ffmpeg ts 封装 aac 中推断出的两个结论

static struct muxer* init_muxer(enum CodecID id, const char* name, const char* ext, struct audio_param* in){
int bufid = buf_length_register(1024);
if(bufid == -1){
return NULL;
}

struct muxer* muxerp = (struct muxer *) malloc(sizeof(struct muxer));
if(muxerp == NULL){
return NULL;
}

INIT_LIST_HEAD(&muxerp->head);
memset(muxerp->line, 0, sizeof(muxerp->line));
muxerp->cur_buf = NULL;
muxerp->bufid = bufid;
muxerp->avctx = NULL;

AVIOContext* ioctx = NULL;
AVStream *st = NULL;
AVFormatContext* fmtctx = NULL;

avformat_alloc_output_context2(&fmtctx, NULL, name, ext);
if(fmtctx == NULL){
goto LABEL;
}

st = av_new_stream(fmtctx, 0);
if(st == NULL){
goto LABEL;
}
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = id;
st->codec->frame_size = in->spf; // this should be the pcm frame size
st->codec->sample_rate = in->rate; // this should be the pcm sample rate
st->codec->bit_rate = in->bps; // this should be output stream's bitrate
st->codec->channels = in->chans;

ioctx = avio_alloc_context(&muxerp->line[0], sizeof(muxerp->line), 0,
(void *) muxerp, NULL, write_packet, NULL);
if(ioctx == NULL){
goto LABEL;
}
fmtctx->pb = ioctx;

if(0 != avformat_write_header(fmtctx, NULL)){
goto LABEL;
}

muxerp->avctx = fmtctx;
return muxerp;
LABEL:
if(fmtctx){
avformat_free_context(fmtctx);
if(ioctx){
av_free(ioctx);
}
}
free(muxerp);
return NULL;
}

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st->codec->frame_size = in->spf; // this should be the pcm frame size
st->codec->sample_rate = in->rate; // this should be the pcm sample rate

由以下代码推导出来：

if (pcr_st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
if (!pcr_st->codec->frame_size) {
av_log(s, AV_LOG_WARNING, "frame size not set\n");
service->pcr_packet_period =
pcr_st->codec->sample_rate/(10*512);
} else {
service->pcr_packet_period =
pcr_st->codec->sample_rate/(10*pcr_st->codec->frame_size);
}

st->codec->bit_rate = in->bps; // this should be output stream's bitrate
由以下代码推导出来：

static int get_audio_frame_size(AVCodecContext *enc, int size)
{
int frame_size;

if(enc->codec_id == CODEC_ID_VORBIS)
return -1;

if (enc->frame_size <= 1) {
int bits_per_sample = av_get_bits_per_sample(enc->codec_id);

if (bits_per_sample) {
if (enc->channels == 0)
return -1;
frame_size = (size << 3) / (bits_per_sample * enc->channels);
} else {
/* used for example by ADPCM codecs */
if (enc->bit_rate == 0)
return -1;
frame_size = ((int64_t)size * 8 * enc->sample_rate) / enc->bit_rate;
}
} else {
frame_size = enc->frame_size;
}
return frame_size;