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| 1 /* | 1 /* |
| 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| 11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
| 12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
| 13 | 13 |
| 14 #include <algorithm> | |
| 15 #include <memory> | 14 #include <memory> |
| 16 #include <string> | 15 #include <string> |
| 17 #include <vector> | 16 #include <vector> |
| 18 | 17 |
| 19 #include "webrtc/api/rtpparameters.h" | 18 #include "webrtc/api/rtpparameters.h" |
| 20 #include "webrtc/base/basictypes.h" | 19 #include "webrtc/base/basictypes.h" |
| 21 #include "webrtc/base/buffer.h" | 20 #include "webrtc/base/buffer.h" |
| 22 #include "webrtc/base/copyonwritebuffer.h" | 21 #include "webrtc/base/copyonwritebuffer.h" |
| 23 #include "webrtc/base/dscp.h" | 22 #include "webrtc/base/dscp.h" |
| 24 #include "webrtc/base/logging.h" | 23 #include "webrtc/base/logging.h" |
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| 74 for (size_t i = 0; i < vals.size(); ++i) { | 73 for (size_t i = 0; i < vals.size(); ++i) { |
| 75 if (i > 0) { | 74 if (i > 0) { |
| 76 ost << ", "; | 75 ost << ", "; |
| 77 } | 76 } |
| 78 ost << vals[i].ToString(); | 77 ost << vals[i].ToString(); |
| 79 } | 78 } |
| 80 ost << "]"; | 79 ost << "]"; |
| 81 return ost.str(); | 80 return ost.str(); |
| 82 } | 81 } |
| 83 | 82 |
| 84 template <typename T> | |
| 85 static T MinPositive(T a, T b) { | |
| 86 if (a <= 0) { | |
| 87 return b; | |
| 88 } | |
| 89 if (b <= 0) { | |
| 90 return a; | |
| 91 } | |
| 92 return std::min(a, b); | |
| 93 } | |
| 94 | |
| 95 // Construction-time settings, passed on when creating | 83 // Construction-time settings, passed on when creating |
| 96 // MediaChannels. | 84 // MediaChannels. |
| 97 struct MediaConfig { | 85 struct MediaConfig { |
| 98 // Set DSCP value on packets. This flag comes from the | 86 // Set DSCP value on packets. This flag comes from the |
| 99 // PeerConnection constraint 'googDscp'. | 87 // PeerConnection constraint 'googDscp'. |
| 100 bool enable_dscp = false; | 88 bool enable_dscp = false; |
| 101 | 89 |
| 102 // Video-specific config. | 90 // Video-specific config. |
| 103 struct Video { | 91 struct Video { |
| 104 // Enable WebRTC CPU Overuse Detection. This flag comes from the | 92 // Enable WebRTC CPU Overuse Detection. This flag comes from the |
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| 1215 const char*, | 1203 const char*, |
| 1216 size_t> SignalDataReceived; | 1204 size_t> SignalDataReceived; |
| 1217 // Signal when the media channel is ready to send the stream. Arguments are: | 1205 // Signal when the media channel is ready to send the stream. Arguments are: |
| 1218 // writable(bool) | 1206 // writable(bool) |
| 1219 sigslot::signal1<bool> SignalReadyToSend; | 1207 sigslot::signal1<bool> SignalReadyToSend; |
| 1220 }; | 1208 }; |
| 1221 | 1209 |
| 1222 } // namespace cricket | 1210 } // namespace cricket |
| 1223 | 1211 |
| 1224 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1212 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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