text: /*
   * 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"
  
  //#define DEBUG
  
  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 },
      { 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_H263:
      case AV_CODEC_ID_H263P:
      case AV_CODEC_ID_H264:
      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;
      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)
          /* Round the NTP time to whole milliseconds. */
          s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
                                   NTP_OFFSET_US;
  
      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 == NULL) {
          return AVERROR(ENOMEM);
      }
      s->max_payload_size = s1->packet_size - 12;
  
      s->max_frames_per_packet = 0;
      if (s1->max_delay > 0) {
          if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
              int frame_size = av_get_audio_frame_duration(st->codec, 0);
              if (!frame_size)
                  frame_size = st->codec->frame_size;
              if (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_q_rnd(s1->max_delay,
                                           AV_TIME_BASE_Q,
                                           (AVRational){ frame_size, st->codec->sample_rate },
                                           AV_ROUND_DOWN);
              }
          }
          if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
              /* FIXME: We should round down here... */
              s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
          }
      }
  
      avpriv_set_pts_info(st, 32, 1, 90000);
      switch(st->codec->codec_id) {
      case AV_CODEC_ID_MP2:
      case AV_CODEC_ID_MP3:
          s->buf_ptr = s->buf + 4;
          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;
          s->buf_ptr = s->buf;
          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_VORBIS:
      case AV_CODEC_ID_THEORA:
          if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
          s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
          s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
          s->num_frames = 0;
          goto defaultcase;
      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;
          }
          if (!s->max_frames_per_packet)
              s->max_frames_per_packet = 1;
          s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
                                           s->max_payload_size / st->codec->block_align);
          goto defaultcase;
      case AV_CODEC_ID_AMR_NB:
      case AV_CODEC_ID_AMR_WB:
          if (!s->max_frames_per_packet)
              s->max_frames_per_packet = 12;
          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;
          }
      case AV_CODEC_ID_AAC:
          s->num_frames = 0;
      default:
  defaultcase:
          if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
              avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
          }
          s->buf_ptr = s->buf;
          break;
      }
  
      return 0;
  
  fail:
      av_freep(&s->buf);
      return AVERROR(EINVAL);
  }
  
  /* send an rtcp sender report packet */
  static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
  {
      RTPMuxContext *s = s1->priv_data;
      uint32_t rtp_ts;
  
      av_dlog(s1, "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_wb32(s1->pb, ntp_time / 1000000);
      avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
      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);
      }
  
      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_dlog(s1, "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->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;
  
      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) {
          int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
  
          if (!s->num_frames) {
              s->buf_ptr = s->buf;
              s->timestamp = s->cur_timestamp;
          }
          memcpy(s->buf_ptr, buf, n * frame_size);
          frames           -= n;
          s->num_frames    += n;
          s->buf_ptr       += n * frame_size;
          buf              += n * frame_size;
          s->cur_timestamp += n * 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_dlog(s1, "%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());
          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(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_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;
  
      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,
  };
  