/*
* RTP definitions
- * Copyright (c) 2002 Fabrice Bellard.
+ * Copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#ifndef FFMPEG_RTP_H
-#define FFMPEG_RTP_H
+#ifndef AVFORMAT_RTP_H
+#define AVFORMAT_RTP_H
-#include "avcodec.h"
-#include "avformat.h"
+#include "libavcodec/avcodec.h"
-#define RTP_MIN_PACKET_LENGTH 12
-#define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
-
-int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
-
-/** return < 0 if unknown payload type */
-int rtp_get_payload_type(AVCodecContext *codec);
+/**
+ * Return the payload type for a given codec.
+ *
+ * @param codec The context of the codec
+ * @return In case of unknown payload type or dynamic payload type, a
+ * negative value is returned; otherwise, the payload type (the 'PT' field
+ * in the RTP header) is returned.
+ */
+int ff_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, 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);
+/**
+ * Initialize a codec context based on the payload type.
+ *
+ * Fill the codec_type and codec_id fields of a codec context with
+ * information depending on the payload type; for audio codecs, the
+ * channels and sample_rate fields are also filled.
+ *
+ * @param codec The context of the codec
+ * @param payload_type The payload type (the 'PT' field in the RTP header)
+ * @return In case of unknown payload type or dynamic payload type, a
+ * negative value is returned; otherwise, 0 is returned
+ */
+int ff_rtp_get_codec_info(AVCodecContext *codec, int payload_type);
-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);
+/**
+ * Return the encoding name (as defined in
+ * http://www.iana.org/assignments/rtp-parameters) for a given payload type.
+ *
+ * @param payload_type The payload type (the 'PT' field in the RTP header)
+ * @return In case of unknown payload type or dynamic payload type, a pointer
+ * to an empty string is returned; otherwise, a pointer to a string containing
+ * the encoding name is returned
+ */
+const char *ff_rtp_enc_name(int payload_type);
/**
- * 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)
+ * Return the codec id for the given encoding name and codec type.
+ *
+ * @param buf A pointer to the string containing the encoding name
+ * @param codec_type The codec type
+ * @return In case of unknown encoding name, CODEC_ID_NONE is returned;
+ * otherwise, the codec id is returned
*/
-int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
+enum CodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type);
#define RTP_PT_PRIVATE 96
#define RTP_VERSION 2
#define RTCP_TX_RATIO_NUM 5
#define RTCP_TX_RATIO_DEN 1000
-/** Structure listing useful 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;
-
-#if 0
-typedef enum {
- RTCP_SR = 200,
- RTCP_RR = 201,
- RTCP_SDES = 202,
- RTCP_BYE = 203,
- RTCP_APP = 204
-} rtcp_type_t;
+/* An arbitrary id value for RTP Xiph streams - only relevant to indicate
+ * the the configuration has changed within a stream (by changing the
+ * ident value sent).
+ */
+#define RTP_XIPH_IDENT 0xfecdba
-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;
-#endif
+/* RTCP packet types */
+enum RTCPType {
+ RTCP_SR = 200,
+ RTCP_RR, // 201
+ RTCP_SDES, // 202
+ RTCP_BYE, // 203
+ RTCP_APP // 204
+};
-extern AVRtpPayloadType_t AVRtpPayloadTypes[];
-#endif /* FFMPEG_RTP_H */
+#endif /* AVFORMAT_RTP_H */