Update to 2.0.0 tree from current Fremantle build

[opencv] / 3rdparty / include / ffmpeg_ / rtp.h

diff --git a/3rdparty/include/ffmpeg_/rtp.h b/3rdparty/include/ffmpeg_/rtp.h
new file mode 100644 (file)
index 0000000..fec7630
--- /dev/null
+++ b/3rdparty/include/ffmpeg_/rtp.h
@@ -0,0 +1,127 @@
+/*
+ * RTP definitions
+ * Copyright (c) 2002 Fabrice Bellard.
+ *
+ * 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 RTP_H
+#define RTP_H
+
+#define RTP_MIN_PACKET_LENGTH 12
+#define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
+
+int rtp_init(void);
+int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
+
+/** return < 0 if unknown payload type */
+int 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);
+
+extern AVOutputFormat rtp_muxer;
+extern AVInputFormat rtp_demuxer;
+
+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);
+
+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 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;
+
+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
+
+extern AVRtpPayloadType_t AVRtpPayloadTypes[];
+#endif /* RTP_H */