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 "libavcodec/bitstream.h"
  #include "avformat.h"
  #include "mpegts.h"
  
  #include <unistd.h>
  #include "network.h"
  
  #include "rtpenc.h"
  
  //#define DEBUG
  
  #define RTCP_SR_SIZE 28
  #define NTP_OFFSET 2208988800ULL
  #define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL)
  
  static uint64_t ntp_time(void)
  {
    return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US;
  }
  
  static int rtp_write_header(AVFormatContext *s1)
  {
      RTPMuxContext *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 = ff_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->buf = av_malloc(max_packet_size);
      if (s->buf == NULL) {
          return AVERROR(ENOMEM);
      }
      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, (AVRational){1, 1000000}, 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->num_frames = 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)
  {
      RTPMuxContext *s = s1->priv_data;
      uint32_t rtp_ts;
  
      dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
  
      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, (AVRational){1, 1000000},
                            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)
  {
      RTPMuxContext *s = s1->priv_data;
  
      dprintf(s1, "rtp_send_data size=%d\n", len);
  
      /* 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)
  {
      RTPMuxContext *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)
  {
      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;
          }
      }
  }
  
  /* write an RTP packet. 'buf1' must contain a single specific frame. */
  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;
      uint8_t *buf1= pkt->data;
  
      dprintf(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) &&
                             (ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
          rtcp_send_sr(s1, 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 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;
      case CODEC_ID_H264:
          ff_rtp_send_h264(s1, buf1, size);
          break;
      default:
          /* better than nothing : send the codec raw data */
          rtp_send_raw(s1, buf1, size);
          break;
      }
      return 0;
  }
  
  static int rtp_write_trailer(AVFormatContext *s1)
  {
      RTPMuxContext *s = s1->priv_data;
  
      av_freep(&s->buf);
  
      return 0;
  }
  
  AVOutputFormat rtp_muxer = {
      "rtp",
      NULL_IF_CONFIG_SMALL("RTP output format"),
      NULL,
      NULL,
      sizeof(RTPMuxContext),
      CODEC_ID_PCM_MULAW,
      CODEC_ID_NONE,
      rtp_write_header,
      rtp_write_packet,
      rtp_write_trailer,
  };
  