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| 1 /* | 1 /* |
| 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 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 |
| 11 #include <numeric> | 11 #include <numeric> |
| 12 #include <vector> | 12 #include <vector> |
| 13 | 13 |
| 14 #include "webrtc/base/array_view.h" | 14 #include "webrtc/base/array_view.h" |
| 15 #include "webrtc/base/random.h" | 15 #include "webrtc/base/random.h" |
| 16 #include "webrtc/modules/audio_processing/audio_buffer.h" | 16 #include "webrtc/modules/audio_processing/audio_buffer.h" |
| 17 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 17 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| 18 #include "webrtc/modules/audio_processing/residual_echo_detector.h" | 18 #include "webrtc/modules/audio_processing/residual_echo_detector.h" |
| 19 #include "webrtc/modules/audio_processing/test/audio_buffer_tools.h" | 19 #include "webrtc/modules/audio_processing/test/audio_buffer_tools.h" |
| 20 #include "webrtc/modules/audio_processing/test/performance_timer.h" | 20 #include "webrtc/modules/audio_processing/test/performance_timer.h" |
| 21 #include "webrtc/modules/audio_processing/test/simulator_buffers.h" | 21 #include "webrtc/modules/audio_processing/test/simulator_buffers.h" |
| 22 #include "webrtc/system_wrappers/include/clock.h" | 22 #include "webrtc/system_wrappers/include/clock.h" |
| 23 #include "webrtc/test/gtest.h" | 23 #include "webrtc/test/gtest.h" |
| 24 #include "webrtc/test/testsupport/perf_test.h" | 24 #include "webrtc/test/testsupport/perf_test.h" |
| 25 | 25 |
| 26 namespace webrtc { | 26 namespace webrtc { |
| 27 namespace { | 27 namespace { |
| 28 | 28 |
| 29 const size_t kNumFramesToProcess = 500; | 29 const size_t kNumFramesToProcess = 20000; |
|
hlundin-webrtc
2017/03/17 08:48:03
Nit: constexprs, please.
ivoc
2017/03/17 13:53:20
Done.
| |
| 30 const size_t kProcessingBatchSize = 20; | 30 const size_t kNumFramesToProcessStandalone = 50 * kNumFramesToProcess; |
| 31 const size_t kWarmupBatchSize = 2 * kProcessingBatchSize; | 31 const size_t kProcessingBatchSize = 300; |
| 32 const size_t kProcessingBatchSizeStandalone = 50 * kProcessingBatchSize; | |
| 33 const size_t kWarmupBatchSize = 50 * kProcessingBatchSize; | |
| 34 const size_t kWarmupBatchSizeStandalone = 50 * kWarmupBatchSize; | |
| 32 const int kSampleRate = AudioProcessing::kSampleRate48kHz; | 35 const int kSampleRate = AudioProcessing::kSampleRate48kHz; |
| 33 const int kNumberOfChannels = 1; | 36 const int kNumberOfChannels = 1; |
| 34 | 37 |
| 35 std::string FormPerformanceMeasureString(const test::PerformanceTimer& timer) { | 38 std::string FormPerformanceMeasureString(const test::PerformanceTimer& timer) { |
| 36 std::string s = std::to_string(timer.GetDurationAverage()); | 39 std::string s = std::to_string(timer.GetDurationAverage()); |
| 37 s += ", "; | 40 s += ", "; |
| 38 s += std::to_string(timer.GetDurationStandardDeviation()); | 41 s += std::to_string(timer.GetDurationStandardDeviation()); |
| 39 return s; | 42 return s; |
| 40 } | 43 } |
| 41 | 44 |
| 42 void RunStandaloneSubmodule() { | 45 void RunStandaloneSubmodule() { |
| 43 test::SimulatorBuffers buffers( | 46 test::SimulatorBuffers buffers( |
| 44 kSampleRate, kSampleRate, kSampleRate, kSampleRate, kNumberOfChannels, | 47 kSampleRate, kSampleRate, kSampleRate, kSampleRate, kNumberOfChannels, |
| 45 kNumberOfChannels, kNumberOfChannels, kNumberOfChannels); | 48 kNumberOfChannels, kNumberOfChannels, kNumberOfChannels); |
| 46 test::PerformanceTimer timer(kNumFramesToProcess); | 49 test::PerformanceTimer timer(kNumFramesToProcessStandalone / |
|
hlundin-webrtc
2017/03/17 08:48:03
I suggest you modify PerformanceTimer to help you
ivoc
2017/03/17 13:53:20
Good idea. I implemented it in a bit of a differen
hlundin-webrtc
2017/03/17 14:16:36
Good solution!
| |
| 50 kProcessingBatchSizeStandalone); | |
| 47 | 51 |
| 48 ResidualEchoDetector echo_detector; | 52 ResidualEchoDetector echo_detector; |
| 49 echo_detector.Initialize(); | 53 echo_detector.Initialize(); |
| 54 float sum = 0.f; | |
| 50 | 55 |
| 51 for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) { | 56 for (size_t frame_no = 0; frame_no < kNumFramesToProcessStandalone; |
| 57 ++frame_no) { | |
| 52 // The first batch of frames are for warming up, and are not part of the | 58 // The first batch of frames are for warming up, and are not part of the |
| 53 // benchmark. After that the processing time is measured in chunks of | 59 // benchmark. After that the processing time is measured in chunks of |
| 54 // kProcessingBatchSize frames. | 60 // kProcessingBatchSize frames. |
| 55 if (frame_no >= kWarmupBatchSize && frame_no % kProcessingBatchSize == 0) { | 61 if (frame_no >= kWarmupBatchSizeStandalone && |
| 62 frame_no % kProcessingBatchSizeStandalone == 0) { | |
|
hlundin-webrtc
2017/03/17 08:48:03
I don't know if this will skew the measurement, bu
ivoc
2017/03/17 13:53:20
Although I agree that modulo operations are pretty
hlundin-webrtc
2017/03/17 14:16:36
Acknowledged.
| |
| 56 timer.StartTimer(); | 63 timer.StartTimer(); |
| 57 } | 64 } |
| 58 | 65 |
| 59 buffers.UpdateInputBuffers(); | 66 buffers.UpdateInputBuffers(); |
| 60 echo_detector.AnalyzeRenderAudio(rtc::ArrayView<const float>( | 67 echo_detector.AnalyzeRenderAudio(rtc::ArrayView<const float>( |
| 61 buffers.render_input_buffer->split_bands_const_f(0)[kBand0To8kHz], | 68 buffers.render_input_buffer->split_bands_const_f(0)[kBand0To8kHz], |
| 62 buffers.render_input_buffer->num_frames_per_band())); | 69 buffers.render_input_buffer->num_frames_per_band())); |
| 63 echo_detector.AnalyzeCaptureAudio(rtc::ArrayView<const float>( | 70 echo_detector.AnalyzeCaptureAudio(rtc::ArrayView<const float>( |
| 64 buffers.capture_input_buffer->split_bands_const_f(0)[kBand0To8kHz], | 71 buffers.capture_input_buffer->split_bands_const_f(0)[kBand0To8kHz], |
| 65 buffers.capture_input_buffer->num_frames_per_band())); | 72 buffers.capture_input_buffer->num_frames_per_band())); |
| 73 sum += echo_detector.echo_likelihood(); | |
| 66 | 74 |
| 67 if (frame_no >= kWarmupBatchSize && | 75 if (frame_no >= kWarmupBatchSizeStandalone && |
| 68 frame_no % kProcessingBatchSize == kProcessingBatchSize - 1) { | 76 frame_no % kProcessingBatchSizeStandalone == |
| 77 kProcessingBatchSizeStandalone - 1) { | |
| 69 timer.StopTimer(); | 78 timer.StopTimer(); |
| 70 } | 79 } |
| 71 } | 80 } |
| 81 EXPECT_EQ(0.0f, sum); | |
|
hlundin-webrtc
2017/03/17 08:48:03
Do we know that this will be exactly 0.0? Are ther
ivoc
2017/03/17 13:53:20
In this test both signals are filled with zeros, s
hlundin-webrtc
2017/03/17 14:16:36
Acknowledged.
| |
| 72 webrtc::test::PrintResultMeanAndError( | 82 webrtc::test::PrintResultMeanAndError( |
| 73 "echo_detector_call_durations", "", "StandaloneEchoDetector", | 83 "echo_detector_call_durations", "", "StandaloneEchoDetector", |
| 74 FormPerformanceMeasureString(timer), "us", false); | 84 FormPerformanceMeasureString(timer), "us", false); |
| 75 } | 85 } |
| 76 | 86 |
| 77 void RunTogetherWithApm(std::string test_description, | 87 void RunTogetherWithApm(std::string test_description, |
| 78 bool use_mobile_aec, | 88 bool use_mobile_aec, |
| 79 bool include_default_apm_processing) { | 89 bool include_default_apm_processing) { |
| 80 test::SimulatorBuffers buffers( | 90 test::SimulatorBuffers buffers( |
| 81 kSampleRate, kSampleRate, kSampleRate, kSampleRate, kNumberOfChannels, | 91 kSampleRate, kSampleRate, kSampleRate, kSampleRate, kNumberOfChannels, |
| 82 kNumberOfChannels, kNumberOfChannels, kNumberOfChannels); | 92 kNumberOfChannels, kNumberOfChannels, kNumberOfChannels); |
| 83 test::PerformanceTimer timer(kNumFramesToProcess); | 93 test::PerformanceTimer timer(kNumFramesToProcess / kProcessingBatchSize); |
| 84 | 94 |
| 85 webrtc::Config config; | 95 webrtc::Config config; |
| 86 AudioProcessing::Config apm_config; | 96 AudioProcessing::Config apm_config; |
| 87 if (include_default_apm_processing) { | 97 if (include_default_apm_processing) { |
| 88 config.Set<DelayAgnostic>(new DelayAgnostic(true)); | 98 config.Set<DelayAgnostic>(new DelayAgnostic(true)); |
| 89 config.Set<ExtendedFilter>(new ExtendedFilter(true)); | 99 config.Set<ExtendedFilter>(new ExtendedFilter(true)); |
| 90 } | 100 } |
| 91 apm_config.level_controller.enabled = include_default_apm_processing; | 101 apm_config.level_controller.enabled = include_default_apm_processing; |
| 92 apm_config.residual_echo_detector.enabled = true; | 102 apm_config.residual_echo_detector.enabled = true; |
| 93 | 103 |
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| 166 TEST(EchoDetectorPerformanceTest, ProcessingViaApm) { | 176 TEST(EchoDetectorPerformanceTest, ProcessingViaApm) { |
| 167 RunTogetherWithApm("SimpleEchoDetectorViaApm", false, false); | 177 RunTogetherWithApm("SimpleEchoDetectorViaApm", false, false); |
| 168 } | 178 } |
| 169 | 179 |
| 170 TEST(EchoDetectorPerformanceTest, InteractionWithDefaultApm) { | 180 TEST(EchoDetectorPerformanceTest, InteractionWithDefaultApm) { |
| 171 RunTogetherWithApm("EchoDetectorAndDefaultDesktopApm", false, true); | 181 RunTogetherWithApm("EchoDetectorAndDefaultDesktopApm", false, true); |
| 172 RunTogetherWithApm("EchoDetectorAndDefaultMobileApm", true, true); | 182 RunTogetherWithApm("EchoDetectorAndDefaultMobileApm", true, true); |
| 173 } | 183 } |
| 174 | 184 |
| 175 } // namespace webrtc | 185 } // namespace webrtc |
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