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ad_lavc: deal with arbitrary decoder delay 

Normally, audio decoder don't have a decoder delay, so the code was
fine. But FFmpeg supports multithreaded decoding for some audio codecs,
which introduces such a delay.

The delay means that we won't get decoded audio for the first few
packets, and that we need to do something to get the trailing audio
still buffered in the decoder when reaching EOF.

Two changes are needed to deal with the delay:
- If EOF is reached, pass a "flush" packet to the decoder to return the
  buffered audio. Such a flush packet is automatically setup when
  calling mp_set_av_packet() with a NULL packet.
- Use the PTS returned by the decoder, instead of the packet's. This is
  important to get correct timestamps for decoded audio. Ignoring this
  would result into offsetting the audio playback time by the decoder
  delay. Note that we can still use the timestamp of the first packet
  to get the timestamp for the start of the audio.
This commit is contained in:
wm4 2013-12-04 20:57:52 +01:00
parent 8a84da8102
commit 9c2858f37f
1 changed files with 24 additions and 16 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

40
audio/decode/ad_lavc.c
View File

@ -321,33 +321,35 @@ static int decode_new_packet(struct dec_audio *da)
struct demux_packet *mpkt = priv->packet;
if (!mpkt)
mpkt = demux_read_packet(da->header);
if (!mpkt)
return -1; // error or EOF
priv->packet = talloc_steal(priv, mpkt);
int in_len = mpkt->len;
int in_len = mpkt ? mpkt->len : 0;
AVPacket pkt;
mp_set_av_packet(&pkt, mpkt, NULL);
if (mpkt->pts != MP_NOPTS_VALUE) {
// If we don't have a PTS yet, use the first packet PTS we can get.
if (da->pts == MP_NOPTS_VALUE && mpkt && mpkt->pts != MP_NOPTS_VALUE) {
da->pts = mpkt->pts;
da->pts_offset = 0;
}
int got_frame = 0;
int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
// At least "shorten" decodes sub-frames, instead of the whole packet.
// At least "mpc8" can return 0 and wants the packet again next time.
if (ret >= 0) {
ret = FFMIN(ret, mpkt->len); // sanity check against decoder overreads
mpkt->buffer += ret;
mpkt->len -= ret;
mpkt->pts = MP_NOPTS_VALUE; // don't reset PTS next time
}
if (mpkt->len == 0 || ret < 0) {
talloc_free(mpkt);
priv->packet = NULL;
if (mpkt) {
// At least "shorten" decodes sub-frames, instead of the whole packet.
// At least "mpc8" can return 0 and wants the packet again next time.
if (ret >= 0) {
ret = FFMIN(ret, mpkt->len); // sanity check against decoder overreads
mpkt->buffer += ret;
mpkt->len -= ret;
mpkt->pts = MP_NOPTS_VALUE; // don't reset PTS next time
}
if (mpkt->len == 0 || ret < 0) {
talloc_free(mpkt);
priv->packet = NULL;
}
}
// LATM may need many packets to find mux info
if (ret == AVERROR(EAGAIN))
@ -357,7 +359,7 @@ static int decode_new_packet(struct dec_audio *da)
return -1;
}
if (!got_frame)
return 0;
return mpkt ? 0 : -1; // -1: eof
if (setup_format(da) < 0)
return -1;
@ -367,6 +369,12 @@ static int decode_new_packet(struct dec_audio *da)
for (int n = 0; n < priv->frame.num_planes; n++)
priv->frame.planes[n] = priv->avframe->data[n];
double out_pts = mp_pts_from_av(priv->avframe->pkt_pts, NULL);
if (out_pts != MP_NOPTS_VALUE) {
da->pts = out_pts;
da->pts_offset = 0;
}
mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d samples\n", in_len,
priv->frame.samples);
return 0;