Hi
I want to realize an application that features a constant but realtime
adjustable delay between playback and capture with mmap alsa lib usage. The
idea is to use up/down button with process messaging to fine tune sync between
audio and video. I am using a commercial DVB-S reciever with AC3-OUT connected
to the Linux PC SPDIF input of the soundcard and a subsequent TV for video
display of the STB. Audio as about 500ms leading the corresponding image
sequence. Which is good in general since I also plan to chain brutefir for DRC
which gives we room for max 22000 samples FIR filter length/channel to be back
in sync.
I know this can be accomplished by proper buffer length settings for a fixed
delay. But it should be adjustable to use it for other purpose as well.
Unfortunately I am not able to find a detailed diagram how alsa uses itsÂ
ring-buffer and how interrupts do affect delay as well. Can I use snd_pcm_wait,
snd_pcm_delay, snd_pcm_forward, and snd_pcm_rewind for that purpose and how? Is
there a code fragment available?
Or in general, exists a code for SPDIF input to analog AC3 output with realtime
adjustable delay? I managed easily to get it to work down to 3ms with arecord |
ffmpeg spdif copy and ac3dec. For the application I am thinking to modify
clalsadrv, ac3dec and irmp3 for my stand alone AC3 receiver Linux project.
Better ideas welcome? But don't say JACK!
Regards
JohnÂ
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