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author | Luca Abeni <lucabe72@email.it> | 2009-02-06 10:35:52 +0000 |
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committer | Luca Abeni <lucabe72@email.it> | 2009-02-06 10:35:52 +0000 |
commit | 302879cb36fe59e7341690d91e0e656b02ba07a1 (patch) | |
tree | a8e4e0c1984dca69cdba723bcc6374d0951350f9 /libavformat/rtpdec.h | |
parent | 1a45a9f4c06bbbaa322ba744e658491df44f2c2a (diff) | |
download | ffmpeg-302879cb36fe59e7341690d91e0e656b02ba07a1.tar.gz |
Split rtp.h in rtp.h, rtpdec.h, and rtpenc.h
Originally committed as revision 17016 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/rtpdec.h')
-rw-r--r-- | libavformat/rtpdec.h | 187 |
1 files changed, 187 insertions, 0 deletions
diff --git a/libavformat/rtpdec.h b/libavformat/rtpdec.h new file mode 100644 index 0000000000..1eeb0ba968 --- /dev/null +++ b/libavformat/rtpdec.h @@ -0,0 +1,187 @@ +/* + * RTP demuxer definitions + * Copyright (c) 2002 Fabrice Bellard + * Copyright (c) 2006 Ryan Martell <rdm4@martellventures.com> + * + * 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 2.1 of the License, or (at your option) any later version. + * + * 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ +#ifndef AVFORMAT_RTPDEC_H +#define AVFORMAT_RTPDEC_H + +#include "libavcodec/avcodec.h" +#include "avformat.h" +#include "rtp.h" + +/** Structure listing useful vars to parse RTP packet payload*/ +typedef struct rtp_payload_data +{ + 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; +} RTPPayloadData; + +typedef struct PayloadContext PayloadContext; +typedef struct RTPDynamicProtocolHandler_s RTPDynamicProtocolHandler; + +#define RTP_MIN_PACKET_LENGTH 12 + +int rtp_get_codec_info(AVCodecContext *codec, int payload_type); + +typedef struct RTPDemuxContext RTPDemuxContext; +RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data); +void rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, + RTPDynamicProtocolHandler *handler); +int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, + const uint8_t *buf, int len); +void rtp_parse_close(RTPDemuxContext *s); + +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); + +/** + * 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); + +// these statistics are used for rtcp receiver reports... +typedef struct { + uint16_t max_seq; ///< highest sequence number seen + uint32_t cycles; ///< shifted count of sequence number cycles + uint32_t base_seq; ///< base sequence number + uint32_t bad_seq; ///< last bad sequence number + 1 + int probation; ///< sequence packets till source is valid + int received; ///< packets received + int expected_prior; ///< packets expected in last interval + int received_prior; ///< packets received in last interval + uint32_t transit; ///< relative transit time for previous packet + uint32_t jitter; ///< estimated jitter. +} RTPStatistics; + +/** + * Packet parsing for "private" payloads in the RTP specs. + * + * @param ctx RTSP demuxer context + * @param s stream context + * @param st stream that this packet belongs to + * @param pkt packet in which to write the parsed data + * @param timestamp pointer in which to write the timestamp of this RTP packet + * @param buf pointer to raw RTP packet data + * @param len length of buf + * @param flags flags from the RTP packet header (PKT_FLAG_*) + */ +typedef int (*DynamicPayloadPacketHandlerProc) (AVFormatContext *ctx, + PayloadContext *s, + AVStream *st, + AVPacket * pkt, + uint32_t *timestamp, + const uint8_t * buf, + int len, int flags); + +struct RTPDynamicProtocolHandler_s { + // fields from AVRtpDynamicPayloadType_s + const char enc_name[50]; /* XXX: still why 50 ? ;-) */ + enum CodecType codec_type; + enum CodecID codec_id; + + // may be null + int (*parse_sdp_a_line) (AVFormatContext *s, + int st_index, + PayloadContext *priv_data, + const char *line); ///< Parse the a= line from the sdp field + PayloadContext *(*open) (); ///< allocate any data needed by the rtp parsing for this dynamic data. + void (*close)(PayloadContext *protocol_data); ///< free any data needed by the rtp parsing for this dynamic data. + DynamicPayloadPacketHandlerProc parse_packet; ///< parse handler for this dynamic packet. + + struct RTPDynamicProtocolHandler_s *next; +}; + +// moved out of rtp.c, because the h264 decoder needs to know about this structure.. +struct RTPDemuxContext { + AVFormatContext *ic; + AVStream *st; + int payload_type; + uint32_t ssrc; + uint16_t seq; + uint32_t timestamp; + uint32_t base_timestamp; + uint32_t cur_timestamp; + int max_payload_size; + struct MpegTSContext *ts; /* only used for MP2T payloads */ + int read_buf_index; + int read_buf_size; + /* used to send back RTCP RR */ + URLContext *rtp_ctx; + char hostname[256]; + + RTPStatistics statistics; ///< Statistics for this stream (used by RTCP receiver reports) + + /* rtcp sender statistics receive */ + int64_t last_rtcp_ntp_time; // TODO: move into statistics + int64_t first_rtcp_ntp_time; // TODO: move into statistics + uint32_t last_rtcp_timestamp; // TODO: move into statistics + + /* rtcp sender statistics */ + unsigned int packet_count; // TODO: move into statistics (outgoing) + unsigned int octet_count; // TODO: move into statistics (outgoing) + unsigned int last_octet_count; // TODO: move into statistics (outgoing) + int first_packet; + /* buffer for output */ + uint8_t buf[RTP_MAX_PACKET_LENGTH]; + uint8_t *buf_ptr; + + /* special infos for au headers parsing */ + RTPPayloadData *rtp_payload_data; // TODO: Move into dynamic payload handlers + + /* dynamic payload stuff */ + DynamicPayloadPacketHandlerProc parse_packet; ///< This is also copied from the dynamic protocol handler structure + PayloadContext *dynamic_protocol_context; ///< This is a copy from the values setup from the sdp parsing, in rtsp.c don't free me. + int max_frames_per_packet; +}; + +extern RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler; +void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler); + +int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size); ///< from rtsp.c, but used by rtp dynamic protocol handlers. + +const char *ff_rtp_enc_name(int payload_type); +enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type); + +void av_register_rtp_dynamic_payload_handlers(void); + +#endif /* AVFORMAT_RTPDEC_H */ |