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[alsa-devel] [PATCH 5/5 v1] ASoC Adds DSP DAI format support for platform driver
by Arun KS 30 Sep '08

30 Sep '08

Enables DSP DAI format for mcbsp in omap platform driver Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/omap-mcbsp.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 35310e1..fb920e1 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -245,6 +245,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_DSP_A: + /* 0-bit data delay */ + regs->rcr2 |= RDATDLY(0); + regs->xcr2 |= XDATDLY(0); + break; default: /* Unsupported data format */ return -EINVAL; -- 1.5.3.4

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[alsa-devel] [PATCH 3/5 v1] Adds I2C board info for tlvaic23
by Arun KS 30 Sep '08

30 Sep '08

Adding I2C board info required for tlvaic23 i2c chip driver Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- arch/arm/mach-omap1/board-osk.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-omap1/board-osk.c b/arch/arm/mach-omap1/board-osk.c index 3e766e4..e506aef 100644 --- a/arch/arm/mach-omap1/board-osk.c +++ b/arch/arm/mach-omap1/board-osk.c @@ -258,8 +258,10 @@ static struct i2c_board_info __initdata osk_i2c_board_info[] = { .platform_data = &tps_board, }, + { + I2C_BOARD_INFO("tlv320aic23", 0x1B), + }, /* TODO when driver support is ready: - * - aic23 audio chip at 0x1a * - on Mistral, 24c04 eeprom at 0x50 * - optionally on Mistral, ov9640 camera sensor at 0x30 */ -- 1.5.3.4

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[alsa-devel] [PATCH 0/5 v1] ASoC Support for TLVaic23b audio codec
by Arun KS 30 Sep '08

30 Sep '08

Hi all, I have incorporated the comments from Mark Brown, Jarkko Nikula and Felipe Balbi. The following series of patches are against the tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git The new series of patches have the following changes from previous patches: 1) The codec is named tlv320aic23 uniformly throughout the code. 2) Removed set_bias_level function from the codec driver and is switching on/off the codec at resume/suspend function. 3) Added mic sidetone and line bypass. 4) Renamed the machine driver file as 5912osk.c instead of osk.c Thanks Arun

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[alsa-devel] [PATCH] ASoC: Fix cs4270 error path
by Jean Delvare 30 Sep '08

30 Sep '08

The error path in cs4270_probe/cs4270_remove is pretty broken: * If cs4270_probe fails, codec is leaked. * If snd_soc_register_card fails, cs4270_i2c_driver stays registered. * If I2C support is enabled but no I2C device is found, i2c_del_driver is never called (neither in cs4270_probe nor in cs4270_remove. Fix all 3 problems by implementing a clean error path in cs4270_probe and jumping to its labels as needed. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Acked-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 25 ++++++++++++++++++------- 1 file changed, 18 insertions(+), 7 deletions(-) --- linux-2.6.27-rc8.orig/sound/soc/codecs/cs4270.c 2008-09-30 11:11:22.000000000 +0200 +++ linux-2.6.27-rc8/sound/soc/codecs/cs4270.c 2008-09-30 11:33:03.000000000 +0200 @@ -681,7 +681,7 @@ static int cs4270_probe(struct platform_ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "cs4270: failed to create PCMs\n"); - return ret; + goto error_free_codec; } #ifdef USE_I2C @@ -690,8 +690,7 @@ static int cs4270_probe(struct platform_ ret = i2c_add_driver(&cs4270_i2c_driver); if (ret) { printk(KERN_ERR "cs4270: failed to attach driver"); - snd_soc_free_pcms(socdev); - return ret; + goto error_free_pcms; } /* Did we find a CS4270 on the I2C bus? */ @@ -713,10 +712,23 @@ static int cs4270_probe(struct platform_ ret = snd_soc_register_card(socdev); if (ret < 0) { printk(KERN_ERR "cs4270: failed to register card\n"); - snd_soc_free_pcms(socdev); - return ret; + goto error_del_driver; } + return 0; + +error_del_driver: +#ifdef USE_I2C + i2c_del_driver(&cs4270_i2c_driver); + +error_free_pcms: +#endif + snd_soc_free_pcms(socdev); + +error_free_codec: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; } @@ -727,8 +739,7 @@ static int cs4270_remove(struct platform snd_soc_free_pcms(socdev); #ifdef USE_I2C - if (socdev->codec->control_data) - i2c_del_driver(&cs4270_i2c_driver); + i2c_del_driver(&cs4270_i2c_driver); #endif kfree(socdev->codec); -- Jean Delvare

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[alsa-devel] Support for ALC1200
by Norbert Piotrowski 30 Sep '08

30 Sep '08

I have a question about ALC1200 which is used by ASUS in P5Q motherboard. Unfortunately I have that board and problems with spdif output ( simply no light, it's not a problem with mute/unmute IEC958 ). ALC1200 is recognized as ALC883, below more details. Realtek don't publish specification for that chip because it was made for Asus request. From what i gathered, 1200 is 890 in disguise, but I'm not sure about that. I would like to know if there is any chance to add support in alsa driver for that kind of codec in the near future ? I will be very grateful for any help or suggestion. Best regards, Norbert norbert@nelson / $ uname -a Linux nelson 2.6.24-16-generic #1 SMP Thu Apr 10 13:23:42 UTC 2008 i686 GNU/Linux norbert@nelson / $ alsactl -v alsactl version 1.0.18rc3 norbert@nelson / $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC883 Digital [ALC883 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 norbert@nelson / $ aplay -L default:CARD=Intel HDA Intel, ALC883 Analog Default Audio Device front:CARD=Intel,DEV=0 HDA Intel, ALC883 Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, ALC883 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, ALC883 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, ALC883 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, ALC883 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, ALC883 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=Intel,DEV=0 HDA Intel HDMI Audio Output null Discard all samples (playback) or generate zero samples (capture) norbert@nelson / $ cat /proc/asound/card0/codec#0 Codec: Realtek ALC888 Address: 0 Vendor Id: 0x10ec0888 Subsystem Id: 0x104382fe Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x13 0x13] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x05 0x0b Node 0x10 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x99430140: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x10 Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Trigger Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Trigger Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0836: IN OUT Detect Trigger Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0836: IN OUT Detect Trigger Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19850: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19d60: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x6, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181305f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x5, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x08373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02214d20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4015e601: [N/A] Speaker at Ext N/A Conn = Optical, Color = White DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x01447130: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Yellow DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x25 0x0b

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[alsa-devel] sound tree slightly rebased
by Takashi Iwai 30 Sep '08

30 Sep '08

Hi, since I replaced Jean's patch to fix Vladimir's e-mail address, I had to rebase topic/asoc, master and for-next branches. If you are based on it, please be careful at the next pull. Sorry for inconvenience. Takashi

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[alsa-devel] AIC23 codec ADC problem
by Troy Kisky 30 Sep '08

30 Sep '08

Hi All I'm using the TLV320AIC23 to record sound from a mic. It sounds fine when I play it back, but when I look at the data, the value in one channel is always exactly double the value in the other. Since the mic gives the same input to both ADC's I was expecting to have the same value in the left & right channels. Has anyone seen this before? Any guesses to what I might be doing wrong? Can someone using an AIC23 check their data for me? I'm using a davinci DM644x platform and spi interface to the codec in dsp_a mode. sample data 00ec40 fffeffff 00040002 00040002 00000000 00ec50 fffeffff fffeffff 00000000 fffafffd 00ec60 fffeffff 00020001 00040002 00080004 00ec70 000a0005 0016000b 0014000a 001a000d 00ec80 001c000e 001e000f 00200010 00260013 00ec90 001e000f 001e000f 001e000f 001c000e 00eca0 001c000e 0018000c 0014000a 0014000a 00ecb0 0014000a 000e0007 00120009 0014000a ^ ^ ^ ^ Always even After swapping the left & right channels at the codec 011af0 fffcfffe fffeffff 00060003 00060003 011b00 000a0005 000c0006 000e0007 000c0006 011b10 00080004 00080004 00060003 00060003 011b20 00080004 00060003 00060003 00040002 011b30 00020001 00020001 00000000 00020001 011b40 fffeffff 00000000 00040002 00060003 011b50 000c0006 000e0007 00100008 0014000a 011b60 00120009 000e0007 00100008 00100008 011b70 000e0007 00080004 00080004 00060003 011b80 00060003 00060003 00060003 000a0005 No change, so the ADC's both do convert to the same value. But there appears to be a 1 bit delay between the two channels. My McBsp is reading 1 32 bit element, so I don't think it can shift only half of the element. After switching to dsp_b mode (an extra bit delay before data starts) 00b130 fff8fff8 fff5fff5 fffafffa fffcfffc 00b140 fffefffe 00010001 00000000 00010001 00b150 00020002 00010001 00010001 00010001 00b160 00000000 ffffffff 00020002 00000000 00b170 00000000 00000000 fffefffe fffefffe 00b180 fffdfffd fffdfffd 00000000 ffffffff 00b190 00020002 00010001 00020002 00030003 00b1a0 00030003 00030003 00050005 00030003 00b1b0 00050005 00030003 00010001 00000000 When I play this back in dsp_b mode I hear random soft popping from the speaker, but it is loud. Playing it in dsp_a mode is quiet with no popping. So, I conclude that recording with the codec is buggy in dsp_a mode and playback is buggy in dsp_b mode. I hope someone will tell me I'm wrong. Thanks Troy

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[alsa-devel] maintaining fixed latency between output and input?
by Aaron J. Grier 30 Sep '08

30 Sep '08

what's the most straight-forward way to maintain fixed latency between output and input without sitting in loop and continuously checking snd_pcm_avail_update() for the output and input descriptors? I was originally going to use callbacks, but I've recently read that is not a good idea. poll() seems to be the reccomended method. I could use separate threads for output and input, but then I have to perform synchronization between them, marshal external requests between the two threads, and make a lot more work for myself. I'm visualizing something like this: for(;;) { poll() if (output_is_ready) { /* write output buffer */ /* do output-side housekeeping */ } if (input_is_ready) { /* read input buffer */ /* do input-side housekeeping */ } /* communicate with other threads */ } combining the output and input snd_pcm_t into a single descriptor list for poll() seems straight-forward, but can I pass this superset list to snd_pcm_poll_descriptors_revents() without it getting confused? -- Aaron J. Grier | "Not your ordinary poofy goof." | agrier(a)poofygoof.com

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[alsa-devel] [PATCH 0/5] [RFC] ALSA ASoC Support for TLVaic23b audio codec
by Arun KS 30 Sep '08

30 Sep '08

Hi all, The following patches adds ASOC driver for TLVaic23b audio codec. Tested playback and capture on omap 5912osk board with some hacks in the omap platform and machine driver. Thanks, Arun -- 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

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Re: [alsa-devel] Secure delivery
by ink＠jurassic.park.msu.ru 29 Sep '08

29 Sep '08

Bad Gateway: The message has been attached. ++++ Attachment: No Virus found ++++ F-Secure AntiVirus - www.f-secure.com

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