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Issue 2525943003: Increase the threshold for RunPlayoutAndRecordingInFullDuplex. Again. (Closed)
Patch Set: Created 4 years ago
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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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975 StartRecording(); 975 StartRecording();
976 StartPlayout(); 976 StartPlayout();
977 test_is_done_->Wait(std::max(kTestTimeOutInMilliseconds, 977 test_is_done_->Wait(std::max(kTestTimeOutInMilliseconds,
978 1000 * kFullDuplexTimeInSec)); 978 1000 * kFullDuplexTimeInSec));
979 StopPlayout(); 979 StopPlayout();
980 StopRecording(); 980 StopRecording();
981 981
982 // These thresholds are set rather high to accomodate differences in hardware 982 // These thresholds are set rather high to accomodate differences in hardware
983 // in several devices, so this test can be used in swarming. 983 // in several devices, so this test can be used in swarming.
984 // See http://bugs.webrtc.org/6464 984 // See http://bugs.webrtc.org/6464
985 EXPECT_LE(fifo_audio_stream->average_size(), 30u); 985 EXPECT_LE(fifo_audio_stream->average_size(), 60u);
986 EXPECT_LE(fifo_audio_stream->largest_size(), 40u); 986 EXPECT_LE(fifo_audio_stream->largest_size(), 70u);
987 } 987 }
988 988
989 // Measures loopback latency and reports the min, max and average values for 989 // Measures loopback latency and reports the min, max and average values for
990 // a full duplex audio session. 990 // a full duplex audio session.
991 // The latency is measured like so: 991 // The latency is measured like so:
992 // - Insert impulses periodically on the output side. 992 // - Insert impulses periodically on the output side.
993 // - Detect the impulses on the input side. 993 // - Detect the impulses on the input side.
994 // - Measure the time difference between the transmit time and receive time. 994 // - Measure the time difference between the transmit time and receive time.
995 // - Store time differences in a vector and calculate min, max and average. 995 // - Store time differences in a vector and calculate min, max and average.
996 // This test requires a special hardware called Audio Loopback Dongle. 996 // This test requires a special hardware called Audio Loopback Dongle.
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1014 StopPlayout(); 1014 StopPlayout();
1015 StopRecording(); 1015 StopRecording();
1016 // Verify that the correct number of transmitted impulses are detected. 1016 // Verify that the correct number of transmitted impulses are detected.
1017 EXPECT_EQ(latency_audio_stream->num_latency_values(), 1017 EXPECT_EQ(latency_audio_stream->num_latency_values(),
1018 static_cast<size_t>( 1018 static_cast<size_t>(
1019 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1)); 1019 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
1020 latency_audio_stream->PrintResults(); 1020 latency_audio_stream->PrintResults();
1021 } 1021 }
1022 1022
1023 } // namespace webrtc 1023 } // namespace webrtc
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