* 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[]=
{
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,
-};
--- /dev/null
+/*
+ * 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,
+};