Split the RTP muxer out of rtp.c, to simplify the RTSP demuxer's dependencies

Originally committed as revision 11408 to svn://svn.ffmpeg.org/ffmpeg/trunk

3 files changed, 357 insertions, 330 deletions

diff --git a/libavformat/Makefile b/libavformat/Makefile
index 7664087bbe..0d69448ad5 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -121,9 +121,9 @@ OBJS-$(CONFIG_RM_DEMUXER)                += rmdec.o
 OBJS-$(CONFIG_RM_MUXER)                  += rmenc.o
 OBJS-$(CONFIG_ROQ_DEMUXER)               += idroq.o
 OBJS-$(CONFIG_ROQ_MUXER)                 += raw.o
-OBJS-$(CONFIG_RTP_MUXER)                 += rtp.o rtp_mpv.o rtp_aac.o
+OBJS-$(CONFIG_RTP_MUXER)                 += rtp.o rtpenc.o rtp_mpv.o rtp_aac.o
 OBJS-$(CONFIG_RTSP_DEMUXER)              += rtsp.o
-OBJS-$(CONFIG_SDP_DEMUXER)               += rtsp.o rtp.o rtpdec.o rtp_h264.o rtp_mpv.o rtp_aac.o
+OBJS-$(CONFIG_SDP_DEMUXER)               += rtsp.o rtp.o rtpdec.o rtp_h264.o
 OBJS-$(CONFIG_SEGAFILM_DEMUXER)          += segafilm.o
 OBJS-$(CONFIG_SHORTEN_DEMUXER)           += raw.o
 OBJS-$(CONFIG_SIFF_DEMUXER)              += siff.o
diff --git a/libavformat/rtp.c b/libavformat/rtp.c
index 7841ef641f..ddd2ad1601 100644
--- a/libavformat/rtp.c
+++ b/libavformat/rtp.c
@@ -19,20 +19,15 @@
  * 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_mpv.h"
-#include "rtp_aac.h"
 
 //#define DEBUG
 
-#define RTCP_SR_SIZE 28
-
 /* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
 AVRtpPayloadType_t AVRtpPayloadTypes[]=
 {
@@ -225,326 +220,3 @@ enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type)
 
     return CODEC_ID_NONE;
 }
-
-/* 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 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
-    s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
-    s->timestamp = s->base_timestamp;
-    s->cur_timestamp = 0;
-    s->ssrc = 0; /* FIXME: was random(), what should this be? */
-    s->first_packet = 1;
-    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
-
-    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;
-
-    s->max_frames_per_packet = 0;
-    if (s1->max_delay) {
-        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
-            if (st->codec->frame_size == 0) {
-                av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
-            } else {
-                s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
-            }
-        }
-        if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
-            /* FIXME: We should round down here... */
-            s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
-        }
-    }
-
-    av_set_pts_info(st, 32, 1, 90000);
-    switch(st->codec->codec_id) {
-    case CODEC_ID_MP2:
-    case CODEC_ID_MP3:
-        s->buf_ptr = s->buf + 4;
-        break;
-    case CODEC_ID_MPEG1VIDEO:
-    case CODEC_ID_MPEG2VIDEO:
-        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;
-    case CODEC_ID_AAC:
-        s->read_buf_index = 0;
-    default:
-        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
-            av_set_pts_info(st, 32, 1, st->codec->sample_rate);
-        }
-        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;
-    uint32_t rtp_ts;
-
-#if defined(DEBUG)
-    printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
-#endif
-
-    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
-    s->last_rtcp_ntp_time = ntp_time;
-    rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
-                          s1->streams[0]->time_base) + s->base_timestamp;
-    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_be32(s1->pb, ntp_time / 1000000);
-    put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
-    put_be32(s1->pb, rtp_ts);
-    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 */
-void ff_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();
-    n = 0;
-    while (size > 0) {
-        s->buf_ptr = s->buf;
-        len = FFMIN(max_packet_size, size);
-
-        /* copy data */
-        memcpy(s->buf_ptr, buf1, len);
-        s->buf_ptr += len;
-        buf1 += len;
-        size -= len;
-        s->timestamp = s->cur_timestamp + n / sample_size;
-        ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
-        n += (s->buf_ptr - s->buf);
-    }
-}
-
-/* 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;
-    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) {
-            ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
-            s->buf_ptr = s->buf + 4;
-        }
-    }
-    if (s->buf_ptr == s->buf + 4) {
-        s->timestamp = s->cur_timestamp;
-    }
-
-    /* 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);
-            ff_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;
-    }
-}
-
-static void rtp_send_raw(AVFormatContext *s1,
-                         const uint8_t *buf1, int size)
-{
-    RTPDemuxContext *s = s1->priv_data;
-    int len, max_packet_size;
-
-    max_packet_size = s->max_payload_size;
-
-    while (size > 0) {
-        len = max_packet_size;
-        if (len > size)
-            len = size;
-
-        s->timestamp = s->cur_timestamp;
-        ff_rtp_send_data(s1, buf1, len, (len == size));
-
-        buf1 += len;
-        size -= len;
-    }
-}
-
-/* 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) {
-            ff_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;
-    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 >= RTCP_SR_SIZE) &&
-                           (av_gettime() - s->last_rtcp_ntp_time > 5000000))) {
-        rtcp_send_sr(s1, av_gettime());
-        s->last_octet_count = s->octet_count;
-        s->first_packet = 0;
-    }
-    s->cur_timestamp = s->base_timestamp + pkt->pts;
-
-    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:
-    case CODEC_ID_MPEG2VIDEO:
-        ff_rtp_send_mpegvideo(s1, buf1, size);
-        break;
-    case CODEC_ID_AAC:
-        ff_rtp_send_aac(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,
-};
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
new file mode 100644
index 0000000000..291d474204
--- /dev/null
+++ b/libavformat/rtpenc.c
@@ -0,0 +1,355 @@
+/*
+ * RTP 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 "network.h"
+
+#include "rtp_internal.h"
+#include "rtp_mpv.h"
+#include "rtp_aac.h"
+
+//#define DEBUG
+
+#define RTCP_SR_SIZE 28
+
+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 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
+    s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
+    s->timestamp = s->base_timestamp;
+    s->cur_timestamp = 0;
+    s->ssrc = 0; /* FIXME: was random(), what should this be? */
+    s->first_packet = 1;
+    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
+
+    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;
+
+    s->max_frames_per_packet = 0;
+    if (s1->max_delay) {
+        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+            if (st->codec->frame_size == 0) {
+                av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
+            } else {
+                s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
+            }
+        }
+        if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
+            /* FIXME: We should round down here... */
+            s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
+        }
+    }
+
+    av_set_pts_info(st, 32, 1, 90000);
+    switch(st->codec->codec_id) {
+    case CODEC_ID_MP2:
+    case CODEC_ID_MP3:
+        s->buf_ptr = s->buf + 4;
+        break;
+    case CODEC_ID_MPEG1VIDEO:
+    case CODEC_ID_MPEG2VIDEO:
+        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;
+    case CODEC_ID_AAC:
+        s->read_buf_index = 0;
+    default:
+        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+            av_set_pts_info(st, 32, 1, st->codec->sample_rate);
+        }
+        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;
+    uint32_t rtp_ts;
+
+#if defined(DEBUG)
+    printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
+#endif
+
+    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
+    s->last_rtcp_ntp_time = ntp_time;
+    rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
+                          s1->streams[0]->time_base) + s->base_timestamp;
+    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_be32(s1->pb, ntp_time / 1000000);
+    put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
+    put_be32(s1->pb, rtp_ts);
+    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 */
+void ff_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();
+    n = 0;
+    while (size > 0) {
+        s->buf_ptr = s->buf;
+        len = FFMIN(max_packet_size, size);
+
+        /* copy data */
+        memcpy(s->buf_ptr, buf1, len);
+        s->buf_ptr += len;
+        buf1 += len;
+        size -= len;
+        s->timestamp = s->cur_timestamp + n / sample_size;
+        ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
+        n += (s->buf_ptr - s->buf);
+    }
+}
+
+/* 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;
+    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) {
+            ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
+            s->buf_ptr = s->buf + 4;
+        }
+    }
+    if (s->buf_ptr == s->buf + 4) {
+        s->timestamp = s->cur_timestamp;
+    }
+
+    /* 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);
+            ff_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;
+    }
+}
+
+static void rtp_send_raw(AVFormatContext *s1,
+                         const uint8_t *buf1, int size)
+{
+    RTPDemuxContext *s = s1->priv_data;
+    int len, max_packet_size;
+
+    max_packet_size = s->max_payload_size;
+
+    while (size > 0) {
+        len = max_packet_size;
+        if (len > size)
+            len = size;
+
+        s->timestamp = s->cur_timestamp;
+        ff_rtp_send_data(s1, buf1, len, (len == size));
+
+        buf1 += len;
+        size -= len;
+    }
+}
+
+/* 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) {
+            ff_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;
+    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 >= RTCP_SR_SIZE) &&
+                           (av_gettime() - s->last_rtcp_ntp_time > 5000000))) {
+        rtcp_send_sr(s1, av_gettime());
+        s->last_octet_count = s->octet_count;
+        s->first_packet = 0;
+    }
+    s->cur_timestamp = s->base_timestamp + pkt->pts;
+
+    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:
+    case CODEC_ID_MPEG2VIDEO:
+        ff_rtp_send_mpegvideo(s1, buf1, size);
+        break;
+    case CODEC_ID_AAC:
+        ff_rtp_send_aac(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,
+};