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Side by Side Diff: remoting/protocol/connection_unittest.cc

Issue 2710483002: Actually pass the stereo=1 parameter to WebRTC for receiving audio (Closed)
Patch Set: err -> parse_error, no braces Created 3 years, 10 months ago
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1 // Copyright 2015 The Chromium Authors. All rights reserved. 1 // Copyright 2015 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #define _USE_MATH_DEFINES // For VC++ to get M_PI. This has to be first. 5 #define _USE_MATH_DEFINES // For VC++ to get M_PI. This has to be first.
6 6
7 #include <utility> 7 #include <utility>
8 8
9 #include "base/bind.h" 9 #include "base/bind.h"
10 #include "base/macros.h" 10 #include "base/macros.h"
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616 stats.host_stats.capture_overhead_delay + 616 stats.host_stats.capture_overhead_delay +
617 stats.host_stats.encode_delay + 617 stats.host_stats.encode_delay +
618 stats.host_stats.send_pending_delay, 618 stats.host_stats.send_pending_delay,
619 stats.client_stats.time_received); 619 stats.client_stats.time_received);
620 EXPECT_LE(stats.client_stats.time_received, stats.client_stats.time_decoded); 620 EXPECT_LE(stats.client_stats.time_received, stats.client_stats.time_decoded);
621 EXPECT_LE(stats.client_stats.time_decoded, stats.client_stats.time_rendered); 621 EXPECT_LE(stats.client_stats.time_decoded, stats.client_stats.time_rendered);
622 EXPECT_LE(stats.client_stats.time_rendered, finish_time); 622 EXPECT_LE(stats.client_stats.time_rendered, finish_time);
623 } 623 }
624 624
625 // Disabling due to failures after WebRTC roll http://crbug.com/685910 625 // Disabling due to failures after WebRTC roll http://crbug.com/685910
626 TEST_P(ConnectionTest, DISABLED_Audio) { 626 TEST_P(ConnectionTest, Audio) {
627 Connect(); 627 Connect();
628 628
629 std::unique_ptr<AudioStream> audio_stream = 629 std::unique_ptr<AudioStream> audio_stream =
630 host_connection_->StartAudioStream(base::MakeUnique<TestAudioSource>()); 630 host_connection_->StartAudioStream(base::MakeUnique<TestAudioSource>());
631 631
632 // Wait for 1 second worth of audio samples. 632 // Wait for 1 second worth of audio samples.
633 client_audio_player_.WaitForSamples(kAudioSampleRate * 2); 633 client_audio_player_.WaitForSamples(kAudioSampleRate * 2);
634 client_audio_player_.Verify(); 634 client_audio_player_.Verify();
635 } 635 }
636 636
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673 auto capturer = base::MakeUnique<TestScreenCapturer>(); 673 auto capturer = base::MakeUnique<TestScreenCapturer>();
674 capturer->FailNthFrame(1); 674 capturer->FailNthFrame(1);
675 auto video_stream = host_connection_->StartVideoStream(std::move(capturer)); 675 auto video_stream = host_connection_->StartVideoStream(std::move(capturer));
676 676
677 WaitNextVideoFrame(); 677 WaitNextVideoFrame();
678 WaitNextVideoFrame(); 678 WaitNextVideoFrame();
679 } 679 }
680 680
681 } // namespace protocol 681 } // namespace protocol
682 } // namespace remoting 682 } // namespace remoting
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