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[alsa-devel] [Via8237] - VT1618 data sheet
by Goran Sterjov 14 May '08

14 May '08

Hello. i tried out the VT1618 patch from john utz but i failed to get the "Smart 5.1" surround sound working. so i figured that i'd go ahead and experiment with the driver to make it work on my via epia en15000 motherboard. i tried requesting the data sheet from VIA ten ga-gillion times but they only responded once with a "the datasheet is a high level engineering document. visit the technical support forums for help". i tried explaining that i needed the information to further develop an ALSA driver for the linux system but it seems like i wont get my hands on this god forsaken document. Is there anyone out there who has any documentation on the vt1618..? via support is shocking. Nay, beyond shocking. in John's patch he mentions various registers but unfortunately they dont help me without the docs.. Thanks Goran

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[alsa-devel] ioctl request code for specific alsa control
by Peter Wurmsdobler 14 May '08

14 May '08

Hello, The driver for our audio hardware provides 6 snd_kcontrol_new_t structures, with their proper callbacks, members and name. One is called pmic_control_pb_vol (.name = "Master Playback Volume") in the driver, but appears somehow as "Master" in amixer and can be successfully written to using the OSS interface and ioctl as: int mixerFd = open("/dev/mixer", O_RDWR); if (mixerFd >= 0) { unsigned int leftright; ioctl(mixerFd, SOUND_MIXER_WRITE_VOLUME, &leftright); close(mixerFd); } It is a big enigma for me that the ioctl with the request code used above arrives automagically in the particular audio driver and calls pmic_pb_volume_put() defined in pmic_control_pb_vol. Another control is called pmic_control_op_sw (.name = "Master Output Playback Volume") and appears in amixer as "Master Output" (in fact it is not a volume control but a switch). It is possible to set the output device using amixer. My questions are: What is the ioctl request code for a particular snd_kcontrol_new_t defined in the driver? Alternatively, how does the OSS layer in ALSA map the different ALSA controls to the predefined and fixed OSS controls? The hard way (and what I am currently trying as last resort) is to write to all of them and see what happens. Many thanks for your help, peter

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[alsa-devel] From snd_kcontrol_new_t to amixer
by Peter Wurmsdobler 14 May '08

14 May '08

Hello, I am puzzled by an observation that runs against my common sense. The pmic-alsa-mixer.c provides 6 (six) snd_kcontrol_new_t controls with their callback functions, name and other members, then registers themusing snd_ctl_add(...). However, amixer shows only 5 (five), and all with different names. What is the process of generating these names if not from the kernel structures? By looking through the mixer related code in alsa-utils and alsa-libs, I see that somehow the library loads the mixer controls and possibly retrieves all the information provided by them, including their names. Either some piece of code shortens the names defined in the driver, or it maps them to some generic controls. Any suggestions? The system I am working on uses a Freescale imx27 in conjunction with an MC13783 power management IC, also responsible for audio (it provides a voice codec and a stereo dac). The kernel used is a 2.6.19.2 as part of ltib from Freescale. amixer tells me: mx27# amixer Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined cvolume Playback channels: Mono Capture channels: Mono Limits: Playback 0 - 100 Capture 0 - 100 Mono: Playback 0 [0%] Capture 0 [0%] Simple mixer control 'Master Balance',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 100 Mono: Playback 50 [50%] Simple mixer control 'Master Input',0 Capabilities: cvolume Capture channels: Mono Limits: Capture 0 - 7 Mono: Capture 1 [14%] Simple mixer control 'Master Monoconfig',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 3 Mono: Playback 0 [0%] Simple mixer control 'Master Output',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 15 Mono: Playback 12 [80%] snd_kcontrol_new_t pmic_control_pb_vol __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .index = 0x00, .info = pmic_pb_volume_info, .get = pmic_pb_volume_get, .put = pmic_pb_volume_put, .private_value = 0xffab1, }; snd_kcontrol_new_t pmic_control_pb_bal __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Balance Playback Volume", .index = 0x00, .info = pmic_pb_balance_info, .get = pmic_pb_balance_get, .put = pmic_pb_balance_put, .private_value = 0xffab2, }; snd_kcontrol_new_t pmic_control_pb_mono __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Monoconfig Playback Volume", .index = 0x00, .info = pmic_pb_monoconfig_info, .get = pmic_pb_monoconfig_get, .put = pmic_pb_monoconfig_put, .private_value = 0xffab2, }; snd_kcontrol_new_t pmic_control_op_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Output Playback Volume", .index = 0x00, .info = pmic_mixer_output_info, .get = pmic_mixer_output_get, .put = pmic_mixer_output_put, .private_value = 0xffab4, }; snd_kcontrol_new_t pmic_control_cap_vol __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Capture Volume", .index = 0x00, .info = pmic_cap_volume_info, .get = pmic_cap_volume_get, .put = pmic_cap_volume_put, .private_value = 0xffab5, }; snd_kcontrol_new_t pmic_control_ip_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Input Capture Volume", .index = 0x00, .info = pmic_cap_input_info, .get = pmic_cap_input_get, .put = pmic_cap_input_put, .private_value = 0xffab5, }; int mxc_alsa_create_ctl(snd_card_t * card, void *p_value) { int err = 0; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_op_sw, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_pb_vol, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_pb_mono, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_pb_bal, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_cap_vol, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_ip_sw, p_value))) < 0) return err; return 0; } Cheers, peter

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Re: [alsa-devel] Problem with aloop
by Benjamin van den Hout 14 May '08

14 May '08

Gustavo da Silva Serra wrote: I had some problems until I could figure it out how to configure alsa > properly in /etc/asound.conf. Maybe this could be your problem? What is > your alsa version? I heard that aloop didn't work well until 1.0.15, or > such (alsa guys: is this true?). > My alsa is 1.0.14 but I'm using the 'HG' version of the alsa-drivers and alsa-kernel packages so that basically gets me the latest-and-greatest drivers from the source repository if I understand correctly. I saw some patches coming by in February and they are already in there so I presume I'm working with the most current source. I'm intrigued by the asound.conf, is there any specific configuration you needed to make the aloop device work properly? I'm using the distro-configured alsa (ubuntu 7.10) and there is no such file somehwere in /etc. Even on my debian server (running Etch), I can't find it. Kind regards, Benjamin

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[alsa-devel] STB TV PCI FM
by Michael Papet 14 May '08

14 May '08

6000704 2.6.24 To: alsa-devel(a)alsa-project.org MIME-Version: 1.0 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: 8bit Message-ID: <934014.92322.qm(a)web62006.mail.re1.yahoo.com> Hi, I'm trying to finish installing an old STB TV PCI FM 6000704 tv capture card working and there's no sound. I'm on Debian Lenny/Testing kernel version 2.6.24. The card itself does not detect or load snd-bt87x when booting. I force it to load with the load_all kernel option when I modprobe. It doesn't seem to use msp3400 in any way. In theory,/proc/asound/card1/pcm0c is the stereo capture, but I can't figure out how to make it capture in stereo. I can set the digital_rate as needed if that will get stereo going. I've tried 36000 and 44100 with no success, but it could be because .asoundrc isn't right. Check http://pastebin.ca/1017578 for alsa-info.sh output. Any advice, or perhaps an .asoundrc would be great. Best regards, Michael modprobe -v snd-bt87x dmesg>> "unknown card 0x878-0x10b4:0x2636 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> bt87x0: Using board 0, analog, digital (rate 32000 Hz"

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[alsa-devel] Auzen X-Fi Prelude
by Kamil Kaminski 14 May '08

14 May '08

Hi, When will be there a working driver for Auzen Prelude card?

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Re: [alsa-devel] Alsa-devel Digest, Vol 15, Issue 30
by Benjamin van den Hout 13 May '08

13 May '08

Gustavo da Silva Serra wrote: I tryied to use the HG versions, but they didn't work well for me. Maybe > you should give the 1.0.16 a try, just to make sure. > I need an asound.conf since my network application uses a very low > latency, that is not default. I am attaching my asound.conf. > Another possible source of problems are the artsd and esd daemons, are > they running? > > Hi Gustavo, I just downloaded the 10.0.16 version and the code for aloop-kernel.c looked quite different from the one in the HG version. I was under the understanding that the HG version is a current development snapshot that eventually gets tagged into an official 1.0.1xx version. That's not the case then? Anyway, I will try with the 10.0.16 version, I see some stuff about a #define called SND_CARD_LOOPBACK_START_SYNC, that sounds promising perhaps. I've also looked at the config file you attached, the stuff about period_time and buffer_size also looks quite interesting and is perhaps something I should play with then :) I'll let you know how things work out! Thanks for your continued help, it is greatly appreciated... Kind regards, Benjamin

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Re: [alsa-devel] [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver
by Mark Brown 13 May '08

13 May '08

On Mon, May 12, 2008 at 06:45:12PM +0800, Bryan Wu wrote: > From: Cliff Cai <cliff.cai(a)analog.com> > > Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> > Signed-off-by: Bryan Wu <cooloney(a)kernel.org> Thanks, I've applied this to the ASoC git tree. CCing in alsa-devel(a)alsa-project.org - ALSA patches should go via there. > +static int ad1980_soc_suspend(struct platform_device *pdev, > + pm_message_t state) > +{ > + return 0; > +} > + > +static int ad1980_soc_resume(struct platform_device *pdev) > +{ > + return 0; > +} Are you sure about these? I would expect the suspend and resume functions to either do some register writes or be omitted if they don't do anything. Standard AC97 codecs would have some power management via register 0x26 if they were doing anything.

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[alsa-devel] [PATCH 01/10] ASoC: build fix for snd_soc_info_bool_ext
by Mark Brown 13 May '08

13 May '08

From: Philipp Zabel <philipp.zabel(a)gmail.com> I suspect that snd_ctl_boolean_mono should have been snd_ctl_boolean_mono_info instead. This fixes the build for magician. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e6ea6f7..d3c8c03 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -238,7 +238,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -#define snd_soc_info_bool_ext snd_ctl_boolean_mono +#define snd_soc_info_bool_ext snd_ctl_boolean_mono_info int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, -- 1.5.5.1

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[alsa-devel] Acer Aspire 8920g / snd-hda-intel / ALC889
by Andy Green 13 May '08

13 May '08

Hi folks - I have a new 8920g on Fedora development running 2.6.25 which has $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.16rc2 (Thu Jan 31 16:40:16 2008 UTC). I get a decent looking set of alsa things in /proc/asound $ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc300000 irq 22 and dmesg says that model=acer was selected reasonably enough. I can play audio down the logical sound devices, including cat /dev/urandom > /dev/dsp and the interrupt count goes up in /proc/interrupts. Alsamixer shows a bunch of things and I twiddled them all to no avail. There has never been even a click from the speakers or headphone jack on this machine. At least three funnies on this machine: 1) It is confused if it has ALC889 or ALC883 in there $ grep ALC /proc/asound/* -R /proc/asound/card0/codec#0:Codec: Realtek ALC889 <==== 889 /proc/asound/card0/pcm2c/sub0/info:id: ALC883 Analog <=== 883... /proc/asound/card0/pcm2c/sub0/info:name: ALC883 Analog /proc/asound/card0/pcm2c/info:id: ALC883 Analog /proc/asound/card0/pcm2c/info:name: ALC883 Analog /proc/asound/card0/pcm0c/sub0/info:id: ALC883 Analog /proc/asound/card0/pcm0c/sub0/info:name: ALC883 Analog /proc/asound/card0/pcm0c/info:id: ALC883 Analog /proc/asound/card0/pcm0c/info:name: ALC883 Analog /proc/asound/card0/pcm0p/sub0/info:id: ALC883 Analog /proc/asound/card0/pcm0p/sub0/info:name: ALC883 Analog /proc/asound/card0/pcm0p/info:id: ALC883 Analog /proc/asound/card0/pcm0p/info:name: ALC883 Analog /proc/asound/Intel/codec#0:Codec: Realtek ALC889 <=== 889 /proc/asound/Intel/pcm2c/sub0/info:id: ALC883 Analog /proc/asound/Intel/pcm2c/sub0/info:name: ALC883 Analog /proc/asound/Intel/pcm2c/info:id: ALC883 Analog /proc/asound/Intel/pcm2c/info:name: ALC883 Analog /proc/asound/Intel/pcm0c/sub0/info:id: ALC883 Analog /proc/asound/Intel/pcm0c/sub0/info:name: ALC883 Analog /proc/asound/Intel/pcm0c/info:id: ALC883 Analog /proc/asound/Intel/pcm0c/info:name: ALC883 Analog /proc/asound/Intel/pcm0p/sub0/info:id: ALC883 Analog /proc/asound/Intel/pcm0p/sub0/info:name: ALC883 Analog /proc/asound/Intel/pcm0p/info:id: ALC883 Analog /proc/asound/Intel/pcm0p/info:name: ALC883 Analog /proc/asound/oss/sndstat:0: ALC883 Analog (DUPLEX) /proc/asound/oss/sndstat:0: Realtek ALC889 /proc/asound/pcm:00-02: ALC883 Analog : ALC883 Analog : capture 1 /proc/asound/pcm:00-00: ALC883 Analog : ALC883 Analog : playback 1 : capture 1 2) from dmesg ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x3 ALSA sound/pci/hda/hda_codec.c:2211: hda_codec: model 'acer' is selected ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Headphone Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Line-Out Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave IEC958 Playback Switch, skipped 3) there is a funny embedded capacitive volume control thing on this device, whether it gates output by itself or is just a bunch of special keyboard keys, I dunno. Any ideas, more information needed? -Andy

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