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author | Martin Storsjö <martin@martin.st> | 2015-02-26 00:00:39 +0200 |
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committer | Martin Storsjö <martin@martin.st> | 2015-02-28 22:54:31 +0200 |
commit | 4f6cd883f06f7893a2b60a41e7a4f8ae633dac2f (patch) | |
tree | 0a140108c19744a4399aa5d5244fd63d8b5e13bf /libavformat/rtpenc_aac.c | |
parent | bde2bba45c2f2df27a8534028bda09a6e7f835e2 (diff) | |
download | ffmpeg-4f6cd883f06f7893a2b60a41e7a4f8ae633dac2f.tar.gz |
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>
Diffstat (limited to 'libavformat/rtpenc_aac.c')
-rw-r--r-- | libavformat/rtpenc_aac.c | 5 |
1 files changed, 4 insertions, 1 deletions
diff --git a/libavformat/rtpenc_aac.c b/libavformat/rtpenc_aac.c index 7805ab9034..d0b4ca0964 100644 --- a/libavformat/rtpenc_aac.c +++ b/libavformat/rtpenc_aac.c @@ -27,6 +27,7 @@ void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size) { RTPMuxContext *s = s1->priv_data; + AVStream *st = s1->streams[0]; const int max_au_headers_size = 2 + 2 * s->max_frames_per_packet; int len, max_packet_size = s->max_payload_size - max_au_headers_size; uint8_t *p; @@ -41,7 +42,9 @@ void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size) len = (s->buf_ptr - s->buf); if (s->num_frames && (s->num_frames == s->max_frames_per_packet || - (len + size) > s->max_payload_size)) { + (len + size) > s->max_payload_size || + av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base, + s1->max_delay, AV_TIME_BASE_Q) >= 0)) { int au_size = s->num_frames * 2; p = s->buf + max_au_headers_size - au_size - 2; |