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  1. /*
  2.  * RTP output format
  3.  * Copyright (c) 2002 Fabrice Bellard
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  7.  * FFmpeg is free software; you can redistribute it and/or
  8.  * modify it under the terms of the GNU Lesser General Public
  9.  * License as published by the Free Software Foundation; either
  10.  * version 2.1 of the License, or (at your option) any later version.
  11.  *
  12.  * FFmpeg is distributed in the hope that it will be useful,
  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15.  * Lesser General Public License for more details.
  16.  *
  17.  * You should have received a copy of the GNU Lesser General Public
  18.  * License along with FFmpeg; if not, write to the Free Software
  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20.  */
  21.  
  22. #include "avformat.h"
  23. #include "mpegts.h"
  24. #include "internal.h"
  25. #include "libavutil/mathematics.h"
  26. #include "libavutil/random_seed.h"
  27. #include "libavutil/opt.h"
  28.  
  29. #include "rtpenc.h"
  30.  
  31. static const AVOption options[] = {
  32.     FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
  33.     { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
  34.     { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
  35.     { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
  36.     { "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
  37.     { NULL },
  38. };
  39.  
  40. static const AVClass rtp_muxer_class = {
  41.     .class_name = "RTP muxer",
  42.     .item_name  = av_default_item_name,
  43.     .option     = options,
  44.     .version    = LIBAVUTIL_VERSION_INT,
  45. };
  46.  
  47. #define RTCP_SR_SIZE 28
  48.  
  49. static int is_supported(enum AVCodecID id)
  50. {
  51.     switch(id) {
  52.     case AV_CODEC_ID_H263:
  53.     case AV_CODEC_ID_H263P:
  54.     case AV_CODEC_ID_H264:
  55.     case AV_CODEC_ID_MPEG1VIDEO:
  56.     case AV_CODEC_ID_MPEG2VIDEO:
  57.     case AV_CODEC_ID_MPEG4:
  58.     case AV_CODEC_ID_AAC:
  59.     case AV_CODEC_ID_MP2:
  60.     case AV_CODEC_ID_MP3:
  61.     case AV_CODEC_ID_PCM_ALAW:
  62.     case AV_CODEC_ID_PCM_MULAW:
  63.     case AV_CODEC_ID_PCM_S8:
  64.     case AV_CODEC_ID_PCM_S16BE:
  65.     case AV_CODEC_ID_PCM_S16LE:
  66.     case AV_CODEC_ID_PCM_U16BE:
  67.     case AV_CODEC_ID_PCM_U16LE:
  68.     case AV_CODEC_ID_PCM_U8:
  69.     case AV_CODEC_ID_MPEG2TS:
  70.     case AV_CODEC_ID_AMR_NB:
  71.     case AV_CODEC_ID_AMR_WB:
  72.     case AV_CODEC_ID_VORBIS:
  73.     case AV_CODEC_ID_THEORA:
  74.     case AV_CODEC_ID_VP8:
  75.     case AV_CODEC_ID_ADPCM_G722:
  76.     case AV_CODEC_ID_ADPCM_G726:
  77.     case AV_CODEC_ID_ILBC:
  78.     case AV_CODEC_ID_MJPEG:
  79.     case AV_CODEC_ID_SPEEX:
  80.     case AV_CODEC_ID_OPUS:
  81.         return 1;
  82.     default:
  83.         return 0;
  84.     }
  85. }
  86.  
  87. static int rtp_write_header(AVFormatContext *s1)
  88. {
  89.     RTPMuxContext *s = s1->priv_data;
  90.     int n;
  91.     AVStream *st;
  92.  
  93.     if (s1->nb_streams != 1) {
  94.         av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
  95.         return AVERROR(EINVAL);
  96.     }
  97.     st = s1->streams[0];
  98.     if (!is_supported(st->codec->codec_id)) {
  99.         av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
  100.  
  101.         return -1;
  102.     }
  103.  
  104.     if (s->payload_type < 0) {
  105.         /* Re-validate non-dynamic payload types */
  106.         if (st->id < RTP_PT_PRIVATE)
  107.             st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
  108.  
  109.         s->payload_type = st->id;
  110.     } else {
  111.         /* private option takes priority */
  112.         st->id = s->payload_type;
  113.     }
  114.  
  115.     s->base_timestamp = av_get_random_seed();
  116.     s->timestamp = s->base_timestamp;
  117.     s->cur_timestamp = 0;
  118.     if (!s->ssrc)
  119.         s->ssrc = av_get_random_seed();
  120.     s->first_packet = 1;
  121.     s->first_rtcp_ntp_time = ff_ntp_time();
  122.     if (s1->start_time_realtime)
  123.         /* Round the NTP time to whole milliseconds. */
  124.         s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
  125.                                  NTP_OFFSET_US;
  126.     // Pick a random sequence start number, but in the lower end of the
  127.     // available range, so that any wraparound doesn't happen immediately.
  128.     // (Immediate wraparound would be an issue for SRTP.)
  129.     if (s->seq < 0) {
  130.         if (st->codec->flags & CODEC_FLAG_BITEXACT) {
  131.             s->seq = 0;
  132.         } else
  133.             s->seq = av_get_random_seed() & 0x0fff;
  134.     } else
  135.         s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
  136.  
  137.     if (s1->packet_size) {
  138.         if (s1->pb->max_packet_size)
  139.             s1->packet_size = FFMIN(s1->packet_size,
  140.                                     s1->pb->max_packet_size);
  141.     } else
  142.         s1->packet_size = s1->pb->max_packet_size;
  143.     if (s1->packet_size <= 12) {
  144.         av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
  145.         return AVERROR(EIO);
  146.     }
  147.     s->buf = av_malloc(s1->packet_size);
  148.     if (s->buf == NULL) {
  149.         return AVERROR(ENOMEM);
  150.     }
  151.     s->max_payload_size = s1->packet_size - 12;
  152.  
  153.     s->max_frames_per_packet = 0;
  154.     if (s1->max_delay > 0) {
  155.         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
  156.             int frame_size = av_get_audio_frame_duration(st->codec, 0);
  157.             if (!frame_size)
  158.                 frame_size = st->codec->frame_size;
  159.             if (frame_size == 0) {
  160.                 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
  161.             } else {
  162.                 s->max_frames_per_packet =
  163.                         av_rescale_q_rnd(s1->max_delay,
  164.                                          AV_TIME_BASE_Q,
  165.                                          (AVRational){ frame_size, st->codec->sample_rate },
  166.                                          AV_ROUND_DOWN);
  167.             }
  168.         }
  169.         if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
  170.             /* FIXME: We should round down here... */
  171.             s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
  172.         }
  173.     }
  174.  
  175.     avpriv_set_pts_info(st, 32, 1, 90000);
  176.     switch(st->codec->codec_id) {
  177.     case AV_CODEC_ID_MP2:
  178.     case AV_CODEC_ID_MP3:
  179.         s->buf_ptr = s->buf + 4;
  180.         break;
  181.     case AV_CODEC_ID_MPEG1VIDEO:
  182.     case AV_CODEC_ID_MPEG2VIDEO:
  183.         break;
  184.     case AV_CODEC_ID_MPEG2TS:
  185.         n = s->max_payload_size / TS_PACKET_SIZE;
  186.         if (n < 1)
  187.             n = 1;
  188.         s->max_payload_size = n * TS_PACKET_SIZE;
  189.         s->buf_ptr = s->buf;
  190.         break;
  191.     case AV_CODEC_ID_H264:
  192.         /* check for H.264 MP4 syntax */
  193.         if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
  194.             s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
  195.         }
  196.         break;
  197.     case AV_CODEC_ID_VORBIS:
  198.     case AV_CODEC_ID_THEORA:
  199.         if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
  200.         s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
  201.         s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
  202.         s->num_frames = 0;
  203.         goto defaultcase;
  204.     case AV_CODEC_ID_ADPCM_G722:
  205.         /* Due to a historical error, the clock rate for G722 in RTP is
  206.          * 8000, even if the sample rate is 16000. See RFC 3551. */
  207.         avpriv_set_pts_info(st, 32, 1, 8000);
  208.         break;
  209.     case AV_CODEC_ID_OPUS:
  210.         if (st->codec->channels > 2) {
  211.             av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
  212.             goto fail;
  213.         }
  214.         /* The opus RTP RFC says that all opus streams should use 48000 Hz
  215.          * as clock rate, since all opus sample rates can be expressed in
  216.          * this clock rate, and sample rate changes on the fly are supported. */
  217.         avpriv_set_pts_info(st, 32, 1, 48000);
  218.         break;
  219.     case AV_CODEC_ID_ILBC:
  220.         if (st->codec->block_align != 38 && st->codec->block_align != 50) {
  221.             av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
  222.             goto fail;
  223.         }
  224.         if (!s->max_frames_per_packet)
  225.             s->max_frames_per_packet = 1;
  226.         s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
  227.                                          s->max_payload_size / st->codec->block_align);
  228.         goto defaultcase;
  229.     case AV_CODEC_ID_AMR_NB:
  230.     case AV_CODEC_ID_AMR_WB:
  231.         if (!s->max_frames_per_packet)
  232.             s->max_frames_per_packet = 12;
  233.         if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
  234.             n = 31;
  235.         else
  236.             n = 61;
  237.         /* max_header_toc_size + the largest AMR payload must fit */
  238.         if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
  239.             av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
  240.             goto fail;
  241.         }
  242.         if (st->codec->channels != 1) {
  243.             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
  244.             goto fail;
  245.         }
  246.     case AV_CODEC_ID_AAC:
  247.         s->num_frames = 0;
  248.     default:
  249. defaultcase:
  250.         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
  251.             avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
  252.         }
  253.         s->buf_ptr = s->buf;
  254.         break;
  255.     }
  256.  
  257.     return 0;
  258.  
  259. fail:
  260.     av_freep(&s->buf);
  261.     return AVERROR(EINVAL);
  262. }
  263.  
  264. /* send an rtcp sender report packet */
  265. static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
  266. {
  267.     RTPMuxContext *s = s1->priv_data;
  268.     uint32_t rtp_ts;
  269.  
  270.     av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
  271.  
  272.     s->last_rtcp_ntp_time = ntp_time;
  273.     rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
  274.                           s1->streams[0]->time_base) + s->base_timestamp;
  275.     avio_w8(s1->pb, (RTP_VERSION << 6));
  276.     avio_w8(s1->pb, RTCP_SR);
  277.     avio_wb16(s1->pb, 6); /* length in words - 1 */
  278.     avio_wb32(s1->pb, s->ssrc);
  279.     avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time));
  280.     avio_wb32(s1->pb, rtp_ts);
  281.     avio_wb32(s1->pb, s->packet_count);
  282.     avio_wb32(s1->pb, s->octet_count);
  283.  
  284.     if (s->cname) {
  285.         int len = FFMIN(strlen(s->cname), 255);
  286.         avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
  287.         avio_w8(s1->pb, RTCP_SDES);
  288.         avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
  289.  
  290.         avio_wb32(s1->pb, s->ssrc);
  291.         avio_w8(s1->pb, 0x01); /* CNAME */
  292.         avio_w8(s1->pb, len);
  293.         avio_write(s1->pb, s->cname, len);
  294.         avio_w8(s1->pb, 0); /* END */
  295.         for (len = (7 + len) % 4; len % 4; len++)
  296.             avio_w8(s1->pb, 0);
  297.     }
  298.  
  299.     avio_flush(s1->pb);
  300. }
  301.  
  302. /* send an rtp packet. sequence number is incremented, but the caller
  303.    must update the timestamp itself */
  304. void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
  305. {
  306.     RTPMuxContext *s = s1->priv_data;
  307.  
  308.     av_dlog(s1, "rtp_send_data size=%d\n", len);
  309.  
  310.     /* build the RTP header */
  311.     avio_w8(s1->pb, (RTP_VERSION << 6));
  312.     avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
  313.     avio_wb16(s1->pb, s->seq);
  314.     avio_wb32(s1->pb, s->timestamp);
  315.     avio_wb32(s1->pb, s->ssrc);
  316.  
  317.     avio_write(s1->pb, buf1, len);
  318.     avio_flush(s1->pb);
  319.  
  320.     s->seq = (s->seq + 1) & 0xffff;
  321.     s->octet_count += len;
  322.     s->packet_count++;
  323. }
  324.  
  325. /* send an integer number of samples and compute time stamp and fill
  326.    the rtp send buffer before sending. */
  327. static int rtp_send_samples(AVFormatContext *s1,
  328.                             const uint8_t *buf1, int size, int sample_size_bits)
  329. {
  330.     RTPMuxContext *s = s1->priv_data;
  331.     int len, max_packet_size, n;
  332.     /* Calculate the number of bytes to get samples aligned on a byte border */
  333.     int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
  334.  
  335.     max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
  336.     /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
  337.     if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
  338.         return AVERROR(EINVAL);
  339.     n = 0;
  340.     while (size > 0) {
  341.         s->buf_ptr = s->buf;
  342.         len = FFMIN(max_packet_size, size);
  343.  
  344.         /* copy data */
  345.         memcpy(s->buf_ptr, buf1, len);
  346.         s->buf_ptr += len;
  347.         buf1 += len;
  348.         size -= len;
  349.         s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
  350.         ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
  351.         n += (s->buf_ptr - s->buf);
  352.     }
  353.     return 0;
  354. }
  355.  
  356. static void rtp_send_mpegaudio(AVFormatContext *s1,
  357.                                const uint8_t *buf1, int size)
  358. {
  359.     RTPMuxContext *s = s1->priv_data;
  360.     int len, count, max_packet_size;
  361.  
  362.     max_packet_size = s->max_payload_size;
  363.  
  364.     /* test if we must flush because not enough space */
  365.     len = (s->buf_ptr - s->buf);
  366.     if ((len + size) > max_packet_size) {
  367.         if (len > 4) {
  368.             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
  369.             s->buf_ptr = s->buf + 4;
  370.         }
  371.     }
  372.     if (s->buf_ptr == s->buf + 4) {
  373.         s->timestamp = s->cur_timestamp;
  374.     }
  375.  
  376.     /* add the packet */
  377.     if (size > max_packet_size) {
  378.         /* big packet: fragment */
  379.         count = 0;
  380.         while (size > 0) {
  381.             len = max_packet_size - 4;
  382.             if (len > size)
  383.                 len = size;
  384.             /* build fragmented packet */
  385.             s->buf[0] = 0;
  386.             s->buf[1] = 0;
  387.             s->buf[2] = count >> 8;
  388.             s->buf[3] = count;
  389.             memcpy(s->buf + 4, buf1, len);
  390.             ff_rtp_send_data(s1, s->buf, len + 4, 0);
  391.             size -= len;
  392.             buf1 += len;
  393.             count += len;
  394.         }
  395.     } else {
  396.         if (s->buf_ptr == s->buf + 4) {
  397.             /* no fragmentation possible */
  398.             s->buf[0] = 0;
  399.             s->buf[1] = 0;
  400.             s->buf[2] = 0;
  401.             s->buf[3] = 0;
  402.         }
  403.         memcpy(s->buf_ptr, buf1, size);
  404.         s->buf_ptr += size;
  405.     }
  406. }
  407.  
  408. static void rtp_send_raw(AVFormatContext *s1,
  409.                          const uint8_t *buf1, int size)
  410. {
  411.     RTPMuxContext *s = s1->priv_data;
  412.     int len, max_packet_size;
  413.  
  414.     max_packet_size = s->max_payload_size;
  415.  
  416.     while (size > 0) {
  417.         len = max_packet_size;
  418.         if (len > size)
  419.             len = size;
  420.  
  421.         s->timestamp = s->cur_timestamp;
  422.         ff_rtp_send_data(s1, buf1, len, (len == size));
  423.  
  424.         buf1 += len;
  425.         size -= len;
  426.     }
  427. }
  428.  
  429. /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
  430. static void rtp_send_mpegts_raw(AVFormatContext *s1,
  431.                                 const uint8_t *buf1, int size)
  432. {
  433.     RTPMuxContext *s = s1->priv_data;
  434.     int len, out_len;
  435.  
  436.     while (size >= TS_PACKET_SIZE) {
  437.         len = s->max_payload_size - (s->buf_ptr - s->buf);
  438.         if (len > size)
  439.             len = size;
  440.         memcpy(s->buf_ptr, buf1, len);
  441.         buf1 += len;
  442.         size -= len;
  443.         s->buf_ptr += len;
  444.  
  445.         out_len = s->buf_ptr - s->buf;
  446.         if (out_len >= s->max_payload_size) {
  447.             ff_rtp_send_data(s1, s->buf, out_len, 0);
  448.             s->buf_ptr = s->buf;
  449.         }
  450.     }
  451. }
  452.  
  453. static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
  454. {
  455.     RTPMuxContext *s = s1->priv_data;
  456.     AVStream *st = s1->streams[0];
  457.     int frame_duration = av_get_audio_frame_duration(st->codec, 0);
  458.     int frame_size = st->codec->block_align;
  459.     int frames = size / frame_size;
  460.  
  461.     while (frames > 0) {
  462.         int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
  463.  
  464.         if (!s->num_frames) {
  465.             s->buf_ptr = s->buf;
  466.             s->timestamp = s->cur_timestamp;
  467.         }
  468.         memcpy(s->buf_ptr, buf, n * frame_size);
  469.         frames           -= n;
  470.         s->num_frames    += n;
  471.         s->buf_ptr       += n * frame_size;
  472.         buf              += n * frame_size;
  473.         s->cur_timestamp += n * frame_duration;
  474.  
  475.         if (s->num_frames == s->max_frames_per_packet) {
  476.             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
  477.             s->num_frames = 0;
  478.         }
  479.     }
  480.     return 0;
  481. }
  482.  
  483. static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
  484. {
  485.     RTPMuxContext *s = s1->priv_data;
  486.     AVStream *st = s1->streams[0];
  487.     int rtcp_bytes;
  488.     int size= pkt->size;
  489.  
  490.     av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
  491.  
  492.     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
  493.         RTCP_TX_RATIO_DEN;
  494.     if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
  495.                             (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
  496.         !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
  497.         rtcp_send_sr(s1, ff_ntp_time());
  498.         s->last_octet_count = s->octet_count;
  499.         s->first_packet = 0;
  500.     }
  501.     s->cur_timestamp = s->base_timestamp + pkt->pts;
  502.  
  503.     switch(st->codec->codec_id) {
  504.     case AV_CODEC_ID_PCM_MULAW:
  505.     case AV_CODEC_ID_PCM_ALAW:
  506.     case AV_CODEC_ID_PCM_U8:
  507.     case AV_CODEC_ID_PCM_S8:
  508.         return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
  509.     case AV_CODEC_ID_PCM_U16BE:
  510.     case AV_CODEC_ID_PCM_U16LE:
  511.     case AV_CODEC_ID_PCM_S16BE:
  512.     case AV_CODEC_ID_PCM_S16LE:
  513.         return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
  514.     case AV_CODEC_ID_ADPCM_G722:
  515.         /* The actual sample size is half a byte per sample, but since the
  516.          * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
  517.          * the correct parameter for send_samples_bits is 8 bits per stream
  518.          * clock. */
  519.         return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
  520.     case AV_CODEC_ID_ADPCM_G726:
  521.         return rtp_send_samples(s1, pkt->data, size,
  522.                                 st->codec->bits_per_coded_sample * st->codec->channels);
  523.     case AV_CODEC_ID_MP2:
  524.     case AV_CODEC_ID_MP3:
  525.         rtp_send_mpegaudio(s1, pkt->data, size);
  526.         break;
  527.     case AV_CODEC_ID_MPEG1VIDEO:
  528.     case AV_CODEC_ID_MPEG2VIDEO:
  529.         ff_rtp_send_mpegvideo(s1, pkt->data, size);
  530.         break;
  531.     case AV_CODEC_ID_AAC:
  532.         if (s->flags & FF_RTP_FLAG_MP4A_LATM)
  533.             ff_rtp_send_latm(s1, pkt->data, size);
  534.         else
  535.             ff_rtp_send_aac(s1, pkt->data, size);
  536.         break;
  537.     case AV_CODEC_ID_AMR_NB:
  538.     case AV_CODEC_ID_AMR_WB:
  539.         ff_rtp_send_amr(s1, pkt->data, size);
  540.         break;
  541.     case AV_CODEC_ID_MPEG2TS:
  542.         rtp_send_mpegts_raw(s1, pkt->data, size);
  543.         break;
  544.     case AV_CODEC_ID_H264:
  545.         ff_rtp_send_h264(s1, pkt->data, size);
  546.         break;
  547.     case AV_CODEC_ID_H263:
  548.         if (s->flags & FF_RTP_FLAG_RFC2190) {
  549.             int mb_info_size = 0;
  550.             const uint8_t *mb_info =
  551.                 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
  552.                                         &mb_info_size);
  553.             ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
  554.             break;
  555.         }
  556.         /* Fallthrough */
  557.     case AV_CODEC_ID_H263P:
  558.         ff_rtp_send_h263(s1, pkt->data, size);
  559.         break;
  560.     case AV_CODEC_ID_VORBIS:
  561.     case AV_CODEC_ID_THEORA:
  562.         ff_rtp_send_xiph(s1, pkt->data, size);
  563.         break;
  564.     case AV_CODEC_ID_VP8:
  565.         ff_rtp_send_vp8(s1, pkt->data, size);
  566.         break;
  567.     case AV_CODEC_ID_ILBC:
  568.         rtp_send_ilbc(s1, pkt->data, size);
  569.         break;
  570.     case AV_CODEC_ID_MJPEG:
  571.         ff_rtp_send_jpeg(s1, pkt->data, size);
  572.         break;
  573.     case AV_CODEC_ID_OPUS:
  574.         if (size > s->max_payload_size) {
  575.             av_log(s1, AV_LOG_ERROR,
  576.                    "Packet size %d too large for max RTP payload size %d\n",
  577.                    size, s->max_payload_size);
  578.             return AVERROR(EINVAL);
  579.         }
  580.         /* Intentional fallthrough */
  581.     default:
  582.         /* better than nothing : send the codec raw data */
  583.         rtp_send_raw(s1, pkt->data, size);
  584.         break;
  585.     }
  586.     return 0;
  587. }
  588.  
  589. static int rtp_write_trailer(AVFormatContext *s1)
  590. {
  591.     RTPMuxContext *s = s1->priv_data;
  592.  
  593.     av_freep(&s->buf);
  594.  
  595.     return 0;
  596. }
  597.  
  598. AVOutputFormat ff_rtp_muxer = {
  599.     .name              = "rtp",
  600.     .long_name         = NULL_IF_CONFIG_SMALL("RTP output"),
  601.     .priv_data_size    = sizeof(RTPMuxContext),
  602.     .audio_codec       = AV_CODEC_ID_PCM_MULAW,
  603.     .video_codec       = AV_CODEC_ID_MPEG4,
  604.     .write_header      = rtp_write_header,
  605.     .write_packet      = rtp_write_packet,
  606.     .write_trailer     = rtp_write_trailer,
  607.     .priv_class        = &rtp_muxer_class,
  608. };
  609.