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avconv: move streamcopy-only code out of decoding loop.

This commit is contained in:
Anton Khirnov 2011-11-21 14:05:38 +01:00
parent 51aeb69452
commit 2a651b719c
1 changed files with 30 additions and 25 deletions
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55
avconv.c
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@ -1605,6 +1605,16 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
av_free_packet(&opkt); av_free_packet(&opkt);
} }
static void rate_emu_sleep(InputStream *ist)
{
if (input_files[ist->file_index].rate_emu) {
int64_t pts = av_rescale(ist->pts, 1000000, AV_TIME_BASE);
int64_t now = av_gettime() - ist->start;
if (pts > now)
usleep(pts - now);
}
}
/* pkt = NULL means EOF (needed to flush decoder buffers) */ /* pkt = NULL means EOF (needed to flush decoder buffers) */
static int output_packet(InputStream *ist, int ist_index, static int output_packet(InputStream *ist, int ist_index,
OutputStream *ost_table, int nb_ostreams, OutputStream *ost_table, int nb_ostreams,
@ -1645,7 +1655,7 @@ static int output_packet(InputStream *ist, int ist_index,
pkt_pts = av_rescale_q(pkt->pts, ist->st->time_base, AV_TIME_BASE_Q); pkt_pts = av_rescale_q(pkt->pts, ist->st->time_base, AV_TIME_BASE_Q);
//while we have more to decode or while the decoder did output something on EOF //while we have more to decode or while the decoder did output something on EOF
while (avpkt.size > 0 || (!pkt && got_output)) { while (ist->decoding_needed && (avpkt.size > 0 || (!pkt && got_output))) {
uint8_t *decoded_data_buf; uint8_t *decoded_data_buf;
int decoded_data_size; int decoded_data_size;
AVFrame *decoded_frame, *filtered_frame; AVFrame *decoded_frame, *filtered_frame;
@ -1662,7 +1672,6 @@ static int output_packet(InputStream *ist, int ist_index,
decoded_data_buf = NULL; /* fail safe */ decoded_data_buf = NULL; /* fail safe */
decoded_data_size= 0; decoded_data_size= 0;
subtitle_to_free = NULL; subtitle_to_free = NULL;
if (ist->decoding_needed) {
switch(ist->st->codec->codec_type) { switch(ist->st->codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:{ case AVMEDIA_TYPE_AUDIO:{
if(pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) { if(pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) {
@ -1733,23 +1742,6 @@ static int output_packet(InputStream *ist, int ist_index,
default: default:
return -1; return -1;
} }
} else {
switch(ist->st->codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
ist->next_pts += ((int64_t)AV_TIME_BASE * ist->st->codec->frame_size) /
ist->st->codec->sample_rate;
break;
case AVMEDIA_TYPE_VIDEO:
if (ist->st->codec->time_base.num != 0) {
int ticks= ist->st->parser ? ist->st->parser->repeat_pict+1 : ist->st->codec->ticks_per_frame;
ist->next_pts += ((int64_t)AV_TIME_BASE *
ist->st->codec->time_base.num * ticks) /
ist->st->codec->time_base.den;
}
break;
}
avpkt.size = 0;
}
// preprocess audio (volume) // preprocess audio (volume)
if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
@ -1810,12 +1802,8 @@ static int output_packet(InputStream *ist, int ist_index,
} }
/* frame rate emulation */ /* frame rate emulation */
if (input_files[ist->file_index].rate_emu) { rate_emu_sleep(ist);
int64_t pts = av_rescale(ist->pts, 1000000, AV_TIME_BASE);
int64_t now = av_gettime() - ist->start;
if (pts > now)
usleep(pts - now);
}
/* if output time reached then transcode raw format, /* if output time reached then transcode raw format,
encode packets and output them */ encode packets and output them */
for (i = 0; i < nb_ostreams; i++) { for (i = 0; i < nb_ostreams; i++) {
@ -1902,6 +1890,23 @@ fail:
discard_packet: discard_packet:
/* handle stream copy */ /* handle stream copy */
if (!ist->decoding_needed) {
rate_emu_sleep(ist);
switch (ist->st->codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
ist->next_pts += ((int64_t)AV_TIME_BASE * ist->st->codec->frame_size) /
ist->st->codec->sample_rate;
break;
case AVMEDIA_TYPE_VIDEO:
if (ist->st->codec->time_base.num != 0) {
int ticks = ist->st->parser ? ist->st->parser->repeat_pict+1 : ist->st->codec->ticks_per_frame;
ist->next_pts += ((int64_t)AV_TIME_BASE *
ist->st->codec->time_base.num * ticks) /
ist->st->codec->time_base.den;
}
break;
}
}
for (i = 0; pkt && i < nb_ostreams; i++) { for (i = 0; pkt && i < nb_ostreams; i++) {
ost = &ost_table[i]; ost = &ost_table[i];