From 44e1c06eb7727171298a9852fc53bd84930c0a9c Mon Sep 17 00:00:00 2001 From: arpi Date: Thu, 21 Feb 2002 02:01:46 +0000 Subject: [PATCH] 3 mails describing hwac3 tech details git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@4779 b3059339-0415-0410-9bf9-f77b7e298cf2 --- DOCS/tech/hwac3.txt | 145 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 145 insertions(+) create mode 100644 DOCS/tech/hwac3.txt diff --git a/DOCS/tech/hwac3.txt b/DOCS/tech/hwac3.txt new file mode 100644 index 0000000000..e5e42670d4 --- /dev/null +++ b/DOCS/tech/hwac3.txt @@ -0,0 +1,145 @@ +mails by A'rpi and Marcus Blomenkamp +describing how this ac3-passtrough hack work under linux and mplayer... +----------------------------------------------------------------------- +Hi, + +> I received the following patch from Steven Brookes . +> 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 +dependant 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