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

[alsa-devel] Realtek ALC268, Acer 4315 ?Use mic port as independent output?
by Eliot Blennerhassett 01 Nov '08

01 Nov '08

Greetings, I have acer 4315 laptop with realtek alc268 codec. (It also has hda modem which I don't use.) Looking at the datasheet, the codec has 2 PCM outs, but alsa hda driver with model=acer shows only one. I want to know if I can enable the 2nd output, and use the external mic jack as a second line out? I can see that PCM2/LOUT2 connects to HP-OUT (widget 0x15), while PCM1/LOUT1 connects to several pins 0x[14,15,16,18,1A] How do I go about testing this. Does the hda model=test enable *every* widget and PCM and allow change of pin direction etc? BTW I want to use the 2nd output as the secondar "preview" (prelisten?) headphone output for my audio player (pytone), while main output is going to speakers. TIA for any advice -- Eliot !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 1025:0133 !!Modprobe options (Sound related) !!-------------------------------- snd-hda-intel: enable=1 index=0 model=acer snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : 0,-1,-1,-1,-1,-1,-1,-1 model : acer,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x10250133 Revision Id: 0x100003 No Modem Function Group found !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC268 Analog [ALC268 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC268 Analog [ALC268 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0

3 3

[alsa-devel] tlv320aic23b soft mute weirdness
by Troy Kisky 01 Nov '08

01 Nov '08

I'm having trouble making the soft mute bit work correctly. I was wondering if anyone else was experiencing this issue. The soft mute (reg 5, bit 3) does mute the left or the right speaker, depending on whether or not channel swap is set reg 7, bit 5). But it makes the other channel louder. Does this sound familiar to anyone? Can anyone verify the problem? I don't see how our board design can cause this, but I can't rule it out. I've duplicated the result on 3 of our boards. Thanks Troy

1 0

[alsa-devel] [PATCH 1/1] [ALSA]TLV320AIC23B: Support more sample rates
by Troy Kisky 01 Nov '08

01 Nov '08

Add support for more sample rates, different crystals and split playback/capture rates. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> Acked-by: Arun KS <arunks(a)mistralsolutions.com> diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 44308da..c1b2060 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -37,12 +37,6 @@ #define AIC23_VERSION "0.1" -struct tlv320aic23_srate_reg_info { - u32 sample_rate; - u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ - u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ -}; - /* * AIC23 register cache */ @@ -261,19 +255,155 @@ static const struct snd_soc_dapm_route intercon[] = { }; -/* tlv320aic23 related */ -static const struct tlv320aic23_srate_reg_info srate_reg_info[] = { - {4000, 0x06, 1}, /* 4000 */ - {8000, 0x06, 0}, /* 8000 */ - {16000, 0x0C, 1}, /* 16000 */ - {22050, 0x11, 1}, /* 22050 */ - {24000, 0x00, 1}, /* 24000 */ - {32000, 0x0C, 0}, /* 32000 */ - {44100, 0x11, 0}, /* 44100 */ - {48000, 0x00, 0}, /* 48000 */ - {88200, 0x1F, 0}, /* 88200 */ - {96000, 0x0E, 0}, /* 96000 */ +/* AIC23 driver data */ +struct aic23 { + struct snd_soc_codec codec; + int mclk; + int requested_adc; + int requested_dac; +}; + +/* + * Common Crystals used + * 11.2896 Mhz /128 = *88.2k /192 = 58.8k + * 12.0000 Mhz /125 = *96k /136 = 88.235K + * 12.2880 Mhz /128 = *96k /192 = 64k + * 16.9344 Mhz /128 = 132.3k /192 = *88.2k + * 18.4320 Mhz /128 = 144k /192 = *96k + */ + +/* + * Normal BOSR 0-256/2 = 128, 1-384/2 = 192 + * USB BOSR 0-250/2 = 125, 1-272/2 = 136 + */ +static const int bosr_usb_divisor_table[] = { + 128, 125, 192, 136 +}; +#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7)) +#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15)) +static const unsigned short sr_valid_mask[] = { + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/ + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ + LOWER_GROUP, /* Usb, bosr - 0*/ + UPPER_GROUP, /* Usb, bosr - 1*/ +}; +/* + * Every divisor is a factor of 11*12 + */ +#define SR_MULT (11*12) +#define A(x) (x) ? (SR_MULT/x) : 0 +static const unsigned char sr_adc_mult_table[] = { + A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1), + A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1) }; +static const unsigned char sr_dac_mult_table[] = { + A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1), + A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1) +}; + +unsigned get_score(int adc, int adc_l, int adc_h, int need_adc, + int dac, int dac_l, int dac_h, int need_dac) +{ + if ((adc >= adc_l) && (adc <= adc_h) && + (dac >= dac_l) && (dac <= dac_h)) { + unsigned diff_adc = need_adc - adc; + unsigned diff_dac = need_dac - dac; + unsigned score; + if (((int)diff_adc) < 0) + diff_adc = -diff_adc; + if (((int)diff_dac) < 0) + diff_dac = -diff_dac; + score = diff_adc + diff_dac; + return score; + } + return 0xffffffff; +} +int find_rate(int mclk, u32 need_adc, u32 need_dac) +{ + int i, j; + int best_i = -1; + int best_j = -1; + int best_div = 0; + unsigned best_score = 0xffffffff; + int adc_l, adc_h, dac_l, dac_h; + + need_adc *= SR_MULT; + need_dac *= SR_MULT; + /* + * rates given are +/- 1/32 + */ + adc_l = need_adc - (need_adc >> 5); + adc_h = need_adc + (need_adc >> 5); + dac_l = need_dac - (need_dac >> 5); + dac_h = need_dac + (need_dac >> 5); + for (i = 0; i < 4; i++) { + int base = mclk / bosr_usb_divisor_table[i]; + int mask = sr_valid_mask[i]; + for (j = 0; j < 16; j++, mask >>= 1) { + int adc; + int dac; + int score; + if ((mask & 1) == 0) + continue; + adc = base * sr_adc_mult_table[j]; + dac = base * sr_dac_mult_table[j]; + score = get_score(adc, adc_l, adc_h, need_adc, + dac, dac_l, dac_h, need_dac); + if (best_score > score) { + best_score = score; + best_i = i; + best_j = j; + best_div = 0; + } + score = get_score((adc >> 1), adc_l, adc_h, need_adc, + (dac >> 1), dac_l, dac_h, need_dac); + /* prefer to have a /2 */ + if ((score != 0xffffffff) && (best_score >= score)) { + best_score = score; + best_i = i; + best_j = j; + best_div = 1; + } + } + } + return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT); +} + +static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, + u32 *sample_rate_adc, u32 *sample_rate_dac) +{ + int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE); + int sr = (src >> 2) & 0x0f; + int val = (mclk / bosr_usb_divisor_table[src & 3]); + int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; + int dac = (val * sr_dac_mult_table[sr]) / SR_MULT; + if (src & TLV320AIC23_CLKIN_HALF) { + adc >>= 1; + dac >>= 1; + } + *sample_rate_adc = adc; + *sample_rate_dac = dac; +} + +static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, + u32 sample_rate_adc, u32 sample_rate_dac) +{ + /* Search for the right sample rate */ + int data = find_rate(mclk, sample_rate_adc, sample_rate_dac); + if (data < 0) { + printk(KERN_ERR "%s:Invalid rate %u,%u requested\n", + __func__, sample_rate_adc, sample_rate_dac); + return -EINVAL; + } + tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + if (1) { + int adc, dac; + get_current_sample_rates(codec, mclk, &adc, &dac); + printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n", + adc, dac, data); + } + return 0; +} static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) { @@ -293,27 +423,30 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; - u16 iface_reg, data; - u8 count = 0; + u16 iface_reg; + int ret; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); + u32 sample_rate_adc = aic23->requested_adc; + u32 sample_rate_dac = aic23->requested_dac; + u32 sample_rate = params_rate(params); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + aic23->requested_dac = sample_rate_dac = sample_rate; + if (!sample_rate_adc) + sample_rate_adc = sample_rate; + } else { + aic23->requested_adc = sample_rate_adc = sample_rate; + if (!sample_rate_dac) + sample_rate_dac = sample_rate; + } + ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc, + sample_rate_dac); + if (ret < 0) + return ret; iface_reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); - - /* Search for the right sample rate */ - /* Verify what happens if the rate is not supported - * now it goes to 96Khz */ - while ((srate_reg_info[count].sample_rate != params_rate(params)) && - (count < ARRAY_SIZE(srate_reg_info))) { - count++; - } - - data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) | - (srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) | - TLV320AIC23_USB_CLK_ON; - - tlv320aic23_write(codec, TLV320AIC23_SRATE, data); - switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; @@ -349,12 +482,17 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); /* deactivate */ if (!codec->active) { udelay(50); tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + aic23->requested_dac = 0; + else + aic23->requested_adc = 0; } static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) @@ -422,12 +560,9 @@ static int tlv320aic23_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; - - switch (freq) { - case 12000000: - return 0; - } - return -EINVAL; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); + aic23->mclk = freq; + return 0; } static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, @@ -659,14 +794,15 @@ static int tlv320aic23_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; + struct aic23 *aic23; int ret = 0; printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) + aic23 = kzalloc(sizeof(struct aic23), GFP_KERNEL); + if (aic23 == NULL) return -ENOMEM; - + codec = &aic23->codec; socdev->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); @@ -687,6 +823,7 @@ static int tlv320aic23_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); if (codec->control_data) tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -697,7 +834,7 @@ static int tlv320aic23_remove(struct platform_device *pdev) i2c_del_driver(&tlv320aic23_i2c_driver); #endif kfree(codec->reg_cache); - kfree(codec); + kfree(aic23); return 0; } -- 1.5.4.3

2 1

[alsa-devel] [PATCH 0/5] ASoC driver updates
by Mark Brown 31 Oct '08

31 Oct '08

The following changes since commit 57b41898c2ecd13a9d338b66ef23f66caab5c4e9: Stephen Rothwell (1): ALSA: ASoC - restore removed variable declaration are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Note that this series was generated with git format-patch -M and so won't apply with ordianary diff. Sedji Gaouaou (2): ASoC: Merge AT91 and AVR32 support into a single atmel architecture ASoC: Add audio support for the Atmel AT91SAM9G20ek board(uing wolfson 8731). Steve Sakoman (3): ASoC: Add support for TWL4030 audio codec ASoC: Add support for Gumstix Overo ASoC: Add support for Beagleboard sound/soc/Kconfig | 3 +- sound/soc/Makefile | 2 +- sound/soc/at32/Kconfig | 34 - sound/soc/at32/Makefile | 11 - sound/soc/at32/at32-pcm.h | 79 -- sound/soc/at32/at32-ssc.c | 849 ---------------------- sound/soc/at32/at32-ssc.h | 59 -- sound/soc/at91/Kconfig | 10 - sound/soc/at91/Makefile | 6 - sound/soc/at91/at91-pcm.c | 434 ----------- sound/soc/at91/at91-pcm.h | 72 -- sound/soc/at91/at91-ssc.c | 791 -------------------- sound/soc/at91/at91-ssc.h | 27 - sound/soc/atmel/Kconfig | 43 ++ sound/soc/atmel/Makefile | 15 + sound/soc/{at32/at32-pcm.c => atmel/atmel-pcm.c} | 376 +++++----- sound/soc/atmel/atmel-pcm.h | 86 +++ sound/soc/atmel/atmel_ssc_dai.c | 782 ++++++++++++++++++++ sound/soc/atmel/atmel_ssc_dai.h | 121 +++ sound/soc/{at32 => atmel}/playpaq_wm8510.c | 4 +- sound/soc/atmel/sam9g20_wm8731.c | 329 +++++++++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl4030.c | 653 +++++++++++++++++ sound/soc/codecs/twl4030.h | 197 +++++ sound/soc/omap/Kconfig | 10 + sound/soc/omap/Makefile | 3 + sound/soc/omap/omap3beagle.c | 149 ++++ sound/soc/omap/overo.c | 148 ++++ 29 files changed, 2731 insertions(+), 2569 deletions(-) delete mode 100644 sound/soc/at32/Kconfig delete mode 100644 sound/soc/at32/Makefile delete mode 100644 sound/soc/at32/at32-pcm.h delete mode 100644 sound/soc/at32/at32-ssc.c delete mode 100644 sound/soc/at32/at32-ssc.h delete mode 100644 sound/soc/at91/Kconfig delete mode 100644 sound/soc/at91/Makefile delete mode 100644 sound/soc/at91/at91-pcm.c delete mode 100644 sound/soc/at91/at91-pcm.h delete mode 100644 sound/soc/at91/at91-ssc.c delete mode 100644 sound/soc/at91/at91-ssc.h create mode 100644 sound/soc/atmel/Kconfig create mode 100644 sound/soc/atmel/Makefile rename sound/soc/{at32/at32-pcm.c => atmel/atmel-pcm.c} (50%) create mode 100644 sound/soc/atmel/atmel-pcm.h create mode 100644 sound/soc/atmel/atmel_ssc_dai.c create mode 100644 sound/soc/atmel/atmel_ssc_dai.h rename sound/soc/{at32 => atmel}/playpaq_wm8510.c (99%) create mode 100644 sound/soc/atmel/sam9g20_wm8731.c create mode 100644 sound/soc/codecs/twl4030.c create mode 100644 sound/soc/codecs/twl4030.h create mode 100644 sound/soc/omap/omap3beagle.c create mode 100644 sound/soc/omap/overo.c

2 6

Re: [alsa-devel] TASCAM US122L
by Philippe CARRIERE 31 Oct '08

31 Oct '08

I discovered your previous email to alsa-devel recently. Had you a look to my further mail http://mailman.alsa-project.org/pipermail/alsa-devel/2008-October/011339.ht… which possibly answer part of your questions ? A patch is required for ehci-hcd module to access pcm emulation. Regards Phil >At Mon, 6 Oct 2008 13:11:48 +0200, >Fernando Perfumo wrote: >> >> >> Hello all, >> >> following is an extract of kern.log when connecting and disconecting a >> TASCAM US122L : >> ( the pcm part of the us122L does not appear in /proc/asound/... but only >> the midi part. >> kernel : 2.6.27-rc5-git11, alsa-driver-1.0.18rc3 compiled separately of >> the kernel sources, alsa-lib-1.0.18rc3 and alsa-utils-1.0.18rc3 ) > >Most messages are from snd_printdd(), so it's no bug but verbose messages. > >> I'd appreciate some ideas about the segfault in alsactl and the ausence of >> the pcm functionality.. > >alsactl segfault seems irrelevant. Are you using the latest git version? > >thanks, > >Takashi

1 0

[alsa-devel] Questions about the changes introduced in alsa-utils 1.0.18
by Ozan Çağlayan 31 Oct '08

31 Oct '08

Hi, AFAIK, we may now have the possibility to drop salsa helper and use alsactl init instead of it in the udev rules file for alsa. I wonder how our distribution's old rules file should be modified in order to reflect this change. I looked at the rules file found in fedora but we were also doing some renaming and grouping in our rules file so I thought that this list would be a right place to get some ideas. Here's our rules file that we're using with 1.0.17: SUBSYSTEM=="sound", GROUP="audio" SUBSYSTEM=="snd", GROUP="audio" KERNEL=="controlC[0-9]*", NAME="snd/%k" KERNEL=="hwC[D0-9]*", NAME="snd/%k" KERNEL=="pcmC[D0-9cp]*", NAME="snd/%k" KERNEL=="midiC[D0-9]*", NAME="snd/%k" KERNEL=="timer", NAME="snd/%k" KERNEL=="seq", NAME="snd/%k" SUBSYSTEM=="sound", KERNEL=="controlC*" RUN+="/usr/sbin/salsa" SUBSYSTEM=="sound", KERNEL=="pcm*" RUN+="/usr/sbin/salsa" Fedora uses only two restore/store actions in their new package: ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", \ RUN+="/sbin/alsactl -E ALSA_CONFIG_PATH=/etc/alsa/alsactl.conf --initfile=/lib/alsa/init/00main restore /dev/$name" ACTION=="remove", SUBSYSTEM=="sound", KERNEL=="controlC*", \ RUN+="/sbin/alsactl -E ALSA_CONFIG_PATH=/etc/alsa/alsactl.conf store /dev/$name" And also ALSA_CONFIG_PATH was pointing to /etc/asound.state in salsa, is it safe and sufficient to change it to /etc/alsa/alsactl.conf in the udev rules file or do we need any other modifications in the code base? And finally, what are the pros and cons of compiling alsa-plugins with speex support? What's the trade off if it is disabled? Thanks, -- Ozan Çağlayan http://www.pardus.org.tr/eng <ozan_at_pardus.org.tr>

1 0

[alsa-devel] Ubuntu - Alsa Installation Script
by Klaus Schulz 31 Oct '08

31 Oct '08

Hi folks. I prepared an unofficial Alsa-Installation (now 1.0.18) script for the Ubuntu Community. The intention is mainly to be up2date on the drivers, since this - the lack or limited functionality of drivers (I am aware of the reasons behind it) is IMO still one of the biggest weaknesses within Linux. Therefore it is IMO crucial to have the latest driver available. As you know most of the distributions make things worse since these are usually one major release behind. http://ubuntuforums.org/showthread.php?t=962695 For standard installations the script seems to work fine. Quite some people have been/are using it. And of course responding to it - mainly about problems on specific sound cards. I am wondering if things can be improved in the script. I would appreciate if somebody of the designers around here could have a quick look at it. I am not so deep into Alsa that I can cover severe problems brought up over there. How would you recommend to handle this? THX. Cheers Klaus

4 5

[alsa-devel] ASOC deprecated warning
by Jon Smirl 31 Oct '08

31 Oct '08

What's the status of this? CC sound/soc/soc-dapm.o sound/soc/soc-dapm.c:1029: warning: 'snd_soc_dapm_connect_input' is deprecated (declared at sound/soc/soc-dapm.c:1026) sound/soc/soc-dapm.c:1029: warning: 'snd_soc_dapm_connect_input' is deprecated (declared at sound/soc/soc-dapm.c:1026) Could we have a Kconfig option of disabling deprecated APIs? If I can get rid of this I'll have a warning free build. -- Jon Smirl jonsmirl(a)gmail.com

3 4

[alsa-devel] [PATCH] i2c: Drop I2C_CLASS_CAM_ANALOG and I2C_CLASS_SOUND
by Jean Delvare 31 Oct '08

31 Oct '08

There are no users left of these two i2c probe class flags so we can drop the now. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> --- include/linux/i2c.h | 2 -- 1 file changed, 2 deletions(-) --- linux-2.6.28-rc2.orig/include/linux/i2c.h 2008-10-31 09:14:39.000000000 +0100 +++ linux-2.6.28-rc2/include/linux/i2c.h 2008-10-31 09:49:27.000000000 +0100 @@ -393,9 +393,7 @@ static inline void i2c_set_adapdata(stru #define I2C_CLASS_TV_ANALOG (1<<1) /* bttv + friends */ #define I2C_CLASS_TV_DIGITAL (1<<2) /* dvb cards */ #define I2C_CLASS_DDC (1<<3) /* DDC bus on graphics adapters */ -#define I2C_CLASS_CAM_ANALOG (1<<4) /* camera with analog CCD */ #define I2C_CLASS_CAM_DIGITAL (1<<5) /* most webcams */ -#define I2C_CLASS_SOUND (1<<6) /* sound devices */ #define I2C_CLASS_SPD (1<<7) /* SPD EEPROMs and similar */ /* i2c_client_address_data is the struct for holding default client -- Jean Delvare

1 0

[alsa-devel] How to set the original/target sample rate?
by cocala 31 Oct '08

31 Oct '08

My audio codec can perform sample rate convert. Then how to tell codec driver the original sample rate and target sample rate through alsa library interface?

1 0