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[alsa-devel] [PATCH 2/2] ac97: YMF743 missing controls support
by Keita Maehara 19 Sep '07

19 Sep '07

From: Keita Maehara <maehara(a)debian.org> These patches enable a few YMF743 controls (Tone/3D/IEC958) that won't be detected with the current version of ALSA. Signed-off-by: Keita Maehara <maehara(a)debian.org> --- The second one contains following changes: - A chip-specific SPDIF support for YMF743 (It doesn't have AC97 standard SPDIF registers seen on YMF753). - The implementation for "IEC958 Playback Source" and "IEC958 Mute" are identical to the ones for YMF753. But there is no "IEC958 Output Pin" for YMF743. diff -r 92978f62cc03 pci/ac97/ac97_codec.c --- a/pci/ac97/ac97_codec.c Tue Sep 18 00:28:51 2007 +0900 +++ b/pci/ac97/ac97_codec.c Tue Sep 18 00:29:26 2007 +0900 @@ -176,7 +176,7 @@ static const struct ac97_codec_id snd_ac { 0x574d4C09, 0xffffffff, "WM9709", NULL, NULL}, { 0x574d4C12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL}, { 0x574d4c13, 0xffffffff, "WM9713,WM9714", patch_wolfson13, NULL, AC97_DEFAULT_POWER_OFF}, -{ 0x594d4800, 0xffffffff, "YMF743", NULL, NULL }, +{ 0x594d4800, 0xffffffff, "YMF743", patch_yamaha_ymf743, NULL }, { 0x594d4802, 0xffffffff, "YMF752", NULL, NULL }, { 0x594d4803, 0xffffffff, "YMF753", patch_yamaha_ymf753, NULL }, { 0x83847600, 0xffffffff, "STAC9700,83,84", patch_sigmatel_stac9700, NULL }, @@ -779,6 +779,12 @@ static int snd_ac97_spdif_default_put(st change |= snd_ac97_update_bits_nolock(ac97, AC97_CXR_AUDIO_MISC, AC97_CXR_SPDIF_MASK | AC97_CXR_COPYRGT, v); + } else if (ac97->id == AC97_ID_YMF743) { + change |= snd_ac97_update_bits_nolock(ac97, + AC97_YMF7X3_DIT_CTRL, + 0xff38, + ((val << 4) & 0xff00) | + ((val << 2) & 0x0038)); } else { unsigned short extst = snd_ac97_read_cache(ac97, AC97_EXTENDED_STATUS); snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, 0); /* turn off */ @@ -1375,7 +1381,8 @@ static int snd_ac97_mixer_build(struct s for (idx = 0; idx < 2; idx++) { if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_tone[idx], ac97))) < 0) return err; - if (ac97->id == AC97_ID_YMF753) { + if (ac97->id == AC97_ID_YMF743 || + ac97->id == AC97_ID_YMF753) { kctl->private_value &= ~(0xff << 16); kctl->private_value |= 7 << 16; } diff -r 92978f62cc03 pci/ac97/ac97_id.h --- a/pci/ac97/ac97_id.h Tue Sep 18 00:28:51 2007 +0900 +++ b/pci/ac97/ac97_id.h Tue Sep 18 00:29:26 2007 +0900 @@ -54,6 +54,7 @@ #define AC97_ID_ALC658 0x414c4780 #define AC97_ID_ALC658D 0x414c4781 #define AC97_ID_ALC850 0x414c4790 +#define AC97_ID_YMF743 0x594d4800 #define AC97_ID_YMF753 0x594d4803 #define AC97_ID_VT1616 0x49434551 #define AC97_ID_CM9738 0x434d4941 diff -r 92978f62cc03 pci/ac97/ac97_patch.c --- a/pci/ac97/ac97_patch.c Tue Sep 18 00:28:51 2007 +0900 +++ b/pci/ac97/ac97_patch.c Tue Sep 18 00:29:26 2007 +0900 @@ -204,6 +204,7 @@ static inline int is_shared_micin(struct /* The following snd_ac97_ymf753_... items added by David Shust (dshust(a)shustring.com) */ +/* Modified for YMF743 by Keita Maehara <maehara(a)debian.org> */ /* It is possible to indicate to the Yamaha YMF7x3 the type of speakers being used. */ static int snd_ac97_ymf7x3_info_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -287,6 +288,63 @@ static int snd_ac97_ymf7x3_spdif_source_ return -EINVAL; val = ucontrol->value.enumerated.item[0] << 1; return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0002, val); +} + +static int patch_yamaha_ymf7x3_3d(struct snd_ac97 * ac97) +{ + struct snd_kcontrol *kctl; + int err; + + if ((err = snd_ctl_add(ac97->bus->card, kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97))) < 0) + return err; + strcpy(kctl->id.name, "3D Control - Wide"); + kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0); + snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000); + if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_ymf7x3_controls_speaker, ac97))) < 0) + return err; + snd_ac97_write_cache(ac97, AC97_YMF7X3_3D_MODE_SEL, 0x0c00); + return 0; +} + +static const struct snd_kcontrol_new snd_ac97_yamaha_ymf743_controls_spdif[3] = { + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), AC97_YMF7X3_DIT_CTRL, 0, 1, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + .info = snd_ac97_ymf7x3_spdif_source_info, + .get = snd_ac97_ymf7x3_spdif_source_get, + .put = snd_ac97_ymf7x3_spdif_source_put, + }, + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",NONE,NONE) "Mute", AC97_YMF7X3_DIT_CTRL, 2, 1, 1) +}; + +static int patch_yamaha_ymf743_build_spdif(struct snd_ac97 * ac97) +{ + int err; + + if ((err = patch_build_controls(ac97, &snd_ac97_controls_spdif[0], 3)) < 0) + return err; + if ((err = patch_build_controls(ac97, snd_ac97_yamaha_ymf743_controls_spdif, ARRAY_SIZE(snd_ac97_yamaha_ymf743_controls_spdif))) < 0) + return err; + /* set default PCM S/PDIF params */ + /* PCM audio,no copyright,no preemphasis,PCM coder,original */ + snd_ac97_write_cache(ac97, AC97_YMF7X3_DIT_CTRL, 0xa201); + return 0; +} + +static struct snd_ac97_build_ops patch_yamaha_ymf743_ops = { + .build_spdif = patch_yamaha_ymf743_build_spdif, + .build_3d = patch_yamaha_ymf7x3_3d, +}; + +static int patch_yamaha_ymf743(struct snd_ac97 * ac97) +{ + ac97->build_ops = &patch_yamaha_ymf743_ops; + ac97->caps |= AC97_BC_BASS_TREBLE; + ac97->caps |= 0x04 << 10; /* Yamaha 3D enhancement */ + ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_48000; /* 48k only */ + ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */ + return 0; } /* The AC'97 spec states that the S/PDIF signal is to be output at pin 48. @@ -347,22 +405,6 @@ static const struct snd_kcontrol_new snd }, AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",NONE,NONE) "Mute", AC97_YMF7X3_DIT_CTRL, 2, 1, 1) }; - -static int patch_yamaha_ymf7x3_3d(struct snd_ac97 * ac97) -{ - struct snd_kcontrol *kctl; - int err; - - if ((err = snd_ctl_add(ac97->bus->card, kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97))) < 0) - return err; - strcpy(kctl->id.name, "3D Control - Wide"); - kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0); - snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000); - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_ymf7x3_controls_speaker, ac97))) < 0) - return err; - snd_ac97_write_cache(ac97, AC97_YMF7X3_3D_MODE_SEL, 0x0c00); - return 0; -} static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 * ac97) { diff -r 92978f62cc03 pci/ac97/ac97_proc.c --- a/pci/ac97/ac97_proc.c Tue Sep 18 00:28:51 2007 +0900 +++ b/pci/ac97/ac97_proc.c Tue Sep 18 00:29:26 2007 +0900 @@ -236,10 +236,14 @@ static void snd_ac97_proc_read_main(stru val = snd_ac97_read(ac97, AC97_PCM_MIC_ADC_RATE); snd_iprintf(buffer, "PCM MIC ADC : %iHz\n", val); } - if ((ext & AC97_EI_SPDIF) || (ac97->flags & AC97_CS_SPDIF)) { + if ((ext & AC97_EI_SPDIF) || (ac97->flags & AC97_CS_SPDIF) || + (ac97->id == AC97_ID_YMF743)) { if (ac97->flags & AC97_CS_SPDIF) val = snd_ac97_read(ac97, AC97_CSR_SPDIF); - else + else if (ac97->id == AC97_ID_YMF743) { + val = snd_ac97_read(ac97, AC97_YMF7X3_DIT_CTRL); + val = 0x2000 | (val & 0xff00) >> 4 | (val & 0x38) >> 2; + } else val = snd_ac97_read(ac97, AC97_SPDIF); snd_iprintf(buffer, "SPDIF Control :%s%s%s%s Category=0x%x Generation=%i%s%s%s\n",

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[alsa-devel] Routing multiple signals to one output with ice1712 (Delta 66)
by Mathis Hofer 19 Sep '07

19 Sep '07

Hi folks I want to setup the following audio routing: pcm + input1 (mic) + input2 (mic) -> output1 pcm + input1 (mic) + input2 (mic) -> output2 ...with pcm being the audio playback and input1&2 being two microphones. With envy24control it is possible to select one signal for each output, but not multiple signals. Is this a restriction of envy24control, the alsa driver or is it simply not supported by the hardware? Regards, Mathis -- DreamLab Technologies AG Monbijoustrasse 36 3011 Bern Switzerland Tel: +41 31 398 66 66 Fax: +41 31 398 66 69 PGP Key ID: 2462240B

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[alsa-devel] Separate default devices for playback and capture
by Lasse Kärkkäinen 19 Sep '07

19 Sep '07

I've been reading the library source code and from what I gather, it seems that it just fundamentally cannot support separate config sections for playback and capture. What should the fix be? I can see three options: 1. allow specifying in asound.conf whether a section is for recording or playback (needs major redesign of conf.c and some changes in pcm.c). 2. if the name is "default", use "capture" (for example) instead, in snd_pcm_open_noupdate, maybe reverting to "default" if the other device cannot be found (dirty, but very easy to implement) 3. tell application developers to use some other device name for capture: "capture", instead of "default", for example (needs a little change in every application program that uses ALSA capture).

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Re: [alsa-devel] Audiophile Delta 24/96 Midi Out continuously sends zeroes (alsa 1.0.14)
by Clemens Ladisch 19 Sep '07

19 Sep '07

Romain Bossart wrote: > Selon Clemens Ladisch <cladisch(a)fastmail.net>: > > That is, the zeroes are _not_ sent when you're not using jack? When > > nothing is connected, or when you use aplaymidi or amidi to send > > something, everything is OK? > > Here is a summary of the various tests I did. I have just one card: Audiophile > Delta 24/96. My kernel is now 2.6.22.1 RT patched & alsa 1.0.14, > Audio works perfectly. I don't think jack is the problem. > > Config #2 = Delta Midi Out -> (Midi In) Instrument (Midi Out/Thru) -> Delta Midi > In and 'amidi -p hw:0,0 -d'. Jack being started or not. > > Symptoms: tons of zeroes are shown by amidi as soon as I launch it. The stream > is so intense I can actually hear it in my speakers/monitors (need to switch to > better MIDI cables, I know!). amidi and /proc/asound/card0/midi0 give the same number > of transmitted bytes. With "transmitted", you meant "received"? When Jack is not running, there is no software to send anything. If the number of _transmitted_ bytes changes anyway, this would be a bug in ALSA. > Config #2b = Delta Midi Out -> Delta Midi In > and 'amidi -p hw:0,0 -d', Jack being started or not. > > Symptoms: identical to config#2. Moreover, if meanwhile I send some Midi commands > with amidi ('amidi -p hw:0,0 -S C01C'), I don't find this data in my dump. Apparently, the output port is broken. While it is theoretically possible that the driver doesn't initialize the card correctly, I cannot imagine a bug that would cause the port to presume that there is data to send. Could you test if the port sends data even before the driver (or Linux) is loaded, or in Windows? Regards, Clemens

2 2

[alsa-devel] [PATCH] ad1838/cs4231 -- fix MCE timeout upon initial load
by Rene Herman 19 Sep '07

19 Sep '07

Hi Takashi. When the ad1848/cs2431 is first being initialized, auto-calibration may not be set causing a timeout waiting for it in snd_ad1848/cs4231_mce_down(). This has no dire consequences other than an alarming printk, but since what we need to wait for is for the calibration to _finish_, let's just check for that instead. The early chips need a slight delay (as commented -- 5 sample periods) to be sure that _if_ calibration is going to happen, it has started when we check While the CS4231A datasheet implies it'll happen immediately on downing MCE, some testing is showing that there's a window there as well, so just do the delay everywhere. Thanks to Krysztof Helt for pinpointing this problem. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> Rene. diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index 8094282..dcd4435 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -227,16 +227,15 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) spin_unlock_irqrestore(&chip->reg_lock, flags); return; } - /* calibration process */ - for (timeout = 500; timeout > 0 && (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) == 0; timeout--); - if ((snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) == 0) { - snd_printd("mce_down - auto calibration time out (1)\n"); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return; - } + /* + * Wait for (possible -- during init auto-calibration may not be set) + * calibration process to start. Needs upto 5 sample periods on AD1848 + * which at the slowest possible rate of 5.5125 kHz means 907 us. + */ + msleep(1); #if 0 - printk("(2) timeout = %i, jiffies = %li\n", timeout, jiffies); + printk("(2) jiffies = %li\n", jiffies); #endif time = HZ / 4; while (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) { diff --git a/sound/isa/cs423x/cs4231_lib.c b/sound/isa/cs423x/cs4231_lib.c index 914d77b..9e8e0f1 100644 --- a/sound/isa/cs423x/cs4231_lib.c +++ b/sound/isa/cs423x/cs4231_lib.c @@ -346,16 +346,14 @@ void snd_cs4231_mce_down(struct snd_cs4231 *chip) } snd_cs4231_busy_wait(chip); - /* calibration process */ - - for (timeout = 500; timeout > 0 && (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0; timeout--) - udelay(10); - if ((snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0) { - snd_printd("cs4231_mce_down - auto calibration time out (1)\n"); - return; - } + /* + * Wait for (possible -- during init auto-calibration may not be set) + * calibration process to start. Needs upto 5 sample periods on AD1848 + * which at the slowest possible rate of 5.5125 kHz means 907 us. + */ + msleep(1); #if 0 - printk("(2) timeout = %i, jiffies = %li\n", timeout, jiffies); + printk("(2) jiffies = %li\n", jiffies); #endif /* in 10 ms increments, check condition, up to 250 ms */ timeout = 25;

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[alsa-devel] Enabling in-kernel synch for M-Audio boards
by Ludovico Verducci 19 Sep '07

19 Sep '07

Hello all! I'm developing a complex multichannel audio distribution system where multiple linux boxes will stream audio data over ethernet and then should play audio at sample level resolution synchronization. The boxes clocks are synchronized over ethernet using PTP. I need to keep in synch the audio board's clocks and I can't use an external wordclock nor s/pdif. At the moment I'm experimenting with m-audio delta 1010, based on envy24 (VT1712). I know these boards can synch their clocks over PCI signalling (in windows drivers this feature support synching up to 4 boards), but ALSA drivers don't implement that feature. I'm trying to implement a similar feature (controlling board clock synch through kernel module), but it's very difficult without a good starting point! Is there anyone that could give me some help, support or infos? I would eventually appreciate also hardware reference designs or application notes about the VIA envy24 VT1712 chip. Obviously, should I ever be successful in the implementation, those features could be eventually added to ALSA drivers. Thank you all in advance. Regards, Ludovico

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Re: [alsa-devel] Intel HDA / AD1986A problems
by Takashi Iwai 18 Sep '07

18 Sep '07

At Tue, 21 Aug 2007 21:45:52 -0400, Peter Skensved wrote: > > On Tue, Aug 21, 2007 at 02:34:22PM +0200, Takashi Iwai wrote: > > At Mon, 20 Aug 2007 22:27:40 -0400, > > Peter Skensved wrote: > > > > > > > > > > > > Configuration : Lenovo 3000 N100 laptop with Intel HDA / Anlog Devices AD1986A > > > soundcard running CentOS5.0 with 2.6.18-8.1.8.el5 kernel ( uses 1.0.12rc1 based > > > alsa driver ). > > > > > > The driver works with the model=laptop-eapd option except for a few things : > > > > > > 1) Plugging in an external headset will not mute the internal speakers ( has to be done > > > manually in alsa-mixer ) > > > > That's an unimplemented feature. It shouldn't be too hard to add it > > if you help debugging well (i.e. you can build and test drivers from > > HG tree without hitch). > > > > Ok - I'm willing to give it a try. Do I need development version of theh library > and the utilities as well ? Recommended, but not mandatory. > > > 2) The PC speaker ( bell ) appears to be permanently muted. > > > > Partly intended (as I dislike it ;) > > > > It is nice to have the option ... Another opinion :) > > > 3) The external microphone input does not appear to be working no matter what I do. > > > > Do you mean for recording or analog loopback? > > > > Yes - for recording. I was hoping to get it to work well enough to use skype - Note that recording and analog-loopback are different things. > > > 4) When recording from the internal microphone the level is very, very low and almost > > > drowns in static noise. > > > > There are "Mic Boost" and "Front Mic Boost" mixer volumes. > > > > Not with model=laptop-eapd - my alas mixer only shows Master, PCM, Mic, Capture, Mix, > External Amp and Internal Amp The Mic Boost control was added _after_ 1.0.14 release. (Correction in the above text: Front Mic Boost doesn't exist for AD1986A but for other codec.) So, please try HG version first. Then let's debug further. Takashi

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[alsa-devel] AD1986A stop working if adjust the volume
by zhejiang 18 Sep '07

18 Sep '07

Hi, I have a ICH8M + AD1986A machine and run alsa-driver-1.0.15rc. I found that : In 6 channel mode,if adjust the front or pcm volume when playing music, all channel will be muted,but if adjust the volume alone(don't play music at the same time),every thing is ok. I use printk to display the verb in the azx_send_cmd(), when adjusting the volume in either case, there are just two verb (0x300) sent to the nid(0x1b) [correct action]. But in the first case, the dac widget amp-out vals are all affected. Some other interesting thing, If i adjust the Sur or CLFE volume when playing music, the printk show that correct verb is sent, but the codec status don't change. If I adjust them alone,every thing is ok. The card0/codec#0 before adjust volume : Codec: Analog Devices AD1986A Address: 0 Vendor Id: 0x11d41986 Subsystem Id: 0xbfd60000 Revision Id: 0x100500 No Modem Function Group found Default PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital PCM: rates [0x60]: 44100 48000 bits [0x2]: 16 formats [0x5]: PCM AC3 Connection: 2 0x01* 0x06 Node 0x03 [Audio Output] wcaps 0x44d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Power: 0x0 Node 0x04 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Power: 0x0 Node 0x05 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Power: 0x0 Node 0x06 [Audio Input] wcaps 0x100511: Stereo PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x6]: 16 20 formats [0x1]: PCM Power: 0x0 Connection: 1 0x12 Node 0x07 [Audio Mixer] wcaps 0x200101: Stereo Connection: 8 0x03 0x09 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x08 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80] [0x80] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Connection: 2 0x04 0x05 Node 0x0a [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x07* 0x04 0x05 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x07* 0x04 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x04* 0x07 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x05* 0x08 Node 0x0e [Audio Selector] wcaps 0x300100: Mono Connection: 2 0x08* 0x11 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 8 0x1f* 0x20 0x1d 0x1d 0x27 0x28 0x29 0x2a Node 0x10 [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x20* 0x1c 0x1f Node 0x11 [Audio Selector] wcaps 0x300941: Stereo Connection: 2 0x0f* 0x2b Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 8 0x11* 0x22 0x00 0x21 0x10 0x07 0x08 0x23 Node 0x13 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x11 Node 0x14 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80] Connection: 1 0x23 Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x22 Node 0x16 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x21 Node 0x17 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x10 Node 0x18 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x8f] Connection: 2 0x19* 0x24 Node 0x19 [Beep Generator Widget] wcaps 0x700000: Mono Node 0x1a [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x081f: OUT HP Detect Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 1 0x0a Node 0x1b [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x10 0x10] Pincap 0x081001f: OUT HP EAPD Detect Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 1 0x0b Node 0x1c [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x10 0x10] Pincap 0x0837: IN OUT Detect Pin Default 0x01013012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x40: OUT Connection: 1 0x0c Node 0x1d [Pin Complex] wcaps 0x400985: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x0e 0x0f] Pincap 0x081737: IN OUT Detect Pin Default 0x01019011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x40: OUT Connection: 1 0x0d Node 0x1e [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x901301f0: [Fixed] Speaker at Int N/A Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x1f [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Node 0x20 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x01813021: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Node 0x21 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0827: IN Detect Pin Default 0x99930123: [Fixed] Aux at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x22 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x99330122: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x23 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90b70130: [Fixed] Telephony at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x24 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90f701f0: [Fixed] Other at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x25 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x014511f0: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x26 [Power Widget] wcaps 0x500500: Mono Power: 0x0 Connection: 8 0x07* 0x08 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x27 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x1d Node 0x28 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x20 Node 0x29 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1d 0x20 Node 0x2a [Audio Mixer] wcaps 0x200101: Stereo Connection: 3 0x1f 0x1d 0x20 Node 0x2b [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x0f The card0/codec#0 after adjust "Front " volume: Codec: Analog Devices AD1986A Address: 0 Vendor Id: 0x11d41986 Subsystem Id: 0xbfd60000 Revision Id: 0x100500 No Modem Function Group found Default PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital PCM: rates [0x60]: 44100 48000 bits [0x2]: 16 formats [0x5]: PCM AC3 Connection: 2 0x01* 0x06 Node 0x03 [Audio Output] wcaps 0x44d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97 0x97] Power: 0x0 Node 0x04 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Power: 0x0 Node 0x05 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Power: 0x0 Node 0x06 [Audio Input] wcaps 0x100511: Stereo PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x6]: 16 20 formats [0x1]: PCM Power: 0x0 Connection: 1 0x12 Node 0x07 [Audio Mixer] wcaps 0x200101: Stereo Connection: 8 0x03 0x09 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x08 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80] [0x80] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Connection: 2 0x04 0x05 Node 0x0a [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x07* 0x04 0x05 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x07* 0x04 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x04* 0x07 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x05* 0x08 Node 0x0e [Audio Selector] wcaps 0x300100: Mono Connection: 2 0x08* 0x11 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 8 0x1f* 0x20 0x1d 0x1d 0x27 0x28 0x29 0x2a Node 0x10 [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x20* 0x1c 0x1f Node 0x11 [Audio Selector] wcaps 0x300941: Stereo Connection: 2 0x0f* 0x2b Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 8 0x11* 0x22 0x00 0x21 0x10 0x07 0x08 0x23 Node 0x13 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97 0x97] Connection: 1 0x11 Node 0x14 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97] Connection: 1 0x23 Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97 0x97] Connection: 1 0x22 Node 0x16 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97 0x97] Connection: 1 0x21 Node 0x17 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97 0x97] Connection: 1 0x10 Node 0x18 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x8f] Connection: 2 0x19* 0x24 Node 0x19 [Beep Generator Widget] wcaps 0x700000: Mono Node 0x1a [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Pincap 0x081f: OUT HP Detect Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 1 0x0a Node 0x1b [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Pincap 0x081001f: OUT HP EAPD Detect Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 1 0x0b Node 0x1c [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Pincap 0x0837: IN OUT Detect Pin Default 0x01013012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x40: OUT Connection: 1 0x0c Node 0x1d [Pin Complex] wcaps 0x400985: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Pincap 0x081737: IN OUT Detect Pin Default 0x01019011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x40: OUT Connection: 1 0x0d Node 0x1e [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f] Pincap 0x0810: OUT Pin Default 0x901301f0: [Fixed] Speaker at Int N/A Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x1f [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Node 0x20 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x01813021: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Node 0x21 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0827: IN Detect Pin Default 0x99930123: [Fixed] Aux at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x22 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x99330122: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x23 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90b70130: [Fixed] Telephony at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x24 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90f701f0: [Fixed] Other at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x25 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x014511f0: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x26 [Power Widget] wcaps 0x500500: Mono Power: 0x0 Connection: 8 0x07* 0x08 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x27 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x1d Node 0x28 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x20 Node 0x29 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1d 0x20 Node 0x2a [Audio Mixer] wcaps 0x200101: Stereo Connection: 3 0x1f 0x1d 0x20 Node 0x2b [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x0f

2 1

[alsa-devel] [PATCH 1/2] ac97: YMF743 missing controls support
by Keita Maehara 18 Sep '07

18 Sep '07

From: Keita Maehara <maehara(a)debian.org> These patches enable some YMF743 controls (Tone/3D/IEC958) that won't be detected with the current version of ALSA. Signed-off-by: Keita Maehara <maehara(a)debian.org> --- The first one contains only cosmetic changes to share a few YMF753-specific symbols with YMF743. diff -r 4a3f9f77e676 include/ac97_codec.h --- a/include/ac97_codec.h Mon Sep 17 16:26:16 2007 +0200 +++ b/include/ac97_codec.h Tue Sep 18 00:27:24 2007 +0900 @@ -345,9 +345,9 @@ #define AC97_ALC650_GPIO_STATUS 0x78 #define AC97_ALC650_CLOCK 0x7a -/* specific - Yamaha YMF753 */ -#define AC97_YMF753_DIT_CTRL2 0x66 /* DIT Control 2 */ -#define AC97_YMF753_3D_MODE_SEL 0x68 /* 3D Mode Select */ +/* specific - Yamaha YMF7x3 */ +#define AC97_YMF7X3_DIT_CTRL 0x66 /* DIT Control (YMF743) / 2 (YMF753) */ +#define AC97_YMF7X3_3D_MODE_SEL 0x68 /* 3D Mode Select */ /* specific - C-Media */ #define AC97_CM9738_VENDOR_CTRL 0x5a diff -r 4a3f9f77e676 pci/ac97/ac97_patch.c --- a/pci/ac97/ac97_patch.c Mon Sep 17 16:26:16 2007 +0200 +++ b/pci/ac97/ac97_patch.c Tue Sep 18 00:27:24 2007 +0900 @@ -205,8 +205,8 @@ static inline int is_shared_micin(struct /* The following snd_ac97_ymf753_... items added by David Shust (dshust(a)shustring.com) */ -/* It is possible to indicate to the Yamaha YMF753 the type of speakers being used. */ -static int snd_ac97_ymf753_info_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +/* It is possible to indicate to the Yamaha YMF7x3 the type of speakers being used. */ +static int snd_ac97_ymf7x3_info_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[3] = { "Standard", "Small", "Smaller" @@ -221,12 +221,12 @@ static int snd_ac97_ymf753_info_speaker( return 0; } -static int snd_ac97_ymf753_get_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_get_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = ac97->regs[AC97_YMF753_3D_MODE_SEL]; + val = ac97->regs[AC97_YMF7X3_3D_MODE_SEL]; val = (val >> 10) & 3; if (val > 0) /* 0 = invalid */ val--; @@ -234,7 +234,7 @@ static int snd_ac97_ymf753_get_speaker(s return 0; } -static int snd_ac97_ymf753_put_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_put_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; @@ -242,20 +242,20 @@ static int snd_ac97_ymf753_put_speaker(s if (ucontrol->value.enumerated.item[0] > 2) return -EINVAL; val = (ucontrol->value.enumerated.item[0] + 1) << 10; - return snd_ac97_update(ac97, AC97_YMF753_3D_MODE_SEL, val); -} - -static const struct snd_kcontrol_new snd_ac97_ymf753_controls_speaker = + return snd_ac97_update(ac97, AC97_YMF7X3_3D_MODE_SEL, val); +} + +static const struct snd_kcontrol_new snd_ac97_ymf7x3_controls_speaker = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "3D Control - Speaker", - .info = snd_ac97_ymf753_info_speaker, - .get = snd_ac97_ymf753_get_speaker, - .put = snd_ac97_ymf753_put_speaker, -}; - -/* It is possible to indicate to the Yamaha YMF753 the source to direct to the S/PDIF output. */ -static int snd_ac97_ymf753_spdif_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) + .info = snd_ac97_ymf7x3_info_speaker, + .get = snd_ac97_ymf7x3_get_speaker, + .put = snd_ac97_ymf7x3_put_speaker, +}; + +/* It is possible to indicate to the Yamaha YMF7x3 the source to direct to the S/PDIF output. */ +static int snd_ac97_ymf7x3_spdif_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[2] = { "AC-Link", "A/D Converter" }; @@ -268,17 +268,17 @@ static int snd_ac97_ymf753_spdif_source_ return 0; } -static int snd_ac97_ymf753_spdif_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_spdif_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = ac97->regs[AC97_YMF753_DIT_CTRL2]; + val = ac97->regs[AC97_YMF7X3_DIT_CTRL]; ucontrol->value.enumerated.item[0] = (val >> 1) & 1; return 0; } -static int snd_ac97_ymf753_spdif_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_spdif_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; @@ -286,7 +286,7 @@ static int snd_ac97_ymf753_spdif_source_ if (ucontrol->value.enumerated.item[0] > 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << 1; - return snd_ac97_update_bits(ac97, AC97_YMF753_DIT_CTRL2, 0x0002, val); + return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0002, val); } /* The AC'97 spec states that the S/PDIF signal is to be output at pin 48. @@ -311,7 +311,7 @@ static int snd_ac97_ymf753_spdif_output_ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = ac97->regs[AC97_YMF753_DIT_CTRL2]; + val = ac97->regs[AC97_YMF7X3_DIT_CTRL]; ucontrol->value.enumerated.item[0] = (val & 0x0008) ? 2 : (val & 0x0020) ? 1 : 0; return 0; } @@ -325,7 +325,7 @@ static int snd_ac97_ymf753_spdif_output_ return -EINVAL; val = (ucontrol->value.enumerated.item[0] == 2) ? 0x0008 : (ucontrol->value.enumerated.item[0] == 1) ? 0x0020 : 0; - return snd_ac97_update_bits(ac97, AC97_YMF753_DIT_CTRL2, 0x0028, val); + return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0028, val); /* The following can be used to direct S/PDIF output to pin 47 (EAPD). snd_ac97_write_cache(ac97, 0x62, snd_ac97_read(ac97, 0x62) | 0x0008); */ } @@ -334,9 +334,9 @@ static const struct snd_kcontrol_new snd { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - .info = snd_ac97_ymf753_spdif_source_info, - .get = snd_ac97_ymf753_spdif_source_get, - .put = snd_ac97_ymf753_spdif_source_put, + .info = snd_ac97_ymf7x3_spdif_source_info, + .get = snd_ac97_ymf7x3_spdif_source_get, + .put = snd_ac97_ymf7x3_spdif_source_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -345,10 +345,10 @@ static const struct snd_kcontrol_new snd .get = snd_ac97_ymf753_spdif_output_pin_get, .put = snd_ac97_ymf753_spdif_output_pin_put, }, - AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",NONE,NONE) "Mute", AC97_YMF753_DIT_CTRL2, 2, 1, 1) -}; - -static int patch_yamaha_ymf753_3d(struct snd_ac97 * ac97) + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",NONE,NONE) "Mute", AC97_YMF7X3_DIT_CTRL, 2, 1, 1) +}; + +static int patch_yamaha_ymf7x3_3d(struct snd_ac97 * ac97) { struct snd_kcontrol *kctl; int err; @@ -358,9 +358,9 @@ static int patch_yamaha_ymf753_3d(struct strcpy(kctl->id.name, "3D Control - Wide"); kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0); snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000); - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_ymf753_controls_speaker, ac97))) < 0) + if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_ymf7x3_controls_speaker, ac97))) < 0) return err; - snd_ac97_write_cache(ac97, AC97_YMF753_3D_MODE_SEL, 0x0c00); + snd_ac97_write_cache(ac97, AC97_YMF7X3_3D_MODE_SEL, 0x0c00); return 0; } @@ -374,7 +374,7 @@ static int patch_yamaha_ymf753_post_spdi } static struct snd_ac97_build_ops patch_yamaha_ymf753_ops = { - .build_3d = patch_yamaha_ymf753_3d, + .build_3d = patch_yamaha_ymf7x3_3d, .build_post_spdif = patch_yamaha_ymf753_post_spdif };

1 0

[alsa-devel] alsa-tools -> usx2yloader
by Elimar Riesebieter 18 Sep '07

18 Sep '07

Hi all, according to Debian #436502 I am asking herefore if there are plans to port asx2yloader from the old hotplug to udev? Debian lenny/sid relies on udev only, though. Thanks in advance Elimar -- It's a good thing we don't get all the government we pay for.

4 9