| 1 | /* |
| 2 | * RTSP muxer |
| 3 | * Copyright (c) 2010 Martin Storsjo |
| 4 | * |
| 5 | * This file is part of FFmpeg. |
| 6 | * |
| 7 | * FFmpeg is free software; you can redistribute it and/or |
| 8 | * modify it under the terms of the GNU Lesser General Public |
| 9 | * License as published by the Free Software Foundation; either |
| 10 | * version 2.1 of the License, or (at your option) any later version. |
| 11 | * |
| 12 | * FFmpeg is distributed in the hope that it will be useful, |
| 13 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| 14 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| 15 | * Lesser General Public License for more details. |
| 16 | * |
| 17 | * You should have received a copy of the GNU Lesser General Public |
| 18 | * License along with FFmpeg; if not, write to the Free Software |
| 19 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| 20 | */ |
| 21 | |
| 22 | #include "avformat.h" |
| 23 | |
| 24 | #if HAVE_POLL_H |
| 25 | #include <poll.h> |
| 26 | #endif |
| 27 | #include "network.h" |
| 28 | #include "os_support.h" |
| 29 | #include "rtsp.h" |
| 30 | #include "internal.h" |
| 31 | #include "avio_internal.h" |
| 32 | #include "libavutil/intreadwrite.h" |
| 33 | #include "libavutil/avstring.h" |
| 34 | #include "libavutil/time.h" |
| 35 | #include "url.h" |
| 36 | |
| 37 | #define SDP_MAX_SIZE 16384 |
| 38 | |
| 39 | static const AVClass rtsp_muxer_class = { |
| 40 | .class_name = "RTSP muxer", |
| 41 | .item_name = av_default_item_name, |
| 42 | .option = ff_rtsp_options, |
| 43 | .version = LIBAVUTIL_VERSION_INT, |
| 44 | }; |
| 45 | |
| 46 | int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr) |
| 47 | { |
| 48 | RTSPState *rt = s->priv_data; |
| 49 | RTSPMessageHeader reply1, *reply = &reply1; |
| 50 | int i; |
| 51 | char *sdp; |
| 52 | AVFormatContext sdp_ctx, *ctx_array[1]; |
| 53 | |
| 54 | if (s->start_time_realtime == 0 || s->start_time_realtime == AV_NOPTS_VALUE) |
| 55 | s->start_time_realtime = av_gettime(); |
| 56 | |
| 57 | /* Announce the stream */ |
| 58 | sdp = av_mallocz(SDP_MAX_SIZE); |
| 59 | if (!sdp) |
| 60 | return AVERROR(ENOMEM); |
| 61 | /* We create the SDP based on the RTSP AVFormatContext where we |
| 62 | * aren't allowed to change the filename field. (We create the SDP |
| 63 | * based on the RTSP context since the contexts for the RTP streams |
| 64 | * don't exist yet.) In order to specify a custom URL with the actual |
| 65 | * peer IP instead of the originally specified hostname, we create |
| 66 | * a temporary copy of the AVFormatContext, where the custom URL is set. |
| 67 | * |
| 68 | * FIXME: Create the SDP without copying the AVFormatContext. |
| 69 | * This either requires setting up the RTP stream AVFormatContexts |
| 70 | * already here (complicating things immensely) or getting a more |
| 71 | * flexible SDP creation interface. |
| 72 | */ |
| 73 | sdp_ctx = *s; |
| 74 | ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename), |
| 75 | "rtsp", NULL, addr, -1, NULL); |
| 76 | ctx_array[0] = &sdp_ctx; |
| 77 | if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) { |
| 78 | av_free(sdp); |
| 79 | return AVERROR_INVALIDDATA; |
| 80 | } |
| 81 | av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp); |
| 82 | ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri, |
| 83 | "Content-Type: application/sdp\r\n", |
| 84 | reply, NULL, sdp, strlen(sdp)); |
| 85 | av_free(sdp); |
| 86 | if (reply->status_code != RTSP_STATUS_OK) |
| 87 | return ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA); |
| 88 | |
| 89 | /* Set up the RTSPStreams for each AVStream */ |
| 90 | for (i = 0; i < s->nb_streams; i++) { |
| 91 | RTSPStream *rtsp_st; |
| 92 | |
| 93 | rtsp_st = av_mallocz(sizeof(RTSPStream)); |
| 94 | if (!rtsp_st) |
| 95 | return AVERROR(ENOMEM); |
| 96 | dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st); |
| 97 | |
| 98 | rtsp_st->stream_index = i; |
| 99 | |
| 100 | av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url)); |
| 101 | /* Note, this must match the relative uri set in the sdp content */ |
| 102 | av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url), |
| 103 | "/streamid=%d", i); |
| 104 | } |
| 105 | |
| 106 | return 0; |
| 107 | } |
| 108 | |
| 109 | static int rtsp_write_record(AVFormatContext *s) |
| 110 | { |
| 111 | RTSPState *rt = s->priv_data; |
| 112 | RTSPMessageHeader reply1, *reply = &reply1; |
| 113 | char cmd[1024]; |
| 114 | |
| 115 | snprintf(cmd, sizeof(cmd), |
| 116 | "Range: npt=0.000-\r\n"); |
| 117 | ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL); |
| 118 | if (reply->status_code != RTSP_STATUS_OK) |
| 119 | return ff_rtsp_averror(reply->status_code, -1); |
| 120 | rt->state = RTSP_STATE_STREAMING; |
| 121 | return 0; |
| 122 | } |
| 123 | |
| 124 | static int rtsp_write_header(AVFormatContext *s) |
| 125 | { |
| 126 | int ret; |
| 127 | |
| 128 | ret = ff_rtsp_connect(s); |
| 129 | if (ret) |
| 130 | return ret; |
| 131 | |
| 132 | if (rtsp_write_record(s) < 0) { |
| 133 | ff_rtsp_close_streams(s); |
| 134 | ff_rtsp_close_connections(s); |
| 135 | return AVERROR_INVALIDDATA; |
| 136 | } |
| 137 | return 0; |
| 138 | } |
| 139 | |
| 140 | int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st) |
| 141 | { |
| 142 | RTSPState *rt = s->priv_data; |
| 143 | AVFormatContext *rtpctx = rtsp_st->transport_priv; |
| 144 | uint8_t *buf, *ptr; |
| 145 | int size; |
| 146 | uint8_t *interleave_header, *interleaved_packet; |
| 147 | |
| 148 | size = avio_close_dyn_buf(rtpctx->pb, &buf); |
| 149 | rtpctx->pb = NULL; |
| 150 | ptr = buf; |
| 151 | while (size > 4) { |
| 152 | uint32_t packet_len = AV_RB32(ptr); |
| 153 | int id; |
| 154 | /* The interleaving header is exactly 4 bytes, which happens to be |
| 155 | * the same size as the packet length header from |
| 156 | * ffio_open_dyn_packet_buf. So by writing the interleaving header |
| 157 | * over these bytes, we get a consecutive interleaved packet |
| 158 | * that can be written in one call. */ |
| 159 | interleaved_packet = interleave_header = ptr; |
| 160 | ptr += 4; |
| 161 | size -= 4; |
| 162 | if (packet_len > size || packet_len < 2) |
| 163 | break; |
| 164 | if (RTP_PT_IS_RTCP(ptr[1])) |
| 165 | id = rtsp_st->interleaved_max; /* RTCP */ |
| 166 | else |
| 167 | id = rtsp_st->interleaved_min; /* RTP */ |
| 168 | interleave_header[0] = '$'; |
| 169 | interleave_header[1] = id; |
| 170 | AV_WB16(interleave_header + 2, packet_len); |
| 171 | ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len); |
| 172 | ptr += packet_len; |
| 173 | size -= packet_len; |
| 174 | } |
| 175 | av_free(buf); |
| 176 | return ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE); |
| 177 | } |
| 178 | |
| 179 | static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt) |
| 180 | { |
| 181 | RTSPState *rt = s->priv_data; |
| 182 | RTSPStream *rtsp_st; |
| 183 | int n; |
| 184 | struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0}; |
| 185 | AVFormatContext *rtpctx; |
| 186 | int ret; |
| 187 | |
| 188 | while (1) { |
| 189 | n = poll(&p, 1, 0); |
| 190 | if (n <= 0) |
| 191 | break; |
| 192 | if (p.revents & POLLIN) { |
| 193 | RTSPMessageHeader reply; |
| 194 | |
| 195 | /* Don't let ff_rtsp_read_reply handle interleaved packets, |
| 196 | * since it would block and wait for an RTSP reply on the socket |
| 197 | * (which may not be coming any time soon) if it handles |
| 198 | * interleaved packets internally. */ |
| 199 | ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL); |
| 200 | if (ret < 0) |
| 201 | return AVERROR(EPIPE); |
| 202 | if (ret == 1) |
| 203 | ff_rtsp_skip_packet(s); |
| 204 | /* XXX: parse message */ |
| 205 | if (rt->state != RTSP_STATE_STREAMING) |
| 206 | return AVERROR(EPIPE); |
| 207 | } |
| 208 | } |
| 209 | |
| 210 | if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams) |
| 211 | return AVERROR_INVALIDDATA; |
| 212 | rtsp_st = rt->rtsp_streams[pkt->stream_index]; |
| 213 | rtpctx = rtsp_st->transport_priv; |
| 214 | |
| 215 | ret = ff_write_chained(rtpctx, 0, pkt, s, 0); |
| 216 | /* ff_write_chained does all the RTP packetization. If using TCP as |
| 217 | * transport, rtpctx->pb is only a dyn_packet_buf that queues up the |
| 218 | * packets, so we need to send them out on the TCP connection separately. |
| 219 | */ |
| 220 | if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) |
| 221 | ret = ff_rtsp_tcp_write_packet(s, rtsp_st); |
| 222 | return ret; |
| 223 | } |
| 224 | |
| 225 | static int rtsp_write_close(AVFormatContext *s) |
| 226 | { |
| 227 | RTSPState *rt = s->priv_data; |
| 228 | |
| 229 | // If we want to send RTCP_BYE packets, these are sent by av_write_trailer. |
| 230 | // Thus call this on all streams before doing the teardown. This is |
| 231 | // done within ff_rtsp_undo_setup. |
| 232 | ff_rtsp_undo_setup(s, 1); |
| 233 | |
| 234 | ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL); |
| 235 | |
| 236 | ff_rtsp_close_streams(s); |
| 237 | ff_rtsp_close_connections(s); |
| 238 | ff_network_close(); |
| 239 | return 0; |
| 240 | } |
| 241 | |
| 242 | AVOutputFormat ff_rtsp_muxer = { |
| 243 | .name = "rtsp", |
| 244 | .long_name = NULL_IF_CONFIG_SMALL("RTSP output"), |
| 245 | .priv_data_size = sizeof(RTSPState), |
| 246 | .audio_codec = AV_CODEC_ID_AAC, |
| 247 | .video_codec = AV_CODEC_ID_MPEG4, |
| 248 | .write_header = rtsp_write_header, |
| 249 | .write_packet = rtsp_write_packet, |
| 250 | .write_trailer = rtsp_write_close, |
| 251 | .flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER, |
| 252 | .priv_class = &rtsp_muxer_class, |
| 253 | }; |