/*
  * 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,

 };