mirror of
https://git.videolan.org/git/ffmpeg.git
synced 2024-08-21 08:45:06 +02:00
250 lines
7.8 KiB
C
250 lines
7.8 KiB
C
/*
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* RTSP muxer
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* Copyright (c) 2010 Martin Storsjo
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avformat.h"
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#include <sys/time.h>
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#if HAVE_SYS_SELECT_H
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#include <sys/select.h>
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#endif
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#include "network.h"
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#include "rtsp.h"
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#include "internal.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/avstring.h"
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#define SDP_MAX_SIZE 16384
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int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
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{
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RTSPState *rt = s->priv_data;
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RTSPMessageHeader reply1, *reply = &reply1;
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int i;
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char *sdp;
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AVFormatContext sdp_ctx, *ctx_array[1];
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s->start_time_realtime = av_gettime();
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/* Announce the stream */
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sdp = av_mallocz(SDP_MAX_SIZE);
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if (sdp == NULL)
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return AVERROR(ENOMEM);
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/* We create the SDP based on the RTSP AVFormatContext where we
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* aren't allowed to change the filename field. (We create the SDP
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* based on the RTSP context since the contexts for the RTP streams
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* don't exist yet.) In order to specify a custom URL with the actual
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* peer IP instead of the originally specified hostname, we create
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* a temporary copy of the AVFormatContext, where the custom URL is set.
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*
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* FIXME: Create the SDP without copying the AVFormatContext.
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* This either requires setting up the RTP stream AVFormatContexts
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* already here (complicating things immensely) or getting a more
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* flexible SDP creation interface.
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*/
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sdp_ctx = *s;
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ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
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"rtsp", NULL, addr, -1, NULL);
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ctx_array[0] = &sdp_ctx;
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if (avf_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
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av_free(sdp);
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return AVERROR_INVALIDDATA;
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}
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av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
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ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
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"Content-Type: application/sdp\r\n",
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reply, NULL, sdp, strlen(sdp));
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av_free(sdp);
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if (reply->status_code != RTSP_STATUS_OK)
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return AVERROR_INVALIDDATA;
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/* Set up the RTSPStreams for each AVStream */
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for (i = 0; i < s->nb_streams; i++) {
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RTSPStream *rtsp_st;
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AVStream *st = s->streams[i];
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rtsp_st = av_mallocz(sizeof(RTSPStream));
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if (!rtsp_st)
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return AVERROR(ENOMEM);
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dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
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st->priv_data = rtsp_st;
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rtsp_st->stream_index = i;
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av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
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/* Note, this must match the relative uri set in the sdp content */
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av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
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"/streamid=%d", i);
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}
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return 0;
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}
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static int rtsp_write_record(AVFormatContext *s)
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{
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RTSPState *rt = s->priv_data;
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RTSPMessageHeader reply1, *reply = &reply1;
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char cmd[1024];
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snprintf(cmd, sizeof(cmd),
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"Range: npt=%0.3f-\r\n",
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(double) 0);
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ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
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if (reply->status_code != RTSP_STATUS_OK)
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return -1;
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rt->state = RTSP_STATE_STREAMING;
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return 0;
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}
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static int rtsp_write_header(AVFormatContext *s)
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{
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int ret;
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ret = ff_rtsp_connect(s);
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if (ret)
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return ret;
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if (rtsp_write_record(s) < 0) {
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ff_rtsp_close_streams(s);
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ff_rtsp_close_connections(s);
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return AVERROR_INVALIDDATA;
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}
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return 0;
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}
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static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
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{
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RTSPState *rt = s->priv_data;
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AVFormatContext *rtpctx = rtsp_st->transport_priv;
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uint8_t *buf, *ptr;
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int size;
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uint8_t *interleave_header, *interleaved_packet;
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size = url_close_dyn_buf(rtpctx->pb, &buf);
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ptr = buf;
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while (size > 4) {
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uint32_t packet_len = AV_RB32(ptr);
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int id;
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/* The interleaving header is exactly 4 bytes, which happens to be
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* the same size as the packet length header from
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* url_open_dyn_packet_buf. So by writing the interleaving header
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* over these bytes, we get a consecutive interleaved packet
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* that can be written in one call. */
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interleaved_packet = interleave_header = ptr;
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ptr += 4;
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size -= 4;
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if (packet_len > size || packet_len < 2)
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break;
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if (ptr[1] >= RTCP_SR && ptr[1] <= RTCP_APP)
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id = rtsp_st->interleaved_max; /* RTCP */
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else
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id = rtsp_st->interleaved_min; /* RTP */
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interleave_header[0] = '$';
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interleave_header[1] = id;
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AV_WB16(interleave_header + 2, packet_len);
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url_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
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ptr += packet_len;
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size -= packet_len;
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}
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av_free(buf);
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url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
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return 0;
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}
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static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
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{
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RTSPState *rt = s->priv_data;
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RTSPStream *rtsp_st;
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fd_set rfds;
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int n, tcp_fd;
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struct timeval tv;
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AVFormatContext *rtpctx;
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int ret;
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tcp_fd = url_get_file_handle(rt->rtsp_hd);
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while (1) {
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FD_ZERO(&rfds);
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FD_SET(tcp_fd, &rfds);
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tv.tv_sec = 0;
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tv.tv_usec = 0;
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n = select(tcp_fd + 1, &rfds, NULL, NULL, &tv);
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if (n <= 0)
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break;
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if (FD_ISSET(tcp_fd, &rfds)) {
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RTSPMessageHeader reply;
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/* Don't let ff_rtsp_read_reply handle interleaved packets,
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* since it would block and wait for an RTSP reply on the socket
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* (which may not be coming any time soon) if it handles
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* interleaved packets internally. */
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ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
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if (ret < 0)
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return AVERROR(EPIPE);
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if (ret == 1)
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ff_rtsp_skip_packet(s);
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/* XXX: parse message */
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if (rt->state != RTSP_STATE_STREAMING)
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return AVERROR(EPIPE);
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}
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}
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if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)
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return AVERROR_INVALIDDATA;
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rtsp_st = rt->rtsp_streams[pkt->stream_index];
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rtpctx = rtsp_st->transport_priv;
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ret = ff_write_chained(rtpctx, 0, pkt, s);
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/* ff_write_chained does all the RTP packetization. If using TCP as
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* transport, rtpctx->pb is only a dyn_packet_buf that queues up the
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* packets, so we need to send them out on the TCP connection separately.
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*/
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if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
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ret = tcp_write_packet(s, rtsp_st);
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return ret;
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}
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static int rtsp_write_close(AVFormatContext *s)
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{
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RTSPState *rt = s->priv_data;
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ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
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ff_rtsp_close_streams(s);
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ff_rtsp_close_connections(s);
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ff_network_close();
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return 0;
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}
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AVOutputFormat rtsp_muxer = {
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"rtsp",
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NULL_IF_CONFIG_SMALL("RTSP output format"),
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NULL,
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NULL,
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sizeof(RTSPState),
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CODEC_ID_AAC,
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CODEC_ID_MPEG4,
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rtsp_write_header,
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rtsp_write_packet,
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rtsp_write_close,
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.flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
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};
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