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webrtc/modules/audio_processing/audio_processing_unittest.cc
Per Åhgren 3e8bf282c4 Increase the maximum supported sample rate to 384000 Hz and add tests 
This CL increases the maximum supported sample rate so that all rates
up to 384000 Hz are handled.

The CL also adds tests that verifies that APM works as intended for
different combinations of number of channels and sample rates.

Bug: webrtc:10882
Change-Id: I98738e33ac21413ae00fec10bb43b8796ae2078c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150532
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29014}
2019-08-29 22:14:25 +00:00

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/*
* 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 <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;
}
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.high_pass_filter.enabled = true;
apm_config.level_estimation.enabled = true;
ap->ApplyConfig(apm_config);
EXPECT_NOERR(ap->level_estimator()->Enable(true));
EXPECT_NOERR(ap->noise_suppression()->Enable(true));
EXPECT_NOERR(ap->voice_detection()->Enable(true));
}
// 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
frame_(NULL),
revframe_(NULL),
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);
frame_ = new AudioFrame();
revframe_ = new AudioFrame();
Init(32000, 32000, 32000, 2, 2, 2, false);
}
void ApmTest::TearDown() {
if (frame_) {
delete frame_;
}
frame_ = NULL;
if (revframe_) {
delete revframe_;
}
revframe_ = NULL;
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));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->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(), frame_->samples_per_channel_,
frame_->sample_rate_hz_, LayoutFromChannels(frame_->num_channels_),
output_sample_rate_hz_, LayoutFromChannels(num_output_channels_),
float_cb_->channels());
}
int ApmTest::AnalyzeReverseStreamChooser(Format format) {
if (format == kIntFormat) {
return apm_->ProcessReverseStream(revframe_);
}
return apm_->AnalyzeReverseStream(
revfloat_cb_->channels(), revframe_->samples_per_channel_,
revframe_->sample_rate_hz_, LayoutFromChannels(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 --
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format));
// Resets after successful ProcessStream().
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format));
// Other stream parameters set correctly.
AudioProcessing::Config apm_config = apm_->GetConfig();
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));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
// -- 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.
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
// -- 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));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->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));
}
}
TEST_F(ApmTest, GainControl) {
// Testing gain modes
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_mode(apm_->gain_control()->mode()));
GainControl::Mode mode[] = {GainControl::kAdaptiveAnalog,
GainControl::kAdaptiveDigital,
GainControl::kFixedDigital};
for (size_t i = 0; i < arraysize(mode); i++) {
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_mode(mode[i]));
EXPECT_EQ(mode[i], apm_->gain_control()->mode());
}
// Testing target levels
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_target_level_dbfs(
apm_->gain_control()->target_level_dbfs()));
int level_dbfs[] = {0, 6, 31};
for (size_t i = 0; i < arraysize(level_dbfs); i++) {
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_target_level_dbfs(level_dbfs[i]));
EXPECT_EQ(level_dbfs[i], apm_->gain_control()->target_level_dbfs());
}
// Testing compression gains
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_compression_gain_db(
apm_->gain_control()->compression_gain_db()));
int gain_db[] = {0, 10, 90};
for (size_t i = 0; i < arraysize(gain_db); i++) {
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_compression_gain_db(gain_db[i]));
ProcessStreamChooser(kFloatFormat);
EXPECT_EQ(gain_db[i], apm_->gain_control()->compression_gain_db());
}
// Testing limiter off/on
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->enable_limiter(false));
EXPECT_FALSE(apm_->gain_control()->is_limiter_enabled());
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->enable_limiter(true));
EXPECT_TRUE(apm_->gain_control()->is_limiter_enabled());
// Testing level limits
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_analog_level_limits(
apm_->gain_control()->analog_level_minimum(),
apm_->gain_control()->analog_level_maximum()));
int min_level[] = {0, 255, 1024};
for (size_t i = 0; i < arraysize(min_level); i++) {
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_analog_level_limits(
min_level[i], 1024));
EXPECT_EQ(min_level[i], apm_->gain_control()->analog_level_minimum());
}
int max_level[] = {0, 1024, 65535};
for (size_t i = 0; i < arraysize(min_level); i++) {
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_analog_level_limits(0, max_level[i]));
EXPECT_EQ(max_level[i], apm_->gain_control()->analog_level_maximum());
}
// TODO(ajm): stream_is_saturated() and stream_analog_level()
// Turn AGC off
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
EXPECT_FALSE(apm_->gain_control()->is_enabled());
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST_F(ApmTest, GainControlDiesOnTooLowTargetLevelDbfs) {
EXPECT_DEATH(apm_->gain_control()->set_target_level_dbfs(-1), "");
}
TEST_F(ApmTest, GainControlDiesOnTooHighTargetLevelDbfs) {
EXPECT_DEATH(apm_->gain_control()->set_target_level_dbfs(32), "");
}
TEST_F(ApmTest, GainControlDiesOnTooLowCompressionGainDb) {
EXPECT_DEATH(apm_->gain_control()->set_compression_gain_db(-1), "");
}
TEST_F(ApmTest, GainControlDiesOnTooHighCompressionGainDb) {
EXPECT_DEATH(apm_->gain_control()->set_compression_gain_db(91), "");
}
TEST_F(ApmTest, GainControlDiesOnTooLowAnalogLevelLowerLimit) {
EXPECT_DEATH(apm_->gain_control()->set_analog_level_limits(-1, 512), "");
}
TEST_F(ApmTest, GainControlDiesOnTooHighAnalogLevelUpperLimit) {
EXPECT_DEATH(apm_->gain_control()->set_analog_level_limits(512, 65536), "");
}
TEST_F(ApmTest, GainControlDiesOnInvertedAnalogLevelLimits) {
EXPECT_DEATH(apm_->gain_control()->set_analog_level_limits(512, 255), "");
}
TEST_F(ApmTest, ApmDiesOnTooLowAnalogLevel) {
apm_->gain_control()->set_analog_level_limits(255, 512);
EXPECT_DEATH(apm_->set_stream_analog_level(254), "");
}
TEST_F(ApmTest, ApmDiesOnTooHighAnalogLevel) {
apm_->gain_control()->set_analog_level_limits(255, 512);
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);
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
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.
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(100));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
out_analog_level = apm_->gain_control()->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);
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
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);
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(out_analog_level));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
out_analog_level = apm_->gain_control()->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);
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(out_analog_level));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
out_analog_level = apm_->gain_control()->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, NoiseSuppression) {
// Test valid suppression levels.
NoiseSuppression::Level level[] = {
NoiseSuppression::kLow, NoiseSuppression::kModerate,
NoiseSuppression::kHigh, NoiseSuppression::kVeryHigh};
for (size_t i = 0; i < arraysize(level); i++) {
EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->set_level(level[i]));
EXPECT_EQ(level[i], apm_->noise_suppression()->level());
}
// Turn NS on/off
EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(true));
EXPECT_TRUE(apm_->noise_suppression()->is_enabled());
EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(false));
EXPECT_FALSE(apm_->noise_suppression()->is_enabled());
}
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, LevelEstimator) {
// Turn level estimator on/off
EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
EXPECT_FALSE(apm_->level_estimator()->is_enabled());
EXPECT_EQ(apm_->kNotEnabledError, apm_->level_estimator()->RMS());
EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
EXPECT_TRUE(apm_->level_estimator()->is_enabled());
// Run this test in wideband; in super-wb, the splitting filter distorts the
// audio enough to cause deviation from the expectation for small values.
frame_->samples_per_channel_ = 160;
frame_->num_channels_ = 2;
frame_->sample_rate_hz_ = 16000;
// Min value if no frames have been processed.
EXPECT_EQ(127, apm_->level_estimator()->RMS());
// Min value on zero frames.
SetFrameTo(frame_, 0);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(127, apm_->level_estimator()->RMS());
// Try a few RMS values.
// (These also test that the value resets after retrieving it.)
SetFrameTo(frame_, 32767);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(0, apm_->level_estimator()->RMS());
SetFrameTo(frame_, 30000);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(1, apm_->level_estimator()->RMS());
SetFrameTo(frame_, 10000);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(10, apm_->level_estimator()->RMS());
SetFrameTo(frame_, 10);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(70, apm_->level_estimator()->RMS());
// Verify reset after enable/disable.
SetFrameTo(frame_, 32767);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
SetFrameTo(frame_, 1);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(90, apm_->level_estimator()->RMS());
// Verify reset after initialize.
SetFrameTo(frame_, 32767);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
SetFrameTo(frame_, 1);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(90, apm_->level_estimator()->RMS());
}
TEST_F(ApmTest, VoiceDetection) {
// Test external VAD
EXPECT_EQ(apm_->kNoError,
apm_->voice_detection()->set_stream_has_voice(true));
EXPECT_TRUE(apm_->voice_detection()->stream_has_voice());
EXPECT_EQ(apm_->kNoError,
apm_->voice_detection()->set_stream_has_voice(false));
EXPECT_FALSE(apm_->voice_detection()->stream_has_voice());
// Test valid likelihoods
VoiceDetection::Likelihood likelihood[] = {
VoiceDetection::kVeryLowLikelihood, VoiceDetection::kLowLikelihood,
VoiceDetection::kModerateLikelihood, VoiceDetection::kHighLikelihood};
for (size_t i = 0; i < arraysize(likelihood); i++) {
EXPECT_EQ(apm_->kNoError,
apm_->voice_detection()->set_likelihood(likelihood[i]));
EXPECT_EQ(likelihood[i], apm_->voice_detection()->likelihood());
}
/* TODO(bjornv): Enable once VAD supports other frame lengths than 10 ms
// Test invalid frame sizes
EXPECT_EQ(apm_->kBadParameterError,
apm_->voice_detection()->set_frame_size_ms(12));
// Test valid frame sizes
for (int i = 10; i <= 30; i += 10) {
EXPECT_EQ(apm_->kNoError,
apm_->voice_detection()->set_frame_size_ms(i));
EXPECT_EQ(i, apm_->voice_detection()->frame_size_ms());
}
*/
// Turn VAD on/off
EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
EXPECT_TRUE(apm_->voice_detection()->is_enabled());
EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
EXPECT_FALSE(apm_->voice_detection()->is_enabled());
// Test that AudioFrame activity is maintained when VAD is disabled.
EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
AudioFrame::VADActivity activity[] = {
AudioFrame::kVadActive, AudioFrame::kVadPassive, AudioFrame::kVadUnknown};
for (size_t i = 0; i < arraysize(activity); i++) {
frame_->vad_activity_ = activity[i];
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(activity[i], frame_->vad_activity_);
}
// Test that AudioFrame activity is set when VAD is enabled.
EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
frame_->vad_activity_ = AudioFrame::kVadUnknown;
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_NE(AudioFrame::kVadUnknown, frame_->vad_activity_);
// TODO(bjornv): Add tests for streamed voice; stream_has_voice()
}
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.level_estimation.enabled);
EXPECT_FALSE(config.voice_detection.enabled);
EXPECT_FALSE(apm_->gain_control()->is_enabled());
EXPECT_FALSE(apm_->level_estimator()->is_enabled());
EXPECT_FALSE(apm_->noise_suppression()->is_enabled());
EXPECT_FALSE(apm_->voice_detection()->is_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, kSamples, sample_rate,
LayoutFromChannels(1), sample_rate,
LayoutFromChannels(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));
ASSERT_EQ(kNoErr,
apm_->gain_control()->set_stream_analog_level(analog_level));
ASSERT_EQ(kNoErr, apm_->ProcessStream(frame_));
analog_level = apm_->gain_control()->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...
SetFrameTo(frame_, 1000);
frame_copy.CopyFrom(*frame_);
EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
// 3. Only VAD is enabled...
SetFrameTo(frame_, 1000);
frame_copy.CopyFrom(*frame_);
EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
// 4. Only GetStatistics-reporting VAD is enabled...
SetFrameTo(frame_, 1000);
frame_copy.CopyFrom(*frame_);
auto apm_config = apm_->GetConfig();
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);
// 5. Both VADs and the level estimator are enabled...
SetFrameTo(frame_, 1000);
frame_copy.CopyFrom(*frame_);
EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
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));
EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
apm_config.voice_detection.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();
EXPECT_NOERR(apm_->gain_control()->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 is_saturated_count = 0;
int analog_level = 127;
int analog_level_average = 0;
int max_output_average = 0;
float ns_speech_prob_average = 0.0f;
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));
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->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_->gain_control()->stream_analog_level();
analog_level_average += analog_level;
if (apm_->gain_control()->stream_is_saturated()) {
is_saturated_count++;
}
if (apm_->voice_detection()->stream_has_voice()) {
has_voice_count++;
EXPECT_EQ(AudioFrame::kVadActive, frame_->vad_activity_);
} else {
EXPECT_EQ(AudioFrame::kVadPassive, frame_->vad_activity_);
}
ns_speech_prob_average += apm_->noise_suppression()->speech_probability();
AudioProcessingStats stats =
apm_->GetStatistics(/*has_remote_tracks=*/false);
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;
ns_speech_prob_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->is_saturated_count(), is_saturated_count, kIntNear);
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->ns_speech_probability_average(), ns_speech_prob_average,
kFloatNear);
EXPECT_NEAR(test->rms_dbfs_average(), rms_dbfs_average, kFloatNear);
#endif
} else {
test->set_has_voice_count(has_voice_count);
test->set_is_saturated_count(is_saturated_count);
test->set_analog_level_average(analog_level_average);
test->set_max_output_average(max_output_average);
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
EXPECT_LE(0.0f, ns_speech_prob_average);
EXPECT_GE(1.0f, ns_speech_prob_average);
test->set_ns_speech_probability_average(ns_speech_prob_average);
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.
ap->noise_suppression()->Enable(true);
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));
// Run over a few chunks.
for (int j = 0; j < 10; ++j) {
EXPECT_NOERR(ap->ProcessStream(in_cb.channels(), in_cb.num_frames(),
in_rate, cf[i].in_layout, out_rate,
cf[i].out_layout, 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));
EXPECT_NOERR(ap->gain_control()->set_stream_analog_level(analog_level));
EXPECT_NOERR(ap->ProcessStream(
fwd_cb.channels(), fwd_cb.num_frames(), input_rate,
LayoutFromChannels(num_input_channels), output_rate,
LayoutFromChannels(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->gain_control()->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, 24, 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, 35, 0),
std::make_tuple(32000, 48000, 32000, 48000, 65, 30),
std::make_tuple(32000, 48000, 16000, 48000, 40, 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, 35, 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, 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, 25, 0),
std::make_tuple(16000, 48000, 32000, 48000, 25, 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, 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, 35, 20),
std::make_tuple(16000, 16000, 32000, 16000, 35, 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::_)).Times(2);
return std::unique_ptr<EchoControl>(ec);
}
};
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_controller2.enabled = false;
apm_config.echo_canceller.enabled = true;
apm_config.echo_canceller.mobile_mode = mobile_aec;
apm->ApplyConfig(apm_config);
EXPECT_EQ(apm->gain_control()->Enable(false), 0);
EXPECT_EQ(apm->level_estimator()->Enable(false), 0);
EXPECT_EQ(apm->noise_suppression()->Enable(false), 0);
EXPECT_EQ(apm->voice_detection()->Enable(false), 0);
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
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