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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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