Improvements to the reliability of the echo detector perf test.
BUG=chromium:673683 Review-Url: https://codereview.webrtc.org/2568883004 Cr-Commit-Position: refs/heads/master@{#15606}
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@ -26,7 +26,9 @@
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namespace webrtc {
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namespace {
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const size_t kNumFramesToProcess = 100;
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const size_t kNumFramesToProcess = 500;
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const size_t kProcessingBatchSize = 20;
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const size_t kWarmupBatchSize = 2 * kProcessingBatchSize;
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const int kSampleRate = AudioProcessing::kSampleRate48kHz;
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const int kNumberOfChannels = 1;
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@ -47,16 +49,25 @@ void RunStandaloneSubmodule() {
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echo_detector.Initialize();
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for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
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buffers.UpdateInputBuffers();
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// The first batch of frames are for warming up, and are not part of the
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// benchmark. After that the processing time is measured in chunks of
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// kProcessingBatchSize frames.
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if (frame_no >= kWarmupBatchSize && frame_no % kProcessingBatchSize == 0) {
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timer.StartTimer();
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}
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timer.StartTimer();
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buffers.UpdateInputBuffers();
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echo_detector.AnalyzeRenderAudio(rtc::ArrayView<const float>(
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buffers.render_input_buffer->split_bands_const_f(0)[kBand0To8kHz],
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buffers.render_input_buffer->num_frames_per_band()));
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echo_detector.AnalyzeCaptureAudio(rtc::ArrayView<const float>(
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buffers.capture_input_buffer->split_bands_const_f(0)[kBand0To8kHz],
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buffers.capture_input_buffer->num_frames_per_band()));
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timer.StopTimer();
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if (frame_no >= kWarmupBatchSize &&
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frame_no % kProcessingBatchSize == kProcessingBatchSize - 1) {
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timer.StopTimer();
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}
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}
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webrtc::test::PrintResultMeanAndError(
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"echo_detector_call_durations", "", "StandaloneEchoDetector",
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@ -69,9 +80,7 @@ void RunTogetherWithApm(std::string test_description,
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test::SimulatorBuffers buffers(
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kSampleRate, kSampleRate, kSampleRate, kSampleRate, kNumberOfChannels,
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kNumberOfChannels, kNumberOfChannels, kNumberOfChannels);
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test::PerformanceTimer render_timer(kNumFramesToProcess);
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test::PerformanceTimer capture_timer(kNumFramesToProcess);
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test::PerformanceTimer total_timer(kNumFramesToProcess);
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test::PerformanceTimer timer(kNumFramesToProcess);
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webrtc::Config config;
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AudioProcessing::Config apm_config;
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@ -112,18 +121,20 @@ void RunTogetherWithApm(std::string test_description,
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StreamConfig stream_config(kSampleRate, kNumberOfChannels, false);
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for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
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// The first batch of frames are for warming up, and are not part of the
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// benchmark. After that the processing time is measured in chunks of
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// kProcessingBatchSize frames.
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if (frame_no >= kWarmupBatchSize && frame_no % kProcessingBatchSize == 0) {
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timer.StartTimer();
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}
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buffers.UpdateInputBuffers();
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total_timer.StartTimer();
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render_timer.StartTimer();
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ASSERT_EQ(
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AudioProcessing::kNoError,
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apm->ProcessReverseStream(&buffers.render_input[0], stream_config,
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stream_config, &buffers.render_output[0]));
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render_timer.StopTimer();
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capture_timer.StartTimer();
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ASSERT_EQ(AudioProcessing::kNoError, apm->set_stream_delay_ms(0));
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if (include_default_apm_processing) {
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apm->gain_control()->set_stream_analog_level(0);
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@ -135,19 +146,15 @@ void RunTogetherWithApm(std::string test_description,
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apm->ProcessStream(&buffers.capture_input[0], stream_config,
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stream_config, &buffers.capture_output[0]));
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capture_timer.StopTimer();
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total_timer.StopTimer();
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if (frame_no >= kWarmupBatchSize &&
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frame_no % kProcessingBatchSize == kProcessingBatchSize - 1) {
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timer.StopTimer();
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}
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}
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webrtc::test::PrintResultMeanAndError(
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"echo_detector_call_durations", "_render", test_description,
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FormPerformanceMeasureString(render_timer), "us", false);
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webrtc::test::PrintResultMeanAndError(
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"echo_detector_call_durations", "_capture", test_description,
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FormPerformanceMeasureString(capture_timer), "us", false);
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webrtc::test::PrintResultMeanAndError(
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"echo_detector_call_durations", "_total", test_description,
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FormPerformanceMeasureString(total_timer), "us", false);
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FormPerformanceMeasureString(timer), "us", false);
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}
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} // namespace
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