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| 1 /* | 1 /* |
| 2 * libjingle | 2 * libjingle |
| 3 * Copyright 2015 Google Inc. | 3 * Copyright 2015 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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| 33 #include "webrtc/base/checks.h" | 33 #include "webrtc/base/checks.h" |
| 34 #include "webrtc/base/gunit.h" | 34 #include "webrtc/base/gunit.h" |
| 35 | 35 |
| 36 namespace cricket { | 36 namespace cricket { |
| 37 FakeAudioReceiveStream::FakeAudioReceiveStream( | 37 FakeAudioReceiveStream::FakeAudioReceiveStream( |
| 38 const webrtc::AudioReceiveStream::Config& config) | 38 const webrtc::AudioReceiveStream::Config& config) |
| 39 : config_(config), received_packets_(0) { | 39 : config_(config), received_packets_(0) { |
| 40 RTC_DCHECK(config.voe_channel_id != -1); | 40 RTC_DCHECK(config.voe_channel_id != -1); |
| 41 } | 41 } |
| 42 | 42 |
| 43 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const { |
| 44 return webrtc::AudioReceiveStream::Stats(); |
| 45 } |
| 46 |
| 43 const webrtc::AudioReceiveStream::Config& | 47 const webrtc::AudioReceiveStream::Config& |
| 44 FakeAudioReceiveStream::GetConfig() const { | 48 FakeAudioReceiveStream::GetConfig() const { |
| 45 return config_; | 49 return config_; |
| 46 } | 50 } |
| 47 | 51 |
| 48 void FakeAudioReceiveStream::SetStats( | |
| 49 const webrtc::AudioReceiveStream::Stats& stats) { | |
| 50 stats_ = stats; | |
| 51 } | |
| 52 | |
| 53 void FakeAudioReceiveStream::IncrementReceivedPackets() { | 52 void FakeAudioReceiveStream::IncrementReceivedPackets() { |
| 54 received_packets_++; | 53 received_packets_++; |
| 55 } | 54 } |
| 56 | 55 |
| 57 FakeVideoSendStream::FakeVideoSendStream( | 56 FakeVideoSendStream::FakeVideoSendStream( |
| 58 const webrtc::VideoSendStream::Config& config, | 57 const webrtc::VideoSendStream::Config& config, |
| 59 const webrtc::VideoEncoderConfig& encoder_config) | 58 const webrtc::VideoEncoderConfig& encoder_config) |
| 60 : sending_(false), | 59 : sending_(false), |
| 61 config_(config), | 60 config_(config), |
| 62 codec_settings_set_(false), | 61 codec_settings_set_(false), |
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| 365 } | 364 } |
| 366 | 365 |
| 367 void FakeCall::SignalNetworkState(webrtc::NetworkState state) { | 366 void FakeCall::SignalNetworkState(webrtc::NetworkState state) { |
| 368 network_state_ = state; | 367 network_state_ = state; |
| 369 } | 368 } |
| 370 | 369 |
| 371 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { | 370 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { |
| 372 last_sent_packet_ = sent_packet; | 371 last_sent_packet_ = sent_packet; |
| 373 } | 372 } |
| 374 } // namespace cricket | 373 } // namespace cricket |
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