blob: 8926f0223f285568a6daf993f9463bf73e7ed5ef [file] [log] [blame]
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include "modules/audio_processing/audio_processing_impl.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/test/test_utils.h"
#include "rtc_base/refcountedobject.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "test/gmock.h"
#include "test/gtest.h"
using ::testing::Invoke;
namespace webrtc {
namespace {
class MockInitialize : public AudioProcessingImpl {
explicit MockInitialize(const webrtc::Config& config)
: AudioProcessingImpl(config) {}
MOCK_METHOD0(InitializeLocked, int());
int RealInitializeLocked() RTC_NO_THREAD_SAFETY_ANALYSIS {
return AudioProcessingImpl::InitializeLocked();
MOCK_CONST_METHOD0(AddRef, void());
MOCK_CONST_METHOD0(Release, rtc::RefCountReleaseStatus());
void InitializeAudioFrame(size_t input_rate,
size_t num_channels,
AudioFrame* frame) {
const size_t samples_per_input_channel = rtc::CheckedDivExact(
input_rate, static_cast<size_t>(rtc::CheckedDivExact(
1000, AudioProcessing::kChunkSizeMs)));
RTC_DCHECK_LE(samples_per_input_channel * num_channels,
frame->samples_per_channel_ = samples_per_input_channel;
frame->sample_rate_hz_ = input_rate;
frame->num_channels_ = num_channels;
void FillFixedFrame(int16_t audio_level, AudioFrame* frame) {
const size_t num_samples = frame->samples_per_channel_ * frame->num_channels_;
for (size_t i = 0; i < num_samples; ++i) {
frame->mutable_data()[i] = audio_level;
// Mocks EchoDetector and records the first samples of the last analyzed render
// stream frame. Used to check what data is read by an EchoDetector
// implementation injected into an APM.
class TestEchoDetector : public EchoDetector {
: analyze_render_audio_called_(false),
last_render_audio_first_sample_(0.f) {}
~TestEchoDetector() override = default;
void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) override {
last_render_audio_first_sample_ = render_audio[0];
analyze_render_audio_called_ = true;
void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio) override {
void Initialize(int capture_sample_rate_hz,
int num_capture_channels,
int render_sample_rate_hz,
int num_render_channels) override {}
EchoDetector::Metrics GetMetrics() const override { return {}; }
// Returns true if AnalyzeRenderAudio() has been called at least once.
bool analyze_render_audio_called() const {
return analyze_render_audio_called_;
// Returns the first sample of the last analyzed render frame.
float last_render_audio_first_sample() const {
return last_render_audio_first_sample_;
bool analyze_render_audio_called_;
float last_render_audio_first_sample_;
// Mocks CustomProcessing and applies ProcessSample() to all the samples.
// Meant to be injected into an APM to modify samples in a known and detectable
// way.
class TestRenderPreProcessor : public CustomProcessing {
TestRenderPreProcessor() = default;
~TestRenderPreProcessor() = default;
void Initialize(int sample_rate_hz, int num_channels) override {}
void Process(AudioBuffer* audio) override {
for (size_t k = 0; k < audio->num_channels(); ++k) {
rtc::ArrayView<float> channel_view(audio->channels_f()[k],
std::transform(channel_view.begin(), channel_view.end(),
channel_view.begin(), ProcessSample);
std::string ToString() const override { return "TestRenderPreProcessor"; }
void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting) override {}
// Modifies a sample. This member is used in Process() to modify a frame and
// it is publicly visible to enable tests.
static constexpr float ProcessSample(float x) { return 2.f * x; }
} // namespace
TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
webrtc::Config config;
MockInitialize mock(config);
ON_CALL(mock, InitializeLocked())
.WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked));
EXPECT_CALL(mock, InitializeLocked()).Times(1);
AudioFrame frame;
// Call with the default parameters; there should be an init.
frame.num_channels_ = 1;
SetFrameSampleRate(&frame, 16000);
EXPECT_CALL(mock, InitializeLocked()).Times(0);
// New sample rate. (Only impacts ProcessStream).
SetFrameSampleRate(&frame, 32000);
EXPECT_CALL(mock, InitializeLocked()).Times(1);
// New number of channels.
// TODO(peah): Investigate why this causes 2 inits.
frame.num_channels_ = 2;
EXPECT_CALL(mock, InitializeLocked()).Times(2);
// ProcessStream sets num_channels_ == num_output_channels.
frame.num_channels_ = 2;
// A new sample rate passed to ProcessReverseStream should cause an init.
SetFrameSampleRate(&frame, 16000);
EXPECT_CALL(mock, InitializeLocked()).Times(1);
TEST(AudioProcessingImplTest, UpdateCapturePreGainRuntimeSetting) {
std::unique_ptr<AudioProcessing> apm(AudioProcessingBuilder().Create());
webrtc::AudioProcessing::Config apm_config;
apm_config.pre_amplifier.enabled = true;
apm_config.pre_amplifier.fixed_gain_factor = 1.f;
AudioFrame frame;
constexpr int16_t kAudioLevel = 10000;
constexpr size_t kSampleRateHz = 48000;
constexpr size_t kNumChannels = 2;
InitializeAudioFrame(kSampleRateHz, kNumChannels, &frame);
FillFixedFrame(kAudioLevel, &frame);
EXPECT_EQ([100], kAudioLevel)
<< "With factor 1, frame shouldn't be modified.";
constexpr float kGainFactor = 2.f;
// Process for two frames to have time to ramp up gain.
for (int i = 0; i < 2; ++i) {
FillFixedFrame(kAudioLevel, &frame);
EXPECT_EQ([100], kGainFactor * kAudioLevel)
<< "Frame should be amplified.";
TEST(AudioProcessingImplTest, RenderPreProcessorBeforeEchoDetector) {
// Make sure that signal changes caused by a render pre-processing sub-module
// take place before any echo detector analysis.
rtc::scoped_refptr<TestEchoDetector> test_echo_detector(
new rtc::RefCountedObject<TestEchoDetector>());
std::unique_ptr<CustomProcessing> test_render_pre_processor(
new TestRenderPreProcessor());
// Create APM injecting the test echo detector and render pre-processor.
std::unique_ptr<AudioProcessing> apm(
webrtc::AudioProcessing::Config apm_config;
apm_config.pre_amplifier.enabled = true;
apm_config.residual_echo_detector.enabled = true;
constexpr int16_t kAudioLevel = 1000;
constexpr int kSampleRateHz = 16000;
constexpr size_t kNumChannels = 1;
AudioFrame frame;
InitializeAudioFrame(kSampleRateHz, kNumChannels, &frame);
constexpr float kAudioLevelFloat = static_cast<float>(kAudioLevel);
constexpr float kExpectedPreprocessedAudioLevel =
ASSERT_NE(kAudioLevelFloat, kExpectedPreprocessedAudioLevel);
// Analyze a render stream frame.
FillFixedFrame(kAudioLevel, &frame);
// Trigger a call to in EchoDetector::AnalyzeRenderAudio() via
// ProcessStream().
FillFixedFrame(kAudioLevel, &frame);
ASSERT_EQ(AudioProcessing::Error::kNoError, apm->ProcessStream(&frame));
// Regardless of how the call to in EchoDetector::AnalyzeRenderAudio() is
// triggered, the line below checks that the call has occurred. If not, the
// APM implementation may have changed and this test might need to be adapted.
// Check that the data read in EchoDetector::AnalyzeRenderAudio() is that
// produced by the render pre-processor.
} // namespace webrtc