diff options
| author | Luca Abeni <lucabe72@email.it> | 2007-10-02 14:48:08 +0000 |
|---|---|---|
| committer | Luca Abeni <lucabe72@email.it> | 2007-10-02 14:48:08 +0000 |
| commit | e0d21bfe835eec87b8c0efe6e9973ae5f588bc07 (patch) | |
| tree | 1031f231b2673252dd031b79c9fc7045d3a3d254 | |
| parent | 0d0447eae09dca047412a03d64ac2bfc11a80d35 (diff) | |
Allow to set the maximum number of frames per RTP packet (and add support for
this in the AAC packetizer)
Originally committed as revision 10647 to svn://svn.ffmpeg.org/ffmpeg/trunk
| -rw-r--r-- | libavformat/rtp.c | 15 | ||||
| -rw-r--r-- | libavformat/rtp_aac.c | 2 | ||||
| -rw-r--r-- | libavformat/rtp_internal.h | 1 |
3 files changed, 17 insertions, 1 deletions
diff --git a/libavformat/rtp.c b/libavformat/rtp.c index 02d3e7be29..14e9189635 100644 --- a/libavformat/rtp.c +++ b/libavformat/rtp.c @@ -747,6 +747,21 @@ static int rtp_write_header(AVFormatContext *s1) return AVERROR(EIO); s->max_payload_size = max_packet_size - 12; + s->max_frames_per_packet = 0; + if (s1->max_delay) { + if (st->codec->codec_type == CODEC_TYPE_AUDIO) { + if (st->codec->frame_size == 0) { + av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); + } else { + s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN); + } + } + if (st->codec->codec_type == CODEC_TYPE_VIDEO) { + /* FIXME: We should round down here... */ + s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base); + } + } + av_set_pts_info(st, 32, 1, 90000); switch(st->codec->codec_id) { case CODEC_ID_MP2: diff --git a/libavformat/rtp_aac.c b/libavformat/rtp_aac.c index 04dc3cfdda..4cd8e129d2 100644 --- a/libavformat/rtp_aac.c +++ b/libavformat/rtp_aac.c @@ -22,7 +22,7 @@ #include "rtp_aac.h" #include "rtp_internal.h" -#define MAX_FRAMES_PER_PACKET 5 +#define MAX_FRAMES_PER_PACKET (s->max_frames_per_packet ? s->max_frames_per_packet : 5) #define MAX_AU_HEADERS_SIZE (2 + 2 * MAX_FRAMES_PER_PACKET) void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size) diff --git a/libavformat/rtp_internal.h b/libavformat/rtp_internal.h index 96477620b3..4bfa4bba62 100644 --- a/libavformat/rtp_internal.h +++ b/libavformat/rtp_internal.h @@ -105,6 +105,7 @@ struct RTPDemuxContext { /* dynamic payload stuff */ DynamicPayloadPacketHandlerProc parse_packet; ///< This is also copied from the dynamic protocol handler structure void *dynamic_protocol_context; ///< This is a copy from the values setup from the sdp parsing, in rtsp.c don't free me. + int max_frames_per_packet; }; extern RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler; |