Mercurial > libavformat.hg
diff rtp.h @ 774:820863425158 libavformat
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)
author | michael |
---|---|
date | Thu, 26 May 2005 07:47:51 +0000 |
parents | 6091b76cfc2a |
children | 021c8bf015c2 |
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--- a/rtp.h Sun May 22 23:58:58 2005 +0000 +++ b/rtp.h Thu May 26 07:47:51 2005 +0000 @@ -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_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 @@ 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 */