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authorLuca Abeni <lucabe72@email.it>2007-11-16 07:59:41 +0000
committerLuca Abeni <lucabe72@email.it>2007-11-16 07:59:41 +0000
commit7ed19d7fbfc0dda44d401388db06b245f07d206f (patch)
tree1d834bdda316cba0989eb7481d53829731748f4a /libavformat/rtp.c
parentb5298028b050d8c20d741c5219c1b0d2304c9391 (diff)
Remove the "AVRtpPayloadTypes[i].pt == i" assumption from RTP and RTSP
code (this is needed for supporting MPEG2 video in the RTP muxer) Originally committed as revision 11046 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/rtp.c')
-rw-r--r--libavformat/rtp.c51
1 files changed, 40 insertions, 11 deletions
diff --git a/libavformat/rtp.c b/libavformat/rtp.c
index a5820d717d..83e7dfb4ed 100644
--- a/libavformat/rtp.c
+++ b/libavformat/rtp.c
@@ -198,15 +198,20 @@ void av_register_rtp_dynamic_payload_handlers(void)
int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
{
- if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) {
- codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type;
- codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id;
- if (AVRtpPayloadTypes[payload_type].audio_channels > 0)
- codec->channels = AVRtpPayloadTypes[payload_type].audio_channels;
- if (AVRtpPayloadTypes[payload_type].clock_rate > 0)
- codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate;
- return 0;
- }
+ int i = 0;
+
+ for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
+ if (AVRtpPayloadTypes[i].pt == payload_type) {
+ if (AVRtpPayloadTypes[i].codec_id != CODEC_ID_NONE) {
+ codec->codec_type = AVRtpPayloadTypes[i].codec_type;
+ codec->codec_id = AVRtpPayloadTypes[i].codec_id;
+ if (AVRtpPayloadTypes[i].audio_channels > 0)
+ codec->channels = AVRtpPayloadTypes[i].audio_channels;
+ if (AVRtpPayloadTypes[i].clock_rate > 0)
+ codec->sample_rate = AVRtpPayloadTypes[i].clock_rate;
+ return 0;
+ }
+ }
return -1;
}
@@ -225,6 +230,30 @@ int rtp_get_payload_type(AVCodecContext *codec)
return payload_type;
}
+const char *ff_rtp_enc_name(int payload_type)
+{
+ int i;
+
+ for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
+ if (AVRtpPayloadTypes[i].pt == payload_type) {
+ return AVRtpPayloadTypes[i].enc_name;
+ }
+
+ return "";
+}
+
+enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type)
+{
+ int i;
+
+ for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
+ if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec_type == AVRtpPayloadTypes[i].codec_type)){
+ return AVRtpPayloadTypes[i].codec_id;
+ }
+
+ return CODEC_ID_NONE;
+}
+
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
{
if (buf[1] != 200)
@@ -448,7 +477,7 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
s->st = st;
s->rtp_payload_data = rtp_payload_data;
rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
- if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
+ if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
s->ts = mpegts_parse_open(s->ic);
if (s->ts == NULL) {
av_free(s);
@@ -712,7 +741,7 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
void rtp_parse_close(RTPDemuxContext *s)
{
// TODO: fold this into the protocol specific data fields.
- if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) {
+ if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
mpegts_parse_close(s->ts);
}
av_free(s);