mirror of
https://git.videolan.org/git/ffmpeg.git
synced 2024-08-28 20:26:14 +02:00
256 lines
8.4 KiB
C
256 lines
8.4 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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#if HAVE_POLL_H
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#include <poll.h>
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#endif
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#include "network.h"
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#include "os_support.h"
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#include "rtsp.h"
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#include "internal.h"
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#include "avio_internal.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/avstring.h"
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#include "libavutil/time.h"
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#include "url.h"
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#define SDP_MAX_SIZE 16384
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static const AVClass rtsp_muxer_class = {
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.class_name = "RTSP muxer",
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.item_name = av_default_item_name,
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.option = ff_rtsp_options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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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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char url[1024];
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if (s->start_time_realtime == 0 || s->start_time_realtime == AV_NOPTS_VALUE)
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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)
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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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sdp_ctx.url = url;
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ff_url_join(url, sizeof(url),
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"rtsp", NULL, addr, -1, NULL);
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ctx_array[0] = &sdp_ctx;
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if (av_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 ff_rtsp_averror(reply->status_code, 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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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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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=0.000-\r\n");
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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 ff_rtsp_averror(reply->status_code, -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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int ff_rtsp_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 = avio_close_dyn_buf(rtpctx->pb, &buf);
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rtpctx->pb = NULL;
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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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* ffio_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 (RTP_PT_IS_RTCP(ptr[1]))
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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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ffurl_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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return ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
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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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int n;
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struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
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AVFormatContext *rtpctx;
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int ret;
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while (1) {
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n = poll(&p, 1, 0);
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if (n <= 0)
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break;
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if (p.revents & POLLIN) {
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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, NULL);
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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, 0);
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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 = ff_rtsp_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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// If we want to send RTCP_BYE packets, these are sent by av_write_trailer.
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// Thus call this on all streams before doing the teardown. This is
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// done within ff_rtsp_undo_setup.
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ff_rtsp_undo_setup(s, 1);
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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 ff_rtsp_muxer = {
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.name = "rtsp",
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.long_name = NULL_IF_CONFIG_SMALL("RTSP output"),
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.priv_data_size = sizeof(RTSPState),
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.audio_codec = AV_CODEC_ID_AAC,
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.video_codec = AV_CODEC_ID_MPEG4,
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.write_header = rtsp_write_header,
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.write_packet = rtsp_write_packet,
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.write_trailer = rtsp_write_close,
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.flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
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.priv_class = &rtsp_muxer_class,
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};
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