[FFmpeg-cvslog] r11408 - in trunk/libavformat: Makefile rtp.c rtpenc.c

lucabe subversion
Fri Jan 4 21:09:48 CET 2008


Author: lucabe
Date: Fri Jan  4 21:09:48 2008
New Revision: 11408

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


Added:
   trunk/libavformat/rtpenc.c
      - copied, changed from r11406, /trunk/libavformat/rtp.c
Modified:
   trunk/libavformat/Makefile
   trunk/libavformat/rtp.c

Modified: trunk/libavformat/Makefile
==============================================================================
--- trunk/libavformat/Makefile	(original)
+++ trunk/libavformat/Makefile	Fri Jan  4 21:09:48 2008
@@ -121,9 +121,9 @@ OBJS-$(CONFIG_RM_DEMUXER)               
 OBJS-$(CONFIG_RM_MUXER)                  += rmenc.o
 OBJS-$(CONFIG_ROQ_DEMUXER)               += idroq.o
 OBJS-$(CONFIG_ROQ_MUXER)                 += raw.o
-OBJS-$(CONFIG_RTP_MUXER)                 += rtp.o rtp_mpv.o rtp_aac.o
+OBJS-$(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

Modified: trunk/libavformat/rtp.c
==============================================================================
--- trunk/libavformat/rtp.c	(original)
+++ trunk/libavformat/rtp.c	Fri Jan  4 21:09:48 2008
@@ -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 
 
     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,
-};

Copied: trunk/libavformat/rtpenc.c (from r11406, /trunk/libavformat/rtp.c)
==============================================================================
--- /trunk/libavformat/rtp.c	(original)
+++ trunk/libavformat/rtpenc.c	Fri Jan  4 21:09:48 2008
@@ -1,5 +1,5 @@
 /*
- * RTP input/output format
+ * RTP output format
  * Copyright (c) 2002 Fabrice Bellard.
  *
  * This file is part of FFmpeg.
@@ -33,201 +33,6 @@
 
 #define RTCP_SR_SIZE 28
 
-/* 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},
-  {32, "MPV",        CODEC_TYPE_VIDEO,   CODEC_ID_MPEG2VIDEO, 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}
-};
-
-int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
-{
-    int i = 0;
-
-    for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
-        if (AVRtpPayloadTypes[i].pt == payload_type) {
-            if (AVRtpPayloadTypes[i].codec_id != CODEC_ID_NONE) {
-                codec->codec_type = AVRtpPayloadTypes[i].codec_type;
-                codec->codec_id = AVRtpPayloadTypes[i].codec_id;
-                if (AVRtpPayloadTypes[i].audio_channels > 0)
-                    codec->channels = AVRtpPayloadTypes[i].audio_channels;
-                if (AVRtpPayloadTypes[i].clock_rate > 0)
-                    codec->sample_rate = AVRtpPayloadTypes[i].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;
-}
-
-const char *ff_rtp_enc_name(int payload_type)
-{
-    int i;
-
-    for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
-        if (AVRtpPayloadTypes[i].pt == payload_type) {
-            return AVRtpPayloadTypes[i].enc_name;
-        }
-
-    return "";
-}
-
-enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type)
-{
-    int i;
-
-    for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
-        if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec_type == AVRtpPayloadTypes[i].codec_type)){
-            return AVRtpPayloadTypes[i].codec_id;
-        }
-
-    return CODEC_ID_NONE;
-}
-
-/* rtp output */
-
 static int rtp_write_header(AVFormatContext *s1)
 {
     RTPDemuxContext *s = s1->priv_data;




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