FFmpeg: libavformat/rtspenc.c Source File

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libavformat

rtspenc.c

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1 /*

2 * RTSP muxer

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 */

22 #include "avformat.h"

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"

37 #define SDP_MAX_SIZE 16384

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 };

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];

54 s->start_time_realtime = av_gettime();

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;

88 /* Set up the RTSPStreams for each AVStream */

89 for (i = 0; i < s->nb_streams; i++) {

90 RTSPStream *rtsp_st;

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);

97 rtsp_st->stream_index = i;

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

108 static 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

123 static 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

139 static 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 rtpctx->pb = NULL;

149 ptr = buf;

150 while (size > 4) {

151 uint32_t packet_len = AV_RB32(ptr);

152 int id;

153 /* The interleaving header is exactly 4 bytes, which happens to be

154 * the same size as the packet length header from

155 * ffio_open_dyn_packet_buf. So by writing the interleaving header

156 * over these bytes, we get a consecutive interleaved packet

157 * that can be written in one call. */

158 interleaved_packet = interleave_header = ptr;

159 ptr += 4;

160 size -= 4;

161 if (packet_len > size || packet_len < 2)

162 break;

163 if (RTP_PT_IS_RTCP(ptr[1]))

164 id = rtsp_st->interleaved_max; /* RTCP */

165 else

166 id = rtsp_st->interleaved_min; /* RTP */

167 interleave_header[0] = '$';

168 interleave_header[1] = id;

169 AV_WB16(interleave_header + 2, packet_len);

170 ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);

171 ptr += packet_len;

172 size -= packet_len;

173 }

174 av_free(buf);

175 return ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);

176 }

177

178 static 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

224 static 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

236 AVOutputFormat ff_rtsp_muxer = {

237 .name = "rtsp",

238 .long_name = NULL_IF_CONFIG_SMALL("RTSP output"),

239 .priv_data_size = sizeof(RTSPState),

240 .audio_codec = AV_CODEC_ID_AAC,

241 .video_codec = AV_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 };

Generated on Sat Jan 25 2014 19:52:05 for FFmpeg by doxygen 1.8.2