I'm using libavformat/libavcodec/x264/libmp3lame to encode/mux
captured pictures/audio samples on iPhone 4. My spec is H.264/MP3 in
MPEG2TS format. The app runs fine if it's video stream only. My frame
rate is 10 fps (i.e., 100 ms for every frame). So, the pts interval
for my video stream is 9000 based on a 90KHz clock. Now I'm trying to
include the audio stream but failing. It seems the audio PTS is messed
up. This is how I'm doing it: I record 100 ms worth of audio data
every time and encode them into MP3 packet; before multiplexing
(av_interleaved_write_frame), I assign the packet a PTS with the same
interval as video (9000). The resulted audio stream sounds garbage and
much short than the video.

Also, I was trying to compile libfaac for iPhone 4 (armv7) but
couldn't get it right. When linking my code, the compiler has been
complaining about the libfaac is not build for armv7. Has anyone made
it work?

Thanks in advance.
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