July 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Enabling HDMI HBR mode on Intel HDA
by Anssi Hannula 30 Jul '10

30 Jul '10

Hi all! We have been trying to get truehd/mlp/mat passthrough to work, with no luck so far. As HDMI standard requires a separate transmission mode (i.e. HBR Audio Stream Packets instead of Audio Sample Packets) for high-bitrate (over 6.144Mbps) compressed formats, a driver change is needed, AFAIU. Attached is a hack we have been using with hda_intel driver (the patch is for generic code; we have been using nvidia 8x codec if that matters). Could anyone familiar with Intel HDA confirm whether the hack looks correct, or if something more is required? (0x05 is the hardcoded pin nid). It would help to know for sure that the problem is in our iso61937/mat encapsulation and not in the temporary driver hack. Thanks, -- Anssi Hannula

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[alsa-devel] [PATCH - usb: Headphone support for cm6206 1/2] usb: move snd_usb_cm106_write_int_reg to mixer_quirks.c
by Adrian Pardini 30 Jul '10

30 Jul '10

Hi all, this series of patches adds another control to cm6206 based cards to let the user select wich ouput the "Headphone" jack mirrors. Please review. Thanks. Signed-off-by: Adrian Pardini <adrian.pardini(a)solar.org.ar> diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index e7df1e5..c4cbbc0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -354,6 +354,22 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, } } +/* + * C-Media CM106/CM106+ have four 16-bit internal registers that are nicely + * documented in the device's data sheet. + */ +int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value) +{ + u8 buf[4]; + buf[0] = 0x20; + buf[1] = value & 0xff; + buf[2] = (value >> 8) & 0xff; + buf[3] = reg; + return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION, + USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT, + 0, 0, &buf, 4, 1000); +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err; diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h index bdbfab0..bc5d577 100644 --- a/sound/usb/mixer_quirks.h +++ b/sound/usb/mixer_quirks.h @@ -9,5 +9,7 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, int unitid); +int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value); + #endif /* SND_USB_MIXER_QUIRKS_H */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 9a9da09..b8a5a18 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -353,22 +353,6 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) return 0; } -/* - * C-Media CM106/CM106+ have four 16-bit internal registers that are nicely - * documented in the device's data sheet. - */ -static int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value) -{ - u8 buf[4]; - buf[0] = 0x20; - buf[1] = value & 0xff; - buf[2] = (value >> 8) & 0xff; - buf[3] = reg; - return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION, - USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT, - 0, 0, &buf, 4, 1000); -} - static int snd_usb_cm106_boot_quirk(struct usb_device *dev) { /* -- 1.5.4.3 -- Adrian. http://elesquinazotango.com.ar http://www.noalcodigodescioli.blogspot.com/

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[alsa-devel] HDMI out with Zepto Nox A14, help for a patch
by Damien Thébault 30 Jul '10

30 Jul '10

Hello, My laptop is a Zepto Nox A14, with an ALC 262 audio chip. HDMI is connected through an NVIDIA GeForce 9600 graphic card. After some investigation, it appears that to output HDMI audio, I have to use the sound card which has a digital out for HDMI connected to the graphic card (there is no nvhdmi thing). I noticed with hda_analyser that there are two digital audio out nodes, the first one is the default ALC 262 one (S/PDIF). With the default configuration, when I enable what's called "HDMI" it's S/PDIF that is enabled (and I can see the red light on the mini-TOSLINK jack when it's enabled). So I was thinking that the second digital out was for the real HDMI. I went into patch_realtek.c and used the id of the second digital out (0x10) instead of ALC_262_DIGOUT_NID (0x06) in alc262_parse_auto_config. That allowed me to have a alsa device hw:0,3 connected to the digital out 0x10 instead of having it on 0x06. Then I had to enable the linked pin too, because otherwise I didn't have any sound. The pin connected to the digital out 0x10 is 0x11 (the one connected to 0x06 is 0x1e). So in the end, I'm able to have HDMI sound! Now here are my questions: - how can I add this with SND_PCI_QUIRK? I think that I have to find the IDs of my setup (PCI ID?) and then create a preset. How can I find this ID on my system? There is a dig_out_nid field in the preset so I would use 0x10 instead of ALC262_DIGOUT_NID (0x06). I don't know how to change the pin from 0x1e to 0x11, is it automatic? - doing this will disable my S/PDIF output, is it possible to juste have both outputs available? Thanks! -- Damien Thebault

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[alsa-devel] Issues with ATI IXP SB4x0
by Simos Xenitellis 30 Jul '10

30 Jul '10

An Acer laptop with the ATI IXP SB4x0 chipset fails to detect the audio card. Here is the alsa-info file with latest Alsa, http://www.alsa-project.org/db/?f=b63339af32dd301eba9f5ef6a0787ccfbeb55778 What should be the next thing to try in order to solve this issue? The "Conexant ID 2c06" string in the file appears to refer to the modem (?). Googling with the subsystem ID '1025:009f only brings cases where the sound does not work. Any suggestions on where to look at? Simos

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[alsa-devel] Fixing ALC269 the right way?
by David Henningsson 30 Jul '10

30 Jul '10

Here's what I try to fix by playing around with it in hda-emu, it seems like more than one thing is broken here, but let's start with playback. This machine has one internal speaker and one line-out. Codec proc file is attached. Here's my analysis so far: 1) If there's one internal speaker and one line-out (rather than headphone-out), should the internal speaker auto-mute? What's standard practice? 2) There are two DAC nids, 0x02 and 0x03, but only 0x02 is used (i e activated on playback). But line-out is connected (via nid 0x0d) to DAC 0x03, so there is no sound on line-out. Should we try to activate DAC nid 0x03 or try to move line-out to DAC 0x02? And is either possible through just fiddling with pin_configs (which Takashi preferred), or should we add fixup verbs, or a completely new model? What would upstream prefer in this case? 3) Another strange thing is the controls, there is both Master Playback and Speaker Playback Volumes and Switches, but they both control Volume=0x02 and Switch=0x14, and setting one then getting the other returns the wrong result since they're not updating each other's cached values (I assume). So we should try to get rid of one of these, but how depends on the answer to question 1... Thanks in advance for helping out! -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[PATCH 5/5] Add analog output NIDs used by Dell Optiplex
by David Henningsson 30 Jul '10

30 Jul '10

This patch enables playback on Dell Optiplex XE, 380 and 980. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b4b7f58..0b87715 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13108,6 +13108,8 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, dac = 0x02; break; case 0x15: + case 0x1a: + case 0x1b: case 0x21: /* ALC269vb has this pin, too */ dac = 0x03; break; -- 1.7.0.4 --------------040609020801060909050209--

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[alsa-devel] PROPOSAL: Layer Signal Processing Architecture
by Ilya D 29 Jul '10

29 Jul '10

Hello. I have a proposal for this Layered Model, which would allow implementation of new audio interfaces with DSP cores, with cross-platform compatibility and audio networking in mind. it would be a sort of OSI for signal processing, but with some extra stuff to it, such as a common language (library) for hardware programming. it's not to replace ALSA, but just to bring a different approach for design engineers of professional audio applications. we all know that these days a lot manufactuares chose Linux for their Mixing Consoles (such as Midas, Lawo, Calrec and others), also other products use Linux, but there might be only a few who use ALSA (as far as i could find out there rather NONE in pro-audio Linux devices). Linux runs on devices such as Blackfin and OMAP, which have DSP cores, but the interface is device-specific, hence low-level. ALSA had been designed for conventional sound-cards (largely) also there drivers for RME, DigiGram and other pro-interfaces, but the hardware is still proprietary and it isn't quite flexible. The idea is to bring new concepts to the world, well not 100% new, but new in a way. One of these would be to implement an "open-source" audio processor interface of a kind. It would have configurable channel strips with (multi-band EQ and dynamics). Another motivating factor is upcomming completition of AVB ethernet standard from IETF/IEEE, there had been no publicaly avaliable code for this, nor i could find any discussions of how it could be implemented in Linux. However i have contact with a professor Richard Foss from Rhodes University, and there they have implemented a library and packet generator for AVB, though they haven't yet published this code. There also hasn't heppend any wide adoption of OSC, and may be OSC is not that great? I am currently working on "Control and Monitoring Protocol" which could take part in this architercture on it's TOP Layer. The fallowing are the ABSTRACTION layers that i have thought being most general: ~>> Control, Monitoring and Information (CMI) ACMP (above mentioned above, it's kind of OSC-based) or AES-x170 (both can fit together) also OSC ..or even MIDI but ACMP is more general, it should be good for lighting and other controll applications .. here is the place for GUI interfaces, hardware controllers, control surfaces ...etc ~>> Signal Processing Subsystem (SPS) ~>> Core Signal Bus (CSB) : our inter-connect it has one common clock one or more control ports and one or more signal ports (a port can be input, output, or both) the control ports are exposed to CMI layer the signal prorts can be: bridged (over Ethernet (i.e. AVB), USB or FireWire) interfaced (to say I2S, AES/EBU, AES-50 ..MADI, ADAT) or streamed to a codec or uncompressed file dumping Let's call it an Port Exposure Layer ~>> Core Signal Elements (CSE) the main abstraction for this is library (our cross-platform API) compiler (to produce platform dependent code) interpreter (to configure and glue the blocks) ~>> General Purpose Operating System Of course it is still there ..it could be on a remote machine or anything ..and it could be everywhere around, even General Purpose CPU could be a fall-back option ..well, this is all about abstraction after all :) This is with parallel audio clustering applications and distributed chain processing in mind. Looking forward to here some discussion, I would like to set up a wiki and discusion list for this! Regards, -- Ilya Dmitrichenko Илья Дмитриченко email: errordeveloper(a)gmail.com

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[alsa-devel] Equalizer plugin
by John Lindgren 29 Jul '10

29 Jul '10

Hi, I know at present it is possible to set up a system-wide equalizer by using a LADSPA effect configured in .asoundrc, but (to my knowledge) adjusting such a setup requires editing .asoundrc and then restarting all applications using ALSA. I've cooked up an external PCM plugin that uses its own config file to store equalizer settings. Since it watches the config file with inotify, it is possible to change the settings without restarting any applications. I plan to eventually write a GUI for it. So ... is this useful? Any suggestions? -- John

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[alsa-devel] Driver for AK4555 codec
by Yoran Brondsema 29 Jul '10

29 Jul '10

Hi, For a system here at work, we have a simple audio system consisting of an I2S-bus audio controller embedded in the CPU (an NXP LPC3250) and a AK4555 codec which communicates with the CPU through the I2S-bus and produces (or captures) analog signals. The codec is extremely simple; it is basically an A/D & D/A converter and it doesn't support a control interface such as I2C. There is a driver available for the LPC3250 but none for the AK4555 codec, so I have to write it myself. I checked the document "Writing an ALSA Driver" by Takashi Iwai but it seems to focus on full-featured sound cards; I'm just dealing with a "simple" codec. I checked at the code of the uda1380 in linux/sound/soc/lpc3xxx/lpc3xxx-uda1380.c and it seems to be calling some functions like platform_device_alloc, platform_set_drvdata,... but I can't find the documentation of what these functions do. Can someone give me some pointers to some documentation or some rough idea of how it needs to be done? Thank you, Yoran

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[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 29 Jul '10

29 Jul '10

The following changes since commit a7e7cd5bd7d1e0134032b8db5e64ceb9dac8b3ca: ASoC: da7210: Add HeadPhone Playback Volume control (2010-07-23 10:17:47 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Kuninori Morimoto (3): ASoC: fsi: remove unnecessary clock processing ASoC: fsi: remove device id check ASoC: fsi: Add new funtion for SPDIF include/sound/sh_fsi.h | 2 + sound/soc/sh/fsi.c | 70 +++++++++++++++++++++++++++++++++++++---------- 2 files changed, 57 insertions(+), 15 deletions(-)

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