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249 lines
7.8 KiB
249 lines
7.8 KiB
/* |
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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, 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); |
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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 ff_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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