/*
  * RTSP muxer
  * Copyright (c) 2010 Martin Storsjo
  *
  * 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"
 

 #if HAVE_POLL_H
 #include <poll.h>

 #endif
 #include "network.h"

 #include "os_support.h"

 #include "rtsp.h"

 #include "internal.h"

 #include "avio_internal.h"

 #include "libavutil/intreadwrite.h"

 #include "libavutil/avstring.h"

 #include "libavutil/time.h"

 #include "url.h"

 
 #define SDP_MAX_SIZE 16384
 

 static const AVClass rtsp_muxer_class = {
     .class_name = "RTSP muxer",
     .item_name  = av_default_item_name,

     .option     = ff_rtsp_options,

     .version    = LIBAVUTIL_VERSION_INT,
 };
 

 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
 {
     RTSPState *rt = s->priv_data;
     RTSPMessageHeader reply1, *reply = &reply1;
     int i;
     char *sdp;
     AVFormatContext sdp_ctx, *ctx_array[1];
 
     s->start_time_realtime = av_gettime();
 
     /* Announce the stream */
     sdp = av_mallocz(SDP_MAX_SIZE);
     if (sdp == NULL)
         return AVERROR(ENOMEM);
     /* We create the SDP based on the RTSP AVFormatContext where we
      * aren't allowed to change the filename field. (We create the SDP
      * based on the RTSP context since the contexts for the RTP streams
      * don't exist yet.) In order to specify a custom URL with the actual
      * peer IP instead of the originally specified hostname, we create
      * a temporary copy of the AVFormatContext, where the custom URL is set.
      *
      * FIXME: Create the SDP without copying the AVFormatContext.
      * This either requires setting up the RTP stream AVFormatContexts
      * already here (complicating things immensely) or getting a more
      * flexible SDP creation interface.
      */
     sdp_ctx = *s;
     ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
                 "rtsp", NULL, addr, -1, NULL);
     ctx_array[0] = &sdp_ctx;

     if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {

         av_free(sdp);
         return AVERROR_INVALIDDATA;
     }
     av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
     ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
                                   "Content-Type: application/sdp\r\n",
                                   reply, NULL, sdp, strlen(sdp));
     av_free(sdp);
     if (reply->status_code != RTSP_STATUS_OK)
         return AVERROR_INVALIDDATA;
 
     /* Set up the RTSPStreams for each AVStream */
     for (i = 0; i < s->nb_streams; i++) {
         RTSPStream *rtsp_st;
 
         rtsp_st = av_mallocz(sizeof(RTSPStream));
         if (!rtsp_st)
             return AVERROR(ENOMEM);
         dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
 
         rtsp_st->stream_index = i;
 
         av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
         /* Note, this must match the relative uri set in the sdp content */
         av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
                     "/streamid=%d", i);
     }
 
     return 0;
 }

 
 static int rtsp_write_record(AVFormatContext *s)
 {
     RTSPState *rt = s->priv_data;
     RTSPMessageHeader reply1, *reply = &reply1;
     char cmd[1024];
 
     snprintf(cmd, sizeof(cmd),

              "Range: npt=0.000-\r\n");

     ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);

     if (reply->status_code != RTSP_STATUS_OK)
         return -1;
     rt->state = RTSP_STATE_STREAMING;
     return 0;
 }
 
 static int rtsp_write_header(AVFormatContext *s)
 {
     int ret;
 

     ret = ff_rtsp_connect(s);

     if (ret)
         return ret;
 
     if (rtsp_write_record(s) < 0) {

         ff_rtsp_close_streams(s);

         ff_rtsp_close_connections(s);

         return AVERROR_INVALIDDATA;
     }
     return 0;
 }
 

 int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)

 {
     RTSPState *rt = s->priv_data;
     AVFormatContext *rtpctx = rtsp_st->transport_priv;
     uint8_t *buf, *ptr;
     int size;

     uint8_t *interleave_header, *interleaved_packet;

     size = avio_close_dyn_buf(rtpctx->pb, &buf);

     rtpctx->pb = NULL;

     ptr = buf;
     while (size > 4) {
         uint32_t packet_len = AV_RB32(ptr);
         int id;

         /* The interleaving header is exactly 4 bytes, which happens to be
          * the same size as the packet length header from

          * ffio_open_dyn_packet_buf. So by writing the interleaving header

          * over these bytes, we get a consecutive interleaved packet
          * that can be written in one call. */
         interleaved_packet = interleave_header = ptr;

         ptr += 4;
         size -= 4;
         if (packet_len > size || packet_len < 2)
             break;

         if (RTP_PT_IS_RTCP(ptr[1]))

             id = rtsp_st->interleaved_max; /* RTCP */
         else
             id = rtsp_st->interleaved_min; /* RTP */
         interleave_header[0] = '$';
         interleave_header[1] = id;
         AV_WB16(interleave_header + 2, packet_len);

         ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);

         ptr += packet_len;
         size -= packet_len;
     }
     av_free(buf);

     return ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);

 }
 

 static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
 {
     RTSPState *rt = s->priv_data;
     RTSPStream *rtsp_st;

     int n;

     struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};

     AVFormatContext *rtpctx;

     int ret;

     while (1) {

         n = poll(&p, 1, 0);

         if (n <= 0)
             break;

         if (p.revents & POLLIN) {

             RTSPMessageHeader reply;
 

             /* Don't let ff_rtsp_read_reply handle interleaved packets,
              * since it would block and wait for an RTSP reply on the socket
              * (which may not be coming any time soon) if it handles
              * interleaved packets internally. */

             ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);

             if (ret < 0)

                 return AVERROR(EPIPE);

             if (ret == 1)
                 ff_rtsp_skip_packet(s);

             /* XXX: parse message */
             if (rt->state != RTSP_STATE_STREAMING)
                 return AVERROR(EPIPE);
         }
     }
 
     if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)
         return AVERROR_INVALIDDATA;
     rtsp_st = rt->rtsp_streams[pkt->stream_index];
     rtpctx = rtsp_st->transport_priv;
 

     ret = ff_write_chained(rtpctx, 0, pkt, s);
     /* ff_write_chained does all the RTP packetization. If using TCP as

      * transport, rtpctx->pb is only a dyn_packet_buf that queues up the
      * packets, so we need to send them out on the TCP connection separately.
      */
     if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)

         ret = ff_rtsp_tcp_write_packet(s, rtsp_st);

     return ret;

 }
 
 static int rtsp_write_close(AVFormatContext *s)
 {
     RTSPState *rt = s->priv_data;
 

     // If we want to send RTCP_BYE packets, these are sent by av_write_trailer.
     // Thus call this on all streams before doing the teardown. This is
     // done within ff_rtsp_undo_setup.
     ff_rtsp_undo_setup(s, 1);
 

     ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);

     ff_rtsp_close_streams(s);

     ff_rtsp_close_connections(s);

     ff_network_close();

     return 0;
 }
 

 AVOutputFormat ff_rtsp_muxer = {

     .name              = "rtsp",

     .long_name         = NULL_IF_CONFIG_SMALL("RTSP output"),

     .priv_data_size    = sizeof(RTSPState),

     .audio_codec       = AV_CODEC_ID_AAC,
     .video_codec       = AV_CODEC_ID_MPEG4,

     .write_header      = rtsp_write_header,
     .write_packet      = rtsp_write_packet,
     .write_trailer     = rtsp_write_close,

     .flags             = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
     .priv_class        = &rtsp_muxer_class,

 };