Re: [Alsa-user] Digital Sound (AC3/DD/DTS) / Yamaha DSP A1 / cs46xx (Hercules Digifire 7.1)
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From: Dominique D. <dom...@fr...> - 2004-06-04 19:19:15
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Philipp Morger <phi...@do...> writes: > Audio: RealTek ALC650 6-channel audio CODEC > > That's the one I have onboard, and this one is working for me... they > look to me the same... what does you iecset show, while you are playing > an ac3 encoded sound? (preferably in a player where dts works....) $ iecset Mode: consumer Data: non-audio Rate: 48000 Hz Copyright: protected Emphasis: none Category: general Original: 1st generation Clock: 1000 ppm (I switched off the copyright to no avail) But iecset -x gives: $ iecset -x AES0=0x02,AES1=0x00,AES2=0x00,AES3=0x02 Notice that the AES0 and AES1 do not fit the ac3dec command: ./ac3dec -C -Diec958:AES0=0x6,AES1=0x82,AES2=0x0,AES3=0x2 dolby.ac3 Is that normal ? > can you give me your xine-config (without the #'s please), version and > some for mplayer? possibly and mplayer output, like what happens with I've removed from mplayer.conf the parameters not related to sound: vo=xv,x11, ao=alsa1x, Dolby digital (but no sound :-( ): $ mplayer -aid 130 -ac hwac3 -ao alsa1x:spdif -vo null -v dvd://4 MPlayer 1.0pre4-3.3.3 (C) 2000-2004 MPlayer Team [...] [open] audio stream: 2 audio format: ac3 (5.1) language: en aid: 130 [...] DEMUXER: freeing demuxer at 0x8730460 ASF_check: not ASF guid! [...] Forced audio codec: hwac3 Opening audio decoder: [hwac3] AC3/DTS pass-through SP/DIF dec_audio: Allocating 8192 bytes for input buffer. dec_audio: Allocating 16384 + 65536 = 81920 bytes for output buffer. hwac3: switched to AC3, 448000 bps, 48000 Hz AUDIO: 48000 Hz, 2 ch, 16 bit (0x400), ratio: 56000->192000 (448.0 kbit) Selected audio codec: [hwac3] afm:hwac3 (AC3 through SPDIF) [...] ========================================================================== Checking audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/8bit... [libaf] Adding filter dummy [dummy] Was reinitialized, rate=48000Hz, nch = 2, format = 0x00000020 and bps = 2 AF_pre: af format: 2 bps, 2 ch, 48000 hz, big endian AC3 AF_pre: 48000Hz 2ch AC3 alsa-init: requested format: 48000 Hz, 2 channels, AC3 alsa-init: compiled for ALSA-1.0.4 alsa-spdif-init: playing AC3, 2 channels alsa-init: soundcard set to iec958:AES0=0x2,AES1=0x82,AES2=0x0,AES3=0x2 alsa-init: pcm opened in block-mode alsa-init: chunksize set to 1024 alsa-init: fragcount=16 alsa-init: got buffersize=65536 alsa1x: 48000 Hz/2 channels/4 bpf/65536 bytes buffer/Signed 16 bit Little Endian AO: [alsa1x] 48000Hz 2ch AC3 (1 bps) AO: Description: ALSA-1.x audio output AO: Author: Alex Beregszaszi, Joy Winter <jo...@pi...> AO: Comment: under developement Building audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/8bit... [dummy] Was reinitialized, rate=48000Hz, nch = 2, format = 0x00000020 and bps = 2 [libaf] Adding filter format Starting playback... DTS (works): $mplayer -aid 139 -ac hwac3 -ao alsa1x:spdif -vo null -v dvd://4 [open] audio stream: 3 audio format: dts (5.1) language: en aid: 139 [...] DEMUXER: freeing demuxer at 0x8730460 ASF_check: not ASF guid! [...] ========================================================================== Forced audio codec: hwac3 Opening audio decoder: [hwac3] AC3/DTS pass-through SP/DIF dec_audio: Allocating 8192 bytes for input buffer. dec_audio: Allocating 16384 + 65536 = 81920 bytes for output buffer. hwac3: switched to DTS, 768000 bps, 48000 Hz AUDIO: 48000 Hz, 2 ch, 16 bit (0x400), ratio: 96000->192000 (768.0 kbit) Selected audio codec: [hwac3] afm:hwac3 (AC3 through SPDIF) ========================================================================== [...] Checking audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/8bit... [libaf] Adding filter dummy [dummy] Was reinitialized, rate=48000Hz, nch = 2, format = 0x00000020 and bps = 2 AF_pre: af format: 2 bps, 2 ch, 48000 hz, big endian AC3 AF_pre: 48000Hz 2ch AC3 alsa-init: requested format: 48000 Hz, 2 channels, AC3 alsa-init: compiled for ALSA-1.0.4 alsa-spdif-init: playing AC3, 2 channels alsa-init: soundcard set to iec958:AES0=0x2,AES1=0x82,AES2=0x0,AES3=0x2 alsa-init: pcm opened in block-mode alsa-init: chunksize set to 1024 alsa-init: fragcount=16 alsa-init: got buffersize=65536 alsa1x: 48000 Hz/2 channels/4 bpf/65536 bytes buffer/Signed 16 bit Little Endian AO: [alsa1x] 48000Hz 2ch AC3 (1 bps) AO: Description: ALSA-1.x audio output AO: Author: Alex Beregszaszi, Joy Winter <jo...@pi...> AO: Comment: under developement Building audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/8bit... [dummy] Was reinitialized, rate=48000Hz, nch = 2, format = 0x00000020 and bps = 2 [libaf] Adding filter format Starting playback... > and if you get sound and what dts light you have. AC3: no sound DTS: sound with red light. > Somehow I can't belive that an intel/nvidia chipset make all the > difference..... Weel, there's always the possibility that the embedded dolby digital encoder (the soundstorm stuff) muck things up. Even though it's not used. > Please also tell me what input you used on the yamaha amp DVD/VCR3 Frankly, I'm stumped. The only lead I can think of is this weird output of iecset -x. Thanks for your time. |