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author | Stefano Sabatini <stefano.sabatini-lala@poste.it> | 2010-03-30 23:30:55 +0000 |
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committer | Stefano Sabatini <stefano.sabatini-lala@poste.it> | 2010-03-30 23:30:55 +0000 |
commit | 72415b2adb2c25f95ceede49001bb97ed9247dbb (patch) | |
tree | 99247889aa61bb7621c966466ea4b84c100207d3 /libavformat/rtpenc.c | |
parent | ca6e7708b42e7d33ba3053bcd447d52a077bca25 (diff) | |
download | ffmpeg-72415b2adb2c25f95ceede49001bb97ed9247dbb.tar.gz |
Define AVMediaType enum, and use it instead of enum CodecType, which
is deprecated and will be dropped at the next major bump.
Originally committed as revision 22735 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/rtpenc.c')
-rw-r--r-- | libavformat/rtpenc.c | 8 |
1 files changed, 4 insertions, 4 deletions
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index c5441b8312..5df101eb04 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -78,7 +78,7 @@ static int rtp_write_header(AVFormatContext *s1) s->payload_type = ff_rtp_get_payload_type(st->codec); if (s->payload_type < 0) - s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == CODEC_TYPE_AUDIO); + s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO); s->base_timestamp = ff_random_get_seed(); s->timestamp = s->base_timestamp; @@ -102,14 +102,14 @@ static int rtp_write_header(AVFormatContext *s1) s->max_frames_per_packet = 0; if (s1->max_delay) { - if (st->codec->codec_type == CODEC_TYPE_AUDIO) { + if (st->codec->codec_type == AVMEDIA_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) { + if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) { /* FIXME: We should round down here... */ s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base); } @@ -151,7 +151,7 @@ static int rtp_write_header(AVFormatContext *s1) case CODEC_ID_AAC: s->num_frames = 0; default: - if (st->codec->codec_type == CODEC_TYPE_AUDIO) { + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { av_set_pts_info(st, 32, 1, st->codec->sample_rate); } s->buf_ptr = s->buf; |