Add test to verify identical input channels result in identical output channels.
BUG=issue411 TEST=audioproc_unittest, trybots Review URL: https://webrtc-codereview.appspot.com/553009 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2182 4adac7df-926f-26a2-2b94-8c16560cd09d
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andrew@webrtc.org
parent
329fcbba29
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07bf9a07f5
@ -10,6 +10,8 @@
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#include <stdio.h>
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#include <algorithm>
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#include "gtest/gtest.h"
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#include "audio_processing.h"
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@ -55,6 +57,15 @@ const size_t kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
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const int kChannels[] = {1, 2};
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const size_t kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
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#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
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// AECM doesn't support super-wb.
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const int kProcessSampleRates[] = {8000, 16000};
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#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
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const int kProcessSampleRates[] = {8000, 16000, 32000};
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#endif
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const size_t kProcessSampleRatesSize = sizeof(kProcessSampleRates) /
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sizeof(*kProcessSampleRates);
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class ApmTest : public ::testing::Test {
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protected:
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ApmTest();
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@ -81,6 +92,8 @@ class ApmTest : public ::testing::Test {
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int num_reverse_channels,
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int num_input_channels,
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int num_output_channels);
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void EnableAllComponents();
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bool ReadFrame(FILE* file, AudioFrame* frame);
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const std::string output_path_;
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const std::string ref_path_;
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@ -238,14 +251,74 @@ void MixStereoToMono(const int16_t* stereo,
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}
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}
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template <class T>
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T MaxValue(T a, T b) {
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return a > b ? a : b;
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void CopyLeftToRightChannel(int16_t* stereo, int samples_per_channel) {
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for (int i = 0; i < samples_per_channel; i++) {
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stereo[i * 2 + 1] = stereo[i * 2];
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}
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}
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template <class T>
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T AbsValue(T a) {
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return a > 0 ? a : -a;
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void VerifyChannelsAreEqual(int16_t* stereo, int samples_per_channel) {
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for (int i = 0; i < samples_per_channel; i++) {
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EXPECT_EQ(stereo[i * 2 + 1], stereo[i * 2]);
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}
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}
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void ApmTest::EnableAllComponents() {
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#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
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EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
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EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->gain_control()->set_mode(GainControl::kAdaptiveDigital));
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EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
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#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
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EXPECT_EQ(apm_->kNoError,
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apm_->echo_cancellation()->enable_drift_compensation(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->echo_cancellation()->enable_metrics(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->echo_cancellation()->enable_delay_logging(true));
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EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
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EXPECT_EQ(apm_->kNoError,
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apm_->gain_control()->set_analog_level_limits(0, 255));
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EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
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#endif
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EXPECT_EQ(apm_->kNoError,
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apm_->high_pass_filter()->Enable(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->level_estimator()->Enable(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->noise_suppression()->Enable(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->voice_detection()->Enable(true));
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}
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bool ApmTest::ReadFrame(FILE* file, AudioFrame* frame) {
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// The files always contain stereo audio.
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size_t frame_size = frame->samples_per_channel_ * 2;
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size_t read_count = fread(frame->data_,
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sizeof(int16_t),
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frame_size,
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file);
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if (read_count != frame_size) {
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// Check that the file really ended.
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EXPECT_NE(0, feof(file));
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return false; // This is expected.
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}
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if (frame->num_channels_ == 1) {
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MixStereoToMono(frame->data_, frame->data_,
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frame->samples_per_channel_);
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}
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return true;
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}
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void SetFrameTo(AudioFrame* frame, int16_t value) {
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@ -263,14 +336,19 @@ void SetFrameTo(AudioFrame* frame, int16_t left, int16_t right) {
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}
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}
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template <class T>
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T AbsValue(T a) {
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return a > 0 ? a: -a;
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}
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int16_t MaxAudioFrame(const AudioFrame& frame) {
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const int length = frame.samples_per_channel_ * frame.num_channels_;
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int16_t max = AbsValue(frame.data_[0]);
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int16_t max_data = AbsValue(frame.data_[0]);
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for (int i = 1; i < length; i++) {
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max = MaxValue(max, AbsValue(frame.data_[i]));
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max_data = std::max(max_data, AbsValue(frame.data_[i]));
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}
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return max;
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return max_data;
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}
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bool FrameDataAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) {
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@ -1009,6 +1087,35 @@ TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabled) {
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}
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}
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TEST_F(ApmTest, IdenticalInputChannelsResultInIdenticalOutputChannels) {
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EnableAllComponents();
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for (size_t i = 0; i < kProcessSampleRatesSize; i++) {
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Init(kProcessSampleRates[i], 2, 2, 2, false);
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int analog_level = 127;
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while (1) {
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if (!ReadFrame(far_file_, revframe_)) break;
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CopyLeftToRightChannel(revframe_->data_, revframe_->samples_per_channel_);
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EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
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if (!ReadFrame(near_file_, frame_)) break;
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CopyLeftToRightChannel(frame_->data_, frame_->samples_per_channel_);
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frame_->vad_activity_ = AudioFrame::kVadUnknown;
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EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
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EXPECT_EQ(apm_->kNoError,
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apm_->echo_cancellation()->set_stream_drift_samples(0));
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EXPECT_EQ(apm_->kNoError,
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apm_->gain_control()->set_stream_analog_level(analog_level));
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EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
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analog_level = apm_->gain_control()->stream_analog_level();
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VerifyChannelsAreEqual(frame_->data_, frame_->samples_per_channel_);
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}
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}
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}
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TEST_F(ApmTest, SplittingFilter) {
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// Verify the filter is not active through undistorted audio when:
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// 1. No components are enabled...
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@ -1129,14 +1236,6 @@ TEST_F(ApmTest, Process) {
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ReadMessageLiteFromFile(ref_filename_, &ref_data);
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} else {
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// Write the desired tests to the protobuf reference file.
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#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
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// AECM doesn't support super-wb.
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const int kProcessSampleRates[] = {8000, 16000};
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#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
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const int kProcessSampleRates[] = {8000, 16000, 32000};
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#endif
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const size_t kProcessSampleRatesSize = sizeof(kProcessSampleRates) /
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sizeof(*kProcessSampleRates);
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for (size_t i = 0; i < kChannelsSize; i++) {
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for (size_t j = 0; j < kChannelsSize; j++) {
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// We can't have more output than input channels.
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@ -1153,40 +1252,7 @@ TEST_F(ApmTest, Process) {
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}
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}
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#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
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EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
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EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->gain_control()->set_mode(GainControl::kAdaptiveDigital));
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EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
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#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
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EXPECT_EQ(apm_->kNoError,
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apm_->echo_cancellation()->enable_drift_compensation(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->echo_cancellation()->enable_metrics(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->echo_cancellation()->enable_delay_logging(true));
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EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
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EXPECT_EQ(apm_->kNoError,
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apm_->gain_control()->set_analog_level_limits(0, 255));
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EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
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#endif
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EXPECT_EQ(apm_->kNoError,
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apm_->high_pass_filter()->Enable(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->level_estimator()->Enable(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->noise_suppression()->Enable(true));
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EXPECT_EQ(apm_->kNoError,
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apm_->voice_detection()->Enable(true));
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EnableAllComponents();
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for (int i = 0; i < ref_data.test_size(); i++) {
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printf("Running test %d of %d...\n", i + 1, ref_data.test_size());
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@ -1195,7 +1261,6 @@ TEST_F(ApmTest, Process) {
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Init(test->sample_rate(), test->num_reverse_channels(),
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test->num_input_channels(), test->num_output_channels(), true);
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const int samples_per_channel = test->sample_rate() / 100;
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int frame_count = 0;
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int has_echo_count = 0;
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int has_voice_count = 0;
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@ -1205,48 +1270,18 @@ TEST_F(ApmTest, Process) {
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int max_output_average = 0;
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while (1) {
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// Read far-end frame, always stereo.
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size_t frame_size = samples_per_channel * 2;
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size_t read_count = fread(revframe_->data_,
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sizeof(int16_t),
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frame_size,
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far_file_);
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if (read_count != frame_size) {
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// Check that the file really ended.
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ASSERT_NE(0, feof(far_file_));
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break; // This is expected.
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}
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if (revframe_->num_channels_ == 1) {
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MixStereoToMono(revframe_->data_, revframe_->data_,
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samples_per_channel);
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}
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if (!ReadFrame(far_file_, revframe_)) break;
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EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
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if (!ReadFrame(near_file_, frame_)) break;
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frame_->vad_activity_ = AudioFrame::kVadUnknown;
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EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
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EXPECT_EQ(apm_->kNoError,
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apm_->echo_cancellation()->set_stream_drift_samples(0));
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EXPECT_EQ(apm_->kNoError,
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apm_->gain_control()->set_stream_analog_level(analog_level));
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// Read near-end frame, always stereo.
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read_count = fread(frame_->data_,
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sizeof(int16_t),
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frame_size,
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near_file_);
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if (read_count != frame_size) {
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// Check that the file really ended.
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ASSERT_NE(0, feof(near_file_));
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break; // This is expected.
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}
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if (frame_->num_channels_ == 1) {
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MixStereoToMono(frame_->data_, frame_->data_,
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samples_per_channel);
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}
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frame_->vad_activity_ = AudioFrame::kVadUnknown;
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EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
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// Ensure the frame was downmixed properly.
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EXPECT_EQ(test->num_output_channels(), frame_->num_channels_);
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@ -1269,7 +1304,7 @@ TEST_F(ApmTest, Process) {
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EXPECT_EQ(AudioFrame::kVadPassive, frame_->vad_activity_);
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}
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frame_size = samples_per_channel * frame_->num_channels_;
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size_t frame_size = frame_->samples_per_channel_ * frame_->num_channels_;
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size_t write_count = fwrite(frame_->data_,
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sizeof(int16_t),
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frame_size,
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