| /* |
| * 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" |
| |
| |
| 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]; |
| char url[MAX_URL_SIZE]; |
| |
| 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; |
| sdp_ctx.url = url; |
| ff_url_join(url, sizeof(url), |
| "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[MAX_URL_SIZE]; |
| |
| 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, |
| }; |