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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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101 gain_ = gain; | 101 gain_ = gain; |
102 } | 102 } |
103 | 103 |
104 FakeVideoSendStream::FakeVideoSendStream( | 104 FakeVideoSendStream::FakeVideoSendStream( |
105 webrtc::VideoSendStream::Config config, | 105 webrtc::VideoSendStream::Config config, |
106 webrtc::VideoEncoderConfig encoder_config) | 106 webrtc::VideoEncoderConfig encoder_config) |
107 : sending_(false), | 107 : sending_(false), |
108 config_(std::move(config)), | 108 config_(std::move(config)), |
109 codec_settings_set_(false), | 109 codec_settings_set_(false), |
110 resolution_scaling_enabled_(false), | 110 resolution_scaling_enabled_(false), |
111 framerate_scaling_enalbed_(false), | |
nisse-webrtc
2017/03/14 09:00:28
Spelling.
sprang_webrtc
2017/03/14 14:15:02
Done.
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111 source_(nullptr), | 112 source_(nullptr), |
112 num_swapped_frames_(0) { | 113 num_swapped_frames_(0) { |
113 RTC_DCHECK(config.encoder_settings.encoder != NULL); | 114 RTC_DCHECK(config.encoder_settings.encoder != NULL); |
114 ReconfigureVideoEncoder(std::move(encoder_config)); | 115 ReconfigureVideoEncoder(std::move(encoder_config)); |
115 } | 116 } |
116 | 117 |
117 FakeVideoSendStream::~FakeVideoSendStream() { | 118 FakeVideoSendStream::~FakeVideoSendStream() { |
118 if (source_) | 119 if (source_) |
119 source_->RemoveSink(this); | 120 source_->RemoveSink(this); |
120 } | 121 } |
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245 } | 246 } |
246 | 247 |
247 void FakeVideoSendStream::SetSource( | 248 void FakeVideoSendStream::SetSource( |
248 rtc::VideoSourceInterface<webrtc::VideoFrame>* source, | 249 rtc::VideoSourceInterface<webrtc::VideoFrame>* source, |
249 const webrtc::VideoSendStream::DegradationPreference& | 250 const webrtc::VideoSendStream::DegradationPreference& |
250 degradation_preference) { | 251 degradation_preference) { |
251 RTC_DCHECK(source != source_); | 252 RTC_DCHECK(source != source_); |
252 if (source_) | 253 if (source_) |
253 source_->RemoveSink(this); | 254 source_->RemoveSink(this); |
254 source_ = source; | 255 source_ = source; |
255 resolution_scaling_enabled_ = | 256 switch (degradation_preference) { |
nisse-webrtc
2017/03/14 09:00:28
This is only ever called once in the object's life
sprang_webrtc
2017/03/14 14:15:02
Right, thanks. I misread this code.
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256 degradation_preference != | 257 case DegradationPreference::kMaintainFramerate: |
257 webrtc::VideoSendStream::DegradationPreference::kMaintainResolution; | 258 resolution_scaling_enabled_ = true; |
259 break; | |
260 case DegradationPreference::kMaintainResolution: | |
261 framerate_scaling_enalbed_ = true; | |
262 break; | |
263 case DegradationPreference::kBalanced: | |
264 resolution_scaling_enabled_ = true; | |
265 framerate_scaling_enalbed_ = true; | |
266 break; | |
267 case DegradationPreference::kDegradationDisabled: | |
268 // No scaling enabled. | |
269 break; | |
270 } | |
258 if (source) | 271 if (source) |
259 source->AddOrUpdateSink(this, resolution_scaling_enabled_ | 272 source->AddOrUpdateSink(this, resolution_scaling_enabled_ |
260 ? sink_wants_ | 273 ? sink_wants_ |
261 : rtc::VideoSinkWants()); | 274 : rtc::VideoSinkWants()); |
262 } | 275 } |
263 | 276 |
264 void FakeVideoSendStream::InjectVideoSinkWants( | 277 void FakeVideoSendStream::InjectVideoSinkWants( |
265 const rtc::VideoSinkWants& wants) { | 278 const rtc::VideoSinkWants& wants) { |
266 sink_wants_ = wants; | 279 sink_wants_ = wants; |
267 source_->AddOrUpdateSink(this, wants); | 280 source_->AddOrUpdateSink(this, wants); |
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326 // TODO(brandtr): Implement when the stats have been designed. | 339 // TODO(brandtr): Implement when the stats have been designed. |
327 webrtc::FlexfecReceiveStream::Stats FakeFlexfecReceiveStream::GetStats() const { | 340 webrtc::FlexfecReceiveStream::Stats FakeFlexfecReceiveStream::GetStats() const { |
328 return webrtc::FlexfecReceiveStream::Stats(); | 341 return webrtc::FlexfecReceiveStream::Stats(); |
329 } | 342 } |
330 | 343 |
331 FakeCall::FakeCall(const webrtc::Call::Config& config) | 344 FakeCall::FakeCall(const webrtc::Call::Config& config) |
332 : config_(config), | 345 : config_(config), |
333 audio_network_state_(webrtc::kNetworkUp), | 346 audio_network_state_(webrtc::kNetworkUp), |
334 video_network_state_(webrtc::kNetworkUp), | 347 video_network_state_(webrtc::kNetworkUp), |
335 num_created_send_streams_(0), | 348 num_created_send_streams_(0), |
336 num_created_receive_streams_(0) {} | 349 num_created_receive_streams_(0), |
350 audio_transport_overhead_(0), | |
351 video_transport_overhead_(0) {} | |
337 | 352 |
338 FakeCall::~FakeCall() { | 353 FakeCall::~FakeCall() { |
339 EXPECT_EQ(0u, video_send_streams_.size()); | 354 EXPECT_EQ(0u, video_send_streams_.size()); |
340 EXPECT_EQ(0u, audio_send_streams_.size()); | 355 EXPECT_EQ(0u, audio_send_streams_.size()); |
341 EXPECT_EQ(0u, video_receive_streams_.size()); | 356 EXPECT_EQ(0u, video_receive_streams_.size()); |
342 EXPECT_EQ(0u, audio_receive_streams_.size()); | 357 EXPECT_EQ(0u, audio_receive_streams_.size()); |
343 } | 358 } |
344 | 359 |
345 webrtc::Call::Config FakeCall::GetConfig() const { | 360 webrtc::Call::Config FakeCall::GetConfig() const { |
346 return config_; | 361 return config_; |
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597 } | 612 } |
598 | 613 |
599 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { | 614 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { |
600 last_sent_packet_ = sent_packet; | 615 last_sent_packet_ = sent_packet; |
601 if (sent_packet.packet_id >= 0) { | 616 if (sent_packet.packet_id >= 0) { |
602 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; | 617 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; |
603 } | 618 } |
604 } | 619 } |
605 | 620 |
606 } // namespace cricket | 621 } // namespace cricket |
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