Start working on a branch where FFmpeg is not copied, patched and carved to be built with automake but instead imported inline and built using its own build system. This is an import of a slightly modified FFmpeg current tree. xine-lib builds, install and run fine with it, but there are of course plenty of things that needs to be fixed before it can even be considered for a 1.2.x series. Work will continue in the next days of course.

CVS patchset: 8397
CVS date: 2006/12/02 01:19:48

1 files changed, 1099 insertions, 0 deletions

diff --git a/contrib/ffmpeg/libavformat/rtp.c b/contrib/ffmpeg/libavformat/rtp.c
new file mode 100644
index 000000000..37a286289
--- /dev/null
+++ b/contrib/ffmpeg/libavformat/rtp.c
@@ -0,0 +1,1099 @@
+/*
+ * RTP input/output format
+ * 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
+ */
+#include "avformat.h"
+#include "mpegts.h"
+#include "bitstream.h"
+
+#include <unistd.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <netinet/in.h>
+#ifndef __BEOS__
+# include <arpa/inet.h>
+#else
+# include "barpainet.h"
+#endif
+#include <netdb.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()
+{
+    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;
+}
+
+/* return < 0 if unknown payload type */
+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 inline uint32_t decode_be32(const uint8_t *p)
+{
+    return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
+}
+
+static inline uint64_t decode_be64(const uint8_t *p)
+{
+    return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
+}
+
+static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
+{
+    if (buf[1] != 200)
+        return -1;
+    s->last_rtcp_ntp_time = decode_be64(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 = decode_be32(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
+
+/**
+ * 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)
+{
+    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 = 1;
+            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 = BE_16(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 doesnt 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  = (buf[2] << 8) | buf[3];
+    timestamp = decode_be32(buf + 4);
+    ssrc = decode_be32(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 = decode_be32(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 = decode_be32(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_IO;
+    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)
+{
+    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)
+{
+    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));
+
+        buf1 += len;
+        size -= len;
+    }
+    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;
+}
+
+static int rtp_write_trailer(AVFormatContext *s1)
+{
+    //    RTPDemuxContext *s = s1->priv_data;
+    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,
+    rtp_write_trailer,
+};