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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
246 lines
10 KiB
C++
246 lines
10 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <numeric>
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#include <vector>
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#include "api/array_view.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/audio_processing/level_controller/level_controller.h"
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#include "modules/audio_processing/test/audio_buffer_tools.h"
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#include "modules/audio_processing/test/bitexactness_tools.h"
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#include "modules/audio_processing/test/performance_timer.h"
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#include "modules/audio_processing/test/simulator_buffers.h"
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#include "rtc_base/random.h"
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#include "system_wrappers/include/clock.h"
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#include "test/gtest.h"
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#include "test/testsupport/perf_test.h"
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namespace webrtc {
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namespace {
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const size_t kNumFramesToProcess = 300;
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const size_t kNumFramesToProcessAtWarmup = 300;
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const size_t kToTalNumFrames =
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kNumFramesToProcess + kNumFramesToProcessAtWarmup;
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std::string FormPerformanceMeasureString(const test::PerformanceTimer& timer) {
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std::string s = std::to_string(timer.GetDurationAverage());
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s += ", ";
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s += std::to_string(timer.GetDurationStandardDeviation());
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return s;
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}
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void RunStandaloneSubmodule(int sample_rate_hz, size_t num_channels) {
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test::SimulatorBuffers buffers(sample_rate_hz, sample_rate_hz, sample_rate_hz,
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sample_rate_hz, num_channels, num_channels,
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num_channels, num_channels);
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test::PerformanceTimer timer(kNumFramesToProcess);
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LevelController level_controller;
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level_controller.Initialize(sample_rate_hz);
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for (size_t frame_no = 0; frame_no < kToTalNumFrames; ++frame_no) {
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buffers.UpdateInputBuffers();
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if (frame_no >= kNumFramesToProcessAtWarmup) {
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timer.StartTimer();
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}
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level_controller.Process(buffers.capture_input_buffer.get());
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if (frame_no >= kNumFramesToProcessAtWarmup) {
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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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"level_controller_call_durations",
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"_" + std::to_string(sample_rate_hz) + "Hz_" +
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std::to_string(num_channels) + "_channels",
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"StandaloneLevelControl", FormPerformanceMeasureString(timer), "us",
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false);
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}
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void RunTogetherWithApm(const std::string& test_description,
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int render_input_sample_rate_hz,
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int render_output_sample_rate_hz,
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int capture_input_sample_rate_hz,
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int capture_output_sample_rate_hz,
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size_t num_channels,
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bool use_mobile_aec,
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bool include_default_apm_processing) {
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test::SimulatorBuffers buffers(
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render_input_sample_rate_hz, capture_input_sample_rate_hz,
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render_output_sample_rate_hz, capture_output_sample_rate_hz, num_channels,
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num_channels, num_channels, num_channels);
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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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webrtc::Config config;
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AudioProcessing::Config apm_config;
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if (include_default_apm_processing) {
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config.Set<DelayAgnostic>(new DelayAgnostic(true));
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config.Set<ExtendedFilter>(new ExtendedFilter(true));
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}
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apm_config.level_controller.enabled = true;
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apm_config.residual_echo_detector.enabled = include_default_apm_processing;
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std::unique_ptr<AudioProcessing> apm;
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apm.reset(AudioProcessing::Create(config));
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ASSERT_TRUE(apm.get());
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apm->ApplyConfig(apm_config);
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ASSERT_EQ(AudioProcessing::kNoError,
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apm->gain_control()->Enable(include_default_apm_processing));
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if (use_mobile_aec) {
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ASSERT_EQ(AudioProcessing::kNoError,
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apm->echo_cancellation()->Enable(false));
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ASSERT_EQ(AudioProcessing::kNoError, apm->echo_control_mobile()->Enable(
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include_default_apm_processing));
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} else {
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ASSERT_EQ(AudioProcessing::kNoError,
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apm->echo_cancellation()->Enable(include_default_apm_processing));
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ASSERT_EQ(AudioProcessing::kNoError,
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apm->echo_control_mobile()->Enable(false));
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}
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apm_config.high_pass_filter.enabled = include_default_apm_processing;
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ASSERT_EQ(AudioProcessing::kNoError,
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apm->noise_suppression()->Enable(include_default_apm_processing));
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ASSERT_EQ(AudioProcessing::kNoError,
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apm->voice_detection()->Enable(include_default_apm_processing));
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ASSERT_EQ(AudioProcessing::kNoError,
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apm->level_estimator()->Enable(include_default_apm_processing));
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StreamConfig render_input_config(render_input_sample_rate_hz, num_channels,
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false);
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StreamConfig render_output_config(render_output_sample_rate_hz, num_channels,
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false);
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StreamConfig capture_input_config(capture_input_sample_rate_hz, num_channels,
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false);
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StreamConfig capture_output_config(capture_output_sample_rate_hz,
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num_channels, false);
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for (size_t frame_no = 0; frame_no < kToTalNumFrames; ++frame_no) {
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buffers.UpdateInputBuffers();
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if (frame_no >= kNumFramesToProcessAtWarmup) {
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total_timer.StartTimer();
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render_timer.StartTimer();
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}
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ASSERT_EQ(AudioProcessing::kNoError,
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apm->ProcessReverseStream(
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&buffers.render_input[0], render_input_config,
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render_output_config, &buffers.render_output[0]));
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if (frame_no >= kNumFramesToProcessAtWarmup) {
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render_timer.StopTimer();
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capture_timer.StartTimer();
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}
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ASSERT_EQ(AudioProcessing::kNoError, apm->set_stream_delay_ms(0));
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ASSERT_EQ(
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AudioProcessing::kNoError,
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apm->ProcessStream(&buffers.capture_input[0], capture_input_config,
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capture_output_config, &buffers.capture_output[0]));
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if (frame_no >= kNumFramesToProcessAtWarmup) {
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capture_timer.StopTimer();
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total_timer.StopTimer();
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}
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}
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webrtc::test::PrintResultMeanAndError(
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"level_controller_call_durations",
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"_" + std::to_string(render_input_sample_rate_hz) + "_" +
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std::to_string(render_output_sample_rate_hz) + "_" +
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std::to_string(capture_input_sample_rate_hz) + "_" +
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std::to_string(capture_output_sample_rate_hz) + "Hz_" +
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std::to_string(num_channels) + "_channels" + "_render",
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test_description, FormPerformanceMeasureString(render_timer), "us",
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false);
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webrtc::test::PrintResultMeanAndError(
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"level_controller_call_durations",
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"_" + std::to_string(render_input_sample_rate_hz) + "_" +
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std::to_string(render_output_sample_rate_hz) + "_" +
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std::to_string(capture_input_sample_rate_hz) + "_" +
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std::to_string(capture_output_sample_rate_hz) + "Hz_" +
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std::to_string(num_channels) + "_channels" + "_capture",
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test_description, FormPerformanceMeasureString(capture_timer), "us",
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false);
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webrtc::test::PrintResultMeanAndError(
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"level_controller_call_durations",
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"_" + std::to_string(render_input_sample_rate_hz) + "_" +
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std::to_string(render_output_sample_rate_hz) + "_" +
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std::to_string(capture_input_sample_rate_hz) + "_" +
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std::to_string(capture_output_sample_rate_hz) + "Hz_" +
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std::to_string(num_channels) + "_channels" + "_total",
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test_description, FormPerformanceMeasureString(total_timer), "us", false);
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}
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} // namespace
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// TODO(peah): Reactivate once issue 7712 has been resolved.
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TEST(LevelControllerPerformanceTest, DISABLED_StandaloneProcessing) {
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int sample_rates_to_test[] = {
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AudioProcessing::kSampleRate8kHz, AudioProcessing::kSampleRate16kHz,
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AudioProcessing::kSampleRate32kHz, AudioProcessing::kSampleRate48kHz};
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for (auto sample_rate : sample_rates_to_test) {
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for (size_t num_channels = 1; num_channels <= 2; ++num_channels) {
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RunStandaloneSubmodule(sample_rate, num_channels);
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}
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}
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}
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void TestSomeSampleRatesWithApm(const std::string& test_name,
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bool use_mobile_agc,
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bool include_default_apm_processing) {
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// Test some stereo combinations first.
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size_t num_channels = 2;
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RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate16kHz,
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AudioProcessing::kSampleRate32kHz, num_channels,
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use_mobile_agc, include_default_apm_processing);
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RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate48kHz,
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AudioProcessing::kSampleRate8kHz, num_channels,
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use_mobile_agc, include_default_apm_processing);
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RunTogetherWithApm(test_name, 48000, 48000, 44100, 44100, num_channels,
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use_mobile_agc, include_default_apm_processing);
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// Then test mono combinations.
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num_channels = 1;
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RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate48kHz,
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AudioProcessing::kSampleRate48kHz, num_channels,
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use_mobile_agc, include_default_apm_processing);
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}
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// TODO(peah): Reactivate once issue 7712 has been resolved.
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#if !defined(WEBRTC_ANDROID)
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TEST(LevelControllerPerformanceTest, DISABLED_ProcessingViaApm) {
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#else
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TEST(LevelControllerPerformanceTest, DISABLED_ProcessingViaApm) {
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#endif
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// Run without default APM processing and desktop AGC.
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TestSomeSampleRatesWithApm("SimpleLevelControlViaApm", false, false);
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}
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// TODO(peah): Reactivate once issue 7712 has been resolved.
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#if !defined(WEBRTC_ANDROID)
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TEST(LevelControllerPerformanceTest, DISABLED_InteractionWithDefaultApm) {
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#else
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TEST(LevelControllerPerformanceTest, DISABLED_InteractionWithDefaultApm) {
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#endif
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bool include_default_apm_processing = true;
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TestSomeSampleRatesWithApm("LevelControlAndDefaultDesktopApm", false,
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include_default_apm_processing);
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TestSomeSampleRatesWithApm("LevelControlAndDefaultMobileApm", true,
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include_default_apm_processing);
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}
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} // namespace webrtc
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