RTP/RTSP and MPEG4-AAC audio

  - preliminary support for mpeg4-aac rtp payload (no interleaving support)
  - use udp transport as default (makes more sense with rtp, doesn't it ?)
  - some code factorization, so adding support for new rtp payload will be easier
  (I hope ;-)
patch by (Romain DEGEZ: romain degez, smartjog com)

Originally committed as revision 4306 to svn://svn.ffmpeg.org/ffmpeg/trunk

libavformat/rtp.h

@@ -19,22 +19,6 @@
 #ifndef RTP_H
 #define RTP_H
 
-enum RTPPayloadType {
-    RTP_PT_ULAW = 0,
-    RTP_PT_GSM = 3,
-    RTP_PT_G723 = 4,
-    RTP_PT_ALAW = 8,
-    RTP_PT_S16BE_STEREO = 10,
-    RTP_PT_S16BE_MONO = 11,
-    RTP_PT_MPEGAUDIO = 14,
-    RTP_PT_JPEG = 26,
-    RTP_PT_H261 = 31,
-    RTP_PT_MPEGVIDEO = 32,
-    RTP_PT_MPEG2TS = 33,
-    RTP_PT_H263 = 34, /* old H263 encapsulation */
-    RTP_PT_PRIVATE = 96,
-};
-
 #define RTP_MIN_PACKET_LENGTH 12
 #define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
 
@@ -43,8 +27,8 @@ int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
 int rtp_get_payload_type(AVCodecContext *codec);
 
 typedef struct RTPDemuxContext RTPDemuxContext;
-
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type);
+typedef struct rtp_payload_data_s rtp_payload_data_s;
+RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_s *rtp_payload_data);
 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, 
                      const uint8_t *buf, int len);
 void rtp_parse_close(RTPDemuxContext *s);
@@ -58,4 +42,82 @@ void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd);
 
 extern URLProtocol rtp_protocol;
 
+#define RTP_PT_PRIVATE 96
+#define RTP_VERSION 2
+#define RTP_MAX_SDES 256   /* maximum text length for SDES */
+
+/* RTCP paquets use 0.5 % of the bandwidth */
+#define RTCP_TX_RATIO_NUM 5
+#define RTCP_TX_RATIO_DEN 1000
+
+/* Structure listing usefull vars to parse RTP packet payload*/
+typedef struct rtp_payload_data_s
+{
+    int sizelength;
+    int indexlength;
+    int indexdeltalength;
+    int profile_level_id;
+    int streamtype;
+    int objecttype;
+    char *mode;
+
+    /* mpeg 4 AU headers */
+    struct AUHeaders {
+        int size;
+        int index;
+        int cts_flag;
+        int cts;
+        int dts_flag;
+        int dts;
+        int rap_flag;
+        int streamstate;
+    } *au_headers;
+    int nb_au_headers;
+    int au_headers_length_bytes;
+    int cur_au_index;
+} rtp_payload_data_t;
+
+typedef struct AVRtpPayloadType_s
+{
+    int pt;
+    const char enc_name[50]; /* XXX: why 50 ? */
+    enum CodecType codec_type;
+    enum CodecID codec_id;
+    int clock_rate;
+    int audio_channels;
+} AVRtpPayloadType_t;
+
+typedef struct AVRtpDynamicPayloadType_s /* payload type >= 96 */
+{
+    const char enc_name[50]; /* XXX: still why 50 ? ;-) */
+    enum CodecType codec_type;
+    enum CodecID codec_id;
+} AVRtpDynamicPayloadType_t;
+
+typedef enum {
+  RTCP_SR   = 200,
+  RTCP_RR   = 201,
+  RTCP_SDES = 202,
+  RTCP_BYE  = 203,
+  RTCP_APP  = 204
+} rtcp_type_t;
+
+typedef enum {
+  RTCP_SDES_END    =  0,
+  RTCP_SDES_CNAME  =  1,
+  RTCP_SDES_NAME   =  2,
+  RTCP_SDES_EMAIL  =  3,
+  RTCP_SDES_PHONE  =  4,
+  RTCP_SDES_LOC    =  5,
+  RTCP_SDES_TOOL   =  6,
+  RTCP_SDES_NOTE   =  7,
+  RTCP_SDES_PRIV   =  8,
+  RTCP_SDES_IMG    =  9,
+  RTCP_SDES_DOOR   = 10,
+  RTCP_SDES_SOURCE = 11
+} rtcp_sdes_type_t;
+
+extern AVRtpPayloadType_t AVRtpPayloadTypes[];
+extern AVRtpDynamicPayloadType_t AVRtpDynamicPayloadTypes[];
+
 #endif /* RTP_H */

libavformat/rtsp.h

@@ -35,6 +35,10 @@ enum RTSPProtocol {
 #define RTSP_DEFAULT_PORT   554
 #define RTSP_MAX_TRANSPORTS 8
 #define RTSP_TCP_MAX_PACKET_SIZE 1472
+#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 2
+#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
+#define RTSP_RTP_PORT_MIN 5000
+#define RTSP_RTP_PORT_MAX 10000
 
 typedef struct RTSPTransportField {
     int interleaved_min, interleaved_max;  /* interleave ids, if TCP transport */