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Side by Side Diff: webrtc/test/call_test.cc

Issue 2060403002: Add task queue to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@move_getpadding
Patch Set: Fix audio thread check when adding audio to bitrateallocator. Created 4 years, 4 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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260 DriftingClock::kNoDrift)); 260 DriftingClock::kNoDrift));
261 } 261 }
262 262
263 void CallTest::CreateVideoStreams() { 263 void CallTest::CreateVideoStreams() {
264 RTC_DCHECK(video_send_stream_ == nullptr); 264 RTC_DCHECK(video_send_stream_ == nullptr);
265 RTC_DCHECK(video_receive_streams_.empty()); 265 RTC_DCHECK(video_receive_streams_.empty());
266 RTC_DCHECK(audio_send_stream_ == nullptr); 266 RTC_DCHECK(audio_send_stream_ == nullptr);
267 RTC_DCHECK(audio_receive_streams_.empty()); 267 RTC_DCHECK(audio_receive_streams_.empty());
268 268
269 video_send_stream_ = sender_call_->CreateVideoSendStream( 269 video_send_stream_ = sender_call_->CreateVideoSendStream(
270 video_send_config_, video_encoder_config_); 270 video_send_config_.Copy(), video_encoder_config_.Copy());
271 for (size_t i = 0; i < video_receive_configs_.size(); ++i) { 271 for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
272 video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream( 272 video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream(
273 video_receive_configs_[i].Copy())); 273 video_receive_configs_[i].Copy()));
274 } 274 }
275 } 275 }
276 276
277 void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) { 277 void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
278 frame_generator_capturer_->SetFakeRotation(rotation); 278 frame_generator_capturer_->SetFakeRotation(rotation);
279 } 279 }
280 280
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430 430
431 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { 431 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
432 } 432 }
433 433
434 bool EndToEndTest::ShouldCreateReceivers() const { 434 bool EndToEndTest::ShouldCreateReceivers() const {
435 return true; 435 return true;
436 } 436 }
437 437
438 } // namespace test 438 } // namespace test
439 } // namespace webrtc 439 } // namespace webrtc
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