/*
* 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];
if (s->start_time_realtime == 0 || s->start_time_realtime == AV_NOPTS_VALUE)
s->start_time_realtime = av_gettime();
/* Announce the stream */
sdp = av_mallocz(SDP_MAX_SIZE);
if (!sdp)
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 ff_rtsp_averror(reply->status_code, 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 ff_rtsp_averror(reply->status_code, -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, 0);
/* 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,
};