if (ret < 0) {
av_free(st);
return -EIO;
- } else {
- /* take real parameters */
- st->codec.codec_type = CODEC_TYPE_AUDIO;
- st->codec.codec_id = s->codec_id;
- st->codec.sample_rate = s->sample_rate;
- st->codec.channels = s->channels;
- return 0;
}
+
+ /* take real parameters */
+ st->codec.codec_type = CODEC_TYPE_AUDIO;
+ st->codec.codec_id = s->codec_id;
+ st->codec.sample_rate = s->sample_rate;
+ st->codec.channels = s->channels;
+
+ av_set_pts_info(s1, 48, 1, 1000000); /* 48 bits pts in us */
+ return 0;
}
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
AudioData *s = s1->priv_data;
- int ret;
-
+ int ret, bdelay;
+ int64_t cur_time;
+ struct audio_buf_info abufi;
+
if (av_new_packet(pkt, s->frame_size) < 0)
return -EIO;
for(;;) {
}
}
pkt->size = ret;
+
+ /* compute pts of the start of the packet */
+ cur_time = av_gettime();
+ bdelay = ret;
+ if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
+ bdelay += abufi.bytes;
+ }
+ /* substract time represented by the number of bytes in the audio fifo */
+ cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
+
+ /* convert to wanted units */
+ pkt->pts = cur_time & ((1LL << 48) - 1);
+
if (s->flip_left && s->channels == 2) {
int i;
short *p = (short *) pkt->data;
st->codec.height = height;
st->codec.frame_rate = frame_rate;
+ av_set_pts_info(s1, 48, 1, 1000000); /* 48 bits pts in us */
+
return 0;
fail:
if (video_fd >= 0)
VideoData *s = s1->priv_data;
INT64 curtime, delay;
struct timespec ts;
- int first;
INT64 per_frame = (INT64_C(1000000) * FRAME_RATE_BASE) / s->frame_rate;
int dropped = 0;
s->time_frame += per_frame;
/* wait based on the frame rate */
- for(first = 1;; first = 0) {
+ for(;;) {
curtime = av_gettime();
delay = s->time_frame - curtime;
if (delay <= 0) {
if (av_new_packet(pkt, s->frame_size) < 0)
return -EIO;
+ pkt->pts = curtime & ((1LL << 48) - 1);
+
if (dropped)
pkt->flags |= PKT_FLAG_DROPPED_FRAME;