yading@11: /*
yading@11:  * RTSP muxer
yading@11:  * Copyright (c) 2010 Martin Storsjo
yading@11:  *
yading@11:  * This file is part of FFmpeg.
yading@11:  *
yading@11:  * FFmpeg is free software; you can redistribute it and/or
yading@11:  * modify it under the terms of the GNU Lesser General Public
yading@11:  * License as published by the Free Software Foundation; either
yading@11:  * version 2.1 of the License, or (at your option) any later version.
yading@11:  *
yading@11:  * FFmpeg is distributed in the hope that it will be useful,
yading@11:  * but WITHOUT ANY WARRANTY; without even the implied warranty of
yading@11:  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
yading@11:  * Lesser General Public License for more details.
yading@11:  *
yading@11:  * You should have received a copy of the GNU Lesser General Public
yading@11:  * License along with FFmpeg; if not, write to the Free Software
yading@11:  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
yading@11:  */
yading@11: 
yading@11: #include "avformat.h"
yading@11: 
yading@11: #if HAVE_POLL_H
yading@11: #include <poll.h>
yading@11: #endif
yading@11: #include "network.h"
yading@11: #include "os_support.h"
yading@11: #include "rtsp.h"
yading@11: #include "internal.h"
yading@11: #include "avio_internal.h"
yading@11: #include "libavutil/intreadwrite.h"
yading@11: #include "libavutil/avstring.h"
yading@11: #include "libavutil/time.h"
yading@11: #include "url.h"
yading@11: 
yading@11: #define SDP_MAX_SIZE 16384
yading@11: 
yading@11: static const AVClass rtsp_muxer_class = {
yading@11:     .class_name = "RTSP muxer",
yading@11:     .item_name  = av_default_item_name,
yading@11:     .option     = ff_rtsp_options,
yading@11:     .version    = LIBAVUTIL_VERSION_INT,
yading@11: };
yading@11: 
yading@11: int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
yading@11: {
yading@11:     RTSPState *rt = s->priv_data;
yading@11:     RTSPMessageHeader reply1, *reply = &reply1;
yading@11:     int i;
yading@11:     char *sdp;
yading@11:     AVFormatContext sdp_ctx, *ctx_array[1];
yading@11: 
yading@11:     s->start_time_realtime = av_gettime();
yading@11: 
yading@11:     /* Announce the stream */
yading@11:     sdp = av_mallocz(SDP_MAX_SIZE);
yading@11:     if (sdp == NULL)
yading@11:         return AVERROR(ENOMEM);
yading@11:     /* We create the SDP based on the RTSP AVFormatContext where we
yading@11:      * aren't allowed to change the filename field. (We create the SDP
yading@11:      * based on the RTSP context since the contexts for the RTP streams
yading@11:      * don't exist yet.) In order to specify a custom URL with the actual
yading@11:      * peer IP instead of the originally specified hostname, we create
yading@11:      * a temporary copy of the AVFormatContext, where the custom URL is set.
yading@11:      *
yading@11:      * FIXME: Create the SDP without copying the AVFormatContext.
yading@11:      * This either requires setting up the RTP stream AVFormatContexts
yading@11:      * already here (complicating things immensely) or getting a more
yading@11:      * flexible SDP creation interface.
yading@11:      */
yading@11:     sdp_ctx = *s;
yading@11:     ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
yading@11:                 "rtsp", NULL, addr, -1, NULL);
yading@11:     ctx_array[0] = &sdp_ctx;
yading@11:     if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
yading@11:         av_free(sdp);
yading@11:         return AVERROR_INVALIDDATA;
yading@11:     }
yading@11:     av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
yading@11:     ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
yading@11:                                   "Content-Type: application/sdp\r\n",
yading@11:                                   reply, NULL, sdp, strlen(sdp));
yading@11:     av_free(sdp);
yading@11:     if (reply->status_code != RTSP_STATUS_OK)
yading@11:         return AVERROR_INVALIDDATA;
yading@11: 
yading@11:     /* Set up the RTSPStreams for each AVStream */
yading@11:     for (i = 0; i < s->nb_streams; i++) {
yading@11:         RTSPStream *rtsp_st;
yading@11: 
yading@11:         rtsp_st = av_mallocz(sizeof(RTSPStream));
yading@11:         if (!rtsp_st)
yading@11:             return AVERROR(ENOMEM);
yading@11:         dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
yading@11: 
yading@11:         rtsp_st->stream_index = i;
yading@11: 
yading@11:         av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
yading@11:         /* Note, this must match the relative uri set in the sdp content */
yading@11:         av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
yading@11:                     "/streamid=%d", i);
yading@11:     }
yading@11: 
yading@11:     return 0;
yading@11: }
yading@11: 
yading@11: static int rtsp_write_record(AVFormatContext *s)
yading@11: {
yading@11:     RTSPState *rt = s->priv_data;
yading@11:     RTSPMessageHeader reply1, *reply = &reply1;
yading@11:     char cmd[1024];
yading@11: 
yading@11:     snprintf(cmd, sizeof(cmd),
yading@11:              "Range: npt=0.000-\r\n");
yading@11:     ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
yading@11:     if (reply->status_code != RTSP_STATUS_OK)
yading@11:         return -1;
yading@11:     rt->state = RTSP_STATE_STREAMING;
yading@11:     return 0;
yading@11: }
yading@11: 
yading@11: static int rtsp_write_header(AVFormatContext *s)
yading@11: {
yading@11:     int ret;
yading@11: 
yading@11:     ret = ff_rtsp_connect(s);
yading@11:     if (ret)
yading@11:         return ret;
yading@11: 
yading@11:     if (rtsp_write_record(s) < 0) {
yading@11:         ff_rtsp_close_streams(s);
yading@11:         ff_rtsp_close_connections(s);
yading@11:         return AVERROR_INVALIDDATA;
yading@11:     }
yading@11:     return 0;
yading@11: }
yading@11: 
yading@11: static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
yading@11: {
yading@11:     RTSPState *rt = s->priv_data;
yading@11:     AVFormatContext *rtpctx = rtsp_st->transport_priv;
yading@11:     uint8_t *buf, *ptr;
yading@11:     int size;
yading@11:     uint8_t *interleave_header, *interleaved_packet;
yading@11: 
yading@11:     size = avio_close_dyn_buf(rtpctx->pb, &buf);
yading@11:     ptr = buf;
yading@11:     while (size > 4) {
yading@11:         uint32_t packet_len = AV_RB32(ptr);
yading@11:         int id;
yading@11:         /* The interleaving header is exactly 4 bytes, which happens to be
yading@11:          * the same size as the packet length header from
yading@11:          * ffio_open_dyn_packet_buf. So by writing the interleaving header
yading@11:          * over these bytes, we get a consecutive interleaved packet
yading@11:          * that can be written in one call. */
yading@11:         interleaved_packet = interleave_header = ptr;
yading@11:         ptr += 4;
yading@11:         size -= 4;
yading@11:         if (packet_len > size || packet_len < 2)
yading@11:             break;
yading@11:         if (RTP_PT_IS_RTCP(ptr[1]))
yading@11:             id = rtsp_st->interleaved_max; /* RTCP */
yading@11:         else
yading@11:             id = rtsp_st->interleaved_min; /* RTP */
yading@11:         interleave_header[0] = '$';
yading@11:         interleave_header[1] = id;
yading@11:         AV_WB16(interleave_header + 2, packet_len);
yading@11:         ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
yading@11:         ptr += packet_len;
yading@11:         size -= packet_len;
yading@11:     }
yading@11:     av_free(buf);
yading@11:     ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
yading@11:     return 0;
yading@11: }
yading@11: 
yading@11: static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
yading@11: {
yading@11:     RTSPState *rt = s->priv_data;
yading@11:     RTSPStream *rtsp_st;
yading@11:     int n;
yading@11:     struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
yading@11:     AVFormatContext *rtpctx;
yading@11:     int ret;
yading@11: 
yading@11:     while (1) {
yading@11:         n = poll(&p, 1, 0);
yading@11:         if (n <= 0)
yading@11:             break;
yading@11:         if (p.revents & POLLIN) {
yading@11:             RTSPMessageHeader reply;
yading@11: 
yading@11:             /* Don't let ff_rtsp_read_reply handle interleaved packets,
yading@11:              * since it would block and wait for an RTSP reply on the socket
yading@11:              * (which may not be coming any time soon) if it handles
yading@11:              * interleaved packets internally. */
yading@11:             ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
yading@11:             if (ret < 0)
yading@11:                 return AVERROR(EPIPE);
yading@11:             if (ret == 1)
yading@11:                 ff_rtsp_skip_packet(s);
yading@11:             /* XXX: parse message */
yading@11:             if (rt->state != RTSP_STATE_STREAMING)
yading@11:                 return AVERROR(EPIPE);
yading@11:         }
yading@11:     }
yading@11: 
yading@11:     if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)
yading@11:         return AVERROR_INVALIDDATA;
yading@11:     rtsp_st = rt->rtsp_streams[pkt->stream_index];
yading@11:     rtpctx = rtsp_st->transport_priv;
yading@11: 
yading@11:     ret = ff_write_chained(rtpctx, 0, pkt, s);
yading@11:     /* ff_write_chained does all the RTP packetization. If using TCP as
yading@11:      * transport, rtpctx->pb is only a dyn_packet_buf that queues up the
yading@11:      * packets, so we need to send them out on the TCP connection separately.
yading@11:      */
yading@11:     if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
yading@11:         ret = tcp_write_packet(s, rtsp_st);
yading@11:     return ret;
yading@11: }
yading@11: 
yading@11: static int rtsp_write_close(AVFormatContext *s)
yading@11: {
yading@11:     RTSPState *rt = s->priv_data;
yading@11: 
yading@11:     ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
yading@11: 
yading@11:     ff_rtsp_close_streams(s);
yading@11:     ff_rtsp_close_connections(s);
yading@11:     ff_network_close();
yading@11:     return 0;
yading@11: }
yading@11: 
yading@11: AVOutputFormat ff_rtsp_muxer = {
yading@11:     .name              = "rtsp",
yading@11:     .long_name         = NULL_IF_CONFIG_SMALL("RTSP output"),
yading@11:     .priv_data_size    = sizeof(RTSPState),
yading@11:     .audio_codec       = AV_CODEC_ID_AAC,
yading@11:     .video_codec       = AV_CODEC_ID_MPEG4,
yading@11:     .write_header      = rtsp_write_header,
yading@11:     .write_packet      = rtsp_write_packet,
yading@11:     .write_trailer     = rtsp_write_close,
yading@11:     .flags             = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
yading@11:     .priv_class        = &rtsp_muxer_class,
yading@11: };