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#include <asm-generic/errno-base.h>
#include <assert.h>
#include <libavcodec/avcodec.h>
#include <libavutil/error.h>
#include <libavutil/frame.h>
#include <pthread.h>
#include <stdint.h>
#include <stdio.h>
#include "muxer.h"
#include "encoder.h"
#include "decoder.h"
#include "resampler.h"
#include "stream.h"
#define NANOSEC (uint64_t)1000000000
struct AudioMuxer {
struct Stream stream;
struct Encoder encoder;
struct Decoder decoder;
struct Resampler resampler;
AVFrame *frame_in;
AVFrame *frame_out;
AVPacket *packet;
/* The amount of output samples that went out of this muxer. Starts at 0 and
* keep on increasing.
*/
unsigned int nb_samples_out;
};
int64_t audiomuxer_flush(AudioMuxer *muxer);
uint64_t audiomuxer_get_runtime(const AudioMuxer *muxer)
{
return (muxer->nb_samples_out * NANOSEC) / muxer->resampler.out_sample_rate;
}
int audiomuxer_init(AudioMuxer **pmuxer, AVFormatContext *s, const struct AudioMuxerOpt *opts)
{
int err;
AudioMuxer *muxer;
AVFrame *frame_out;
AVFrame *frame_in;
AVPacket *pkt;
*pmuxer = NULL;
if ((muxer = malloc(sizeof(AudioMuxer))) == NULL)
return AVERROR(ENOMEM);
muxer->decoder.avctx = NULL;
muxer->encoder.avctx = NULL;
muxer->resampler.swr = NULL;
err = stream_init(
&muxer->stream,
s,
opts->codec_id,
opts->sample_rate,
opts->sample_format,
&opts->ch_layout,
opts->opts_muxer);
if (err) goto error;
err = encoder_init_for_stream(&muxer->encoder, muxer->stream.av_stream, opts->opts_encoder);
if (err) goto error;
err = resampler_init_for_encoder(&muxer->resampler, &muxer->encoder);
if (err) goto error;
if ((frame_out = av_frame_alloc()) == NULL)
goto enomem;
if ((frame_in = av_frame_alloc()) == NULL)
goto enomem;
if ((pkt = av_packet_alloc()) == NULL)
goto enomem;
frame_out->format = muxer->encoder.avctx->sample_fmt;
frame_out->nb_samples = muxer->encoder.avctx->frame_size;
frame_out->ch_layout = muxer->encoder.avctx->ch_layout;
frame_out->sample_rate = muxer->encoder.avctx->sample_rate;
if ((err = av_frame_get_buffer(frame_out, 0)))
goto error;
muxer->packet = pkt;
muxer->frame_out = frame_out;
muxer->frame_in = frame_in;
*pmuxer = muxer;
return 0;
enomem:
err = AVERROR(ENOMEM);
error:
audiomuxer_free(muxer);
return err;
}
int audiomuxer_conf (AudioMuxer *muxer, const AVStream *input)
{
int err, src_rate, dst_rate;
const AVCodec *codec;
SwrContext *resampler;
src_rate = input->codecpar->sample_rate;
err = decoder_init_for_stream(&muxer->decoder, input, NULL); // TODO: Pass options // TODO: Pass options
if (err) {
fprintf(stderr, "Failed to configure decoder: %s\n", av_err2str(err));
return err;
}
err = resampler_conf_for_decoder(&muxer->resampler, &muxer->decoder);
if (err) {
fprintf(stderr, "Failed to configure resampler: %s\n", av_err2str(err));
return err;
}
muxer->frame_in->format = muxer->decoder.avctx->sample_fmt;
muxer->frame_in->nb_samples = muxer->decoder.avctx->frame_size;
muxer->frame_in->ch_layout = muxer->decoder.avctx->ch_layout;
muxer->frame_in->sample_rate = muxer->decoder.avctx->sample_rate;
if ((err = av_frame_get_buffer(muxer->frame_in, 0)))
return err;
return 0;
}
int audiomuxer_send_packet (AudioMuxer *muxer, const AVPacket *pkt_in)
{
int err, duration;
AVFrame *frame_in;
frame_in = muxer->frame_in;
if ((err = decoder_send(&muxer->decoder, pkt_in)) < 0)
return err;
/* Since a packet might contains several frames, we call decoder_convert
* repeatedly until we're out of data.
*/
while ((err = decoder_convert(&muxer->decoder, frame_in)) != AVERROR(EAGAIN)) {
/* Errors other than EAGAIN should be repported */
if (err) {
fprintf(stderr, "Failed to decode frame: %s\n", av_err2str(err));
return err;
}
/* Immediately send frame to resampler and postpone silent injection */
if ((err = resampler_send_frame(&muxer->resampler, frame_in)) < 0) {
fprintf(stderr, "Failed to send data to resampler: %s\n", av_err2str(err));
return err;
}
/* Flush frame if possible */
if ((err = audiomuxer_flush(muxer)) < 0)
return err;
}
return 0;
}
int audiomuxer_send_silence (AudioMuxer *muxer, unsigned int microsec)
{
int err;
if ((err = resampler_send_empty(&muxer->resampler, microsec)))
return err;
return audiomuxer_flush(muxer);
}
int64_t audiomuxer_flush(AudioMuxer *muxer)
{
int err;
AVPacket *pkt_out;
AVFrame *frame_out;
frame_out = muxer->frame_out;
pkt_out = muxer->packet;
if ((err = resampler_convert(&muxer->resampler, frame_out))) {
if (err == AVERROR(EAGAIN)) return 0;
fprintf(stderr, "Failed to resample: %s\n", av_err2str(err));
return err;
}
if ((err = encoder_send(&muxer->encoder, frame_out))) {
fprintf(stderr, "Failed to send frame to encoder: %s\n", av_err2str(err));
return err;
}
if ((err = encoder_convert(&muxer->encoder, pkt_out))) {
if (err == AVERROR(EAGAIN)) return 0;
fprintf(stderr, "Failed to encode packet: %s\n", av_err2str(err));
return err;
}
/* pkt_out->duration is the "Duration of this packet in AVStream->time_base
* units, 0 if unknown." In practice, the timebase appears to be 1 /
* sample_rate, so, in other words, pkt_out->duration is the amount of
* samples stored in that packet.
*
* TODO: This behaviour is however undocumented, and asserts should be put
* in place to catch potential regressions.
*
* NOTE: duration is usually 1152 with a default MP3 audio stream.
*/
assert(pkt_out->duration > 0);
muxer->nb_samples_out += pkt_out->duration;
/* Finally send packet to stream. */
if ((err = stream_send(&muxer->stream, pkt_out)) < 0) {
fprintf(stderr, "Failed stream packet: %s\n", av_err2str(err));
return err;
}
return 0;
}
void audiomuxer_free (AudioMuxer *muxer)
{
if (muxer == NULL) return;
/* Flush decoder */
avcodec_send_packet(muxer->decoder.avctx, NULL);
encoder_free(&muxer->encoder);
decoder_free(&muxer->decoder);
resampler_free(&muxer->resampler);
free(muxer);
}
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