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Diffstat (limited to 'DOCS/OUTDATED-tech/hwac3.txt')
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diff --git a/DOCS/OUTDATED-tech/hwac3.txt b/DOCS/OUTDATED-tech/hwac3.txt deleted file mode 100644 index 25a2c5bc7a..0000000000 --- a/DOCS/OUTDATED-tech/hwac3.txt +++ /dev/null @@ -1,145 +0,0 @@ -mails by A'rpi and Marcus Blomenkamp <Marcus.Blomenkamp@epost.de> -describing how this ac3-passtrough hack work under linux and mplayer... ------------------------------------------------------------------------ -Hi, - -> I received the following patch from Steven Brookes <stevenjb@mda.co.uk>. -> He is working on fixing the digital audio output of the dxr3 driver and -> told me he fixed some bugs in mplayer along the way. I don't know shit -> about hwac3 output so all I did was to make sure the patch applied -> against latest cvs. -> This is from his e-mail to me: -> -> "Secondly there is a patch to dec_audio.c and -> ac3-iec958 to fix the -ac hwac3 codec stuff and to use liba52 to sync it. - -> Seems to work for everything I've thrown at and maintains sync for all audio -> types through the DXR3." - -patch applied (with some comments added and an unwanted change (in software -a52 decoder) removed) - -now i understand how this whole hwac3 mess work. -it's very very tricky. it virtually decodes ac3 to LPCM packets, but really -it keeps the original compressed data padded by zeros. this way it's -constant bitrate, and sync is calculated just like for stereo PCM. -(so it bypass LPCM-capable media converters...) - -so, every ac3 frame is translated to 6144 byte long tricky LPCM packet. -6144 = 4*(6*256) = 4 * samples_per_ac3_frame = LPCM size of uncompressed ac3 -frame. - -i wanna know if it works for sblive and other ac3-capable cards too? -(i can't test it, lack of ac3 decoder) - -A'rpi / Astral & ESP-team - ------------------------------------------------------------------------ -Hi folks. -I spend some time fiddling with ac3 passthrough in mplayer. The -traditional way of setting the output format to AFMT_AC3 was no ideal -solution since not all digital io cards/drivers supported this format or -honoured it to set the spdif non-audio bit. To make it short, it only -worked with oss sblive driver IIRC. - -Inspired by alsa's ac3dec program I found an alternative way by -inspecting to which format the alsa device had been set. Suprise: it was -simple 16bit_le 2_channel pcm. So setting the non-audio bit doesn't -necessarily mean the point. The only important thing seems to be -bit-identical output at the correct samplerate. Modern AV-Receivers seem -to be quite tolerant/compatible. - -So I changed the output format of hwac3 from - -AFMT_AC3 channels=1 - to -AFMT_S16_LE channels=2 - -and corrected the absolute time calculation. That was all to get it -running for me. - ------------------------------------------------------------------------ -Hi there. - -Perhaps I can clear up some mystification about AC3 passthrough in -general and mplayer in special: - -To get the external decoder solution working, it must be fed with data -which is bitidentical to the chunks in the source ac3 file (compressed -data is very picky about bit errors). Additionally - or better to say -'historically' - the non-audio bit should be set in the spdif status -fields to prevent old spdif hardware from reproducing ugly scratchy -noise. Note: for current decoders (probably those with DTS capability) -this safety bit isn't needed anymore. At least I can state that for my -Sherwood RVD-6095RDS. I think it is due to DTS because DTS sound can -reside on a ordinary AudioCD and an ordinary AudioCD-Player will always -have it's audio-bit set. - -The sample format of the data must be 2channel 16bit (little endian -IIRC). Samplerates are 48kHz - although my receiver also accepts -44100Hz. I do not know if this is due to an over-compatability of my -receiver or if 44100 is also possible in the ac3 specs. For safety's -sake lets keep this at 48000Hz. AC3 data chunks are inserted into the -stream every 0x1600 bytes (don't bite me on that, look into -'ac3-iec958.c': 'ac3_iec958_build_burst'). - -To come back to the problem: data must be played bit-identically through -the soundcard at the correct samplerate and should optionally have it's -non-audio bit set. There are two ways to accomplish this: - -1) Some OSS guy invented the format AFMT_AC3. Soundcard drivers -implementing this format should therefore adjust it's mixers and -switches to produce the desired output. Unfortunately some soundcard -drivers do not support this format correctly and most do not even -support it at all (including ALSA). - -2) The alternative approach currently in mplayer CVS is to simply set -the output format to 48kHz16bitLE and rely on the user to have the -soundcard mixers adjusted properly. - -I do have two soundcards with digital IO facilities (CMI8738 and -Trident4DWaveNX based) plus the mentioned decoder. I'm currently running -Linux-2.4.17. Following configurations are happily running here: - -1. Trident with ALSA drivers (OSS does not support Hoontech's dig. IO) -2. CMI with ALSA drivers -3. CMI with OSS drivers - -For Linux I'd suggest using ALSA because of it's cleaner architecture -and more consitent user interface. Not to mention that it'll be the -standard sound support in Linux soon. - -For those who want to stick to OSS drivers: The CMI8738 drivers works -out-of-the-box, if the PCM/Wave mixer is set to 100%. - -For ALSA I'd suggest using its OSS emulation. More on that later. -ALSA-0.9 invented the idea of cards, devices and dubdevices. You can -reach the digital interface of all supported cards consitently by using -the device 'hw:x,2' (x counting from 0 is the number of your soundcard). -So most people would end up at 'hw:0,2'. This device can only be opened -in sample formats and rates which are directly supported in hardware -hence no samplerate conversion is done keeping the stream as-is. However -most consumer soundcards do not support 44kHz so it would definitively -be a bad idea to use this as your standard device if you wanted to -listen to some mp3s (most of them are 44kHz due to CD source). Here the -OSS comes to play again. You can configure which OSS device (/dev/dsp -and /dev/adsp) uses which ALSA device. So I'd suggest pointing the -standard '/dev/dsp' to standard 'hw:0,0' which suports mixing and -samplerate conversion. No further reconfiguration would be needed for -your sound apps. For movies I'd point '/dev/adsp' to 'hw:0,2' and -configure mplayer to use adsp instead of dsp. The samplerate constrain -is no big deal here since movies usually are in 48Khz anyway. The -configuration in '/etc/modules.conf' is no big deal also: - -alias snd-card-0 snd-card-cmipci # insert your card here -alias snd-card-1 snd-pcm-oss # load OSS emulation -options snd-pcm-oss snd_dsp_map=0 snd_adsp_map=2 # do the mapping - -This works flawlessly in combination with alsa's native -SysVrc-init-script 'alsasound'. Be sure to disable any distribution -dependent script (e.g. Mandrake-8.1 has an 'alsa' script which depends -on ALSA-0.5). - -Sorry for you *BSD'lers out there. I have no grasp on sound support there. - -HTH Marcus |