adding ffmpeg specific version

1 files changed, 219 insertions, 0 deletions

diff --git a/ffmpeg1/libavformat/rtpdec.h b/ffmpeg1/libavformat/rtpdec.h
new file mode 100644
index 0000000..e469c0f
--- /dev/null
+++ b/ffmpeg1/libavformat/rtpdec.h
@@ -0,0 +1,219 @@
+/*
+ * 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 1500
+
+#define RTP_REORDER_QUEUE_DEFAULT_SIZE 10
+
+#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,
+                                       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);
+int ff_rtp_get_local_rtp_port(URLContext *h);
+int ff_rtp_get_local_rtcp_port(URLContext *h);
+
+int ff_rtp_set_remote_url(URLContext *h, const char *uri);
+
+/**
+ * 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[50];
+    enum AVMediaType codec_type;
+    enum AVCodecID codec_id;
+    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. */
+
+    /** 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);
+    /** Allocate any data needed by the rtp parsing for this dynamic data. */
+    PayloadContext *(*alloc)(void);
+    /** Free any data needed by the rtp parsing for this dynamic data. */
+    void (*free)(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;
+    uint32_t cur_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 */
+    int64_t last_rtcp_ntp_time;
+    int64_t last_rtcp_reception_time;
+    int64_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;
+};
+
+void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler);
+RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
+                                                  enum AVMediaType codec_type);
+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(AVStream *stream, PayloadContext *data, const char *p,
+                  int (*parse_fmtp)(AVStream *stream,
+                                    PayloadContext *data,
+                                    char *attr, char *value));
+
+void av_register_rtp_dynamic_payload_handlers(void);
+
+/**
+ * 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 */