* RTP definitions
* Copyright (c) 2002 Fabrice Bellard.
*
- * This library is free software; you can redistribute it and/or
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#ifndef RTP_H
-#define RTP_H
+#ifndef AVFORMAT_RTP_H
+#define AVFORMAT_RTP_H
+
+#include "libavcodec/avcodec.h"
+#include "avformat.h"
#define RTP_MIN_PACKET_LENGTH 12
#define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
-int rtp_init(void);
int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
+
+/** return < 0 if unknown payload type */
int rtp_get_payload_type(AVCodecContext *codec);
typedef struct RTPDemuxContext RTPDemuxContext;
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);
+RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, 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);
-extern AVOutputFormat rtp_mux;
-extern AVInputFormat rtp_demux;
-
int rtp_get_local_port(URLContext *h);
int rtp_set_remote_url(URLContext *h, const char *uri);
void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd);
-extern URLProtocol rtp_protocol;
+/**
+ * some rtp servers assume client is dead if they don't hear from them...
+ * so we send a Receiver Report to the provided ByteIO context
+ * (we don't have access to the rtcp handle from here)
+ */
+int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
#define RTP_PT_PRIVATE 96
#define RTP_VERSION 2
-#define RTP_MAX_SDES 256 /* maximum text length for SDES */
+#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*/
+/** Structure listing useful vars to parse RTP packet payload*/
typedef struct rtp_payload_data_s
{
int sizelength;
int objecttype;
char *mode;
- /* mpeg 4 AU headers */
+ /** mpeg 4 AU headers */
struct AUHeaders {
int size;
int index;
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;
-
#if 0
typedef enum {
RTCP_SR = 200,
} rtcp_sdes_type_t;
#endif
-extern AVRtpPayloadType_t AVRtpPayloadTypes[];
-extern AVRtpDynamicPayloadType_t AVRtpDynamicPayloadTypes[];
-
-#endif /* RTP_H */
+#endif /* AVFORMAT_RTP_H */