rtpenc: Don't set max_frames_per_packet based on the packet frame size or frame rate

Instead check the timestamps while muxing, to avoid buffering a
too long timestamp range into one single packet.

This makes the AMR and AAC packetization slightly less efficient,
since we set a possibly unnecessarily high max_frames_per_packet.
(These packetizers end up doing a memmove of the TOC bytes if
sending a packet before max_frames_per_packet is achieved, and
we end up setting max_frames_per_packet to a value that should
be high enough for most uses.)

All packetizers that use max_frames_per_packet now set it either
to a default value, or to a value calculated based on other
parameters, so none of them rely on the previous default setting.

For iLBC, copy one frame at a time, to allow checking the timestamp
range for each of them - basically doing potentially multiple
loops to simplify the code instead of trying to calculate the
number of frames to buffer while honoring s1->max_delay.

This is in preparation for reducing the coupling between libavformat
and libavcodec, by not having the muxers use the encoder field
frame_size (which may not be available during e.g. stream copy).

Signed-off-by: Martin Storsjö <martin@martin.st>
This commit is contained in:
Martin Storsjö
2015-02-26 00:00:39 +02:00
parent bde2bba45c
commit 4f6cd883f0
4 changed files with 29 additions and 49 deletions

View File

@@ -32,6 +32,7 @@
void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size)
{
RTPMuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
int max_pkt_size, xdt, frag;
uint8_t *q;
@@ -77,8 +78,10 @@ void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size)
assert(s->num_frames <= s->max_frames_per_packet);
if (s->num_frames > 0 &&
(remaining < 0 ||
s->num_frames == s->max_frames_per_packet)) {
// send previous packets now; no room for new data
s->num_frames == s->max_frames_per_packet ||
av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
s1->max_delay, AV_TIME_BASE_Q) >= 0)) {
// send previous packets now; no room for new data, or too much delay
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
s->num_frames = 0;
}