Re: [Alsa-user] Combined USB-Serial Serial-MIDI device
Rangel Reale wrote: modprobe snd-serialmidi sdev=/dev/ttyUSB0 Hmm the serialmidi driver seems to be broken with the kernel Ubuntu 8.04 uses (2.6.24-17-generic) (I had to enable CONFIG_BROKEN on configure to be allowed to compile it) /usr/src/alsa-driver-1.0.16/drivers/serialmidi.c:117: error: conflicting types for ‘tty_ioctl’ Well, there is a reason it's marked as broken ... It should be possible to fix this driver, or to write a user-space driver. I'll see what I can do. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] nvidia + HDMI = no sound
hello list! I have my LCD TV connected to my pc with HDMI cable. The problem is there is no sound from TV speakers. OS: Ubuntu 6.06 LTS kernel: 2.6.15-27-686 nvidia card: GeForce 6200 nvidia driver version: 169.12 (latest) sound card: SoundBlaster Live! alsa versions: - alsa-driver 1.0.16 (latest) - alsa-lib 1.0.16 (latest) - alsa-utils 1.0.16 (latest) Latest alsa driver was build with: ./configure --with-cards=emu10k1 --with-card-options=all --with-kernel=/usr/src/linux-headers-2.6.15-27-686 Sound is working perfectly on the pc. My question is, does latest ALSA drivers (+ nvidia) supports sound over HDMI cable? thank you Anze - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] please help with audio problem from hauppauge wintv go fm card
hi there ppl i got a analogue wintv go with fm card (i think). It has radio for sure, 1x composite-in, audio out, and an IR. I run ubuntu hardy 64bit system. tvtime xawtv etc work without a problem, only i can't adjust the volume from within the applications. I have atm the line out of my tv card connected to the line-in of my soundcard. That's how i get my audio from my tv card. In my old computer however, with the same tv card and ubuntu feisty i got it working to get the audio from the tvcard itself and not using the line-in of my soundcard, but i have no idea how i got that working, i only know it took me while. Can someone help me with that please? here is some info about my system: uname -r 2.6.24-16-generic lspci 00:00.0 Host bridge: Intel Corporation 82Q35 Express DRAM Controller (rev 02) 00:01.0 PCI bridge: Intel Corporation 82Q35 Express PCI Express Root Port (rev 02) 00:03.0 Communication controller: Intel Corporation 82Q35 Express MEI Controller (rev 02) 00:03.2 IDE interface: Intel Corporation 82Q35 Express PT IDER Controller (rev 02) 00:03.3 Serial controller: Intel Corporation 82Q35 Express Serial KT Controller (rev 02) 00:19.0 Ethernet controller: Intel Corporation 82566DM-2 Gigabit Network Connection (rev 02) 00:1a.0 USB Controller: Intel Corporation 82801I (ICH9 Family) USB UHCI Controller #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801I (ICH9 Family) USB UHCI Controller #5 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801I (ICH9 Family) USB2 EHCI Controller #2 (rev 02) 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801I (ICH9 Family) PCI Express Port 1 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801I (ICH9 Family) USB UHCI Controller #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801I (ICH9 Family) USB UHCI Controller #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801I (ICH9 Family) USB UHCI Controller #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801I (ICH9 Family) USB2 EHCI Controller #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev 92) 00:1f.0 ISA bridge: Intel Corporation 82801IO (ICH9DO) LPC Interface Controller (rev 02) 00:1f.2 SATA controller: Intel Corporation 82801IR/IO/IH (ICH9R/DO/DH) 6 port SATA AHCI Controller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801I (ICH9 Family) SMBus Controller (rev 02) 01:00.0 VGA compatible controller: nVidia Corporation GeForce 8400 GS (rev a1) 03:00.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 02) 03:00.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 02) lspci -v | grep a4 Brooktree 03:00.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 02) Subsystem: Hauppauge computer works Inc. WinTV Series Flags: bus master, medium devsel, latency 64, IRQ 16 Memory at d0001000 (32-bit, prefetchable) [size=4K] 03:00.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 02) Subsystem: Hauppauge computer works Inc. WinTV Series Flags: bus master, medium devsel, latency 64, IRQ 16 Memory at d000 (32-bit, prefetchable) [size=4K] lsmod Module Size Used by nls_cp437 8320 10 cifs 251152 10 binfmt_misc 14860 1 ppdev 11400 0 acpi_cpufreq 10832 0 cpufreq_powersave 3200 0 cpufreq_conservative 10632 0 cpufreq_userspace 6180 0 cpufreq_stats 8416 0 cpufreq_ondemand 11152 2 freq_table 6464 3 acpi_cpufreq,cpufreq_stats,cpufreq_ondemand video 23444 0 output 5632 1 video bay 8064 0 container 6656 0 sbs 17808 0 sbshc 8960 1 sbs dock 12960 1 bay battery 16776 0 iptable_filter 4608 0 ip_tables 24104 1 iptable_filter x_tables 23560 1 ip_tables ac 8328 0 coretemp 9856 0 lp 14916 0 ipv6 311720 28 bt878 13672 0 tuner 49056 0 tea5767 7812 1 tuner tda8290 13828 1 tuner tuner_simple 10632 1 tuner mt20xx 14600 1 tuner tea5761 6916 1 tuner tvaudio 28188 0 snd_bt87x 19076 1 bttv 214772 1 bt878 snd_hda_intel 440408 5 ir_common 39812 1 bttv snd_pcm_oss 47648 0 snd_mixer_oss 20224 1 snd_pcm_oss compat_ioctl32 11136 1 bttv i2c_algo_bit 8452 1 bttv psmouse 46236 0 videobuf_dma_sg 17028 1 bttv videobuf_core 22020 2 bttv,videobuf_dma_sg btcx_risc 6792 1 bttv dcdbas 11312 0 serio_raw 9092 0 parport_pc 41128 1 parport 44300 3 ppdev,lp,parport_pc pcspkr 4992 0 snd_pcm 92168 3 snd_bt87x,snd_hda_intel,snd_pcm_oss tveeprom 20624 1 bttv videodev 30720 1 bttv v4l2_common 21888 5 tuner,tvaudio,bttv,compat_ioctl32,videodev v4l1_compat 15492 2 bttv,videodev nvidia 8858052 34 snd_page_alloc 13200 3 snd_bt87x,snd_hda_intel,snd_pcm snd_hwdep 12552 1 snd_hda_intel evdev 14976 4 snd_seq_dummy 5764 0 snd_seq_oss 38912 0 i2c_core 28544 11 tuner,tea5767,tda8290,tuner_simple,mt20xx,tea5761, tvaudio,bttv,i2c_algo_bit,tveeprom,nvidia snd_seq_midi 10688 0 snd_rawmidi 29856 1 snd_seq_midi snd_seq_midi_event 10112 2 snd_seq_oss,snd_seq_midi snd_seq 63232 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_mid i_event
[Alsa-user] Any way to count samples written out to the ADC?
Is there any way to divine a count of samples that leave the ALSA ring buffer during playback? My application requires that I send a callback when a buffer-full of data is actually played for it to do A/V sync correctly. Right now I'm doing math to figure out the right clock time to issue the callback, using the latency value from snd_pcm_delay(), but this is not working very well. Thanks in advance. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Any way to count samples written out to the ADC?
2008/5/28 Paul Adolph [EMAIL PROTECTED]: Is there any way to divine a count of samples that leave the ALSA ring buffer during playback? My application requires that I send a callback when a buffer-full of data is actually played for it to do A/V sync correctly. Right now I'm doing math to figure out the right clock time to issue the callback, using the latency value from snd_pcm_delay(), but this is not working very well. Thanks in advance. I don't know what sort of A/V method you are using, but needing a callback for it seems wrong to me. snd_pcm_delay() gives you a way to calculate the delay between you writing samples to the buffer and them actually arriving at the speakers. You can therefore calculate the exact time a particular sample will reach the speakers, and work out the A/V sync required from that. This is how xine works to achieve A/V sync. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Any way to count samples written out to the ADC?
On Wed, May 28, 2008 at 9:04 AM, James Courtier-Dutton [EMAIL PROTECTED] wrote: 2008/5/28 Paul Adolph [EMAIL PROTECTED]: Is there any way to divine a count of samples that leave the ALSA ring buffer during playback? My application requires that I send a callback when a buffer-full of data is actually played for it to do A/V sync correctly. Right now I'm doing math to figure out the right clock time to issue the callback, using the latency value from snd_pcm_delay(), but this is not working very well. Thanks in advance. I don't know what sort of A/V method you are using, but needing a callback for it seems wrong to me. Tell me about it. I'm porting an existing system to a new platform and unfortunately this is the model I'm stuck with. snd_pcm_delay() gives you a way to calculate the delay between you writing samples to the buffer and them actually arriving at the speakers. You can therefore calculate the exact time a particular sample will reach the speakers, and work out the A/V sync required from that. This is how xine works to achieve A/V sync. OK - that is the method I'm using. Good to know that it is used elsewhere with success. There must be a bug in my code somewhere. Thanks for your help. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] SOLUTION: 2-ch from SPDIF works, AC3 passthrough doesn't on ICE1724
All, Found the problem, so I thought I would post. Although it is rarely mentioned in Digital Sound HowTos, the IEC958 (aka SPDIF) interface has field that indicates whether its sending 'audio' or not. This field can be set by a parameter called AES0 or a tool called iecset. My SPDIF was working, and I could hear audio in 2 channels, but when you use passthrough dolby digital it needs the audio bit *unset* -- and neither Xine or Mplayer was doing this automatically. I made a little script that solved the problem, that looked like this: #/bin/sh iecset audio 0 xine parameters iecset audio 1 ...This disables the audio bit before xine plays (using passthrough) then enables it back again so MP3s and standard digital PCM can play when Xine is done. Now everything works great. :) Regards, -- Joe - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] raw spdif frame capture
Hi: I would like to perform raw spdif frame capture (all 64-bits, in 2 32-bit subframes) using an EMU 1212M under ALSA. Can this be done? If so, has anyone else already done it? If not, does anyone have any idea how to do it? EMU has claimed that it is possible with the hardware, but that the driver would need to support it (windows driver does not, they didn't know about the ALSA driver). Does the current emu10k1 driver support this? How would I enter this mode? Thanks, Brett Andrews Software Engineer BSE Reference Software - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] nvidia + HDMI = no sound
On Wed, May 28, 2008 at 5:32 AM, Anže Vidmar [EMAIL PROTECTED] wrote: My question is, does latest ALSA drivers (+ nvidia) supports sound over HDMI cable? No. It's not an ALSA issue but an nvidia issue. They need to fix their driver. Welcome to the wonderful world of DRM. Lee - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Default values in .asoundrc
On Mon, May 26, 2008 at 10:18 PM, Alejandro Benitez [EMAIL PROTECTED] wrote: Hi, First, let me thank klondike for the help, my mic is working again. I have another question: there are some variables like period_time, period_size, buffer_size, rate, etc that have some obscure values for me. Is it possible to get rid of these so that default values take place? Which are the ones I can get rid of? Many thanks in advance again. Why do you use an .asoundrc at all? It shouldn't be needed unless you have a REALLY old alsa version. Lee - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] dmix and mic
On Mon, May 26, 2008 at 6:02 PM, klondike [EMAIL PROTECTED] wrote: I use to talk on this on my speeches, you just lack an asym pcm with can join the microphone with the dmixed output. This conf also adds dsnoop so varios aplications can read data from the same microphone at the same time :) You should not need an .asoundrc at all, this all works by default since ALSA 1.0.9 or so. Lee - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA + SigmaTel STAC92xx sound chip - Device busy
On Sun, May 25, 2008 at 9:10 PM, Polo Talnir [EMAIL PROTECTED] wrote: I have a SigmaTel STAC9221D A2 chip embedded in my Intel motherboard The full ALSA info is uploaded to http://pastebin.ca/1024025 I am running Fedora Core 5 (Bordeaux) with kernel 2.6.18-1.2257_FC5smp on an Intel mobo D955XBK with a Pentium D 940 (dual core processor) @ 3.2GHz and 4GB of RAM. I am using ALSA 1.0.6 which I compiled. That is a REALLY old ALSA version. With a recent one you won't get the device busy issues because it does mixing in software by default. I have sound when playing a file or CD with XMMS. I can also play what comes in through the Line-In, using the following command line: arecord -D hw:1,0,0 -f S16_LE -c2 -r48000 | aplay -D hw:1 Question 0: why should I need to use that command line?! The command-line info I got here: http://people.atrpms.net/~pcavalcanti/alsa-1.0.15rc2_snd-hda-intel.html I thought I should be able to mix-in the signal from the selected input (the chip seems to accept only one ext input at a time) and hear it together (mixed) with, e.g. the PCM channel using just the ALSA mixer. I seem to have a configuration problem right here. Try ALSA 1.0.16. Possibly the issue has been fixed. Lee - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] dmix and mic
I'm running Ubuntu 8.04. So there must be a problem with this distro because I need an .asoundrc, otherwise things work (badly) as described in my first post. On Wed, May 28, 2008 at 9:48 PM, Lee Revell [EMAIL PROTECTED] wrote: On Mon, May 26, 2008 at 6:02 PM, klondike [EMAIL PROTECTED] wrote: I use to talk on this on my speeches, you just lack an asym pcm with can join the microphone with the dmixed output. This conf also adds dsnoop so varios aplications can read data from the same microphone at the same time :) You should not need an .asoundrc at all, this all works by default since ALSA 1.0.9 or so. Lee - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] dmix and mic
On Wed, May 28, 2008 at 9:45 PM, Alejandro Benitez [EMAIL PROTECTED] wrote: I'm running Ubuntu 8.04. So there must be a problem with this distro because I need an .asoundrc, otherwise things work (badly) as described in my first post. What exactly happens if you remove the .asoundrc? What is the output of lsmod | grep snd? Lee - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
