OLD | NEW |
1 /* | 1 /* |
2 * libjingle | 2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. |
3 * Copyright 2004 Google Inc. | |
4 * | 3 * |
5 * Redistribution and use in source and binary forms, with or without | 4 * Use of this source code is governed by a BSD-style license |
6 * modification, are permitted provided that the following conditions are met: | 5 * that can be found in the LICENSE file in the root of the source |
7 * | 6 * tree. An additional intellectual property rights grant can be found |
8 * 1. Redistributions of source code must retain the above copyright notice, | 7 * in the file PATENTS. All contributing project authors may |
9 * this list of conditions and the following disclaimer. | 8 * be found in the AUTHORS file in the root of the source tree. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | |
11 * this list of conditions and the following disclaimer in the documentation | |
12 * and/or other materials provided with the distribution. | |
13 * 3. The name of the author may not be used to endorse or promote products | |
14 * derived from this software without specific prior written permission. | |
15 * | |
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED | |
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF | |
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO | |
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, | |
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, | |
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; | |
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, | |
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR | |
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | |
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | |
26 */ | 9 */ |
27 | 10 |
28 #ifndef TALK_MEDIA_BASE_MEDIACHANNEL_H_ | 11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
29 #define TALK_MEDIA_BASE_MEDIACHANNEL_H_ | 12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
30 | 13 |
31 #include <string> | 14 #include <string> |
32 #include <vector> | 15 #include <vector> |
33 | 16 |
34 #include "talk/media/base/codec.h" | |
35 #include "talk/media/base/constants.h" | |
36 #include "talk/media/base/streamparams.h" | |
37 #include "webrtc/base/basictypes.h" | 17 #include "webrtc/base/basictypes.h" |
38 #include "webrtc/base/buffer.h" | 18 #include "webrtc/base/buffer.h" |
39 #include "webrtc/base/dscp.h" | 19 #include "webrtc/base/dscp.h" |
40 #include "webrtc/base/logging.h" | 20 #include "webrtc/base/logging.h" |
41 #include "webrtc/base/optional.h" | 21 #include "webrtc/base/optional.h" |
42 #include "webrtc/base/sigslot.h" | 22 #include "webrtc/base/sigslot.h" |
43 #include "webrtc/base/socket.h" | 23 #include "webrtc/base/socket.h" |
44 #include "webrtc/base/window.h" | 24 #include "webrtc/base/window.h" |
| 25 #include "webrtc/media/base/codec.h" |
| 26 #include "webrtc/media/base/constants.h" |
| 27 #include "webrtc/media/base/streamparams.h" |
45 // TODO(juberti): re-evaluate this include | 28 // TODO(juberti): re-evaluate this include |
46 #include "talk/session/media/audiomonitor.h" | 29 #include "talk/session/media/audiomonitor.h" |
47 | 30 |
48 namespace rtc { | 31 namespace rtc { |
49 class Buffer; | 32 class Buffer; |
50 class RateLimiter; | 33 class RateLimiter; |
51 class Timing; | 34 class Timing; |
52 } | 35 } |
53 | 36 |
54 namespace webrtc { | 37 namespace webrtc { |
(...skipping 1162 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
1217 size_t> SignalDataReceived; | 1200 size_t> SignalDataReceived; |
1218 // Signal when the media channel is ready to send the stream. Arguments are: | 1201 // Signal when the media channel is ready to send the stream. Arguments are: |
1219 // writable(bool) | 1202 // writable(bool) |
1220 sigslot::signal1<bool> SignalReadyToSend; | 1203 sigslot::signal1<bool> SignalReadyToSend; |
1221 // Signal for notifying that the remote side has closed the DataChannel. | 1204 // Signal for notifying that the remote side has closed the DataChannel. |
1222 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; | 1205 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; |
1223 }; | 1206 }; |
1224 | 1207 |
1225 } // namespace cricket | 1208 } // namespace cricket |
1226 | 1209 |
1227 #endif // TALK_MEDIA_BASE_MEDIACHANNEL_H_ | 1210 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
OLD | NEW |