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}
This commit is contained in:
ivoc 2016-12-14 05:48:16 -08:00 committed by Commit bot
parent a701469a93
commit 930959d261

View File

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