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authorStefano Sabatini <stefano.sabatini-lala@poste.it>2010-03-30 23:30:55 +0000
committerStefano Sabatini <stefano.sabatini-lala@poste.it>2010-03-30 23:30:55 +0000
commit72415b2adb2c25f95ceede49001bb97ed9247dbb (patch)
tree99247889aa61bb7621c966466ea4b84c100207d3 /libavformat/rtpenc.c
parentca6e7708b42e7d33ba3053bcd447d52a077bca25 (diff)
downloadffmpeg-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.c8
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;