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 39static 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 46int 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 == NULL) 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 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 109static 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 -1; 120 rt->state = RTSP_STATE_STREAMING; 121 return 0; 122} 123 124static 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 140int 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 179static 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); 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 225static 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 242AVOutputFormat 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}; 254