3 mails describing hwac3 tech details

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@4779 b3059339-0415-0410-9bf9-f77b7e298cf2

1 files changed, 145 insertions, 0 deletions

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 <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 
+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