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webrtc / src / 3f75209656c5081158e0b80e9919cb286ac2a872 / . / modules / audio_processing / audio_processing_unittest.cc
blob: 199f3780df3c1bdfa9d03ccd301461ea55223264 [file] [log] [blame]
/*
* Copyright (c) 2012 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/include/audio_processing.h"
#include <math.h>
#include <stdio.h>
#include <algorithm>
#include <cmath>
#include <limits>
#include <memory>
#include <numeric>
#include <queue>
#include "absl/flags/flag.h"
#include "common_audio/include/audio_util.h"
#include "common_audio/resampler/include/push_resampler.h"
#include "common_audio/resampler/push_sinc_resampler.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/audio_processing_impl.h"
#include "modules/audio_processing/common.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "modules/audio_processing/test/protobuf_utils.h"
#include "modules/audio_processing/test/test_utils.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/fake_clock.h"
#include "rtc_base/gtest_prod_util.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/protobuf_utils.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/swap_queue.h"
#include "rtc_base/system/arch.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/thread.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_processing/test/unittest.pb.h"
#else
#include "modules/audio_processing/test/unittest.pb.h"
#endif
RTC_POP_IGNORING_WUNDEF()
ABSL_FLAG(bool,
write_apm_ref_data,
false,
"Write ApmTest.Process results to file, instead of comparing results "
"to the existing reference data file.");
namespace webrtc {
namespace {
// TODO(ekmeyerson): Switch to using StreamConfig and ProcessingConfig where
// applicable.
const int32_t kChannels[] = {1, 2};
const int kSampleRates[] = {8000, 16000, 32000, 48000};
#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
// Android doesn't support 48kHz.
const int kProcessSampleRates[] = {8000, 16000, 32000};
#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
const int kProcessSampleRates[] = {8000, 16000, 32000, 48000};
#endif
enum StreamDirection { kForward = 0, kReverse };
void ConvertToFloat(const int16_t* int_data, ChannelBuffer<float>* cb) {
ChannelBuffer<int16_t> cb_int(cb->num_frames(), cb->num_channels());
Deinterleave(int_data, cb->num_frames(), cb->num_channels(),
cb_int.channels());
for (size_t i = 0; i < cb->num_channels(); ++i) {
S16ToFloat(cb_int.channels()[i], cb->num_frames(), cb->channels()[i]);
}
}
void ConvertToFloat(const AudioFrame& frame, ChannelBuffer<float>* cb) {
ConvertToFloat(frame.data(), cb);
}
// Number of channels including the keyboard channel.
size_t TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
switch (layout) {
case AudioProcessing::kMono:
return 1;
case AudioProcessing::kMonoAndKeyboard:
case AudioProcessing::kStereo:
return 2;
case AudioProcessing::kStereoAndKeyboard:
return 3;
}
RTC_NOTREACHED();
return 0;
}
void MixStereoToMono(const float* stereo,
float* mono,
size_t samples_per_channel) {
for (size_t i = 0; i < samples_per_channel; ++i)
mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) / 2;
}
void MixStereoToMono(const int16_t* stereo,
int16_t* mono,
size_t samples_per_channel) {
for (size_t i = 0; i < samples_per_channel; ++i)
mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) >> 1;
}
void CopyLeftToRightChannel(int16_t* stereo, size_t samples_per_channel) {
for (size_t i = 0; i < samples_per_channel; i++) {
stereo[i * 2 + 1] = stereo[i * 2];
}
}
void VerifyChannelsAreEqual(const int16_t* stereo, size_t samples_per_channel) {
for (size_t i = 0; i < samples_per_channel; i++) {
EXPECT_EQ(stereo[i * 2 + 1], stereo[i * 2]);
}
}
void SetFrameTo(AudioFrame* frame, int16_t value) {
int16_t* frame_data = frame->mutable_data();
for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
++i) {
frame_data[i] = value;
}
}
void SetFrameTo(AudioFrame* frame, int16_t left, int16_t right) {
ASSERT_EQ(2u, frame->num_channels_);
int16_t* frame_data = frame->mutable_data();
for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
frame_data[i] = left;
frame_data[i + 1] = right;
}
}
void ScaleFrame(AudioFrame* frame, float scale) {
int16_t* frame_data = frame->mutable_data();
for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
++i) {
frame_data[i] = FloatS16ToS16(frame_data[i] * scale);
}
}
bool FrameDataAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) {
if (frame1.samples_per_channel_ != frame2.samples_per_channel_) {
return false;
}
if (frame1.num_channels_ != frame2.num_channels_) {
return false;
}
if (memcmp(frame1.data(), frame2.data(),
frame1.samples_per_channel_ * frame1.num_channels_ *
sizeof(int16_t))) {
return false;
}
return true;
}
rtc::ArrayView<int16_t> GetMutableFrameData(AudioFrame* frame) {
int16_t* ptr = frame->mutable_data();
const size_t len = frame->samples_per_channel() * frame->num_channels();
return rtc::ArrayView<int16_t>(ptr, len);
}
rtc::ArrayView<const int16_t> GetFrameData(const AudioFrame& frame) {
const int16_t* ptr = frame.data();
const size_t len = frame.samples_per_channel() * frame.num_channels();
return rtc::ArrayView<const int16_t>(ptr, len);
}
void EnableAllAPComponents(AudioProcessing* ap) {
AudioProcessing::Config apm_config = ap->GetConfig();
apm_config.echo_canceller.enabled = true;
#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
apm_config.echo_canceller.mobile_mode = true;
apm_config.gain_controller1.enabled = true;
apm_config.gain_controller1.mode =
AudioProcessing::Config::GainController1::kAdaptiveDigital;
#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
apm_config.echo_canceller.mobile_mode = false;
apm_config.gain_controller1.enabled = true;
apm_config.gain_controller1.mode =
AudioProcessing::Config::GainController1::kAdaptiveAnalog;
apm_config.gain_controller1.analog_level_minimum = 0;
apm_config.gain_controller1.analog_level_maximum = 255;
#endif
apm_config.noise_suppression.enabled = true;
apm_config.high_pass_filter.enabled = true;
apm_config.level_estimation.enabled = true;
apm_config.voice_detection.enabled = true;
ap->ApplyConfig(apm_config);
}
// These functions are only used by ApmTest.Process.
template <class T>
T AbsValue(T a) {
return a > 0 ? a : -a;
}
int16_t MaxAudioFrame(const AudioFrame& frame) {
const size_t length = frame.samples_per_channel_ * frame.num_channels_;
const int16_t* frame_data = frame.data();
int16_t max_data = AbsValue(frame_data[0]);
for (size_t i = 1; i < length; i++) {
max_data = std::max(max_data, AbsValue(frame_data[i]));
}
return max_data;
}
void OpenFileAndWriteMessage(const std::string& filename,
const MessageLite& msg) {
FILE* file = fopen(filename.c_str(), "wb");
ASSERT_TRUE(file != NULL);
int32_t size = rtc::checked_cast<int32_t>(msg.ByteSizeLong());
ASSERT_GT(size, 0);
std::unique_ptr<uint8_t[]> array(new uint8_t[size]);
ASSERT_TRUE(msg.SerializeToArray(array.get(), size));
ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
ASSERT_EQ(static_cast<size_t>(size),
fwrite(array.get(), sizeof(array[0]), size, file));
fclose(file);
}
std::string ResourceFilePath(const std::string& name, int sample_rate_hz) {
rtc::StringBuilder ss;
// Resource files are all stereo.
ss << name << sample_rate_hz / 1000 << "_stereo";
return test::ResourcePath(ss.str(), "pcm");
}
// Temporary filenames unique to this process. Used to be able to run these
// tests in parallel as each process needs to be running in isolation they can't
// have competing filenames.
std::map<std::string, std::string> temp_filenames;
std::string OutputFilePath(const std::string& name,
int input_rate,
int output_rate,
int reverse_input_rate,
int reverse_output_rate,
size_t num_input_channels,
size_t num_output_channels,
size_t num_reverse_input_channels,
size_t num_reverse_output_channels,
StreamDirection file_direction) {
rtc::StringBuilder ss;
ss << name << "_i" << num_input_channels << "_" << input_rate / 1000 << "_ir"
<< num_reverse_input_channels << "_" << reverse_input_rate / 1000 << "_";
if (num_output_channels == 1) {
ss << "mono";
} else if (num_output_channels == 2) {
ss << "stereo";
} else {
RTC_NOTREACHED();
}
ss << output_rate / 1000;
if (num_reverse_output_channels == 1) {
ss << "_rmono";
} else if (num_reverse_output_channels == 2) {
ss << "_rstereo";
} else {
RTC_NOTREACHED();
}
ss << reverse_output_rate / 1000;
ss << "_d" << file_direction << "_pcm";
std::string filename = ss.str();
if (temp_filenames[filename].empty())
temp_filenames[filename] = test::TempFilename(test::OutputPath(), filename);
return temp_filenames[filename];
}
void ClearTempFiles() {
for (auto& kv : temp_filenames)
remove(kv.second.c_str());
}
// Only remove "out" files. Keep "ref" files.
void ClearTempOutFiles() {
for (auto it = temp_filenames.begin(); it != temp_filenames.end();) {
const std::string& filename = it->first;
if (filename.substr(0, 3).compare("out") == 0) {
remove(it->second.c_str());
temp_filenames.erase(it++);
} else {
it++;
}
}
}
void OpenFileAndReadMessage(const std::string& filename, MessageLite* msg) {
FILE* file = fopen(filename.c_str(), "rb");
ASSERT_TRUE(file != NULL);
ReadMessageFromFile(file, msg);
fclose(file);
}
// Reads a 10 ms chunk of int16 interleaved audio from the given (assumed
// stereo) file, converts to deinterleaved float (optionally downmixing) and
// returns the result in |cb|. Returns false if the file ended (or on error) and
// true otherwise.
//
// |int_data| and |float_data| are just temporary space that must be
// sufficiently large to hold the 10 ms chunk.
bool ReadChunk(FILE* file,
int16_t* int_data,
float* float_data,
ChannelBuffer<float>* cb) {
// The files always contain stereo audio.
size_t frame_size = cb->num_frames() * 2;
size_t read_count = fread(int_data, sizeof(int16_t), frame_size, file);
if (read_count != frame_size) {
// Check that the file really ended.
RTC_DCHECK(feof(file));
return false; // This is expected.
}
S16ToFloat(int_data, frame_size, float_data);
if (cb->num_channels() == 1) {
MixStereoToMono(float_data, cb->channels()[0], cb->num_frames());
} else {
Deinterleave(float_data, cb->num_frames(), 2, cb->channels());
}
return true;
}
class ApmTest : public ::testing::Test {
protected:
ApmTest();
virtual void SetUp();
virtual void TearDown();
static void SetUpTestSuite() {}
static void TearDownTestSuite() { ClearTempFiles(); }
// Used to select between int and float interface tests.
enum Format { kIntFormat, kFloatFormat };
void Init(int sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
size_t num_input_channels,
size_t num_output_channels,
size_t num_reverse_channels,
bool open_output_file);
void Init(AudioProcessing* ap);
void EnableAllComponents();
bool ReadFrame(FILE* file, AudioFrame* frame);
bool ReadFrame(FILE* file, AudioFrame* frame, ChannelBuffer<float>* cb);
void ReadFrameWithRewind(FILE* file, AudioFrame* frame);
void ReadFrameWithRewind(FILE* file,
AudioFrame* frame,
ChannelBuffer<float>* cb);
void ProcessWithDefaultStreamParameters(AudioFrame* frame);
void ProcessDelayVerificationTest(int delay_ms,
int system_delay_ms,
int delay_min,
int delay_max);
void TestChangingChannelsInt16Interface(
size_t num_channels,
AudioProcessing::Error expected_return);
void TestChangingForwardChannels(size_t num_in_channels,
size_t num_out_channels,
AudioProcessing::Error expected_return);
void TestChangingReverseChannels(size_t num_rev_channels,
AudioProcessing::Error expected_return);
void RunQuantizedVolumeDoesNotGetStuckTest(int sample_rate);
void RunManualVolumeChangeIsPossibleTest(int sample_rate);
void StreamParametersTest(Format format);
int ProcessStreamChooser(Format format);
int AnalyzeReverseStreamChooser(Format format);
void ProcessDebugDump(const std::string& in_filename,
const std::string& out_filename,
Format format,
int max_size_bytes);
void VerifyDebugDumpTest(Format format);
const std::string output_path_;
const std::string ref_filename_;
std::unique_ptr<AudioProcessing> apm_;
AudioFrame frame_;
AudioFrame revframe_;
std::unique_ptr<ChannelBuffer<float> > float_cb_;
std::unique_ptr<ChannelBuffer<float> > revfloat_cb_;
int output_sample_rate_hz_;
size_t num_output_channels_;
FILE* far_file_;
FILE* near_file_;
FILE* out_file_;
};
ApmTest::ApmTest()
: output_path_(test::OutputPath()),
#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
ref_filename_(
test::ResourcePath("audio_processing/output_data_fixed", "pb")),
#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
ref_filename_(
test::ResourcePath("audio_processing/output_data_float", "pb")),
#endif
output_sample_rate_hz_(0),
num_output_channels_(0),
far_file_(NULL),
near_file_(NULL),
out_file_(NULL) {
Config config;
config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
apm_.reset(AudioProcessingBuilder().Create(config));
}
void ApmTest::SetUp() {
ASSERT_TRUE(apm_.get() != NULL);
Init(32000, 32000, 32000, 2, 2, 2, false);
}
void ApmTest::TearDown() {
if (far_file_) {
ASSERT_EQ(0, fclose(far_file_));
}
far_file_ = NULL;
if (near_file_) {
ASSERT_EQ(0, fclose(near_file_));
}
near_file_ = NULL;
if (out_file_) {
ASSERT_EQ(0, fclose(out_file_));
}
out_file_ = NULL;
}
void ApmTest::Init(AudioProcessing* ap) {
ASSERT_EQ(
kNoErr,
ap->Initialize({{{frame_.sample_rate_hz_, frame_.num_channels_},
{output_sample_rate_hz_, num_output_channels_},
{revframe_.sample_rate_hz_, revframe_.num_channels_},
{revframe_.sample_rate_hz_, revframe_.num_channels_}}}));
}
void ApmTest::Init(int sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
size_t num_input_channels,
size_t num_output_channels,
size_t num_reverse_channels,
bool open_output_file) {
SetContainerFormat(sample_rate_hz, num_input_channels, &frame_, &float_cb_);
output_sample_rate_hz_ = output_sample_rate_hz;
num_output_channels_ = num_output_channels;
SetContainerFormat(reverse_sample_rate_hz, num_reverse_channels, &revframe_,
&revfloat_cb_);
Init(apm_.get());
if (far_file_) {
ASSERT_EQ(0, fclose(far_file_));
}
std::string filename = ResourceFilePath("far", sample_rate_hz);
far_file_ = fopen(filename.c_str(), "rb");
ASSERT_TRUE(far_file_ != NULL) << "Could not open file " << filename << "\n";
if (near_file_) {
ASSERT_EQ(0, fclose(near_file_));
}
filename = ResourceFilePath("near", sample_rate_hz);
near_file_ = fopen(filename.c_str(), "rb");
ASSERT_TRUE(near_file_ != NULL) << "Could not open file " << filename << "\n";
if (open_output_file) {
if (out_file_) {
ASSERT_EQ(0, fclose(out_file_));
}
filename = OutputFilePath(
"out", sample_rate_hz, output_sample_rate_hz, reverse_sample_rate_hz,
reverse_sample_rate_hz, num_input_channels, num_output_channels,
num_reverse_channels, num_reverse_channels, kForward);
out_file_ = fopen(filename.c_str(), "wb");
ASSERT_TRUE(out_file_ != NULL)
<< "Could not open file " << filename << "\n";
}
}
void ApmTest::EnableAllComponents() {
EnableAllAPComponents(apm_.get());
}
bool ApmTest::ReadFrame(FILE* file,
AudioFrame* frame,
ChannelBuffer<float>* cb) {
// The files always contain stereo audio.
size_t frame_size = frame->samples_per_channel_ * 2;
size_t read_count =
fread(frame->mutable_data(), sizeof(int16_t), frame_size, file);
if (read_count != frame_size) {
// Check that the file really ended.
EXPECT_NE(0, feof(file));
return false; // This is expected.
}
if (frame->num_channels_ == 1) {
MixStereoToMono(frame->data(), frame->mutable_data(),
frame->samples_per_channel_);
}
if (cb) {
ConvertToFloat(*frame, cb);
}
return true;
}
bool ApmTest::ReadFrame(FILE* file, AudioFrame* frame) {
return ReadFrame(file, frame, NULL);
}
// If the end of the file has been reached, rewind it and attempt to read the
// frame again.
void ApmTest::ReadFrameWithRewind(FILE* file,
AudioFrame* frame,
ChannelBuffer<float>* cb) {
if (!ReadFrame(near_file_, &frame_, cb)) {
rewind(near_file_);
ASSERT_TRUE(ReadFrame(near_file_, &frame_, cb));
}
}
void ApmTest::ReadFrameWithRewind(FILE* file, AudioFrame* frame) {
ReadFrameWithRewind(file, frame, NULL);
}
void ApmTest::ProcessWithDefaultStreamParameters(AudioFrame* frame) {
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
apm_->set_stream_analog_level(127);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame));
}
int ApmTest::ProcessStreamChooser(Format format) {
if (format == kIntFormat) {
return apm_->ProcessStream(&frame_);
}
return apm_->ProcessStream(
float_cb_->channels(),
StreamConfig(frame_.sample_rate_hz_, frame_.num_channels_),
StreamConfig(output_sample_rate_hz_, num_output_channels_),
float_cb_->channels());
}
int ApmTest::AnalyzeReverseStreamChooser(Format format) {
if (format == kIntFormat) {
return apm_->ProcessReverseStream(&revframe_);
}
return apm_->AnalyzeReverseStream(
revfloat_cb_->channels(),
StreamConfig(revframe_.sample_rate_hz_, revframe_.num_channels_));
}
void ApmTest::ProcessDelayVerificationTest(int delay_ms,
int system_delay_ms,
int delay_min,
int delay_max) {
// The |revframe_| and |frame_| should include the proper frame information,
// hence can be used for extracting information.
AudioFrame tmp_frame;
std::queue<AudioFrame*> frame_queue;
bool causal = true;
tmp_frame.CopyFrom(revframe_);
SetFrameTo(&tmp_frame, 0);
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
// Initialize the |frame_queue| with empty frames.
int frame_delay = delay_ms / 10;
while (frame_delay < 0) {
AudioFrame* frame = new AudioFrame();
frame->CopyFrom(tmp_frame);
frame_queue.push(frame);
frame_delay++;
causal = false;
}
while (frame_delay > 0) {
AudioFrame* frame = new AudioFrame();
frame->CopyFrom(tmp_frame);
frame_queue.push(frame);
frame_delay--;
}
// Run for 4.5 seconds, skipping statistics from the first 2.5 seconds. We
// need enough frames with audio to have reliable estimates, but as few as
// possible to keep processing time down. 4.5 seconds seemed to be a good
// compromise for this recording.
for (int frame_count = 0; frame_count < 450; ++frame_count) {
AudioFrame* frame = new AudioFrame();
frame->CopyFrom(tmp_frame);
// Use the near end recording, since that has more speech in it.
ASSERT_TRUE(ReadFrame(near_file_, frame));
frame_queue.push(frame);
AudioFrame* reverse_frame = frame;
AudioFrame* process_frame = frame_queue.front();
if (!causal) {
reverse_frame = frame_queue.front();
// When we call ProcessStream() the frame is modified, so we can't use the
// pointer directly when things are non-causal. Use an intermediate frame
// and copy the data.
process_frame = &tmp_frame;
process_frame->CopyFrom(*frame);
}
EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(reverse_frame));
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(system_delay_ms));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(process_frame));
frame = frame_queue.front();
frame_queue.pop();
delete frame;
if (frame_count == 250) {
// Discard the first delay metrics to avoid convergence effects.
static_cast<void>(apm_->GetStatistics(true /* has_remote_tracks */));
}
}
rewind(near_file_);
while (!frame_queue.empty()) {
AudioFrame* frame = frame_queue.front();
frame_queue.pop();
delete frame;
}
// Calculate expected delay estimate and acceptable regions. Further,
// limit them w.r.t. AEC delay estimation support.
const size_t samples_per_ms =
rtc::SafeMin<size_t>(16u, frame_.samples_per_channel_ / 10);
const int expected_median =
rtc::SafeClamp<int>(delay_ms - system_delay_ms, delay_min, delay_max);
const int expected_median_high = rtc::SafeClamp<int>(
expected_median + rtc::dchecked_cast<int>(96 / samples_per_ms), delay_min,
delay_max);
const int expected_median_low = rtc::SafeClamp<int>(
expected_median - rtc::dchecked_cast<int>(96 / samples_per_ms), delay_min,
delay_max);
// Verify delay metrics.
AudioProcessingStats stats =
apm_->GetStatistics(true /* has_remote_tracks */);
ASSERT_TRUE(stats.delay_median_ms.has_value());
int32_t median = *stats.delay_median_ms;
EXPECT_GE(expected_median_high, median);
EXPECT_LE(expected_median_low, median);
}
void ApmTest::StreamParametersTest(Format format) {
// No errors when the components are disabled.
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
// -- Missing AGC level --
AudioProcessing::Config apm_config = apm_->GetConfig();
apm_config.gain_controller1.enabled = true;
apm_->ApplyConfig(apm_config);
EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format));
// Resets after successful ProcessStream().
apm_->set_stream_analog_level(127);
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format));
// Other stream parameters set correctly.
apm_config.echo_canceller.enabled = true;
apm_config.echo_canceller.mobile_mode = false;
apm_->ApplyConfig(apm_config);
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format));
apm_config.gain_controller1.enabled = false;
apm_->ApplyConfig(apm_config);
// -- Missing delay --
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
// Resets after successful ProcessStream().
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
// Other stream parameters set correctly.
apm_config.gain_controller1.enabled = true;
apm_->ApplyConfig(apm_config);
apm_->set_stream_analog_level(127);
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
apm_config.gain_controller1.enabled = false;
apm_->ApplyConfig(apm_config);
// -- No stream parameters --
EXPECT_EQ(apm_->kNoError, AnalyzeReverseStreamChooser(format));
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
// -- All there --
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
apm_->set_stream_analog_level(127);
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
}
TEST_F(ApmTest, StreamParametersInt) {
StreamParametersTest(kIntFormat);
}
TEST_F(ApmTest, StreamParametersFloat) {
StreamParametersTest(kFloatFormat);
}
TEST_F(ApmTest, DefaultDelayOffsetIsZero) {
EXPECT_EQ(0, apm_->delay_offset_ms());
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(50));
EXPECT_EQ(50, apm_->stream_delay_ms());
}
TEST_F(ApmTest, DelayOffsetWithLimitsIsSetProperly) {
// High limit of 500 ms.
apm_->set_delay_offset_ms(100);
EXPECT_EQ(100, apm_->delay_offset_ms());
EXPECT_EQ(apm_->kBadStreamParameterWarning, apm_->set_stream_delay_ms(450));
EXPECT_EQ(500, apm_->stream_delay_ms());
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
EXPECT_EQ(200, apm_->stream_delay_ms());
// Low limit of 0 ms.
apm_->set_delay_offset_ms(-50);
EXPECT_EQ(-50, apm_->delay_offset_ms());
EXPECT_EQ(apm_->kBadStreamParameterWarning, apm_->set_stream_delay_ms(20));
EXPECT_EQ(0, apm_->stream_delay_ms());
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
EXPECT_EQ(50, apm_->stream_delay_ms());
}
void ApmTest::TestChangingChannelsInt16Interface(
size_t num_channels,
AudioProcessing::Error expected_return) {
frame_.num_channels_ = num_channels;
EXPECT_EQ(expected_return, apm_->ProcessStream(&frame_));
EXPECT_EQ(expected_return, apm_->ProcessReverseStream(&frame_));
}
void ApmTest::TestChangingForwardChannels(
size_t num_in_channels,
size_t num_out_channels,
AudioProcessing::Error expected_return) {
const StreamConfig input_stream = {frame_.sample_rate_hz_, num_in_channels};
const StreamConfig output_stream = {output_sample_rate_hz_, num_out_channels};
EXPECT_EQ(expected_return,
apm_->ProcessStream(float_cb_->channels(), input_stream,
output_stream, float_cb_->channels()));
}
void ApmTest::TestChangingReverseChannels(
size_t num_rev_channels,
AudioProcessing::Error expected_return) {
const ProcessingConfig processing_config = {
{{frame_.sample_rate_hz_, apm_->num_input_channels()},
{output_sample_rate_hz_, apm_->num_output_channels()},
{frame_.sample_rate_hz_, num_rev_channels},
{frame_.sample_rate_hz_, num_rev_channels}}};
EXPECT_EQ(
expected_return,
apm_->ProcessReverseStream(
float_cb_->channels(), processing_config.reverse_input_stream(),
processing_config.reverse_output_stream(), float_cb_->channels()));
}
TEST_F(ApmTest, ChannelsInt16Interface) {
// Testing number of invalid and valid channels.
Init(16000, 16000, 16000, 4, 4, 4, false);
TestChangingChannelsInt16Interface(0, apm_->kBadNumberChannelsError);
for (size_t i = 1; i < 4; i++) {
TestChangingChannelsInt16Interface(i, kNoErr);
EXPECT_EQ(i, apm_->num_input_channels());
}
}
TEST_F(ApmTest, Channels) {
// Testing number of invalid and valid channels.
Init(16000, 16000, 16000, 4, 4, 4, false);
TestChangingForwardChannels(0, 1, apm_->kBadNumberChannelsError);
TestChangingReverseChannels(0, apm_->kBadNumberChannelsError);
for (size_t i = 1; i < 4; ++i) {
for (size_t j = 0; j < 1; ++j) {
// Output channels much be one or match input channels.
if (j == 1 || i == j) {
TestChangingForwardChannels(i, j, kNoErr);
TestChangingReverseChannels(i, kNoErr);
EXPECT_EQ(i, apm_->num_input_channels());
EXPECT_EQ(j, apm_->num_output_channels());
// The number of reverse channels used for processing to is always 1.
EXPECT_EQ(1u, apm_->num_reverse_channels());
} else {
TestChangingForwardChannels(i, j,
AudioProcessing::kBadNumberChannelsError);
}
}
}
}
TEST_F(ApmTest, SampleRatesInt) {
// Testing invalid sample rates
SetContainerFormat(10000, 2, &frame_, &float_cb_);
EXPECT_EQ(apm_->kBadSampleRateError, ProcessStreamChooser(kIntFormat));
// Testing valid sample rates
int fs[] = {8000, 16000, 32000, 48000};
for (size_t i = 0; i < arraysize(fs); i++) {
SetContainerFormat(fs[i], 2, &frame_, &float_cb_);
EXPECT_NOERR(ProcessStreamChooser(kIntFormat));
}
}
// This test repeatedly reconfigures the pre-amplifier in APM, processes a
// number of frames, and checks that output signal has the right level.
TEST_F(ApmTest, PreAmplifier) {
// Fill the audio frame with a sawtooth pattern.
rtc::ArrayView<int16_t> frame_data = GetMutableFrameData(&frame_);
const size_t samples_per_channel = frame_.samples_per_channel();
for (size_t i = 0; i < samples_per_channel; i++) {
for (size_t ch = 0; ch < frame_.num_channels(); ++ch) {
frame_data[i + ch * samples_per_channel] = 10000 * ((i % 3) - 1);
}
}
// Cache the frame in tmp_frame.
AudioFrame tmp_frame;
tmp_frame.CopyFrom(frame_);
auto compute_power = [](const AudioFrame& frame) {
rtc::ArrayView<const int16_t> data = GetFrameData(frame);
return std::accumulate(data.begin(), data.end(), 0.0f,
[](float a, float b) { return a + b * b; }) /
data.size() / 32768 / 32768;
};
const float input_power = compute_power(tmp_frame);
// Double-check that the input data is large compared to the error kEpsilon.
constexpr float kEpsilon = 1e-4f;
RTC_DCHECK_GE(input_power, 10 * kEpsilon);
// 1. Enable pre-amp with 0 dB gain.
AudioProcessing::Config config = apm_->GetConfig();
config.pre_amplifier.enabled = true;
config.pre_amplifier.fixed_gain_factor = 1.0f;
apm_->ApplyConfig(config);
for (int i = 0; i < 20; ++i) {
frame_.CopyFrom(tmp_frame);
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat));
}
float output_power = compute_power(frame_);
EXPECT_NEAR(output_power, input_power, kEpsilon);
config = apm_->GetConfig();
EXPECT_EQ(config.pre_amplifier.fixed_gain_factor, 1.0f);
// 2. Change pre-amp gain via ApplyConfig.
config.pre_amplifier.fixed_gain_factor = 2.0f;
apm_->ApplyConfig(config);
for (int i = 0; i < 20; ++i) {
frame_.CopyFrom(tmp_frame);
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat));
}
output_power = compute_power(frame_);
EXPECT_NEAR(output_power, 4 * input_power, kEpsilon);
config = apm_->GetConfig();
EXPECT_EQ(config.pre_amplifier.fixed_gain_factor, 2.0f);
// 3. Change pre-amp gain via a RuntimeSetting.
apm_->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCapturePreGain(1.5f));
for (int i = 0; i < 20; ++i) {
frame_.CopyFrom(tmp_frame);
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat));
}
output_power = compute_power(frame_);
EXPECT_NEAR(output_power, 2.25 * input_power, kEpsilon);
config = apm_->GetConfig();
EXPECT_EQ(config.pre_amplifier.fixed_gain_factor, 1.5f);
}
TEST_F(ApmTest, GainControl) {
AudioProcessing::Config config = apm_->GetConfig();
config.gain_controller1.enabled = false;
apm_->ApplyConfig(config);
config.gain_controller1.enabled = true;
apm_->ApplyConfig(config);
// Testing gain modes
for (auto mode :
{AudioProcessing::Config::GainController1::kAdaptiveDigital,
AudioProcessing::Config::GainController1::kFixedDigital,
AudioProcessing::Config::GainController1::kAdaptiveAnalog}) {
config.gain_controller1.mode = mode;
apm_->ApplyConfig(config);
apm_->set_stream_analog_level(100);
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat));
}
// Testing target levels
for (int target_level_dbfs : {0, 15, 31}) {
config.gain_controller1.target_level_dbfs = target_level_dbfs;
apm_->ApplyConfig(config);
apm_->set_stream_analog_level(100);
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat));
}
// Testing compression gains
for (int compression_gain_db : {0, 10, 90}) {
config.gain_controller1.compression_gain_db = compression_gain_db;
apm_->ApplyConfig(config);
apm_->set_stream_analog_level(100);
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat));
}
// Testing limiter off/on
for (bool enable : {false, true}) {
config.gain_controller1.enable_limiter = enable;
apm_->ApplyConfig(config);
apm_->set_stream_analog_level(100);
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat));
}
// Testing level limits
std::array<int, 4> kMinLevels = {0, 0, 255, 65000};
std::array<int, 4> kMaxLevels = {255, 1024, 65535, 65535};
for (size_t i = 0; i < kMinLevels.size(); ++i) {
int min_level = kMinLevels[i];
int max_level = kMaxLevels[i];
config.gain_controller1.analog_level_minimum = min_level;
config.gain_controller1.analog_level_maximum = max_level;
apm_->ApplyConfig(config);
apm_->set_stream_analog_level((min_level + max_level) / 2);
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat));
}
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST_F(ApmTest, GainControlDiesOnTooLowTargetLevelDbfs) {
auto config = apm_->GetConfig();
config.gain_controller1.target_level_dbfs = -1;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
TEST_F(ApmTest, GainControlDiesOnTooHighTargetLevelDbfs) {
auto config = apm_->GetConfig();
config.gain_controller1.target_level_dbfs = 32;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
TEST_F(ApmTest, GainControlDiesOnTooLowCompressionGainDb) {
auto config = apm_->GetConfig();
config.gain_controller1.compression_gain_db = -1;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
TEST_F(ApmTest, GainControlDiesOnTooHighCompressionGainDb) {
auto config = apm_->GetConfig();
config.gain_controller1.compression_gain_db = 91;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
TEST_F(ApmTest, GainControlDiesOnTooLowAnalogLevelLowerLimit) {
auto config = apm_->GetConfig();
config.gain_controller1.analog_level_minimum = -1;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
TEST_F(ApmTest, GainControlDiesOnTooHighAnalogLevelUpperLimit) {
auto config = apm_->GetConfig();
config.gain_controller1.analog_level_maximum = 65536;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
TEST_F(ApmTest, GainControlDiesOnInvertedAnalogLevelLimits) {
auto config = apm_->GetConfig();
config.gain_controller1.analog_level_minimum = 512;
config.gain_controller1.analog_level_maximum = 255;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
TEST_F(ApmTest, ApmDiesOnTooLowAnalogLevel) {
auto config = apm_->GetConfig();
config.gain_controller1.analog_level_minimum = 255;
config.gain_controller1.analog_level_maximum = 512;
apm_->ApplyConfig(config);
EXPECT_DEATH(apm_->set_stream_analog_level(254), "");
}
TEST_F(ApmTest, ApmDiesOnTooHighAnalogLevel) {
auto config = apm_->GetConfig();
config.gain_controller1.analog_level_minimum = 255;
config.gain_controller1.analog_level_maximum = 512;
apm_->ApplyConfig(config);
EXPECT_DEATH(apm_->set_stream_analog_level(513), "");
}
#endif
void ApmTest::RunQuantizedVolumeDoesNotGetStuckTest(int sample_rate) {
Init(sample_rate, sample_rate, sample_rate, 2, 2, 2, false);
auto config = apm_->GetConfig();
config.gain_controller1.enabled = true;
config.gain_controller1.mode =
AudioProcessing::Config::GainController1::kAdaptiveAnalog;
apm_->ApplyConfig(config);
int out_analog_level = 0;
for (int i = 0; i < 2000; ++i) {
ReadFrameWithRewind(near_file_, &frame_);
// Ensure the audio is at a low level, so the AGC will try to increase it.
ScaleFrame(&frame_, 0.25);
// Always pass in the same volume.
apm_->set_stream_analog_level(100);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
out_analog_level = apm_->recommended_stream_analog_level();
}
// Ensure the AGC is still able to reach the maximum.
EXPECT_EQ(255, out_analog_level);
}
// Verifies that despite volume slider quantization, the AGC can continue to
// increase its volume.
TEST_F(ApmTest, QuantizedVolumeDoesNotGetStuck) {
for (size_t i = 0; i < arraysize(kSampleRates); ++i) {
RunQuantizedVolumeDoesNotGetStuckTest(kSampleRates[i]);
}
}
void ApmTest::RunManualVolumeChangeIsPossibleTest(int sample_rate) {
Init(sample_rate, sample_rate, sample_rate, 2, 2, 2, false);
auto config = apm_->GetConfig();
config.gain_controller1.enabled = true;
config.gain_controller1.mode =
AudioProcessing::Config::GainController1::kAdaptiveAnalog;
apm_->ApplyConfig(config);
int out_analog_level = 100;
for (int i = 0; i < 1000; ++i) {
ReadFrameWithRewind(near_file_, &frame_);
// Ensure the audio is at a low level, so the AGC will try to increase it.
ScaleFrame(&frame_, 0.25);
apm_->set_stream_analog_level(out_analog_level);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
out_analog_level = apm_->recommended_stream_analog_level();
}
// Ensure the volume was raised.
EXPECT_GT(out_analog_level, 100);
int highest_level_reached = out_analog_level;
// Simulate a user manual volume change.
out_analog_level = 100;
for (int i = 0; i < 300; ++i) {
ReadFrameWithRewind(near_file_, &frame_);
ScaleFrame(&frame_, 0.25);
apm_->set_stream_analog_level(out_analog_level);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
out_analog_level = apm_->recommended_stream_analog_level();
// Check that AGC respected the manually adjusted volume.
EXPECT_LT(out_analog_level, highest_level_reached);
}
// Check that the volume was still raised.
EXPECT_GT(out_analog_level, 100);
}
TEST_F(ApmTest, ManualVolumeChangeIsPossible) {
for (size_t i = 0; i < arraysize(kSampleRates); ++i) {
RunManualVolumeChangeIsPossibleTest(kSampleRates[i]);
}
}
TEST_F(ApmTest, HighPassFilter) {
// Turn HP filter on/off
AudioProcessing::Config apm_config;
apm_config.high_pass_filter.enabled = true;
apm_->ApplyConfig(apm_config);
apm_config.high_pass_filter.enabled = false;
apm_->ApplyConfig(apm_config);
}
TEST_F(ApmTest, AllProcessingDisabledByDefault) {
AudioProcessing::Config config = apm_->GetConfig();
EXPECT_FALSE(config.echo_canceller.enabled);
EXPECT_FALSE(config.high_pass_filter.enabled);
EXPECT_FALSE(config.gain_controller1.enabled);
EXPECT_FALSE(config.level_estimation.enabled);
EXPECT_FALSE(config.noise_suppression.enabled);
EXPECT_FALSE(config.voice_detection.enabled);
}
TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabled) {
for (size_t i = 0; i < arraysize(kSampleRates); i++) {
Init(kSampleRates[i], kSampleRates[i], kSampleRates[i], 2, 2, 2, false);
SetFrameTo(&frame_, 1000, 2000);
AudioFrame frame_copy;
frame_copy.CopyFrom(frame_);
for (int j = 0; j < 1000; j++) {
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy));
EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(&frame_));
EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy));
}
}
}
TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabledFloat) {
// Test that ProcessStream copies input to output even with no processing.
const size_t kSamples = 160;
const int sample_rate = 16000;
const float src[kSamples] = {-1.0f, 0.0f, 1.0f};
float dest[kSamples] = {};
auto src_channels = &src[0];
auto dest_channels = &dest[0];
apm_.reset(AudioProcessingBuilder().Create());
EXPECT_NOERR(apm_->ProcessStream(&src_channels, StreamConfig(sample_rate, 1),
StreamConfig(sample_rate, 1),
&dest_channels));
for (size_t i = 0; i < kSamples; ++i) {
EXPECT_EQ(src[i], dest[i]);
}
// Same for ProcessReverseStream.
float rev_dest[kSamples] = {};
auto rev_dest_channels = &rev_dest[0];
StreamConfig input_stream = {sample_rate, 1};
StreamConfig output_stream = {sample_rate, 1};
EXPECT_NOERR(apm_->ProcessReverseStream(&src_channels, input_stream,
output_stream, &rev_dest_channels));
for (size_t i = 0; i < kSamples; ++i) {
EXPECT_EQ(src[i], rev_dest[i]);
}
}
TEST_F(ApmTest, IdenticalInputChannelsResultInIdenticalOutputChannels) {
EnableAllComponents();
for (size_t i = 0; i < arraysize(kProcessSampleRates); i++) {
Init(kProcessSampleRates[i], kProcessSampleRates[i], kProcessSampleRates[i],
2, 2, 2, false);
int analog_level = 127;
ASSERT_EQ(0, feof(far_file_));
ASSERT_EQ(0, feof(near_file_));
while (ReadFrame(far_file_, &revframe_) && ReadFrame(near_file_, &frame_)) {
CopyLeftToRightChannel(revframe_.mutable_data(),
revframe_.samples_per_channel_);
ASSERT_EQ(kNoErr, apm_->ProcessReverseStream(&revframe_));
CopyLeftToRightChannel(frame_.mutable_data(),
frame_.samples_per_channel_);
frame_.vad_activity_ = AudioFrame::kVadUnknown;
ASSERT_EQ(kNoErr, apm_->set_stream_delay_ms(0));
apm_->set_stream_analog_level(analog_level);
ASSERT_EQ(kNoErr, apm_->ProcessStream(&frame_));
analog_level = apm_->recommended_stream_analog_level();
VerifyChannelsAreEqual(frame_.data(), frame_.samples_per_channel_);
}
rewind(far_file_);
rewind(near_file_);
}
}
TEST_F(ApmTest, SplittingFilter) {
// Verify the filter is not active through undistorted audio when:
// 1. No components are enabled...
SetFrameTo(&frame_, 1000);
AudioFrame frame_copy;
frame_copy.CopyFrom(frame_);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy));
// 2. Only the level estimator is enabled...
auto apm_config = apm_->GetConfig();
SetFrameTo(&frame_, 1000);
frame_copy.CopyFrom(frame_);
apm_config.level_estimation.enabled = true;
apm_->ApplyConfig(apm_config);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy));
apm_config.level_estimation.enabled = false;
apm_->ApplyConfig(apm_config);
// 3. Only GetStatistics-reporting VAD is enabled...
SetFrameTo(&frame_, 1000);
frame_copy.CopyFrom(frame_);
apm_config.voice_detection.enabled = true;
apm_->ApplyConfig(apm_config);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy));
apm_config.voice_detection.enabled = false;
apm_->ApplyConfig(apm_config);
// 4. Both the VAD and the level estimator are enabled...
SetFrameTo(&frame_, 1000);
frame_copy.CopyFrom(frame_);
apm_config.voice_detection.enabled = true;
apm_config.level_estimation.enabled = true;
apm_->ApplyConfig(apm_config);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy));
apm_config.voice_detection.enabled = false;
apm_config.level_estimation.enabled = false;
apm_->ApplyConfig(apm_config);
// Check the test is valid. We should have distortion from the filter
// when AEC is enabled (which won't affect the audio).
apm_config.echo_canceller.enabled = true;
apm_config.echo_canceller.mobile_mode = false;
apm_->ApplyConfig(apm_config);
frame_.samples_per_channel_ = 320;
frame_.num_channels_ = 2;
frame_.sample_rate_hz_ = 32000;
SetFrameTo(&frame_, 1000);
frame_copy.CopyFrom(frame_);
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
EXPECT_FALSE(FrameDataAreEqual(frame_, frame_copy));
}
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
void ApmTest::ProcessDebugDump(const std::string& in_filename,
const std::string& out_filename,
Format format,
int max_size_bytes) {
TaskQueueForTest worker_queue("ApmTest_worker_queue");
FILE* in_file = fopen(in_filename.c_str(), "rb");
ASSERT_TRUE(in_file != NULL);
audioproc::Event event_msg;
bool first_init = true;
while (ReadMessageFromFile(in_file, &event_msg)) {
if (event_msg.type() == audioproc::Event::INIT) {
const audioproc::Init msg = event_msg.init();
int reverse_sample_rate = msg.sample_rate();
if (msg.has_reverse_sample_rate()) {
reverse_sample_rate = msg.reverse_sample_rate();
}
int output_sample_rate = msg.sample_rate();
if (msg.has_output_sample_rate()) {
output_sample_rate = msg.output_sample_rate();
}
Init(msg.sample_rate(), output_sample_rate, reverse_sample_rate,
msg.num_input_channels(), msg.num_output_channels(),
msg.num_reverse_channels(), false);
if (first_init) {
// AttachAecDump() writes an additional init message. Don't start
// recording until after the first init to avoid the extra message.
auto aec_dump =
AecDumpFactory::Create(out_filename, max_size_bytes, &worker_queue);
EXPECT_TRUE(aec_dump);
apm_->AttachAecDump(std::move(aec_dump));
first_init = false;
}
} else if (event_msg.type() == audioproc::Event::REVERSE_STREAM) {
const audioproc::ReverseStream msg = event_msg.reverse_stream();
if (msg.channel_size() > 0) {
ASSERT_EQ(revframe_.num_channels_,
static_cast<size_t>(msg.channel_size()));
for (int i = 0; i < msg.channel_size(); ++i) {
memcpy(revfloat_cb_->channels()[i], msg.channel(i).data(),
msg.channel(i).size());
}
} else {
memcpy(revframe_.mutable_data(), msg.data().data(), msg.data().size());
if (format == kFloatFormat) {
// We're using an int16 input file; convert to float.
ConvertToFloat(revframe_, revfloat_cb_.get());
}
}
AnalyzeReverseStreamChooser(format);
} else if (event_msg.type() == audioproc::Event::STREAM) {
const audioproc::Stream msg = event_msg.stream();
// ProcessStream could have changed this for the output frame.
frame_.num_channels_ = apm_->num_input_channels();
apm_->set_stream_analog_level(msg.level());
EXPECT_NOERR(apm_->set_stream_delay_ms(msg.delay()));
if (msg.has_keypress()) {
apm_->set_stream_key_pressed(msg.keypress());
} else {
apm_->set_stream_key_pressed(true);
}
if (msg.input_channel_size() > 0) {
ASSERT_EQ(frame_.num_channels_,
static_cast<size_t>(msg.input_channel_size()));
for (int i = 0; i < msg.input_channel_size(); ++i) {
memcpy(float_cb_->channels()[i], msg.input_channel(i).data(),
msg.input_channel(i).size());
}
} else {
memcpy(frame_.mutable_data(), msg.input_data().data(),
msg.input_data().size());
if (format == kFloatFormat) {
// We're using an int16 input file; convert to float.
ConvertToFloat(frame_, float_cb_.get());
}
}
ProcessStreamChooser(format);
}
}
apm_->DetachAecDump();
fclose(in_file);
}
void ApmTest::VerifyDebugDumpTest(Format format) {
rtc::ScopedFakeClock fake_clock;
const std::string in_filename = test::ResourcePath("ref03", "aecdump");
std::string format_string;
switch (format) {
case kIntFormat:
format_string = "_int";
break;
case kFloatFormat:
format_string = "_float";
break;
}
const std::string ref_filename = test::TempFilename(
test::OutputPath(), std::string("ref") + format_string + "_aecdump");
const std::string out_filename = test::TempFilename(
test::OutputPath(), std::string("out") + format_string + "_aecdump");
const std::string limited_filename = test::TempFilename(
test::OutputPath(), std::string("limited") + format_string + "_aecdump");
const size_t logging_limit_bytes = 100000;
// We expect at least this many bytes in the created logfile.
const size_t logging_expected_bytes = 95000;
EnableAllComponents();
ProcessDebugDump(in_filename, ref_filename, format, -1);
ProcessDebugDump(ref_filename, out_filename, format, -1);
ProcessDebugDump(ref_filename, limited_filename, format, logging_limit_bytes);
FILE* ref_file = fopen(ref_filename.c_str(), "rb");
FILE* out_file = fopen(out_filename.c_str(), "rb");
FILE* limited_file = fopen(limited_filename.c_str(), "rb");
ASSERT_TRUE(ref_file != NULL);
ASSERT_TRUE(out_file != NULL);
ASSERT_TRUE(limited_file != NULL);
std::unique_ptr<uint8_t[]> ref_bytes;
std::unique_ptr<uint8_t[]> out_bytes;
std::unique_ptr<uint8_t[]> limited_bytes;
size_t ref_size = ReadMessageBytesFromFile(ref_file, &ref_bytes);
size_t out_size = ReadMessageBytesFromFile(out_file, &out_bytes);
size_t limited_size = ReadMessageBytesFromFile(limited_file, &limited_bytes);
size_t bytes_read = 0;
size_t bytes_read_limited = 0;
while (ref_size > 0 && out_size > 0) {
bytes_read += ref_size;
bytes_read_limited += limited_size;
EXPECT_EQ(ref_size, out_size);
EXPECT_GE(ref_size, limited_size);
EXPECT_EQ(0, memcmp(ref_bytes.get(), out_bytes.get(), ref_size));
EXPECT_EQ(0, memcmp(ref_bytes.get(), limited_bytes.get(), limited_size));
ref_size = ReadMessageBytesFromFile(ref_file, &ref_bytes);
out_size = ReadMessageBytesFromFile(out_file, &out_bytes);
limited_size = ReadMessageBytesFromFile(limited_file, &limited_bytes);
}
EXPECT_GT(bytes_read, 0u);
EXPECT_GT(bytes_read_limited, logging_expected_bytes);
EXPECT_LE(bytes_read_limited, logging_limit_bytes);
EXPECT_NE(0, feof(ref_file));
EXPECT_NE(0, feof(out_file));
EXPECT_NE(0, feof(limited_file));
ASSERT_EQ(0, fclose(ref_file));
ASSERT_EQ(0, fclose(out_file));
ASSERT_EQ(0, fclose(limited_file));
remove(ref_filename.c_str());
remove(out_filename.c_str());
remove(limited_filename.c_str());
}
TEST_F(ApmTest, VerifyDebugDumpInt) {
VerifyDebugDumpTest(kIntFormat);
}
TEST_F(ApmTest, VerifyDebugDumpFloat) {
VerifyDebugDumpTest(kFloatFormat);
}
#endif
// TODO(andrew): expand test to verify output.
TEST_F(ApmTest, DebugDump) {
TaskQueueForTest worker_queue("ApmTest_worker_queue");
const std::string filename =
test::TempFilename(test::OutputPath(), "debug_aec");
{
auto aec_dump = AecDumpFactory::Create("", -1, &worker_queue);
EXPECT_FALSE(aec_dump);
}
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// Stopping without having started should be OK.
apm_->DetachAecDump();
auto aec_dump = AecDumpFactory::Create(filename, -1, &worker_queue);
EXPECT_TRUE(aec_dump);
apm_->AttachAecDump(std::move(aec_dump));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(&revframe_));
apm_->DetachAecDump();
// Verify the file has been written.
FILE* fid = fopen(filename.c_str(), "r");
ASSERT_TRUE(fid != NULL);
// Clean it up.
ASSERT_EQ(0, fclose(fid));
ASSERT_EQ(0, remove(filename.c_str()));
#else
// Verify the file has NOT been written.
ASSERT_TRUE(fopen(filename.c_str(), "r") == NULL);
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
// TODO(andrew): expand test to verify output.
TEST_F(ApmTest, DebugDumpFromFileHandle) {
TaskQueueForTest worker_queue("ApmTest_worker_queue");
const std::string filename =
test::TempFilename(test::OutputPath(), "debug_aec");
FileWrapper f = FileWrapper::OpenWriteOnly(filename.c_str());
ASSERT_TRUE(f.is_open());
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// Stopping without having started should be OK.
apm_->DetachAecDump();
auto aec_dump = AecDumpFactory::Create(std::move(f), -1, &worker_queue);
EXPECT_TRUE(aec_dump);
apm_->AttachAecDump(std::move(aec_dump));
EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(&revframe_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
apm_->DetachAecDump();
// Verify the file has been written.
FILE* fid = fopen(filename.c_str(), "r");
ASSERT_TRUE(fid != NULL);
// Clean it up.
ASSERT_EQ(0, fclose(fid));
ASSERT_EQ(0, remove(filename.c_str()));
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
// TODO(andrew): Add a test to process a few frames with different combinations
// of enabled components.
TEST_F(ApmTest, Process) {
GOOGLE_PROTOBUF_VERIFY_VERSION;
audioproc::OutputData ref_data;
if (!absl::GetFlag(FLAGS_write_apm_ref_data)) {
OpenFileAndReadMessage(ref_filename_, &ref_data);
} else {
// Write the desired tests to the protobuf reference file.
for (size_t i = 0; i < arraysize(kChannels); i++) {
for (size_t j = 0; j < arraysize(kChannels); j++) {
for (size_t l = 0; l < arraysize(kProcessSampleRates); l++) {
audioproc::Test* test = ref_data.add_test();
test->set_num_reverse_channels(kChannels[i]);
test->set_num_input_channels(kChannels[j]);
test->set_num_output_channels(kChannels[j]);
test->set_sample_rate(kProcessSampleRates[l]);
test->set_use_aec_extended_filter(false);
}
}
}
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
// To test the extended filter mode.
audioproc::Test* test = ref_data.add_test();
test->set_num_reverse_channels(2);
test->set_num_input_channels(2);
test->set_num_output_channels(2);
test->set_sample_rate(AudioProcessing::kSampleRate32kHz);
test->set_use_aec_extended_filter(true);
#endif
}
for (int i = 0; i < ref_data.test_size(); i++) {
printf("Running test %d of %d...\n", i + 1, ref_data.test_size());
audioproc::Test* test = ref_data.mutable_test(i);
// TODO(ajm): We no longer allow different input and output channels. Skip
// these tests for now, but they should be removed from the set.
if (test->num_input_channels() != test->num_output_channels())
continue;
Config config;
config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
config.Set<ExtendedFilter>(
new ExtendedFilter(test->use_aec_extended_filter()));
apm_.reset(AudioProcessingBuilder().Create(config));
EnableAllComponents();
Init(test->sample_rate(), test->sample_rate(), test->sample_rate(),
static_cast<size_t>(test->num_input_channels()),
static_cast<size_t>(test->num_output_channels()),
static_cast<size_t>(test->num_reverse_channels()), true);
int frame_count = 0;
int has_voice_count = 0;
int analog_level = 127;
int analog_level_average = 0;
int max_output_average = 0;
float rms_dbfs_average = 0.0f;
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
int stats_index = 0;
#endif
while (ReadFrame(far_file_, &revframe_) && ReadFrame(near_file_, &frame_)) {
EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(&revframe_));
frame_.vad_activity_ = AudioFrame::kVadUnknown;
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
apm_->set_stream_analog_level(analog_level);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&frame_));
// Ensure the frame was downmixed properly.
EXPECT_EQ(static_cast<size_t>(test->num_output_channels()),
frame_.num_channels_);
max_output_average += MaxAudioFrame(frame_);
analog_level = apm_->recommended_stream_analog_level();
analog_level_average += analog_level;
AudioProcessingStats stats =
apm_->GetStatistics(/*has_remote_tracks=*/false);
EXPECT_TRUE(stats.voice_detected);
EXPECT_TRUE(stats.output_rms_dbfs);
has_voice_count += *stats.voice_detected ? 1 : 0;
rms_dbfs_average += *stats.output_rms_dbfs;
size_t frame_size = frame_.samples_per_channel_ * frame_.num_channels_;
size_t write_count =
fwrite(frame_.data(), sizeof(int16_t), frame_size, out_file_);
ASSERT_EQ(frame_size, write_count);
// Reset in case of downmixing.
frame_.num_channels_ = static_cast<size_t>(test->num_input_channels());
frame_count++;
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
const int kStatsAggregationFrameNum = 100; // 1 second.
if (frame_count % kStatsAggregationFrameNum == 0) {
// Get echo and delay metrics.
AudioProcessingStats stats =
apm_->GetStatistics(true /* has_remote_tracks */);
// Echo metrics.
const float echo_return_loss = stats.echo_return_loss.value_or(-1.0f);
const float echo_return_loss_enhancement =
stats.echo_return_loss_enhancement.value_or(-1.0f);
const float residual_echo_likelihood =
stats.residual_echo_likelihood.value_or(-1.0f);
const float residual_echo_likelihood_recent_max =
stats.residual_echo_likelihood_recent_max.value_or(-1.0f);
if (!absl::GetFlag(FLAGS_write_apm_ref_data)) {
const audioproc::Test::EchoMetrics& reference =
test->echo_metrics(stats_index);
constexpr float kEpsilon = 0.01;
EXPECT_NEAR(echo_return_loss, reference.echo_return_loss(), kEpsilon);
EXPECT_NEAR(echo_return_loss_enhancement,
reference.echo_return_loss_enhancement(), kEpsilon);
EXPECT_NEAR(residual_echo_likelihood,
reference.residual_echo_likelihood(), kEpsilon);
EXPECT_NEAR(residual_echo_likelihood_recent_max,
reference.residual_echo_likelihood_recent_max(),
kEpsilon);
++stats_index;
} else {
audioproc::Test::EchoMetrics* message_echo = test->add_echo_metrics();
message_echo->set_echo_return_loss(echo_return_loss);
message_echo->set_echo_return_loss_enhancement(
echo_return_loss_enhancement);
message_echo->set_residual_echo_likelihood(residual_echo_likelihood);
message_echo->set_residual_echo_likelihood_recent_max(
residual_echo_likelihood_recent_max);
}
}
#endif // defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE).
}
max_output_average /= frame_count;
analog_level_average /= frame_count;
rms_dbfs_average /= frame_count;
if (!absl::GetFlag(FLAGS_write_apm_ref_data)) {
const int kIntNear = 1;
// When running the test on a N7 we get a {2, 6} difference of
// |has_voice_count| and |max_output_average| is up to 18 higher.
// All numbers being consistently higher on N7 compare to ref_data.
// TODO(bjornv): If we start getting more of these offsets on Android we
// should consider a different approach. Either using one slack for all,
// or generate a separate android reference.
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
const int kHasVoiceCountOffset = 3;
const int kHasVoiceCountNear = 8;
const int kMaxOutputAverageOffset = 9;
const int kMaxOutputAverageNear = 26;
#else
const int kHasVoiceCountOffset = 0;
const int kHasVoiceCountNear = kIntNear;
const int kMaxOutputAverageOffset = 0;
const int kMaxOutputAverageNear = kIntNear;
#endif
EXPECT_NEAR(test->has_voice_count(),
has_voice_count - kHasVoiceCountOffset, kHasVoiceCountNear);
EXPECT_NEAR(test->analog_level_average(), analog_level_average, kIntNear);
EXPECT_NEAR(test->max_output_average(),
max_output_average - kMaxOutputAverageOffset,
kMaxOutputAverageNear);
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
const double kFloatNear = 0.0005;
EXPECT_NEAR(test->rms_dbfs_average(), rms_dbfs_average, kFloatNear);
#endif
} else {
test->set_has_voice_count(has_voice_count);
test->set_analog_level_average(analog_level_average);
test->set_max_output_average(max_output_average);
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
test->set_rms_dbfs_average(rms_dbfs_average);
#endif
}
rewind(far_file_);
rewind(near_file_);
}
if (absl::GetFlag(FLAGS_write_apm_ref_data)) {
OpenFileAndWriteMessage(ref_filename_, ref_data);
}
}
TEST_F(ApmTest, NoErrorsWithKeyboardChannel) {
struct ChannelFormat {
AudioProcessing::ChannelLayout in_layout;
AudioProcessing::ChannelLayout out_layout;
};
ChannelFormat cf[] = {
{AudioProcessing::kMonoAndKeyboard, AudioProcessing::kMono},
{AudioProcessing::kStereoAndKeyboard, AudioProcessing::kMono},
{AudioProcessing::kStereoAndKeyboard, AudioProcessing::kStereo},
};
std::unique_ptr<AudioProcessing> ap(AudioProcessingBuilder().Create());
// Enable one component just to ensure some processing takes place.
AudioProcessing::Config config;
config.noise_suppression.enabled = true;
ap->ApplyConfig(config);
for (size_t i = 0; i < arraysize(cf); ++i) {
const int in_rate = 44100;
const int out_rate = 48000;
ChannelBuffer<float> in_cb(SamplesFromRate(in_rate),
TotalChannelsFromLayout(cf[i].in_layout));
ChannelBuffer<float> out_cb(SamplesFromRate(out_rate),
ChannelsFromLayout(cf[i].out_layout));
bool has_keyboard = cf[i].in_layout == AudioProcessing::kMonoAndKeyboard ||
cf[i].in_layout == AudioProcessing::kStereoAndKeyboard;
StreamConfig in_sc(in_rate, ChannelsFromLayout(cf[i].in_layout),
has_keyboard);
StreamConfig out_sc(out_rate, ChannelsFromLayout(cf[i].out_layout));
// Run over a few chunks.
for (int j = 0; j < 10; ++j) {
EXPECT_NOERR(ap->ProcessStream(in_cb.channels(), in_sc, out_sc,
out_cb.channels()));
}
}
}
// Compares the reference and test arrays over a region around the expected
// delay. Finds the highest SNR in that region and adds the variance and squared
// error results to the supplied accumulators.
void UpdateBestSNR(const float* ref,
const float* test,
size_t length,
int expected_delay,
double* variance_acc,
double* sq_error_acc) {
double best_snr = std::numeric_limits<double>::min();
double best_variance = 0;
double best_sq_error = 0;
// Search over a region of eight samples around the expected delay.
for (int delay = std::max(expected_delay - 4, 0); delay <= expected_delay + 4;
++delay) {
double sq_error = 0;
double variance = 0;
for (size_t i = 0; i < length - delay; ++i) {
double error = test[i + delay] - ref[i];
sq_error += error * error;
variance += ref[i] * ref[i];
}
if (sq_error == 0) {
*variance_acc += variance;
return;
}
double snr = variance / sq_error;
if (snr > best_snr) {
best_snr = snr;
best_variance = variance;
best_sq_error = sq_error;
}
}
*variance_acc += best_variance;
*sq_error_acc += best_sq_error;
}
// Used to test a multitude of sample rate and channel combinations. It works
// by first producing a set of reference files (in SetUpTestCase) that are
// assumed to be correct, as the used parameters are verified by other tests
// in this collection. Primarily the reference files are all produced at
// "native" rates which do not involve any resampling.
// Each test pass produces an output file with a particular format. The output
// is matched against the reference file closest to its internal processing
// format. If necessary the output is resampled back to its process format.
// Due to the resampling distortion, we don't expect identical results, but
// enforce SNR thresholds which vary depending on the format. 0 is a special
// case SNR which corresponds to inf, or zero error.
typedef std::tuple<int, int, int, int, double, double> AudioProcessingTestData;
class AudioProcessingTest
: public ::testing::TestWithParam<AudioProcessingTestData> {
public:
AudioProcessingTest()
: input_rate_(std::get<0>(GetParam())),
output_rate_(std::get<1>(GetParam())),
reverse_input_rate_(std::get<2>(GetParam())),
reverse_output_rate_(std::get<3>(GetParam())),
expected_snr_(std::get<4>(GetParam())),
expected_reverse_snr_(std::get<5>(GetParam())) {}
virtual ~AudioProcessingTest() {}
static void SetUpTestSuite() {
// Create all needed output reference files.
const int kNativeRates[] = {8000, 16000, 32000, 48000};
const size_t kNumChannels[] = {1, 2};
for (size_t i = 0; i < arraysize(kNativeRates); ++i) {
for (size_t j = 0; j < arraysize(kNumChannels); ++j) {
for (size_t k = 0; k < arraysize(kNumChannels); ++k) {
// The reference files always have matching input and output channels.
ProcessFormat(kNativeRates[i], kNativeRates[i], kNativeRates[i],
kNativeRates[i], kNumChannels[j], kNumChannels[j],
kNumChannels[k], kNumChannels[k], "ref");
}
}
}
}
void TearDown() {
// Remove "out" files after each test.
ClearTempOutFiles();
}
static void TearDownTestSuite() { ClearTempFiles(); }
// Runs a process pass on files with the given parameters and dumps the output
// to a file specified with |output_file_prefix|. Both forward and reverse
// output streams are dumped.
static void ProcessFormat(int input_rate,
int output_rate,
int reverse_input_rate,
int reverse_output_rate,
size_t num_input_channels,
size_t num_output_channels,
size_t num_reverse_input_channels,
size_t num_reverse_output_channels,
const std::string& output_file_prefix) {
Config config;
config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
std::unique_ptr<AudioProcessing> ap(
AudioProcessingBuilder().Create(config));
EnableAllAPComponents(ap.get());
ProcessingConfig processing_config = {
{{input_rate, num_input_channels},
{output_rate, num_output_channels},
{reverse_input_rate, num_reverse_input_channels},
{reverse_output_rate, num_reverse_output_channels}}};
ap->Initialize(processing_config);
FILE* far_file =
fopen(ResourceFilePath("far", reverse_input_rate).c_str(), "rb");
FILE* near_file = fopen(ResourceFilePath("near", input_rate).c_str(), "rb");
FILE* out_file = fopen(
OutputFilePath(
output_file_prefix, input_rate, output_rate, reverse_input_rate,
reverse_output_rate, num_input_channels, num_output_channels,
num_reverse_input_channels, num_reverse_output_channels, kForward)
.c_str(),
"wb");
FILE* rev_out_file = fopen(
OutputFilePath(
output_file_prefix, input_rate, output_rate, reverse_input_rate,
reverse_output_rate, num_input_channels, num_output_channels,
num_reverse_input_channels, num_reverse_output_channels, kReverse)
.c_str(),
"wb");
ASSERT_TRUE(far_file != NULL);
ASSERT_TRUE(near_file != NULL);
ASSERT_TRUE(out_file != NULL);
ASSERT_TRUE(rev_out_file != NULL);
ChannelBuffer<float> fwd_cb(SamplesFromRate(input_rate),
num_input_channels);
ChannelBuffer<float> rev_cb(SamplesFromRate(reverse_input_rate),
num_reverse_input_channels);
ChannelBuffer<float> out_cb(SamplesFromRate(output_rate),
num_output_channels);
ChannelBuffer<float> rev_out_cb(SamplesFromRate(reverse_output_rate),
num_reverse_output_channels);
// Temporary buffers.
const int max_length =
2 * std::max(std::max(out_cb.num_frames(), rev_out_cb.num_frames()),
std::max(fwd_cb.num_frames(), rev_cb.num_frames()));
std::unique_ptr<float[]> float_data(new float[max_length]);
std::unique_ptr<int16_t[]> int_data(new int16_t[max_length]);
int analog_level = 127;
while (ReadChunk(far_file, int_data.get(), float_data.get(), &rev_cb) &&
ReadChunk(near_file, int_data.get(), float_data.get(), &fwd_cb)) {
EXPECT_NOERR(ap->ProcessReverseStream(
rev_cb.channels(), processing_config.reverse_input_stream(),
processing_config.reverse_output_stream(), rev_out_cb.channels()));
EXPECT_NOERR(ap->set_stream_delay_ms(0));
ap->set_stream_analog_level(analog_level);
EXPECT_NOERR(ap->ProcessStream(
fwd_cb.channels(), StreamConfig(input_rate, num_input_channels),
StreamConfig(output_rate, num_output_channels), out_cb.channels()));
// Dump forward output to file.
Interleave(out_cb.channels(), out_cb.num_frames(), out_cb.num_channels(),
float_data.get());
size_t out_length = out_cb.num_channels() * out_cb.num_frames();
ASSERT_EQ(out_length, fwrite(float_data.get(), sizeof(float_data[0]),
out_length, out_file));
// Dump reverse output to file.
Interleave(rev_out_cb.channels(), rev_out_cb.num_frames(),
rev_out_cb.num_channels(), float_data.get());
size_t rev_out_length =
rev_out_cb.num_channels() * rev_out_cb.num_frames();
ASSERT_EQ(rev_out_length, fwrite(float_data.get(), sizeof(float_data[0]),
rev_out_length, rev_out_file));
analog_level = ap->recommended_stream_analog_level();
}
fclose(far_file);
fclose(near_file);
fclose(out_file);
fclose(rev_out_file);
}
protected:
int input_rate_;
int output_rate_;
int reverse_input_rate_;
int reverse_output_rate_;
double expected_snr_;
double expected_reverse_snr_;
};
TEST_P(AudioProcessingTest, Formats) {
struct ChannelFormat {
int num_input;
int num_output;
int num_reverse_input;
int num_reverse_output;
};
ChannelFormat cf[] = {
{1, 1, 1, 1}, {1, 1, 2, 1}, {2, 1, 1, 1},
{2, 1, 2, 1}, {2, 2, 1, 1}, {2, 2, 2, 2},
};
for (size_t i = 0; i < arraysize(cf); ++i) {
ProcessFormat(input_rate_, output_rate_, reverse_input_rate_,
reverse_output_rate_, cf[i].num_input, cf[i].num_output,
cf[i].num_reverse_input, cf[i].num_reverse_output, "out");
// Verify output for both directions.
std::vector<StreamDirection> stream_directions;
stream_directions.push_back(kForward);
stream_directions.push_back(kReverse);
for (StreamDirection file_direction : stream_directions) {
const int in_rate = file_direction ? reverse_input_rate_ : input_rate_;
const int out_rate = file_direction ? reverse_output_rate_ : output_rate_;
const int out_num =
file_direction ? cf[i].num_reverse_output : cf[i].num_output;
const double expected_snr =
file_direction ? expected_reverse_snr_ : expected_snr_;
const int min_ref_rate = std::min(in_rate, out_rate);
int ref_rate;
if (min_ref_rate > 32000) {
ref_rate = 48000;
} else if (min_ref_rate > 16000) {
ref_rate = 32000;
} else if (min_ref_rate > 8000) {
ref_rate = 16000;
} else {
ref_rate = 8000;
}
#ifdef WEBRTC_ARCH_ARM_FAMILY
if (file_direction == kForward) {
ref_rate = std::min(ref_rate, 32000);
}
#endif
FILE* out_file = fopen(
OutputFilePath("out", input_rate_, output_rate_, reverse_input_rate_,
reverse_output_rate_, cf[i].num_input,
cf[i].num_output, cf[i].num_reverse_input,
cf[i].num_reverse_output, file_direction)
.c_str(),
"rb");
// The reference files always have matching input and output channels.
FILE* ref_file =
fopen(OutputFilePath("ref", ref_rate, ref_rate, ref_rate, ref_rate,
cf[i].num_output, cf[i].num_output,
cf[i].num_reverse_output,
cf[i].num_reverse_output, file_direction)
.c_str(),
"rb");
ASSERT_TRUE(out_file != NULL);
ASSERT_TRUE(ref_file != NULL);
const size_t ref_length = SamplesFromRate(ref_rate) * out_num;
const size_t out_length = SamplesFromRate(out_rate) * out_num;
// Data from the reference file.
std::unique_ptr<float[]> ref_data(new float[ref_length]);
// Data from the output file.
std::unique_ptr<float[]> out_data(new float[out_length]);
// Data from the resampled output, in case the reference and output rates
// don't match.
std::unique_ptr<float[]> cmp_data(new float[ref_length]);
PushResampler<float> resampler;
resampler.InitializeIfNeeded(out_rate, ref_rate, out_num);
// Compute the resampling delay of the output relative to the reference,
// to find the region over which we should search for the best SNR.
float expected_delay_sec = 0;
if (in_rate != ref_rate) {
// Input resampling delay.
expected_delay_sec +=
PushSincResampler::AlgorithmicDelaySeconds(in_rate);
}
if (out_rate != ref_rate) {
// Output resampling delay.
expected_delay_sec +=
PushSincResampler::AlgorithmicDelaySeconds(ref_rate);
// Delay of converting the output back to its processing rate for
// testing.
expected_delay_sec +=
PushSincResampler::AlgorithmicDelaySeconds(out_rate);
}
int expected_delay =
std::floor(expected_delay_sec * ref_rate + 0.5f) * out_num;
double variance = 0;
double sq_error = 0;
while (fread(out_data.get(), sizeof(out_data[0]), out_length, out_file) &&
fread(ref_data.get(), sizeof(ref_data[0]), ref_length, ref_file)) {
float* out_ptr = out_data.get();
if (out_rate != ref_rate) {
// Resample the output back to its internal processing rate if
// necssary.
ASSERT_EQ(ref_length,
static_cast<size_t>(resampler.Resample(
out_ptr, out_length, cmp_data.get(), ref_length)));
out_ptr = cmp_data.get();
}
// Update the |sq_error| and |variance| accumulators with the highest
// SNR of reference vs output.
UpdateBestSNR(ref_data.get(), out_ptr, ref_length, expected_delay,
&variance, &sq_error);
}
std::cout << "(" << input_rate_ << ", " << output_rate_ << ", "
<< reverse_input_rate_ << ", " << reverse_output_rate_ << ", "
<< cf[i].num_input << ", " << cf[i].num_output << ", "
<< cf[i].num_reverse_input << ", " << cf[i].num_reverse_output
<< ", " << file_direction << "): ";
if (sq_error > 0) {
double snr = 10 * log10(variance / sq_error);
EXPECT_GE(snr, expected_snr);
EXPECT_NE(0, expected_snr);
std::cout << "SNR=" << snr << " dB" << std::endl;
} else {
std::cout << "SNR=inf dB" << std::endl;
}
fclose(out_file);
fclose(ref_file);
}
}
}
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
INSTANTIATE_TEST_SUITE_P(
CommonFormats,
AudioProcessingTest,
::testing::Values(std::make_tuple(48000, 48000, 48000, 48000, 0, 0),
std::make_tuple(48000, 48000, 32000, 48000, 40, 30),
std::make_tuple(48000, 48000, 16000, 48000, 40, 20),
std::make_tuple(48000, 44100, 48000, 44100, 20, 20),
std::make_tuple(48000, 44100, 32000, 44100, 20, 15),
std::make_tuple(48000, 44100, 16000, 44100, 20, 15),
std::make_tuple(48000, 32000, 48000, 32000, 30, 35),
std::make_tuple(48000, 32000, 32000, 32000, 30, 0),
std::make_tuple(48000, 32000, 16000, 32000, 30, 20),
std::make_tuple(48000, 16000, 48000, 16000, 25, 20),
std::make_tuple(48000, 16000, 32000, 16000, 25, 20),
std::make_tuple(48000, 16000, 16000, 16000, 25, 0),
std::make_tuple(44100, 48000, 48000, 48000, 30, 0),
std::make_tuple(44100, 48000, 32000, 48000, 30, 30),
std::make_tuple(44100, 48000, 16000, 48000, 30, 20),
std::make_tuple(44100, 44100, 48000, 44100, 20, 20),
std::make_tuple(44100, 44100, 32000, 44100, 20, 15),
std::make_tuple(44100, 44100, 16000, 44100, 20, 15),
std::make_tuple(44100, 32000, 48000, 32000, 30, 35),
std::make_tuple(44100, 32000, 32000, 32000, 30, 0),
std::make_tuple(44100, 32000, 16000, 32000, 30, 20),
std::make_tuple(44100, 16000, 48000, 16000, 25, 20),
std::make_tuple(44100, 16000, 32000, 16000, 25, 20),
std::make_tuple(44100, 16000, 16000, 16000, 25, 0),
std::make_tuple(32000, 48000, 48000, 48000, 30, 0),
std::make_tuple(32000, 48000, 32000, 48000, 32, 30),
std::make_tuple(32000, 48000, 16000, 48000, 30, 20),
std::make_tuple(32000, 44100, 48000, 44100, 19, 20),
std::make_tuple(32000, 44100, 32000, 44100, 19, 15),
std::make_tuple(32000, 44100, 16000, 44100, 19, 15),
std::make_tuple(32000, 32000, 48000, 32000, 40, 35),
std::make_tuple(32000, 32000, 32000, 32000, 0, 0),
std::make_tuple(32000, 32000, 16000, 32000, 40, 20),
std::make_tuple(32000, 16000, 48000, 16000, 25, 20),
std::make_tuple(32000, 16000, 32000, 16000, 25, 20),
std::make_tuple(32000, 16000, 16000, 16000, 25, 0),
std::make_tuple(16000, 48000, 48000, 48000, 23, 0),
std::make_tuple(16000, 48000, 32000, 48000, 24, 30),
std::make_tuple(16000, 48000, 16000, 48000, 24, 20),
std::make_tuple(16000, 44100, 48000, 44100, 15, 20),
std::make_tuple(16000, 44100, 32000, 44100, 15, 15),
std::make_tuple(16000, 44100, 16000, 44100, 15, 15),
std::make_tuple(16000, 32000, 48000, 32000, 25, 35),
std::make_tuple(16000, 32000, 32000, 32000, 25, 0),
std::make_tuple(16000, 32000, 16000, 32000, 25, 20),
std::make_tuple(16000, 16000, 48000, 16000, 39, 20),
std::make_tuple(16000, 16000, 32000, 16000, 40, 20),
std::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
#elif defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
INSTANTIATE_TEST_SUITE_P(
CommonFormats,
AudioProcessingTest,
::testing::Values(std::make_tuple(48000, 48000, 48000, 48000, 19, 0),
std::make_tuple(48000, 48000, 32000, 48000, 19, 30),
std::make_tuple(48000, 48000, 16000, 48000, 19, 20),
std::make_tuple(48000, 44100, 48000, 44100, 15, 20),
std::make_tuple(48000, 44100, 32000, 44100, 15, 15),
std::make_tuple(48000, 44100, 16000, 44100, 15, 15),
std::make_tuple(48000, 32000, 48000, 32000, 19, 35),
std::make_tuple(48000, 32000, 32000, 32000, 19, 0),
std::make_tuple(48000, 32000, 16000, 32000, 19, 20),
std::make_tuple(48000, 16000, 48000, 16000, 20, 20),
std::make_tuple(48000, 16000, 32000, 16000, 20, 20),
std::make_tuple(48000, 16000, 16000, 16000, 20, 0),
std::make_tuple(44100, 48000, 48000, 48000, 15, 0),
std::make_tuple(44100, 48000, 32000, 48000, 15, 30),
std::make_tuple(44100, 48000, 16000, 48000, 15, 20),
std::make_tuple(44100, 44100, 48000, 44100, 15, 20),
std::make_tuple(44100, 44100, 32000, 44100, 15, 15),
std::make_tuple(44100, 44100, 16000, 44100, 15, 15),
std::make_tuple(44100, 32000, 48000, 32000, 18, 35),
std::make_tuple(44100, 32000, 32000, 32000, 18, 0),
std::make_tuple(44100, 32000, 16000, 32000, 18, 20),
std::make_tuple(44100, 16000, 48000, 16000, 19, 20),
std::make_tuple(44100, 16000, 32000, 16000, 19, 20),
std::make_tuple(44100, 16000, 16000, 16000, 19, 0),
std::make_tuple(32000, 48000, 48000, 48000, 27, 0),
std::make_tuple(32000, 48000, 32000, 48000, 65, 30),
std::make_tuple(32000, 48000, 16000, 48000, 30, 20),
std::make_tuple(32000, 44100, 48000, 44100, 20, 20),
std::make_tuple(32000, 44100, 32000, 44100, 20, 15),
std::make_tuple(32000, 44100, 16000, 44100, 20, 15),
std::make_tuple(32000, 32000, 48000, 32000, 27, 35),
std::make_tuple(32000, 32000, 32000, 32000, 0, 0),
std::make_tuple(32000, 32000, 16000, 32000, 30, 20),
std::make_tuple(32000, 16000, 48000, 16000, 20, 20),
std::make_tuple(32000, 16000, 32000, 16000, 20, 20),
std::make_tuple(32000, 16000, 16000, 16000, 20, 0),
std::make_tuple(16000, 48000, 48000, 48000, 23, 0),
std::make_tuple(16000, 48000, 32000, 48000, 24, 30),
std::make_tuple(16000, 48000, 16000, 48000, 25, 20),
std::make_tuple(16000, 44100, 48000, 44100, 15, 20),
std::make_tuple(16000, 44100, 32000, 44100, 15, 15),
std::make_tuple(16000, 44100, 16000, 44100, 15, 15),
std::make_tuple(16000, 32000, 48000, 32000, 24, 35),
std::make_tuple(16000, 32000, 32000, 32000, 24, 0),
std::make_tuple(16000, 32000, 16000, 32000, 25, 20),
std::make_tuple(16000, 16000, 48000, 16000, 28, 20),
std::make_tuple(16000, 16000, 32000, 16000, 28, 20),
std::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
#endif
// Produces a scoped trace debug output.
std::string ProduceDebugText(int render_input_sample_rate_hz,
int render_output_sample_rate_hz,
int capture_input_sample_rate_hz,
int capture_output_sample_rate_hz,
size_t render_input_num_channels,
size_t render_output_num_channels,
size_t capture_input_num_channels,
size_t capture_output_num_channels) {
rtc::StringBuilder ss;
ss << "Sample rates:"
<< "\n"
<< " Render input: " << render_input_sample_rate_hz << " Hz"
<< "\n"
<< " Render output: " << render_output_sample_rate_hz << " Hz"
<< "\n"
<< " Capture input: " << capture_input_sample_rate_hz << " Hz"
<< "\n"
<< " Capture output: " << capture_output_sample_rate_hz << " Hz"
<< "\n"
<< "Number of channels:"
<< "\n"
<< " Render input: " << render_input_num_channels << "\n"
<< " Render output: " << render_output_num_channels << "\n"
<< " Capture input: " << capture_input_num_channels << "\n"
<< " Capture output: " << capture_output_num_channels;
return ss.Release();
}
// Validates that running the audio processing module using various combinations
// of sample rates and number of channels works as intended.
void RunApmRateAndChannelTest(
rtc::ArrayView<const int> sample_rates_hz,
rtc::ArrayView<const int> render_channel_counts,
rtc::ArrayView<const int> capture_channel_counts) {
std::unique_ptr<AudioProcessing> apm(AudioProcessingBuilder().Create());
webrtc::AudioProcessing::Config apm_config;
apm_config.echo_canceller.enabled = true;
apm->ApplyConfig(apm_config);
StreamConfig render_input_stream_config;
StreamConfig render_output_stream_config;
StreamConfig capture_input_stream_config;
StreamConfig capture_output_stream_config;
std::vector<float> render_input_frame_channels;
std::vector<float*> render_input_frame;
std::vector<float> render_output_frame_channels;
std::vector<float*> render_output_frame;
std::vector<float> capture_input_frame_channels;
std::vector<float*> capture_input_frame;
std::vector<float> capture_output_frame_channels;
std::vector<float*> capture_output_frame;
for (auto render_input_sample_rate_hz : sample_rates_hz) {
for (auto render_output_sample_rate_hz : sample_rates_hz) {
for (auto capture_input_sample_rate_hz : sample_rates_hz) {
for (auto capture_output_sample_rate_hz : sample_rates_hz) {
for (size_t render_input_num_channels : render_channel_counts) {
for (size_t capture_input_num_channels : capture_channel_counts) {
size_t render_output_num_channels = render_input_num_channels;
size_t capture_output_num_channels = capture_input_num_channels;
auto populate_audio_frame = [](int sample_rate_hz,
size_t num_channels,
StreamConfig* cfg,
std::vector<float>* channels_data,
std::vector<float*>* frame_data) {
cfg->set_sample_rate_hz(sample_rate_hz);
cfg->set_num_channels(num_channels);
cfg->set_has_keyboard(false);
size_t max_frame_size = ceil(sample_rate_hz / 100.f);
channels_data->resize(num_channels * max_frame_size);
std::fill(channels_data->begin(), channels_data->end(), 0.5f);
frame_data->resize(num_channels);
for (size_t channel = 0; channel < num_channels; ++channel) {
(*frame_data)[channel] =
&(*channels_data)[channel * max_frame_size];
}
};
populate_audio_frame(
render_input_sample_rate_hz, render_input_num_channels,
&render_input_stream_config, &render_input_frame_channels,
&render_input_frame);
populate_audio_frame(
render_output_sample_rate_hz, render_output_num_channels,
&render_output_stream_config, &render_output_frame_channels,
&render_output_frame);
populate_audio_frame(
capture_input_sample_rate_hz, capture_input_num_channels,
&capture_input_stream_config, &capture_input_frame_channels,
&capture_input_frame);
populate_audio_frame(
capture_output_sample_rate_hz, capture_output_num_channels,
&capture_output_stream_config, &capture_output_frame_channels,
&capture_output_frame);
for (size_t frame = 0; frame < 2; ++frame) {
SCOPED_TRACE(ProduceDebugText(
render_input_sample_rate_hz, render_output_sample_rate_hz,
capture_input_sample_rate_hz, capture_output_sample_rate_hz,
render_input_num_channels, render_output_num_channels,
render_input_num_channels, capture_output_num_channels));
int result = apm->ProcessReverseStream(
&render_input_frame[0], render_input_stream_config,
render_output_stream_config, &render_output_frame[0]);
EXPECT_EQ(result, AudioProcessing::kNoError);
result = apm->ProcessStream(
&capture_input_frame[0], capture_input_stream_config,
capture_output_stream_config, &capture_output_frame[0]);
EXPECT_EQ(result, AudioProcessing::kNoError);
}
}
}
}
}
}
}
}
} // namespace
TEST(RuntimeSettingTest, TestDefaultCtor) {
auto s = AudioProcessing::RuntimeSetting();
EXPECT_EQ(AudioProcessing::RuntimeSetting::Type::kNotSpecified, s.type());
}
TEST(RuntimeSettingTest, TestCapturePreGain) {
using Type = AudioProcessing::RuntimeSetting::Type;
{
auto s = AudioProcessing::RuntimeSetting::CreateCapturePreGain(1.25f);
EXPECT_EQ(Type::kCapturePreGain, s.type());
float v;
s.GetFloat(&v);
EXPECT_EQ(1.25f, v);
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
EXPECT_DEATH(AudioProcessing::RuntimeSetting::CreateCapturePreGain(0.1f), "");
#endif
}
TEST(RuntimeSettingTest, TestCaptureFixedPostGain) {
using Type = AudioProcessing::RuntimeSetting::Type;
{
auto s = AudioProcessing::RuntimeSetting::CreateCaptureFixedPostGain(1.25f);
EXPECT_EQ(Type::kCaptureFixedPostGain, s.type());
float v;
s.GetFloat(&v);
EXPECT_EQ(1.25f, v);
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
EXPECT_DEATH(AudioProcessing::RuntimeSetting::CreateCapturePreGain(0.1f), "");
#endif
}
TEST(RuntimeSettingTest, TestUsageWithSwapQueue) {
SwapQueue<AudioProcessing::RuntimeSetting> q(1);
auto s = AudioProcessing::RuntimeSetting();
ASSERT_TRUE(q.Insert(&s));
ASSERT_TRUE(q.Remove(&s));
EXPECT_EQ(AudioProcessing::RuntimeSetting::Type::kNotSpecified, s.type());
}
TEST(ApmConfiguration, EnablePostProcessing) {
// Verify that apm uses a capture post processing module if one is provided.
auto mock_post_processor_ptr =
new ::testing::NiceMock<test::MockCustomProcessing>();
auto mock_post_processor =
std::unique_ptr<CustomProcessing>(mock_post_processor_ptr);
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilder()
.SetCapturePostProcessing(std::move(mock_post_processor))
.Create();
AudioFrame audio;
audio.num_channels_ = 1;
SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz);
EXPECT_CALL(*mock_post_processor_ptr, Process(::testing::_)).Times(1);
apm->ProcessStream(&audio);
}
TEST(ApmConfiguration, EnablePreProcessing) {
// Verify that apm uses a capture post processing module if one is provided.
auto mock_pre_processor_ptr =
new ::testing::NiceMock<test::MockCustomProcessing>();
auto mock_pre_processor =
std::unique_ptr<CustomProcessing>(mock_pre_processor_ptr);
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilder()
.SetRenderPreProcessing(std::move(mock_pre_processor))
.Create();
AudioFrame audio;
audio.num_channels_ = 1;
SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz);
EXPECT_CALL(*mock_pre_processor_ptr, Process(::testing::_)).Times(1);
apm->ProcessReverseStream(&audio);
}
TEST(ApmConfiguration, EnableCaptureAnalyzer) {
// Verify that apm uses a capture analyzer if one is provided.
auto mock_capture_analyzer_ptr =
new ::testing::NiceMock<test::MockCustomAudioAnalyzer>();
auto mock_capture_analyzer =
std::unique_ptr<CustomAudioAnalyzer>(mock_capture_analyzer_ptr);
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilder()
.SetCaptureAnalyzer(std::move(mock_capture_analyzer))
.Create();
AudioFrame audio;
audio.num_channels_ = 1;
SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz);
EXPECT_CALL(*mock_capture_analyzer_ptr, Analyze(::testing::_)).Times(1);
apm->ProcessStream(&audio);
}
TEST(ApmConfiguration, PreProcessingReceivesRuntimeSettings) {
auto mock_pre_processor_ptr =
new ::testing::NiceMock<test::MockCustomProcessing>();
auto mock_pre_processor =
std::unique_ptr<CustomProcessing>(mock_pre_processor_ptr);
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilder()
.SetRenderPreProcessing(std::move(mock_pre_processor))
.Create();
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCustomRenderSetting(0));
// RuntimeSettings forwarded during 'Process*Stream' calls.
// Therefore we have to make one such call.
AudioFrame audio;
audio.num_channels_ = 1;
SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz);
EXPECT_CALL(*mock_pre_processor_ptr, SetRuntimeSetting(::testing::_))
.Times(1);
apm->ProcessReverseStream(&audio);
}
class MyEchoControlFactory : public EchoControlFactory {
public:
std::unique_ptr<EchoControl> Create(int sample_rate_hz) {
auto ec = new test::MockEchoControl();
EXPECT_CALL(*ec, AnalyzeRender(::testing::_)).Times(1);
EXPECT_CALL(*ec, AnalyzeCapture(::testing::_)).Times(2);
EXPECT_CALL(*ec, ProcessCapture(::testing::_, ::testing::_, ::testing::_))
.Times(2);
return std::unique_ptr<EchoControl>(ec);
}
std::unique_ptr<EchoControl> Create(int sample_rate_hz,
int num_render_channels,
int num_capture_channels) {
return Create(sample_rate_hz);
}
};
TEST(ApmConfiguration, EchoControlInjection) {
// Verify that apm uses an injected echo controller if one is provided.
webrtc::Config webrtc_config;
std::unique_ptr<EchoControlFactory> echo_control_factory(
new MyEchoControlFactory());
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilder()
.SetEchoControlFactory(std::move(echo_control_factory))
.Create(webrtc_config);
AudioFrame audio;
audio.num_channels_ = 1;
SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz);
apm->ProcessStream(&audio);
apm->ProcessReverseStream(&audio);
apm->ProcessStream(&audio);
}
std::unique_ptr<AudioProcessing> CreateApm(bool mobile_aec) {
Config old_config;
std::unique_ptr<AudioProcessing> apm(
AudioProcessingBuilder().Create(old_config));
if (!apm) {
return apm;
}
ProcessingConfig processing_config = {
{{32000, 1}, {32000, 1}, {32000, 1}, {32000, 1}}};
if (apm->Initialize(processing_config) != 0) {
return nullptr;
}
// Disable all components except for an AEC and the residual echo detector.
AudioProcessing::Config apm_config;
apm_config.residual_echo_detector.enabled = true;
apm_config.high_pass_filter.enabled = false;
apm_config.gain_controller1.enabled = false;
apm_config.gain_controller2.enabled = false;
apm_config.echo_canceller.enabled = true;
apm_config.echo_canceller.mobile_mode = mobile_aec;
apm_config.noise_suppression.enabled = false;
apm_config.level_estimation.enabled = false;
apm_config.voice_detection.enabled = false;
apm->ApplyConfig(apm_config);
return apm;
}
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_MAC)
#define MAYBE_ApmStatistics DISABLED_ApmStatistics
#else
#define MAYBE_ApmStatistics ApmStatistics
#endif
TEST(MAYBE_ApmStatistics, AECEnabledTest) {
// Set up APM with AEC3 and process some audio.
std::unique_ptr<AudioProcessing> apm = CreateApm(false);
ASSERT_TRUE(apm);
AudioProcessing::Config apm_config;
apm_config.echo_canceller.enabled = true;
apm->ApplyConfig(apm_config);
// Set up an audioframe.
AudioFrame frame;
frame.num_channels_ = 1;
SetFrameSampleRate(&frame, AudioProcessing::NativeRate::kSampleRate32kHz);
// Fill the audio frame with a sawtooth pattern.
int16_t* ptr = frame.mutable_data();
for (size_t i = 0; i < frame.kMaxDataSizeSamples; i++) {
ptr[i] = 10000 * ((i % 3) - 1);
}
// Do some processing.
for (int i = 0; i < 200; i++) {
EXPECT_EQ(apm->ProcessReverseStream(&frame), 0);
EXPECT_EQ(apm->set_stream_delay_ms(0), 0);
EXPECT_EQ(apm->ProcessStream(&frame), 0);
}
// Test statistics interface.
AudioProcessingStats stats = apm->GetStatistics(true);
// We expect all statistics to be set and have a sensible value.
ASSERT_TRUE(stats.residual_echo_likelihood);
EXPECT_GE(*stats.residual_echo_likelihood, 0.0);
EXPECT_LE(*stats.residual_echo_likelihood, 1.0);
ASSERT_TRUE(stats.residual_echo_likelihood_recent_max);
EXPECT_GE(*stats.residual_echo_likelihood_recent_max, 0.0);
EXPECT_LE(*stats.residual_echo_likelihood_recent_max, 1.0);
ASSERT_TRUE(stats.echo_return_loss);
EXPECT_NE(*stats.echo_return_loss, -100.0);
ASSERT_TRUE(stats.echo_return_loss_enhancement);
EXPECT_NE(*stats.echo_return_loss_enhancement, -100.0);
// If there are no receive streams, we expect the stats not to be set. The
// 'false' argument signals to APM that no receive streams are currently
// active. In that situation the statistics would get stuck at their last
// calculated value (AEC and echo detection need at least one stream in each
// direction), so to avoid that, they should not be set by APM.
stats = apm->GetStatistics(false);
EXPECT_FALSE(stats.residual_echo_likelihood);
EXPECT_FALSE(stats.residual_echo_likelihood_recent_max);
EXPECT_FALSE(stats.echo_return_loss);
EXPECT_FALSE(stats.echo_return_loss_enhancement);
}
TEST(MAYBE_ApmStatistics, AECMEnabledTest) {
// Set up APM with AECM and process some audio.
std::unique_ptr<AudioProcessing> apm = CreateApm(true);
ASSERT_TRUE(apm);
// Set up an audioframe.
AudioFrame frame;
frame.num_channels_ = 1;
SetFrameSampleRate(&frame, AudioProcessing::NativeRate::kSampleRate32kHz);
// Fill the audio frame with a sawtooth pattern.
int16_t* ptr = frame.mutable_data();
for (size_t i = 0; i < frame.kMaxDataSizeSamples; i++) {
ptr[i] = 10000 * ((i % 3) - 1);
}
// Do some processing.
for (int i = 0; i < 200; i++) {
EXPECT_EQ(apm->ProcessReverseStream(&frame), 0);
EXPECT_EQ(apm->set_stream_delay_ms(0), 0);
EXPECT_EQ(apm->ProcessStream(&frame), 0);
}
// Test statistics interface.
AudioProcessingStats stats = apm->GetStatistics(true);
// We expect only the residual echo detector statistics to be set and have a
// sensible value.
EXPECT_TRUE(stats.residual_echo_likelihood);
if (stats.residual_echo_likelihood) {
EXPECT_GE(*stats.residual_echo_likelihood, 0.0);
EXPECT_LE(*stats.residual_echo_likelihood, 1.0);
}
EXPECT_TRUE(stats.residual_echo_likelihood_recent_max);
if (stats.residual_echo_likelihood_recent_max) {
EXPECT_GE(*stats.residual_echo_likelihood_recent_max, 0.0);
EXPECT_LE(*stats.residual_echo_likelihood_recent_max, 1.0);
}
EXPECT_FALSE(stats.echo_return_loss);
EXPECT_FALSE(stats.echo_return_loss_enhancement);
// If there are no receive streams, we expect the stats not to be set.
stats = apm->GetStatistics(false);
EXPECT_FALSE(stats.residual_echo_likelihood);
EXPECT_FALSE(stats.residual_echo_likelihood_recent_max);
EXPECT_FALSE(stats.echo_return_loss);
EXPECT_FALSE(stats.echo_return_loss_enhancement);
}
TEST(ApmStatistics, ReportOutputRmsDbfs) {
ProcessingConfig processing_config = {
{{32000, 1}, {32000, 1}, {32000, 1}, {32000, 1}}};
AudioProcessing::Config config;
// Set up an audioframe.
AudioFrame frame;
frame.num_channels_ = 1;
SetFrameSampleRate(&frame, AudioProcessing::NativeRate::kSampleRate32kHz);
// Fill the audio frame with a sawtooth pattern.
int16_t* ptr = frame.mutable_data();
for (size_t i = 0; i < frame.kMaxDataSizeSamples; i++) {
ptr[i] = 10000 * ((i % 3) - 1);
}
std::unique_ptr<AudioProcessing> apm(AudioProcessingBuilder().Create());
apm->Initialize(processing_config);
// If not enabled, no metric should be reported.
EXPECT_EQ(apm->ProcessStream(&frame), 0);
EXPECT_FALSE(apm->GetStatistics(false).output_rms_dbfs);
// If enabled, metrics should be reported.
config.level_estimation.enabled = true;
apm->ApplyConfig(config);
EXPECT_EQ(apm->ProcessStream(&frame), 0);
auto stats = apm->GetStatistics(false);
EXPECT_TRUE(stats.output_rms_dbfs);
EXPECT_GE(*stats.output_rms_dbfs, 0);
// If re-disabled, the value is again not reported.
config.level_estimation.enabled = false;
apm->ApplyConfig(config);
EXPECT_EQ(apm->ProcessStream(&frame), 0);
EXPECT_FALSE(apm->GetStatistics(false).output_rms_dbfs);
}
TEST(ApmStatistics, ReportHasVoice) {
ProcessingConfig processing_config = {
{{32000, 1}, {32000, 1}, {32000, 1}, {32000, 1}}};
AudioProcessing::Config config;
// Set up an audioframe.
AudioFrame frame;
frame.num_channels_ = 1;
SetFrameSampleRate(&frame, AudioProcessing::NativeRate::kSampleRate32kHz);
// Fill the audio frame with a sawtooth pattern.
int16_t* ptr = frame.mutable_data();
for (size_t i = 0; i < frame.kMaxDataSizeSamples; i++) {
ptr[i] = 10000 * ((i % 3) - 1);
}
std::unique_ptr<AudioProcessing> apm(AudioProcessingBuilder().Create());
apm->Initialize(processing_config);
// If not enabled, no metric should be reported.
EXPECT_EQ(apm->ProcessStream(&frame), 0);
EXPECT_FALSE(apm->GetStatistics(false).voice_detected);
// If enabled, metrics should be reported.
config.voice_detection.enabled = true;
apm->ApplyConfig(config);
EXPECT_EQ(apm->ProcessStream(&frame), 0);
auto stats = apm->GetStatistics(false);
EXPECT_TRUE(stats.voice_detected);
// If re-disabled, the value is again not reported.
config.voice_detection.enabled = false;
apm->ApplyConfig(config);
EXPECT_EQ(apm->ProcessStream(&frame), 0);
EXPECT_FALSE(apm->GetStatistics(false).voice_detected);
}
TEST(ApmConfiguration, HandlingOfRateAndChannelCombinations) {
std::array<int, 3> sample_rates_hz = {16000, 32000, 48000};
std::array<int, 2> render_channel_counts = {1, 7};
std::array<int, 2> capture_channel_counts = {1, 7};
RunApmRateAndChannelTest(sample_rates_hz, render_channel_counts,
capture_channel_counts);
}
TEST(ApmConfiguration, HandlingOfChannelCombinations) {
std::array<int, 1> sample_rates_hz = {48000};
std::array<int, 8> render_channel_counts = {1, 2, 3, 4, 5, 6, 7, 8};
std::array<int, 8> capture_channel_counts = {1, 2, 3, 4, 5, 6, 7, 8};
RunApmRateAndChannelTest(sample_rates_hz, render_channel_counts,
capture_channel_counts);
}
TEST(ApmConfiguration, HandlingOfRateCombinations) {
std::array<int, 9> sample_rates_hz = {8000, 11025, 16000, 22050, 32000,
48000, 96000, 192000, 384000};
std::array<int, 1> render_channel_counts = {2};
std::array<int, 1> capture_channel_counts = {2};
RunApmRateAndChannelTest(sample_rates_hz, render_channel_counts,
capture_channel_counts);
}
} // namespace webrtc