libavformat/rtpenc.c - nest-android-app/ffmpeg - Git at Google

 /*
  * 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 "internal.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/random_seed.h"
 #include "libavutil/opt.h"

 #include "rtpenc.h"

 static const AVOption options[] = {
     FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
     { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
     { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
     { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
     { "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
     { NULL },
 };

 static const AVClass rtp_muxer_class = {
     .class_name = "RTP muxer",
     .item_name  = av_default_item_name,
     .option     = options,
     .version    = LIBAVUTIL_VERSION_INT,
 };

 #define RTCP_SR_SIZE 28

 static int is_supported(enum AVCodecID id)
 {
     switch(id) {
     case AV_CODEC_ID_H261:
     case AV_CODEC_ID_H263:
     case AV_CODEC_ID_H263P:
     case AV_CODEC_ID_H264:
     case AV_CODEC_ID_HEVC:
     case AV_CODEC_ID_MPEG1VIDEO:
     case AV_CODEC_ID_MPEG2VIDEO:
     case AV_CODEC_ID_MPEG4:
     case AV_CODEC_ID_AAC:
     case AV_CODEC_ID_MP2:
     case AV_CODEC_ID_MP3:
     case AV_CODEC_ID_PCM_ALAW:
     case AV_CODEC_ID_PCM_MULAW:
     case AV_CODEC_ID_PCM_S8:
     case AV_CODEC_ID_PCM_S16BE:
     case AV_CODEC_ID_PCM_S16LE:
     case AV_CODEC_ID_PCM_U16BE:
     case AV_CODEC_ID_PCM_U16LE:
     case AV_CODEC_ID_PCM_U8:
     case AV_CODEC_ID_MPEG2TS:
     case AV_CODEC_ID_AMR_NB:
     case AV_CODEC_ID_AMR_WB:
     case AV_CODEC_ID_VORBIS:
     case AV_CODEC_ID_THEORA:
     case AV_CODEC_ID_VP8:
     case AV_CODEC_ID_ADPCM_G722:
     case AV_CODEC_ID_ADPCM_G726:
     case AV_CODEC_ID_ILBC:
     case AV_CODEC_ID_MJPEG:
     case AV_CODEC_ID_SPEEX:
     case AV_CODEC_ID_OPUS:
         return 1;
     default:
         return 0;
     }
 }

 static int rtp_write_header(AVFormatContext *s1)
 {
     RTPMuxContext *s = s1->priv_data;
     int n, ret = AVERROR(EINVAL);
     AVStream *st;

     if (s1->nb_streams != 1) {
         av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
         return AVERROR(EINVAL);
     }
     st = s1->streams[0];
     if (!is_supported(st->codec->codec_id)) {
         av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));

         return -1;
     }

     if (s->payload_type < 0) {
         /* Re-validate non-dynamic payload types */
         if (st->id < RTP_PT_PRIVATE)
             st->id = ff_rtp_get_payload_type(s1, st->codec, -1);

         s->payload_type = st->id;
     } else {
         /* private option takes priority */
         st->id = s->payload_type;
     }

     s->base_timestamp = av_get_random_seed();
     s->timestamp = s->base_timestamp;
     s->cur_timestamp = 0;
     if (!s->ssrc)
         s->ssrc = av_get_random_seed();
     s->first_packet = 1;
     s->first_rtcp_ntp_time = ff_ntp_time();
     if (s1->start_time_realtime != 0  &&  s1->start_time_realtime != AV_NOPTS_VALUE)
         /* Round the NTP time to whole milliseconds. */
         s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
                                  NTP_OFFSET_US;
     // Pick a random sequence start number, but in the lower end of the
     // available range, so that any wraparound doesn't happen immediately.
     // (Immediate wraparound would be an issue for SRTP.)
     if (s->seq < 0) {
         if (s1->flags & AVFMT_FLAG_BITEXACT) {
             s->seq = 0;
         } else
             s->seq = av_get_random_seed() & 0x0fff;
     } else
         s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval

     if (s1->packet_size) {
         if (s1->pb->max_packet_size)
             s1->packet_size = FFMIN(s1->packet_size,
                                     s1->pb->max_packet_size);
     } else
         s1->packet_size = s1->pb->max_packet_size;
     if (s1->packet_size <= 12) {
         av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
         return AVERROR(EIO);
     }
     s->buf = av_malloc(s1->packet_size);
     if (!s->buf) {
         return AVERROR(ENOMEM);
     }
     s->max_payload_size = s1->packet_size - 12;

     if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
         avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
     } else {
         avpriv_set_pts_info(st, 32, 1, 90000);
     }
     s->buf_ptr = s->buf;
     switch(st->codec->codec_id) {
     case AV_CODEC_ID_MP2:
     case AV_CODEC_ID_MP3:
         s->buf_ptr = s->buf + 4;
         avpriv_set_pts_info(st, 32, 1, 90000);
         break;
     case AV_CODEC_ID_MPEG1VIDEO:
     case AV_CODEC_ID_MPEG2VIDEO:
         break;
     case AV_CODEC_ID_MPEG2TS:
         n = s->max_payload_size / TS_PACKET_SIZE;
         if (n < 1)
             n = 1;
         s->max_payload_size = n * TS_PACKET_SIZE;
         break;
     case AV_CODEC_ID_H261:
         if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
             av_log(s, AV_LOG_ERROR,
                    "Packetizing H261 is experimental and produces incorrect "
                    "packetization for cases where GOBs don't fit into packets "
                    "(even though most receivers may handle it just fine). "
                    "Please set -f_strict experimental in order to enable it.\n");
             ret = AVERROR_EXPERIMENTAL;
             goto fail;
         }
         break;
     case AV_CODEC_ID_H264:
         /* check for H.264 MP4 syntax */
         if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
             s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
         }
         break;
     case AV_CODEC_ID_HEVC:
         /* Only check for the standardized hvcC version of extradata, keeping
          * things simple and similar to the avcC/H264 case above, instead
          * of trying to handle the pre-standardization versions (as in
          * libavcodec/hevc.c). */
         if (st->codec->extradata_size > 21 && st->codec->extradata[0] == 1) {
             s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1;
         }
         break;
     case AV_CODEC_ID_VORBIS:
     case AV_CODEC_ID_THEORA:
         s->max_frames_per_packet = 15;
         break;
     case AV_CODEC_ID_ADPCM_G722:
         /* Due to a historical error, the clock rate for G722 in RTP is
          * 8000, even if the sample rate is 16000. See RFC 3551. */
         avpriv_set_pts_info(st, 32, 1, 8000);
         break;
     case AV_CODEC_ID_OPUS:
         if (st->codec->channels > 2) {
             av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
             goto fail;
         }
         /* The opus RTP RFC says that all opus streams should use 48000 Hz
          * as clock rate, since all opus sample rates can be expressed in
          * this clock rate, and sample rate changes on the fly are supported. */
         avpriv_set_pts_info(st, 32, 1, 48000);
         break;
     case AV_CODEC_ID_ILBC:
         if (st->codec->block_align != 38 && st->codec->block_align != 50) {
             av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
             goto fail;
         }
         s->max_frames_per_packet = s->max_payload_size / st->codec->block_align;
         break;
     case AV_CODEC_ID_AMR_NB:
     case AV_CODEC_ID_AMR_WB:
         s->max_frames_per_packet = 50;
         if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
             n = 31;
         else
             n = 61;
         /* max_header_toc_size + the largest AMR payload must fit */
         if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
             av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
             goto fail;
         }
         if (st->codec->channels != 1) {
             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
             goto fail;
         }
         break;
     case AV_CODEC_ID_AAC:
         s->max_frames_per_packet = 50;
         break;
     default:
         break;
     }

     return 0;

 fail:
     av_freep(&s->buf);
     return ret;
 }

 /* send an rtcp sender report packet */
 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
 {
     RTPMuxContext *s = s1->priv_data;
     uint32_t rtp_ts;

     av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);

     s->last_rtcp_ntp_time = ntp_time;
     rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
                           s1->streams[0]->time_base) + s->base_timestamp;
     avio_w8(s1->pb, RTP_VERSION << 6);
     avio_w8(s1->pb, RTCP_SR);
     avio_wb16(s1->pb, 6); /* length in words - 1 */
     avio_wb32(s1->pb, s->ssrc);
     avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time));
     avio_wb32(s1->pb, rtp_ts);
     avio_wb32(s1->pb, s->packet_count);
     avio_wb32(s1->pb, s->octet_count);

     if (s->cname) {
         int len = FFMIN(strlen(s->cname), 255);
         avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
         avio_w8(s1->pb, RTCP_SDES);
         avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */

         avio_wb32(s1->pb, s->ssrc);
         avio_w8(s1->pb, 0x01); /* CNAME */
         avio_w8(s1->pb, len);
         avio_write(s1->pb, s->cname, len);
         avio_w8(s1->pb, 0); /* END */
         for (len = (7 + len) % 4; len % 4; len++)
             avio_w8(s1->pb, 0);
     }

     if (bye) {
         avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
         avio_w8(s1->pb, RTCP_BYE);
         avio_wb16(s1->pb, 1); /* length in words - 1 */
         avio_wb32(s1->pb, s->ssrc);
     }

     avio_flush(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)
 {
     RTPMuxContext *s = s1->priv_data;

     av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);

     /* build the RTP header */
     avio_w8(s1->pb, RTP_VERSION << 6);
     avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
     avio_wb16(s1->pb, s->seq);
     avio_wb32(s1->pb, s->timestamp);
     avio_wb32(s1->pb, s->ssrc);

     avio_write(s1->pb, buf1, len);
     avio_flush(s1->pb);

     s->seq = (s->seq + 1) & 0xffff;
     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 int rtp_send_samples(AVFormatContext *s1,
                             const uint8_t *buf1, int size, int sample_size_bits)
 {
     RTPMuxContext *s = s1->priv_data;
     int len, max_packet_size, n;
     /* Calculate the number of bytes to get samples aligned on a byte border */
     int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);

     max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
     /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
     if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
         return AVERROR(EINVAL);
     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 * 8 / sample_size_bits;
         ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
         n += (s->buf_ptr - s->buf);
     }
     return 0;
 }

 static void rtp_send_mpegaudio(AVFormatContext *s1,
                                const uint8_t *buf1, int size)
 {
     RTPMuxContext *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)
 {
     RTPMuxContext *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)
 {
     RTPMuxContext *s = s1->priv_data;
     int len, out_len;

     s->timestamp = s->cur_timestamp;
     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;
         }
     }
 }

 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
 {
     RTPMuxContext *s = s1->priv_data;
     AVStream *st = s1->streams[0];
     int frame_duration = av_get_audio_frame_duration(st->codec, 0);
     int frame_size = st->codec->block_align;
     int frames = size / frame_size;

     while (frames > 0) {
         if (s->num_frames > 0 &&
             av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
                           s1->max_delay, AV_TIME_BASE_Q) >= 0) {
             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
             s->num_frames = 0;
         }

         if (!s->num_frames) {
             s->buf_ptr = s->buf;
             s->timestamp = s->cur_timestamp;
         }
         memcpy(s->buf_ptr, buf, frame_size);
         frames--;
         s->num_frames++;
         s->buf_ptr       += frame_size;
         buf              += frame_size;
         s->cur_timestamp += frame_duration;

         if (s->num_frames == s->max_frames_per_packet) {
             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
             s->num_frames = 0;
         }
     }
     return 0;
 }

 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
 {
     RTPMuxContext *s = s1->priv_data;
     AVStream *st = s1->streams[0];
     int rtcp_bytes;
     int size= pkt->size;

     av_log(s1, AV_LOG_TRACE, "%d: write len=%d\n", pkt->stream_index, size);

     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) &&
                             (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
         !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
         rtcp_send_sr(s1, ff_ntp_time(), 0);
         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 AV_CODEC_ID_PCM_MULAW:
     case AV_CODEC_ID_PCM_ALAW:
     case AV_CODEC_ID_PCM_U8:
     case AV_CODEC_ID_PCM_S8:
         return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
     case AV_CODEC_ID_PCM_U16BE:
     case AV_CODEC_ID_PCM_U16LE:
     case AV_CODEC_ID_PCM_S16BE:
     case AV_CODEC_ID_PCM_S16LE:
         return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
     case AV_CODEC_ID_ADPCM_G722:
         /* The actual sample size is half a byte per sample, but since the
          * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
          * the correct parameter for send_samples_bits is 8 bits per stream
          * clock. */
         return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
     case AV_CODEC_ID_ADPCM_G726:
         return rtp_send_samples(s1, pkt->data, size,
                                 st->codec->bits_per_coded_sample * st->codec->channels);
     case AV_CODEC_ID_MP2:
     case AV_CODEC_ID_MP3:
         rtp_send_mpegaudio(s1, pkt->data, size);
         break;
     case AV_CODEC_ID_MPEG1VIDEO:
     case AV_CODEC_ID_MPEG2VIDEO:
         ff_rtp_send_mpegvideo(s1, pkt->data, size);
         break;
     case AV_CODEC_ID_AAC:
         if (s->flags & FF_RTP_FLAG_MP4A_LATM)
             ff_rtp_send_latm(s1, pkt->data, size);
         else
             ff_rtp_send_aac(s1, pkt->data, size);
         break;
     case AV_CODEC_ID_AMR_NB:
     case AV_CODEC_ID_AMR_WB:
         ff_rtp_send_amr(s1, pkt->data, size);
         break;
     case AV_CODEC_ID_MPEG2TS:
         rtp_send_mpegts_raw(s1, pkt->data, size);
         break;
     case AV_CODEC_ID_H264:
         ff_rtp_send_h264_hevc(s1, pkt->data, size);
         break;
     case AV_CODEC_ID_H261:
         ff_rtp_send_h261(s1, pkt->data, size);
         break;
     case AV_CODEC_ID_H263:
         if (s->flags & FF_RTP_FLAG_RFC2190) {
             int mb_info_size = 0;
             const uint8_t *mb_info =
                 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
                                         &mb_info_size);
             ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
             break;
         }
         /* Fallthrough */
     case AV_CODEC_ID_H263P:
         ff_rtp_send_h263(s1, pkt->data, size);
         break;
     case AV_CODEC_ID_HEVC:
         ff_rtp_send_h264_hevc(s1, pkt->data, size);
         break;
     case AV_CODEC_ID_VORBIS:
     case AV_CODEC_ID_THEORA:
         ff_rtp_send_xiph(s1, pkt->data, size);
         break;
     case AV_CODEC_ID_VP8:
         ff_rtp_send_vp8(s1, pkt->data, size);
         break;
     case AV_CODEC_ID_ILBC:
         rtp_send_ilbc(s1, pkt->data, size);
         break;
     case AV_CODEC_ID_MJPEG:
         ff_rtp_send_jpeg(s1, pkt->data, size);
         break;
     case AV_CODEC_ID_OPUS:
         if (size > s->max_payload_size) {
             av_log(s1, AV_LOG_ERROR,
                    "Packet size %d too large for max RTP payload size %d\n",
                    size, s->max_payload_size);
             return AVERROR(EINVAL);
         }
         /* Intentional fallthrough */
     default:
         /* better than nothing : send the codec raw data */
         rtp_send_raw(s1, pkt->data, size);
         break;
     }
     return 0;
 }

 static int rtp_write_trailer(AVFormatContext *s1)
 {
     RTPMuxContext *s = s1->priv_data;

     /* If the caller closes and recreates ->pb, this might actually
      * be NULL here even if it was successfully allocated at the start. */
     if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
         rtcp_send_sr(s1, ff_ntp_time(), 1);
     av_freep(&s->buf);

     return 0;
 }

 AVOutputFormat ff_rtp_muxer = {
     .name              = "rtp",
     .long_name         = NULL_IF_CONFIG_SMALL("RTP output"),
     .priv_data_size    = sizeof(RTPMuxContext),
     .audio_codec       = AV_CODEC_ID_PCM_MULAW,
     .video_codec       = AV_CODEC_ID_MPEG4,
     .write_header      = rtp_write_header,
     .write_packet      = rtp_write_packet,
     .write_trailer     = rtp_write_trailer,
     .priv_class        = &rtp_muxer_class,
     .flags             = AVFMT_TS_NONSTRICT,
 };
	/*
	* 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 "internal.h"
	#include "libavutil/mathematics.h"
	#include "libavutil/random_seed.h"
	#include "libavutil/opt.h"

	#include "rtpenc.h"

	static const AVOption options[] = {
	FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
	{ "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
	{ "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
	{ "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
	{ "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
	{ NULL },
	};

	static const AVClass rtp_muxer_class = {
	.class_name = "RTP muxer",
	.item_name = av_default_item_name,
	.option = options,
	.version = LIBAVUTIL_VERSION_INT,
	};

	#define RTCP_SR_SIZE 28

	static int is_supported(enum AVCodecID id)
	{
	switch(id) {
	case AV_CODEC_ID_H261:
	case AV_CODEC_ID_H263:
	case AV_CODEC_ID_H263P:
	case AV_CODEC_ID_H264:
	case AV_CODEC_ID_HEVC:
	case AV_CODEC_ID_MPEG1VIDEO:
	case AV_CODEC_ID_MPEG2VIDEO:
	case AV_CODEC_ID_MPEG4:
	case AV_CODEC_ID_AAC:
	case AV_CODEC_ID_MP2:
	case AV_CODEC_ID_MP3:
	case AV_CODEC_ID_PCM_ALAW:
	case AV_CODEC_ID_PCM_MULAW:
	case AV_CODEC_ID_PCM_S8:
	case AV_CODEC_ID_PCM_S16BE:
	case AV_CODEC_ID_PCM_S16LE:
	case AV_CODEC_ID_PCM_U16BE:
	case AV_CODEC_ID_PCM_U16LE:
	case AV_CODEC_ID_PCM_U8:
	case AV_CODEC_ID_MPEG2TS:
	case AV_CODEC_ID_AMR_NB:
	case AV_CODEC_ID_AMR_WB:
	case AV_CODEC_ID_VORBIS:
	case AV_CODEC_ID_THEORA:
	case AV_CODEC_ID_VP8:
	case AV_CODEC_ID_ADPCM_G722:
	case AV_CODEC_ID_ADPCM_G726:
	case AV_CODEC_ID_ILBC:
	case AV_CODEC_ID_MJPEG:
	case AV_CODEC_ID_SPEEX:
	case AV_CODEC_ID_OPUS:
	return 1;
	default:
	return 0;
	}
	}

	static int rtp_write_header(AVFormatContext *s1)
	{
	RTPMuxContext *s = s1->priv_data;
	int n, ret = AVERROR(EINVAL);
	AVStream *st;

	if (s1->nb_streams != 1) {
	av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
	return AVERROR(EINVAL);
	}
	st = s1->streams[0];
	if (!is_supported(st->codec->codec_id)) {
	av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));

	return -1;
	}

	if (s->payload_type < 0) {
	/* Re-validate non-dynamic payload types */
	if (st->id < RTP_PT_PRIVATE)
	st->id = ff_rtp_get_payload_type(s1, st->codec, -1);

	s->payload_type = st->id;
	} else {
	/* private option takes priority */
	st->id = s->payload_type;
	}

	s->base_timestamp = av_get_random_seed();
	s->timestamp = s->base_timestamp;
	s->cur_timestamp = 0;
	if (!s->ssrc)
	s->ssrc = av_get_random_seed();
	s->first_packet = 1;
	s->first_rtcp_ntp_time = ff_ntp_time();
	if (s1->start_time_realtime != 0 && s1->start_time_realtime != AV_NOPTS_VALUE)
	/* Round the NTP time to whole milliseconds. */
	s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
	NTP_OFFSET_US;
	// Pick a random sequence start number, but in the lower end of the
	// available range, so that any wraparound doesn't happen immediately.
	// (Immediate wraparound would be an issue for SRTP.)
	if (s->seq < 0) {
	if (s1->flags & AVFMT_FLAG_BITEXACT) {
	s->seq = 0;
	} else
	s->seq = av_get_random_seed() & 0x0fff;
	} else
	s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval

	if (s1->packet_size) {
	if (s1->pb->max_packet_size)
	s1->packet_size = FFMIN(s1->packet_size,
	s1->pb->max_packet_size);
	} else
	s1->packet_size = s1->pb->max_packet_size;
	if (s1->packet_size <= 12) {
	av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
	return AVERROR(EIO);
	}
	s->buf = av_malloc(s1->packet_size);
	if (!s->buf) {
	return AVERROR(ENOMEM);
	}
	s->max_payload_size = s1->packet_size - 12;

	if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
	avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
	} else {
	avpriv_set_pts_info(st, 32, 1, 90000);
	}
	s->buf_ptr = s->buf;
	switch(st->codec->codec_id) {
	case AV_CODEC_ID_MP2:
	case AV_CODEC_ID_MP3:
	s->buf_ptr = s->buf + 4;
	avpriv_set_pts_info(st, 32, 1, 90000);
	break;
	case AV_CODEC_ID_MPEG1VIDEO:
	case AV_CODEC_ID_MPEG2VIDEO:
	break;
	case AV_CODEC_ID_MPEG2TS:
	n = s->max_payload_size / TS_PACKET_SIZE;
	if (n < 1)
	n = 1;
	s->max_payload_size = n * TS_PACKET_SIZE;
	break;
	case AV_CODEC_ID_H261:
	if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
	av_log(s, AV_LOG_ERROR,
	"Packetizing H261 is experimental and produces incorrect "
	"packetization for cases where GOBs don't fit into packets "
	"(even though most receivers may handle it just fine). "
	"Please set -f_strict experimental in order to enable it.\n");
	ret = AVERROR_EXPERIMENTAL;
	goto fail;
	}
	break;
	case AV_CODEC_ID_H264:
	/* check for H.264 MP4 syntax */
	if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
	s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
	}
	break;
	case AV_CODEC_ID_HEVC:
	/* Only check for the standardized hvcC version of extradata, keeping
	* things simple and similar to the avcC/H264 case above, instead
	* of trying to handle the pre-standardization versions (as in
	* libavcodec/hevc.c). */
	if (st->codec->extradata_size > 21 && st->codec->extradata[0] == 1) {
	s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1;
	}
	break;
	case AV_CODEC_ID_VORBIS:
	case AV_CODEC_ID_THEORA:
	s->max_frames_per_packet = 15;
	break;
	case AV_CODEC_ID_ADPCM_G722:
	/* Due to a historical error, the clock rate for G722 in RTP is
	* 8000, even if the sample rate is 16000. See RFC 3551. */
	avpriv_set_pts_info(st, 32, 1, 8000);
	break;
	case AV_CODEC_ID_OPUS:
	if (st->codec->channels > 2) {
	av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
	goto fail;
	}
	/* The opus RTP RFC says that all opus streams should use 48000 Hz
	* as clock rate, since all opus sample rates can be expressed in
	* this clock rate, and sample rate changes on the fly are supported. */
	avpriv_set_pts_info(st, 32, 1, 48000);
	break;
	case AV_CODEC_ID_ILBC:
	if (st->codec->block_align != 38 && st->codec->block_align != 50) {
	av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
	goto fail;
	}
	s->max_frames_per_packet = s->max_payload_size / st->codec->block_align;
	break;
	case AV_CODEC_ID_AMR_NB:
	case AV_CODEC_ID_AMR_WB:
	s->max_frames_per_packet = 50;
	if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
	n = 31;
	else
	n = 61;
	/* max_header_toc_size + the largest AMR payload must fit */
	if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
	av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
	goto fail;
	}
	if (st->codec->channels != 1) {
	av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
	goto fail;
	}
	break;
	case AV_CODEC_ID_AAC:
	s->max_frames_per_packet = 50;
	break;
	default:
	break;
	}

	return 0;

	fail:
	av_freep(&s->buf);
	return ret;
	}

	/* send an rtcp sender report packet */
	static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
	{
	RTPMuxContext *s = s1->priv_data;
	uint32_t rtp_ts;

	av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);

	s->last_rtcp_ntp_time = ntp_time;
	rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
	s1->streams[0]->time_base) + s->base_timestamp;
	avio_w8(s1->pb, RTP_VERSION << 6);
	avio_w8(s1->pb, RTCP_SR);
	avio_wb16(s1->pb, 6); /* length in words - 1 */
	avio_wb32(s1->pb, s->ssrc);
	avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time));
	avio_wb32(s1->pb, rtp_ts);
	avio_wb32(s1->pb, s->packet_count);
	avio_wb32(s1->pb, s->octet_count);

	if (s->cname) {
	int len = FFMIN(strlen(s->cname), 255);
	avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
	avio_w8(s1->pb, RTCP_SDES);
	avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */

	avio_wb32(s1->pb, s->ssrc);
	avio_w8(s1->pb, 0x01); /* CNAME */
	avio_w8(s1->pb, len);
	avio_write(s1->pb, s->cname, len);
	avio_w8(s1->pb, 0); /* END */
	for (len = (7 + len) % 4; len % 4; len++)
	avio_w8(s1->pb, 0);
	}

	if (bye) {
	avio_w8(s1->pb, (RTP_VERSION << 6) \| 1);
	avio_w8(s1->pb, RTCP_BYE);
	avio_wb16(s1->pb, 1); /* length in words - 1 */
	avio_wb32(s1->pb, s->ssrc);
	}

	avio_flush(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)
	{
	RTPMuxContext *s = s1->priv_data;

	av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);

	/* build the RTP header */
	avio_w8(s1->pb, RTP_VERSION << 6);
	avio_w8(s1->pb, (s->payload_type & 0x7f) \| ((m & 0x01) << 7));
	avio_wb16(s1->pb, s->seq);
	avio_wb32(s1->pb, s->timestamp);
	avio_wb32(s1->pb, s->ssrc);

	avio_write(s1->pb, buf1, len);
	avio_flush(s1->pb);

	s->seq = (s->seq + 1) & 0xffff;
	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 int rtp_send_samples(AVFormatContext *s1,
	const uint8_t *buf1, int size, int sample_size_bits)
	{
	RTPMuxContext *s = s1->priv_data;
	int len, max_packet_size, n;
	/* Calculate the number of bytes to get samples aligned on a byte border */
	int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);

	max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
	/* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
	if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
	return AVERROR(EINVAL);
	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 * 8 / sample_size_bits;
	ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
	n += (s->buf_ptr - s->buf);
	}
	return 0;
	}

	static void rtp_send_mpegaudio(AVFormatContext *s1,
	const uint8_t *buf1, int size)
	{
	RTPMuxContext *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)
	{
	RTPMuxContext *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)
	{
	RTPMuxContext *s = s1->priv_data;
	int len, out_len;

	s->timestamp = s->cur_timestamp;
	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;
	}
	}
	}

	static int rtp_send_ilbc(AVFormatContext s1, const uint8_t buf, int size)
	{
	RTPMuxContext *s = s1->priv_data;
	AVStream *st = s1->streams[0];
	int frame_duration = av_get_audio_frame_duration(st->codec, 0);
	int frame_size = st->codec->block_align;
	int frames = size / frame_size;

	while (frames > 0) {
	if (s->num_frames > 0 &&
	av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
	s1->max_delay, AV_TIME_BASE_Q) >= 0) {
	ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
	s->num_frames = 0;
	}

	if (!s->num_frames) {
	s->buf_ptr = s->buf;
	s->timestamp = s->cur_timestamp;
	}
	memcpy(s->buf_ptr, buf, frame_size);
	frames--;
	s->num_frames++;
	s->buf_ptr += frame_size;
	buf += frame_size;
	s->cur_timestamp += frame_duration;

	if (s->num_frames == s->max_frames_per_packet) {
	ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
	s->num_frames = 0;
	}
	}
	return 0;
	}

	static int rtp_write_packet(AVFormatContext s1, AVPacket pkt)
	{
	RTPMuxContext *s = s1->priv_data;
	AVStream *st = s1->streams[0];
	int rtcp_bytes;
	int size= pkt->size;

	av_log(s1, AV_LOG_TRACE, "%d: write len=%d\n", pkt->stream_index, size);

	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) &&
	(ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
	!(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
	rtcp_send_sr(s1, ff_ntp_time(), 0);
	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 AV_CODEC_ID_PCM_MULAW:
	case AV_CODEC_ID_PCM_ALAW:
	case AV_CODEC_ID_PCM_U8:
	case AV_CODEC_ID_PCM_S8:
	return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
	case AV_CODEC_ID_PCM_U16BE:
	case AV_CODEC_ID_PCM_U16LE:
	case AV_CODEC_ID_PCM_S16BE:
	case AV_CODEC_ID_PCM_S16LE:
	return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
	case AV_CODEC_ID_ADPCM_G722:
	/* The actual sample size is half a byte per sample, but since the
	* stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
	* the correct parameter for send_samples_bits is 8 bits per stream
	* clock. */
	return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
	case AV_CODEC_ID_ADPCM_G726:
	return rtp_send_samples(s1, pkt->data, size,
	st->codec->bits_per_coded_sample * st->codec->channels);
	case AV_CODEC_ID_MP2:
	case AV_CODEC_ID_MP3:
	rtp_send_mpegaudio(s1, pkt->data, size);
	break;
	case AV_CODEC_ID_MPEG1VIDEO:
	case AV_CODEC_ID_MPEG2VIDEO:
	ff_rtp_send_mpegvideo(s1, pkt->data, size);
	break;
	case AV_CODEC_ID_AAC:
	if (s->flags & FF_RTP_FLAG_MP4A_LATM)
	ff_rtp_send_latm(s1, pkt->data, size);
	else
	ff_rtp_send_aac(s1, pkt->data, size);
	break;
	case AV_CODEC_ID_AMR_NB:
	case AV_CODEC_ID_AMR_WB:
	ff_rtp_send_amr(s1, pkt->data, size);
	break;
	case AV_CODEC_ID_MPEG2TS:
	rtp_send_mpegts_raw(s1, pkt->data, size);
	break;
	case AV_CODEC_ID_H264:
	ff_rtp_send_h264_hevc(s1, pkt->data, size);
	break;
	case AV_CODEC_ID_H261:
	ff_rtp_send_h261(s1, pkt->data, size);
	break;
	case AV_CODEC_ID_H263:
	if (s->flags & FF_RTP_FLAG_RFC2190) {
	int mb_info_size = 0;
	const uint8_t *mb_info =
	av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
	&mb_info_size);
	ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
	break;
	}
	/* Fallthrough */
	case AV_CODEC_ID_H263P:
	ff_rtp_send_h263(s1, pkt->data, size);
	break;
	case AV_CODEC_ID_HEVC:
	ff_rtp_send_h264_hevc(s1, pkt->data, size);
	break;
	case AV_CODEC_ID_VORBIS:
	case AV_CODEC_ID_THEORA:
	ff_rtp_send_xiph(s1, pkt->data, size);
	break;
	case AV_CODEC_ID_VP8:
	ff_rtp_send_vp8(s1, pkt->data, size);
	break;
	case AV_CODEC_ID_ILBC:
	rtp_send_ilbc(s1, pkt->data, size);
	break;
	case AV_CODEC_ID_MJPEG:
	ff_rtp_send_jpeg(s1, pkt->data, size);
	break;
	case AV_CODEC_ID_OPUS:
	if (size > s->max_payload_size) {
	av_log(s1, AV_LOG_ERROR,
	"Packet size %d too large for max RTP payload size %d\n",
	size, s->max_payload_size);
	return AVERROR(EINVAL);
	}
	/* Intentional fallthrough */
	default:
	/* better than nothing : send the codec raw data */
	rtp_send_raw(s1, pkt->data, size);
	break;
	}
	return 0;
	}

	static int rtp_write_trailer(AVFormatContext *s1)
	{
	RTPMuxContext *s = s1->priv_data;

	/* If the caller closes and recreates ->pb, this might actually
	* be NULL here even if it was successfully allocated at the start. */
	if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
	rtcp_send_sr(s1, ff_ntp_time(), 1);
	av_freep(&s->buf);

	return 0;
	}

	AVOutputFormat ff_rtp_muxer = {
	.name = "rtp",
	.long_name = NULL_IF_CONFIG_SMALL("RTP output"),
	.priv_data_size = sizeof(RTPMuxContext),
	.audio_codec = AV_CODEC_ID_PCM_MULAW,
	.video_codec = AV_CODEC_ID_MPEG4,
	.write_header = rtp_write_header,
	.write_packet = rtp_write_packet,
	.write_trailer = rtp_write_trailer,
	.priv_class = &rtp_muxer_class,
	.flags = AVFMT_TS_NONSTRICT,
	};