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Issue 1886113003: Add rotation to EncodedImage and make sure it is passed through encoders. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix merge mistake. Created 4 years, 8 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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 #include "webrtc/base/checks.h" 10 #include "webrtc/base/checks.h"
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268 RTC_DCHECK(audio_receive_streams_.empty()); 268 RTC_DCHECK(audio_receive_streams_.empty());
269 269
270 video_send_stream_ = sender_call_->CreateVideoSendStream( 270 video_send_stream_ = sender_call_->CreateVideoSendStream(
271 video_send_config_, video_encoder_config_); 271 video_send_config_, video_encoder_config_);
272 for (size_t i = 0; i < video_receive_configs_.size(); ++i) { 272 for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
273 video_receive_streams_.push_back( 273 video_receive_streams_.push_back(
274 receiver_call_->CreateVideoReceiveStream(video_receive_configs_[i])); 274 receiver_call_->CreateVideoReceiveStream(video_receive_configs_[i]));
275 } 275 }
276 } 276 }
277 277
278 void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
279 frame_generator_capturer_->SetFakeRotation(rotation);
280 }
281
278 void CallTest::CreateAudioStreams() { 282 void CallTest::CreateAudioStreams() {
279 audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_); 283 audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
280 for (size_t i = 0; i < audio_receive_configs_.size(); ++i) { 284 for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
281 audio_receive_streams_.push_back( 285 audio_receive_streams_.push_back(
282 receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i])); 286 receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
283 } 287 }
284 CodecInst isac = {kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000}; 288 CodecInst isac = {kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
285 EXPECT_EQ(0, voe_send_.codec->SetSendCodec(voe_send_.channel_id, isac)); 289 EXPECT_EQ(0, voe_send_.codec->SetSendCodec(voe_send_.channel_id, isac));
286 } 290 }
287 291
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446 450
447 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { 451 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
448 } 452 }
449 453
450 bool EndToEndTest::ShouldCreateReceivers() const { 454 bool EndToEndTest::ShouldCreateReceivers() const {
451 return true; 455 return true;
452 } 456 }
453 457
454 } // namespace test 458 } // namespace test
455 } // namespace webrtc 459 } // namespace webrtc
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