April 2011 - Alsa-devel - mailman.alsa-project.org

[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

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[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,

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[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--

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[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

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