[FFmpeg-cvslog] r10201 - in trunk/libavformat: Makefile rtp.c rtp_internal.h rtp_mpv.c rtp_mpv.h

lucabe subversion
Fri Aug 24 09:13:34 CEST 2007


Author: lucabe
Date: Fri Aug 24 09:13:34 2007
New Revision: 10201

Log:
Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c)


Added:
   trunk/libavformat/rtp_mpv.c
      - copied, changed from r10200, /trunk/libavformat/rtp.c
   trunk/libavformat/rtp_mpv.h
Modified:
   trunk/libavformat/Makefile
   trunk/libavformat/rtp.c
   trunk/libavformat/rtp_internal.h

Modified: trunk/libavformat/Makefile
==============================================================================
--- trunk/libavformat/Makefile	(original)
+++ trunk/libavformat/Makefile	Fri Aug 24 09:13:34 2007
@@ -119,7 +119,7 @@ OBJS-$(CONFIG_RM_DEMUXER)               
 OBJS-$(CONFIG_RM_MUXER)                  += rmenc.o
 OBJS-$(CONFIG_ROQ_DEMUXER)               += idroq.o
 OBJS-$(CONFIG_ROQ_MUXER)                 += raw.o
-OBJS-$(CONFIG_RTP_MUXER)                 += rtp.o rtp_h264.o
+OBJS-$(CONFIG_RTP_MUXER)                 += rtp.o rtp_h264.o rtp_mpv.o
 OBJS-$(CONFIG_RTSP_DEMUXER)              += rtsp.o
 OBJS-$(CONFIG_SDP_DEMUXER)               += rtsp.o
 OBJS-$(CONFIG_SEGAFILM_DEMUXER)          += segafilm.o

Modified: trunk/libavformat/rtp.c
==============================================================================
--- trunk/libavformat/rtp.c	(original)
+++ trunk/libavformat/rtp.c	Fri Aug 24 09:13:34 2007
@@ -27,6 +27,7 @@
 
 #include "rtp_internal.h"
 #include "rtp_h264.h"
+#include "rtp_mpv.h"
 
 //#define DEBUG
 
@@ -788,7 +789,7 @@ static void rtcp_send_sr(AVFormatContext
 
 /* send an rtp packet. sequence number is incremented, but the caller
    must update the timestamp itself */
-static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
+void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
 {
     RTPDemuxContext *s = s1->priv_data;
 
@@ -836,7 +837,7 @@ static void rtp_send_samples(AVFormatCon
         n = (s->buf_ptr - s->buf);
         /* if buffer full, then send it */
         if (n >= max_packet_size) {
-            rtp_send_data(s1, s->buf, n, 0);
+            ff_rtp_send_data(s1, s->buf, n, 0);
             s->buf_ptr = s->buf;
             /* update timestamp */
             s->timestamp += n / sample_size;
@@ -859,7 +860,7 @@ static void rtp_send_mpegaudio(AVFormatC
     len = (s->buf_ptr - s->buf);
     if ((len + size) > max_packet_size) {
         if (len > 4) {
-            rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
+            ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
             s->buf_ptr = s->buf + 4;
             /* 90 KHz time stamp */
             s->timestamp = s->base_timestamp +
@@ -881,7 +882,7 @@ static void rtp_send_mpegaudio(AVFormatC
             s->buf[2] = count >> 8;
             s->buf[3] = count;
             memcpy(s->buf + 4, buf1, len);
-            rtp_send_data(s1, s->buf, len + 4, 0);
+            ff_rtp_send_data(s1, s->buf, len + 4, 0);
             size -= len;
             buf1 += len;
             count += len;
@@ -900,55 +901,6 @@ static void rtp_send_mpegaudio(AVFormatC
     s->cur_timestamp += st->codec->frame_size;
 }
 
-/* NOTE: a single frame must be passed with sequence header if
-   needed. XXX: use slices. */
-static void rtp_send_mpegvideo(AVFormatContext *s1,
-                               const uint8_t *buf1, int size)
-{
-    RTPDemuxContext *s = s1->priv_data;
-    AVStream *st = s1->streams[0];
-    int len, h, max_packet_size;
-    uint8_t *q;
-
-    max_packet_size = s->max_payload_size;
-
-    while (size > 0) {
-        /* XXX: more correct headers */
-        h = 0;
-        if (st->codec->sub_id == 2)
-            h |= 1 << 26; /* mpeg 2 indicator */
-        q = s->buf;
-        *q++ = h >> 24;
-        *q++ = h >> 16;
-        *q++ = h >> 8;
-        *q++ = h;
-
-        if (st->codec->sub_id == 2) {
-            h = 0;
-            *q++ = h >> 24;
-            *q++ = h >> 16;
-            *q++ = h >> 8;
-            *q++ = h;
-        }
-
-        len = max_packet_size - (q - s->buf);
-        if (len > size)
-            len = size;
-
-        memcpy(q, buf1, len);
-        q += len;
-
-        /* 90 KHz time stamp */
-        s->timestamp = s->base_timestamp +
-            av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
-        rtp_send_data(s1, s->buf, q - s->buf, (len == size));
-
-        buf1 += len;
-        size -= len;
-    }
-    s->cur_timestamp++;
-}
-
 static void rtp_send_raw(AVFormatContext *s1,
                          const uint8_t *buf1, int size)
 {
@@ -966,7 +918,7 @@ static void rtp_send_raw(AVFormatContext
         /* 90 KHz time stamp */
         s->timestamp = s->base_timestamp +
             av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
-        rtp_send_data(s1, buf1, len, (len == size));
+        ff_rtp_send_data(s1, buf1, len, (len == size));
 
         buf1 += len;
         size -= len;
@@ -992,7 +944,7 @@ static void rtp_send_mpegts_raw(AVFormat
 
         out_len = s->buf_ptr - s->buf;
         if (out_len >= s->max_payload_size) {
-            rtp_send_data(s1, s->buf, out_len, 0);
+            ff_rtp_send_data(s1, s->buf, out_len, 0);
             s->buf_ptr = s->buf;
         }
     }
@@ -1042,7 +994,7 @@ static int rtp_write_packet(AVFormatCont
         rtp_send_mpegaudio(s1, buf1, size);
         break;
     case CODEC_ID_MPEG1VIDEO:
-        rtp_send_mpegvideo(s1, buf1, size);
+        ff_rtp_send_mpegvideo(s1, buf1, size);
         break;
     case CODEC_ID_MPEG2TS:
         rtp_send_mpegts_raw(s1, buf1, size);

Modified: trunk/libavformat/rtp_internal.h
==============================================================================
--- trunk/libavformat/rtp_internal.h	(original)
+++ trunk/libavformat/rtp_internal.h	Fri Aug 24 09:13:34 2007
@@ -110,5 +110,7 @@ struct RTPDemuxContext {
 extern RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler;
 
 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.
+
+void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m);
 #endif /* RTP_INTERNAL_H */
 

Copied: trunk/libavformat/rtp_mpv.c (from r10200, /trunk/libavformat/rtp.c)
==============================================================================
--- /trunk/libavformat/rtp.c	(original)
+++ trunk/libavformat/rtp_mpv.c	Fri Aug 24 09:13:34 2007
@@ -1,5 +1,5 @@
 /*
- * RTP input/output format
+ * RTP packetization for MPEG video
  * Copyright (c) 2002 Fabrice Bellard.
  *
  * This file is part of FFmpeg.
@@ -19,891 +19,11 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 #include "avformat.h"
-#include "mpegts.h"
-#include "bitstream.h"
-
-#include <unistd.h>
-#include "network.h"
-
 #include "rtp_internal.h"
-#include "rtp_h264.h"
-
-//#define DEBUG
-
-
-/* TODO: - add RTCP statistics reporting (should be optional).
-
-         - add support for h263/mpeg4 packetized output : IDEA: send a
-         buffer to 'rtp_write_packet' contains all the packets for ONE
-         frame. Each packet should have a four byte header containing
-         the length in big endian format (same trick as
-         'url_open_dyn_packet_buf')
-*/
-
-/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
-AVRtpPayloadType_t AVRtpPayloadTypes[]=
-{
-  {0, "PCMU",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_MULAW, 8000, 1},
-  {1, "Reserved",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {2, "Reserved",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {3, "GSM",         CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
-  {4, "G723",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
-  {5, "DVI4",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
-  {6, "DVI4",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 16000, 1},
-  {7, "LPC",         CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
-  {8, "PCMA",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_ALAW, 8000, 1},
-  {9, "G722",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
-  {10, "L16",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_S16BE, 44100, 2},
-  {11, "L16",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_S16BE, 44100, 1},
-  {12, "QCELP",      CODEC_TYPE_AUDIO,   CODEC_ID_QCELP, 8000, 1},
-  {13, "CN",         CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
-  {14, "MPA",        CODEC_TYPE_AUDIO,   CODEC_ID_MP2, 90000, -1},
-  {15, "G728",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
-  {16, "DVI4",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 11025, 1},
-  {17, "DVI4",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 22050, 1},
-  {18, "G729",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
-  {19, "reserved",   CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
-  {20, "unassigned", CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
-  {21, "unassigned", CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
-  {22, "unassigned", CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
-  {23, "unassigned", CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
-  {24, "unassigned", CODEC_TYPE_VIDEO,   CODEC_ID_NONE, -1, -1},
-  {25, "CelB",       CODEC_TYPE_VIDEO,   CODEC_ID_NONE, 90000, -1},
-  {26, "JPEG",       CODEC_TYPE_VIDEO,   CODEC_ID_MJPEG, 90000, -1},
-  {27, "unassigned", CODEC_TYPE_VIDEO,   CODEC_ID_NONE, -1, -1},
-  {28, "nv",         CODEC_TYPE_VIDEO,   CODEC_ID_NONE, 90000, -1},
-  {29, "unassigned", CODEC_TYPE_VIDEO,   CODEC_ID_NONE, -1, -1},
-  {30, "unassigned", CODEC_TYPE_VIDEO,   CODEC_ID_NONE, -1, -1},
-  {31, "H261",       CODEC_TYPE_VIDEO,   CODEC_ID_H261, 90000, -1},
-  {32, "MPV",        CODEC_TYPE_VIDEO,   CODEC_ID_MPEG1VIDEO, 90000, -1},
-  {33, "MP2T",       CODEC_TYPE_DATA,    CODEC_ID_MPEG2TS, 90000, -1},
-  {34, "H263",       CODEC_TYPE_VIDEO,   CODEC_ID_H263, 90000, -1},
-  {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {96, "dynamic",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {97, "dynamic",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {98, "dynamic",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {99, "dynamic",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {100, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {101, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {102, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {103, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {104, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {105, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {106, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {107, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {108, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {109, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {110, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {111, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {112, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {113, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {114, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {115, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {116, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {117, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {118, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {119, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {120, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {121, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {122, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {123, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {124, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {125, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {126, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {127, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
-  {-1, "",           CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}
-};
-
-/* statistics functions */
-RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
-
-static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
-static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
-
-static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
-{
-    handler->next= RTPFirstDynamicPayloadHandler;
-    RTPFirstDynamicPayloadHandler= handler;
-}
-
-void av_register_rtp_dynamic_payload_handlers(void)
-{
-    register_dynamic_payload_handler(&mp4v_es_handler);
-    register_dynamic_payload_handler(&mpeg4_generic_handler);
-    register_dynamic_payload_handler(&ff_h264_dynamic_handler);
-}
-
-int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
-{
-    if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) {
-        codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type;
-        codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id;
-        if (AVRtpPayloadTypes[payload_type].audio_channels > 0)
-            codec->channels = AVRtpPayloadTypes[payload_type].audio_channels;
-        if (AVRtpPayloadTypes[payload_type].clock_rate > 0)
-            codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate;
-        return 0;
-    }
-    return -1;
-}
-
-int rtp_get_payload_type(AVCodecContext *codec)
-{
-    int i, payload_type;
-
-    /* compute the payload type */
-    for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
-        if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) {
-            if (codec->codec_id == CODEC_ID_PCM_S16BE)
-                if (codec->channels != AVRtpPayloadTypes[i].audio_channels)
-                    continue;
-            payload_type = AVRtpPayloadTypes[i].pt;
-        }
-    return payload_type;
-}
-
-static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
-{
-    if (buf[1] != 200)
-        return -1;
-    s->last_rtcp_ntp_time = AV_RB64(buf + 8);
-    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
-        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
-    s->last_rtcp_timestamp = AV_RB32(buf + 16);
-    return 0;
-}
-
-#define RTP_SEQ_MOD (1<<16)
-
-/**
-* called on parse open packet
-*/
-static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
-{
-    memset(s, 0, sizeof(RTPStatistics));
-    s->max_seq= base_sequence;
-    s->probation= 1;
-}
-
-/**
-* called whenever there is a large jump in sequence numbers, or when they get out of probation...
-*/
-static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
-{
-    s->max_seq= seq;
-    s->cycles= 0;
-    s->base_seq= seq -1;
-    s->bad_seq= RTP_SEQ_MOD + 1;
-    s->received= 0;
-    s->expected_prior= 0;
-    s->received_prior= 0;
-    s->jitter= 0;
-    s->transit= 0;
-}
-
-/**
-* returns 1 if we should handle this packet.
-*/
-static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
-{
-    uint16_t udelta= seq - s->max_seq;
-    const int MAX_DROPOUT= 3000;
-    const int MAX_MISORDER = 100;
-    const int MIN_SEQUENTIAL = 2;
-
-    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
-    if(s->probation)
-    {
-        if(seq==s->max_seq + 1) {
-            s->probation--;
-            s->max_seq= seq;
-            if(s->probation==0) {
-                rtp_init_sequence(s, seq);
-                s->received++;
-                return 1;
-            }
-        } else {
-            s->probation= MIN_SEQUENTIAL - 1;
-            s->max_seq = seq;
-        }
-    } else if (udelta < MAX_DROPOUT) {
-        // in order, with permissible gap
-        if(seq < s->max_seq) {
-            //sequence number wrapped; count antother 64k cycles
-            s->cycles += RTP_SEQ_MOD;
-        }
-        s->max_seq= seq;
-    } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
-        // sequence made a large jump...
-        if(seq==s->bad_seq) {
-            // two sequential packets-- assume that the other side restarted without telling us; just resync.
-            rtp_init_sequence(s, seq);
-        } else {
-            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
-            return 0;
-        }
-    } else {
-        // duplicate or reordered packet...
-    }
-    s->received++;
-    return 1;
-}
-
-#if 0
-/**
-* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
-* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
-* never change.  I left this in in case someone else can see a way. (rdm)
-*/
-static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
-{
-    uint32_t transit= arrival_timestamp - sent_timestamp;
-    int d;
-    s->transit= transit;
-    d= FFABS(transit - s->transit);
-    s->jitter += d - ((s->jitter + 8)>>4);
-}
-#endif
-
-int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
-{
-    ByteIOContext pb;
-    uint8_t *buf;
-    int len;
-    int rtcp_bytes;
-    RTPStatistics *stats= &s->statistics;
-    uint32_t lost;
-    uint32_t extended_max;
-    uint32_t expected_interval;
-    uint32_t received_interval;
-    uint32_t lost_interval;
-    uint32_t expected;
-    uint32_t fraction;
-    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
-
-    if (!s->rtp_ctx || (count < 1))
-        return -1;
-
-    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
-    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
-    s->octet_count += count;
-    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
-        RTCP_TX_RATIO_DEN;
-    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
-    if (rtcp_bytes < 28)
-        return -1;
-    s->last_octet_count = s->octet_count;
-
-    if (url_open_dyn_buf(&pb) < 0)
-        return -1;
-
-    // Receiver Report
-    put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
-    put_byte(&pb, 201);
-    put_be16(&pb, 7); /* length in words - 1 */
-    put_be32(&pb, s->ssrc); // our own SSRC
-    put_be32(&pb, s->ssrc); // XXX: should be the server's here!
-    // some placeholders we should really fill...
-    // RFC 1889/p64
-    extended_max= stats->cycles + stats->max_seq;
-    expected= extended_max - stats->base_seq + 1;
-    lost= expected - stats->received;
-    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
-    expected_interval= expected - stats->expected_prior;
-    stats->expected_prior= expected;
-    received_interval= stats->received - stats->received_prior;
-    stats->received_prior= stats->received;
-    lost_interval= expected_interval - received_interval;
-    if (expected_interval==0 || lost_interval<=0) fraction= 0;
-    else fraction = (lost_interval<<8)/expected_interval;
-
-    fraction= (fraction<<24) | lost;
-
-    put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
-    put_be32(&pb, extended_max); /* max sequence received */
-    put_be32(&pb, stats->jitter>>4); /* jitter */
-
-    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
-    {
-        put_be32(&pb, 0); /* last SR timestamp */
-        put_be32(&pb, 0); /* delay since last SR */
-    } else {
-        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
-        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
-
-        put_be32(&pb, middle_32_bits); /* last SR timestamp */
-        put_be32(&pb, delay_since_last); /* delay since last SR */
-    }
-
-    // CNAME
-    put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
-    put_byte(&pb, 202);
-    len = strlen(s->hostname);
-    put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */
-    put_be32(&pb, s->ssrc);
-    put_byte(&pb, 0x01);
-    put_byte(&pb, len);
-    put_buffer(&pb, s->hostname, len);
-    // padding
-    for (len = (6 + len) % 4; len % 4; len++) {
-        put_byte(&pb, 0);
-    }
-
-    put_flush_packet(&pb);
-    len = url_close_dyn_buf(&pb, &buf);
-    if ((len > 0) && buf) {
-        int result;
-#if defined(DEBUG)
-        printf("sending %d bytes of RR\n", len);
-#endif
-        result= url_write(s->rtp_ctx, buf, len);
-#if defined(DEBUG)
-        printf("result from url_write: %d\n", result);
-#endif
-        av_free(buf);
-    }
-    return 0;
-}
-
-/**
- * open a new RTP parse context for stream 'st'. 'st' can be NULL for
- * MPEG2TS streams to indicate that they should be demuxed inside the
- * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
- * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
- */
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
-{
-    RTPDemuxContext *s;
-
-    s = av_mallocz(sizeof(RTPDemuxContext));
-    if (!s)
-        return NULL;
-    s->payload_type = payload_type;
-    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
-    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
-    s->ic = s1;
-    s->st = st;
-    s->rtp_payload_data = rtp_payload_data;
-    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
-    if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
-        s->ts = mpegts_parse_open(s->ic);
-        if (s->ts == NULL) {
-            av_free(s);
-            return NULL;
-        }
-    } else {
-        switch(st->codec->codec_id) {
-        case CODEC_ID_MPEG1VIDEO:
-        case CODEC_ID_MPEG2VIDEO:
-        case CODEC_ID_MP2:
-        case CODEC_ID_MP3:
-        case CODEC_ID_MPEG4:
-        case CODEC_ID_H264:
-            st->need_parsing = AVSTREAM_PARSE_FULL;
-            break;
-        default:
-            break;
-        }
-    }
-    // needed to send back RTCP RR in RTSP sessions
-    s->rtp_ctx = rtpc;
-    gethostname(s->hostname, sizeof(s->hostname));
-    return s;
-}
-
-static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
-{
-    int au_headers_length, au_header_size, i;
-    GetBitContext getbitcontext;
-    rtp_payload_data_t *infos;
-
-    infos = s->rtp_payload_data;
-
-    if (infos == NULL)
-        return -1;
-
-    /* decode the first 2 bytes where are stored the AUHeader sections
-       length in bits */
-    au_headers_length = AV_RB16(buf);
-
-    if (au_headers_length > RTP_MAX_PACKET_LENGTH)
-      return -1;
-
-    infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
-
-    /* skip AU headers length section (2 bytes) */
-    buf += 2;
-
-    init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
-
-    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
-    au_header_size = infos->sizelength + infos->indexlength;
-    if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
-        return -1;
-
-    infos->nb_au_headers = au_headers_length / au_header_size;
-    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
-
-    /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
-       In my test, the FAAD decoder does not behave correctly when sending each AU one by one
-       but does when sending the whole as one big packet...  */
-    infos->au_headers[0].size = 0;
-    infos->au_headers[0].index = 0;
-    for (i = 0; i < infos->nb_au_headers; ++i) {
-        infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
-        infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
-    }
-
-    infos->nb_au_headers = 1;
-
-    return 0;
-}
-
-/**
- * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
- */
-static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
-{
-    switch(s->st->codec->codec_id) {
-        case CODEC_ID_MP2:
-        case CODEC_ID_MPEG1VIDEO:
-            if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
-                int64_t addend;
-
-                int delta_timestamp;
-                /* XXX: is it really necessary to unify the timestamp base ? */
-                /* compute pts from timestamp with received ntp_time */
-                delta_timestamp = timestamp - s->last_rtcp_timestamp;
-                /* convert to 90 kHz without overflow */
-                addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
-                addend = (addend * 5625) >> 14;
-                pkt->pts = addend + delta_timestamp;
-            }
-            break;
-        case CODEC_ID_AAC:
-        case CODEC_ID_H264:
-        case CODEC_ID_MPEG4:
-            pkt->pts = timestamp;
-            break;
-        default:
-            /* no timestamp info yet */
-            break;
-    }
-    pkt->stream_index = s->st->index;
-}
-
-/**
- * Parse an RTP or RTCP packet directly sent as a buffer.
- * @param s RTP parse context.
- * @param pkt returned packet
- * @param buf input buffer or NULL to read the next packets
- * @param len buffer len
- * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
- * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
- */
-int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
-                     const uint8_t *buf, int len)
-{
-    unsigned int ssrc, h;
-    int payload_type, seq, ret;
-    AVStream *st;
-    uint32_t timestamp;
-    int rv= 0;
-
-    if (!buf) {
-        /* return the next packets, if any */
-        if(s->st && s->parse_packet) {
-            timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
-            rv= s->parse_packet(s, pkt, &timestamp, NULL, 0);
-            finalize_packet(s, pkt, timestamp);
-            return rv;
-        } else {
-            // TODO: Move to a dynamic packet handler (like above)
-            if (s->read_buf_index >= s->read_buf_size)
-                return -1;
-            ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
-                                      s->read_buf_size - s->read_buf_index);
-            if (ret < 0)
-                return -1;
-            s->read_buf_index += ret;
-            if (s->read_buf_index < s->read_buf_size)
-                return 1;
-            else
-                return 0;
-        }
-    }
-
-    if (len < 12)
-        return -1;
-
-    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
-        return -1;
-    if (buf[1] >= 200 && buf[1] <= 204) {
-        rtcp_parse_packet(s, buf, len);
-        return -1;
-    }
-    payload_type = buf[1] & 0x7f;
-    seq  = AV_RB16(buf + 2);
-    timestamp = AV_RB32(buf + 4);
-    ssrc = AV_RB32(buf + 8);
-    /* store the ssrc in the RTPDemuxContext */
-    s->ssrc = ssrc;
-
-    /* NOTE: we can handle only one payload type */
-    if (s->payload_type != payload_type)
-        return -1;
-
-    st = s->st;
-    // only do something with this if all the rtp checks pass...
-    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
-    {
-        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
-               payload_type, seq, ((s->seq + 1) & 0xffff));
-        return -1;
-    }
-
-    s->seq = seq;
-    len -= 12;
-    buf += 12;
-
-    if (!st) {
-        /* specific MPEG2TS demux support */
-        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
-        if (ret < 0)
-            return -1;
-        if (ret < len) {
-            s->read_buf_size = len - ret;
-            memcpy(s->buf, buf + ret, s->read_buf_size);
-            s->read_buf_index = 0;
-            return 1;
-        }
-    } else {
-        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
-        switch(st->codec->codec_id) {
-        case CODEC_ID_MP2:
-            /* better than nothing: skip mpeg audio RTP header */
-            if (len <= 4)
-                return -1;
-            h = AV_RB32(buf);
-            len -= 4;
-            buf += 4;
-            av_new_packet(pkt, len);
-            memcpy(pkt->data, buf, len);
-            break;
-        case CODEC_ID_MPEG1VIDEO:
-            /* better than nothing: skip mpeg video RTP header */
-            if (len <= 4)
-                return -1;
-            h = AV_RB32(buf);
-            buf += 4;
-            len -= 4;
-            if (h & (1 << 26)) {
-                /* mpeg2 */
-                if (len <= 4)
-                    return -1;
-                buf += 4;
-                len -= 4;
-            }
-            av_new_packet(pkt, len);
-            memcpy(pkt->data, buf, len);
-            break;
-            // moved from below, verbatim.  this is because this section handles packets, and the lower switch handles
-            // timestamps.
-            // TODO: Put this into a dynamic packet handler...
-        case CODEC_ID_AAC:
-            if (rtp_parse_mp4_au(s, buf))
-                return -1;
-            {
-                rtp_payload_data_t *infos = s->rtp_payload_data;
-                if (infos == NULL)
-                    return -1;
-                buf += infos->au_headers_length_bytes + 2;
-                len -= infos->au_headers_length_bytes + 2;
-
-                /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
-                    one au_header */
-                av_new_packet(pkt, infos->au_headers[0].size);
-                memcpy(pkt->data, buf, infos->au_headers[0].size);
-                buf += infos->au_headers[0].size;
-                len -= infos->au_headers[0].size;
-            }
-            s->read_buf_size = len;
-            s->buf_ptr = buf;
-            rv= 0;
-            break;
-        default:
-            if(s->parse_packet) {
-                rv= s->parse_packet(s, pkt, &timestamp, buf, len);
-            } else {
-                av_new_packet(pkt, len);
-                memcpy(pkt->data, buf, len);
-            }
-            break;
-        }
-
-        // now perform timestamp things....
-        finalize_packet(s, pkt, timestamp);
-    }
-    return rv;
-}
-
-void rtp_parse_close(RTPDemuxContext *s)
-{
-    // TODO: fold this into the protocol specific data fields.
-    if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) {
-        mpegts_parse_close(s->ts);
-    }
-    av_free(s);
-}
-
-/* rtp output */
-
-static int rtp_write_header(AVFormatContext *s1)
-{
-    RTPDemuxContext *s = s1->priv_data;
-    int payload_type, max_packet_size, n;
-    AVStream *st;
-
-    if (s1->nb_streams != 1)
-        return -1;
-    st = s1->streams[0];
-
-    payload_type = rtp_get_payload_type(st->codec);
-    if (payload_type < 0)
-        payload_type = RTP_PT_PRIVATE; /* private payload type */
-    s->payload_type = payload_type;
-
-// following 2 FIXMies could be set based on the current time, theres normaly no info leak, as rtp will likely be transmitted immedeatly
-    s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
-    s->timestamp = s->base_timestamp;
-    s->ssrc = 0; /* FIXME: was random(), what should this be? */
-    s->first_packet = 1;
-
-    max_packet_size = url_fget_max_packet_size(&s1->pb);
-    if (max_packet_size <= 12)
-        return AVERROR(EIO);
-    s->max_payload_size = max_packet_size - 12;
-
-    switch(st->codec->codec_id) {
-    case CODEC_ID_MP2:
-    case CODEC_ID_MP3:
-        s->buf_ptr = s->buf + 4;
-        s->cur_timestamp = 0;
-        break;
-    case CODEC_ID_MPEG1VIDEO:
-        s->cur_timestamp = 0;
-        break;
-    case CODEC_ID_MPEG2TS:
-        n = s->max_payload_size / TS_PACKET_SIZE;
-        if (n < 1)
-            n = 1;
-        s->max_payload_size = n * TS_PACKET_SIZE;
-        s->buf_ptr = s->buf;
-        break;
-    default:
-        s->buf_ptr = s->buf;
-        break;
-    }
-
-    return 0;
-}
-
-/* send an rtcp sender report packet */
-static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
-{
-    RTPDemuxContext *s = s1->priv_data;
-#if defined(DEBUG)
-    printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
-#endif
-    put_byte(&s1->pb, (RTP_VERSION << 6));
-    put_byte(&s1->pb, 200);
-    put_be16(&s1->pb, 6); /* length in words - 1 */
-    put_be32(&s1->pb, s->ssrc);
-    put_be64(&s1->pb, ntp_time);
-    put_be32(&s1->pb, s->timestamp);
-    put_be32(&s1->pb, s->packet_count);
-    put_be32(&s1->pb, s->octet_count);
-    put_flush_packet(&s1->pb);
-}
-
-/* send an rtp packet. sequence number is incremented, but the caller
-   must update the timestamp itself */
-static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
-{
-    RTPDemuxContext *s = s1->priv_data;
-
-#ifdef DEBUG
-    printf("rtp_send_data size=%d\n", len);
-#endif
-
-    /* build the RTP header */
-    put_byte(&s1->pb, (RTP_VERSION << 6));
-    put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
-    put_be16(&s1->pb, s->seq);
-    put_be32(&s1->pb, s->timestamp);
-    put_be32(&s1->pb, s->ssrc);
-
-    put_buffer(&s1->pb, buf1, len);
-    put_flush_packet(&s1->pb);
-
-    s->seq++;
-    s->octet_count += len;
-    s->packet_count++;
-}
-
-/* send an integer number of samples and compute time stamp and fill
-   the rtp send buffer before sending. */
-static void rtp_send_samples(AVFormatContext *s1,
-                             const uint8_t *buf1, int size, int sample_size)
-{
-    RTPDemuxContext *s = s1->priv_data;
-    int len, max_packet_size, n;
-
-    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
-    /* not needed, but who nows */
-    if ((size % sample_size) != 0)
-        av_abort();
-    while (size > 0) {
-        len = (max_packet_size - (s->buf_ptr - s->buf));
-        if (len > size)
-            len = size;
-
-        /* copy data */
-        memcpy(s->buf_ptr, buf1, len);
-        s->buf_ptr += len;
-        buf1 += len;
-        size -= len;
-        n = (s->buf_ptr - s->buf);
-        /* if buffer full, then send it */
-        if (n >= max_packet_size) {
-            rtp_send_data(s1, s->buf, n, 0);
-            s->buf_ptr = s->buf;
-            /* update timestamp */
-            s->timestamp += n / sample_size;
-        }
-    }
-}
-
-/* NOTE: we suppose that exactly one frame is given as argument here */
-/* XXX: test it */
-static void rtp_send_mpegaudio(AVFormatContext *s1,
-                               const uint8_t *buf1, int size)
-{
-    RTPDemuxContext *s = s1->priv_data;
-    AVStream *st = s1->streams[0];
-    int len, count, max_packet_size;
-
-    max_packet_size = s->max_payload_size;
-
-    /* test if we must flush because not enough space */
-    len = (s->buf_ptr - s->buf);
-    if ((len + size) > max_packet_size) {
-        if (len > 4) {
-            rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
-            s->buf_ptr = s->buf + 4;
-            /* 90 KHz time stamp */
-            s->timestamp = s->base_timestamp +
-                (s->cur_timestamp * 90000LL) / st->codec->sample_rate;
-        }
-    }
-
-    /* add the packet */
-    if (size > max_packet_size) {
-        /* big packet: fragment */
-        count = 0;
-        while (size > 0) {
-            len = max_packet_size - 4;
-            if (len > size)
-                len = size;
-            /* build fragmented packet */
-            s->buf[0] = 0;
-            s->buf[1] = 0;
-            s->buf[2] = count >> 8;
-            s->buf[3] = count;
-            memcpy(s->buf + 4, buf1, len);
-            rtp_send_data(s1, s->buf, len + 4, 0);
-            size -= len;
-            buf1 += len;
-            count += len;
-        }
-    } else {
-        if (s->buf_ptr == s->buf + 4) {
-            /* no fragmentation possible */
-            s->buf[0] = 0;
-            s->buf[1] = 0;
-            s->buf[2] = 0;
-            s->buf[3] = 0;
-        }
-        memcpy(s->buf_ptr, buf1, size);
-        s->buf_ptr += size;
-    }
-    s->cur_timestamp += st->codec->frame_size;
-}
 
 /* NOTE: a single frame must be passed with sequence header if
    needed. XXX: use slices. */
-static void rtp_send_mpegvideo(AVFormatContext *s1,
-                               const uint8_t *buf1, int size)
+void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size)
 {
     RTPDemuxContext *s = s1->priv_data;
     AVStream *st = s1->streams[0];
@@ -941,32 +61,7 @@ static void rtp_send_mpegvideo(AVFormatC
         /* 90 KHz time stamp */
         s->timestamp = s->base_timestamp +
             av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
-        rtp_send_data(s1, s->buf, q - s->buf, (len == size));
-
-        buf1 += len;
-        size -= len;
-    }
-    s->cur_timestamp++;
-}
-
-static void rtp_send_raw(AVFormatContext *s1,
-                         const uint8_t *buf1, int size)
-{
-    RTPDemuxContext *s = s1->priv_data;
-    AVStream *st = s1->streams[0];
-    int len, max_packet_size;
-
-    max_packet_size = s->max_payload_size;
-
-    while (size > 0) {
-        len = max_packet_size;
-        if (len > size)
-            len = size;
-
-        /* 90 KHz time stamp */
-        s->timestamp = s->base_timestamp +
-            av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
-        rtp_send_data(s1, buf1, len, (len == size));
+        ff_rtp_send_data(s1, s->buf, q - s->buf, (len == size));
 
         buf1 += len;
         size -= len;
@@ -974,95 +69,4 @@ static void rtp_send_raw(AVFormatContext
     s->cur_timestamp++;
 }
 
-/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
-static void rtp_send_mpegts_raw(AVFormatContext *s1,
-                                const uint8_t *buf1, int size)
-{
-    RTPDemuxContext *s = s1->priv_data;
-    int len, out_len;
 
-    while (size >= TS_PACKET_SIZE) {
-        len = s->max_payload_size - (s->buf_ptr - s->buf);
-        if (len > size)
-            len = size;
-        memcpy(s->buf_ptr, buf1, len);
-        buf1 += len;
-        size -= len;
-        s->buf_ptr += len;
-
-        out_len = s->buf_ptr - s->buf;
-        if (out_len >= s->max_payload_size) {
-            rtp_send_data(s1, s->buf, out_len, 0);
-            s->buf_ptr = s->buf;
-        }
-    }
-}
-
-/* write an RTP packet. 'buf1' must contain a single specific frame. */
-static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
-{
-    RTPDemuxContext *s = s1->priv_data;
-    AVStream *st = s1->streams[0];
-    int rtcp_bytes;
-    int64_t ntp_time;
-    int size= pkt->size;
-    uint8_t *buf1= pkt->data;
-
-#ifdef DEBUG
-    printf("%d: write len=%d\n", pkt->stream_index, size);
-#endif
-
-    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
-    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
-        RTCP_TX_RATIO_DEN;
-    if (s->first_packet || rtcp_bytes >= 28) {
-        /* compute NTP time */
-        /* XXX: 90 kHz timestamp hardcoded */
-        ntp_time = (pkt->pts << 28) / 5625;
-        rtcp_send_sr(s1, ntp_time);
-        s->last_octet_count = s->octet_count;
-        s->first_packet = 0;
-    }
-
-    switch(st->codec->codec_id) {
-    case CODEC_ID_PCM_MULAW:
-    case CODEC_ID_PCM_ALAW:
-    case CODEC_ID_PCM_U8:
-    case CODEC_ID_PCM_S8:
-        rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
-        break;
-    case CODEC_ID_PCM_U16BE:
-    case CODEC_ID_PCM_U16LE:
-    case CODEC_ID_PCM_S16BE:
-    case CODEC_ID_PCM_S16LE:
-        rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
-        break;
-    case CODEC_ID_MP2:
-    case CODEC_ID_MP3:
-        rtp_send_mpegaudio(s1, buf1, size);
-        break;
-    case CODEC_ID_MPEG1VIDEO:
-        rtp_send_mpegvideo(s1, buf1, size);
-        break;
-    case CODEC_ID_MPEG2TS:
-        rtp_send_mpegts_raw(s1, buf1, size);
-        break;
-    default:
-        /* better than nothing : send the codec raw data */
-        rtp_send_raw(s1, buf1, size);
-        break;
-    }
-    return 0;
-}
-
-AVOutputFormat rtp_muxer = {
-    "rtp",
-    "RTP output format",
-    NULL,
-    NULL,
-    sizeof(RTPDemuxContext),
-    CODEC_ID_PCM_MULAW,
-    CODEC_ID_NONE,
-    rtp_write_header,
-    rtp_write_packet,
-};

Added: trunk/libavformat/rtp_mpv.h
==============================================================================
--- (empty file)
+++ trunk/libavformat/rtp_mpv.h	Fri Aug 24 09:13:34 2007
@@ -0,0 +1,26 @@
+/*
+ * 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_MPV_H
+#define RTP_MPV_H
+
+void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size);
+
+#endif /* RTP_MPV_H */




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