1/* 2 * RTSP muxer 3 * Copyright (c) 2010 Martin Storsjo 4 * 5 * This file is part of Libav. 6 * 7 * Libav 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 * Libav 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 Libav; 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#include <sys/time.h> 25#if HAVE_POLL_H 26#include <poll.h> 27#endif 28#include "network.h" 29#include "os_support.h" 30#include "rtsp.h" 31#include "internal.h" 32#include "avio_internal.h" 33#include "libavutil/intreadwrite.h" 34#include "libavutil/avstring.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 s->start_time_realtime = av_gettime(); 55 56 /* Announce the stream */ 57 sdp = av_mallocz(SDP_MAX_SIZE); 58 if (sdp == NULL) 59 return AVERROR(ENOMEM); 60 /* We create the SDP based on the RTSP AVFormatContext where we 61 * aren't allowed to change the filename field. (We create the SDP 62 * based on the RTSP context since the contexts for the RTP streams 63 * don't exist yet.) In order to specify a custom URL with the actual 64 * peer IP instead of the originally specified hostname, we create 65 * a temporary copy of the AVFormatContext, where the custom URL is set. 66 * 67 * FIXME: Create the SDP without copying the AVFormatContext. 68 * This either requires setting up the RTP stream AVFormatContexts 69 * already here (complicating things immensely) or getting a more 70 * flexible SDP creation interface. 71 */ 72 sdp_ctx = *s; 73 ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename), 74 "rtsp", NULL, addr, -1, NULL); 75 ctx_array[0] = &sdp_ctx; 76 if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) { 77 av_free(sdp); 78 return AVERROR_INVALIDDATA; 79 } 80 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp); 81 ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri, 82 "Content-Type: application/sdp\r\n", 83 reply, NULL, sdp, strlen(sdp)); 84 av_free(sdp); 85 if (reply->status_code != RTSP_STATUS_OK) 86 return AVERROR_INVALIDDATA; 87 88 /* Set up the RTSPStreams for each AVStream */ 89 for (i = 0; i < s->nb_streams; i++) { 90 RTSPStream *rtsp_st; 91 92 rtsp_st = av_mallocz(sizeof(RTSPStream)); 93 if (!rtsp_st) 94 return AVERROR(ENOMEM); 95 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st); 96 97 rtsp_st->stream_index = i; 98 99 av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url)); 100 /* Note, this must match the relative uri set in the sdp content */ 101 av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url), 102 "/streamid=%d", i); 103 } 104 105 return 0; 106} 107 108static int rtsp_write_record(AVFormatContext *s) 109{ 110 RTSPState *rt = s->priv_data; 111 RTSPMessageHeader reply1, *reply = &reply1; 112 char cmd[1024]; 113 114 snprintf(cmd, sizeof(cmd), 115 "Range: npt=0.000-\r\n"); 116 ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL); 117 if (reply->status_code != RTSP_STATUS_OK) 118 return -1; 119 rt->state = RTSP_STATE_STREAMING; 120 return 0; 121} 122 123static int rtsp_write_header(AVFormatContext *s) 124{ 125 int ret; 126 127 ret = ff_rtsp_connect(s); 128 if (ret) 129 return ret; 130 131 if (rtsp_write_record(s) < 0) { 132 ff_rtsp_close_streams(s); 133 ff_rtsp_close_connections(s); 134 return AVERROR_INVALIDDATA; 135 } 136 return 0; 137} 138 139static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st) 140{ 141 RTSPState *rt = s->priv_data; 142 AVFormatContext *rtpctx = rtsp_st->transport_priv; 143 uint8_t *buf, *ptr; 144 int size; 145 uint8_t *interleave_header, *interleaved_packet; 146 147 size = avio_close_dyn_buf(rtpctx->pb, &buf); 148 ptr = buf; 149 while (size > 4) { 150 uint32_t packet_len = AV_RB32(ptr); 151 int id; 152 /* The interleaving header is exactly 4 bytes, which happens to be 153 * the same size as the packet length header from 154 * ffio_open_dyn_packet_buf. So by writing the interleaving header 155 * over these bytes, we get a consecutive interleaved packet 156 * that can be written in one call. */ 157 interleaved_packet = interleave_header = ptr; 158 ptr += 4; 159 size -= 4; 160 if (packet_len > size || packet_len < 2) 161 break; 162 if (ptr[1] >= RTCP_SR && ptr[1] <= RTCP_APP) 163 id = rtsp_st->interleaved_max; /* RTCP */ 164 else 165 id = rtsp_st->interleaved_min; /* RTP */ 166 interleave_header[0] = '$'; 167 interleave_header[1] = id; 168 AV_WB16(interleave_header + 2, packet_len); 169 ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len); 170 ptr += packet_len; 171 size -= packet_len; 172 } 173 av_free(buf); 174 ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE); 175 return 0; 176} 177 178static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt) 179{ 180 RTSPState *rt = s->priv_data; 181 RTSPStream *rtsp_st; 182 int n; 183 struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0}; 184 AVFormatContext *rtpctx; 185 int ret; 186 187 while (1) { 188 n = poll(&p, 1, 0); 189 if (n <= 0) 190 break; 191 if (p.revents & POLLIN) { 192 RTSPMessageHeader reply; 193 194 /* Don't let ff_rtsp_read_reply handle interleaved packets, 195 * since it would block and wait for an RTSP reply on the socket 196 * (which may not be coming any time soon) if it handles 197 * interleaved packets internally. */ 198 ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL); 199 if (ret < 0) 200 return AVERROR(EPIPE); 201 if (ret == 1) 202 ff_rtsp_skip_packet(s); 203 /* XXX: parse message */ 204 if (rt->state != RTSP_STATE_STREAMING) 205 return AVERROR(EPIPE); 206 } 207 } 208 209 if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams) 210 return AVERROR_INVALIDDATA; 211 rtsp_st = rt->rtsp_streams[pkt->stream_index]; 212 rtpctx = rtsp_st->transport_priv; 213 214 ret = ff_write_chained(rtpctx, 0, pkt, s); 215 /* ff_write_chained does all the RTP packetization. If using TCP as 216 * transport, rtpctx->pb is only a dyn_packet_buf that queues up the 217 * packets, so we need to send them out on the TCP connection separately. 218 */ 219 if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) 220 ret = tcp_write_packet(s, rtsp_st); 221 return ret; 222} 223 224static int rtsp_write_close(AVFormatContext *s) 225{ 226 RTSPState *rt = s->priv_data; 227 228 ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL); 229 230 ff_rtsp_close_streams(s); 231 ff_rtsp_close_connections(s); 232 ff_network_close(); 233 return 0; 234} 235 236AVOutputFormat ff_rtsp_muxer = { 237 .name = "rtsp", 238 .long_name = NULL_IF_CONFIG_SMALL("RTSP output format"), 239 .priv_data_size = sizeof(RTSPState), 240 .audio_codec = CODEC_ID_AAC, 241 .video_codec = CODEC_ID_MPEG4, 242 .write_header = rtsp_write_header, 243 .write_packet = rtsp_write_packet, 244 .write_trailer = rtsp_write_close, 245 .flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER, 246 .priv_class = &rtsp_muxer_class, 247}; 248 249