int audio_decode_frame(VideoState *is, double *pts_ptr) {
/* For example with wma audio package size can be
like 100 000 bytes */
long len1, data_size = 0;
AVPacket *pkt = &is->audio_pkt;
double pts;
int n = 0;
#ifdef __RESAMPLER__
long resample_size = 0;
#endif
for(;;) {
while(is->audio_pkt_size > 0) {
int got_frame = 0;
len1 = avcodec_decode_audio4(is->audio_st->codec, &is->audio_frame, &got_frame, pkt);
if(len1 < 0) {
/* if error, skip frame */
is->audio_pkt_size = 0;
break;
}
if(got_frame) {
data_size =
av_samples_get_buffer_size
(
NULL,
is->audio_st->codec->channels,
is->audio_frame.nb_samples,
is->audio_st->codec->sample_fmt,
1
);
#ifdef __RESAMPLER__
if(is->audio_need_resample == 1) {
resample_size = audio_tutorial_resample(is, &is->audio_frame);
if( resample_size > 0 ) {
memcpy(is->audio_buf, is->pResampledOut, resample_size);
memset(is->pResampledOut, 0x00, resample_size);
}
} else {
#endif
memcpy(is->audio_buf, is->audio_frame.data[0], data_size);
#ifdef __RESAMPLER__
}
#endif
}
is->audio_pkt_data += len1;
is->audio_pkt_size -= len1;
if(data_size <= 0) {
/* No data yet, get more frames */
continue;
}
pts = is->audio_clock;
*pts_ptr = pts;
n = 2 * is->audio_st->codec->channels;
#ifdef __RESAMPLER__
/* If you just return original data_size you will suffer
for clicks because you don't have that much data in
queue incoming so return resampled size. */
if(is->audio_need_resample == 1) {
is->audio_clock += (double)resample_size /
(double)(n * is->audio_st->codec->sample_rate);
return resample_size;
} else {
#endif
/* We have data, return it and come back for more later */
is->audio_clock += (double)data_size /
(double)(n * is->audio_st->codec->sample_rate);
return data_size;
#ifdef __RESAMPLER__
}
#endif
}
if(pkt->data) {
av_free_packet(pkt);
}
if(is->quit) {
return -1;
}
/* next packet */
if(packet_queue_get(&is->audioq, pkt, 1) < 0) {
return -1;
}
if(pkt->data == flush_pkt.data) {
avcodec_flush_buffers(is->audio_st->codec);
continue;
}
is->audio_pkt_data = pkt->data;
is->audio_pkt_size = pkt->size;
/* if update, update the audio clock w/pts */
if(pkt->pts != AV_NOPTS_VALUE) {
is->audio_clock = av_q2d(is->audio_st->time_base) * pkt->pts;
}
}
}
int audio_decode_frame(VideoState *is, double *pts_ptr) {
int len1, data_size = 0, n;
AVPacket *pkt = &is->audio_pkt;
double pts;
for(;;) {
while(is->audio_pkt_size > 0) {
int got_frame = 0;
int resample_size = 0;
len1 = avcodec_decode_audio4(is->audio_st->codec, &is->audio_frame, &got_frame,
pkt);
if(len1 < 0) {
/* if error, skip frame */
is->audio_pkt_size = 0;
break;
}
if (got_frame) {
data_size =
av_samples_get_buffer_size
(
NULL,
is->audio_st->codec->channels,
is->audio_frame.nb_samples,
is->audio_st->codec->sample_fmt,
1
);
#ifdef __RESAMPLER__
if(is->audio_need_resample == 1) {
resample_size = audio_tutorial_resample(is, &is->audio_frame);
if( resample_size > 0 ) {
memcpy(is->audio_buf, is->pResampledOut, resample_size);
memset(is->pResampledOut, 0x00, resample_size);
}
} else {
#endif
memcpy(is->audio_buf, is->audio_frame.data[0], data_size);
#ifdef __RESAMPLER__
}
#endif
}
is->audio_pkt_data += len1;
is->audio_pkt_size -= len1;
if(data_size <= 0) {
/* No data yet, get more frames */
continue;
}
pts = is->audio_clock;
*pts_ptr = pts;
n = 2 * is->audio_st->codec->channels;
is->audio_clock += (double)data_size /
(double)(n * is->audio_st->codec->sample_rate);
/* We have data, return it and come back for more later */
return data_size;
}
if(pkt->data)
av_free_packet(pkt);
if(is->quit) {
return -1;
}
/* next packet */
if(packet_queue_get(&is->audioq, pkt, 1) < 0) {
return -1;
}
if(pkt->data == flush_pkt.data) {
avcodec_flush_buffers(is->audio_st->codec);
continue;
}
is->audio_pkt_data = pkt->data;
is->audio_pkt_size = pkt->size;
/* if update, update the audio clock w/pts */
if(pkt->pts != AV_NOPTS_VALUE) {
is->audio_clock = av_q2d(is->audio_st->time_base)*pkt->pts;
}
}
}
