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| 1 /* | 1 /* |
| 2 * libjingle | 2 * libjingle |
| 3 * Copyright 2010 Google Inc. | 3 * Copyright 2010 Google Inc. |
| 4 * | 4 * |
| 5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
| 6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
| 7 * | 7 * |
| 8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
| 9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
| 10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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| 82 32, // int medianWaitingTimeMs; | 82 32, // int medianWaitingTimeMs; |
| 83 1, // int minWaitingTimeMs; | 83 1, // int minWaitingTimeMs; |
| 84 98, // int maxWaitingTimeMs; | 84 98, // int maxWaitingTimeMs; |
| 85 7654, // int addedSamples; | 85 7654, // int addedSamples; |
| 86 }; // These random but non-trivial numbers are used for testing. | 86 }; // These random but non-trivial numbers are used for testing. |
| 87 | 87 |
| 88 #define WEBRTC_CHECK_CHANNEL(channel) \ | 88 #define WEBRTC_CHECK_CHANNEL(channel) \ |
| 89 if (channels_.find(channel) == channels_.end()) return -1; | 89 if (channels_.find(channel) == channels_.end()) return -1; |
| 90 | 90 |
| 91 #define WEBRTC_ASSERT_CHANNEL(channel) \ | 91 #define WEBRTC_ASSERT_CHANNEL(channel) \ |
| 92 DCHECK(channels_.find(channel) != channels_.end()); | 92 RTC_DCHECK(channels_.find(channel) != channels_.end()); |
| 93 | 93 |
| 94 // Verify the header extension ID, if enabled, is within the bounds specified in | 94 // Verify the header extension ID, if enabled, is within the bounds specified in |
| 95 // [RFC5285]: 1-14 inclusive. | 95 // [RFC5285]: 1-14 inclusive. |
| 96 #define WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id) \ | 96 #define WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id) \ |
| 97 do { \ | 97 do { \ |
| 98 if (enable && (id < 1 || id > 14)) { \ | 98 if (enable && (id < 1 || id > 14)) { \ |
| 99 return -1; \ | 99 return -1; \ |
| 100 } \ | 100 } \ |
| 101 } while (0); | 101 } while (0); |
| 102 | 102 |
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| 376 std::string packet = channels_[channel]->packets.front(); | 376 std::string packet = channels_[channel]->packets.front(); |
| 377 result = (packet == std::string(static_cast<const char*>(data), len)); | 377 result = (packet == std::string(static_cast<const char*>(data), len)); |
| 378 channels_[channel]->packets.pop_front(); | 378 channels_[channel]->packets.pop_front(); |
| 379 } | 379 } |
| 380 return result; | 380 return result; |
| 381 } | 381 } |
| 382 bool CheckNoPacket(int channel) { | 382 bool CheckNoPacket(int channel) { |
| 383 return channels_[channel]->packets.empty(); | 383 return channels_[channel]->packets.empty(); |
| 384 } | 384 } |
| 385 void TriggerCallbackOnError(int channel_num, int err_code) { | 385 void TriggerCallbackOnError(int channel_num, int err_code) { |
| 386 DCHECK(observer_ != NULL); | 386 RTC_DCHECK(observer_ != NULL); |
| 387 observer_->CallbackOnError(channel_num, err_code); | 387 observer_->CallbackOnError(channel_num, err_code); |
| 388 } | 388 } |
| 389 void set_playout_fail_channel(int channel) { | 389 void set_playout_fail_channel(int channel) { |
| 390 playout_fail_channel_ = channel; | 390 playout_fail_channel_ = channel; |
| 391 } | 391 } |
| 392 void set_send_fail_channel(int channel) { | 392 void set_send_fail_channel(int channel) { |
| 393 send_fail_channel_ = channel; | 393 send_fail_channel_ = channel; |
| 394 } | 394 } |
| 395 void set_fail_start_recording_microphone( | 395 void set_fail_start_recording_microphone( |
| 396 bool fail_start_recording_microphone) { | 396 bool fail_start_recording_microphone) { |
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| 1284 DtmfInfo dtmf_info_; | 1284 DtmfInfo dtmf_info_; |
| 1285 webrtc::VoEMediaProcess* media_processor_; | 1285 webrtc::VoEMediaProcess* media_processor_; |
| 1286 FakeAudioProcessing audio_processing_; | 1286 FakeAudioProcessing audio_processing_; |
| 1287 }; | 1287 }; |
| 1288 | 1288 |
| 1289 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID | 1289 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID |
| 1290 | 1290 |
| 1291 } // namespace cricket | 1291 } // namespace cricket |
| 1292 | 1292 |
| 1293 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ | 1293 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ |
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