FFmpeg: libavformat/rtspenc.c Source File

FFmpeg

[フレーム]

libavformat

rtspenc.c

Go to the documentation of this file.

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 if (s->start_time_realtime == 0 || s->start_time_realtime == AV_NOPTS_VALUE)

55 s->start_time_realtime = av_gettime();

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);

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

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

91 RTSPStream *rtsp_st;

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);

98 rtsp_st->stream_index = i;

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 };

Generated on Sun Mar 8 2015 02:35:12 for FFmpeg by doxygen 1.8.2