG_MACH_CPU9260 is not set # CONFIG_MACH_FLEXIBITY is not set # # AT91SAM9260/AT91SAM9G20 boards # # CONFIG_MACH_SNAPPER_9260 is not set # # AT91 Board Options # # # AT91 Feature Selections # CONFIG_AT91_PROGRAMMABLE_CLOCKS=y CONFIG_AT91_TIMER_HZ=100 CONFIG_AT91_EARLY_DBGU=y # CONFIG_AT91_EARLY_USART0 is not set # CONFIG_AT91_EARLY_USART1 is not set # CONFIG_AT91_EARLY_USART2 is not set # CONFIG_AT91_EARLY_USART3 is not set # CONFIG_AT91_EARLY_USART4 is not set # CONFIG_AT91_EARLY_USART5 is not set # # Processor Type # CONFIG_CPU_ARM926T=y CONFIG_CPU_32v5=y CONFIG_CPU_ABRT_EV5TJ=y CONFIG_CPU_PABRT_LEGACY=y CONFIG_CPU_CACHE_VIVT=y CONFIG_CPU_COPY_V4WB=y CONFIG_CPU_TLB_V4WBI=y CONFIG_CPU_CP15=y CONFIG_CPU_CP15_MMU=y # # Processor Features # CONFIG_ARM_THUMB=y # CONFIG_CPU_ICACHE_DISABLE is not set # CONFIG_CPU_DCACHE_DISABLE is not set # CONFIG_CPU_DCACHE_WRITETHROUGH is not set # CONFIG_CPU_CACHE_ROUND_ROBIN is not set CONFIG_ARM_L1_CACHE_SHIFT=5 # # Bus support # # CONFIG_PCI_SYSCALL is not set # CONFIG_ARCH_SUPPORTS_MSI is not set # CONFIG_PCCARD is not set # # Kernel Features # CONFIG_TICK_ONESHOT=y # CONFIG_NO_HZ is not set CONFIG_HIGH_RES_TIMERS=y CONFIG_GENERIC_CLOCKEVENTS_BUILD=y CONFIG_VMSPLIT_3G=y # CONFIG_VMSPLIT_2G is not set # CONFIG_VMSPLIT_1G is not set CONFIG_PAGE_OFFSET=0xC0000000 # CONFIG_PREEMPT_NONE is not set # CONFIG_PREEMPT_VOLUNTARY is not set CONFIG_PREEMPT=y CONFIG_HZ=100 CONFIG_AEABI=y # CONFIG_OABI_COMPAT is not set # CONFIG_ARCH_SPARSEMEM_DEFAULT is not set # CONFIG_ARCH_SELECT_MEMORY_MODEL is not set # CONFIG_HIGHMEM is not set CONFIG_SELECT_MEMORY_MODEL=y CONFIG_FLATMEM_MANUAL=y CONFIG_FLATMEM=y CONFIG_FLAT_NODE_MEM_MAP=y CONFIG_HAVE_MEMBLOCK=y CONFIG_PAGEFLAGS_EXTENDED=y CONFIG_SPLIT_PTLOCK_CPUS=999999 # CONFIG_PHYS_ADDR_T_64BIT is not set CONFIG_ZONE_DMA_FLAG=0 CONFIG_VIRT_TO_BUS=y # CONFIG_KSM is not set CONFIG_DEFAULT_MMAP_MIN_ADDR=4096 CONFIG_NEED_PER_CPU_KM=y CONFIG_FORCE_MAX_ZONEORDER=11 # CONFIG_LEDS is not set CONFIG_ALIGNMENT_TRAP=y # CONFIG_UACCESS_WITH_MEMCPY is not set # CONFIG_SECCOMP is not set # CONFIG_CC_STACKPROTECTOR is not set # CONFIG_DEPRECATED_PARAM_STRUCT is not set # # Boot options # CONFIG_ZBOOT_ROM_TEXT=0x0 CONFIG_ZBOOT_ROM_BSS=0x0 CONFIG_CMDLINE="mem=64M console=ttyS0,115200 initrd=0x21100000,3145728 root=/dev/ram0 rw" # CONFIG_CMDLINE_FORCE is not set # CONFIG_XIP_KERNEL is not set # CONFIG_KEXEC is not set # CONFIG_AUTO_ZRELADDR is not set # # CPU Power Management # # CONFIG_CPU_IDLE is not set # # Floating point emulation # # # At least one emulation must be selected # CONFIG_VFP=y # # Userspace binary formats # CONFIG_BINFMT_ELF=y CONFIG_CORE_DUMP_DEFAULT_ELF_HEADERS=y CONFIG_HAVE_AOUT=y # CONFIG_BINFMT_AOUT is not set # CONFIG_BINFMT_MISC is not set # # Power management options # # CONFIG_PM is not set CONFIG_ARCH_SUSPEND_POSSIBLE=y CONFIG_NET=y # # Networking options # CONFIG_PACKET=y CONFIG_UNIX=y # CONFIG_NET_KEY is not set CONFIG_INET=y CONFIG_IP_MULTICAST=y CONFIG_IP_ADVANCED_ROUTER=y CONFIG_ASK_IP_FIB_HASH=y # CONFIG_IP_FIB_TRIE is not set CONFIG_IP_FIB_HASH=y # CONFIG_IP_MULTIPLE_TABLES is not set # CONFIG_IP_ROUTE_MULTIPATH is not set # CONFIG_IP_ROUTE_VERBOSE is not set CONFIG_IP_PNP=y # CONFIG_IP_PNP_DHCP is not set CONFIG_IP_PNP_BOOTP=y # CONFIG_IP_PNP_RARP is not set # CONFIG_NET_IPIP is not set # CONFIG_NET_IPGRE_DEMUX is not set # CONFIG_IP_MROUTE is not set # CONFIG_ARPD is not set # CONFIG_SYN_COOKIES is not set # CONFIG_INET_AH is not set # CONFIG_INET_ESP is not set # CONFIG_INET_IPCOMP is not set # CONFIG_INET_XFRM_TUNNEL is not set # CONFIG_INET_TUNNEL is not set # CONFIG_INET_XFRM_MODE_TRANSPORT is not set # CONFIG_INET_XFRM_MODE_TUNNEL is not set # CONFIG_INET_XFRM_MODE_BEET is not set # CONFIG_INET_LRO is not set CONFIG_INET_DIAG=y CONFIG_INET_TCP_DIAG=y # CONFIG_TCP_CONG_ADVANCED is not set CONFIG_TCP_CONG_CUBIC=y CONFIG_DEFAULT_TCP_CONG="cubic" # CONFIG_TCP_MD5SIG is not set # CONFIG_IPV6 is not set # CONFIG_NETWORK_SECMARK is not set # CONFIG_NETWORK_PHY_TIMESTAMPING is not set # CONFIG_NETFILTER is not set # CONFIG_IP_DCCP is not set # CONFIG_IP_SCTP is not set # CONFIG_RDS is not set # CONFIG_TIPC is not set # CONFIG_ATM is not set # CONFIG_L2TP is not set # CONFIG_BRIDGE is not set # CONFIG_NET_DSA is not set # CONFIG_VLAN_8021Q is not set # CONFIG_DECNET is not set # CONFIG_LLC2 is not set # CONFIG_IPX is not set # CONFIG_ATALK is not set # CONFIG_X25 is not set # CONFIG_LAPB is not set # CONFIG_ECONET is not set # CONFIG_WAN_ROUTER is not set # CONFIG_PHONET is not set # CONFIG_IEEE802154 is not set # CONFIG_NET_SCHED is not set # CONFIG_DCB is not set # # Network testing # # CONFIG_NET_PKTGEN is not set # CONFIG_HAMRADIO is not set # CONFIG_CAN is not set # CONFIG_IRDA is not set # CONFIG_BT is not set # CONFIG_AF_RXRPC is not set # CONFIG_WIRELESS is not set # CONFIG_WIMAX is not set # CONFIG_RFKILL is not set # CONFIG_NET_9P is not set # CONFIG_CAIF is not set # CONFIG_CEPH_LIB is not set # # Device Drivers # # # Generic Driver Options # CONFIG_UEVENT_HELPER_PATH="/sbin/hotplug" CONFIG_DEVTMPFS=y CONFIG_DEVTMPFS_MOUNT=y CONFIG_STANDALONE=y CONFIG_PREVENT_FIRMWARE_BUILD=y CONFIG_FW_LOADER=y # CONFIG_FIRMWARE_IN_KERNEL is not set CONFIG_EXTRA_FIRMWARE="" # CONFIG_DEBUG_DRIVER is not set # CONFIG_DEBUG_DEVRES is not set # CONFIG_SYS_HYPERVISOR is not set # CONFIG_CONNECTOR is not set CONFIG_MTD=y CONFIG_MTD_DEBUG=y CONFIG_MTD_DEBUG_VERBOSE=1 # CONFIG_MTD_TESTS is not set # CONFIG_MTD_CONCAT is not set CONFIG_MTD_PARTITIONS=y # CONFIG_MTD_REDBOOT_PARTS is not set CONFIG_MTD_CMDLINE_PARTS=y # CONFIG_MTD_AFS_PARTS is not set # CONFIG_MTD_AR7_PARTS is not set # # User Modules And Translation Layers # # CONFIG_MTD_CHAR is not set CONFIG_MTD_BLKDEVS=y CONFIG_MTD_BLOCK=y # CONFIG_FTL is not set # CONFIG_NFTL is not set # CONFIG_INFTL is not set # CONFIG_RFD_FTL is not set # CONFIG_SSFDC is not set # CONFIG_SM_FTL is not set # CONFIG_MTD_OOPS is not set # # RAM/ROM/Flash chip drivers # # CONFIG_MTD_CFI is not set # CONFIG_MTD_JEDECPROBE is not set CONFIG_MTD_MAP_BANK_WIDTH_1=y CONFIG_MTD_MAP_BANK_WIDTH_2=y CONFIG_MTD_MAP_BANK_WIDTH_4=y # CONFIG_MTD_MAP_BANK_WIDTH_8 is not set # CONFIG_MTD_MAP_BANK_WIDTH_16 is not set # CONFIG_MTD_MAP_BANK_WIDTH_32 is not set CONFIG_MTD_CFI_I1=y CONFIG_MTD_CFI_I2=y # CONFIG_MTD_CFI_I4 is not set # CONFIG_MTD_CFI_I8 is not set # CONFIG_MTD_RAM is not set # CONFIG_MTD_ROM is not set # CONFIG_MTD_ABSENT is not set # # Mapping drivers for chip access # # CONFIG_MTD_COMPLEX_MAPPINGS is not set # CONFIG_MTD_PLATRAM is not set # # Self-contained MTD device drivers # # CONFIG_MTD_SLRAM is not set # CONFIG_MTD_PHRAM is not set # CONFIG_MTD_MTDRAM is not set # CONFIG_MTD_BLOCK2MTD is not set # # Disk-On-Chip Device Drivers # # CONFIG_MTD_DOC2000 is not set # CONFIG_MTD_DOC2001 is not set # CONFIG_MTD_DOC2001PLUS is not set CONFIG_MTD_NAND_ECC=y # CONFIG_MTD_NAND_ECC_SMC is not set CONFIG_MTD_NAND=y CONFIG_MTD_NAND_VERIFY_WRITE=y # CONFIG_MTD_SM_COMMON is not set # CONFIG_MTD_NAND_MUSEUM_IDS is not set # CONFIG_MTD_NAND_GPIO is not set CONFIG_MTD_NAND_IDS=y # CONFIG_MTD_NAND_DISKONCHIP is not set CONFIG_MTD_NAND_ATMEL=y # CONFIG_MTD_NAND_ATMEL_ECC_HW is not set CONFIG_MTD_NAND_ATMEL_ECC_SOFT=y # CONFIG_MTD_NAND_ATMEL_ECC_NONE is not set # CONFIG_MTD_NAND_NANDSIM is not set # CONFIG_MTD_NAND_PLATFORM is not set # CONFIG_MTD_ALAUDA is not set # CONFIG_MTD_ONENAND is not set # # LPDDR flash memory drivers # # CONFIG_MTD_LPDDR is not set # CONFIG_MTD_UBI is not set # CONFIG_PARPORT is not set # CONFIG_BLK_DEV is not set CONFIG_MISC_DEVICES=y # CONFIG_AD525X_DPOT is not set # CONFIG_ATMEL_TCLIB is not set # CONFIG_ICS932S401 is not set CONFIG_ATMEL_SSC=y # CONFIG_ENCLOSURE_SERVICES is not set # CONFIG_APDS9802ALS is not set # CONFIG_ISL29003 is not set # CONFIG_ISL29020 is not set # CONFIG_SENSORS_TSL2550 is not set # CONFIG_SENSORS_BH1780 is not set # CONFIG_SENSORS_BH1770 is not set # CONFIG_SENSORS_APDS990X is not set # CONFIG_HMC6352 is not set # CONFIG_DS1682 is not set # CONFIG_BMP085 is not set # CONFIG_C2PORT is not set # # EEPROM support # CONFIG_EEPROM_AT24=y # CONFIG_EEPROM_LEGACY is not set # CONFIG_EEPROM_MAX6875 is not set # CONFIG_EEPROM_93CX6 is not set # CONFIG_IWMC3200TOP is not set # # Texas Instruments shared transport line discipline # CONFIG_HAVE_IDE=y # CONFIG_IDE is not set # # SCSI device support # CONFIG_SCSI_MOD=y # CONFIG_RAID_ATTRS is not set CONFIG_SCSI=y CONFIG_SCSI_DMA=y # CONFIG_SCSI_TGT is not set # CONFIG_SCSI_NETLINK is not set # CONFIG_SCSI_PROC_FS is not set # # SCSI support type (disk, tape, CD-ROM) # CONFIG_BLK_DEV_SD=y # CONFIG_CHR_DEV_ST is not set # CONFIG_CHR_DEV_OSST is not set # CONFIG_BLK_DEV_SR is not set # CONFIG_CHR_DEV_SG is not set # CONFIG_CHR_DEV_SCH is not set # CONFIG_SCSI_MULTI_LUN is not set # CONFIG_SCSI_CONSTANTS is not set # CONFIG_SCSI_LOGGING is not set # CONFIG_SCSI_SCAN_ASYNC is not set CONFIG_SCSI_WAIT_SCAN=m # # SCSI Transports # # CONFIG_SCSI_SPI_ATTRS is not set # CONFIG_SCSI_FC_ATTRS is not set # CONFIG_SCSI_ISCSI_ATTRS is not set # CONFIG_SCSI_SAS_ATTRS is not set # CONFIG_SCSI_SAS_LIBSAS is not set # CONFIG_SCSI_SRP_ATTRS is not set # CONFIG_SCSI_LOWLEVEL is not set # CONFIG_SCSI_DH is not set # CONFIG_SCSI_OSD_INITIATOR is not set # CONFIG_ATA is not set # CONFIG_MD is not set CONFIG_HAVE_NET_MACB=y CONFIG_NETDEVICES=y # CONFIG_DUMMY is not set # CONFIG_BONDING is not set # CONFIG_MACVLAN is not set # CONFIG_EQUALIZER is not set # CONFIG_TUN is not set # CONFIG_VETH is not set # CONFIG_MII is not set CONFIG_PHYLIB=y # # MII PHY device drivers # # CONFIG_MARVELL_PHY is not set CONFIG_DAVICOM_PHY=y # CONFIG_QSEMI_PHY is not set # CONFIG_LXT_PHY is not set # CONFIG_CICADA_PHY is not set # CONFIG_VITESSE_PHY is not set # CONFIG_SMSC_PHY is not set # CONFIG_BROADCOM_PHY is not set # CONFIG_BCM63XX_PHY is not set # CONFIG_ICPLUS_PHY is not set # CONFIG_REALTEK_PHY is not set # CONFIG_NATIONAL_PHY is not set # CONFIG_STE10XP is not set # CONFIG_LSI_ET1011C_PHY is not set # CONFIG_MICREL_PHY is not set # CONFIG_FIXED_PHY is not set # CONFIG_MDIO_BITBANG is not set CONFIG_NET_ETHERNET=y CONFIG_MACB=y # CONFIG_AX88796 is not set # CONFIG_SMC91X is not set # CONFIG_DM9000 is not set # CONFIG_ETHOC is not set # CONFIG_SMC911X is not set # CONFIG_SMSC911X is not set # CONFIG_DNET is not set # CONFIG_IBM_NEW_EMAC_ZMII is not set # CONFIG_IBM_NEW_EMAC_RGMII is not set # CONFIG_IBM_NEW_EMAC_TAH is not set # CONFIG_IBM_NEW_EMAC_EMAC4 is not set # CONFIG_IBM_NEW_EMAC_NO_FLOW_CTRL is not set # CONFIG_IBM_NEW_EMAC_MAL_CLR_ICINTSTAT is not set # CONFIG_IBM_NEW_EMAC_MAL_COMMON_ERR is not set # CONFIG_B44 is not set # CONFIG_KS8851_MLL is not set # CONFIG_NETDEV_1000 is not set # CONFIG_NETDEV_10000 is not set # CONFIG_WLAN is not set # # Enable WiMAX (Networking options) to see the WiMAX drivers # # # USB Network Adapters # # CONFIG_USB_CATC is not set # CONFIG_USB_KAWETH is not set # CONFIG_USB_PEGASUS is not set # CONFIG_USB_RTL8150 is not set # CONFIG_USB_USBNET is not set # CONFIG_USB_IPHETH is not set # CONFIG_WAN is not set # # CAIF transport drivers # CONFIG_PPP=y # CONFIG_PPP_MULTILINK is not set # CONFIG_PPP_FILTER is not set CONFIG_PPP_ASYNC=y # CONFIG_PPP_SYNC_TTY is not set CONFIG_PPP_DEFLATE=y CONFIG_PPP_BSDCOMP=y CONFIG_PPP_MPPE=y # CONFIG_PPPOE is not set # CONFIG_SLIP is not set CONFIG_SLHC=y # CONFIG_NETCONSOLE is not set # CONFIG_NETPOLL is not set # CONFIG_NET_POLL_CONTROLLER is not set # CONFIG_ISDN is not set # CONFIG_PHONE is not set # # Input device support # CONFIG_INPUT=y # CONFIG_INPUT_FF_MEMLESS is not set # CONFIG_INPUT_POLLDEV is not set # CONFIG_INPUT_SPARSEKMAP is not set # # Userland interfaces # CONFIG_INPUT_MOUSEDEV=y # CONFIG_INPUT_MOUSEDEV_PSAUX is not set CONFIG_INPUT_MOUSEDEV_SCREEN_X=1024 CONFIG_INPUT_MOUSEDEV_SCREEN_Y=768 # CONFIG_INPUT_JOYDEV is not set # CONFIG_INPUT_EVDEV is not set # CONFIG_INPUT_EVBUG is not set # # Input Device Drivers # # CONFIG_INPUT_KEYBOARD is not set # CONFIG_INPUT_MOUSE is not set # CONFIG_INPUT_JOYSTICK is not set # CONFIG_INPUT_TABLET is not set # CONFIG_INPUT_TOUCHSCREEN is not set # CONFIG_INPUT_MISC is not set # # Hardware I/O ports # # CONFIG_SERIO is not set # CONFIG_GAMEPORT is not set # # Character devices # CONFIG_VT=y CONFIG_CONSOLE_TRANSLATIONS=y CONFIG_VT_CONSOLE=y CONFIG_HW_CONSOLE=y # CONFIG_VT_HW_CONSOLE_BINDING is not set CONFIG_DEVKMEM=y # CONFIG_SERIAL_NONSTANDARD is not set # CONFIG_N_GSM is not set # # Serial drivers # # CONFIG_SERIAL_8250 is not set # # Non-8250 serial port support # CONFIG_SERIAL_ATMEL=y CONFIG_SERIAL_ATMEL_CONSOLE=y CONFIG_SERIAL_ATMEL_PDC=y # CONFIG_SERIAL_ATMEL_TTYAT is not set CONFIG_SERIAL_CORE=y CONFIG_SERIAL_CORE_CONSOLE=y # CONFIG_SERIAL_TIMBERDALE is not set # CONFIG_SERIAL_ALTERA_JTAGUART is not set # CONFIG_SERIAL_ALTERA_UART is not set CONFIG_UNIX98_PTYS=y # CONFIG_DEVPTS_MULTIPLE_INSTANCES is not set # CONFIG_LEGACY_PTYS is not set # CONFIG_IPMI_HANDLER is not set # CONFIG_HW_RANDOM is not set # CONFIG_R3964 is not set # CONFIG_RAW_DRIVER is not set # CONFIG_TCG_TPM is not set # CONFIG_RAMOOPS is not set CONFIG_I2C=y CONFIG_I2C_BOARDINFO=y # CONFIG_I2C_COMPAT is not set CONFIG_I2C_CHARDEV=y # CONFIG_I2C_MUX is not set # CONFIG_I2C_HELPER_AUTO is not set # CONFIG_I2C_SMBUS is not set # # I2C Algorithms # CONFIG_I2C_ALGOBIT=y # CONFIG_I2C_ALGOPCF is not set # CONFIG_I2C_ALGOPCA is not set # # I2C Hardware Bus support # # # I2C system bus drivers (mostly embedded / system-on-chip) # # CONFIG_I2C_DESIGNWARE is not set CONFIG_I2C_GPIO=y # CONFIG_I2C_OCORES is not set # CONFIG_I2C_PCA_PLATFORM is not set # CONFIG_I2C_SIMTEC is not set # CONFIG_I2C_XILINX is not set # # External I2C/SMBus adapter drivers # # CONFIG_I2C_PARPORT_LIGHT is not set # CONFIG_I2C_TAOS_EVM is not set # CONFIG_I2C_TINY_USB is not set # # Other I2C/SMBus bus drivers # # CONFIG_I2C_STUB is not set CONFIG_I2C_DEBUG_CORE=y CONFIG_I2C_DEBUG_ALGO=y CONFIG_I2C_DEBUG_BUS=y # CONFIG_SPI is not set # # PPS support # # CONFIG_PPS is not set CONFIG_ARCH_REQUIRE_GPIOLIB=y CONFIG_GPIOLIB=y CONFIG_DEBUG_GPIO=y CONFIG_GPIO_SYSFS=y # # Memory mapped GPIO expanders: # # CONFIG_GPIO_BASIC_MMIO is not set # CONFIG_GPIO_IT8761E is not set # CONFIG_GPIO_VX855 is not set # # I2C GPIO expanders: # # CONFIG_GPIO_MAX7300 is not set # CONFIG_GPIO_MAX732X is not set # CONFIG_GPIO_PCA953X is not set # CONFIG_GPIO_PCF857X is not set # CONFIG_GPIO_SX150X is not set # CONFIG_GPIO_ADP5588 is not set # # PCI GPIO expanders: # # # SPI GPIO expanders: # # # AC97 GPIO expanders: # # # MODULbus GPIO expanders: # # CONFIG_W1 is not set # CONFIG_POWER_SUPPLY is not set # CONFIG_HWMON is not set # CONFIG_THERMAL is not set CONFIG_WATCHDOG=y CONFIG_WATCHDOG_NOWAYOUT=y # # Watchdog Device Drivers # # CONFIG_SOFT_WATCHDOG is not set CONFIG_AT91SAM9X_WATCHDOG=y # CONFIG_MAX63XX_WATCHDOG is not set # # USB-based Watchdog Cards # # CONFIG_USBPCWATCHDOG is not set CONFIG_SSB_POSSIBLE=y # # Sonics Silicon Backplane # # CONFIG_SSB is not set # CONFIG_MFD_SUPPORT is not set # CONFIG_REGULATOR is not set # CONFIG_MEDIA_SUPPORT is not set # # Graphics support # # CONFIG_DRM is not set # CONFIG_VGASTATE is not set # CONFIG_VIDEO_OUTPUT_CONTROL is not set # CONFIG_FB is not set # CONFIG_BACKLIGHT_LCD_SUPPORT is not set # # Display device support # # CONFIG_DISPLAY_SUPPORT is not set # # Console display driver support # CONFIG_DUMMY_CONSOLE=y CONFIG_SOUND=y # CONFIG_SOUND_OSS_CORE is not set CONFIG_SND=y CONFIG_SND_TIMER=y CONFIG_SND_PCM=y CONFIG_SND_JACK=y # CONFIG_SND_SEQUENCER is not set # CONFIG_SND_MIXER_OSS is not set # CONFIG_SND_PCM_OSS is not set # CONFIG_SND_HRTIMER is not set # CONFIG_SND_DYNAMIC_MINORS is not set # CONFIG_SND_SUPPORT_OLD_API is not set CONFIG_SND_VERBOSE_PROCFS=y CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_DEBUG_VERBOSE=y CONFIG_SND_PCM_XRUN_DEBUG=y # CONFIG_SND_RAWMIDI_SEQ is not set # CONFIG_SND_OPL3_LIB_SEQ is not set # CONFIG_SND_OPL4_LIB_SEQ is not set # CONFIG_SND_SBAWE_SEQ is not set # CONFIG_SND_EMU10K1_SEQ is not set # CONFIG_SND_DRIVERS is not set # CONFIG_SND_ARM is not set # # Atmel devices (AVR32 and AT91) # # CONFIG_SND_ATMEL_AC97C is not set # CONFIG_SND_USB is not set CONFIG_SND_SOC=y CONFIG_SND_ATMEL_SOC=y CONFIG_SND_ATMEL_SOC_SSC=y CONFIG_SND_AT91_SOC_CDU=y CONFIG_SND_SOC_I2C_AND_SPI=y # CONFIG_SND_SOC_ALL_CODECS is not set CONFIG_SND_SOC_TLV320AIC3X=y # CONFIG_SOUND_PRIME is not set # CONFIG_HID_SUPPORT is not set CONFIG_USB_SUPPORT=y CONFIG_USB_ARCH_HAS_HCD=y CONFIG_USB_ARCH_HAS_OHCI=y # CONFIG_USB_ARCH_HAS_EHCI is not set CONFIG_USB=y # CONFIG_USB_DEBUG is not set CONFIG_USB_ANNOUNCE_NEW_DEVICES=y # # Miscellaneous USB options # # CONFIG_USB_DEVICEFS is not set # CONFIG_USB_DEVICE_CLASS is not set # CONFIG_USB_DYNAMIC_MINORS is not set # CONFIG_USB_MON is not set # CONFIG_USB_WUSB is not set # CONFIG_USB_WUSB_CBAF is not set # # USB Host Controller Drivers # # CONFIG_USB_C67X00_HCD is not set # CONFIG_USB_OXU210HP_HCD is not set # CONFIG_USB_ISP116X_HCD is not set # CONFIG_USB_ISP1760_HCD is not set # CONFIG_USB_ISP1362_HCD is not set CONFIG_USB_OHCI_HCD=y # CONFIG_USB_OHCI_BIG_ENDIAN_DESC is not set # CONFIG_USB_OHCI_BIG_ENDIAN_MMIO is not set CONFIG_USB_OHCI_LITTLE_ENDIAN=y # CONFIG_USB_SL811_HCD is not set # CONFIG_USB_R8A66597_HCD is not set # CONFIG_USB_HWA_HCD is not set # CONFIG_USB_MUSB_HDRC is not set # # USB Device Class drivers # # CONFIG_USB_ACM is not set # CONFIG_USB_PRINTER is not set # CONFIG_USB_WDM is not set # CONFIG_USB_TMC is not set # # NOTE: USB_STORAGE depends on SCSI but BLK_DEV_SD may # # # also be needed; see USB_STORAGE Help for more info # CONFIG_USB_STORAGE=y # CONFIG_USB_STORAGE_DEBUG is not set # CONFIG_USB_STORAGE_DATAFAB is not set # CONFIG_USB_STORAGE_FREECOM is not set # CONFIG_USB_STORAGE_ISD200 is not set # CONFIG_USB_STORAGE_USBAT is not set # CONFIG_USB_STORAGE_SDDR09 is not set # CONFIG_USB_STORAGE_SDDR55 is not set # CONFIG_USB_STORAGE_JUMPSHOT is not set # CONFIG_USB_STORAGE_ALAUDA is not set # CONFIG_USB_STORAGE_ONETOUCH is not set # CONFIG_USB_STORAGE_KARMA is not set # CONFIG_USB_STORAGE_CYPRESS_ATACB is not set # CONFIG_USB_UAS is not set # CONFIG_USB_LIBUSUAL is not set # # USB Imaging devices # # CONFIG_USB_MDC800 is not set # CONFIG_USB_MICROTEK is not set # # USB port drivers # # CONFIG_USB_SERIAL is not set # # USB Miscellaneous drivers # # CONFIG_USB_EMI62 is not set # CONFIG_USB_EMI26 is not set # CONFIG_USB_ADUTUX is not set # CONFIG_USB_SEVSEG is not set # CONFIG_USB_RIO500 is not set # CONFIG_USB_LEGOTOWER is not set # CONFIG_USB_LCD is not set # CONFIG_USB_LED is not set # CONFIG_USB_CYPRESS_CY7C63 is not set # CONFIG_USB_CYTHERM is not set # CONFIG_USB_IDMOUSE is not set # CONFIG_USB_FTDI_ELAN is not set # CONFIG_USB_APPLEDISPLAY is not set # CONFIG_USB_LD is not set # CONFIG_USB_TRANCEVIBRATOR is not set # CONFIG_USB_IOWARRIOR is not set # CONFIG_USB_TEST is not set # CONFIG_USB_ISIGHTFW is not set # CONFIG_USB_YUREX is not set # CONFIG_USB_GADGET is not set # # OTG and related infrastructure # # CONFIG_USB_GPIO_VBUS is not set # CONFIG_USB_ULPI is not set # CONFIG_NOP_USB_XCEIV is not set CONFIG_MMC=y # CONFIG_MMC_DEBUG is not set CONFIG_MMC_UNSAFE_RESUME=y # # MMC/SD/SDIO Card Drivers # CONFIG_MMC_BLOCK=y CONFIG_MMC_BLOCK_MINORS=8 CONFIG_MMC_BLOCK_BOUNCE=y # CONFIG_SDIO_UART is not set # CONFIG_MMC_TEST is not set # # MMC/SD/SDIO Host Controller Drivers # # CONFIG_MMC_SDHCI is not set # CONFIG_MMC_AT91 is not set CONFIG_MMC_ATMELMCI=y # CONFIG_MMC_USHC is not set # CONFIG_MEMSTICK is not set CONFIG_NEW_LEDS=y CONFIG_LEDS_CLASS=y # # LED drivers # # CONFIG_LEDS_PCA9532 is not set CONFIG_LEDS_GPIO=y CONFIG_LEDS_GPIO_PLATFORM=y # CONFIG_LEDS_LP3944 is not set # CONFIG_LEDS_LP5521 is not set # CONFIG_LEDS_LP5523 is not set # CONFIG_LEDS_PCA955X is not set # CONFIG_LEDS_BD2802 is not set # CONFIG_LEDS_LT3593 is not set CONFIG_LEDS_TRIGGERS=y # # LED Triggers # # CONFIG_LEDS_TRIGGER_TIMER is not set CONFIG_LEDS_TRIGGER_HEARTBEAT=y # CONFIG_LEDS_TRIGGER_BACKLIGHT is not set # CONFIG_LEDS_TRIGGER_GPIO is not set # CONFIG_LEDS_TRIGGER_DEFAULT_ON is not set # # iptables trigger is under Netfilter config (LED target) # # CONFIG_ACCESSIBILITY is not set CONFIG_RTC_LIB=y CONFIG_RTC_CLASS=y CONFIG_RTC_HCTOSYS=y CONFIG_RTC_HCTOSYS_DEVICE="rtc0" # CONFIG_RTC_DEBUG is not set # # RTC interfaces # CONFIG_RTC_INTF_SYSFS=y CONFIG_RTC_INTF_PROC=y CONFIG_RTC_INTF_DEV=y # CONFIG_RTC_INTF_DEV_UIE_EMUL is not set # CONFIG_RTC_DRV_TEST is not set # # I2C RTC drivers # # CONFIG_RTC_DRV_DS1307 is not set # CONFIG_RTC_DRV_DS1374 is not set # CONFIG_RTC_DRV_DS1672 is not set # CONFIG_RTC_DRV_DS3232 is not set # CONFIG_RTC_DRV_MAX6900 is not set # CONFIG_RTC_DRV_RS5C372 is not set # CONFIG_RTC_DRV_ISL1208 is not set # CONFIG_RTC_DRV_ISL12022 is not set # CONFIG_RTC_DRV_X1205 is not set # CONFIG_RTC_DRV_PCF8563 is not set # CONFIG_RTC_DRV_PCF8583 is not set # CONFIG_RTC_DRV_M41T80 is not set # CONFIG_RTC_DRV_BQ32K is not set # CONFIG_RTC_DRV_S35390A is not set # CONFIG_RTC_DRV_FM3130 is not set # CONFIG_RTC_DRV_RX8581 is not set # CONFIG_RTC_DRV_RX8025 is not set # # SPI RTC drivers # # # Platform RTC drivers # # CONFIG_RTC_DRV_CMOS is not set # CONFIG_RTC_DRV_DS1286 is not set # CONFIG_RTC_DRV_DS1511 is not set # CONFIG_RTC_DRV_DS1553 is not set # CONFIG_RTC_DRV_DS1742 is not set # CONFIG_RTC_DRV_STK17TA8 is not set # CONFIG_RTC_DRV_M48T86 is not set # CONFIG_RTC_DRV_M48T35 is not set # CONFIG_RTC_DRV_M48T59 is not set # CONFIG_RTC_DRV_MSM6242 is not set # CONFIG_RTC_DRV_BQ4802 is not set # CONFIG_RTC_DRV_RP5C01 is not set # CONFIG_RTC_DRV_V3020 is not set # # on-CPU RTC drivers # CONFIG_RTC_DRV_AT91SAM9=y CONFIG_RTC_DRV_AT91SAM9_RTT=0 CONFIG_RTC_DRV_AT91SAM9_GPBR=0 # CONFIG_DMADEVICES is not set # CONFIG_AUXDISPLAY is not set # CONFIG_UIO is not set # CONFIG_STAGING is not set # # File systems # # CONFIG_EXT2_FS is not set # CONFIG_EXT3_FS is not set CONFIG_EXT4_FS=y CONFIG_EXT4_USE_FOR_EXT23=y CONFIG_EXT4_FS_XATTR=y # CONFIG_EXT4_FS_POSIX_ACL is not set # CONFIG_EXT4_FS_SECURITY is not set # CONFIG_EXT4_DEBUG is not set CONFIG_JBD2=y # CONFIG_JBD2_DEBUG is not set CONFIG_FS_MBCACHE=y # CONFIG_REISERFS_FS is not set # CONFIG_JFS_FS is not set # CONFIG_FS_POSIX_ACL is not set # CONFIG_XFS_FS is not set # CONFIG_GFS2_FS is not set # CONFIG_OCFS2_FS is not set # CONFIG_BTRFS_FS is not set # CONFIG_NILFS2_FS is not set CONFIG_FILE_LOCKING=y # CONFIG_FSNOTIFY is not set # CONFIG_DNOTIFY is not set # CONFIG_INOTIFY_USER is not set # CONFIG_FANOTIFY is not set # CONFIG_QUOTA is not set # CONFIG_QUOTACTL is not set # CONFIG_AUTOFS4_FS is not set # CONFIG_FUSE_FS is not set # # Caches # # CONFIG_FSCACHE is not set # # CD-ROM/DVD Filesystems # # CONFIG_ISO9660_FS is not set # CONFIG_UDF_FS is not set # # DOS/FAT/NT Filesystems # CONFIG_FAT_FS=y # CONFIG_MSDOS_FS is not set CONFIG_VFAT_FS=y CONFIG_FAT_DEFAULT_CODEPAGE=437 CONFIG_FAT_DEFAULT_IOCHARSET="iso8859-1" # CONFIG_NTFS_FS is not set # # Pseudo filesystems # CONFIG_PROC_FS=y CONFIG_PROC_SYSCTL=y CONFIG_PROC_PAGE_MONITOR=y CONFIG_SYSFS=y # CONFIG_TMPFS is not set # CONFIG_HUGETLB_PAGE is not set # CONFIG_CONFIGFS_FS is not set CONFIG_MISC_FILESYSTEMS=y # CONFIG_ADFS_FS is not set # CONFIG_AFFS_FS is not set # CONFIG_HFS_FS is not set # CONFIG_HFSPLUS_FS is not set # CONFIG_BEFS_FS is not set # CONFIG_BFS_FS is not set # CONFIG_EFS_FS is not set CONFIG_JFFS2_FS=y CONFIG_JFFS2_FS_DEBUG=0 CONFIG_JFFS2_FS_WRITEBUFFER=y CONFIG_JFFS2_FS_WBUF_VERIFY=y # CONFIG_JFFS2_SUMMARY is not set # CONFIG_JFFS2_FS_XATTR is not set CONFIG_JFFS2_COMPRESSION_OPTIONS=y CONFIG_JFFS2_ZLIB=y CONFIG_JFFS2_LZO=y CONFIG_JFFS2_RTIME=y CONFIG_JFFS2_RUBIN=y # CONFIG_JFFS2_CMODE_NONE is not set CONFIG_JFFS2_CMODE_PRIORITY=y # CONFIG_JFFS2_CMODE_SIZE is not set # CONFIG_JFFS2_CMODE_FAVOURLZO is not set # CONFIG_LOGFS is not set # CONFIG_CRAMFS is not set # CONFIG_SQUASHFS is not set # CONFIG_VXFS_FS is not set # CONFIG_MINIX_FS is not set # CONFIG_OMFS_FS is not set # CONFIG_HPFS_FS is not set # CONFIG_QNX4FS_FS is not set # CONFIG_ROMFS_FS is not set # CONFIG_SYSV_FS is not set # CONFIG_UFS_FS is not set # CONFIG_NETWORK_FILESYSTEMS is not set # # Partition Types # # CONFIG_PARTITION_ADVANCED is not set CONFIG_MSDOS_PARTITION=y CONFIG_NLS=y CONFIG_NLS_DEFAULT="iso8859-1" CONFIG_NLS_CODEPAGE_437=y # CONFIG_NLS_CODEPAGE_737 is not set # CONFIG_NLS_CODEPAGE_775 is not set CONFIG_NLS_CODEPAGE_850=y CONFIG_NLS_CODEPAGE_852=y # CONFIG_NLS_CODEPAGE_855 is not set # CONFIG_NLS_CODEPAGE_857 is not set # CONFIG_NLS_CODEPAGE_860 is not set # CONFIG_NLS_CODEPAGE_861 is not set # CONFIG_NLS_CODEPAGE_862 is not set # CONFIG_NLS_CODEPAGE_863 is not set # CONFIG_NLS_CODEPAGE_864 is not set # CONFIG_NLS_CODEPAGE_865 is not set # CONFIG_NLS_CODEPAGE_866 is not set # CONFIG_NLS_CODEPAGE_869 is not set # CONFIG_NLS_CODEPAGE_936 is not set # CONFIG_NLS_CODEPAGE_950 is not set # CONFIG_NLS_CODEPAGE_932 is not set # CONFIG_NLS_CODEPAGE_949 is not set # CONFIG_NLS_CODEPAGE_874 is not set # CONFIG_NLS_ISO8859_8 is not set # CONFIG_NLS_CODEPAGE_1250 is not set # CONFIG_NLS_CODEPAGE_1251 is not set # CONFIG_NLS_ASCII is not set CONFIG_NLS_ISO8859_1=y CONFIG_NLS_ISO8859_2=y # CONFIG_NLS_ISO8859_3 is not set # CONFIG_NLS_ISO8859_4 is not set # CONFIG_NLS_ISO8859_5 is not set # CONFIG_NLS_ISO8859_6 is not set # CONFIG_NLS_ISO8859_7 is not set # CONFIG_NLS_ISO8859_9 is not set # CONFIG_NLS_ISO8859_13 is not set # CONFIG_NLS_ISO8859_14 is not set # CONFIG_NLS_ISO8859_15 is not set # CONFIG_NLS_KOI8_R is not set # CONFIG_NLS_KOI8_U is not set # CONFIG_NLS_UTF8 is not set # CONFIG_DLM is not set # # Kernel hacking # CONFIG_PRINTK_TIME=y CONFIG_ENABLE_WARN_DEPRECATED=y CONFIG_ENABLE_MUST_CHECK=y CONFIG_FRAME_WARN=1024 # CONFIG_MAGIC_SYSRQ is not set # CONFIG_STRIP_ASM_SYMS is not set # CONFIG_UNUSED_SYMBOLS is not set CONFIG_DEBUG_FS=y # CONFIG_HEADERS_CHECK is not set CONFIG_DEBUG_KERNEL=y # CONFIG_DEBUG_SHIRQ is not set # CONFIG_LOCKUP_DETECTOR is not set # CONFIG_HARDLOCKUP_DETECTOR is not set # CONFIG_DETECT_HUNG_TASK is not set CONFIG_SCHED_DEBUG=y # CONFIG_SCHEDSTATS is not set # CONFIG_TIMER_STATS is not set # CONFIG_DEBUG_OBJECTS is not set # CONFIG_DEBUG_SLAB is not set # CONFIG_DEBUG_KMEMLEAK is not set CONFIG_DEBUG_PREEMPT=y # CONFIG_DEBUG_RT_MUTEXES is not set # CONFIG_RT_MUTEX_TESTER is not set # CONFIG_DEBUG_SPINLOCK is not set # CONFIG_DEBUG_MUTEXES is not set CONFIG_BKL=y # CONFIG_DEBUG_LOCK_ALLOC is not set # CONFIG_PROVE_LOCKING is not set # CONFIG_SPARSE_RCU_POINTER is not set # CONFIG_LOCK_STAT is not set # CONFIG_DEBUG_SPINLOCK_SLEEP is not set # CONFIG_DEBUG_LOCKING_API_SELFTESTS is not set # CONFIG_DEBUG_KOBJECT is not set CONFIG_DEBUG_BUGVERBOSE=y CONFIG_DEBUG_INFO=y # CONFIG_DEBUG_INFO_REDUCED is not set # CONFIG_DEBUG_VM is not set # CONFIG_DEBUG_WRITECOUNT is not set CONFIG_DEBUG_MEMORY_INIT=y # CONFIG_DEBUG_LIST is not set # CONFIG_TEST_LIST_SORT is not set # CONFIG_DEBUG_SG is not set # CONFIG_DEBUG_NOTIFIERS is not set # CONFIG_DEBUG_CREDENTIALS is not set # CONFIG_BOOT_PRINTK_DELAY is not set # CONFIG_RCU_TORTURE_TEST is not set CONFIG_RCU_CPU_STALL_DETECTOR=y CONFIG_RCU_CPU_STALL_TIMEOUT=60 CONFIG_RCU_CPU_STALL_DETECTOR_RUNNABLE=y CONFIG_RCU_CPU_STALL_VERBOSE=y # CONFIG_BACKTRACE_SELF_TEST is not set # CONFIG_DEBUG_BLOCK_EXT_DEVT is not set # CONFIG_DEBUG_FORCE_WEAK_PER_CPU is not set # CONFIG_LKDTM is not set # CONFIG_FAULT_INJECTION is not set # CONFIG_LATENCYTOP is not set CONFIG_SYSCTL_SYSCALL_CHECK=y # CONFIG_PAGE_POISONING is not set CONFIG_HAVE_FUNCTION_TRACER=y CONFIG_HAVE_DYNAMIC_FTRACE=y CONFIG_HAVE_FTRACE_MCOUNT_RECORD=y CONFIG_TRACING_SUPPORT=y # CONFIG_FTRACE is not set # CONFIG_DYNAMIC_DEBUG is not set # CONFIG_ATOMIC64_SELFTEST is not set # CONFIG_SAMPLES is not set CONFIG_HAVE_ARCH_KGDB=y # CONFIG_KGDB is not set # CONFIG_STRICT_DEVMEM is not set CONFIG_ARM_UNWIND=y CONFIG_DEBUG_USER=y # CONFIG_DEBUG_ERRORS is not set # CONFIG_DEBUG_STACK_USAGE is not set CONFIG_DEBUG_LL=y # CONFIG_EARLY_PRINTK is not set # CONFIG_DEBUG_ICEDCC is not set # CONFIG_OC_ETM is not set # # Security options # # CONFIG_KEYS is not set # CONFIG_SECURITY_DMESG_RESTRICT is not set # CONFIG_SECURITY is not set # CONFIG_SECURITYFS is not set CONFIG_DEFAULT_SECURITY_DAC=y CONFIG_DEFAULT_SECURITY="" CONFIG_CRYPTO=y # # Crypto core or helper # CONFIG_CRYPTO_ALGAPI=y CONFIG_CRYPTO_ALGAPI2=y CONFIG_CRYPTO_AEAD2=y CONFIG_CRYPTO_BLKCIPHER=y CONFIG_CRYPTO_BLKCIPHER2=y CONFIG_CRYPTO_HASH=y CONFIG_CRYPTO_HASH2=y CONFIG_CRYPTO_RNG=y CONFIG_CRYPTO_RNG2=y CONFIG_CRYPTO_PCOMP2=y CONFIG_CRYPTO_MANAGER=y CONFIG_CRYPTO_MANAGER2=y CONFIG_CRYPTO_MANAGER_DISABLE_TESTS=y # CONFIG_CRYPTO_GF128MUL is not set # CONFIG_CRYPTO_NULL is not set CONFIG_CRYPTO_WORKQUEUE=y # CONFIG_CRYPTO_CRYPTD is not set # CONFIG_CRYPTO_AUTHENC is not set # CONFIG_CRYPTO_TEST is not set # # Authenticated Encryption with Associated Data # # CONFIG_CRYPTO_CCM is not set # CONFIG_CRYPTO_GCM is not set # CONFIG_CRYPTO_SEQIV is not set # # Block modes # # CONFIG_CRYPTO_CBC is not set # CONFIG_CRYPTO_CTR is not set # CONFIG_CRYPTO_CTS is not set CONFIG_CRYPTO_ECB=y # CONFIG_CRYPTO_LRW is not set # CONFIG_CRYPTO_PCBC is not set # CONFIG_CRYPTO_XTS is not set # # Hash modes # # CONFIG_CRYPTO_HMAC is not set # CONFIG_CRYPTO_XCBC is not set # CONFIG_CRYPTO_VMAC is not set # # Digest # # CONFIG_CRYPTO_CRC32C is not set # CONFIG_CRYPTO_GHASH is not set # CONFIG_CRYPTO_MD4 is not set # CONFIG_CRYPTO_MD5 is not set # CONFIG_CRYPTO_MICHAEL_MIC is not set # CONFIG_CRYPTO_RMD128 is not set # CONFIG_CRYPTO_RMD160 is not set # CONFIG_CRYPTO_RMD256 is not set # CONFIG_CRYPTO_RMD320 is not set CONFIG_CRYPTO_SHA1=y # CONFIG_CRYPTO_SHA256 is not set # CONFIG_CRYPTO_SHA512 is not set # CONFIG_CRYPTO_TGR192 is not set # CONFIG_CRYPTO_WP512 is not set # # Ciphers # CONFIG_CRYPTO_AES=y # CONFIG_CRYPTO_ANUBIS is not set CONFIG_CRYPTO_ARC4=y # CONFIG_CRYPTO_BLOWFISH is not set # CONFIG_CRYPTO_CAMELLIA is not set # CONFIG_CRYPTO_CAST5 is not set # CONFIG_CRYPTO_CAST6 is not set # CONFIG_CRYPTO_DES is not set # CONFIG_CRYPTO_FCRYPT is not set # CONFIG_CRYPTO_KHAZAD is not set # CONFIG_CRYPTO_SALSA20 is not set # CONFIG_CRYPTO_SEED is not set # CONFIG_CRYPTO_SERPENT is not set # CONFIG_CRYPTO_TEA is not set # CONFIG_CRYPTO_TWOFISH is not set # # Compression # # CONFIG_CRYPTO_DEFLATE is not set # CONFIG_CRYPTO_ZLIB is not set # CONFIG_CRYPTO_LZO is not set # # Random Number Generation # CONFIG_CRYPTO_ANSI_CPRNG=y CONFIG_CRYPTO_HW=y # CONFIG_BINARY_PRINTF is not set # # Library routines # CONFIG_BITREVERSE=y CONFIG_GENERIC_FIND_LAST_BIT=y CONFIG_CRC_CCITT=y CONFIG_CRC16=y CONFIG_CRC_T10DIF=y # CONFIG_CRC_ITU_T is not set CONFIG_CRC32=y # CONFIG_CRC7 is not set # CONFIG_LIBCRC32C is not set CONFIG_ZLIB_INFLATE=y CONFIG_ZLIB_DEFLATE=y CONFIG_LZO_COMPRESS=y CONFIG_LZO_DECOMPRESS=y CONFIG_HAS_IOMEM=y CONFIG_HAS_IOPORT=y CONFIG_HAS_DMA=y CONFIG_NLATTR=y CONFIG_GENERIC_ATOMIC64=y [5.] Most recent kernel version which did not have the bug: Linux version 2.6.35.11-cdu (prchal@prchal) (gcc version 4.4.5 (GCC) ) #50 PREEMPT Thu Mar 10 16:47:25 CET 2011 [6.] Output of Oops.. message (if applicable) with symbolic information resolved (see Documentation/oops-tracing.txt) dmesg: Linux version 2.6.37.3-cdu (prchal@prchal) (gcc version 4.4.5 (GCC) ) #17 PREEMPT Thu Mar 10 15:06:24 CET 2011 CPU: ARM926EJ-S [41069265] revision 5 (ARMv5TEJ), cr=00053177 CPU: VIVT data cache, VIVT instruction cache Machine: CDU Memory policy: ECC disabled, Data cache writeback Clocks: PLLA overclocked, 210 MHz Clocks: CPU 210 MHz, master 105 MHz, main 6.000 MHz On node 0 totalpages: 16384 free_area_init_node: node 0, pgdat c036983c, node_mem_map c037b000 Normal zone: 128 pages used for memmap Normal zone: 0 pages reserved Normal zone: 16256 pages, LIFO batch:3 pcpu-alloc: s0 r0 d32768 u32768 alloc=1*32768 pcpu-alloc: [0] 0 Built 1 zonelists in Zone order, mobility grouping on. Total pages: 16256 Kernel command line: console=ttyS0,115200 root=/dev/mtdblock4 mtdparts=atmel_nand:256k(bootstrap)ro,512k(uboot)ro,256k(env)ro,3M(linux),-(root) rw rootfstype=jffs2 PID hash table entries: 256 (order: -2, 1024 bytes) Dentry cache hash table entries: 8192 (order: 3, 32768 bytes) Inode-cache hash table entries: 4096 (order: 2, 16384 bytes) Memory: 64MB = 64MB total Memory: 61364k/61364k available, 4172k reserved, 0K highmem Virtual kernel memory layout: vector : 0xffff0000 - 0xffff1000 ( 4 kB) fixmap : 0xfff00000 - 0xfffe0000 ( 896 kB) DMA : 0xffc00000 - 0xffe00000 ( 2 MB) vmalloc : 0xc4800000 - 0xfee00000 ( 934 MB) lowmem : 0xc0000000 - 0xc4000000 ( 64 MB) modules : 0xbf000000 - 0xc0000000 ( 16 MB) .init : 0xc0008000 - 0xc0021000 ( 100 kB) .text : 0xc0021000 - 0xc034812c (3229 kB) .data : 0xc034a000 - 0xc0369ee0 ( 128 kB) Preemptable hierarchical RCU implementation. NR_IRQS:192 AT91: 96 gpio irqs in 3 banks Console: colour dummy device 80x30 console [ttyS0] enabled Calibrating delay loop... 104.44 BogoMIPS (lpj=522240) pid_max: default: 32768 minimum: 301 Mount-cache hash table entries: 512 CPU: Testing write buffer coherency: ok NET: Registered protocol family 16 i2c-core: driver [dummy] registered gpio_request: gpio-128 (busadrp) status -22 gpio_export_name: gpio128 status -22 gpio_request: gpio-130 (busadr5) status -22 gpio_export_name: gpio130 status -22 gpio_request: gpio-131 (busadr4) status -22 gpio_export_name: gpio131 status -22 gpio_request: gpio-132 (busadr3) status -22 gpio_export_name: gpio132 status -22 gpio_request: gpio-133 (busadr2) status -22 gpio_export_name: gpio133 status -22 gpio_request: gpio-134 (busadr1) status -22 gpio_export_name: gpio134 status -22 gpio_request: gpio-135 (busadr0) status -22 gpio_export_name: gpio135 status -22 bio: create slab <bio-0> at 0 SCSI subsystem initialized usbcore: registered new interface driver usbfs usbcore: registered new interface driver hub usbcore: registered new device driver usb i2c i2c-0: adapter [i2c-gpio-1] registered i2c 0-001a: uevent i2c i2c-0: client [tlv320aic3x] registered with bus id 0-001a i2c-gpio i2c-gpio: using pins 55 (SDA) and 56 (SCL) Advanced Linux Sound Architecture Driver Version 1.0.23. Switching to clocksource pit NET: Registered protocol family 2 IP route cache hash table entries: 1024 (order: 0, 4096 bytes) TCP established hash table entries: 2048 (order: 2, 16384 bytes) TCP bind hash table entries: 2048 (order: 1, 8192 bytes) TCP: Hash tables configured (established 2048 bind 2048) TCP reno registered UDP hash table entries: 256 (order: 0, 4096 bytes) UDP-Lite hash table entries: 256 (order: 0, 4096 bytes) NET: Registered protocol family 1 JFFS2 version 2.2. (NAND) ÂŠ 2001-2006 Red Hat, Inc. msgmni has been set to 119 io scheduler noop registered (default) atmel_usart.0: ttyS0 at MMIO 0xfefff200 (irq = 1) is a ATMEL_SERIAL atmel_usart.1: ttyS1 at MMIO 0xfffb0000 (irq = 6) is a ATMEL_SERIAL atmel_usart.2: ttyS2 at MMIO 0xfffb4000 (irq = 7) is a ATMEL_SERIAL atmel_usart.3: ttyS3 at MMIO 0xfffb8000 (irq = 8) is a ATMEL_SERIAL atmel_usart.4: ttyS4 at MMIO 0xfffd0000 (irq = 23) is a ATMEL_SERIAL ssc ssc.0: Atmel SSC device at 0xc4890000 (irq 14) ONFI flash detected ONFI param page 0 valid NAND device: Manufacturer ID: 0x2c, Chip ID: 0xa1 (Micron NAND 128MiB 1,8V 8-bit) Scanning device for bad blocks 5 cmdlinepart partitions found on MTD device atmel_nand Creating 5 MTD partitions on "atmel_nand": 0x000000000000-0x000000040000 : "bootstrap" mtd: Giving out device 0 to bootstrap 0x000000040000-0x0000000c0000 : "uboot" mtd: Giving out device 1 to uboot 0x0000000c0000-0x000000100000 : "env" mtd: Giving out device 2 to env 0x000000100000-0x000000400000 : "linux" mtd: Giving out device 3 to linux 0x000000400000-0x000008000000 : "root" mtd: Giving out device 4 to root PPP generic driver version 2.4.2 PPP Deflate Compression module registered PPP BSD Compression module registered PPP MPPE Compression module registered MACB_mii_bus: probed eth0: Atmel MACB at 0xfffc4000 irq 21 (02:00:de:ad:be:ef) eth0: attached PHY driver [Generic PHY] (mii_bus:phy_addr=ffffffff:00, irq=-1) ohci_hcd: USB 1.1 'Open' Host Controller (OHCI) Driver at91_ohci at91_ohci: AT91 OHCI at91_ohci at91_ohci: new USB bus registered, assigned bus number 1 at91_ohci at91_ohci: irq 20, io mem 0x00500000 usb usb1: New USB device found, idVendor=1d6b, idProduct=0001 usb usb1: New USB device strings: Mfr=3, Product=2, SerialNumber=1 usb usb1: Product: AT91 OHCI usb usb1: Manufacturer: Linux 2.6.37.3-cdu ohci_hcd usb usb1: SerialNumber: at91 hub 1-0:1.0: USB hub found hub 1-0:1.0: 2 ports detected Initializing USB Mass Storage driver... usbcore: registered new interface driver usb-storage USB Mass Storage support registered. mice: PS/2 mouse device common for all mice rtc-at91sam9 at91_rtt.0: rtc core: registered at91_rtt as rtc0 i2c /dev entries driver i2c-core: driver [dev_driver] registered i2c-dev: adapter [i2c-gpio-1] registered as minor 0 AT91SAM9 Watchdog enabled (heartbeat=15 sec, nowayout=1) Registered led device: red Registered led device: green Registered led device: yellow Registered led device: blue tlv320aic3x-codec 0-001a: probe i2c-core: driver [tlv320aic3x-codec] registered ALSA device list: No soundcards found. TCP cubic registered NET: Registered protocol family 17 VFP support v0.3: not present rtc-at91sam9 at91_rtt.0: setting system clock to 2011-03-10 14:19:41 UTC (1299766781) gpio_request: gpio-0 (mmc_detect) status -22 gpio_request: gpio-0 (mmc_wp) status -22 atmel_mci atmel_mci: Atmel MCI controller at 0xfffa8000 irq 9, 1 slots MTDSB: lookup_bdev() returned 0 MTDSB: New superblock for device 4 ("root") mmc0: host does not support reading read-only switch. assuming write-enable. mmc0: new SDHC card at address 1234 mmcblk0: mmc0:1234 SA04G 3.67 GiB mmcblk0: p1 Empty flash at 0x02446464 ends at 0x02446800 Empty flash at 0x0244b130 ends at 0x0244b800 Empty flash at 0x029c2fb4 ends at 0x029c3000 Empty flash at 0x029c86f8 ends at 0x029c8800 Empty flash at 0x029c9ae0 ends at 0x029ca000 Empty flash at 0x029d3d54 ends at 0x029d4000 Empty flash at 0x029d933c ends at 0x029d9800 Empty flash at 0x029de144 ends at 0x029de800 Empty flash at 0x069a8020 ends at 0x069a8800 Empty flash at 0x069afdc0 ends at 0x069b0000 jffs2_scan_dirent_node(): Node CRC failed on node at 0x069b6ff0: Read 0xffffffff, calculated 0x6055a656 Empty flash at 0x069b7020 ends at 0x069b7800 Empty flash at 0x069bdf40 ends at 0x069be000 Empty flash at 0x06b84fa8 ends at 0x06b85000 VFS: Mounted root (jffs2 filesystem) on device 31:4. Freeing init memory: 100K JFFS2 notice: (1) check_node_data: wrong data CRC in data node at 0x029c9060: read 0x5bfb4489, calculated 0x299d675. EXT4-fs (mmcblk0p1): recovery complete EXT4-fs (mmcblk0p1): mounted filesystem with ordered data mode. Opts: (null) eth0: link up (100/Full) [8.] Environment [8.1.] Software (add the output of the ver_linux script here) crosscompiling [8.2.] Processor information (from /proc/cpuinfo): Processor : ARM926EJ-S rev 5 (v5l) BogoMIPS : 104.44 Features : swp half thumb fastmult edsp java CPU implementer : 0x41 CPU architecture: 5TEJ CPU variant : 0x0 CPU part : 0x926 CPU revision : 5 Hardware : CDU Revision : 0000 Serial : 0000000000000000 [8.3.] Module information (from /proc/modules): all in kernel [8.4.] Loaded driver and hardware information (/proc/ioports, /proc/iomem) 00500000-005fffff : at91_ohci 00500000-005fffff : ohci_hcd 20000000-23ffffff : System RAM 20021000-203491ab : Kernel text 2034a000-2037ac7f : Kernel data 40000000-4fffffff : atmel_nand fefff200-fefff3ff : atmel_usart.0 fefff200-fefff3ff : atmel_serial fffa8000-fffabfff : atmel_mci fffb0000-fffb3fff : atmel_usart.1 fffb0000-fffb3fff : atmel_serial fffb4000-fffb7fff : atmel_usart.2 fffb4000-fffb7fff : atmel_serial fffb8000-fffbbfff : atmel_usart.3 fffb8000-fffbbfff : atmel_serial fffbc000-fffbffff : ssc.0 fffc4000-fffc7fff : macb fffd0000-fffd3fff : atmel_usart.4 fffd0000-fffd3fff : atmel_serial ffffe800-ffffe9ff : atmel_nand fffffd20-fffffd2f : at91_rtt.0 [8.7.] Other information that might be relevant to the problem (please look in /proc and include all information that you think to be relevant): dmesg of 2.6.35.11: Linux version 2.6.35.11-cdu (prchal@prchal) (gcc version 4.4.5 (GCC) ) #48 PREEMPT Thu Mar 10 15:13:00 CET 2011 CPU: ARM926EJ-S [41069265] revision 5 (ARMv5TEJ), cr=00053177 CPU: VIVT data cache, VIVT instruction cache Machine: CDU Memory policy: ECC disabled, Data cache writeback On node 0 totalpages: 16384 free_area_init_node: node 0, pgdat c0371cfc, node_mem_map c0390000 Normal zone: 128 pages used for memmap Normal zone: 0 pages reserved Normal zone: 16256 pages, LIFO batch:3 Clocks: PLLA overclocked, 210 MHz Clocks: CPU 210 MHz, master 105 MHz, main 6.000 MHz Built 1 zonelists in Zone order, mobility grouping on. Total pages: 16256 Kernel command line: console=ttyS0,115200 root=/dev/mtdblock4 mtdparts=atmel_nand:256k(bootstrap)ro,512k(uboot)ro,256k(env)ro,3M(linux),-(root) rw rootfstype=jffs2 PID hash table entries: 256 (order: -2, 1024 bytes) Dentry cache hash table entries: 8192 (order: 3, 32768 bytes) Inode-cache hash table entries: 4096 (order: 2, 16384 bytes) Memory: 64MB = 64MB total Memory: 61320k/61320k available, 4216k reserved, 0K highmem Virtual kernel memory layout: vector : 0xffff0000 - 0xffff1000 ( 4 kB) fixmap : 0xfff00000 - 0xfffe0000 ( 896 kB) DMA : 0xffc00000 - 0xffe00000 ( 2 MB) vmalloc : 0xc4800000 - 0xfee00000 ( 934 MB) lowmem : 0xc0000000 - 0xc4000000 ( 64 MB) modules : 0xbf000000 - 0xc0000000 ( 16 MB) .init : 0xc0008000 - 0xc001f000 ( 92 kB) .text : 0xc001f000 - 0xc0338000 (3172 kB) .data : 0xc034e000 - 0xc0372320 ( 145 kB) Hierarchical RCU implementation. Verbose stalled-CPUs detection is disabled. NR_IRQS:192 AT91: 96 gpio irqs in 3 banks Console: colour dummy device 80x30 console [ttyS0] enabled Calibrating delay loop... 104.44 BogoMIPS (lpj=522240) pid_max: default: 32768 minimum: 301 Mount-cache hash table entries: 512 CPU: Testing write buffer coherency: ok devtmpfs: initialized NET: Registered protocol family 16 i2c-core: driver [dummy] registered gpio_request: gpio-128 (busadrp) status -22 gpio_request: gpio-130 (busadr5) status -22 gpio_request: gpio-131 (busadr4) status -22 gpio_request: gpio-132 (busadr3) status -22 gpio_request: gpio-133 (busadr2) status -22 gpio_request: gpio-134 (busadr1) status -22 gpio_request: gpio-135 (busadr0) status -22 bio: create slab <bio-0> at 0 SCSI subsystem initialized usbcore: registered new interface driver usbfs usbcore: registered new interface driver hub usbcore: registered new device driver usb i2c i2c-0: adapter [i2c-gpio-1] registered i2c 0-001a: uevent i2c i2c-0: client [tlv320aic3x] registered with bus id 0-001a i2c-gpio i2c-gpio: using pins 55 (SDA) and 56 (SCL) Advanced Linux Sound Architecture Driver Version 1.0.23. Switching to clocksource pit NET: Registered protocol family 2 IP route cache hash table entries: 1024 (order: 0, 4096 bytes) TCP established hash table entries: 2048 (order: 2, 16384 bytes) TCP bind hash table entries: 2048 (order: 1, 8192 bytes) TCP: Hash tables configured (established 2048 bind 2048) TCP reno registered UDP hash table entries: 256 (order: 0, 4096 bytes) UDP-Lite hash table entries: 256 (order: 0, 4096 bytes) NET: Registered protocol family 1 JFFS2 version 2.2. (NAND) ÂŠ 2001-2006 Red Hat, Inc. msgmni has been set to 119 alg: No test for stdrng (krng) io scheduler noop registered (default) atmel_usart.0: ttyS0 at MMIO 0xfefff200 (irq = 1) is a ATMEL_SERIAL atmel_usart.1: ttyS1 at MMIO 0xfffb0000 (irq = 6) is a ATMEL_SERIAL atmel_usart.2: ttyS2 at MMIO 0xfffb4000 (irq = 7) is a ATMEL_SERIAL atmel_usart.3: ttyS3 at MMIO 0xfffb8000 (irq = 8) is a ATMEL_SERIAL atmel_usart.4: ttyS4 at MMIO 0xfffd0000 (irq = 23) is a ATMEL_SERIAL loop: module loaded mGine mflash driver, (c) 2008 mGine Co. ssc ssc.0: Atmel SSC device at 0xc4890000 (irq 14) NAND device: Manufacturer ID: 0x2c, Chip ID: 0xa1 (Micron NAND 128MiB 1,8V 8-bit) Scanning device for bad blocks 5 cmdlinepart partitions found on MTD device atmel_nand Creating 5 MTD partitions on "atmel_nand": 0x000000000000-0x000000040000 : "bootstrap" mtd: Giving out device 0 to bootstrap 0x000000040000-0x0000000c0000 : "uboot" mtd: Giving out device 1 to uboot 0x0000000c0000-0x000000100000 : "env" mtd: Giving out device 2 to env 0x000000100000-0x000000400000 : "linux" mtd: Giving out device 3 to linux 0x000000400000-0x000008000000 : "root" mtd: Giving out device 4 to root PPP generic driver version 2.4.2 PPP Deflate Compression module registered PPP BSD Compression module registered PPP MPPE Compression module registered MACB_mii_bus: probed eth0: Atmel MACB at 0xfffc4000 irq 21 (02:00:de:ad:be:ef) eth0: attached PHY driver [Generic PHY] (mii_bus:phy_addr=ffffffff:00, irq=-1) ohci_hcd: USB 1.1 'Open' Host Controller (OHCI) Driver at91_ohci at91_ohci: AT91 OHCI at91_ohci at91_ohci: new USB bus registered, assigned bus number 1 at91_ohci at91_ohci: irq 20, io mem 0x00500000 usb usb1: New USB device found, idVendor=1d6b, idProduct=0001 usb usb1: New USB device strings: Mfr=3, Product=2, SerialNumber=1 usb usb1: Product: AT91 OHCI usb usb1: Manufacturer: Linux 2.6.35.11-cdu ohci_hcd usb usb1: SerialNumber: at91 hub 1-0:1.0: USB hub found hub 1-0:1.0: 2 ports detected Initializing USB Mass Storage driver... usbcore: registered new interface driver usb-storage USB Mass Storage support registered. mice: PS/2 mouse device common for all mice rtc-at91sam9 at91_rtt.0: rtc core: registered at91_rtt as rtc0 i2c /dev entries driver i2c-core: driver [dev_driver] registered i2c-dev: adapter [i2c-gpio-1] registered as minor 0 AT91SAM9 Watchdog enabled (heartbeat=15 sec, nowayout=0) Registered led device: red Registered led device: green Registered led device: yellow Registered led device: blue aic3x I2C Codec 0-001a: probe i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 No device for DAI tlv320aic3x i2c-core: driver [aic3x I2C Codec] registered No device for DAI atmel-ssc0 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 asoc: tlv320aic3x <-> atmel-ssc0 mapping ok ALSA device list: #0: AFEB9260 (tlv320aic3x) TCP cubic registered NET: Registered protocol family 17 VFP support v0.3: not present rtc-at91sam9 at91_rtt.0: setting system clock to 2011-03-10 14:13:40 UTC (1299766420) MTDSB: lookup_bdev() returned 0 MTDSB: New superblock for device 4 ("root") Empty flash at 0x02446464 ends at 0x02446800 Empty flash at 0x0244b130 ends at 0x0244b800 Empty flash at 0x029c2fb4 ends at 0x029c3000 Empty flash at 0x029c86f8 ends at 0x029c8800 Empty flash at 0x029c9ae0 ends at 0x029ca000 Empty flash at 0x029d3d54 ends at 0x029d4000 Empty flash at 0x029d933c ends at 0x029d9800 Empty flash at 0x029de144 ends at 0x029de800 Empty flash at 0x069a8020 ends at 0x069a8800 Empty flash at 0x069afdc0 ends at 0x069b0000 jffs2_scan_dirent_node(): Node CRC failed on node at 0x069b6ff0: Read 0xffffffff, calculated 0x6055a656 Empty flash at 0x069b7020 ends at 0x069b7800 Empty flash at 0x069bdf40 ends at 0x069be000 Empty flash at 0x06b84fa8 ends at 0x06b85000 VFS: Mounted root (jffs2 filesystem) on device 31:4. devtmpfs: mounted Freeing init memory: 92K JFFS2 notice: (1) check_node_data: wrong data CRC in data node at 0x029c9060: read 0x5bfb4489, calculated 0x299d675. eth0: link up (100/Full) i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 i2c i2c-0: master_xfer[0] W, addr=0x1a, len=2 JFFS2 notice: (316) check_node_data: wrong data CRC in data node at 0x06b84788: read 0x5cd0398c, calculated 0x79b2602. JFFS2 notice: (316) check_node_data: wrong data CRC in data node at 0x029c7d70: read 0x6258a55d, calculated 0x7b3d45f6. JFFS2 notice: (316) check_node_data: wrong data CRC in data node at 0x069a7800: read 0x5cd0398c, calculated 0xd59c68aa. JFFS2 notice: (316) check_node_data: wrong data CRC in data node at 0x029d89a0: read 0x865f1b6f, calculated 0xfe6d110c. JFFS2 notice: (316) check_node_data: wrong data CRC in data node at 0x029dd874: read 0xa9e45a0, calculated 0xc8106360. JFFS2 notice: (316) check_node_data: wrong data CRC in data node at 0x029d2fdc: read 0x8378e0a5, calculated 0xe157d191. JFFS2 notice: (316) check_node_data: wrong data CRC in data node at 0x069af138: read 0x6009610e, calculated 0x3d3910eb. [X.] Other notes, patches, fixes, workarounds: In older version 2.6.35.11 I had to set codec to have clock on BCLK. I thing this is not good way to set it, but it works. --- tmp/linux/linux-2.6.35.11/sound/soc/codecs/tlv320aic3x.c 2011-02-06 20:04:07.000000000 +0100 +++ arm/fw-cdu/linux/linux-2.6.35.11/sound/soc/codecs/tlv320aic3x.c 2011-03-08 15:20:07.942613211 +0100 @@ -911,6 +911,12 @@ found: data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + + //nastavit externi hodiny na BCLK, jinak to neumim + // melo by se nastavit jen kdyz codec ma hodiny na BCLK + aic3x_write(codec, AIC3X_CLKGEN_CTRL_REG, 0xa2); + //*** + aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); aic3x_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); aic3x_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); Thanks for any help. -- ************************************************** Jiøí Prchal vývojový pracovník AK signal Brno, a.s. Kulkova 26 61400 Brno Czech Republic tel: +420-545 233 484 kl. 215 fax: +420-545 233 487 www.aksignal.cz IÈO: 26245507 DIÈ: CZ26245507 GPS: 49°12'55.862"N, 16°39'14.723"E **************************************************

3 4

[alsa-devel] Problem and dirty Fix for hda-intel on ALC892
by Ilya Demyanov 23 Mar '11

23 Mar '11

I confirm this bug on 2.6.38 (debian sid) for hda-intel on ALC883 my alsa-info: http://www.alsa-project.org/db/?f=de285de868e6da935a9e2aad545c7f1bc0afa2aa

1 1

[alsa-devel] [PATCH] sound/oss/midi_synth: prevent underflow, use of uninitialized value, and signedness issue
by Dan Rosenberg 23 Mar '11

23 Mar '11

The offset passed to midi_synth_load_patch() can be essentially arbitrary. If it's greater than the header length, this will result in a copy_from_user(dst, src, negative_val). While this will just return -EFAULT on x86, on other architectures this may cause memory corruption. Additionally, the length field of the sysex_info structure may not be initialized prior to its use. Finally, a signed comparison may result in an unintentionally large loop. This patch fixes all these issues, as well as some cleanup to prevent checkpatch.pl from complaining. Signed-off-by: Dan Rosenberg <drosenberg(a)vsecurity.com> Cc: stable(a)kernel.org --- sound/oss/midi_synth.c | 27 ++++++++++++++------------- 1 files changed, 14 insertions(+), 13 deletions(-) diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c index 3c09374..3500f80 100644 --- a/sound/oss/midi_synth.c +++ b/sound/oss/midi_synth.c @@ -491,16 +491,18 @@ midi_synth_load_patch(int dev, int format, const char __user *addr, if (!prefix_cmd(orig_dev, 0xf0)) return 0; + /* Invalid patch format */ if (format != SYSEX_PATCH) - { -/* printk("MIDI Error: Invalid patch format (key) 0x%x\n", format);*/ return -EINVAL; - } + + /* Patch header too short */ if (count < hdr_size) - { -/* printk("MIDI Error: Patch header too short\n");*/ return -EINVAL; - } + + /* Offset too high */ + if (offs > offsetof(struct sysex_info, len) || offs < 0) + return -EINVAL; + count -= hdr_size; /* @@ -510,14 +512,13 @@ midi_synth_load_patch(int dev, int format, const char __user *addr, if(copy_from_user(&((char *) &sysex)[offs], &(addr)[offs], hdr_size - offs)) return -EFAULT; - - if (count < sysex.len) - { -/* printk(KERN_WARNING "MIDI Warning: Sysex record too short (%d<%d)\n", count, (int) sysex.len);*/ + + /* Sysex record too short */ + if ((unsigned)count < (unsigned)sysex.len) sysex.len = count; - } - left = sysex.len; - src_offs = 0; + + left = sysex.len; + src_offs = 0; for (i = 0; i < left && !signal_pending(current); i++) {

2 3

[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 23 Mar '11

23 Mar '11

The following changes since commit e58a8947b0264ec1f43ce5deef563b72acf1284f: Merge branch 'for-2.6.39' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2011-03-16 12:14:46 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Marek Belisko (1): ASoC: mini2440: Fix uda134x codec problem. Mark Brown (3): Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.39 ASoC: Remove bogus check for register validity in debugfs write ASoC: Support !REGULATOR build for sgtl5000 arch/arm/mach-s3c2440/mach-mini2440.c | 7 +++++++ sound/soc/codecs/sgtl5000.c | 14 ++++++++++++++ sound/soc/codecs/uda134x.c | 3 ++- sound/soc/samsung/s3c24xx_uda134x.c | 3 ++- sound/soc/soc-core.c | 2 -- 5 files changed, 25 insertions(+), 4 deletions(-)

2 1

[alsa-devel] pci/hda/patch_via.c - broken between 2.6.36 and 2.6.37 (bisected)
by Wolfgang Scheicher 23 Mar '11

23 Mar '11

After the mentioned Kernel upgrade quite a view things went wrong on my hardware ( http://www.asrock.com/mb/overview.asp?Model=890GM%20Pro3 ): * Microphone doesn't work anymore. neither the front mic nor the rear one. Tried all thinkable settings. * Increasing Microphone levels leads to some strange internal feedback. That means, at some point when the feedback starts, it just starts, but i have no clue what audio source did actually trigger it. * Touching the mixer channel "Smart 5.1" in alsamixer makes alsamicer crash. After quite some days of frustration and the inability to get help i however managed to learn how to build a kernel and how to git bisect. So this is what i did: root@Blackrock ~src/alsa-kernel (git)-[7b315bb...|bisect] # git bisect log git bisect start '--' 'sound/pci' # bad: [81a2f603f94d6da108ddd8e3535294445097f730] ALSA: ac97: replace open- coded, error-prone stuff with AC97 bit defines git bisect bad 81a2f603f94d6da108ddd8e3535294445097f730 # good: [f6f94e2ab1b33f0082ac22d71f66385a60d8157f] Linux 2.6.36 git bisect good f6f94e2ab1b33f0082ac22d71f66385a60d8157f # bad: [28c4edb71d21623f1e47422194d865d2b6712fd4] ALSA: HDA: Rename "Int Mic" to "Internal Mic" git bisect bad 28c4edb71d21623f1e47422194d865d2b6712fd4 # bad: [0e7adbe263f89ea2ef15b5af5e80a812b2a85025] ALSA: hda - Disable sticky PCM stream assignment for AD codecs git bisect bad 0e7adbe263f89ea2ef15b5af5e80a812b2a85025 # bad: [5637edb2e1c2d13b276748508ae17f319fb7f066] ALSA: HDA: Sort CXT5066 quirk table git bisect bad 5637edb2e1c2d13b276748508ae17f319fb7f066 # bad: [18675e4283f575594d55ef1239c14ab5b4de53b6] ALSA: hda - Add fixup for FSC Celsius H270 git bisect bad 18675e4283f575594d55ef1239c14ab5b4de53b6 # good: [eea7dc932bfa802ad0377755ea821f416f4f8623] ALSA: hda - Use new inputs[] field to parse input-pins for STAC/IDT codecs git bisect good eea7dc932bfa802ad0377755ea821f416f4f8623 # bad: [263d0328c46995d8e4fb478005177839104483d2] ALSA: hda - Improve input control names for IDT/STAC codecs git bisect bad 263d0328c46995d8e4fb478005177839104483d2 # bad: [9737731d64bfa64ab4fc04b46ae5b650f6432222] Merge branch 'fix/hda' into topic/hda git bisect bad 9737731d64bfa64ab4fc04b46ae5b650f6432222 # bad: [7b315bb4980448250c80a7464c256b54d546cb26] ALSA: hda - Use new inputs[] field to parse input-pins for VIA codecs git bisect bad 7b315bb4980448250c80a7464c256b54d546cb26 root@Blackrock ~src/alsa-kernel (git)-[7b315bb...|bisect] # git bisect bad 7b315bb4980448250c80a7464c256b54d546cb26 is the first bad commit commit 7b315bb4980448250c80a7464c256b54d546cb26 Author: Takashi Iwai <tiwai(a)suse.de> Date: Mon Aug 30 13:06:30 2010 +0200 ALSA: hda - Use new inputs[] field to parse input-pins for VIA codecs Signed-off-by: Takashi Iwai <tiwai(a)suse.de> :040000 040000 fb32bef12d78f90f00020cf375fddd36be1cdfbe 02f61291093ce6c7c8e7e1bd98fd87f9338702a1 M sound Oh, and here are debug infos from when i first tried to report the problem: bad: http://www.alsa-project.org/db/?f=0f29e6de98ea9fefae5c6e2adb43ff8f8eede697 good: http://www.alsa-project.org/db/?f=11290fe50066e9bcd6a244f2c83b1c66c9dd201b I hope someone can take it from here. I'm not much more than a user. Just because i managed to build a kernel doesn't mean i can fix this bug. Wolfgang Scheicher

5 13

[alsa-devel] [PATCH 1/2] ALSA: VIA HDA: Add support for VT1705
by Lydia Wang 23 Mar '11

23 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add support for VT1705. Add support for VT1705 codec, which is similiar with VT1708S except it has 6 channels output. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 100 ++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 87 insertions(+), 13 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3181,7 +3181,8 @@ int imux_is_smixer; unsigned int parm; int is_8ch = 0; - if (spec->codec_type != VT1708B_4CH) + if ((spec->codec_type != VT1708B_4CH) && + (codec->vendor_id != 0x11064397)) is_8ch = 1; /* SW0 (17h) = stereo mixer */ @@ -3220,6 +3221,16 @@ AC_VERB_SET_POWER_STATE, parm); snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_POWER_STATE, parm); + } else if (codec->vendor_id == 0x11064397) { + /* PW7(23h), SW2(27h), AOW2(25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x23, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); } /* PW 3/4/7 (1ch/1dh/23h) */ @@ -3239,7 +3250,9 @@ AC_VERB_SET_POWER_STATE, parm); snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); - } + } else if (codec->vendor_id == 0x11064397 && spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); } static int patch_vt1708S(struct hda_codec *codec); @@ -3414,6 +3427,18 @@ { } }; +static struct hda_verb vt1705_uniwill_init_verbs[] = { + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + static struct hda_pcm_stream vt1708S_pcm_analog_playback = { .substreams = 2, .channels_min = 2, @@ -3427,6 +3452,19 @@ }, }; +static struct hda_pcm_stream vt1705_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 6, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close + }, +}; + static struct hda_pcm_stream vt1708S_pcm_analog_capture = { .substreams = 2, .channels_min = 2, @@ -3473,7 +3511,10 @@ spec->multiout.dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: - spec->multiout.dac_nids[i] = 0x24; + if (spec->codec->vendor_id == 0x11064397) + spec->multiout.dac_nids[i] = 0x25; + else + spec->multiout.dac_nids[i] = 0x24; break; case AUTO_SEQ_SURROUND: spec->multiout.dac_nids[i] = 0x11; @@ -3489,22 +3530,28 @@ if (cfg->line_outs == 1) { spec->multiout.num_dacs = 3; spec->multiout.dac_nids[AUTO_SEQ_SURROUND] = 0x11; - spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x24; + if (spec->codec->vendor_id == 0x11064397) + spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x25; + else + spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x24; } return 0; } /* add playback controls from the parsed DAC table */ -static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec, +static int vt1708S_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { + struct via_spec *spec = codec->spec; char name[32]; static const char * const chname[4] = { "Front", "Surround", "C/LFE", "Side" }; - hda_nid_t nid_vols[] = {0x10, 0x11, 0x24, 0x25}; - hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x26, 0x27}; + hda_nid_t nid_vols[2][4] = { {0x10, 0x11, 0x24, 0x25}, + {0x10, 0x11, 0x25, 0} }; + hda_nid_t nid_mutes[2][4] = { {0x1C, 0x18, 0x26, 0x27}, + {0x1C, 0x18, 0x27, 0} }; hda_nid_t nid, nid_vol, nid_mute; int i, err; @@ -3515,8 +3562,15 @@ if (!nid && i > AUTO_SEQ_CENLFE) continue; - nid_vol = nid_vols[i]; - nid_mute = nid_mutes[i]; + if (codec->vendor_id == 0x11064397) { + nid_vol = nid_vols[1][i]; + nid_mute = nid_mutes[1][i]; + } else { + nid_vol = nid_vols[0][i]; + nid_mute = nid_mutes[0][i]; + } + if (!nid_vol && !nid_mute) + continue; if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ @@ -3670,7 +3724,7 @@ if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) return 0; /* can't find valid BIOS pin config */ - err = vt1708S_auto_create_multi_out_ctls(spec, &spec->autocfg); + err = vt1708S_auto_create_multi_out_ctls(codec, &spec->autocfg); if (err < 0) return err; err = vt1708S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); @@ -3737,17 +3791,29 @@ } spec->init_verbs[spec->num_iverbs++] = vt1708S_volume_init_verbs; - spec->init_verbs[spec->num_iverbs++] = vt1708S_uniwill_init_verbs; + if (codec->vendor_id == 0x11064397) + spec->init_verbs[spec->num_iverbs++] = + vt1705_uniwill_init_verbs; + else + spec->init_verbs[spec->num_iverbs++] = + vt1708S_uniwill_init_verbs; if (codec->vendor_id == 0x11060440) spec->stream_name_analog = "VT1818S Analog"; + else if (codec->vendor_id == 0x11064397) + spec->stream_name_analog = "VT1705 Analog"; else spec->stream_name_analog = "VT1708S Analog"; - spec->stream_analog_playback = &vt1708S_pcm_analog_playback; + if (codec->vendor_id == 0x11064397) + spec->stream_analog_playback = &vt1705_pcm_analog_playback; + else + spec->stream_analog_playback = &vt1708S_pcm_analog_playback; spec->stream_analog_capture = &vt1708S_pcm_analog_capture; if (codec->vendor_id == 0x11060440) spec->stream_name_digital = "VT1818S Digital"; + else if (codec->vendor_id == 0x11064397) + spec->stream_name_digital = "VT1705 Digital"; else spec->stream_name_digital = "VT1708S Digital"; spec->stream_digital_playback = &vt1708S_pcm_digital_playback; @@ -3785,6 +3851,14 @@ spec->stream_name_analog = "VT1818S Analog"; spec->stream_name_digital = "VT1818S Digital"; } + /* correct names for VT1705 */ + if (codec->vendor_id == 0x11064397) { + kfree(codec->chip_name); + codec->chip_name = kstrdup("VT1705", GFP_KERNEL); + snprintf(codec->bus->card->mixername, + sizeof(codec->bus->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); + } spec->set_widgets_power_state = set_widgets_power_state_vt1708B; return 0; } @@ -6003,7 +6077,7 @@ .patch = patch_vt1708S}, { .id = 0x11063397, .name = "VT1708S", .patch = patch_vt1708S}, - { .id = 0x11064397, .name = "VT1708S", + { .id = 0x11064397, .name = "VT1705", .patch = patch_vt1708S}, { .id = 0x11065397, .name = "VT1708S", .patch = patch_vt1708S},

2 1

[alsa-devel] snd-bt87x unknown soundcard, detected but no sound.
by Grzegorz Orciuch 23 Mar '11

23 Mar '11

Hi, I've the TV card which is Kworld KW-TV878RF and tv module bttv gives the vision well. It registers also the sound device in lspci -vv: ---- 01:08.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (1000ns min, 63750ns max) Interrupt: pin A routed to IRQ 3 Region 0: Memory at f8000000 (32-bit, prefetchable) [size=4K] Capabilities: [44] Vital Product Data pcilib: sysfs_read_vpd: read failed: Connection timed out Not readable Capabilities: [4c] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- ----- then I load the module snd-bt87x with no argument, then it says the card is not detected. Then I load the module as "modprobe snd-bt87x load_all=1 digital_rate=44100" (also 32k is not giving sound) ---- ALSA sound/pci/bt87x.c:866: unknown card 0x878-0x00:0x00 ALSA sound/pci/bt87x.c:869: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> Bt87x 0000:01:08.1: PCI INT A -> GSI 17 (level, low) -> IRQ 17 ALSA sound/pci/bt87x.c:937: bt87x0: Using board 0, analog, digital (rate 44100 Hz) ----- Can you somehow help me ? Now I'm going to connect audio-out with soundblaster line-in, as workaround. BR, Gregory

1 0

[alsa-devel] ALC888 model detection on MSI GX740 laptop
by K900 23 Mar '11

23 Mar '11

Filed everything in the bugtracker. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5321 If you need any more info, email me or comment there.

2 1

[alsa-devel] [PATCH 2/2] ALSA: VIA HDA: Add support for VT1802
by Lydia Wang 23 Mar '11

23 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add support for VT1802. Add support for VT1802 codec, which is similiar with VT2002P except VT1802 has no Class-D and has some different pin widget id. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 195 ++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 163 insertions(+), 32 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -98,9 +98,15 @@ VT1716S, VT2002P, VT1812, + VT1802, CODEC_TYPES, }; +#define VT2002P_COMPATIBLE(spec) \ + ((spec)->codec_type == VT2002P ||\ + (spec)->codec_type == VT1812 ||\ + (spec)->codec_type == VT1802) + struct via_spec { /* codec parameterization */ struct snd_kcontrol_new *mixers[6]; @@ -221,6 +227,8 @@ codec_type = VT1812; else if (dev_id == 0x0440) codec_type = VT1708S; + else if ((dev_id & 0xfff) == 0x446) + codec_type = VT1802; else codec_type = UNKNOWN; return codec_type; @@ -749,8 +757,7 @@ || spec->codec_type == VT1702 || spec->codec_type == VT1718S || spec->codec_type == VT1716S - || spec->codec_type == VT2002P - || spec->codec_type == VT1812) { + || VT2002P_COMPATIBLE(spec)) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -788,6 +795,7 @@ nid = 0x34; break; case VT2002P: + case VT1802: nid = 0x35; break; case VT1812: @@ -1071,6 +1079,7 @@ break; case VT2002P: case VT1812: + case VT1802: nid_mixer = 0x21; start_idx = 0; end_idx = 2; @@ -1135,6 +1144,7 @@ break; case VT2002P: case VT1812: + case VT1802: verb = 0xf93; parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; @@ -2151,7 +2161,8 @@ via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_analog_input(codec); - if (spec->codec_type == VT2002P || spec->codec_type == VT1812) { + + if (VT2002P_COMPATIBLE(spec)) { via_hp_bind_automute(codec); } else { via_hp_automute(codec); @@ -5291,6 +5302,57 @@ {0x1, 0xfb8, 0x88}, { } }; +static struct hda_verb vt1802_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Mic = 0 */ + {0x1e, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0}, + + /* PW9 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + + /* Enable Boost Volume backdoor */ + {0x1, 0xfb9, 0x24}, + + /* MW0/1/4/8: un-mute index 0 (MUXx), un-mute index 1 (MW9) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* set MUX0/1/4/8 = 0 (AOW0) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0}, + {0x38, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3c, AC_VERB_SET_CONNECT_SEL, 0}, + + /* set PW0 index=0 (MW0) */ + {0x24, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Enable AOW0 to MW9 */ + {0x1, 0xfb8, 0x88}, + { } +}; static struct hda_verb vt2002P_uniwill_init_verbs[] = { @@ -5303,6 +5365,16 @@ {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; +static struct hda_verb vt1802_uniwill_init_verbs[] = { + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; static struct hda_pcm_stream vt2002P_pcm_analog_playback = { .substreams = 2, @@ -5359,10 +5431,15 @@ const struct auto_pin_cfg *cfg) { int err; + hda_nid_t sw_nid; if (!cfg->line_out_pins[0]) return -1; + if (spec->codec_type == VT1802) + sw_nid = 0x28; + else + sw_nid = 0x26; /* Line-Out: PortE */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, @@ -5372,7 +5449,7 @@ return err; err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x26, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(sw_nid, 3, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -5507,21 +5584,41 @@ /* AOW0 (8h)*/ snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); - /* PW4 (26h), MW4 (1ch), MUX4(37h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x37, 0, - AC_VERB_SET_POWER_STATE, parm); + if (spec->codec_type == VT1802) { + /* PW4 (28h), MW4 (18h), MUX4(38h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x38, 0, + AC_VERB_SET_POWER_STATE, parm); + } else { + /* PW4 (26h), MW4 (1ch), MUX4(37h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x37, 0, + AC_VERB_SET_POWER_STATE, parm); + } - /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); + if (spec->codec_type == VT1802) { + /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x15, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + } else { + /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + } if (spec->hp_independent_mode) snd_hda_codec_write(codec, 0x9, 0, @@ -5534,22 +5631,35 @@ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x24, &parm); parm = present ? AC_PWRST_D3 : AC_PWRST_D0; - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); + if (spec->codec_type == VT1802) + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, parm); + else + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm); /* Mono Out */ present = snd_hda_jack_detect(codec, 0x26); parm = present ? AC_PWRST_D3 : AC_PWRST_D0; - /* PW15 (31h), MW8(17h), MUX8(3bh) */ - snd_hda_codec_write(codec, 0x31, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, parm); - + if (spec->codec_type == VT1802) { + /* PW15 (33h), MW8(1ch), MUX8(3ch) */ + snd_hda_codec_write(codec, 0x33, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, parm); + } else { + /* PW15 (31h), MW8(17h), MUX8(3bh) */ + snd_hda_codec_write(codec, 0x31, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, parm); + } /* MW9 (21h) */ if (imux_is_smixer || !is_aa_path_mute(codec)) snd_hda_codec_write(codec, 0x21, 0, @@ -5580,14 +5690,31 @@ "from BIOS. Using genenic mode...\n"); } - spec->init_verbs[spec->num_iverbs++] = vt2002P_volume_init_verbs; - spec->init_verbs[spec->num_iverbs++] = vt2002P_uniwill_init_verbs; + if (spec->codec_type == VT1802) + spec->init_verbs[spec->num_iverbs++] = + vt1802_volume_init_verbs; + else + spec->init_verbs[spec->num_iverbs++] = + vt2002P_volume_init_verbs; - spec->stream_name_analog = "VT2002P Analog"; + if (spec->codec_type == VT1802) + spec->init_verbs[spec->num_iverbs++] = + vt1802_uniwill_init_verbs; + else + spec->init_verbs[spec->num_iverbs++] = + vt2002P_uniwill_init_verbs; + + if (spec->codec_type == VT1802) + spec->stream_name_analog = "VT1802 Analog"; + else + spec->stream_name_analog = "VT2002P Analog"; spec->stream_analog_playback = &vt2002P_pcm_analog_playback; spec->stream_analog_capture = &vt2002P_pcm_analog_capture; - spec->stream_name_digital = "VT2002P Digital"; + if (spec->codec_type == VT1802) + spec->stream_name_digital = "VT1802 Digital"; + else + spec->stream_name_digital = "VT2002P Digital"; spec->stream_digital_playback = &vt2002P_pcm_digital_playback; if (!spec->adc_nids && spec->input_mux) { @@ -6118,6 +6245,10 @@ { .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812}, { .id = 0x11060440, .name = "VT1818S", .patch = patch_vt1708S}, + { .id = 0x11060446, .name = "VT1802", + .patch = patch_vt2002P}, + { .id = 0x11068446, .name = "VT1802", + .patch = patch_vt2002P}, {} /* terminator */ };

1 0

Re: [alsa-devel] 答复: 答复: [PATCH] ALSA: VIA HDA: Add new power management function
by Takashi Iwai 23 Mar '11

23 Mar '11

At Wed, 23 Mar 2011 16:39:16 +0800, <LydiaWang(a)viatech.com.cn> wrote: > > > > > -----邮件原件----- > > 发件人: Takashi Iwai [mailto:tiwai@suse.de] > > 发送时间: 2011年3月23日 15:58 > > 收件人: Lydia Wang > > 抄送: alsa-devel(a)alsa-project.org; Harald Welte > > 主题: Re: 答复: [PATCH] ALSA: VIA HDA: Add new power management function > > > > At Wed, 23 Mar 2011 15:52:04 +0800, > > <LydiaWang(a)viatech.com.cn> wrote: > > > > > > > > > > -----邮件原件----- > > > > 发件人: Takashi Iwai [mailto:tiwai@suse.de] > > > > 发送时间: 2011年3月23日 15:49 > > > > 收件人: Lydia Wang > > > > 抄送: alsa-devel(a)alsa-project.org; Harald Welte > > > > 主题: Re: [PATCH] ALSA: VIA HDA: Add new power management function > > > > > > > > At Wed, 23 Mar 2011 15:13:28 +0800, > > > > Lydia Wang wrote: > > > > > > > > > > From: Lydia Wang <lydiawang(a)viatech.com.cn> > > > > > Subject: ALSA: VIA HDA: Add new power management function. > > > > > > > > > > Use set_widgets_power_state() function to seperately control different > > > > > codecs' power management actions and to replace the original large > > > > > function. Also fix some wrong widgets power up sequence which caused > > > > > no sound issue under Smart5.1 mode and Independent HP mode. > > > > > > > > > > Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> > > > > > > > > Applied now. Thanks. > > > > > > > > > > > > Takashi > > > > > > > Thank you! > > > Can I continue to submit the rest patches? > > > > Sure. > > > > > > Takashi > > Sorry. I found forget pm function for VT1708S, so I submit a patch to add it. Is it ok or should I submit a new patch to replace the first one? I fixed the original commit (amened). thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: VIA HDA: Add VT1708S power management function
by Lydia Wang 23 Mar '11

23 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add VT1708S power management function. Define set_widgets_power_state() function for VT1708S. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> sound/pci/hda/patch_via.c | 1 + 1 file changed, 1 insertion(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3785,6 +3785,7 @@ spec->stream_name_analog = "VT1818S Analog"; spec->stream_name_digital = "VT1818S Digital"; } + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; return 0; }

1 0

Re: [alsa-devel] 答复: [PATCH] ALSA: VIA HDA: Add new power management function
by Takashi Iwai 23 Mar '11

23 Mar '11

At Wed, 23 Mar 2011 15:52:04 +0800, <LydiaWang(a)viatech.com.cn> wrote: > > > > -----邮件原件----- > > 发件人: Takashi Iwai [mailto:tiwai@suse.de] > > 发送时间: 2011年3月23日 15:49 > > 收件人: Lydia Wang > > 抄送: alsa-devel(a)alsa-project.org; Harald Welte > > 主题: Re: [PATCH] ALSA: VIA HDA: Add new power management function > > > > At Wed, 23 Mar 2011 15:13:28 +0800, > > Lydia Wang wrote: > > > > > > From: Lydia Wang <lydiawang(a)viatech.com.cn> > > > Subject: ALSA: VIA HDA: Add new power management function. > > > > > > Use set_widgets_power_state() function to seperately control different > > > codecs' power management actions and to replace the original large > > > function. Also fix some wrong widgets power up sequence which caused > > > no sound issue under Smart5.1 mode and Independent HP mode. > > > > > > Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> > > > > Applied now. Thanks. > > > > > > Takashi > > > Thank you! > Can I continue to submit the rest patches? Sure. Takashi

1 0

[alsa-devel] [PATCH] ALSA: VIA HDA: Add new power management function
by Lydia Wang 23 Mar '11

23 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add new power management function. Use set_widgets_power_state() function to seperately control different codecs' power management actions and to replace the original large function. Also fix some wrong widgets power up sequence which caused no sound issue under Smart5.1 mode and Independent HP mode. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 937 ++++++++++++++++++++++------------------------ 1 file changed, 452 insertions(+), 485 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -154,6 +154,9 @@ struct delayed_work vt1708_hp_work; int vt1708_jack_detectect; int vt1708_hp_present; + + void (*set_widgets_power_state)(struct hda_codec *codec); + #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -245,7 +248,6 @@ }; static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); -static void set_jack_power_state(struct hda_codec *codec); static int is_aa_path_mute(struct hda_codec *codec); static void vt1708_start_hp_work(struct via_spec *spec) @@ -271,6 +273,12 @@ cancel_delayed_work_sync(&spec->vt1708_hp_work); } +static void set_widgets_power_state(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + if (spec->set_widgets_power_state) + spec->set_widgets_power_state(codec); +} static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -278,7 +286,7 @@ int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - set_jack_power_state(codec); + set_widgets_power_state(codec); analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { if (is_aa_path_mute(codec)) @@ -602,482 +610,6 @@ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); } -static void set_jack_power_state(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - - if (spec->codec_type == VT1702) { - imux_is_smixer = snd_hda_codec_read( - codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - /* inputs */ - /* PW 1/2/5 (14h/15h/18h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x14, &parm); - set_pin_power_state(codec, 0x15, &parm); - set_pin_power_state(codec, 0x18, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; /* SW0 = stereo mixer (idx 3) */ - /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW 3/4 (16h/17h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x16, &parm); - set_pin_power_state(codec, 0x17, &parm); - /* MW0 (1ah), AOW 0/1 (10h/1dh) */ - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, - parm); - } else if (spec->codec_type == VT1708B_8CH - || spec->codec_type == VT1708B_4CH - || spec->codec_type == VT1708S) { - /* SW0 (17h) = stereo mixer */ - int is_8ch = spec->codec_type != VT1708B_4CH; - imux_is_smixer = snd_hda_codec_read( - codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) - == ((spec->codec_type == VT1708S) ? 5 : 0); - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW 0/1 (13h/14h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - if (spec->smart51_enabled) - parm = AC_PWRST_D0; - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW6 (22h), SW2 (26h), AOW2 (24h) */ - if (is_8ch) { - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x22, &parm); - if (spec->smart51_enabled) - parm = AC_PWRST_D0; - snd_hda_codec_write(codec, 0x26, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x24, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* PW 3/4/7 (1ch/1dh/23h) */ - parm = AC_PWRST_D3; - /* force to D0 for internal Speaker */ - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - if (is_8ch) - set_pin_power_state(codec, 0x23, &parm); - /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - if (is_8ch) { - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x27, 0, - AC_VERB_SET_POWER_STATE, parm); - } - } else if (spec->codec_type == VT1718S) { - /* MUX6 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x27, &parm); - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW2 (26h), AOW2 (ah) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW0/1 (24h/25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - set_pin_power_state(codec, 0x25, &parm); - if (!spec->hp_independent_mode) /* check for redirected HP */ - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, - parm); - /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - if (spec->hp_independent_mode) { - /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x1b, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0xc, 0, - AC_VERB_SET_POWER_STATE, parm); - } - } else if (spec->codec_type == VT1716S) { - unsigned int mono_out, present; - /* SW0 (17h) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW0(13h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1e, &parm); - /* PW11 (22h) */ - if (spec->dmic_enabled) - set_pin_power_state(codec, 0x22, &parm); - else - snd_hda_codec_write( - codec, 0x22, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - - /* SW2(26h), AIW1(14h) */ - snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - /* Smart 5.1 PW2(1bh) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1b, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW7 (23h), SW3 (27h), AOW3 (25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x23, &parm); - /* Smart 5.1 PW1(1ah) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* Smart 5.1 PW5(1eh) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1e, &parm); - snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* Mono out */ - /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ - present = snd_hda_jack_detect(codec, 0x1c); - if (present) - mono_out = 0; - else { - present = snd_hda_jack_detect(codec, 0x1d); - if (!spec->hp_independent_mode && present) - mono_out = 0; - else - mono_out = 1; - } - parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; - snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW 3/4 (1ch/1dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - /* HP Independent Mode, power on AOW3 */ - if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* force to D0 for internal Speaker */ - /* MW0 (16h), AOW0 (10h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - mono_out ? AC_PWRST_D0 : parm); - } else if (spec->codec_type == VT2002P) { - unsigned int present; - /* MUX9 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* outputs */ - /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - - /* PW4 (26h), MW4 (1ch), MUX4(37h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x37, - 0, AC_VERB_SET_POWER_STATE, parm); - - /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); - if (spec->hp_independent_mode) { - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* Class-D */ - /* PW0 (24h), MW0(18h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (present) { - snd_hda_codec_write( - codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write( - codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else { - snd_hda_codec_write( - codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write( - codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - } - - /* Mono Out */ - /* PW15 (31h), MW8(17h), MUX8(3bh) */ - present = snd_hda_jack_detect(codec, 0x26); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x31, &parm); - if (present) { - snd_hda_codec_write( - codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write( - codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else { - snd_hda_codec_write( - codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write( - codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - } - - /* MW9 (21h) */ - if (imux_is_smixer || !is_aa_path_mute(codec)) - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - else - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else if (spec->codec_type == VT1812) { - unsigned int present; - /* MUX10 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* outputs */ - /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - - /* PW4 (28h), MW4 (18h), MUX4(38h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x38, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); - if (spec->hp_independent_mode) { - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* Internal Speaker */ - /* PW0 (24h), MW0(14h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (present) { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - } else { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - } - /* Mono Out */ - /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ - present = snd_hda_jack_detect(codec, 0x28); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x31, &parm); - if (present) { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - } else { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - } - - /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x33, &parm); - snd_hda_codec_write(codec, 0x1d, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3d, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* MW9 (21h) */ - if (imux_is_smixer || !is_aa_path_mute(codec)) - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - else - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } -} - /* * input MUX handling */ @@ -1120,7 +652,7 @@ spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); /* update jack power state */ - set_jack_power_state(codec); + set_widgets_power_state(codec); return ret; } @@ -1225,7 +757,7 @@ spec->hp_independent_mode); } /* update jack power state */ - set_jack_power_state(codec); + set_widgets_power_state(codec); return 0; } @@ -1443,7 +975,7 @@ } } spec->smart51_enabled = *ucontrol->value.integer.value; - set_jack_power_state(codec); + set_widgets_power_state(codec); return 1; } @@ -1967,7 +1499,7 @@ } /* init power states */ - set_jack_power_state(codec); + set_widgets_power_state(codec); analog_low_current_mode(codec, 1); via_free_kctls(codec); /* no longer needed */ @@ -2195,7 +1727,7 @@ if (res & VIA_GPIO_EVENT) via_gpio_control(codec); if (res & VIA_JACK_EVENT) - set_jack_power_state(codec); + set_widgets_power_state(codec); if (res & VIA_MONO_EVENT) via_mono_automute(codec); if (res & VIA_SPEAKER_EVENT) @@ -3642,6 +3174,74 @@ { } /* end */ }; #endif + +static void set_widgets_power_state_vt1708B(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + int is_8ch = 0; + if (spec->codec_type != VT1708B_4CH) + is_8ch = 1; + + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = + (snd_hda_codec_read(codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) + == ((spec->codec_type == VT1708S) ? 5 : 0)); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW 0/1 (13h/14h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW6 (22h), SW2 (26h), AOW2 (24h) */ + if (is_8ch) { + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x22, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x24, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* PW 3/4/7 (1ch/1dh/23h) */ + parm = AC_PWRST_D3; + /* force to D0 for internal Speaker */ + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + if (is_8ch) + set_pin_power_state(codec, 0x23, &parm); + + /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + if (is_8ch) { + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + } +} + static int patch_vt1708S(struct hda_codec *codec); static int patch_vt1708B_8ch(struct hda_codec *codec) { @@ -3692,6 +3292,8 @@ spec->loopback.amplist = vt1708B_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; + return 0; } @@ -3742,6 +3344,8 @@ spec->loopback.amplist = vt1708B_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; + return 0; } @@ -4438,6 +4042,37 @@ }; #endif +static void set_widgets_power_state_vt1702(struct hda_codec *codec) +{ + int imux_is_smixer = + snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + unsigned int parm; + /* inputs */ + /* PW 1/2/5 (14h/15h/18h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x14, &parm); + set_pin_power_state(codec, 0x15, &parm); + set_pin_power_state(codec, 0x18, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */ + /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW 3/4 (16h/17h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x17, &parm); + set_pin_power_state(codec, 0x16, &parm); + /* MW0 (1ah), AOW 0/1 (10h/1dh) */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); +} + static int patch_vt1702(struct hda_codec *codec) { struct via_spec *spec; @@ -4484,6 +4119,7 @@ spec->loopback.amplist = vt1702_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1702; return 0; } @@ -4825,6 +4461,72 @@ }; #endif +static void set_widgets_power_state_vt1718S(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + /* MUX6 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x27, &parm); + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW2 (26h), AOW2 (ah) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x2b, &parm); + snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW0 (24h), AOW0 (8h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (!spec->hp_independent_mode) /* check for redirected HP */ + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + + /* PW1 (25h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x2a, &parm); + snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, parm); + + if (spec->hp_independent_mode) { + /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xc, 0, + AC_VERB_SET_POWER_STATE, parm); + } +} + static int patch_vt1718S(struct hda_codec *codec) { struct via_spec *spec; @@ -4886,6 +4588,8 @@ spec->loopback.amplist = vt1718S_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1718S; + return 0; } @@ -4925,8 +4629,7 @@ snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_CONNECT_SEL, index); spec->dmic_enabled = index; - set_jack_power_state(codec); - + set_widgets_power_state(codec); return 1; } @@ -5285,6 +4988,99 @@ }; #endif +static void set_widgets_power_state_vt1716S(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + unsigned int mono_out, present; + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = + (snd_hda_codec_read(codec, 0x17, 0, + AC_VERB_GET_CONNECT_SEL, 0x00) == 5); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW0(13h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1e, &parm); + /* PW11 (22h) */ + if (spec->dmic_enabled) + set_pin_power_state(codec, 0x22, &parm); + else + snd_hda_codec_write(codec, 0x22, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + /* SW2(26h), AIW1(14h) */ + snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + /* Smart 5.1 PW2(1bh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW7 (23h), SW3 (27h), AOW3 (25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x23, &parm); + /* Smart 5.1 PW1(1ah) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Smart 5.1 PW5(1eh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1e, &parm); + snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Mono out */ + /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ + present = snd_hda_jack_detect(codec, 0x1c); + + if (present) + mono_out = 0; + else { + present = snd_hda_jack_detect(codec, 0x1d); + if (!spec->hp_independent_mode && present) + mono_out = 0; + else + mono_out = 1; + } + parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; + snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW 3/4 (1ch/1dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + /* HP Independent Mode, power on AOW3 */ + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* force to D0 for internal Speaker */ + /* MW0 (16h), AOW0 (10h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + mono_out ? AC_PWRST_D0 : parm); +} + static int patch_vt1716S(struct hda_codec *codec) { struct via_spec *spec; @@ -5339,6 +5135,7 @@ spec->loopback.amplist = vt1716S_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1716S; return 0; } @@ -5609,6 +5406,83 @@ }; #endif +static void set_widgets_power_state_vt2002P(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + unsigned int present; + /* MUX9 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + parm = AC_PWRST_D0; + /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW4 (26h), MW4 (1ch), MUX4(37h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x37, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* Class-D */ + /* PW0 (24h), MW0(18h/14h), MUX0(34h) */ + present = snd_hda_jack_detect(codec, 0x25); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + parm = present ? AC_PWRST_D3 : AC_PWRST_D0; + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Mono Out */ + present = snd_hda_jack_detect(codec, 0x26); + + parm = present ? AC_PWRST_D3 : AC_PWRST_D0; + /* PW15 (31h), MW8(17h), MUX8(3bh) */ + snd_hda_codec_write(codec, 0x31, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); +} /* patch for vt2002P */ static int patch_vt2002P(struct hda_codec *codec) @@ -5660,6 +5534,7 @@ spec->loopback.amplist = vt2002P_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt2002P; return 0; } @@ -5931,6 +5806,97 @@ }; #endif +static void set_widgets_power_state_vt1812(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer = + snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + unsigned int parm; + unsigned int present; + /* MUX10 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + parm = AC_PWRST_D0; + /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (28h), MW4 (18h), MUX4(38h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x38, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* Internal Speaker */ + /* PW0 (24h), MW0(14h), MUX0(34h) */ + present = snd_hda_jack_detect(codec, 0x25); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + + /* Mono Out */ + /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ + present = snd_hda_jack_detect(codec, 0x28); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x33, &parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3d, 0, AC_VERB_SET_POWER_STATE, parm); + +} /* patch for vt1812 */ static int patch_vt1812(struct hda_codec *codec) @@ -5984,6 +5950,7 @@ spec->loopback.amplist = vt1812_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1812; return 0; }

2 1

[PATCH] ALSA: HDA: Realtek: Avoid unnecessary volume control index on Surround/Side
by David Henningsson 23 Mar '11

23 Mar '11

Similar to commit 7e59e097c09b82760bb0fe08b0fa2b704d76c3f4, this patch avoids unnecessary volume control indices for more Realtek auto-parsers, e g the ALC66x family, on the "Surround" and "Side" controls. These indices cause these volume controls to be ignored by PulseAudio and vmaster and should be removed whenever possible. Cc: stable(a)kernel.org Reported-by: Jan Losinski <losinski(a)wh2.tu-dresden.de> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 25 +++++++++++++++++-------- 1 files changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f1a03f2..dc67c5d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16006,9 +16006,12 @@ static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; - err = __alc861_create_out_sw(codec, name, nid, i, 3); + index = 0; + } + err = __alc861_create_out_sw(codec, name, nid, index, 3); if (err < 0) return err; } @@ -17159,16 +17162,19 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; + index = 0; + } err = __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT)); if (err < 0) return err; err = __add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT)); if (err < 0) @@ -19217,12 +19223,15 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; - err = __alc662_add_vol_ctl(spec, name, nid, i, 3); + index = 0; + } + err = __alc662_add_vol_ctl(spec, name, nid, index, 3); if (err < 0) return err; - err = __alc662_add_sw_ctl(spec, name, mix, i, 3); + err = __alc662_add_sw_ctl(spec, name, mix, index, 3); if (err < 0) return err; } -- 1.7.1 --------------010401080105070700060709--

1 0

Re: [alsa-devel] issue with AD1989A HDA codec and alsamixer
by Raymond Yau 23 Mar '11

23 Mar '11

2011/2/7 Vincent Meserette <vmeserette(a)aldebaran-robotics.com> > Hi Raymond, > > Thanks for your reply. > > To reply to your question, indeed the AD1989A codec is seen by alsa as a > device containing three subdevices (the three ADC) : > > result of arecord -l : > List of CAPTURE Hardware Devices : > > card 0: MID [HDA Intel MID], device 0: AD198x Analog [AD198x Analog] > > Subdevices: 3/3 > Subdevice #0: subdevice #0 > Subdevice #1: subdevice #1 > Subdevice #2: subdevice #2 > > > To record four channels simultaneously, I have created the following alsa configuration file : > > *# output device > pcm.AD1989A{ > type hw > card 0 > } > > # input device* * > pcm.quad { > type multi > > slaves.a.pcm "hw:0,0,0" # ADC 0 of the codec (left and right mics) > slaves.a.channels 2 > slaves.b.pcm "hw:0,0,1" # ADC 1 of the codec (front and rear mics) > 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 > } > > # input device containing interleaved data > pcm.quad2 { > type route > slave.pcm "quad" > ttable.0.0 1 > ttable.1.1 1 > ttable.2.2 1 > ttable.3.3 1 > } > > ### CTL DEVICES (controls) > ctl.mixer0 { > type hw > card 0 > }* > > With this configuration file, I am able to record the four channels > simultaneously with the following command : > > arecord -c4 -fS16_LE -r48000 -D «quad2 » -d5 /tmp/recording.wav > > > I found from where came my problem with the capture gains by adding some > logs in the hda_codec.c file. > When I set one of the capture gain, the put_vol_mute function well write in > the AMP_GAIN_MUTE verb of the corresponding ADC selector (0x0c, 0x0d or 0x0e > widget) but the mute bit is always set to 1. > Indeed, the value associated to the AMP_GAIN_MUTE verb is formated as folow > : > bits 0 to 6 : gain > bit 7 : mute > bits 8 to 11 : index > bit 12 : set right amp > bit 13 : set left amp > bit 14 : set input amp > bit 15 : set output amp > > and alsa sets the value of this register from a080 / b080 to a0b6 / b0b6 > when i change the gain from -58.5 dB to 22.5dB > > Do you know how alsa and thus the alsamixer determine the minimum and > maximum values for the capture gain, and how to say to alsa the the gain is > only a 7bits step value ? > > Thanks in advance. > > Best regards > > Vincent > > > > > > 2011/2/5 Raymond Yau <superquad.vortex2(a)gmail.com> > >> 2011/2/3 Vincent Meserette <vmeserette(a)aldebaran-robotics.com> >> >> > Hi Raymond, >> > >> > Many thanks for your answer. >> > >> > Looking the AD1989A datasheet (which can be found here : >> > http://www.conexant.com/products/entry.jsp?id=599) the vendor id of the >> > codec seems to be 11D4989A >> > So, I think the codec taken into account in the "patch_analog.c" file is >> > well the AD1989A because I found these lines in the source code : >> > >> > /* >> > * patch entries >> > */ >> > static struct hda_codec_preset snd_hda_preset_analog[] = { >> > ... >> > { .id = 0x11d4989a, .name = "AD1989A", .patch = patch_ad1988 }, >> > ... >> > {} /* terminator */ >> > }; >> > >> > Concerning the digital microphone input of the codec, I don't use it. I >> > have four electret microphones connected to the B and C ports of the >> codec >> > (one on the PORT-B_L pin, one on the PORT-B_R pin, one on the PORT-C_L >> pin >> > and one on the PORT-C_R pin). I have connected by software configuration >> the >> > port B to the ADC1 and the port C to the ADC0. >> > Maybe my problem comes from here, because I don't use the codec in a >> > standard configuration ? >> > if it is the case, do you know how can i resolve it ? >> > Do you also know which registers control the three capture gains of the >> > alsamixerl (i thought it was the gains of the ADC0, ADC1 and ADC2 )? >> > >> >> can you post the output of alsa-info.sh >> >> Six 192kHz 92dB ADCs >> - simultaneous record of up to 3 stereo channels >> >> It does not imply you can record 4 channels with the existing hda driver >> which create three subdevices >> > when recording with 4 channels using audacity with my ad1988a, audacity hang after a few seconds or half minute Does snd-hda-inel really suport SNDRV_PCM_INFO_SYNC_START ? ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=0/1216/4864, hwptr=1216/314944/316160/311296 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1/1216/4864, hwptr=1216/314945/316161/311296 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=1216/1216/4864, hwptr=1207/316169/317376/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1217/1216/4864, hwptr=1216/316161/317377/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=2432/1216/4864, hwptr=1216/317376/318592/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=2433/1216/4864, hwptr=1216/317377/318593/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=3648/1216/4864, hwptr=1199/318609/319808/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=3649/1216/4864, hwptr=1216/318593/319809/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=0/1216/4864, hwptr=1216/319808/321024/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1/1216/4864, hwptr=1216/319809/321025/316160 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=1216/1216/4864, hwptr=1207/321033/322240/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1217/1216/4864, hwptr=1216/321025/322241/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=2432/1216/4864, hwptr=1216/322240/323456/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=2433/1216/4864, hwptr=1216/322241/323457/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=3648/1216/4864, hwptr=1199/323473/324672/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=3649/1216/4864, hwptr=1216/323457/324673/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=0/1216/4864, hwptr=1216/324672/325888/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1/1216/4864, hwptr=1216/324673/325889/321024 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=1216/1216/4864, hwptr=1184/325920/327104/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1217/1216/4864, hwptr=1216/325889/327105/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=2432/1216/4864, hwptr=1216/327104/328320/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=2433/1216/4864, hwptr=1216/327105/328321/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=3648/1216/4864, hwptr=1215/328321/329536/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=3649/1216/4864, hwptr=1216/328321/329537/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:0: pos=0/1216/4864, hwptr=1216/329536/330752/325888 ALSA pcm_lib.c:374: period_update: pcmC0D0c:1: pos=1/1216/4864, hwptr=1216/329537/330753/325888 ALSA pcm_lib.c:172: XRUN: pcmC0D0c:1 [<f895f201>] xrun+0x97/0x9e [snd_pcm] [<f895f859>] snd_pcm_update_state+0x74/0xc1 [snd_pcm] [<f895fe03>] snd_pcm_update_hw_ptr0+0x55d/0x569 [snd_pcm] [<c042823f>] ? default_wake_function+0xb/0xd [<c0421fd3>] ? __wake_up+0x31/0x3b [<f895fe89>] snd_pcm_period_elapsed+0x7a/0xe9 [snd_pcm] [<f8a41744>] azx_interrupt+0x99/0x120 [snd_hda_intel] [<c046548f>] handle_IRQ_event+0x2f/0x64 [<c04664f8>] handle_fasteoi_irq+0x85/0xc0 [<c0466473>] ? handle_fasteoi_irq+0x0/0xc0 [<c0406e6e>] do_IRQ+0xc7/0xfe [<c0405668>] common_interrupt+0x28/0x30 [<c040a112>] ? mwait_idle+0x38/0x4b [<c0403c61>] cpu_idle+0x101/0x134 [<c06a7db4>] start_secondary+0x197/0x19f ======================= ALSA hda_intel.c:1679: azx_pcm_prepare: bufsize=0x4c00, format=0x11 ALSA hda_codec.c:1228: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x11 ALSA hda_intel.c:1679: azx_pcm_prepare: bufsize=0x4c00, format=0x11 ALSA hda_codec.c:1228: hda_codec_setup_stream: NID=0x9, stream=0x2, channel=0, format=0x11 ALSA hda_intel.c:1679: azx_pcm_prepare: bufsize=0x4c00, format=0x11 ALSA hda_codec.c:1228: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x11 ALSA hda_intel.c:1679: azx_pcm_prepare: bufsize=0x4c00, format=0x11 ALSA hda_codec.c:1228: hda_codec_setup_stream: NID=0x9, stream=0x2, channel=0, format=0x11 ALSA hda_codec.c:1291: hda_codec_cleanup_stream: NID=0x8 ALSA hda_codec.c:1291: hda_codec_cleanup_stream: NID=0x9 ALSA hda_codec.c:1291: hda_codec_cleanup_stream: NID=0x8 ALSA hda_codec.c:1291: hda_codec_cleanup_stream: NID=0x9

1 0

[alsa-devel] [PATCH 1/4] ASoC: Support download of WM8958 MBC firmware
by Mark Brown 22 Mar '11

22 Mar '11

Allow userspace to supply an update to the ROM firmware. The firmware request is non-blocking so userspace can load the firmware at its leisure without delaying startup, the driver will begin using the firmware the next time MBC is started after it has been supplied. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8958-dsp2.c | 223 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8994.c | 4 + sound/soc/codecs/wm8994.h | 4 + 3 files changed, 231 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index c4e4baa..88fd4a9 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -30,10 +30,212 @@ #include "wm8994.h" +#define WM_FW_BLOCK_INFO 0xff +#define WM_FW_BLOCK_PM 0x00 +#define WM_FW_BLOCK_X 0x01 +#define WM_FW_BLOCK_Y 0x02 +#define WM_FW_BLOCK_Z 0x03 +#define WM_FW_BLOCK_I 0x06 +#define WM_FW_BLOCK_A 0x08 +#define WM_FW_BLOCK_C 0x0c + +static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, + const struct firmware *fw, bool check) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + u64 data64; + u32 data32; + const u8 *data; + char *str; + size_t block_len, len; + int ret = 0; + + /* Suppress unneeded downloads */ + if (wm8994->cur_fw == fw) + return 0; + + if (fw->size < 32) { + dev_err(codec->dev, "%s: firmware too short\n", name); + goto err; + } + + if (memcmp(fw->data, "WMFW", 4) != 0) { + dev_err(codec->dev, "%s: firmware has bad file magic %08x\n", + name, data32); + goto err; + } + + memcpy(&data32, fw->data + 4, sizeof(data32)); + len = be32_to_cpu(data32); + + memcpy(&data32, fw->data + 8, sizeof(data32)); + data32 = be32_to_cpu(data32); + if ((data32 >> 24) & 0xff) { + dev_err(codec->dev, "%s: unsupported firmware version %d\n", + name, (data32 >> 24) & 0xff); + goto err; + } + if ((data32 & 0xffff) != 8958) { + dev_err(codec->dev, "%s: unsupported target device %d\n", + name, data32 & 0xffff); + goto err; + } + if (((data32 >> 16) & 0xff) != 0xc) { + dev_err(codec->dev, "%s: unsupported target core %d\n", + name, (data32 >> 16) & 0xff); + goto err; + } + + if (check) { + memcpy(&data64, fw->data + 24, sizeof(u64)); + dev_info(codec->dev, "%s timestamp %llx\n", + name, be64_to_cpu(data64)); + } else { + snd_soc_write(codec, 0x102, 0x2); + snd_soc_write(codec, 0x900, 0x2); + } + + data = fw->data + len; + len = fw->size - len; + while (len) { + if (len < 12) { + dev_err(codec->dev, "%s short data block of %d\n", + name, len); + goto err; + } + + memcpy(&data32, data + 4, sizeof(data32)); + block_len = be32_to_cpu(data32); + if (block_len + 8 > len) { + dev_err(codec->dev, "%d byte block longer than file\n", + block_len); + goto err; + } + if (block_len == 0) { + dev_err(codec->dev, "Zero length block\n"); + goto err; + } + + memcpy(&data32, data, sizeof(data32)); + data32 = be32_to_cpu(data32); + + switch ((data32 >> 24) & 0xff) { + case WM_FW_BLOCK_INFO: + /* Informational text */ + if (!check) + break; + + str = kzalloc(block_len + 1, GFP_KERNEL); + if (str) { + memcpy(str, data + 8, block_len); + dev_info(codec->dev, "%s: %s\n", name, str); + kfree(str); + } else { + dev_err(codec->dev, "Out of memory\n"); + } + break; + case WM_FW_BLOCK_PM: + case WM_FW_BLOCK_X: + case WM_FW_BLOCK_Y: + case WM_FW_BLOCK_Z: + case WM_FW_BLOCK_I: + case WM_FW_BLOCK_A: + case WM_FW_BLOCK_C: + dev_dbg(codec->dev, "%s: %d bytes of %x@%x\n", name, + block_len, (data32 >> 24) & 0xff, + data32 & 0xffffff); + + if (check) + break; + + data32 &= 0xffffff; + + wm8994_bulk_write(codec->control_data, + data32 & 0xffffff, + block_len / 2, + (void *)(data + 8)); + + break; + default: + dev_warn(codec->dev, "%s: unknown block type %d\n", + name, (data32 >> 24) & 0xff); + break; + } + + /* Round up to the next 32 bit word */ + block_len += block_len % 4; + + data += block_len + 8; + len -= block_len + 8; + } + + if (!check) { + dev_dbg(codec->dev, "%s: download done\n", name); + wm8994->cur_fw = fw; + } else { + dev_info(codec->dev, "%s: got firmware\n", name); + } + + goto ok; + +err: + ret = -EINVAL; +ok: + if (!check) { + snd_soc_write(codec, 0x900, 0x0); + snd_soc_write(codec, 0x102, 0x0); + } + + return ret; +} + static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994_pdata *pdata = wm8994->pdata; + int i; + + /* If the DSP is already running then noop */ + if (snd_soc_read(codec, WM8958_DSP2_PROGRAM) & WM8958_DSP2_ENA) + return; + + /* If we have MBC firmware download it */ + if (wm8994->mbc) + wm8958_dsp2_fw(codec, "MBC", wm8994->mbc, false); + + snd_soc_update_bits(codec, WM8958_DSP2_PROGRAM, + WM8958_DSP2_ENA, WM8958_DSP2_ENA); + + /* If we've got user supplied MBC settings use them */ + if (pdata && pdata->num_mbc_cfgs) { + struct wm8958_mbc_cfg *cfg + = &pdata->mbc_cfgs[wm8994->mbc_cfg]; + + for (i = 0; i < ARRAY_SIZE(cfg->coeff_regs); i++) + snd_soc_write(codec, i + WM8958_MBC_BAND_1_K_1, + cfg->coeff_regs[i]); + + for (i = 0; i < ARRAY_SIZE(cfg->cutoff_regs); i++) + snd_soc_write(codec, + i + WM8958_MBC_BAND_2_LOWER_CUTOFF_C1_1, + cfg->cutoff_regs[i]); + } + + /* Run the DSP */ + snd_soc_write(codec, WM8958_DSP2_EXECCONTROL, + WM8958_DSP2_RUNR); + + /* And we're off! */ + snd_soc_update_bits(codec, WM8958_DSP2_CONFIG, + WM8958_MBC_ENA | + WM8958_MBC_SEL_MASK, + path << WM8958_MBC_SEL_SHIFT | + WM8958_MBC_ENA); +} + +static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int pwr_reg = snd_soc_read(codec, WM8994_POWER_MANAGEMENT_5); int ena, reg, aif, i; @@ -76,6 +278,10 @@ static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start) & WM8994_AIF2CLK_ENA_MASK)) return; + /* If we have MBC firmware download it */ + if (wm8994->mbc && wm8994->mbc_ena[mbc]) + wm8958_dsp2_fw(codec, "MBC", wm8994->mbc, false); + /* Switch the clock over to the appropriate AIF */ snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8958_DSP2CLK_SRC | WM8958_DSP2CLK_ENA, @@ -242,6 +448,18 @@ WM8958_MBC_SWITCH("AIF1DAC2 MBC Switch", 1), WM8958_MBC_SWITCH("AIF2DAC MBC Switch", 2), }; +static void wm8958_mbc_loaded(const struct firmware *fw, void *context) +{ + struct snd_soc_codec *codec = context; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + if (fw && wm8958_dsp2_fw(codec, "MBC", fw, true) != 0) { + mutex_lock(&codec->mutex); + wm8994->mbc = fw; + mutex_unlock(&codec->mutex); + } +} + void wm8958_dsp2_init(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -251,6 +469,11 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8958_mbc_snd_controls, ARRAY_SIZE(wm8958_mbc_snd_controls)); + /* We don't require firmware and don't want to delay boot */ + request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + "wm8958_mbc.wfw", codec->dev, GFP_KERNEL, + codec, wm8958_mbc_loaded); + if (!pdata) return; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bdd1ac7..f622ff6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1922,6 +1922,8 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8994_VMID_BUF_ENA | WM8994_VMID_RAMP_MASK, 0); + wm8994->cur_fw = NULL; + pm_runtime_put(codec->dev); } break; @@ -3136,6 +3138,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) free_irq(wm8994->micdet_irq, wm8994); break; } + if (wm8994->mbc) + release_firmware(wm8994->mbc); kfree(wm8994->retune_mobile_texts); kfree(wm8994->drc_texts); kfree(wm8994); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 93a6cf1..1aa365b 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -10,6 +10,7 @@ #define _WM8994_H #include <sound/soc.h> +#include <linux/firmware.h> #include "wm_hubs.h" @@ -114,6 +115,9 @@ struct wm8994_priv { unsigned int aif1clk_disable:1; unsigned int aif2clk_disable:1; + + const struct firmware *cur_fw; + const struct firmware *mbc; }; #endif -- 1.7.4.1

2 4

[alsa-devel] [PATCH] ASoC: Support !REGULATOR build for sgtl5000
by Mark Brown 22 Mar '11

22 Mar '11

The regulator is optional depending on board design. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sgtl5000.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f7217f..ff29380 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -772,6 +772,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } +#ifdef CONFIG_REGULATOR static int ldo_regulator_is_enabled(struct regulator_dev *dev) { struct ldo_regulator *ldo = rdev_get_drvdata(dev); @@ -901,6 +902,19 @@ static int ldo_regulator_remove(struct snd_soc_codec *codec) return 0; } +#else +static int ldo_regulator_register(struct snd_soc_codec *codec, + struct regulator_init_data *init_data, + int voltage) +{ + return -EINVAL; +} + +static int ldo_regulator_remove(struct snd_soc_codec *codec) +{ + return 0; +} +#endif /* * set dac bias -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: tlv320dac33: Fix inconsistent spinlock usage
by Peter Ujfalusi 22 Mar '11

22 Mar '11

The lock is used within the interrupt handler. Correct the spinlock usage, and use irqsave/irqrestore flavour of spin_lock/unlock. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 28 ++++++++++++++++------------ 1 files changed, 16 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 71d7be8..93f38b1 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -670,6 +670,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) { struct snd_soc_codec *codec = dac33->codec; unsigned int delay; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: @@ -677,10 +678,10 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) DAC33_THRREG(dac33->nsample)); /* Take the timestamps */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp2 = ktime_to_us(ktime_get()); dac33->t_stamp1 = dac33->t_stamp2; - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_PREFILL_MSB, DAC33_THRREG(dac33->alarm_threshold)); @@ -692,11 +693,11 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) break; case DAC33_FIFO_MODE7: /* Take the timestamp */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp1 = ktime_to_us(ktime_get()); /* Move back the timestamp with drain time */ dac33->t_stamp1 -= dac33->mode7_us_to_lthr; - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_PREFILL_MSB, DAC33_THRREG(DAC33_MODE7_MARGIN)); @@ -714,13 +715,14 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) static inline void dac33_playback_handler(struct tlv320dac33_priv *dac33) { struct snd_soc_codec *codec = dac33->codec; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: /* Take the timestamp */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp2 = ktime_to_us(ktime_get()); - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_NSAMPLE_MSB, DAC33_THRREG(dac33->nsample)); @@ -773,10 +775,11 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev) { struct snd_soc_codec *codec = dev; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + unsigned long flags; - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp1 = ktime_to_us(ktime_get()); - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); /* Do not schedule the workqueue in Mode7 */ if (dac33->fifo_mode != DAC33_FIFO_MODE7) @@ -1173,15 +1176,16 @@ static snd_pcm_sframes_t dac33_dai_delay( unsigned int time_delta, uthr; int samples_out, samples_in, samples; snd_pcm_sframes_t delay = 0; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_BYPASS: break; case DAC33_FIFO_MODE1: - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); t0 = dac33->t_stamp1; t1 = dac33->t_stamp2; - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); t_now = ktime_to_us(ktime_get()); /* We have not started to fill the FIFO yet, delay is 0 */ @@ -1246,10 +1250,10 @@ static snd_pcm_sframes_t dac33_dai_delay( } break; case DAC33_FIFO_MODE7: - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); t0 = dac33->t_stamp1; uthr = dac33->uthr; - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); t_now = ktime_to_us(ktime_get()); /* We have not started to fill the FIFO yet, delay is 0 */ -- 1.7.4.1

3 2

[alsa-devel] [PATCH 1/2 v4] ALSA: Add snd_ctl_replace() to dynamically replace a control
by Dimitris Papastamos 22 Mar '11

22 Mar '11

Add a function to dynamically replace a given control. If the control does not already exist, a third parameter is used to determine whether to actually add that control. This is useful in cases where downloadable firmware at runtime can add or replace existing controls. A separate patch needs to be made to allow ALSA Mixer to render the replaced controls on the fly. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/control.h | 1 + sound/core/control.c | 65 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 66 insertions(+), 0 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index e67db28..1c8b3ee 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -113,6 +113,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new * kcontrolnew, v void snd_ctl_free_one(struct snd_kcontrol * kcontrol); int snd_ctl_add(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol); +int snd_ctl_replace(struct snd_card * card, struct snd_kcontrol * kcontrol, bool add_on_replace); int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id); int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id); int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, diff --git a/sound/core/control.c b/sound/core/control.c index db51e4e..a8cae8a 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -368,6 +368,71 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) EXPORT_SYMBOL(snd_ctl_add); /** + * snd_ctl_replace - replace the control instance of the card + * @card: the card instance + * @kcontrol: the control instance to replace + * @add_on_replace: add the control if not already added + * + * Replaces the given control. If the given control does not exist + * and the add_on_replace flag is set, the control is added. If the + * control exists, it is destroyed first. + * + * Returns zero if successful, or a negative error code on failure. + * + * It frees automatically the control which cannot be added or replaced. + */ +int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, + bool add_on_replace) +{ + struct snd_ctl_elem_id id; + unsigned int idx; + struct snd_kcontrol *old; + int ret; + + if (!kcontrol) + return -EINVAL; + if (snd_BUG_ON(!card || !kcontrol->info)) { + ret = -EINVAL; + goto error; + } + id = kcontrol->id; + down_write(&card->controls_rwsem); + old = snd_ctl_find_id(card, &id); + if (!old) { + if (add_on_replace) + goto add; + up_write(&card->controls_rwsem); + ret = -EINVAL; + goto error; + } + ret = snd_ctl_remove(card, old); + if (ret < 0) { + up_write(&card->controls_rwsem); + goto error; + } +add: + if (snd_ctl_find_hole(card, kcontrol->count) < 0) { + up_write(&card->controls_rwsem); + ret = -ENOMEM; + goto error; + } + list_add_tail(&kcontrol->list, &card->controls); + card->controls_count += kcontrol->count; + kcontrol->id.numid = card->last_numid + 1; + card->last_numid += kcontrol->count; + up_write(&card->controls_rwsem); + for (idx = 0; idx < kcontrol->count; idx++, id.index++, id.numid++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id); + return 0; + +error: + snd_ctl_free_one(kcontrol); + return ret; +} + +EXPORT_SYMBOL(snd_ctl_replace); + +/** * snd_ctl_remove - remove the control from the card and release it * @card: the card instance * @kcontrol: the control instance to remove -- 1.7.4.1

3 7

[alsa-devel] Compressed Audio Playback/Capture through ALSA framework
by Patrick Lai 22 Mar '11

22 Mar '11

Hi, Is there a precedent for playback/capture compressed audio stream through ALSA playback/capture interface if underlying hardware supports decoder + sink & encoder + source capabilities? Which source file can I find an example? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

10 30

[alsa-devel] [PATCH 0/7] ALSA: VIA HDA: patch series
by Lydia Wang 22 Mar '11

22 Mar '11

Following patch series mainly includes: -fix stereo mixer recording no sound issue -fix independent headphone no sound issue -fix invalid a-a path volume adjust issue for some codecs -fix side channel has sound under independent hp mode of VT1718S issue -correct display stream names for VT1818S -fix hp mixer build fail of VT1708A issue

2 1

[alsa-devel] [PATCH 1/5] hdspmixer: Fix toggle of master mute and solo buttons
by Adrian Knoth 22 Mar '11

22 Mar '11

The sensitive area for master mute and solo buttons was slightly off. Discovered and fixed by Reuben Martin. Thanks for the patch. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/hdspmixer/src/HDSPMixerMaster.cxx b/hdspmixer/src/HDSPMixerMaster.cxx index 77c4a28..f4399eb 100644 --- a/hdspmixer/src/HDSPMixerMaster.cxx +++ b/hdspmixer/src/HDSPMixerMaster.cxx @@ -31,21 +31,21 @@ HDSPMixerMaster::HDSPMixerMaster(int x, int y, int w, int h):Fl_Widget(x, y, 62, void HDSPMixerMaster::draw() { if (mute && mute_active) { - fl_push_clip(x(), y(), 29, 10); + fl_push_clip(x(), y(), 30, 11); fl_draw_pixmap(b_mute_xpm, x(), y()); fl_pop_clip(); } else if (mute) { - fl_push_clip(x(), y(), 29, 10); - fl_draw_pixmap(b_mute_xpm, x(), y()-10); + fl_push_clip(x(), y(), 30, 11); + fl_draw_pixmap(b_mute_xpm, x(), y()-11); fl_pop_clip(); } if (solo && solo_active) { - fl_push_clip(x()+32, y(), 29, 10); + fl_push_clip(x()+32, y(), 30, 11); fl_draw_pixmap(b_solo_xpm, x()+32, y()); fl_pop_clip(); } else if (solo) { - fl_push_clip(x()+32, y(), 29, 10); - fl_draw_pixmap(b_solo_xpm, x()+32, y()-10); + fl_push_clip(x()+32, y(), 30, 11); + fl_draw_pixmap(b_solo_xpm, x()+32, y()-11); fl_pop_clip(); } } @@ -55,7 +55,7 @@ int HDSPMixerMaster::handle(int e) int xpos = Fl::event_x()-x(); switch (e) { case FL_PUSH: - if (xpos >= 0 && xpos <= 29) { + if (xpos >= 0 && xpos <= 30) { if (mute) { mute = 0; } else { -- 1.7.4.1

2 5

[alsa-devel] [PATCH 7/7] ALSA: VIA HDA: Fix VT1708 can't build up Headphone control issue
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix VT1708 can't build up Headphone control issue. Since VT1708 didn't support the control of getting connection number, building of headphone control will fail in via_hp_build() function. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1266,9 +1266,12 @@ break; } - nums = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); - if (nums <= 1) - return 0; + if (spec->codec_type != VT1708) { + nums = snd_hda_get_connections(codec, nid, + conn, HDA_MAX_CONNECTIONS); + if (nums <= 1) + return 0; + } knew = via_clone_control(spec, &via_hp_mixer[0]); if (knew == NULL)

3 2

[alsa-devel] [PATCH 6/7] ALSA: VIA HDA: Correct stream names for VT1818S
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Correct stream names for VT1818S. Correct stream names of analog playback and capture streams for VT1818S. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 5 +++++ 1 file changed, 5 insertions(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -4173,6 +4173,11 @@ spec->stream_name_analog = "VT1708BCE Analog"; spec->stream_name_digital = "VT1708BCE Digital"; } + /* correct names for VT1818S */ + if (codec->vendor_id == 0x11060440) { + spec->stream_name_analog = "VT1818S Analog"; + spec->stream_name_digital = "VT1818S Digital"; + } return 0; }

1 0

[alsa-devel] [PATCH 5/7] ALSA: VIA HDA: Get codec's type in via_new_spec() function
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Get codec's type in via_new_spec() function. Add get_codec_type() in via_new_spec() function to make sure getting correct codec type before building mixer controls. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -159,6 +159,7 @@ #endif }; +static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec); static struct via_spec * via_new_spec(struct hda_codec *codec) { struct via_spec *spec; @@ -169,6 +170,10 @@ codec->spec = spec; spec->codec = codec; + spec->codec_type = get_codec_type(codec); + /* VT1708BCE & VT1708S are almost same */ + if (spec->codec_type == VT1708BCE) + spec->codec_type = VT1708S; return spec; } @@ -2203,10 +2208,6 @@ for (i = 0; i < spec->num_iverbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); - spec->codec_type = get_codec_type(codec); - if (spec->codec_type == VT1708BCE) - spec->codec_type = VT1708S; /* VT1708BCE & VT1708S are almost - same */ /* Lydia Add for EAPD enable */ if (!spec->dig_in_nid) { /* No Digital In connection */ if (spec->dig_in_pin) {

1 0

[alsa-devel] [PATCH 4/7] ALSA: VIA HDA: Fix invalid A-A path volume adjust issue
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix invalid A-A path volume adjust issue. Modify vt_auto_create_analog_input_ctls() function to fix invalid a-a path volume adjust issue for VT1708S, VT1702 and VT1716S codecs. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2456,7 +2456,14 @@ else type_idx = 0; label = hda_get_autocfg_input_label(codec, cfg, i); - err = via_new_analog_input(spec, label, type_idx, idx, cap_nid); + if (spec->codec_type == VT1708S || + spec->codec_type == VT1702 || + spec->codec_type == VT1716S) + err = via_new_analog_input(spec, label, type_idx, + idx+1, cap_nid); + else + err = via_new_analog_input(spec, label, type_idx, + idx, cap_nid); if (err < 0) return err; snd_hda_add_imux_item(imux, label, idx, NULL);

1 0

[alsa-devel] [PATCH 3/7] ALSA: VIA HDA: Modify mute_aa_path() function
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Modify mute_aa_path() function. Modify mute_aa_path() function to support VT1718S codec. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 5 +++++ 1 file changed, 5 insertions(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1323,6 +1323,11 @@ start_idx = 2; end_idx = 4; break; + case VT1718S: + nid_mixer = 0x21; + start_idx = 1; + end_idx = 3; + break; default: return; }

1 0

[alsa-devel] [PATCH 2/7] ALSA: VIA HDA: Fix independent headphone no sound issue
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix independent headphone no sound issue. Modify via_independent_hp_put() function to support VT1718S and VT1812 codecs, and fix independent headphone no sound issue. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1191,8 +1191,16 @@ /* Get Independent Mode index of headphone pin widget */ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); + if (spec->codec_type == VT1718S) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0); + else + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, pinsel); + if (spec->codec_type == VT1812) + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_CONNECT_SEL, pinsel); if (spec->multiout.hp_nid && spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, @@ -1211,6 +1219,8 @@ activate_ctl(codec, "Headphone Playback Switch", spec->hp_independent_mode); } + /* update jack power state */ + set_jack_power_state(codec); return 0; }

1 0

[alsa-devel] [PATCH 1/7] ALSA: VIA HDA: Fix stereo mixer recording no sound issue
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix stereo mixer recording no sound issue. Modify function via_mux_enum_put() to fix stereo mixer recording no sound issue. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1101,6 +1101,7 @@ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + int ret; if (!spec->mux_nids[adc_idx]) return -EINVAL; @@ -1109,12 +1110,14 @@ AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - /* update jack power state */ - set_jack_power_state(codec); - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + ret = snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); + /* update jack power state */ + set_jack_power_state(codec); + + return ret; } static int via_independent_hp_info(struct snd_kcontrol *kcontrol,

1 0

[alsa-devel] [PATCH 2/8] ALSA: VIA HDA: Fix stereo mixer recording no sound issue
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix stereo mixer recording no sound issue. Modify function via_mux_enum_put() to fix stereo mixer recording no sound issue. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -633,6 +633,7 @@ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + int ret; if (!spec->mux_nids[adc_idx]) return -EINVAL; @@ -641,12 +642,14 @@ AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - /* update jack power state */ - set_widgets_power_state(codec); - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + ret = snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); + /* update jack power state */ + set_widgets_power_state(codec); + + return ret; } static int via_independent_hp_info(struct snd_kcontrol *kcontrol,

2 2

[alsa-devel] [PATCH 7/8] ALSA: VIA HDA: Get codec's type in via_new_spec() function
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Get codec's type in via_new_spec() function. Add get_codec_type() in via_new_spec() function to make sure getting correct codec type before building mixer controls. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -162,6 +162,7 @@ #endif }; +static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec); static struct via_spec * via_new_spec(struct hda_codec *codec) { struct via_spec *spec; @@ -172,6 +173,10 @@ codec->spec = spec; spec->codec = codec; + spec->codec_type = get_codec_type(codec); + /* VT1708BCE & VT1708S are almost same */ + if (spec->codec_type == VT1708BCE) + spec->codec_type = VT1708S; return spec; } @@ -1735,10 +1740,6 @@ for (i = 0; i < spec->num_iverbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); - spec->codec_type = get_codec_type(codec); - if (spec->codec_type == VT1708BCE) - spec->codec_type = VT1708S; /* VT1708BCE & VT1708S are almost - same */ /* Lydia Add for EAPD enable */ if (!spec->dig_in_nid) { /* No Digital In connection */ if (spec->dig_in_pin) {

2 1

[alsa-devel] [PATCH 8/8] ALSA: VIA HDA: Fix VT1708 can't build up Headphone control issue
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix VT1708 can't build up Headphone control issue. Since VT1708 didn't support the control of getting connection number, building of headphone control will fail in via_hp_build() function. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -798,9 +798,12 @@ break; } - nums = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); - if (nums <= 1) - return 0; + if (spec->codec_type != VT1708) { + nums = snd_hda_get_connections(codec, nid, + conn, HDA_MAX_CONNECTIONS); + if (nums <= 1) + return 0; + } knew = via_clone_control(spec, &via_hp_mixer[0]); if (knew == NULL)

1 0

[alsa-devel] [PATCH 6/8] ALSA: VIA HDA: Correct stream names for VT1818S
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Correct stream names for VT1818S. Correct stream names of analog playback and capture streams for VT1818S. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 5 +++++ 1 file changed, 5 insertions(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3777,6 +3777,11 @@ spec->stream_name_analog = "VT1708BCE Analog"; spec->stream_name_digital = "VT1708BCE Digital"; } + /* correct names for VT1818S */ + if (codec->vendor_id == 0x11060440) { + spec->stream_name_analog = "VT1818S Analog"; + spec->stream_name_digital = "VT1818S Digital"; + } spec->set_widgets_power_state = set_widgets_power_state_vt1708B; return 0; }

1 0

[alsa-devel] [PATCH 5/8] ALSA: VIA HDA: Modify mute_aa_path() function
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Modify mute_aa_path() function. Modify mute_aa_path() function to support VT1718S codec. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 5 +++++ 1 file changed, 5 insertions(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -855,6 +855,11 @@ start_idx = 2; end_idx = 4; break; + case VT1718S: + nid_mixer = 0x21; + start_idx = 1; + end_idx = 3; + break; default: return; }

1 0

[alsa-devel] [PATCH 4/8] ALSA: VIA HDA: Fix invalid A-A path volume adjust issue
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix invalid A-A path volume adjust issue. Modify vt_auto_create_analog_input_ctls() function to fix invalid a-a path volume adjust issue for VT1708S, VT1702 and VT1716S codecs. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1983,7 +1983,14 @@ else type_idx = 0; label = hda_get_autocfg_input_label(codec, cfg, i); - err = via_new_analog_input(spec, label, type_idx, idx, cap_nid); + if (spec->codec_type == VT1708S || + spec->codec_type == VT1702 || + spec->codec_type == VT1716S) + err = via_new_analog_input(spec, label, type_idx, + idx+1, cap_nid); + else + err = via_new_analog_input(spec, label, type_idx, + idx, cap_nid); if (err < 0) return err; snd_hda_add_imux_item(imux, label, idx, NULL);

1 0

[alsa-devel] [PATCH 3/8] ALSA: VIA HDA: Fix independent headphone no sound issue
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix independent headphone no sound issue. Modify via_independent_hp_put() function to support VT1718S and VT1812 codecs, and fix independent headphone no sound issue. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -723,8 +723,16 @@ /* Get Independent Mode index of headphone pin widget */ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); + if (spec->codec_type == VT1718S) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0); + else + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, pinsel); + if (spec->codec_type == VT1812) + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_CONNECT_SEL, pinsel); if (spec->multiout.hp_nid && spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, @@ -743,6 +751,8 @@ activate_ctl(codec, "Headphone Playback Switch", spec->hp_independent_mode); } + /* update jack power state */ + set_widgets_power_state(codec); return 0; }

1 0

[alsa-devel] [PATCH 1/8] ALSA: VIA HDA: Add new power management function
by Lydia Wang 22 Mar '11

22 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add new power management function. Use set_widgets_power_state() function to seperately control different codecs' power management actions. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 940 ++++++++++++++++++++++------------------------ 1 file changed, 456 insertions(+), 484 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -154,6 +154,9 @@ struct delayed_work vt1708_hp_work; int vt1708_jack_detectect; int vt1708_hp_present; + + void (*set_widgets_power_state)(struct hda_codec *codec); + #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -240,7 +243,6 @@ }; static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); -static void set_jack_power_state(struct hda_codec *codec); static int is_aa_path_mute(struct hda_codec *codec); static void vt1708_start_hp_work(struct via_spec *spec) @@ -266,6 +268,12 @@ cancel_delayed_work_sync(&spec->vt1708_hp_work); } +static void set_widgets_power_state(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + if (spec->set_widgets_power_state) + spec->set_widgets_power_state(codec); +} static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -273,7 +281,7 @@ int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - set_jack_power_state(codec); + set_widgets_power_state(codec); analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { if (is_aa_path_mute(codec)) @@ -597,482 +605,6 @@ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); } -static void set_jack_power_state(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - - if (spec->codec_type == VT1702) { - imux_is_smixer = snd_hda_codec_read( - codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - /* inputs */ - /* PW 1/2/5 (14h/15h/18h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x14, &parm); - set_pin_power_state(codec, 0x15, &parm); - set_pin_power_state(codec, 0x18, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; /* SW0 = stereo mixer (idx 3) */ - /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW 3/4 (16h/17h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x16, &parm); - set_pin_power_state(codec, 0x17, &parm); - /* MW0 (1ah), AOW 0/1 (10h/1dh) */ - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, - parm); - } else if (spec->codec_type == VT1708B_8CH - || spec->codec_type == VT1708B_4CH - || spec->codec_type == VT1708S) { - /* SW0 (17h) = stereo mixer */ - int is_8ch = spec->codec_type != VT1708B_4CH; - imux_is_smixer = snd_hda_codec_read( - codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) - == ((spec->codec_type == VT1708S) ? 5 : 0); - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW 0/1 (13h/14h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - if (spec->smart51_enabled) - parm = AC_PWRST_D0; - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW6 (22h), SW2 (26h), AOW2 (24h) */ - if (is_8ch) { - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x22, &parm); - if (spec->smart51_enabled) - parm = AC_PWRST_D0; - snd_hda_codec_write(codec, 0x26, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x24, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* PW 3/4/7 (1ch/1dh/23h) */ - parm = AC_PWRST_D3; - /* force to D0 for internal Speaker */ - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - if (is_8ch) - set_pin_power_state(codec, 0x23, &parm); - /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - if (is_8ch) { - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x27, 0, - AC_VERB_SET_POWER_STATE, parm); - } - } else if (spec->codec_type == VT1718S) { - /* MUX6 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x27, &parm); - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW2 (26h), AOW2 (ah) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW0/1 (24h/25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - set_pin_power_state(codec, 0x25, &parm); - if (!spec->hp_independent_mode) /* check for redirected HP */ - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, - parm); - /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - if (spec->hp_independent_mode) { - /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x1b, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0xc, 0, - AC_VERB_SET_POWER_STATE, parm); - } - } else if (spec->codec_type == VT1716S) { - unsigned int mono_out, present; - /* SW0 (17h) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW0(13h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1e, &parm); - /* PW11 (22h) */ - if (spec->dmic_enabled) - set_pin_power_state(codec, 0x22, &parm); - else - snd_hda_codec_write( - codec, 0x22, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - - /* SW2(26h), AIW1(14h) */ - snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - /* Smart 5.1 PW2(1bh) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1b, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW7 (23h), SW3 (27h), AOW3 (25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x23, &parm); - /* Smart 5.1 PW1(1ah) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* Smart 5.1 PW5(1eh) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1e, &parm); - snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* Mono out */ - /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ - present = snd_hda_jack_detect(codec, 0x1c); - if (present) - mono_out = 0; - else { - present = snd_hda_jack_detect(codec, 0x1d); - if (!spec->hp_independent_mode && present) - mono_out = 0; - else - mono_out = 1; - } - parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; - snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW 3/4 (1ch/1dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - /* HP Independent Mode, power on AOW3 */ - if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* force to D0 for internal Speaker */ - /* MW0 (16h), AOW0 (10h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - mono_out ? AC_PWRST_D0 : parm); - } else if (spec->codec_type == VT2002P) { - unsigned int present; - /* MUX9 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* outputs */ - /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - - /* PW4 (26h), MW4 (1ch), MUX4(37h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x37, - 0, AC_VERB_SET_POWER_STATE, parm); - - /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); - if (spec->hp_independent_mode) { - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* Class-D */ - /* PW0 (24h), MW0(18h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (present) { - snd_hda_codec_write( - codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write( - codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else { - snd_hda_codec_write( - codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write( - codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - } - - /* Mono Out */ - /* PW15 (31h), MW8(17h), MUX8(3bh) */ - present = snd_hda_jack_detect(codec, 0x26); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x31, &parm); - if (present) { - snd_hda_codec_write( - codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write( - codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else { - snd_hda_codec_write( - codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write( - codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - } - - /* MW9 (21h) */ - if (imux_is_smixer || !is_aa_path_mute(codec)) - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - else - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else if (spec->codec_type == VT1812) { - unsigned int present; - /* MUX10 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* outputs */ - /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - - /* PW4 (28h), MW4 (18h), MUX4(38h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x38, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); - if (spec->hp_independent_mode) { - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* Internal Speaker */ - /* PW0 (24h), MW0(14h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (present) { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - } else { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - } - /* Mono Out */ - /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ - present = snd_hda_jack_detect(codec, 0x28); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x31, &parm); - if (present) { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - } else { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - } - - /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x33, &parm); - snd_hda_codec_write(codec, 0x1d, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3d, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* MW9 (21h) */ - if (imux_is_smixer || !is_aa_path_mute(codec)) - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - else - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } -} - /* * input MUX handling */ @@ -1110,7 +642,7 @@ snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); /* update jack power state */ - set_jack_power_state(codec); + set_widgets_power_state(codec); return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], @@ -1417,7 +949,7 @@ } } spec->smart51_enabled = *ucontrol->value.integer.value; - set_jack_power_state(codec); + set_widgets_power_state(codec); return 1; } @@ -1941,7 +1473,7 @@ } /* init power states */ - set_jack_power_state(codec); + set_widgets_power_state(codec); analog_low_current_mode(codec, 1); via_free_kctls(codec); /* no longer needed */ @@ -2169,7 +1701,7 @@ if (res & VIA_GPIO_EVENT) via_gpio_control(codec); if (res & VIA_JACK_EVENT) - set_jack_power_state(codec); + set_widgets_power_state(codec); if (res & VIA_MONO_EVENT) via_mono_automute(codec); if (res & VIA_SPEAKER_EVENT) @@ -3613,6 +3145,75 @@ { } /* end */ }; #endif + +static void set_widgets_power_state_vt1708B(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + int is_8ch = 0; + if ((spec->codec_type != VT1708B_4CH) && + (codec->vendor_id != 0x11064397)) + is_8ch = 1; + + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = + (snd_hda_codec_read(codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) + == ((spec->codec_type == VT1708S) ? 5 : 0)); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW 0/1 (13h/14h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW6 (22h), SW2 (26h), AOW2 (24h) */ + if (is_8ch) { + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x22, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x24, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* PW 3/4/7 (1ch/1dh/23h) */ + parm = AC_PWRST_D3; + /* force to D0 for internal Speaker */ + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + if (is_8ch) + set_pin_power_state(codec, 0x23, &parm); + + /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + if (is_8ch) { + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + } +} + static int patch_vt1708S(struct hda_codec *codec); static int patch_vt1708B_8ch(struct hda_codec *codec) { @@ -3663,6 +3264,8 @@ spec->loopback.amplist = vt1708B_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; + return 0; } @@ -3713,6 +3316,8 @@ spec->loopback.amplist = vt1708B_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; + return 0; } @@ -4147,6 +3752,7 @@ spec->stream_name_analog = "VT1708BCE Analog"; spec->stream_name_digital = "VT1708BCE Digital"; } + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; return 0; } @@ -4404,6 +4010,37 @@ }; #endif +static void set_widgets_power_state_vt1702(struct hda_codec *codec) +{ + int imux_is_smixer = + snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + unsigned int parm; + /* inputs */ + /* PW 1/2/5 (14h/15h/18h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x14, &parm); + set_pin_power_state(codec, 0x15, &parm); + set_pin_power_state(codec, 0x18, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */ + /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW 3/4 (16h/17h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x17, &parm); + set_pin_power_state(codec, 0x16, &parm); + /* MW0 (1ah), AOW 0/1 (10h/1dh) */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); +} + static int patch_vt1702(struct hda_codec *codec) { struct via_spec *spec; @@ -4450,6 +4087,7 @@ spec->loopback.amplist = vt1702_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1702; return 0; } @@ -4791,6 +4429,73 @@ }; #endif +static void set_widgets_power_state_vt1718S(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer = + snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + unsigned int parm; + /* MUX6 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x27, &parm); + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW2 (26h), AOW2 (ah) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x2b, &parm); + snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW0 (24h), AOW0 (8h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (!spec->hp_independent_mode) /* check for redirected HP */ + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + + /* PW1 (25h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x2a, &parm); + snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, parm); + + if (spec->hp_independent_mode) { + /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xc, 0, + AC_VERB_SET_POWER_STATE, parm); + } +} + static int patch_vt1718S(struct hda_codec *codec) { struct via_spec *spec; @@ -4852,6 +4557,8 @@ spec->loopback.amplist = vt1718S_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1718S; + return 0; } @@ -4891,8 +4598,7 @@ snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_CONNECT_SEL, index); spec->dmic_enabled = index; - set_jack_power_state(codec); - + set_widgets_power_state(codec); return 1; } @@ -5251,6 +4957,99 @@ }; #endif +static void set_widgets_power_state_vt1716S(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + unsigned int mono_out, present; + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = + (snd_hda_codec_read(codec, 0x17, 0, + AC_VERB_GET_CONNECT_SEL, 0x00) == 5); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW0(13h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1e, &parm); + /* PW11 (22h) */ + if (spec->dmic_enabled) + set_pin_power_state(codec, 0x22, &parm); + else + snd_hda_codec_write(codec, 0x22, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + /* SW2(26h), AIW1(14h) */ + snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + /* Smart 5.1 PW2(1bh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW7 (23h), SW3 (27h), AOW3 (25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x23, &parm); + /* Smart 5.1 PW1(1ah) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Smart 5.1 PW5(1eh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1e, &parm); + snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Mono out */ + /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ + present = snd_hda_jack_detect(codec, 0x1c); + + if (present) + mono_out = 0; + else { + present = snd_hda_jack_detect(codec, 0x1d); + if (!spec->hp_independent_mode && present) + mono_out = 0; + else + mono_out = 1; + } + parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; + snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW 3/4 (1ch/1dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + /* HP Independent Mode, power on AOW3 */ + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* force to D0 for internal Speaker */ + /* MW0 (16h), AOW0 (10h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + mono_out ? AC_PWRST_D0 : parm); +} + static int patch_vt1716S(struct hda_codec *codec) { struct via_spec *spec; @@ -5305,6 +5104,7 @@ spec->loopback.amplist = vt1716S_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1716S; return 0; } @@ -5575,6 +5375,85 @@ }; #endif +static void set_widgets_power_state_vt2002P(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + unsigned int present; + /* MUX9 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + parm = AC_PWRST_D0; + /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW4 (26h), MW4 (1ch), MUX4(37h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x37, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* Class-D */ + /* PW0 (24h), MW0(18h/14h), MUX0(34h) */ + present = snd_hda_jack_detect(codec, 0x25); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + parm = present ? AC_PWRST_D3 : AC_PWRST_D0; + + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Mono Out */ + present = snd_hda_jack_detect(codec, 0x26); + + parm = present ? AC_PWRST_D3 : AC_PWRST_D0; + + /* PW15 (31h), MW8(17h), MUX8(3bh) */ + snd_hda_codec_write(codec, 0x31, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); +} /* patch for vt2002P */ static int patch_vt2002P(struct hda_codec *codec) @@ -5626,6 +5505,7 @@ spec->loopback.amplist = vt2002P_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt2002P; return 0; } @@ -5897,6 +5777,97 @@ }; #endif +static void set_widgets_power_state_vt1812(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer = + snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + unsigned int parm; + unsigned int present; + /* MUX10 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + parm = AC_PWRST_D0; + /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (28h), MW4 (18h), MUX4(38h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x38, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* Internal Speaker */ + /* PW0 (24h), MW0(14h), MUX0(34h) */ + present = snd_hda_jack_detect(codec, 0x25); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + + /* Mono Out */ + /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ + present = snd_hda_jack_detect(codec, 0x28); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x33, &parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3d, 0, AC_VERB_SET_POWER_STATE, parm); + +} /* patch for vt1812 */ static int patch_vt1812(struct hda_codec *codec) @@ -5950,6 +5921,7 @@ spec->loopback.amplist = vt1812_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1812; return 0; }

1 0

[alsa-devel] [PATCH 0/8] ALSA: VIA HDA: patch series
by Lydia Wang 22 Mar '11

22 Mar '11

Following patch series mainly includes: -add new power management functions of different codecs -fix stereo mixer recording no sound issue -fix independent headphone no sound issue -fix invalid a-a path volume adjust issue for some codecs -fix side channel has sound under independent hp mode of VT1718S issue -correct display stream names for VT1818S -fix hp mixer build fail of VT1708A issue

1 0

[alsa-devel] [PATCH RFC] arm: mini2440: Fix uda134x codec problem.
by Marek Belisko 22 Mar '11

22 Mar '11

ASoC audio for mini2440 platform in current kenrel doesn't work. First problem is samsung_asoc_dma device is missing in initialization. Next problem is with codec. Codec is initialized but never probed because no platform_device exist for codec driver. It leads to errors during codec binding to asoc dai. Next problem was platform data which was passed from board to asoc main driver but not passed to codec when called codec_soc_probe(). Following patch should fix issues. But not sure if in correct way. Please review. Signed-off-by: Marek Belisko <marek.belisko(a)open-nandra.com> --- arch/arm/mach-s3c2440/mach-mini2440.c | 7 +++++++ sound/soc/codecs/uda134x.c | 3 ++- sound/soc/samsung/s3c24xx_uda134x.c | 3 ++- 3 files changed, 11 insertions(+), 2 deletions(-) diff --git a/arch/arm/mach-s3c2440/mach-mini2440.c b/arch/arm/mach-s3c2440/mach-mini2440.c index f62bb4c..7c3fb07 100644 --- a/arch/arm/mach-s3c2440/mach-mini2440.c +++ b/arch/arm/mach-s3c2440/mach-mini2440.c @@ -506,6 +506,11 @@ static struct i2c_board_info mini2440_i2c_devs[] __initdata = { }, }; +static struct platform_device uda1340_codec = { + .name = "uda134x-codec", + .id = -1, +}; + static struct platform_device *mini2440_devices[] __initdata = { &s3c_device_ohci, &s3c_device_wdt, @@ -521,7 +526,9 @@ static struct platform_device *mini2440_devices[] __initdata = { &s3c_device_nand, &s3c_device_sdi, &s3c_device_iis, + &uda1340_codec, &mini2440_audio, + &samsung_asoc_dma, }; static void __init mini2440_map_io(void) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index e76847a..48ffd40 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -486,7 +486,8 @@ static struct snd_soc_dai_driver uda134x_dai = { static int uda134x_soc_probe(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x; - struct uda134x_platform_data *pd = dev_get_drvdata(codec->card->dev); + struct uda134x_platform_data *pd = codec->card->dev->platform_data; + int ret; printk(KERN_INFO "UDA134X SoC Audio Codec\n"); diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 2c09e93..86f1dc4 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -226,7 +226,7 @@ static struct snd_soc_ops s3c24xx_uda134x_ops = { static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { .name = "UDA134X", .stream_name = "UDA134X", - .codec_name = "uda134x-hifi", + .codec_name = "uda134x-codec", .codec_dai_name = "uda134x-hifi", .cpu_dai_name = "s3c24xx-iis", .ops = &s3c24xx_uda134x_ops, @@ -321,6 +321,7 @@ static int s3c24xx_uda134x_probe(struct platform_device *pdev) platform_set_drvdata(s3c24xx_uda134x_snd_device, &snd_soc_s3c24xx_uda134x); + platform_device_add_data(s3c24xx_uda134x_snd_device, &s3c24xx_uda134x, sizeof(s3c24xx_uda134x)); ret = platform_device_add(s3c24xx_uda134x_snd_device); if (ret) { printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n"); -- 1.7.1

5 5

Re: [alsa-devel] [PATCH 1/16] ALSA: VIA HDA - suggesting additional patch
by Mark Goldstein 21 Mar '11

21 Mar '11

Hello, I do not want to hijack the thread, but since it looks like many changes are going to be applied to VIA HDA, I'd like to ask you to look into potential patch that fixed my problem with VIA VT1708S. There was a long thread "Problem with VIA VT1708S and git versions of alsa-driver" on this list. I'll give a brief summary: I have ASUS P5QL/EPU motherboard with VIA VT1708S on-board audio. My distribution is openSUSE. The sound worked satisfactory with alsa driver 1.0.23. The problem occurred first in oS 11.1. After one of the updates from git snapshot repo (end of September / beginning of October 2010), the functionality of controls got messed up: Line control did not work, Front Mic control actually changed the volume of Line In. Front Mic itself did not work at all. I tried to figure out what happened with the help of Raymond Yau, but at some stage gave up and reverted to working driver 1.0.23. Later this year I had to try git version again (because I've got new USB device). Still the same problem. So I decided to compare patch_via.c from 1.0.23 that worked and from git version (and now 1.0.24) that did not. It turned out that the cause of my problem was in the patch (identified by Raimond) http://git.alsa-project.org/?p=alsa-kernel.git;a=commitdiff;h=f3268512c3a5d… This patch introduced common function vt_auto_create_analog_input_ctls and codec-specific functions passing the array pin_idxs to it. The problem, as I understood it, is that vt_auto_create_analog_input_ctls uses the same pin index in calls to via_new_analog_input and snd_hda_add_imux_item, while in the old code (1.0.23) for some codecs, including VT1708S, VT1702 and VT1716S, via_new_analog_input was called with pin index + 1. I do not understand the details, so I just did straightforward patch (attached here, based on the code of 1.0.24) to restore the old behavior. In my case this patch resolved the problem and I have working controls now. Could you please check it? Maybe it should be done differently, but the problem is real (at least for me) and needs a fix. Regards, -- Mark Goldstein

1 0

[alsa-devel] ALSA issue on DA850/OMAP-L138/AM18x
by Rajashekhara, Sudhakar 21 Mar '11

21 Mar '11

Resending with proper $SUBJECT... Hi, I was testing Audio with 2.6.37 on DA850 from Kevin Hilman Linux tree at [1] and found that audio is broken. Below patch fixes the issue. --- From: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> davinci: fixes for audio on da850/omap-l138/am18x On DA850/OMAP-L138/AM18x, AIC3x codec is at 0x18 slave address. But in sound/soc/davinci/davinci-evm.c file, "struct snd_soc_dai_link" has the wrong AIC3x codec slave address. This patch fixes this issue. Also, this patch registers the platform device for davinci-pcm-audio. Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> --- arch/arm/mach-davinci/devices-da8xx.c | 12 ++++++++++++ sound/soc/davinci/davinci-evm.c | 2 +- 2 files changed, 13 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c index 9eec630..17c0dbc 100644 --- a/arch/arm/mach-davinci/devices-da8xx.c +++ b/arch/arm/mach-davinci/devices-da8xx.c @@ -473,6 +473,11 @@ static struct resource da850_mcasp_resources[] = { }, }; +struct platform_device davinci_pcm_device = { + .name = "davinci-pcm-audio", + .id = -1, +}; + static struct platform_device da850_mcasp_device = { .name = "davinci-mcasp", .id = 0, @@ -480,8 +485,15 @@ static struct platform_device da850_mcasp_device = { .resource = da850_mcasp_resources, }; +static void davinci_init_pcm(void) +{ + platform_device_register(&davinci_pcm_device); +} + void __init da8xx_register_mcasp(int id, struct snd_platform_data *pdata) { + davinci_init_pcm(); + /* DA830/OMAP-L137 has 3 instances of McASP */ if (cpu_is_davinci_da830() && id == 1) { da830_mcasp1_device.dev.platform_data = pdata; diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index bc9e6b0..07db881 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -224,7 +224,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", - .codec_name = "tlv320aic3x-codec.0-001a", + .codec_name = "tlv320aic3x-codec.1-0018", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, --- Also, I found that either CONFIG_REGULATOR should not be defined or if CONFIG_REGULATOR is defined then CONFIG_REGULATOR_DUMMY should also be defined. Without this menuconfig fix, Soundcard does not get detected. With the above fixes, arecord and aplay does not work for the first time. Couple of times I get the below error: root@da850-omapl138-evm:~# arecord -f dat | aplay -f dat arecord: main:608: audio open error: Invalid argument aplay: playback:2297: read error root@da850-omapl138-evm:~# arecord -f dat | aplay -f dat aplay: main:608: audio open error: Invalid argument Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Third time arecord and aplay work normally. Has anyone seen such issues on DA850 or any other platform? I am currently debugging this issue. I'll submit the above patch to community once the issue of fixed. [1] http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git;a=summa… Regards, Sudhakar _______________________________________________ Davinci-linux-open-source mailing list Davinci-linux-open-source(a)linux.davincidsp.com http://linux.davincidsp.com/mailman/listinfo/davinci-linux-open-source

5 11

[alsa-devel] [PATCH 16/16] ALSA: VIA HDA: Update VT2002P initial verbs
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Update VT2002P initial verbs. Add some hardware related verbs in VT2002P initial verbs. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 4 ++++ 1 file changed, 4 insertions(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -5375,6 +5375,10 @@ }; static struct hda_verb vt2002P_volume_init_verbs[] = { + /* Hardware verbs */ + {0x1, 0xfe0, 0x4}, + {0x1, 0xfe9, 0x80}, + {0x1, 0xfe2, 0x22}, /* * Unmute ADC0-1 and set the default input to mic-in */

2 1

[alsa-devel] [PATCH 14/16] ALSA: VIA HDA: Update VT1708 initial verbs
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Update VT1708 initial verbs. Add a verb of power down jack detect function in VT1708 initial verbs. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 2 ++ 1 file changed, 2 insertions(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1205,6 +1205,8 @@ {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* power down jack detect function */ + {0x1, 0xf81, 0x1}, { } };

2 1

[alsa-devel] [PATCH 10/16] ALSA: VIA HDA: Add support for VT1705
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add support for VT1705. Add support for VT1705 codec, which is similiar with VT1708S. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 179 +++++++++++++++++++++++++++++-- 1 file changed, 172 insertions(+), 7 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3253,6 +3253,16 @@ AC_VERB_SET_POWER_STATE, parm); snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_POWER_STATE, parm); + } else if (codec->vendor_id == 0x11064397) { + /* PW7(23h), SW2(27h), AOW2(25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x23, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); } /* PW 3/4/7 (1ch/1dh/23h) */ @@ -3272,7 +3282,9 @@ AC_VERB_SET_POWER_STATE, parm); snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); - } + } else if (codec->vendor_id == 0x11064397 && spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); } static int patch_vt1708S(struct hda_codec *codec); @@ -3447,6 +3459,18 @@ { } }; +static struct hda_verb vt1705_uniwill_init_verbs[] = { + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + static struct hda_pcm_stream vt1708S_pcm_analog_playback = { .substreams = 2, .channels_min = 2, @@ -3460,6 +3484,19 @@ }, }; +static struct hda_pcm_stream vt1705_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 6, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close + }, +}; + static struct hda_pcm_stream vt1708S_pcm_analog_capture = { .substreams = 2, .channels_min = 2, @@ -3506,7 +3543,10 @@ spec->multiout.dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: - spec->multiout.dac_nids[i] = 0x24; + if (spec->codec->vendor_id == 0x11064397) + spec->multiout.dac_nids[i] = 0x25; + else + spec->multiout.dac_nids[i] = 0x24; break; case AUTO_SEQ_SURROUND: spec->multiout.dac_nids[i] = 0x11; @@ -3522,7 +3562,10 @@ if (cfg->line_outs == 1) { spec->multiout.num_dacs = 3; spec->multiout.dac_nids[AUTO_SEQ_SURROUND] = 0x11; - spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x24; + if (spec->codec->vendor_id == 0x11064397) + spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x25; + else + spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x24; } return 0; @@ -3628,6 +3671,104 @@ return 0; } +/* add playback controls from the parsed DAC table */ +static int vt1705_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char * const chname[3] = { "Front", "Surround", "C/LFE" }; + hda_nid_t nid_vols[] = {0x10, 0x11, 0x25}; + hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x27}; + hda_nid_t nid, nid_vol, nid_mute; + int i, err; + + for (i = 0; i < AUTO_SEQ_SIDE; i++) { + nid = cfg->line_out_pins[i]; + + /* for Smart 5.1, there are always at least six channels */ + if (!nid && i > AUTO_SEQ_CENLFE) + continue; + + nid_vol = nid_vols[i]; + nid_mute = nid_mutes[i]; + + if (i == AUTO_SEQ_CENLFE) { + /* Center/LFE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, + 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, + 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (i == AUTO_SEQ_FRONT) { + /* add control to mixer index 0 */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, + HDA_INPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, + HDA_INPUT)); + if (err < 0) + return err; + + /* Front */ + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, + 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, + 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + + return 0; +} + static int vt1708S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) { int err; @@ -3703,7 +3844,10 @@ if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) return 0; /* can't find valid BIOS pin config */ - err = vt1708S_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (codec->vendor_id == 0x11064397) + err = vt1705_auto_create_multi_out_ctls(spec, &spec->autocfg); + else + err = vt1708S_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; err = vt1708S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); @@ -3770,17 +3914,30 @@ } spec->init_verbs[spec->num_iverbs++] = vt1708S_volume_init_verbs; - spec->init_verbs[spec->num_iverbs++] = vt1708S_uniwill_init_verbs; + if (codec->vendor_id == 0x11064397) + spec->init_verbs[spec->num_iverbs++] = + vt1705_uniwill_init_verbs; + else + spec->init_verbs[spec->num_iverbs++] = + vt1708S_uniwill_init_verbs; if (codec->vendor_id == 0x11060440) spec->stream_name_analog = "VT1818S Analog"; + else if (codec->vendor_id == 0x11064397) + spec->stream_name_analog = "VT1705 Analog"; else spec->stream_name_analog = "VT1708S Analog"; - spec->stream_analog_playback = &vt1708S_pcm_analog_playback; + + if (codec->vendor_id == 0x11064397) + spec->stream_analog_playback = &vt1705_pcm_analog_playback; + else + spec->stream_analog_playback = &vt1708S_pcm_analog_playback; spec->stream_analog_capture = &vt1708S_pcm_analog_capture; if (codec->vendor_id == 0x11060440) spec->stream_name_digital = "VT1818S Digital"; + else if (codec->vendor_id == 0x11064397) + spec->stream_name_digital = "VT1705 Digital"; else spec->stream_name_digital = "VT1708S Digital"; spec->stream_digital_playback = &vt1708S_pcm_digital_playback; @@ -3813,6 +3970,14 @@ spec->stream_name_analog = "VT1708BCE Analog"; spec->stream_name_digital = "VT1708BCE Digital"; } + /* correct names for VT1705 */ + if (codec->vendor_id == 0x11064397) { + kfree(codec->chip_name); + codec->chip_name = kstrdup("VT1705", GFP_KERNEL); + snprintf(codec->bus->card->mixername, + sizeof(codec->bus->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); + } spec->set_widgets_power_state = set_widgets_power_state_vt1708B; return 0; } @@ -6164,7 +6329,7 @@ .patch = patch_vt1708S}, { .id = 0x11063397, .name = "VT1708S", .patch = patch_vt1708S}, - { .id = 0x11064397, .name = "VT1708S", + { .id = 0x11064397, .name = "VT1705", .patch = patch_vt1708S}, { .id = 0x11065397, .name = "VT1708S", .patch = patch_vt1708S},

2 1

[alsa-devel] [PATCH 6/16] ALSA: VIA HDA: Fix independent headphone no sound issue
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix independent headphone no sound issue. Modify via_independent_hp_put() function to support VT1718S and VT1812 codecs, and fix independent headphone no sound issue. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -740,8 +740,15 @@ /* Get Independent Mode index of headphone pin widget */ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); - + if (spec->codec_type == VT1718S) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0); + else + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, pinsel); + if (spec->codec_type == VT1812) + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_CONNECT_SEL, pinsel); if (spec->multiout.hp_nid && spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, @@ -761,6 +768,8 @@ activate_ctl(codec, "Headphone Playback Switch", spec->hp_independent_mode); } + /* update jack power state */ + set_widgets_power_state(codec); return 0; }

2 1

[alsa-devel] [PATCH 4/16] ALSA: VIA HDA: Fix stereo mixer recording no sound issue
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix stereo mixer recording no sound issue. Modify function via_mux_enum_put() to fix stereo mixer recording no sound issue. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -636,6 +636,7 @@ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + int ret; if (!spec->mux_nids[adc_idx]) return -EINVAL; @@ -644,12 +645,14 @@ AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - /* update jack power state */ - set_widgets_power_state(codec); - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + ret = snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); + /* update jack power state */ + set_widgets_power_state(codec); + + return ret; } static int via_independent_hp_info(struct snd_kcontrol *kcontrol,

2 1

[alsa-devel] [PATCH 2/16] ALSA: VIA HDA: Add support for VT1802
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add support for VT1802. Add support for VT1802 codec, which is similiar with VT2002P. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> sound/pci/hda/patch_via.c | 208 +++++++++++++++++++++++++------ 1 file changed, 174 insertions(+), 34 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -98,6 +98,7 @@ VT1716S, VT2002P, VT1812, + VT1802, CODEC_TYPES, }; @@ -216,6 +217,8 @@ codec_type = VT1812; else if (dev_id == 0x0440) codec_type = VT1708S; + else if ((dev_id & 0xfff) == 0x446) + codec_type = VT1802; else codec_type = UNKNOWN; return codec_type; @@ -734,7 +737,8 @@ || spec->codec_type == VT1718S || spec->codec_type == VT1716S || spec->codec_type == VT2002P - || spec->codec_type == VT1812) { + || spec->codec_type == VT1812 + || spec->codec_type == VT1802) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -770,6 +774,7 @@ nid = 0x34; break; case VT2002P: + case VT1802: nid = 0x35; break; case VT1812: @@ -1045,6 +1050,7 @@ break; case VT2002P: case VT1812: + case VT1802: nid_mixer = 0x21; start_idx = 0; end_idx = 2; @@ -1109,6 +1115,7 @@ break; case VT2002P: case VT1812: + case VT1802: verb = 0xf93; parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; @@ -1637,7 +1644,9 @@ unsigned int hp_present; struct via_spec *spec = codec->spec; - if (spec->codec_type != VT2002P && spec->codec_type != VT1812) + if (spec->codec_type != VT2002P && + spec->codec_type != VT1812 && + spec->codec_type != VT1802) return; hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]); @@ -2122,7 +2131,10 @@ via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_analog_input(codec); - if (spec->codec_type == VT2002P || spec->codec_type == VT1812) { + + if (spec->codec_type == VT2002P || + spec->codec_type == VT1812 || + spec->codec_type == VT1802) { via_hp_bind_automute(codec); } else { via_hp_automute(codec); @@ -5187,6 +5199,67 @@ {0x1, 0xfb8, 0x88}, { } }; +static struct hda_verb vt1802_volume_init_verbs[] = { + /* Special verbs */ + /*{0x1, 0xf93, 0xc3}, + {0x1, 0xfa2, 0x70}, + {0x1, 0xfe0, 0x04}, + {0x1, 0xfe3, 0xfc}, + {0x1, 0xff0, 0x39}, + {0x1, 0xfa1, 0x58}, + {0x1, 0xfd9, 0x0b}, + {0x1, 0xfdb, 0x20}, + {0x1, 0xfb0, 0x2b},*/ + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Mic = 0 */ + {0x1e, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0}, + + /* PW9 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + + /* Enable Boost Volume backdoor */ + {0x1, 0xfb9, 0x24}, + + /* MW0/1/4/8: un-mute index 0 (MUXx), un-mute index 1 (MW9) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* set MUX0/1/4/8 = 0 (AOW0) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0}, + {0x38, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3c, AC_VERB_SET_CONNECT_SEL, 0}, + + /* set PW0 index=0 (MW0) */ + {0x24, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Enable AOW0 to MW9 */ + {0x1, 0xfb8, 0x88}, + { } +}; static struct hda_verb vt2002P_uniwill_init_verbs[] = { @@ -5199,6 +5272,16 @@ {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; +static struct hda_verb vt1802_uniwill_init_verbs[] = { + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; static struct hda_pcm_stream vt2002P_pcm_analog_playback = { .substreams = 2, @@ -5255,10 +5338,15 @@ const struct auto_pin_cfg *cfg) { int err; + hda_nid_t sw_nid; if (!cfg->line_out_pins[0]) return -1; + if (spec->codec_type == VT1802) + sw_nid = 0x28; + else + sw_nid = 0x26; /* Line-Out: PortE */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, @@ -5268,7 +5356,7 @@ return err; err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x26, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(sw_nid, 3, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -5403,22 +5491,41 @@ /* AOW0 (8h)*/ snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); - /* PW4 (26h), MW4 (1ch), MUX4(37h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x37, 0, - AC_VERB_SET_POWER_STATE, parm); - + if (spec->codec_type == VT1802) { + /* PW4 (28h), MW4 (18h), MUX4(38h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x38, 0, + AC_VERB_SET_POWER_STATE, parm); + } else { + /* PW4 (26h), MW4 (1ch), MUX4(37h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x37, 0, + AC_VERB_SET_POWER_STATE, parm); + } - /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); + if (spec->codec_type == VT1802) { + /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x15, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + } else { + /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + } if (spec->hp_independent_mode) snd_hda_codec_write(codec, 0x9, 0, @@ -5432,8 +5539,12 @@ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x24, &parm); parm = present ? AC_PWRST_D3 : AC_PWRST_D0; - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); + if (spec->codec_type == VT1802) + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, parm); + else + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm); /* Mono Out */ @@ -5441,15 +5552,23 @@ snd_hda_codec_read(codec, 0x26, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; parm = present ? AC_PWRST_D3 : AC_PWRST_D0; - - /* PW15 (31h), MW8(17h), MUX8(3bh) */ - snd_hda_codec_write(codec, 0x31, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, parm); - + if (spec->codec_type == VT1802) { + /* PW15 (33h), MW8(1ch), MUX8(3ch) */ + snd_hda_codec_write(codec, 0x33, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, parm); + } else { + /* PW15 (31h), MW8(17h), MUX8(3bh) */ + snd_hda_codec_write(codec, 0x31, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, parm); + } /* MW9 (21h) */ if (imux_is_smixer || !is_aa_path_mute(codec)) snd_hda_codec_write(codec, 0x21, 0, @@ -5480,14 +5599,31 @@ "from BIOS. Using genenic mode...\n"); } - spec->init_verbs[spec->num_iverbs++] = vt2002P_volume_init_verbs; - spec->init_verbs[spec->num_iverbs++] = vt2002P_uniwill_init_verbs; + if (spec->codec_type == VT1802) + spec->init_verbs[spec->num_iverbs++] = + vt1802_volume_init_verbs; + else + spec->init_verbs[spec->num_iverbs++] = + vt2002P_volume_init_verbs; + + if (spec->codec_type == VT1802) + spec->init_verbs[spec->num_iverbs++] = + vt1802_uniwill_init_verbs; + else + spec->init_verbs[spec->num_iverbs++] = + vt2002P_uniwill_init_verbs; - spec->stream_name_analog = "VT2002P Analog"; + if (spec->codec_type == VT1802) + spec->stream_name_analog = "VT1802 Analog"; + else + spec->stream_name_analog = "VT2002P Analog"; spec->stream_analog_playback = &vt2002P_pcm_analog_playback; spec->stream_analog_capture = &vt2002P_pcm_analog_capture; - spec->stream_name_digital = "VT2002P Digital"; + if (spec->codec_type == VT1802) + spec->stream_name_digital = "VT1802 Digital"; + else + spec->stream_name_digital = "VT2002P Digital"; spec->stream_digital_playback = &vt2002P_pcm_digital_playback; if (!spec->adc_nids && spec->input_mux) { @@ -6020,6 +6156,10 @@ { .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812}, { .id = 0x11060440, .name = "VT1818S", .patch = patch_vt1708S}, + { .id = 0x11060446, .name = "VT1802", + .patch = patch_vt2002P}, + { .id = 0x11068446, .name = "VT1802", + .patch = patch_vt2002P}, {} /* terminator */ };

2 1

[alsa-devel] [PATCH 1/16] ALSA: VIA HDA: Add new power management function
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add new power management function. Use set_widgets_power_state() function to seperately control different codecs' power management actions. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 946 +++++++++++++++---------------- 1 file changed, 462 insertions(+), 484 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -154,6 +154,9 @@ struct delayed_work vt1708_hp_work; int vt1708_jack_detectect; int vt1708_hp_present; + + void (*set_widgets_power_state)(struct hda_codec *codec); + #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -240,7 +243,6 @@ }; static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); -static void set_jack_power_state(struct hda_codec *codec); static int is_aa_path_mute(struct hda_codec *codec); static void vt1708_start_hp_work(struct via_spec *spec) @@ -266,6 +268,12 @@ cancel_delayed_work_sync(&spec->vt1708_hp_work); } +static void set_widgets_power_state(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + if (spec->set_widgets_power_state) + spec->set_widgets_power_state(codec); +} static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -273,7 +281,7 @@ int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - set_jack_power_state(codec); + set_widgets_power_state(codec); analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { if (is_aa_path_mute(codec)) @@ -597,482 +605,6 @@ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); } -static void set_jack_power_state(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - - if (spec->codec_type == VT1702) { - imux_is_smixer = snd_hda_codec_read( - codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - /* inputs */ - /* PW 1/2/5 (14h/15h/18h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x14, &parm); - set_pin_power_state(codec, 0x15, &parm); - set_pin_power_state(codec, 0x18, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; /* SW0 = stereo mixer (idx 3) */ - /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW 3/4 (16h/17h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x16, &parm); - set_pin_power_state(codec, 0x17, &parm); - /* MW0 (1ah), AOW 0/1 (10h/1dh) */ - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, - parm); - } else if (spec->codec_type == VT1708B_8CH - || spec->codec_type == VT1708B_4CH - || spec->codec_type == VT1708S) { - /* SW0 (17h) = stereo mixer */ - int is_8ch = spec->codec_type != VT1708B_4CH; - imux_is_smixer = snd_hda_codec_read( - codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) - == ((spec->codec_type == VT1708S) ? 5 : 0); - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW 0/1 (13h/14h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - if (spec->smart51_enabled) - parm = AC_PWRST_D0; - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW6 (22h), SW2 (26h), AOW2 (24h) */ - if (is_8ch) { - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x22, &parm); - if (spec->smart51_enabled) - parm = AC_PWRST_D0; - snd_hda_codec_write(codec, 0x26, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x24, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* PW 3/4/7 (1ch/1dh/23h) */ - parm = AC_PWRST_D3; - /* force to D0 for internal Speaker */ - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - if (is_8ch) - set_pin_power_state(codec, 0x23, &parm); - /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - if (is_8ch) { - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x27, 0, - AC_VERB_SET_POWER_STATE, parm); - } - } else if (spec->codec_type == VT1718S) { - /* MUX6 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x27, &parm); - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW2 (26h), AOW2 (ah) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW0/1 (24h/25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - set_pin_power_state(codec, 0x25, &parm); - if (!spec->hp_independent_mode) /* check for redirected HP */ - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, - parm); - /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - if (spec->hp_independent_mode) { - /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x1b, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0xc, 0, - AC_VERB_SET_POWER_STATE, parm); - } - } else if (spec->codec_type == VT1716S) { - unsigned int mono_out, present; - /* SW0 (17h) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW0(13h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1e, &parm); - /* PW11 (22h) */ - if (spec->dmic_enabled) - set_pin_power_state(codec, 0x22, &parm); - else - snd_hda_codec_write( - codec, 0x22, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - - /* SW2(26h), AIW1(14h) */ - snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - /* Smart 5.1 PW2(1bh) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1b, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW7 (23h), SW3 (27h), AOW3 (25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x23, &parm); - /* Smart 5.1 PW1(1ah) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* Smart 5.1 PW5(1eh) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1e, &parm); - snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* Mono out */ - /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ - present = snd_hda_jack_detect(codec, 0x1c); - if (present) - mono_out = 0; - else { - present = snd_hda_jack_detect(codec, 0x1d); - if (!spec->hp_independent_mode && present) - mono_out = 0; - else - mono_out = 1; - } - parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; - snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW 3/4 (1ch/1dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - /* HP Independent Mode, power on AOW3 */ - if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* force to D0 for internal Speaker */ - /* MW0 (16h), AOW0 (10h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - mono_out ? AC_PWRST_D0 : parm); - } else if (spec->codec_type == VT2002P) { - unsigned int present; - /* MUX9 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* outputs */ - /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - - /* PW4 (26h), MW4 (1ch), MUX4(37h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x37, - 0, AC_VERB_SET_POWER_STATE, parm); - - /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); - if (spec->hp_independent_mode) { - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* Class-D */ - /* PW0 (24h), MW0(18h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (present) { - snd_hda_codec_write( - codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write( - codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else { - snd_hda_codec_write( - codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write( - codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - } - - /* Mono Out */ - /* PW15 (31h), MW8(17h), MUX8(3bh) */ - present = snd_hda_jack_detect(codec, 0x26); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x31, &parm); - if (present) { - snd_hda_codec_write( - codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write( - codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else { - snd_hda_codec_write( - codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write( - codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - } - - /* MW9 (21h) */ - if (imux_is_smixer || !is_aa_path_mute(codec)) - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - else - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else if (spec->codec_type == VT1812) { - unsigned int present; - /* MUX10 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* outputs */ - /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - - /* PW4 (28h), MW4 (18h), MUX4(38h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x38, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); - if (spec->hp_independent_mode) { - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* Internal Speaker */ - /* PW0 (24h), MW0(14h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (present) { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - } else { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - } - /* Mono Out */ - /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ - present = snd_hda_jack_detect(codec, 0x28); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x31, &parm); - if (present) { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - } else { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - } - - /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x33, &parm); - snd_hda_codec_write(codec, 0x1d, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3d, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* MW9 (21h) */ - if (imux_is_smixer || !is_aa_path_mute(codec)) - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - else - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } -} - /* * input MUX handling */ @@ -1110,7 +642,7 @@ snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); /* update jack power state */ - set_jack_power_state(codec); + set_widgets_power_state(codec); return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], @@ -1417,7 +949,7 @@ } } spec->smart51_enabled = *ucontrol->value.integer.value; - set_jack_power_state(codec); + set_widgets_power_state(codec); return 1; } @@ -1941,7 +1473,7 @@ } /* init power states */ - set_jack_power_state(codec); + set_widgets_power_state(codec); analog_low_current_mode(codec, 1); via_free_kctls(codec); /* no longer needed */ @@ -2169,7 +1701,7 @@ if (res & VIA_GPIO_EVENT) via_gpio_control(codec); if (res & VIA_JACK_EVENT) - set_jack_power_state(codec); + set_widgets_power_state(codec); if (res & VIA_MONO_EVENT) via_mono_automute(codec); if (res & VIA_SPEAKER_EVENT) @@ -3613,6 +3145,75 @@ { } /* end */ }; #endif + +static void set_widgets_power_state_vt1708B(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + int is_8ch = 0; + if ((spec->codec_type != VT1708B_4CH) && + (codec->vendor_id != 0x11064397)) + is_8ch = 1; + + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = + (snd_hda_codec_read(codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) + == ((spec->codec_type == VT1708S) ? 5 : 0)); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW 0/1 (13h/14h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW6 (22h), SW2 (26h), AOW2 (24h) */ + if (is_8ch) { + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x22, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x24, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* PW 3/4/7 (1ch/1dh/23h) */ + parm = AC_PWRST_D3; + /* force to D0 for internal Speaker */ + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + if (is_8ch) + set_pin_power_state(codec, 0x23, &parm); + + /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + if (is_8ch) { + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + } +} + static int patch_vt1708S(struct hda_codec *codec); static int patch_vt1708B_8ch(struct hda_codec *codec) { @@ -3663,6 +3264,8 @@ spec->loopback.amplist = vt1708B_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; + return 0; } @@ -3713,6 +3316,8 @@ spec->loopback.amplist = vt1708B_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; + return 0; } @@ -4147,6 +3752,7 @@ spec->stream_name_analog = "VT1708BCE Analog"; spec->stream_name_digital = "VT1708BCE Digital"; } + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; return 0; } @@ -4404,6 +4010,37 @@ }; #endif +static void set_widgets_power_state_vt1702(struct hda_codec *codec) +{ + int imux_is_smixer = + snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + unsigned int parm; + /* inputs */ + /* PW 1/2/5 (14h/15h/18h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x14, &parm); + set_pin_power_state(codec, 0x15, &parm); + set_pin_power_state(codec, 0x18, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */ + /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW 3/4 (16h/17h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x17, &parm); + set_pin_power_state(codec, 0x16, &parm); + /* MW0 (1ah), AOW 0/1 (10h/1dh) */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); +} + static int patch_vt1702(struct hda_codec *codec) { struct via_spec *spec; @@ -4450,6 +4087,7 @@ spec->loopback.amplist = vt1702_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1702; return 0; } @@ -4791,6 +4429,73 @@ }; #endif +static void set_widgets_power_state_vt1718S(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer = + snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + unsigned int parm; + /* MUX6 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x27, &parm); + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW2 (26h), AOW2 (ah) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x2b, &parm); + snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW0 (24h), AOW0 (8h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (!spec->hp_independent_mode) /* check for redirected HP */ + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + + /* PW1 (25h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x2a, &parm); + snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, parm); + + if (spec->hp_independent_mode) { + /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xc, 0, + AC_VERB_SET_POWER_STATE, parm); + } +} + static int patch_vt1718S(struct hda_codec *codec) { struct via_spec *spec; @@ -4852,6 +4557,8 @@ spec->loopback.amplist = vt1718S_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1718S; + return 0; } @@ -4891,8 +4598,7 @@ snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_CONNECT_SEL, index); spec->dmic_enabled = index; - set_jack_power_state(codec); - + set_widgets_power_state(codec); return 1; } @@ -5251,6 +4957,101 @@ }; #endif +void set_widgets_power_state_vt1716S(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + unsigned int mono_out, present; + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = + (snd_hda_codec_read(codec, 0x17, 0, + AC_VERB_GET_CONNECT_SEL, 0x00) == 5); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW0(13h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1e, &parm); + /* PW11 (22h) */ + if (spec->dmic_enabled) + set_pin_power_state(codec, 0x22, &parm); + else + snd_hda_codec_write(codec, 0x22, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + /* SW2(26h), AIW1(14h) */ + snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + /* Smart 5.1 PW2(1bh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW7 (23h), SW3 (27h), AOW3 (25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x23, &parm); + /* Smart 5.1 PW1(1ah) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Smart 5.1 PW5(1eh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1e, &parm); + snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Mono out */ + /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ + present = snd_hda_codec_read(codec, 0x1c, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) + mono_out = 0; + else { + present = + snd_hda_codec_read(codec, 0x1d, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (!spec->hp_independent_mode && present) + mono_out = 0; + else + mono_out = 1; + } + parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; + snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW 3/4 (1ch/1dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + /* HP Independent Mode, power on AOW3 */ + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* force to D0 for internal Speaker */ + /* MW0 (16h), AOW0 (10h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + mono_out ? AC_PWRST_D0 : parm); +} + static int patch_vt1716S(struct hda_codec *codec) { struct via_spec *spec; @@ -5305,6 +5106,7 @@ spec->loopback.amplist = vt1716S_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1716S; return 0; } @@ -5575,6 +5377,87 @@ }; #endif +void set_widgets_power_state_vt2002P(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + unsigned int present; + /* MUX9 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + parm = AC_PWRST_D0; + /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW4 (26h), MW4 (1ch), MUX4(37h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x37, 0, + AC_VERB_SET_POWER_STATE, parm); + + + /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* Class-D */ + /* PW0 (24h), MW0(18h/14h), MUX0(34h) */ + present = + snd_hda_codec_read(codec, 0x25, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + parm = present ? AC_PWRST_D3 : AC_PWRST_D0; + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Mono Out */ + present = + snd_hda_codec_read(codec, 0x26, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = present ? AC_PWRST_D3 : AC_PWRST_D0; + + /* PW15 (31h), MW8(17h), MUX8(3bh) */ + snd_hda_codec_write(codec, 0x31, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); +} /* patch for vt2002P */ static int patch_vt2002P(struct hda_codec *codec) @@ -5626,6 +5509,7 @@ spec->loopback.amplist = vt2002P_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt2002P; return 0; } @@ -5897,6 +5781,99 @@ }; #endif +void set_widgets_power_state_vt1812(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer = + snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + unsigned int parm; + unsigned int present; + /* MUX10 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + parm = AC_PWRST_D0; + /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (28h), MW4 (18h), MUX4(38h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x38, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* Internal Speaker */ + /* PW0 (24h), MW0(14h), MUX0(34h) */ + present = + snd_hda_codec_read(codec, 0x25, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + + /* Mono Out */ + /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ + present = + snd_hda_codec_read(codec, 0x28, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x33, &parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3d, 0, AC_VERB_SET_POWER_STATE, parm); + +} /* patch for vt1812 */ static int patch_vt1812(struct hda_codec *codec) @@ -5950,6 +5927,7 @@ spec->loopback.amplist = vt1812_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1812; return 0; }

2 1

[alsa-devel] [PATCH - Set EAPD for Realtek ALC665 (Vendor Id: 0x10eSet EAPD for Realtek ALC665 (Vendor Id: 0x10ec0665).
by Andres Mejia 21 Mar '11

21 Mar '11

Signed-off-by: Andres Mejia <mcitadel(a)gmail.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 50855cf..be910fa6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1264,6 +1264,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: + case 0x10ec0665: case 0x10ec0862: case 0x10ec0889: set_eapd(codec, 0x14, 1); @@ -4236,6 +4237,7 @@ static void alc_power_eapd(struct hda_codec *codec) case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: + case 0x10ec0665: case 0x10ec0862: case 0x10ec0889: set_eapd(codec, 0x14, 0); -- 1.7.4.1

2 2

[alsa-devel] Silent speakers after resume on nx6325
by Adrian Knoth 21 Mar '11

21 Mar '11

Hi! This isn't new, but for a couple of months now, the nx6325 (builtin ATIIXP SB400) sometimes disables the internal speakers after suspend/resume. (the headphones jack works fine) It's not exactly jack sensing, the jack states are correct. I then rmmod snd_hda_intel and modprobe it again, the mute-button-LED shortly flashes and the speakers are back to life. Is this a common problem that can be solved by adding a quirk? Cheers -- mail: adi(a)thur.de http://adi.thur.de PGP/GPG: key via keyserver

1 0

[alsa-devel] [PATCH 7/16] ALSA: VIA HDA: Modify notify_aa_path_ctls() function
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Modify notify_aa_path_ctls() function. Modify notify_aa_path_ctls() function to get volume control items name dynamically instead of by giving static item name. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 27 +++++++++++++++++---------- 1 file changed, 17 insertions(+), 10 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -832,16 +832,23 @@ static void notify_aa_path_ctls(struct hda_codec *codec) { - int i; - struct snd_ctl_elem_id id; - const char *labels[] = {"Mic", "Front Mic", "Line"}; - - memset(&id, 0, sizeof(id)); - id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - for (i = 0; i < ARRAY_SIZE(labels); i++) { - sprintf(id.name, "%s Playback Volume", labels[i]); - snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, - &id); + struct snd_card *card = codec->bus->card; + struct snd_kcontrol *kctl; + struct via_spec *spec = codec->spec; + const struct auto_pin_cfg *cfg = &spec->autocfg; + list_for_each_entry(kctl, &card->controls, list) { + int i; + for (i = 0; i < cfg->num_inputs; i++) { + char name[32]; + const char *label; + label = hda_get_autocfg_input_label(codec, cfg, i); + sprintf(name, "%s Playback Volume", label); + if (strcmp(kctl->id.name, name) == 0) { + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &kctl->id); + break; + } + } } }

3 2

[alsa-devel] Intel HDA: Realtek ALC260 - jack plugging event interface
by Bent Bisballle Nyeng 21 Mar '11

21 Mar '11

Hello I have a FitPC2 with an Realtek ALC260 chip. In the kernel config I have enabled CONFIG_SND_HDA_INPUT_JACK which reads "enable the jack plugging notification via input layer" in the kernel help text. I do however not see any /dev/input/eventX node for the events. Is there another way in which I can read the jack plugging notification events from a C program? I have run til ALSA Iformation script with the following result: http://www.alsa-project.org/db/?f=7f6d755a504980cd0494ee9b87850c1dee7f8f03 Thank you in advance. Bent Bisballe Nyeng

1 0

[alsa-devel] [PATCH] alsa-lib: fixed coverity reported issues under "FORWARD_NULL" checker.
by sudarshan.bisht＠nokia.com 21 Mar '11

21 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> --- modules/mixer/simple/sbasedl.c | 2 +- src/conf.c | 1 + src/hwdep/hwdep.c | 1 + src/pcm/pcm_hooks.c | 1 + src/pcm/pcm_simple.c | 2 +- src/rawmidi/rawmidi.c | 1 + src/rawmidi/rawmidi_virt.c | 8 +++++--- 7 files changed, 11 insertions(+), 5 deletions(-) diff --git a/modules/mixer/simple/sbasedl.c b/modules/mixer/simple/sbasedl.c index 0137586..494802f 100644 --- a/modules/mixer/simple/sbasedl.c +++ b/modules/mixer/simple/sbasedl.c @@ -99,7 +99,7 @@ int mixer_simple_basic_dlopen(snd_mixer_class_t *class, __error: if (initflag) free(priv); - if (h == NULL) + if (h) snd_dlclose(h); free(xlib); return -ENXIO; diff --git a/src/conf.c b/src/conf.c index 8939d62..ddefff6 100644 --- a/src/conf.c +++ b/src/conf.c @@ -3268,6 +3268,7 @@ static int snd_config_hooks_call(snd_config_t *root, snd_config_t *config, snd_c snd_config_iterator_t i, next; if (snd_config_get_type(func_conf) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for func %s definition", str); + err = -EINVAL; goto _err; } snd_config_for_each(i, next, func_conf) { diff --git a/src/hwdep/hwdep.c b/src/hwdep/hwdep.c index b882b35..5dc791c 100644 --- a/src/hwdep/hwdep.c +++ b/src/hwdep/hwdep.c @@ -78,6 +78,7 @@ static int snd_hwdep_open_conf(snd_hwdep_t **hwdep, if (err >= 0) { if (snd_config_get_type(type_conf) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for HWDEP type %s definition", str); + err = -EINVAL; goto _err; } snd_config_for_each(i, next, type_conf) { diff --git a/src/pcm/pcm_hooks.c b/src/pcm/pcm_hooks.c index 3a99d55..404d51e 100644 --- a/src/pcm/pcm_hooks.c +++ b/src/pcm/pcm_hooks.c @@ -385,6 +385,7 @@ static int snd_pcm_hook_add_conf(snd_pcm_t *pcm, snd_config_t *root, snd_config_ if (err >= 0) { if (snd_config_get_type(type) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for PCM type %s definition", str); + err = -EINVAL; goto _err; } snd_config_for_each(i, next, type) { diff --git a/src/pcm/pcm_simple.c b/src/pcm/pcm_simple.c index 975f699..f943ec0 100644 --- a/src/pcm/pcm_simple.c +++ b/src/pcm/pcm_simple.c @@ -89,7 +89,7 @@ static int set_hw_params(snd_pcm_t *pcm, return err; if (periods == 1) return -EINVAL; - if (*period_time == 0) { + if (period_time) { err = INTERNAL(snd_pcm_hw_params_get_period_time)(hw_params, period_time, NULL); if (err < 0) return err; diff --git a/src/rawmidi/rawmidi.c b/src/rawmidi/rawmidi.c index b28488a..0bd6b96 100644 --- a/src/rawmidi/rawmidi.c +++ b/src/rawmidi/rawmidi.c @@ -201,6 +201,7 @@ static int snd_rawmidi_open_conf(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp if (err >= 0) { if (snd_config_get_type(type_conf) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for RAWMIDI type %s definition", str); + err = -EINVAL; goto _err; } snd_config_for_each(i, next, type_conf) { diff --git a/src/rawmidi/rawmidi_virt.c b/src/rawmidi/rawmidi_virt.c index 52b8984..e5b17e4 100644 --- a/src/rawmidi/rawmidi_virt.c +++ b/src/rawmidi/rawmidi_virt.c @@ -383,9 +383,11 @@ int snd_rawmidi_virtual_open(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp, _err: if (seq_handle) snd_seq_close(seq_handle); - if (virt->midi_event) - snd_midi_event_free(virt->midi_event); - free(virt); + if (virt) { + if (virt->midi_event) + snd_midi_event_free(virt->midi_event); + free(virt); + } if (inputp) free(*inputp); if (outputp) -- 1.7.0.4

1 0

[alsa-devel] [PATCH 0/1] Fixed issues/defects reported by Coverity tool.
by sudarshan.bisht＠nokia.com 21 Mar '11

21 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> Coverity Static Analysis helps developers find hard-to-spot, yet potentially crash-causing defects early in the development phase, reducing the cost,time, and risk of software errors. This patch has fix for situations where variable can be NULL but not been checked beforehand. Sudarshan (1): alsa-lib: fixed coverity reported issues under "FORWARD_NULL" checker. modules/mixer/simple/sbasedl.c | 2 +- src/conf.c | 8 +++++--- src/hwdep/hwdep.c | 2 +- src/pcm/pcm_hooks.c | 5 +++-- src/pcm/pcm_simple.c | 2 +- src/rawmidi/rawmidi.c | 2 +- src/rawmidi/rawmidi_virt.c | 8 +++++--- 7 files changed, 17 insertions(+), 12 deletions(-)

3 5

[alsa-devel] [PATCH 15/16] ALSA: VIA HDA: Update VT1718S initial verbs
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Update VT1718S initial verbs. Add a verb of unmute MW0 gain 5 in VT1718S initial verbs. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -4360,7 +4360,8 @@ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - + /* Enable MW0 adjust Gain 5 */ + {0x1, 0xfb2, 0x10}, /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback * mixer widget */ @@ -4369,7 +4370,7 @@ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, /* Setup default input of Front HP to MW9 */ {0x28, AC_VERB_SET_CONNECT_SEL, 0x1},

1 0

[alsa-devel] [PATCH 13/16] ALSA: VIA HDA: Fix VT1708 can't build up Headphone control issue
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix VT1708 can't build up Headphone control issue. Since VT1708 didn't support the control of getting connection number, building of headphone control will fail in via_hp_build() function. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -816,9 +816,12 @@ break; } - nums = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); - if (nums <= 1) - return 0; + if (spec->codec_type != VT1708) { + nums = snd_hda_get_connections(codec, nid, + conn, HDA_MAX_CONNECTIONS); + if (nums <= 1) + return 0; + } knew = via_clone_control(spec, &via_hp_mixer[0]); if (knew == NULL)

1 0

[alsa-devel] [PATCH 12/16] ALSA: VIA HDA: Get codec's type in via_new_spec() function
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Get codec's type in via_new_spec() function. Add get_codec_type() in via_new_spec() function to make sure getting correct codec type before building mixer controls. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> sound/pci/hda/patch_via.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -163,6 +163,7 @@ #endif }; +static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec); static struct via_spec * via_new_spec(struct hda_codec *codec) { struct via_spec *spec; @@ -173,6 +174,10 @@ codec->spec = spec; spec->codec = codec; + spec->codec_type = get_codec_type(codec); + /* VT1708BCE & VT1708S are almost same */ + if (spec->codec_type == VT1708BCE) + spec->codec_type = VT1708S; return spec; } @@ -1768,10 +1773,6 @@ for (i = 0; i < spec->num_iverbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); - spec->codec_type = get_codec_type(codec); - if (spec->codec_type == VT1708BCE) - spec->codec_type = VT1708S; /* VT1708BCE & VT1708S are almost - same */ /* Lydia Add for EAPD enable */ if (!spec->dig_in_nid) { /* No Digital In connection */ if (spec->dig_in_pin) {

1 0

[alsa-devel] [PATCH 11/16] ALSA: VIA HDA: Correct stream names for VT1818S
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Correct stream names for VT1818S. Correct stream names of analog playback and capture streams for VT1818S. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 5 +++++ 1 file changed, 5 insertions(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3970,6 +3970,11 @@ spec->stream_name_analog = "VT1708BCE Analog"; spec->stream_name_digital = "VT1708BCE Digital"; } + /* correct names for VT1818S */ + if (codec->vendor_id == 0x11060440) { + spec->stream_name_analog = "VT1818S Analog"; + spec->stream_name_digital = "VT1818S Digital"; + } /* correct names for VT1705 */ if (codec->vendor_id == 0x11064397) { kfree(codec->chip_name);

1 0

[alsa-devel] [PATCH 9/16] ALSA: VIA HDA: Fix invalid A-A path volume adjust issue
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix invalid A-A path volume adjust issue. Modify vt_auto_create_analog_input_ctls() function to fix invalid a-a path volume adjust issue for VT1708S, VT1702 and VT1716S codecs. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2021,7 +2021,14 @@ else type_idx = 0; label = hda_get_autocfg_input_label(codec, cfg, i); - err = via_new_analog_input(spec, label, type_idx, idx, cap_nid); + if (spec->codec_type == VT1708S || + spec->codec_type == VT1702 || + spec->codec_type == VT1716S) + err = via_new_analog_input(spec, label, type_idx, + idx+1, cap_nid); + else + err = via_new_analog_input(spec, label, type_idx, + idx, cap_nid); if (err < 0) return err; snd_hda_add_imux_item(imux, label, idx, NULL);

1 0

[alsa-devel] [PATCH 8/16] ALSA: VIA HDA: Modify mute_aa_path() function
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Modify mute_aa_path() function. Modify mute_aa_path() function to support VT1718S codec. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 5 +++++ 1 file changed, 5 insertions(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -880,6 +880,11 @@ start_idx = 2; end_idx = 4; break; + case VT1718S: + nid_mixer = 0x21; + start_idx = 1; + end_idx = 3; + break; default: return; }

1 0

[alsa-devel] [PATCH 5/16] ALSA: VIA HDA: Fix side channel mute invalid issue
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix side channel mute invalid issue. Modify side_mute_channel() and update_side_mute_status() functions to fix invalid side channel mute issue of VT2002P, VT1812 and VT1802 codecs. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 24 +++++++++++++++++++----- 1 file changed, 19 insertions(+), 5 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -698,6 +698,9 @@ case VT1709_10CH: return 0x29; case VT1708B_8CH: /* fall thru */ case VT1708S: return 0x27; + case VT2002P: return 0x19; + case VT1802: return 0x15; + case VT1812: return 0x15; default: return 0; } } @@ -706,13 +709,24 @@ { /* mute side channel */ struct via_spec *spec = codec->spec; - unsigned int parm = spec->hp_independent_mode - ? AMP_OUT_MUTE : AMP_OUT_UNMUTE; + unsigned int parm; hda_nid_t sw3 = side_mute_channel(spec); - if (sw3) - snd_hda_codec_write(codec, sw3, 0, AC_VERB_SET_AMP_GAIN_MUTE, - parm); + if (sw3) { + if (spec->codec_type == VT2002P || + spec->codec_type == VT1802 || + spec->codec_type == VT1812) + parm = spec->hp_independent_mode ? + AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1); + else + parm = spec->hp_independent_mode ? + AMP_OUT_MUTE : AMP_OUT_UNMUTE; + snd_hda_codec_write(codec, sw3, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm); + if (spec->codec_type == VT1812) + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm); + } return 0; }

1 0

[alsa-devel] [PATCH 3/16] ALSA: VIA HDA: Update unsolicited event function
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Update unsolicited event function. Update unsolicited event function via_unsol_event() to process more unsolicited events. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -224,12 +224,12 @@ return codec_type; }; +#define VIA_JACK_EVENT 0x20 #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 -#define VIA_JACK_EVENT 0x04 -#define VIA_MONO_EVENT 0x08 -#define VIA_SPEAKER_EVENT 0x10 -#define VIA_BIND_HP_EVENT 0x20 +#define VIA_MONO_EVENT 0x03 +#define VIA_SPEAKER_EVENT 0x04 +#define VIA_BIND_HP_EVENT 0x05 enum { VIA_CTL_WIDGET_VOL, @@ -1705,17 +1705,21 @@ unsigned int res) { res >>= 26; - if (res & VIA_HP_EVENT) - via_hp_automute(codec); - if (res & VIA_GPIO_EVENT) - via_gpio_control(codec); + if (res & VIA_JACK_EVENT) set_widgets_power_state(codec); - if (res & VIA_MONO_EVENT) + + res &= ~VIA_JACK_EVENT; + + if (res == VIA_HP_EVENT) + via_hp_automute(codec); + else if (res == VIA_GPIO_EVENT) + via_gpio_control(codec); + else if (res == VIA_MONO_EVENT) via_mono_automute(codec); - if (res & VIA_SPEAKER_EVENT) + else if (res == VIA_SPEAKER_EVENT) via_speaker_automute(codec); - if (res & VIA_BIND_HP_EVENT) + else if (res == VIA_BIND_HP_EVENT) via_hp_bind_automute(codec); }

1 0

[alsa-devel] [PATCH 0/16] ALSA: HDA VIA: patch series
by Lydia Wang 21 Mar '11

21 Mar '11

Following patch series mainly includes: -support VT1802, VT1705,... -new power management functions of different codecs -fix some issues of old driver

1 0

Re: [alsa-devel] VIA VT1708 bugs
by Raymond Yau 21 Mar '11

21 Mar '11

>> *Independent* headphone >> >> http://www.viaarena.com/forums/showthread.p*hp*?t=41015 <http://www.viaarena.com/forums/showthread.php?t=41015> >Ok, that makes some sense. It translates to me as >* With *independent* *HP* turned off - activates auto-mute between >speaker-out/line-out and headphone out. Note the difference between VIA >and Realtek/Sigmatel here as line-out are auto-muted as well. >* With *independent* *HP* turned on - headphone out and speaker-out/line-out >should take output from different PCM subdevices. Practically, it is more complex than the above since the desktop motherboard can have 3 , 5 or 6 audio jacks at rear panel since those hda codecs only have 8 channels instead of 10 Enable Independent HP on a 6stack model should decrease spec->mult_out.max_channels from 8 to 6 for subdevice 0 For 5 jacks model , the driver need to retask blue jack as side only when independent HP switch is off To support 8 channels for a 3stack model need to retask the front HP as side

1 0

[alsa-devel] QasMixer 0.10.0 released
by Sebastian H. 20 Mar '11

20 Mar '11

Hello Alsa and Qt users Version 0.10.0 of QasMixer is now available. This is a polishing release Changes: * Start minimized if a session is restored * Alsa configuration browser in a separate top menu entry * Slider style tweaks - curves instead of triangles * Check button style tweaks Homepage with more informations and screenshots http://xwmw.org/qasmixer Files are hosted at sourceforge http://sourceforge.net/projects/qasmixer/ Happy volume changing! -- Sebastian Holtermann

1 0

[alsa-devel] [PATCH - Set EAPD for Realtek ALC665 (Vendor Id: 0x10eSet EAPD for Realtek ALC665 (Vendor Id: 0x10ec0665).
by Andres Mejia 20 Mar '11

20 Mar '11

Signed-off-by: Andres Mejia <mcitadel(a)gmail.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 50855cf..be910fa6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1264,6 +1264,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: + case 0x10ec0665: case 0x10ec0862: case 0x10ec0889: set_eapd(codec, 0x14, 1); @@ -4236,6 +4237,7 @@ static void alc_power_eapd(struct hda_codec *codec) case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: + case 0x10ec0665: case 0x10ec0862: case 0x10ec0889: set_eapd(codec, 0x14, 0); -- 1.7.4.1

1 0

[alsa-devel] snd_pcm_writei returns -EAGAIN after poll
by Ivan Sorokin 20 Mar '11

20 Mar '11

Hello! I use slightly modified test/pcm.c in write_and_poll mode. My program looks like this: snd_pcm_open with SND_PCM_NONBLOCK snd_pcm_set_params infinite loop poll for snd_pcm_poll_descriptors snd_pcm_poll_descriptors_revents if POLLOUT flag is set snd_pcm_writei Everything works, but snd_pcm_writei sometimes returns -EAGAIN. The probability of when snd_pcm_writei returns -EAGAIN depends on buffer size that I pass to it. ExperimentallyI found out that when I call snd_pcm_writei with 940 bytes buffers (maybe something related to period size?) it never returns -EAGAIN. Why poll returns when there is no place for new data? Is it ok to ignore EAGAIN and just start another poll? Is there any recommended buffer size for snd_pcm_writei? My test program: https://github.com/sorokin/asoundpp/blob/master/sine_async.cpp (94 LOC)

2 1

[alsa-devel] ALSA/pulseaudio problem workaround
by Craig Bourne 19 Mar '11

19 Mar '11

Here is a workaround for the problem I described in my email to you yesterday: Background My past experience with ALSA, JACK and many of the Linux audio tools that depend on them is that often some few applications in the tool-chain of a complex task will not work properly (or at all) unless invoked with *root* privileges. One is now frustrated in getting such things to work Fedora 14 as they have for the past decade or so in that the most basic tools (e.g., alsactl, QjackCtl) no longer work for root. Indeed, audio also does not work for unprivileged users on Fedora 14 either (though this evidently fails at a different level and will be taken up in a subsequent communication). Symptoms alsactl fails carping about pulseaudio, audacity can be started on its own but no longer wants to cooperate with JACK and ALSA resulting in horribly distorted sound from something horked together in the background as a "favor" to the user, QjackCtl runs, produces messages, and blinks its cute little meter display but evidently does this off on the sidelines reporting 0 XRUNS even as every little jot and tiddle of computation produces a surprising audio result. Strategy Based on my observation that PULSEAUDIO is evidently being allowed to interpose itself between the user (even the superuser) and ALSA, and since this user has been kept largely ignorant of the workings of all this and does not know the secret handshake, I will attempt to at least minimize the damage to audio and regain some measure of control while not attempting to disable the PULSEAUDIO/ALSA cabal. Workaround Install the following if they are not already on your system: - avahi - padevchooser - pulseaudio-libs-zeroconf - pulseaudio-module-zeroconf Enable and start the Avahi Daemon Create these two shell scripts (that will be used by QjackCtl), make them executable, and place them in a directory where they will stay to be invoked each time QjackCtl is started: PAjack_startup #!/bin/bash # Name: PAjack_startup pulseaudio -D --system #perhaps more fastidious than the above line is #pulseaudio -D --system --disallow-exit --disallow-module-loading padevchooser& PAjack_shutdown #!/bin/bash #Name: PAjack_shutdown pulseaudio --kill killall padevchooser Invoke QjackCtl click Setup select the Options tab tick the box "Execute script after Startup" and enter ~/bin/PAjack_startup [Note: using, of course the path to your script] tick the box "Execute script on Shutdown" and enter ~/bin/PAjack_shutdown [Note: as before, using your path to the file] Quit QjackCtl -- your scripts will now be invoked the next time it is used. A partial workaround for the unprivileged user, at least to the point of getting QjackCtl and audacious working places a slightly modified PAjack_startup script in a place where QjackCtl can find it when it starts with the changes used above for root (e.g., insure that both acounts have a subdirectory named "bin" in their home directory and have QjackCtl look for ~/bin/PAjack_startup when it is invoked). Here is the script for unprivileged users: #!/bin/bash #Name: PAjack_startup pulseaudio -D padevchooser& Limitations This is not a general solution to the ALSA/PULSEAUDIO problems I am experiencing but it is a workaround that at least allows me to have some minimal functionality back from basic audio tools that I have come to rely on in my work as a composer. -- Comradely, Craig Bourne

3 10

[alsa-devel] syslog requested reporting this to ALSA developers (ubuntu 10.04 inside virtualbox)
by Jason E. Aten 19 Mar '11

19 Mar '11

from my syslog, it requests a report to ALSA developers. Let me know if I can provide further assistance. -Jason $ tail /var/log/syslog Mar 17 13:58:25 virtub pulseaudio[1038]: alsa-source.c: ALSA woke us up to read new data from the device, but there was actually nothing to read! Mar 17 13:58:25 virtub pulseaudio[1038]: alsa-source.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Mar 17 13:58:25 virtub pulseaudio[1038]: alsa-source.c: We were woken up with POLLIN set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Mar 17 13:58:26 virtub pulseaudio[1038]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Mar 17 13:58:26 virtub pulseaudio[1038]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Mar 17 13:58:26 virtub pulseaudio[1038]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Mar 17 13:58:29 virtub gdm-session-worker[1030]: GLib-GObject-CRITICAL: g_value_get_boolean: assertion `G_VALUE_HOLDS_BOOLEAN (value)' failed Mar 17 13:58:35 virtub rtkit-daemon[1043]: Sucessfully made thread 1256 of process 1256 (n/a) owned by '1000' high priority at nice level -11. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Supervising 4 threads of 2 processes of 2 users. Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-util.c: Disabling timer-based scheduling because running inside a VM. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Sucessfully made thread 1257 of process 1256 (n/a) owned by '1000' RT at priority 5. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Supervising 5 threads of 2 processes of 2 users. Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-util.c: Disabling timer-based scheduling because running inside a VM. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Sucessfully made thread 1258 of process 1256 (n/a) owned by '1000' RT at priority 5. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Supervising 6 threads of 2 processes of 2 users. Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Mar 17 13:59:40 virtub AptDaemon: INFO: Initializing daemon root@virtub:/var/log# uname -a Linux virtub 2.6.32-29-generic #58-Ubuntu SMP Fri Feb 11 20:52:10 UTC 2011 x86_64 GNU/Linux root@virtub:/var/log# dpkg -s pulseaudio Package: pulseaudio Status: install ok installed Priority: optional Section: sound Installed-Size: 4540 Maintainer: Ubuntu Core Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Architecture: amd64 Version: 1:0.9.22~0.9.21+stable-queue-32-g8478-0ubuntu14 Replaces: pulseaudio-module-hal, pulseaudio-module-rygel-media-server (<< 1:0.9.22~0.9.21+stable-queue-32-g8478-0ubuntu14), pulseaudio-module-udev (<< 1:0.9.22~0.9.21+stable-qu\ eue-32-g8478-0ubuntu14) Provides: pulseaudio-module-hal, pulseaudio-module-rygel-media-server, pulseaudio-module-udev Depends: libasound2 (>> 1.0.22), libc6 (>= 2.9), libdbus-1-3 (>= 1.0.2), libice6 (>= 1:1.0.0), libltdl7 (>= 2.2.6b), libpulse0 (= 1:0.9.22~0.9.21+stable-queue-32-g8478-0ubuntu1\ 4), libsamplerate0, libsm6, libsndfile1 (>= 1.0.20), libspeexdsp1 (>= 1.2~beta3.2-1), libtdb1 (>= 1.1.2~git20080520), libudev0 (>= 147), libx11-6 (>= 0), libxtst6, adduser, lsb\ -base (>= 3), consolekit, libasound2-plugins, pulseaudio-utils Recommends: pulseaudio-module-x11, gstreamer0.10-pulseaudio, pulseaudio-esound-compat, rtkit Suggests: pavumeter, paman, paprefs Conflicts: libltdl3 (<< 1.5.24-1), pulseaudio-module-rygel-media-server (<< 1:0.9.22~0.9.21+stable-queue-32-g8478-0ubuntu14), pulseaudio-module-udev (<< 1:0.9.22~0.9.21+stable-\ queue-32-g8478-0ubuntu14) Conffiles: /etc/default/pulseaudio 5ac1e3004508adbb816e9b1af352617e /etc/init.d/pulseaudio bd6ccc065d494d8baac1d823521833b4 /etc/pulse/default.pa d2d623c97aef9f81f5b55f2d7e736c87 /etc/pulse/system.pa 182fbc3ba3449027c4d1e39a150c8396 /etc/pulse/daemon.conf ed3746b9cd3ba486184128fd611b73c9 /etc/dbus-1/system.d/pulseaudio-system.conf 69d56dc5342bf589a2e199541709a452 /etc/xdg/autostart/pulseaudio.desktop 1310d377ad0d534f699e864037f66bd2 Description: PulseAudio sound server PulseAudio, previously known as Polypaudio, is a sound server for POSIX and WIN32 systems. It is a drop in replacement for the ESD sound server with much better latency, mixing/re-sampling quality and overall architecture. . These are some of PulseAudio's features: . * High quality software mixing of multiple audio streams with support for more than one sink/source. May be used to combine multiple sound cards into one (with sample rate adjustment). . * Wide range of supported client libraries. ESD, ALSA, oss, libao and GStreamer client applications are supported as-is. Native PulseAudio plug-ins are also available for xmms and mplayer. . * Good low latency behaviour and very accurate latency measurement for playback and recording. Ability to fully synchronize multiple playback streams. . * Network transparency, allowing an application to play back or record audio on a different machine than the one it is running on. . * Extensible plug-in architecture with plug-ins for jackd, multicast-rtp lirc and avahi, just to name a few. . This package contains the daemon and basic module set. Homepage: http://www.pulseaudio.org Original-Maintainer: Pulseaudio maintenance team < pkg-pulseaudio-devel(a)lists.alioth.debian.org> root@virtub:/var/log#

2 1

Re: [alsa-devel] Microphone not working on PCI ID 8086:3b56
by Jeff Chua 18 Mar '11

18 Mar '11

On Tue, Sep 28, 2010 at 9:42 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > Enable all CONFIG_SND_HDA_*. Done that. Still no mic. On Tue, Sep 28, 2010 at 9:42 PM, Wu Fengguang <fengguang.wu(a)intel.com> wrote: > It's easier to enable all of them. They are small modules. Tried just about everything, still no mic. Sounds works in all cases. I found that with SND_HDA_CODEC_INTELHDMI selected, SND_DYNAMIC_MINORS will be enabled by default which won't work in my case as I'm not using udev. Here's a patch to take away the default so non-udev will still works. Do I really need udev to get mic to work? Thanks, Jeff --- lx/sound/pci/hda/Kconfig.org 2010-09-14 16:34:52.000000000 +0800 +++ lx/sound/pci/hda/Kconfig 2010-09-14 16:35:09.000000000 +0800 @@ -145,7 +145,6 @@ config SND_HDA_CODEC_INTELHDMI bool "Build INTEL HDMI HD-audio codec support" - select SND_DYNAMIC_MINORS default y help Say Y here to include INTEL HDMI HD-audio codec support in

3 16

[alsa-devel] [PATCHv4] alsa-lib: ucm: allow values to be read from devices
by Margarita Olaya 18 Mar '11

18 Mar '11

This allows devices value to be read using the get_value(). Also fix a little formatting. Signed-off-by: Margarita Olaya Cabrera <magi(a)slimlogic.co.uk> Acked-by: Liam Girdwood <lrg(a)ti.com> --- src/ucm/main.c | 21 ++++++++++++++------- 1 files changed, 14 insertions(+), 7 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 42fd373..88f8bf6 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -1059,24 +1059,31 @@ static int get_value3(const char **value, * \param uc_mgr Use case manager * \param identifier Value identifier (string) * \param value Returned value string - * \param modifier modifier name (string) + * \param item Modifier or Device name (string) * \return Zero on success (value is filled), otherwise a negative error code */ static int get_value(snd_use_case_mgr_t *uc_mgr, - const char *identifier, - const char **value, - const char *modifier) + const char *identifier, + const char **value, + const char *item) { - struct use_case_modifier *mod; + struct use_case_modifier *mod; + struct use_case_device *dev; int err; - if (modifier != NULL) { - mod = find_modifier(uc_mgr, modifier); + if (item != NULL) { + mod = find_modifier(uc_mgr, item); if (mod != NULL) { err = get_value1(value, &mod->value_list, identifier); if (err >= 0 || err != -ENOENT) return err; } + dev = find_device(uc_mgr->active_verb, item); + if (dev != NULL) { + err = get_value1(value, &dev->value_list, identifier); + if (err >= 0 || err != -ENOENT) + return err; + } } err = get_value1(value, &uc_mgr->active_verb->value_list, identifier); if (err >= 0 || err != -ENOENT) -- 1.7.0.4

2 2

[alsa-devel] [PATCHv3] alsa-lib: ucm: allow values to be read from devices
by Margarita Olaya 18 Mar '11

18 Mar '11

This allows devices value to be read using the get_value(). Also fix a little formatting. Signed-off-by: Margarita Olaya Cabrera <magi(a)slimlogic.co.uk> Acked-by: Liam Girdwood <lrg(a)ti.com> --- src/ucm/main.c | 21 ++++++++++++++------- 1 files changed, 14 insertions(+), 7 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 42fd373..1246aa8 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -1059,24 +1059,31 @@ static int get_value3(const char **value, * \param uc_mgr Use case manager * \param identifier Value identifier (string) * \param value Returned value string - * \param modifier modifier name (string) + * \param item Modifier of Device name (string) * \return Zero on success (value is filled), otherwise a negative error code */ static int get_value(snd_use_case_mgr_t *uc_mgr, - const char *identifier, - const char **value, - const char *modifier) + const char *identifier, + const char **value, + const char *item) { - struct use_case_modifier *mod; + struct use_case_modifier *mod; + struct use_case_device *dev; int err; - if (modifier != NULL) { - mod = find_modifier(uc_mgr, modifier); + if (item != NULL) { + mod = find_modifier(uc_mgr, item); if (mod != NULL) { err = get_value1(value, &mod->value_list, identifier); if (err >= 0 || err != -ENOENT) return err; } + dev = find_device(uc_mgr->active_verb, item); + if (dev != NULL) { + err = get_value1(value, &dev->value_list, identifier); + if (err >= 0 || err != -ENOENT) + return err; + } } err = get_value1(value, &uc_mgr->active_verb->value_list, identifier); if (err >= 0 || err != -ENOENT) -- 1.7.0.4

2 1

[alsa-devel] [PATCHv2] alsa-lib: ucm: allow values to be read from devices
by Margarita Olaya 18 Mar '11

18 Mar '11

This allows devices value to be read using the get_value(). Also fix a little formatting. Signed-off-by: Margarita Olaya Cabrera <magi(a)slimlogic.co.uk> Acked-by: Liam Girdwood <lrg(a)ti.com> --- src/ucm/main.c | 21 ++++++++++++++------- 1 files changed, 14 insertions(+), 7 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 42fd373..0393601 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -1059,24 +1059,31 @@ static int get_value3(const char **value, * \param uc_mgr Use case manager * \param identifier Value identifier (string) * \param value Returned value string - * \param modifier modifier name (string) + * \param item item name (string) * \return Zero on success (value is filled), otherwise a negative error code */ static int get_value(snd_use_case_mgr_t *uc_mgr, - const char *identifier, - const char **value, - const char *modifier) + const char *identifier, + const char **value, + const char *item) { - struct use_case_modifier *mod; + struct use_case_modifier *mod; + struct use_case_device *dev; int err; - if (modifier != NULL) { - mod = find_modifier(uc_mgr, modifier); + if (item != NULL) { + mod = find_modifier(uc_mgr, item); if (mod != NULL) { err = get_value1(value, &mod->value_list, identifier); if (err >= 0 || err != -ENOENT) return err; } + dev = find_device(uc_mgr->active_verb, item); + if (dev != NULL) { + err = get_value1(value, &dev->value_list, identifier); + if (err >= 0 || err != -ENOENT) + return err; + } } err = get_value1(value, &uc_mgr->active_verb->value_list, identifier); if (err >= 0 || err != -ENOENT) -- 1.7.0.4

2 1

[alsa-devel] [PATCH] alsa-lib: ucm: allow values to be read from devices
by Margarita Olaya 18 Mar '11

18 Mar '11

This allows devices value to be read using the get_value(). Also fix a little formatting. Signed-off-by: Margarita Olaya Cabrera <magi(a)slimlogic.co.uk> --- src/ucm/main.c | 19 +++++++++++++------ 1 files changed, 13 insertions(+), 6 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 42fd373..fb11e61 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -1063,20 +1063,27 @@ static int get_value3(const char **value, * \return Zero on success (value is filled), otherwise a negative error code */ static int get_value(snd_use_case_mgr_t *uc_mgr, - const char *identifier, - const char **value, - const char *modifier) + const char *identifier, + const char **value, + const char *item) { - struct use_case_modifier *mod; + struct use_case_modifier *mod; + struct use_case_device *dev; int err; - if (modifier != NULL) { - mod = find_modifier(uc_mgr, modifier); + if (item != NULL) { + mod = find_modifier(uc_mgr, item); if (mod != NULL) { err = get_value1(value, &mod->value_list, identifier); if (err >= 0 || err != -ENOENT) return err; } + dev = find_device(uc_mgr->active_verb, item); + if (dev != NULL) { + err = get_value1(value, &dev->value_list, identifier); + if (err >=0 || err != -ENOENT) + return err; + } } err = get_value1(value, &uc_mgr->active_verb->value_list, identifier); if (err >= 0 || err != -ENOENT) -- 1.7.0.4

3 4

[alsa-devel] Keyclick latency
by Marc-André Hébert 18 Mar '11

18 Mar '11

Hello, I am working on an embedded device which does audio playback. Now when the user presses a key, we want to play a tone to provide a keyclick feedback to the user. What we always did (we were using OSS before) was pretty much the simplest thing possible which is for the application to: -stop sending playback samples -send the keyclick samples -resume sending the playback samples With our new hardware this approach does not produce good results because of the size of the device's ringbuffer. The larger buffer introduces a significant latency issue for the user feedback. Reducing the ring buffer size is not a viable option to ensure no underruns are generated on the system. Now I was wondering if there a good approach for this using alsa. Checking the api, a simple solution would be to: -call snd_pcm_drop -send the keyclick samples -resume sending the playback samples I've tried this and the latency is good, unfortunately there is the drawback that some samples were dropped. So I guess the application could : -keep a copy of the samples sent, -check the amount of samples left in the ringbuffer before calling snd_pcm_drop, -send the keyclick -resend dropped samples -resume sending the playback samples Seems a little overcomplicated so I was wondering if I was missing something. Would there be a simpler approach? Something like pausing the stream or reading the contents of the ringbuffer before dropping. Thanks Marc

2 1

[alsa-devel] [PATCH] ASoC: Mark WM8958 DSP2 registers readable
by Mark Brown 18 Mar '11

18 Mar '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 328f328..bdd1ac7 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -55,6 +55,9 @@ static int wm8994_retune_mobile_base[] = { static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = wm8994->control_data; + switch (reg) { case WM8994_GPIO_1: case WM8994_GPIO_2: @@ -71,6 +74,15 @@ static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) case WM8994_INTERRUPT_STATUS_2: case WM8994_INTERRUPT_RAW_STATUS_2: return 1; + + case WM8958_DSP2_PROGRAM: + case WM8958_DSP2_CONFIG: + case WM8958_DSP2_EXECCONTROL: + if (control->type == WM8958) + return 1; + else + return 0; + default: break; } -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: Remove bogus check for register validity in debugfs write
by Mark Brown 18 Mar '11

18 Mar '11

Since not all registers need to be cached and the cache is entirely optional anyway we shouldn't be checking that a register is in the cached range. If the register is invalid then the actual I/O code can determine that and report an error. Similarly, the step size can and should be enforced by the lower level code if it's important. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 17efacd..4dda589 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -259,8 +259,6 @@ static ssize_t codec_reg_write_file(struct file *file, while (*start == ' ') start++; reg = simple_strtoul(start, &start, 16); - if ((reg >= codec->driver->reg_cache_size) || (reg % step)) - return -EINVAL; while (*start == ' ') start++; if (strict_strtoul(start, 16, &value)) -- 1.7.4.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: Treat WM8958 revision A as WM8994 revision D
by Mark Brown 18 Mar '11

18 Mar '11

The first WM8958 revision requires similar treatment. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 36 ++++++++++++++++++++++++++---------- 1 files changed, 26 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3dc64c8..9458289 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3327,14 +3327,23 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8958: snd_soc_add_controls(codec, wm8958_snd_controls, ARRAY_SIZE(wm8958_snd_controls)); - snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, - ARRAY_SIZE(wm8994_lateclk_widgets)); - snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets, - ARRAY_SIZE(wm8994_adc_widgets)); - snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, - ARRAY_SIZE(wm8994_dac_widgets)); snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, ARRAY_SIZE(wm8958_dapm_widgets)); + if (wm8994->revision < 1) { + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, + ARRAY_SIZE(wm8994_lateclk_revd_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets, + ARRAY_SIZE(wm8994_adc_revd_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_dac_revd_widgets, + ARRAY_SIZE(wm8994_dac_revd_widgets)); + } else { + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, + ARRAY_SIZE(wm8994_lateclk_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets, + ARRAY_SIZE(wm8994_adc_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, + ARRAY_SIZE(wm8994_dac_widgets)); + } break; } @@ -3358,10 +3367,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } break; case WM8958: - snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, - ARRAY_SIZE(wm8994_lateclk_intercon)); - snd_soc_dapm_add_routes(dapm, wm8958_intercon, - ARRAY_SIZE(wm8958_intercon)); + if (wm8994->revision < 1) { + snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, + ARRAY_SIZE(wm8994_revd_intercon)); + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, + ARRAY_SIZE(wm8994_lateclk_revd_intercon)); + } else { + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, + ARRAY_SIZE(wm8994_lateclk_intercon)); + snd_soc_dapm_add_routes(dapm, wm8958_intercon, + ARRAY_SIZE(wm8958_intercon)); + } break; } -- 1.7.4.1

2 3

[alsa-devel] [PATCH] alsa-lib: fixed coverity reported issues under "RESOURCE_LEAK" checker.
by sudarshan.bisht＠nokia.com 18 Mar '11

18 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> --- modules/mixer/simple/sbase.c | 1 + modules/mixer/simple/sbasedl.c | 2 +- src/conf.c | 4 ++++ src/control/control_hw.c | 4 +++- src/pcm/pcm_file.c | 4 +++- src/pcm/pcm_ladspa.c | 20 ++++++++++++++------ src/pcm/pcm_plug.c | 9 +++++++-- src/pcm/pcm_rate.c | 4 ++++ src/rawmidi/rawmidi.c | 4 +++- 9 files changed, 40 insertions(+), 12 deletions(-) diff --git a/modules/mixer/simple/sbase.c b/modules/mixer/simple/sbase.c index 97feee8..bb2f59d 100644 --- a/modules/mixer/simple/sbase.c +++ b/modules/mixer/simple/sbase.c @@ -377,6 +377,7 @@ static int simple_event_add1(snd_mixer_class_t *class, if (ctype != SND_CTL_ELEM_TYPE_BOOLEAN) { __invalid_type: snd_mixer_selem_id_free(id); + free(hsimple); return -EINVAL; } break; diff --git a/modules/mixer/simple/sbasedl.c b/modules/mixer/simple/sbasedl.c index 0137586..494802f 100644 --- a/modules/mixer/simple/sbasedl.c +++ b/modules/mixer/simple/sbasedl.c @@ -99,7 +99,7 @@ int mixer_simple_basic_dlopen(snd_mixer_class_t *class, __error: if (initflag) free(priv); - if (h == NULL) + if (h) snd_dlclose(h); free(xlib); return -ENXIO; diff --git a/src/conf.c b/src/conf.c index 8939d62..c4daae7 100644 --- a/src/conf.c +++ b/src/conf.c @@ -638,6 +638,7 @@ static int get_char_skip_comments(input_t *input) fd = malloc(sizeof(*fd)); if (!fd) { free(str); + snd_input_close(in); return -ENOMEM; } fd->name = str; @@ -1847,6 +1848,7 @@ int snd_config_delete(snd_config_t *config) list_del(&config->list); free(config->id); free(config); + config = NULL; return 0; } @@ -4589,6 +4591,8 @@ static int parse_args(snd_config_t *subs, const char *str, snd_config_t *defs) if (err < 0) { _err: free(val); + if (sub) + snd_config_delete(sub); return err; } free(val); diff --git a/src/control/control_hw.c b/src/control/control_hw.c index cf258b4..29cbbc6 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -230,8 +230,10 @@ static int snd_ctl_hw_elem_tlv(snd_ctl_t *handle, int op_flag, return -errno; } if (op_flag == 0) { - if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) + if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) { + free(xtlv); return -EFAULT; + } memcpy(tlv, xtlv->tlv, xtlv->tlv[1] + 2 * sizeof(unsigned int)); } free(xtlv); diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index bfa1cc8..adba7c0 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -226,7 +226,9 @@ static int snd_pcm_file_open_output_file(snd_pcm_file_t *file) file->final_fname); return -errno; } - fd = fileno(pipe); + fd = dup(fileno(pipe)); + if (fd >= 0) + fclose(pipe); } else { if (file->trunc) fd = open(file->final_fname, O_WRONLY|O_CREAT|O_TRUNC, diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index c413c10..41e1d38 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -750,8 +750,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads if (instance->input.data == NULL || instance->input.m_data == NULL || instance->output.data == NULL || - instance->output.m_data == NULL) + instance->output.m_data == NULL) { + free(pchannels); return -ENOMEM; + } for (idx = 0; idx < instance->input.channels.size; idx++) { chn = instance->output.channels.array[idx]; if (pchannels[chn] == NULL && chn < ichannels) { @@ -761,8 +763,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads instance->input.data[idx] = pchannels[chn]; if (instance->input.data[idx] == NULL) { instance->input.data[idx] = snd_pcm_ladspa_allocate_zero(ladspa, 0); - if (instance->input.data[idx] == NULL) - return -ENOMEM; + if (instance->input.data[idx] == NULL) { + free(pchannels); + return -ENOMEM; + } } } for (idx = 0; idx < instance->output.channels.size; idx++) { @@ -770,8 +774,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads /* FIXME/OPTIMIZE: check if we can remove double alloc */ /* if LADSPA plugin has no broken inplace */ instance->output.data[idx] = malloc(sizeof(LADSPA_Data) * ladspa->allocated); - if (instance->output.data[idx] == NULL) - return -ENOMEM; + if (instance->output.data[idx] == NULL) { + free(pchannels); + return -ENOMEM; + } pchannels[chn] = instance->output.m_data[idx] = instance->output.data[idx]; } } @@ -793,8 +799,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads instance->output.data[idx] = NULL; } else { instance->output.data[idx] = snd_pcm_ladspa_allocate_zero(ladspa, 1); - if (instance->output.data[idx] == NULL) + if (instance->output.data[idx] == NULL) { + free(pchannels); return -ENOMEM; + } } } } diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index e9d2923..e80bd3e 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -1298,6 +1298,7 @@ int _snd_pcm_plug_open(snd_pcm_t **pcmp, const char *name, err = snd_pcm_route_load_ttable(tt, ttable, csize, ssize, &cused, &sused, -1); if (err < 0) { snd_config_delete(sconf); + free(ttable); return err; } } @@ -1310,12 +1311,16 @@ int _snd_pcm_plug_open(snd_pcm_t **pcmp, const char *name, err = snd_pcm_open_slave(&spcm, root, sconf, stream, mode, conf); snd_config_delete(sconf); - if (err < 0) + if (err < 0) { + free(ttable); return err; + } err = snd_pcm_plug_open(pcmp, name, sformat, schannels, srate, rate_converter, route_policy, ttable, ssize, cused, sused, spcm, 1); - if (err < 0) + if (err < 0) { + free(ttable); snd_pcm_close(spcm); + } return err; } #ifndef DOC_HIDDEN diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 70e30e5..cf218ae 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -1392,11 +1392,13 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, } else { SNDERR("Invalid type for rate converter"); snd_pcm_close(pcm); + free(rate); return -EINVAL; } if (err < 0) { SNDERR("Cannot find rate converter"); snd_pcm_close(pcm); + free(rate); return -ENOENT; } #else @@ -1405,6 +1407,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, err = open_func(SND_PCM_RATE_PLUGIN_VERSION, &rate->obj, &rate->ops); if (err < 0) { snd_pcm_close(pcm); + free(rate); return err; } #endif @@ -1413,6 +1416,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, ! rate->ops.input_frames || ! rate->ops.output_frames) { SNDERR("Inproper rate plugin %s initialization", type); snd_pcm_close(pcm); + free(rate); return err; } diff --git a/src/rawmidi/rawmidi.c b/src/rawmidi/rawmidi.c index b28488a..25732d2 100644 --- a/src/rawmidi/rawmidi.c +++ b/src/rawmidi/rawmidi.c @@ -255,8 +255,10 @@ static int snd_rawmidi_open_conf(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp snd_config_delete(type_conf); if (err >= 0) err = open_func(inputp, outputp, name, rawmidi_root, rawmidi_conf, mode); - if (err < 0) + if (err < 0) { + snd_dlclose(h); return err; + } if (inputp) { (*inputp)->dl_handle = h; h = NULL; snd_rawmidi_params_default(*inputp, &params); -- 1.7.0.4

1 0

[alsa-devel] [PATCH 0/1] Fixed issues/defects reported by Coverity tool
by sudarshan.bisht＠nokia.com 18 Mar '11

18 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> Coverity Static Analysis helps developers find hard-to-spot, yet potentially crash-causing defects early in the development phase, reducing the cost,time, and risk of software errors. This patch has got fixes for many potential memory leaks. Sudarshan (1): alsa-lib: fixed coverity reported issues under "RESOURCE_LEAK" checker. modules/mixer/simple/sbase.c | 1 + modules/mixer/simple/sbasedl.c | 2 +- src/conf.c | 2 ++ src/control/control_hw.c | 4 +++- src/pcm/pcm_file.c | 5 +++-- src/pcm/pcm_ladspa.c | 20 ++++++++++++++------ src/pcm/pcm_plug.c | 5 ++++- src/pcm/pcm_rate.c | 1 + src/rawmidi/rawmidi.c | 4 +++- 9 files changed, 32 insertions(+), 12 deletions(-)

3 3

[alsa-devel] Destruction of the sound after prolonged output
by Максим 18 Mar '11

18 Mar '11

Hi, My name is Max. I've got some problems using ALSA, a certain type of destruction of sound appears after a prolonged sound output. The destruction is like "bit crush", stutterings, interruptions, etc. Once started, it no longer stops. It starts almost always but only after a prolonged sound output, nearby 24 hours. To reproduce it shurely I use my test wav file and standard speaker-test utility with following options: speaker-test -t wav -w test.wav -r 16000 I've attached test.wav to this mail. To reproduce this I use a fresh minimal installation of Ubuntu 10.10, command line only, with standard Ubuntu 2.6.35 generic kernel. Minimal installation is made by installing Ubuntu from an "alternate" version of the installation CD. It provide option "command line minimal install" that allow install only base system with command line. So there isn't any desktop applications like Gnome, Pulseaudio, etc. I've used standard Ubuntu package of ALSA with version 1.0.23, and also with compiled 1.0.24. I've tried to reproduce this behavior on Centos and Fedora (both in single mode, command line only) with Intel HDA chipset, and on Ubuntu with a Realtek chipset. The results was the same, sound destruction after a day of sound output. Above is the output of speaker-test with parameters: speaker-test -d -c 2 -t wav Before this command my sound test was used during about 6 hours to reproduce the sound destruction: speaker-test -t wav -w test.wav -r 16000 Plug PCM: Linear conversion PCM (S32_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Soft volume PCM Control: PCM Playback Volume min_dB: -51 max_dB: 0 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 1073741824 silence_threshold: 0 silence_size : 1073741824 boundary : 1073741824 appl_ptr : 0 hw_ptr : 926558209

2 1

Re: [alsa-devel] Question about your DSP topic branch
by Patrick Lai 18 Mar '11

18 Mar '11

Hi Liam, I have two more questions about your DSP topic branch 7. I see in sdp4430.c, SDP4430 MODEM front-end dailink no_host_mode is set to SND_SOC_DAI_LINK_NO_HOST. What is the purpose of no_host_mode? Is it for use case that two physical devices can exchange audio data without host-processor intervention? If so, if user-space application tries to write to PCM buffer, will framework reject the buffer? 8. I see there is dmic codec(dmic.c) under sound/soc/codec which is pretty much just a dummy codec driver. I supposed the configuration of DMIC is done in other driver. Would it be better if we could have something like fixed voltage regulator? So, there is no need to duplicate the effort. Look forward to seeing your reply soon Thanks On 1/6/2011 3:39 PM, Patrick Lai wrote: > Hi Liam, > > I sync to your kernel DSP topic branch two days back in attempt to > understand the up-coming ASOC DSP design. I have few questions to get > clarification from you. > > 1. In the sdp4430.c, there are both FE dai link and BE dai link have > omap-aess-audio as platform driver. How is omap-aess-audio platform > driver used in both front-end and backend? > > 2. Front-end dai with stream name "Multimedia" has platform driver as > omap-pcm-audio which is the DMA based. This front-end dai is mapped to > backend (i.e PDM-DL1) with platform driver as omap-aess-audio. > Omap-aess-audio looks to me is DSP platform driver. If a stream is DMA > based, it seems strange to have DSP-based backend. > > 3. To best of my knowledge, I see in omap-abe.c which is the front end > implementation. Playback_trigger() seems to go ahead enable all > back-ends linked to the given front-end. However, front end may not be > routed to all back-ends. Where in the code to make sure BE is only > activated when a stream is being routed to that particular back-end? > > 4. omap-abe.c manages activation of BE DAIS and omap-abe-dsp.c manages > routing of front-end to back-end DAPM widgets and routing map. Am I > correct? This leads to previous question. How are two drivers working > together to make sure BEs are only activated if Front-end has been > routed to them? > > 5. Is there mechanism for front-end to switch between DMA and DSP > platform driver? It looks to me that mapping of frond-end and platform > driver is predetermined based on use case. For example, HDMI goes > through DMA as legacy dailink. > > 6. struct abe_data has array member called dapm. It looks to me that > this array simply tracks dapm components status but I don't see it's > really being used in meaningful way in omap-abe-adsp.c. > > Thanks > -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum. -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

3 13

[alsa-devel] [PATCH] alsa-lib: fixed coverity reported issues under "REVERSE_INULL" checker.
by sudarshan.bisht＠nokia.com 17 Mar '11

17 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> --- src/control/setup.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/src/control/setup.c b/src/control/setup.c index eecda45..3606e5c 100644 --- a/src/control/setup.c +++ b/src/control/setup.c @@ -398,7 +398,7 @@ static int snd_config_get_ctl_elem_value(snd_config_t *conf, static int add_elem(snd_sctl_t *h, snd_config_t *_conf, snd_config_t *private_data) { - snd_config_t *conf; + snd_config_t *conf = NULL ; snd_config_iterator_t i, next; char *tmp; int iface = SND_CTL_ELEM_IFACE_MIXER; @@ -412,6 +412,9 @@ static int add_elem(snd_sctl_t *h, snd_config_t *_conf, snd_config_t *private_da snd_config_t *value = NULL, *mask = NULL; snd_sctl_elem_t *elem = NULL; int err; + /* The last parameter of snd_config_expand() is an output parameter, and + * it is guaranteed that this parameter is set when the function succeeds. + */ err = snd_config_expand(_conf, _conf, NULL, private_data, &conf); if (err < 0) return err; -- 1.7.0.4

1 0

[alsa-devel] [PATCH 0/1] Fixed issues/defects reported by Coverity tool
by sudarshan.bisht＠nokia.com 17 Mar '11

17 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> Coverity Static Analysis helps developers find hard-to-spot, yet potentially crash-causing defects early in the development phase, reducing the cost,time, and risk of software errors. This patch has a fix for a situation where variable can become NULL after certain function call, but later not been checked before passing to the next function call. Sudarshan (1): alsa-lib: fixed coverity reported issues under "REVERSE_INULL" checker. src/control/setup.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-)

3 3

[alsa-devel] [PATCH] alsa-lib: fixed coverity reported issues under "USE_AFTER_FREE" checker.
by sudarshan.bisht＠nokia.com 17 Mar '11

17 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> --- modules/mixer/simple/python.c | 1 + src/control/control_hw.c | 1 + src/mixer/mixer.c | 1 - src/mixer/simple_abst.c | 2 ++ 4 files changed, 4 insertions(+), 1 deletions(-) diff --git a/modules/mixer/simple/python.c b/modules/mixer/simple/python.c index c822c52..784d9a0 100644 --- a/modules/mixer/simple/python.c +++ b/modules/mixer/simple/python.c @@ -663,6 +663,7 @@ pymixer_attach_hctl(struct pymixer *pymixer, PyObject *args) return NULL; err = snd_mixer_attach_hctl(pymixer->mixer, hctl); if (err < 0) { + snd_hctl_close(hctl); PyErr_Format(PyExc_RuntimeError, "Cannot attach hctl: %s", snd_strerror(err)); return NULL; } diff --git a/src/control/control_hw.c b/src/control/control_hw.c index cf258b4..16c4987 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -414,6 +414,7 @@ int snd_ctl_hw_open(snd_ctl_t **handle, const char *name, int card, int mode) if (err < 0) { close(fd); free(hw); + return err; } ctl->ops = &snd_ctl_hw_ops; ctl->private_data = hw; diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 85d843f..5e5789a 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -228,7 +228,6 @@ int snd_mixer_attach_hctl(snd_mixer_t *mixer, snd_hctl_t *hctl) return -ENOMEM; err = snd_hctl_nonblock(hctl, 1); if (err < 0) { - snd_hctl_close(hctl); free(slave); return err; } diff --git a/src/mixer/simple_abst.c b/src/mixer/simple_abst.c index 9e9aaf5..127d8d2 100644 --- a/src/mixer/simple_abst.c +++ b/src/mixer/simple_abst.c @@ -336,6 +336,8 @@ int snd_mixer_simple_basic_register(snd_mixer_t *mixer, err = find_module(class, top); if (err >= 0) err = snd_mixer_attach_hctl(mixer, priv->hctl); + if(err < 0) + goto __error; if (err >= 0) { priv->attach_flag = 1; err = snd_mixer_class_register(class, mixer); -- 1.7.0.4

1 0

[alsa-devel] [PATCH] *** Fixed issues/defects reported by Coverity tool ***
by sudarshan.bisht＠nokia.com 17 Mar '11

17 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> Coverity Static Analysis helps developers find hard-to-spot, yet potentially crash-causing defects early in the development phase, reducing the cost,time, and risk of software errors. This patch has a fix for conditions where 1) variable is used evern after getting freed thus causing junk assignment and 2) double free of memory. Sudarshan (1): alsa-lib: fixed coverity reported issues under "USE_AFTER_FREE" checker. src/control/control_hw.c | 1 + src/mixer/mixer.c | 2 +- 2 files changed, 2 insertions(+), 1 deletions(-)

3 3

[alsa-devel] some questions about sound system
by loody 17 Mar '11

17 Mar '11

Dear all: I just install Ubuntu9.10 on my laptop and I try to update my sound system and recompile the utility. there are some questions I met: 1. when configure the utility it says checking panel.h presence... no checking for panel.h... no configure: error: required curses helper header not found and config.log is "conftest.c:10:42: error: CoreFoundation/CFPreferences.h: No such file or directory conftest.c: In function 'main': conftest.c:14: error: 'NULL' undeclared (first use in this function) conftest.c:14: error: (Each undeclared identifier is repor" the above error seems I miss install some package before. I google the name but I have no improvement. "conftest.c:20:28: error: ac_nonexistent.h: No such file or directory configure:7397: $? = 1" the above error seems the header located where the conftest.c doesn't expect. how could solve them all? 2. there are some configure file in share when I compile advance linux sound architecture. like alsa.conf. have anyone know where I can find the naming rule? appreciate your help, miloody

4 7

[alsa-devel] ALC663 on ASUS G72GX Laptop
by nibs 16 Mar '11

16 Mar '11

I was advised to inquire about this here. Should the driver create two digital devices? _From the ALC663 datasheet: "In addition to the standard (primary) S/PDIF output function, the ALC663 features another independent (secondary) S/PDIF-OUT output and converters that transport digital audio output to a High Definition Media Interface (HDMI) transmitter (becoming more common in high-end PCs." More information here. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5310

1 0

[alsa-devel] UCM list 'modifiers' Seg-Fault when list is empty
by Alejandro Gonzalez 16 Mar '11

16 Mar '11

Hi, I've been testing the Alsa UCM and find that when listing 'modifiers' there is a segmentation fault if the list is empty. $ alsaucm -c Panda set _verb HiFi list _modifiers Im setting defaults list is empty Segmentation fault (core dumped) Using 'gdb' I see that the issue is trying to free a not NULL pointer. Program received signal SIGSEGV, Segmentation fault. 0x402a096c in free () from /lib/libc.so.6 (gdb) bt #0 0x402a096c in free () from /lib/libc.so.6 #1 0x40221224 in snd_use_case_free_list () from /usr/lib/libasound.so.2 #2 0x00008d8a in do_one (context=0x13008) at usecase.c:256 #3 do_commands (context=0x13008) at usecase.c:329 #4 0x000091aa in main (argc=<value optimized out>, argv=0xbe9986c4) at usecase.c:440 (gdb) quit But this can be easily resolved with the correct initialization of the '**list' pointer on the 'do_one' function in 'alsaucm/usecase.c'. You can find a patch attached to the mail. Regards, Alejandro

1 0

[alsa-devel] [PATCH] sound/core: merge list_del()/list_add_tail() to list_move_tail()
by Nicolas Kaiser 16 Mar '11

16 Mar '11

Merge list_del() + list_add_tail() to list_move_tail(). Signed-off-by: Nicolas Kaiser <nikai(a)nikai.net> --- sound/core/timer.c | 8 +++----- 1 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index ed01632..7c1cbf0 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -186,9 +186,8 @@ static void snd_timer_check_slave(struct snd_timer_instance *slave) list_for_each_entry(master, &timer->open_list_head, open_list) { if (slave->slave_class == master->slave_class && slave->slave_id == master->slave_id) { - list_del(&slave->open_list); - list_add_tail(&slave->open_list, - &master->slave_list_head); + list_move_tail(&slave->open_list, + &master->slave_list_head); spin_lock_irq(&slave_active_lock); slave->master = master; slave->timer = master->timer; @@ -414,8 +413,7 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) static int snd_timer_start1(struct snd_timer *timer, struct snd_timer_instance *timeri, unsigned long sticks) { - list_del(&timeri->active_list); - list_add_tail(&timeri->active_list, &timer->active_list_head); + list_move_tail(&timeri->active_list, &timer->active_list_head); if (timer->running) { if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) goto __start_now; -- 1.7.3.4

1 0

[alsa-devel] [PATCH] ASoC: Add National Semiconductor LM49352 Audio Codec support
by Reddy, MR Swami 16 Mar '11

16 Mar '11

Hello, Patch for adding National Semiconductor LM49352 Audio Codec (http://www.national.com/ds/LM/LM49352.pdf) support along with Samsung_6410 (i.e. SMDK6410) platform support for LM49352. [Tested this patch on SMDK6410 platform with Kernel- 2.6.24 and ALSA version is - 1.0.15]. Please review and let me know the comments/suggestion on this patch. And also let me know the forward-porting (to the latest ALSA version APIs) steps/process. Thanks in advance. Thanks Swami

2 6

[alsa-devel] aseqnet ipv6 and embedded clients
by errordeveloper＠gmail.com 16 Mar '11

16 Mar '11

Hi, I am looking at implementing wireless MIDI network using Contiki OS. I would run it on relativelly powerfull microcontroller (Freescale MC13224). I looked at sourcode of aseqnet command and it does look fairly small. Contiki has uIP 6loWPAN for networking and it connects to linux using SLIP-like 'tun' virtual userspace interface. I wonder if there could be any potential issues in terms of IPv6 compatibility, would aseqnet on Linux work with IPv6 or I will certainly need a tunnel ? Would be greate to hear _ANY_ concerns, of course I do imagine there will be a latency (hopefully not too huge) and I hope to work on this further. I also though of MIDI RTP, but that seems quite a bit bigger thing .. Could anyone give a little insight on what aseqnet really does? I would imagine it is quite plain, am I wrong ? Well, I could perhaps implement some very simplistic netcat/telnet kind of data dumping, but would be greate to get working smothly end- to-end, i.e. all the way from my wireless node (which will have a MIDI din5 connector) to an app on Linux, even amidi --dump ..hm Also would be greate to get some idea how clocking works w/ aseqnet. Cheers, -- Ilya

1 0

[alsa-devel] Wrong email in author-tag.
by Peter Rosin 16 Mar '11

16 Mar '11

Hi Takashi, This is not really important to /me/, on /this/ occation. But I thought I'd ask anyway as it seems strange that the author tag ended up wrong. If it is a symptom of more errors, others may care more. Personally I'm mainly wondering how it could happen, and would like for it to not happen again (especially if I ever send more substantial patches). The commit e2e9566230 ALSA: AT73C213: Rectify misleading comment. was first sent as a git patch [1] and then again amended with a pair of other (seemingly) trivial comment fixes [2] (the amended fixes were not commited for some reason). Both patches were sent from my axentia account and both carried signed-off-by tags using the axentia account. Yet the commit is made naming my lysator account as author (the sign-off is still intact). The patch that seems to have ended up being commited is [1], and what I think has happened is that I used the lysator account in the discussion following the 2nd patch, but I fail to understand how the sender of that one message (or was it a few?) could "leak" into the author tag. I'm assuming it was just a silly mistake, I'm mearly pointing out the mishap and understand that there is little one can do to fix this buglet this late. Cheers, Peter [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2010-November/033550.h… [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2010-November/033645.h…

2 1

[alsa-devel] [GIT PULL] Updates for 2.6.39
by Liam Girdwood 16 Mar '11

16 Mar '11

These should be the last updates for 2.6.39. Thanks Liam --- The following changes since commit 273de37655369e57919f19e157fafb2b4b1e4472: Merge branch 'for-2.6.39' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into for-2.6.39 (2011-02-01 14:55:10 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.39 Jarkko Nikula (5): ASoC: omap: rx51: Use gpio_request_one to configure tvout_sel gpio ASoC: omap: rx51: Add headset support ASoC: omap: rx51: Report headset insertion instead of video out cable ASoC: omap: rx51: Add FM transmitter support ASoC: omap: rx51: Enable McBSP2 sidetone sound/soc/omap/rx51.c | 42 ++++++++++++++++++++++++++++++++++++------ 1 files changed, 36 insertions(+), 6 deletions(-)

2 1

[alsa-devel] [PATCH 1/2 v3] ALSA: Add snd_ctl_update() to dynamically update a control
by Dimitris Papastamos 16 Mar '11

16 Mar '11

Add a function to dynamically update a given control. If the control does not already exist, a third parameter is used to determine whether to actually add that control. This is useful in cases where downloadable firmware at runtime can add or update existing controls. A separate patch needs to be made to allow ALSA Mixer to update the controls dynamically. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- Hi, Any potentially locking related issues have been fixed now. Any comments on whether it would be advisable to use the same `numid' when I update an existing control? Thanks, Dimitris include/sound/control.h | 1 + sound/core/control.c | 65 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 66 insertions(+), 0 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index e67db28..2a2fd82 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -113,6 +113,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new * kcontrolnew, v void snd_ctl_free_one(struct snd_kcontrol * kcontrol); int snd_ctl_add(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol); +int snd_ctl_update(struct snd_card * card, struct snd_kcontrol * kcontrol, bool add_on_update); int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id); int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id); int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, diff --git a/sound/core/control.c b/sound/core/control.c index db51e4e..1d2b524 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -368,6 +368,71 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) EXPORT_SYMBOL(snd_ctl_add); /** + * snd_ctl_update - update the control instance of the card + * @card: the card instance + * @kcontrol: the control instance to update + * @add_on_update: add the control if not already added + * + * Updates the given control. If the given control does not exist + * and the add_on_update flag is set, the control is added. If the + * control exists, it is destroyed first. + * + * Returns zero if successful, or a negative error code on failure. + * + * It frees automatically the control which cannot be added or updated. + */ +int snd_ctl_update(struct snd_card *card, struct snd_kcontrol *kcontrol, + bool add_on_update) +{ + struct snd_ctl_elem_id id; + unsigned int idx; + struct snd_kcontrol *old; + int ret; + + if (!kcontrol) + return -EINVAL; + if (snd_BUG_ON(!card || !kcontrol->info)) { + ret = -EINVAL; + goto error; + } + id = kcontrol->id; + down_write(&card->controls_rwsem); + old = snd_ctl_find_id(card, &id); + if (!old) { + if (add_on_update) + goto add; + up_write(&card->controls_rwsem); + ret = -EINVAL; + goto error; + } + ret = snd_ctl_remove(card, old); + if (ret < 0) { + up_write(&card->controls_rwsem); + goto error; + } +add: + if (snd_ctl_find_hole(card, kcontrol->count) < 0) { + up_write(&card->controls_rwsem); + ret = -ENOMEM; + goto error; + } + list_add_tail(&kcontrol->list, &card->controls); + card->controls_count += kcontrol->count; + kcontrol->id.numid = card->last_numid + 1; + card->last_numid += kcontrol->count; + up_write(&card->controls_rwsem); + for (idx = 0; idx < kcontrol->count; idx++, id.index++, id.numid++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id); + return 0; + +error: + snd_ctl_free_one(kcontrol); + return ret; +} + +EXPORT_SYMBOL(snd_ctl_update); + +/** * snd_ctl_remove - remove the control from the card and release it * @card: the card instance * @kcontrol: the control instance to remove -- 1.7.4.1

2 2

[alsa-devel] [PATCH] ALSA: HDA: Correct address for SSYNC register
by David Henningsson 16 Mar '11

16 Mar '11

Here's something I stumbled upon, while debugging something probably unrelated. I don't exactly know the implications of failing multiple stream synchronisation (is that something we commonly use?), or the implications of writing to a register that is "Reserved" in the HDA Spec (can there be crazy vendor magic in there we're unwillingly messing with?). Perhaps someone else knows more and can tell if this is something that is actually important. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] RFC: ST specific ASoC sound driver
by rajeev 16 Mar '11

16 Mar '11

Hi, I have developed a ASoC sound driver for a ST specific platform (SPEAr). This module actually uses the Designware (Synopsys) "dw_apb_i2s_db" IP + some ST specific glue logic on top of it. Now I am confused as to whether I should submit a single patch for the SPEAr ASoC driver or should I try to break the driver such that the Designware and glue logic parts are handled separately (which given the current shape of design seems improbable) Any pointers on the same will be highly appreciated. Best Regards Rajeev

2 1

[PATCH] ALSA: HDA: Correct address for SYNC register
by David Henningsson 16 Mar '11

16 Mar '11

The SSYNC register is located at address 0x38 according to the HDA specification. Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 70a9d32..ab3f654 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -176,7 +176,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #define ICH6_REG_INTCTL 0x20 #define ICH6_REG_INTSTS 0x24 #define ICH6_REG_WALLCLK 0x30 /* 24Mhz source */ -#define ICH6_REG_SYNC 0x34 +#define ICH6_REG_SYNC 0x38 #define ICH6_REG_CORBLBASE 0x40 #define ICH6_REG_CORBUBASE 0x44 #define ICH6_REG_CORBWP 0x48 -- 1.7.1 --------------050206000100040505070002--

1 0

[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 16 Mar '11

16 Mar '11

The following changes since commit 80e1b84ffe1354f314fd85c939900b816f394e5a: Merge branch 'for-2.6.38' into for-2.6.39 (2011-03-10 14:16:35 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 A final batch of updates for 2.6.39 - these have been sitting in my -next tree for several days now. Christian Glindkamp (1): ASoC: Add MAX9850 codec driver Mark Brown (1): ASoC: Fix spacing in MAX8950 Takashi Iwai (1): ALSA: Add snd_ctl_activate_id() Vasily Khoruzhick (2): ASoC: PXA: z2: Mute internal speaker when headphones are connected ASoC: PXA: Z2: Fix codec pin name include/sound/control.h | 2 + sound/core/control.c | 46 +++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9850.c | 389 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max9850.h | 38 +++++ sound/soc/pxa/z2.c | 7 +- 7 files changed, 487 insertions(+), 1 deletions(-) create mode 100644 sound/soc/codecs/max9850.c create mode 100644 sound/soc/codecs/max9850.h

3 3

[alsa-devel] [PATCHv2] omap3: alsa-soc: Remove restrictive checks for cpu type
by Sanjeev Premi 16 Mar '11

16 Mar '11

Current checks for cpu type were too restrictive leading to failures for other silicons in same family. The problem was found while testing audio playback on AM37x and AM35x processors. But should exist on OMAP36xx as well. Signed-off-by: Sanjeev Premi <premi(a)ti.com> --- Changes in this version: 1) Added one more check to the changes. 2) Updated commit message to use plurals. sound/soc/omap/omap-mcbsp.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2175f09..a88a0bf 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -146,7 +146,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) */ - if (cpu_is_omap343x() || cpu_is_omap44xx()) { + if (cpu_is_omap34xx() || cpu_is_omap44xx()) { /* * Rule for the buffer size. We should not allow * smaller buffer than the FIFO size to avoid underruns @@ -258,7 +258,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (cpu_is_omap343x()) { + if (cpu_is_omap34xx()) { dma_data->set_threshold = omap_mcbsp_set_threshold; /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ if (omap_mcbsp_get_dma_op_mode(bus_id) == -- 1.7.2.2

1 0

Re: [alsa-devel] [PATCHv2] omap3: alsa-soc: Remove restrictive checks for cpu type
by Premi, Sanjeev 16 Mar '11

16 Mar '11

> -----Original Message----- > From: Premi, Sanjeev > Sent: Monday, March 14, 2011 8:53 PM > To: linux-omap(a)vger.kernel.org; alsa-devel(a)alsa-project.org; > linux-arm-kernel(a)lists.infradead.org > Cc: Premi, Sanjeev > Subject: [PATCHv2] omap3: alsa-soc: Remove restrictive checks > for cpu type > > Current checks for cpu type were too restrictive leading > to failures for other silicons in same family. > > The problem was found while testing audio playback on > AM37x and AM35x processors. But should exist on OMAP36xx > as well. > > Signed-off-by: Sanjeev Premi <premi(a)ti.com> > --- > Changes in this version: > 1) Added one more check to the changes. > 2) Updated commit message to use plurals. > > sound/soc/omap/omap-mcbsp.c | 4 ++-- > 1 files changed, 2 insertions(+), 2 deletions(-) [sp] This patch didn't seem to be lost in transit! Sending it again. Did anyone else ever face this issue? ~sanjeev

1 0

[alsa-devel] [PATCH 1/2 v2] ALSA: Add snd_ctl_update() to dynamically update a control
by Dimitris Papastamos 16 Mar '11

16 Mar '11

Add a function to dynamically update a given control. If the control does not already exist, a third parameter is used to determine whether to actually add that control. This is useful in cases where downloadable firmware at runtime can add or update existing controls. A separate patch needs to be made to make ALSA Mixer cope with this. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/control.h | 1 + sound/core/control.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 47 insertions(+), 0 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index e67db28..12e4de7 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -113,6 +113,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new * kcontrolnew, v void snd_ctl_free_one(struct snd_kcontrol * kcontrol); int snd_ctl_add(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol); +int snd_ctl_update(struct snd_card * card, struct snd_kcontrol * kcontrol, int add_on_update); int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id); int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id); int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, diff --git a/sound/core/control.c b/sound/core/control.c index db51e4e..9c5d7a9 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -397,6 +397,52 @@ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) EXPORT_SYMBOL(snd_ctl_remove); /** + * snd_ctl_update - update the control instance of the card + * @card: the card instance + * @kcontrol: the control instance to update + * @add_on_update: add the control if not already added + * + * Updates the control instance created via snd_ctl_new() or + * snd_ctl_new1() of the given card. + * + * Returns zero if successful, or a negative error code on failure. + * + */ +int snd_ctl_update(struct snd_card *card, struct snd_kcontrol *kcontrol, + int add_on_update) +{ + struct snd_ctl_elem_id id; + struct snd_kcontrol *old; + int ret; + + if (!kcontrol) + return -EINVAL; + if (snd_BUG_ON(!card || !kcontrol->info)) + return -EINVAL; + id = kcontrol->id; + down_write(&card->controls_rwsem); + old = snd_ctl_find_id(card, &id); + if (!old) { + up_write(&card->controls_rwsem); + if (add_on_update) + return snd_ctl_add(card, kcontrol); + return -EINVAL; + } + ret = snd_ctl_remove(card, old); + if (ret < 0) { + up_write(&card->controls_rwsem); + return ret; + } + up_write(&card->controls_rwsem); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) + return ret; + return 0; +} + +EXPORT_SYMBOL(snd_ctl_update); + +/** * snd_ctl_remove_id - remove the control of the given id and release it * @card: the card instance * @id: the control id to remove -- 1.7.4.1

4 7