| /* |
| * 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" |
| #include "url.h" |
| #include "srtp.h" |
| |
| typedef struct PayloadContext PayloadContext; |
| typedef struct RTPDynamicProtocolHandler RTPDynamicProtocolHandler; |
| |
| #define RTP_MIN_PACKET_LENGTH 12 |
| #define RTP_MAX_PACKET_LENGTH 8192 |
| |
| #define RTP_REORDER_QUEUE_DEFAULT_SIZE 500 |
| |
| #define RTP_NOTS_VALUE ((uint32_t)-1) |
| |
| typedef struct RTPDemuxContext RTPDemuxContext; |
| RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, |
| int payload_type, int queue_size); |
| void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, |
| const RTPDynamicProtocolHandler *handler); |
| void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, |
| const char *params); |
| int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
| uint8_t **buf, int len); |
| void ff_rtp_parse_close(RTPDemuxContext *s); |
| int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s); |
| void ff_rtp_reset_packet_queue(RTPDemuxContext *s); |
| |
| /** |
| * Send a dummy packet on both port pairs to set up the connection |
| * state in potential NAT routers, so that we're able to receive |
| * packets. |
| * |
| * Note, this only works if the NAT router doesn't remap ports. This |
| * isn't a standardized procedure, but it works in many cases in practice. |
| * |
| * The same routine is used with RDT too, even if RDT doesn't use normal |
| * RTP packets otherwise. |
| */ |
| void ff_rtp_send_punch_packets(URLContext* rtp_handle); |
| |
| /** |
| * some rtp servers assume client is dead if they don't hear from them... |
| * so we send a Receiver Report to the provided URLContext or AVIOContext |
| * (we don't have access to the rtcp handle from here) |
| */ |
| int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, |
| AVIOContext *avio, int count); |
| int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, |
| AVIOContext *avio); |
| |
| // these statistics are used for rtcp receiver reports... |
| typedef struct RTPStatistics { |
| 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 |
| uint32_t received; ///< packets received |
| uint32_t expected_prior; ///< packets expected in last interval |
| uint32_t received_prior; ///< packets received in last interval |
| uint32_t transit; ///< relative transit time for previous packet |
| uint32_t jitter; ///< estimated jitter. |
| } RTPStatistics; |
| |
| #define RTP_FLAG_KEY 0x1 ///< RTP packet contains a keyframe |
| #define RTP_FLAG_MARKER 0x2 ///< RTP marker bit was set for this packet |
| /** |
| * 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 to the RTP timestamp of the input data, can be |
| * updated by the function if returning older, buffered data |
| * @param buf pointer to raw RTP packet data |
| * @param len length of buf |
| * @param seq RTP sequence number of the packet |
| * @param flags flags from the RTP packet header (RTP_FLAG_*) |
| */ |
| typedef int (*DynamicPayloadPacketHandlerProc)(AVFormatContext *ctx, |
| PayloadContext *s, |
| AVStream *st, AVPacket *pkt, |
| uint32_t *timestamp, |
| const uint8_t * buf, |
| int len, uint16_t seq, int flags); |
| |
| struct RTPDynamicProtocolHandler { |
| const char *enc_name; |
| enum AVMediaType codec_type; |
| enum AVCodecID codec_id; |
| enum AVStreamParseType need_parsing; |
| int static_payload_id; /* 0 means no payload id is set. 0 is a valid |
| * payload ID (PCMU), too, but that format doesn't |
| * require any custom depacketization code. */ |
| int priv_data_size; |
| |
| /** Initialize dynamic protocol handler, called after the full rtpmap line is parsed, may be null */ |
| int (*init)(AVFormatContext *s, int st_index, PayloadContext *priv_data); |
| /** Parse the a= line from the sdp field */ |
| int (*parse_sdp_a_line)(AVFormatContext *s, int st_index, |
| PayloadContext *priv_data, const char *line); |
| /** Free any data needed by the rtp parsing for this dynamic data. |
| * Don't free the protocol_data pointer itself, that is freed by the |
| * caller. This is called even if the init method failed. */ |
| void (*close)(PayloadContext *protocol_data); |
| /** Parse handler for this dynamic packet */ |
| DynamicPayloadPacketHandlerProc parse_packet; |
| int (*need_keyframe)(PayloadContext *context); |
| |
| struct RTPDynamicProtocolHandler *next; |
| }; |
| |
| typedef struct RTPPacket { |
| uint16_t seq; |
| uint8_t *buf; |
| int len; |
| int64_t recvtime; |
| struct RTPPacket *next; |
| } RTPPacket; |
| |
| struct RTPDemuxContext { |
| AVFormatContext *ic; |
| AVStream *st; |
| int payload_type; |
| uint32_t ssrc; |
| uint16_t seq; |
| uint32_t timestamp; |
| uint32_t base_timestamp; |
| int64_t unwrapped_timestamp; |
| int64_t range_start_offset; |
| int max_payload_size; |
| /* used to send back RTCP RR */ |
| char hostname[256]; |
| |
| int srtp_enabled; |
| struct SRTPContext srtp; |
| |
| /** Statistics for this stream (used by RTCP receiver reports) */ |
| RTPStatistics statistics; |
| |
| /** Fields for packet reordering @{ */ |
| int prev_ret; ///< The return value of the actual parsing of the previous packet |
| RTPPacket* queue; ///< A sorted queue of buffered packets not yet returned |
| int queue_len; ///< The number of packets in queue |
| int queue_size; ///< The size of queue, or 0 if reordering is disabled |
| /*@}*/ |
| |
| /* rtcp sender statistics receive */ |
| uint64_t last_rtcp_ntp_time; |
| int64_t last_rtcp_reception_time; |
| uint64_t first_rtcp_ntp_time; |
| uint32_t last_rtcp_timestamp; |
| int64_t rtcp_ts_offset; |
| |
| /* rtcp sender statistics */ |
| unsigned int packet_count; |
| unsigned int octet_count; |
| unsigned int last_octet_count; |
| int64_t last_feedback_time; |
| |
| /* dynamic payload stuff */ |
| const RTPDynamicProtocolHandler *handler; |
| PayloadContext *dynamic_protocol_context; |
| }; |
| |
| /** |
| * Iterate over all registered rtp dynamic protocol handlers. |
| * |
| * @param opaque a pointer where libavformat will store the iteration state. Must |
| * point to NULL to start the iteration. |
| * |
| * @return the next registered rtp dynamic protocol handler or NULL when the iteration is |
| * finished |
| */ |
| const RTPDynamicProtocolHandler *ff_rtp_handler_iterate(void **opaque); |
| /** |
| * Find a registered rtp dynamic protocol handler with the specified name. |
| * |
| * @param name name of the requested rtp dynamic protocol handler |
| * @return A rtp dynamic protocol handler if one was found, NULL otherwise. |
| */ |
| const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name, |
| enum AVMediaType codec_type); |
| /** |
| * Find a registered rtp dynamic protocol handler with a matching codec ID. |
| * |
| * @param id AVCodecID of the requested rtp dynamic protocol handler. |
| * @return A rtp dynamic protocol handler if one was found, NULL otherwise. |
| */ |
| const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id, |
| enum AVMediaType codec_type); |
| |
| /* from rtsp.c, but used by rtp dynamic protocol handlers. */ |
| int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, |
| char *value, int value_size); |
| |
| int ff_parse_fmtp(AVFormatContext *s, |
| AVStream *stream, PayloadContext *data, const char *p, |
| int (*parse_fmtp)(AVFormatContext *s, |
| AVStream *stream, |
| PayloadContext *data, |
| const char *attr, const char *value)); |
| |
| /** |
| * Close the dynamic buffer and make a packet from it. |
| */ |
| int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx); |
| |
| #endif /* AVFORMAT_RTPDEC_H */ |