/*
  * 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, {.dbl = -1 }, -1, 127, 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 CodecID id)
 {
     switch(id) {

     case CODEC_ID_H263:
     case CODEC_ID_H263P:

     case CODEC_ID_H264:
     case CODEC_ID_MPEG1VIDEO:
     case CODEC_ID_MPEG2VIDEO:
     case CODEC_ID_MPEG4:
     case CODEC_ID_AAC:
     case CODEC_ID_MP2:
     case CODEC_ID_MP3:
     case CODEC_ID_PCM_ALAW:
     case CODEC_ID_PCM_MULAW:
     case CODEC_ID_PCM_S8:
     case CODEC_ID_PCM_S16BE:
     case CODEC_ID_PCM_S16LE:
     case CODEC_ID_PCM_U16BE:
     case CODEC_ID_PCM_U16LE:
     case CODEC_ID_PCM_U8:
     case CODEC_ID_MPEG2TS:

     case CODEC_ID_AMR_NB:
     case CODEC_ID_AMR_WB:

     case CODEC_ID_VORBIS:
     case CODEC_ID_THEORA:

     case CODEC_ID_VP8:

     case CODEC_ID_ADPCM_G722:

     case CODEC_ID_ADPCM_G726:

         return 1;
     default:
         return 0;
     }
 }
 

 static int rtp_write_header(AVFormatContext *s1)
 {

     RTPMuxContext *s = s1->priv_data;

     int max_packet_size, n;

     AVStream *st;
 
     if (s1->nb_streams != 1)
         return -1;
     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)
         s->payload_type = ff_rtp_get_payload_type(s1, st->codec);

     s->base_timestamp = av_get_random_seed();

     s->timestamp = s->base_timestamp;
     s->cur_timestamp = 0;

     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;

     max_packet_size = s1->pb->max_packet_size;

     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 == AVMEDIA_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 * (int64_t)st->codec->frame_size, 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 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_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 CODEC_ID_VORBIS:
     case 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 CODEC_ID_VP8:

         av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
                                  "incompatible with the latest spec drafts.\n");

         break;

     case 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 CODEC_ID_AMR_NB:
     case CODEC_ID_AMR_WB:
         if (!s->max_frames_per_packet)
             s->max_frames_per_packet = 12;
         if (st->codec->codec_id == 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");
             return -1;
         }
         if (st->codec->channels != 1) {
             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
             return -1;
         }

     case 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;
 }
 
 /* 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);

     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 void 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)

         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 * 8 / sample_size_bits;

         ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
         n += (s->buf_ptr - s->buf);
     }
 }
 
 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_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))) {
         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 CODEC_ID_PCM_MULAW:
     case CODEC_ID_PCM_ALAW:
     case CODEC_ID_PCM_U8:
     case CODEC_ID_PCM_S8:

         rtp_send_samples(s1, pkt->data, size, 8 * 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, pkt->data, size, 16 * st->codec->channels);

         break;

     case 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. */
         rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);

         break;

     case CODEC_ID_ADPCM_G726:
         rtp_send_samples(s1, pkt->data, size,
                          st->codec->bits_per_coded_sample * st->codec->channels);

         break;

     case CODEC_ID_MP2:
     case CODEC_ID_MP3:

         rtp_send_mpegaudio(s1, pkt->data, size);

         break;
     case CODEC_ID_MPEG1VIDEO:
     case CODEC_ID_MPEG2VIDEO:

         ff_rtp_send_mpegvideo(s1, pkt->data, size);

         break;
     case 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 CODEC_ID_AMR_NB:
     case CODEC_ID_AMR_WB:

         ff_rtp_send_amr(s1, pkt->data, size);

         break;

     case CODEC_ID_MPEG2TS:

         rtp_send_mpegts_raw(s1, pkt->data, size);

         break;

     case CODEC_ID_H264:

         ff_rtp_send_h264(s1, pkt->data, size);

         break;

     case CODEC_ID_H263:
     case CODEC_ID_H263P:

         ff_rtp_send_h263(s1, pkt->data, size);

         break;

     case CODEC_ID_VORBIS:
     case CODEC_ID_THEORA:
         ff_rtp_send_xiph(s1, pkt->data, size);
         break;

     case CODEC_ID_VP8:
         ff_rtp_send_vp8(s1, pkt->data, size);
         break;

     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 format"),
     .priv_data_size    = sizeof(RTPMuxContext),
     .audio_codec       = CODEC_ID_PCM_MULAW,

     .video_codec       = CODEC_ID_MPEG4,

     .write_header      = rtp_write_header,
     .write_packet      = rtp_write_packet,
     .write_trailer     = rtp_write_trailer,

     .priv_class = &rtp_muxer_class,

 };