[Libav-user] ffmpeg for matlab (mex function)
researchiteng at gmail.com
Tue Apr 8 14:28:05 CEST 2014
I am implementing ffmpeg as a mex function in matlab as part of a research
project (with main focus on the audio, not video like OpenCV, etc).
I managed to have an initial version of audio encoding working, but I need
to clarify few things before considering it usable by others with other
codecs that I tried.
1. Newbie questions section:
1.1: Few definitions which the documentation expects are very clear to the
developer (which is not this case).
avctx->sample_fmt => is the input or the output format? (for my current
codec I am lucky both input and output formats are the same, but I need to
know for other situations).
avctx->sample_rate => for the input or for the output? From what I
understand it's for output, correct?
avctx->bit_rate => of the input of of the output? From what I understand
it's for output, correct?
avctx->bits_per_raw_sample => I understand it's for input. Shall I set it,
or it defaults to s16 by itself? In the examples I did not find this being
avctx->frame_size => I understood is for input, correct?
1.2: how to pass from an external program, things like AVCodecID, which are
enum type. It int good enough (size wise) or I should send int64_t to be
1.3: bit_rate seems to be defined as int in the avcodecontext. Is it able
to accept higher bit_rate values as well?
1.4: When is required to flush the encoder and how to do it? Simply reading
at the end the delayed frames like this: avcodec_encode_audio2(c, &pkt,
NULL, &got_output) means flushing the encoder?
2. bit more advance questions:
2.1.1 I have a loop parsing an input buffer, shall I call inside the loop
av_init_packet(&pkt) each time before avcodec_encode_audio2, or it's enough
only once before the loop?
I am asking because in the avcodec.c example it shows as if I need to
run it each time.
http://www.ffmpeg.org/doxygen/trunk/avcodec_8c-example.html#a43 (note: I am
focusing on performance of the code as well).
2.1.2 I am writing the output in a buffer, by pointing the pkt.data to
the output buffer: pkt.data=&audio_outbuf[audio_outbuf_actual_encoded_size];
The question shall I run av_free_packet(&pkt); inside the loop (as in
question 2.1.1) ? (of course this time after the avcodec_encode_audio2).
If so, will av_free_packet-> temper with my output buffer to which
pkt.data is pointing to?
Currently, when I do: av_free(audio_outbuf) at the end of the
application, and all seems fine, but just to make sure.
2.2 amrwb (like other many codecs), encoding of each frame depends on the
adjacent frames, is there something special I should to while feeding the
encoder, in order to keep the "context"?
AVFrame *frame = av_frame_alloc();
avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,(const
uint8_t*)audio_bufin_single_frame, audio_bufin_single_frame_buffer_size, 0);
3. Is there some magic way of allocating the right size of the required
output buffer? As of now, I am using an non-VBR codec, I encode the 1st
frame, I get the size of the 1st output package and I multiply with the
number of frames I have to encode in total. Is there some better option?
4. I have set av_log_set_level(AV_LOG_VERBOSE), but none of the methods I
call from the ffmpeg libraries show any debug output. Should I expect them
to say anything or not?
Initial version (as of now only for audio encoding) can be found at:
https://github.com/ReSearchITEng/ffmpeg_matlab (utils.h-> main_encode2 is
"the" function) (Open Source, suggestions and contributions are welcome).
I removed the matlab sections, so I can compile and debug it in a IDE.
Thanks in advance for replies,
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