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|    1 /* |    1 /* | 
|    2  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |    2  *  Copyright (c) 2015 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  | 
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|  320 void FakeVideoReceiveStream::SetStats( |  320 void FakeVideoReceiveStream::SetStats( | 
|  321     const webrtc::VideoReceiveStream::Stats& stats) { |  321     const webrtc::VideoReceiveStream::Stats& stats) { | 
|  322   stats_ = stats; |  322   stats_ = stats; | 
|  323 } |  323 } | 
|  324  |  324  | 
|  325 void FakeVideoReceiveStream::EnableEncodedFrameRecording(rtc::PlatformFile file, |  325 void FakeVideoReceiveStream::EnableEncodedFrameRecording(rtc::PlatformFile file, | 
|  326                                                          size_t byte_limit) { |  326                                                          size_t byte_limit) { | 
|  327   rtc::ClosePlatformFile(file); |  327   rtc::ClosePlatformFile(file); | 
|  328 } |  328 } | 
|  329  |  329  | 
 |  330 void FakeVideoReceiveStream::AddSecondarySink( | 
 |  331     webrtc::RtpPacketSinkInterface* sink) { | 
 |  332   RTC_NOTREACHED() << "Not implemented."; | 
 |  333 } | 
 |  334  | 
 |  335 void FakeVideoReceiveStream::RemoveSecondarySink( | 
 |  336     const webrtc::RtpPacketSinkInterface* sink) { | 
 |  337   RTC_NOTREACHED() << "Not implemented."; | 
 |  338 } | 
 |  339  | 
|  330 FakeFlexfecReceiveStream::FakeFlexfecReceiveStream( |  340 FakeFlexfecReceiveStream::FakeFlexfecReceiveStream( | 
|  331     const webrtc::FlexfecReceiveStream::Config& config) |  341     const webrtc::FlexfecReceiveStream::Config& config) | 
|  332     : config_(config), receiving_(false) {} |  342     : config_(config), receiving_(false) {} | 
|  333  |  343  | 
|  334 const webrtc::FlexfecReceiveStream::Config& |  344 const webrtc::FlexfecReceiveStream::Config& | 
|  335 FakeFlexfecReceiveStream::GetConfig() const { |  345 FakeFlexfecReceiveStream::GetConfig() const { | 
|  336   return config_; |  346   return config_; | 
|  337 } |  347 } | 
|  338  |  348  | 
|  339 void FakeFlexfecReceiveStream::Start() { |  349 void FakeFlexfecReceiveStream::Start() { | 
|  340   receiving_ = true; |  350   receiving_ = true; | 
|  341 } |  351 } | 
|  342  |  352  | 
|  343 void FakeFlexfecReceiveStream::Stop() { |  353 void FakeFlexfecReceiveStream::Stop() { | 
|  344   receiving_ = false; |  354   receiving_ = false; | 
|  345 } |  355 } | 
|  346  |  356  | 
|  347 // TODO(brandtr): Implement when the stats have been designed. |  357 // TODO(brandtr): Implement when the stats have been designed. | 
|  348 webrtc::FlexfecReceiveStream::Stats FakeFlexfecReceiveStream::GetStats() const { |  358 webrtc::FlexfecReceiveStream::Stats FakeFlexfecReceiveStream::GetStats() const { | 
|  349   return webrtc::FlexfecReceiveStream::Stats(); |  359   return webrtc::FlexfecReceiveStream::Stats(); | 
|  350 } |  360 } | 
|  351  |  361  | 
 |  362 void FakeFlexfecReceiveStream::OnRtpPacket(const webrtc::RtpPacketReceived&) { | 
 |  363   RTC_NOTREACHED() << "Not implemented."; | 
 |  364 } | 
 |  365  | 
|  352 FakeCall::FakeCall(const webrtc::Call::Config& config) |  366 FakeCall::FakeCall(const webrtc::Call::Config& config) | 
|  353     : config_(config), |  367     : config_(config), | 
|  354       audio_network_state_(webrtc::kNetworkUp), |  368       audio_network_state_(webrtc::kNetworkUp), | 
|  355       video_network_state_(webrtc::kNetworkUp), |  369       video_network_state_(webrtc::kNetworkUp), | 
|  356       num_created_send_streams_(0), |  370       num_created_send_streams_(0), | 
|  357       num_created_receive_streams_(0), |  371       num_created_receive_streams_(0), | 
|  358       audio_transport_overhead_(0), |  372       audio_transport_overhead_(0), | 
|  359       video_transport_overhead_(0) {} |  373       video_transport_overhead_(0) {} | 
|  360  |  374  | 
|  361 FakeCall::~FakeCall() { |  375 FakeCall::~FakeCall() { | 
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|  626 } |  640 } | 
|  627  |  641  | 
|  628 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { |  642 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { | 
|  629   last_sent_packet_ = sent_packet; |  643   last_sent_packet_ = sent_packet; | 
|  630   if (sent_packet.packet_id >= 0) { |  644   if (sent_packet.packet_id >= 0) { | 
|  631     last_sent_nonnegative_packet_id_ = sent_packet.packet_id; |  645     last_sent_nonnegative_packet_id_ = sent_packet.packet_id; | 
|  632   } |  646   } | 
|  633 } |  647 } | 
|  634  |  648  | 
|  635 }  // namespace cricket |  649 }  // namespace cricket | 
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