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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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30 #include <algorithm> | 30 #include <algorithm> |
31 | 31 |
32 #include "talk/media/base/rtputils.h" | 32 #include "talk/media/base/rtputils.h" |
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 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 { | 43 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const { |
44 return webrtc::AudioReceiveStream::Stats(); | 44 return webrtc::AudioReceiveStream::Stats(); |
45 } | 45 } |
46 | 46 |
47 const webrtc::AudioReceiveStream::Config& | 47 const webrtc::AudioReceiveStream::Config& |
48 FakeAudioReceiveStream::GetConfig() const { | 48 FakeAudioReceiveStream::GetConfig() const { |
49 return config_; | 49 return config_; |
50 } | 50 } |
51 | 51 |
52 void FakeAudioReceiveStream::IncrementReceivedPackets() { | 52 void FakeAudioReceiveStream::IncrementReceivedPackets() { |
53 received_packets_++; | 53 received_packets_++; |
54 } | 54 } |
55 | 55 |
56 FakeVideoSendStream::FakeVideoSendStream( | 56 FakeVideoSendStream::FakeVideoSendStream( |
57 const webrtc::VideoSendStream::Config& config, | 57 const webrtc::VideoSendStream::Config& config, |
58 const webrtc::VideoEncoderConfig& encoder_config) | 58 const webrtc::VideoEncoderConfig& encoder_config) |
59 : sending_(false), | 59 : sending_(false), |
60 config_(config), | 60 config_(config), |
61 codec_settings_set_(false), | 61 codec_settings_set_(false), |
62 num_swapped_frames_(0) { | 62 num_swapped_frames_(0) { |
63 DCHECK(config.encoder_settings.encoder != NULL); | 63 RTC_DCHECK(config.encoder_settings.encoder != NULL); |
64 ReconfigureVideoEncoder(encoder_config); | 64 ReconfigureVideoEncoder(encoder_config); |
65 } | 65 } |
66 | 66 |
67 webrtc::VideoSendStream::Config FakeVideoSendStream::GetConfig() const { | 67 webrtc::VideoSendStream::Config FakeVideoSendStream::GetConfig() const { |
68 return config_; | 68 return config_; |
69 } | 69 } |
70 | 70 |
71 webrtc::VideoEncoderConfig FakeVideoSendStream::GetEncoderConfig() const { | 71 webrtc::VideoEncoderConfig FakeVideoSendStream::GetEncoderConfig() const { |
72 return encoder_config_; | 72 return encoder_config_; |
73 } | 73 } |
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106 | 106 |
107 int FakeVideoSendStream::GetLastWidth() const { | 107 int FakeVideoSendStream::GetLastWidth() const { |
108 return last_frame_.width(); | 108 return last_frame_.width(); |
109 } | 109 } |
110 | 110 |
111 int FakeVideoSendStream::GetLastHeight() const { | 111 int FakeVideoSendStream::GetLastHeight() const { |
112 return last_frame_.height(); | 112 return last_frame_.height(); |
113 } | 113 } |
114 | 114 |
115 int64_t FakeVideoSendStream::GetLastTimestamp() const { | 115 int64_t FakeVideoSendStream::GetLastTimestamp() const { |
116 DCHECK(last_frame_.ntp_time_ms() == 0); | 116 RTC_DCHECK(last_frame_.ntp_time_ms() == 0); |
117 return last_frame_.render_time_ms(); | 117 return last_frame_.render_time_ms(); |
118 } | 118 } |
119 | 119 |
120 void FakeVideoSendStream::IncomingCapturedFrame( | 120 void FakeVideoSendStream::IncomingCapturedFrame( |
121 const webrtc::VideoFrame& frame) { | 121 const webrtc::VideoFrame& frame) { |
122 ++num_swapped_frames_; | 122 ++num_swapped_frames_; |
123 last_frame_.ShallowCopy(frame); | 123 last_frame_.ShallowCopy(frame); |
124 } | 124 } |
125 | 125 |
126 void FakeVideoSendStream::SetStats( | 126 void FakeVideoSendStream::SetStats( |
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361 | 361 |
362 void FakeCall::SetBitrateConfig( | 362 void FakeCall::SetBitrateConfig( |
363 const webrtc::Call::Config::BitrateConfig& bitrate_config) { | 363 const webrtc::Call::Config::BitrateConfig& bitrate_config) { |
364 config_.bitrate_config = bitrate_config; | 364 config_.bitrate_config = bitrate_config; |
365 } | 365 } |
366 | 366 |
367 void FakeCall::SignalNetworkState(webrtc::NetworkState state) { | 367 void FakeCall::SignalNetworkState(webrtc::NetworkState state) { |
368 network_state_ = state; | 368 network_state_ = state; |
369 } | 369 } |
370 } // namespace cricket | 370 } // namespace cricket |
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