annotate DOCS/tech/hwac3.txt @ 31290:f60cd3b9d453

libmpeg2: Move pending_buffer stuff to local decoder context. It is only used in our wrapper code, so there is no point to patch it into our libmpeg2 copy. This also helps when trying to use external libmpeg2. patch by Luca Barbato

mails by A'rpi and Marcus Blomenkamp <[e-mail]>
describing how this ac3-passtrough hack work under linux and mplayer...
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Hi,

> I received the following patch from Steven Brookes <[e-mail]>.
> 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

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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.

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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