blob: 892927f223d634f2fe98a177dbd7156670d7646d [file] [log] [blame]
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <math.h>
#include <stdio.h>
#include <algorithm>
#include <limits>
#include <queue>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h"
#include "webrtc/modules/audio_processing/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "gtest/gtest.h"
#include "external/webrtc/webrtc/modules/audio_processing/test/unittest.pb.h"
#else
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio_processing/unittest.pb.h"
#endif
namespace webrtc {
namespace {
// TODO(bjornv): This is not feasible until the functionality has been
// re-implemented; see comment at the bottom of this file. For now, the user has
// to hard code the |write_ref_data| value.
// When false, this will compare the output data with the results stored to
// file. This is the typical case. When the file should be updated, it can
// be set to true with the command-line switch --write_ref_data.
bool write_ref_data = false;
const int kChannels[] = {1, 2};
const size_t kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
const int kSampleRates[] = {8000, 16000, 32000};
const size_t kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
// AECM doesn't support super-wb.
const int kProcessSampleRates[] = {8000, 16000};
#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
const int kProcessSampleRates[] = {8000, 16000, 32000};
#endif
const size_t kProcessSampleRatesSize = sizeof(kProcessSampleRates) /
sizeof(*kProcessSampleRates);
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 (int 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.
int TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
switch (layout) {
case AudioProcessing::kMono:
return 1;
case AudioProcessing::kMonoAndKeyboard:
case AudioProcessing::kStereo:
return 2;
case AudioProcessing::kStereoAndKeyboard:
return 3;
}
assert(false);
return -1;
}
int TruncateToMultipleOf10(int value) {
return (value / 10) * 10;
}
void MixStereoToMono(const float* stereo, float* mono,
int samples_per_channel) {
for (int 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,
int samples_per_channel) {
for (int i = 0; i < samples_per_channel; ++i)
mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) >> 1;
}
void CopyLeftToRightChannel(int16_t* stereo, int samples_per_channel) {
for (int i = 0; i < samples_per_channel; i++) {
stereo[i * 2 + 1] = stereo[i * 2];
}
}
void VerifyChannelsAreEqual(int16_t* stereo, int samples_per_channel) {
for (int i = 0; i < samples_per_channel; i++) {
EXPECT_EQ(stereo[i * 2 + 1], stereo[i * 2]);
}
}
void SetFrameTo(AudioFrame* frame, int16_t value) {
for (int 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(2, frame->num_channels_);
for (int 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) {
for (int 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) {
#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
EXPECT_NOERR(ap->echo_control_mobile()->Enable(true));
EXPECT_NOERR(ap->gain_control()->set_mode(GainControl::kAdaptiveDigital));
EXPECT_NOERR(ap->gain_control()->Enable(true));
#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
EXPECT_NOERR(ap->echo_cancellation()->enable_drift_compensation(true));
EXPECT_NOERR(ap->echo_cancellation()->enable_metrics(true));
EXPECT_NOERR(ap->echo_cancellation()->enable_delay_logging(true));
EXPECT_NOERR(ap->echo_cancellation()->Enable(true));
EXPECT_NOERR(ap->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
EXPECT_NOERR(ap->gain_control()->set_analog_level_limits(0, 255));
EXPECT_NOERR(ap->gain_control()->Enable(true));
#endif
EXPECT_NOERR(ap->high_pass_filter()->Enable(true));
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 int length = frame.samples_per_channel_ * frame.num_channels_;
int16_t max_data = AbsValue(frame.data_[0]);
for (int i = 1; i < length; i++) {
max_data = std::max(max_data, AbsValue(frame.data_[i]));
}
return max_data;
}
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
void TestStats(const AudioProcessing::Statistic& test,
const audioproc::Test::Statistic& reference) {
EXPECT_EQ(reference.instant(), test.instant);
EXPECT_EQ(reference.average(), test.average);
EXPECT_EQ(reference.maximum(), test.maximum);
EXPECT_EQ(reference.minimum(), test.minimum);
}
void WriteStatsMessage(const AudioProcessing::Statistic& output,
audioproc::Test::Statistic* msg) {
msg->set_instant(output.instant);
msg->set_average(output.average);
msg->set_maximum(output.maximum);
msg->set_minimum(output.minimum);
}
#endif
void OpenFileAndWriteMessage(const std::string filename,
const ::google::protobuf::MessageLite& msg) {
#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
FILE* file = fopen(filename.c_str(), "wb");
ASSERT_TRUE(file != NULL);
int32_t size = msg.ByteSize();
ASSERT_GT(size, 0);
rtc::scoped_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);
#else
std::cout << "Warning: Writing new reference is only allowed on Linux!"
<< std::endl;
#endif
}
std::string ResourceFilePath(std::string name, int sample_rate_hz) {
std::ostringstream 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(std::string name,
int input_rate,
int output_rate,
int reverse_rate,
int num_input_channels,
int num_output_channels,
int num_reverse_channels) {
std::ostringstream ss;
ss << name << "_i" << num_input_channels << "_" << input_rate / 1000
<< "_r" << num_reverse_channels << "_" << reverse_rate / 1000 << "_";
if (num_output_channels == 1) {
ss << "mono";
} else if (num_output_channels == 2) {
ss << "stereo";
} else {
assert(false);
}
ss << output_rate / 1000 << "_pcm";
std::string filename = ss.str();
if (temp_filenames[filename] == "")
temp_filenames[filename] = test::TempFilename(test::OutputPath(), filename);
return temp_filenames[filename];
}
void ClearTempFiles() {
for (auto& kv : temp_filenames)
remove(kv.second.c_str());
}
void OpenFileAndReadMessage(const std::string filename,
::google::protobuf::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.
assert(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 SetUpTestCase() {
Trace::CreateTrace();
}
static void TearDownTestCase() {
Trace::ReturnTrace();
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,
int num_reverse_channels,
int num_input_channels,
int num_output_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 TestChangingChannels(int num_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);
void VerifyDebugDumpTest(Format format);
const std::string output_path_;
const std::string ref_path_;
const std::string ref_filename_;
rtc::scoped_ptr<AudioProcessing> apm_;
AudioFrame* frame_;
AudioFrame* revframe_;
rtc::scoped_ptr<ChannelBuffer<float> > float_cb_;
rtc::scoped_ptr<ChannelBuffer<float> > revfloat_cb_;
int output_sample_rate_hz_;
int num_output_channels_;
FILE* far_file_;
FILE* near_file_;
FILE* out_file_;
};
ApmTest::ApmTest()
: output_path_(test::OutputPath()),
ref_path_(test::ProjectRootPath() + "data/audio_processing/"),
#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
ref_filename_(ref_path_ + "output_data_fixed.pb"),
#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
#if defined(WEBRTC_MAC)
// A different file for Mac is needed because on this platform the AEC
// constant |kFixedDelayMs| value is 20 and not 50 as it is on the rest.
ref_filename_(ref_path_ + "output_data_mac.pb"),
#else
ref_filename_(ref_path_ + "output_data_float.pb"),
#endif
#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(AudioProcessing::Create(config));
}
void ApmTest::SetUp() {
ASSERT_TRUE(apm_.get() != NULL);
frame_ = new AudioFrame();
revframe_ = new AudioFrame();
#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
Init(16000, 16000, 16000, 2, 2, 2, false);
#else
Init(32000, 32000, 32000, 2, 2, 2, false);
#endif
}
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_,
output_sample_rate_hz_,
revframe_->sample_rate_hz_,
LayoutFromChannels(frame_->num_channels_),
LayoutFromChannels(num_output_channels_),
LayoutFromChannels(revframe_->num_channels_)));
}
void ApmTest::Init(int sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
int num_input_channels,
int num_output_channels,
int 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,
num_input_channels,
num_output_channels,
num_reverse_channels);
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->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->data_,
frame->samples_per_channel_);
}
if (cb) {
ConvertToFloat(*frame, cb);
}
return true;
}
bool ApmTest::ReadFrame(FILE* file, AudioFrame* frame) {
return ReadFrame(file, frame, NULL);
}
// If the end of the file has been reached, rewind it and attempt to read the
// frame again.
void ApmTest::ReadFrameWithRewind(FILE* file, AudioFrame* frame,
ChannelBuffer<float>* cb) {
if (!ReadFrame(near_file_, frame_, cb)) {
rewind(near_file_);
ASSERT_TRUE(ReadFrame(near_file_, frame_, cb));
}
}
void ApmTest::ReadFrameWithRewind(FILE* file, AudioFrame* frame) {
ReadFrameWithRewind(file, frame, NULL);
}
void ApmTest::ProcessWithDefaultStreamParameters(AudioFrame* frame) {
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
apm_->echo_cancellation()->set_stream_drift_samples(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_->AnalyzeReverseStream(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_->AnalyzeReverseStream(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) {
int median;
int std;
float poor_fraction;
// Discard the first delay metrics to avoid convergence effects.
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->GetDelayMetrics(&median, &std,
&poor_fraction));
}
}
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 int samples_per_ms = std::min(16, frame_->samples_per_channel_ / 10);
int expected_median = std::min(std::max(delay_ms - system_delay_ms,
delay_min), delay_max);
int expected_median_high = std::min(std::max(
expected_median + 96 / samples_per_ms, delay_min), delay_max);
int expected_median_low = std::min(std::max(
expected_median - 96 / samples_per_ms, delay_min), delay_max);
// Verify delay metrics.
int median;
int std;
float poor_fraction;
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->GetDelayMetrics(&median, &std,
&poor_fraction));
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.
EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_drift_compensation(true));
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kStreamParameterNotSetError,
ProcessStreamChooser(format));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_drift_compensation(false));
// -- Missing delay --
EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
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_->kStreamParameterNotSetError,
ProcessStreamChooser(format));
// Other stream parameters set correctly.
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_drift_compensation(true));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
ProcessStreamChooser(format));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
// -- Missing drift --
EXPECT_EQ(apm_->kStreamParameterNotSetError,
ProcessStreamChooser(format));
// Resets after successful ProcessStream().
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
ProcessStreamChooser(format));
// Other stream parameters set correctly.
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
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_->kStreamParameterNotSetError,
ProcessStreamChooser(format));
// -- No stream parameters --
EXPECT_EQ(apm_->kNoError,
AnalyzeReverseStreamChooser(format));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
ProcessStreamChooser(format));
// -- All there --
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
apm_->echo_cancellation()->set_stream_drift_samples(0);
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::TestChangingChannels(int num_channels,
AudioProcessing::Error expected_return) {
frame_->num_channels_ = num_channels;
EXPECT_EQ(expected_return, apm_->ProcessStream(frame_));
EXPECT_EQ(expected_return, apm_->AnalyzeReverseStream(frame_));
}
TEST_F(ApmTest, Channels) {
// Testing number of invalid channels.
TestChangingChannels(0, apm_->kBadNumberChannelsError);
TestChangingChannels(3, apm_->kBadNumberChannelsError);
// Testing number of valid channels.
for (int i = 1; i < 3; i++) {
TestChangingChannels(i, kNoErr);
EXPECT_EQ(i, apm_->num_input_channels());
// We always force the number of reverse channels used for processing to 1.
EXPECT_EQ(1, apm_->num_reverse_channels());
}
}
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};
for (size_t i = 0; i < sizeof(fs) / sizeof(*fs); i++) {
SetContainerFormat(fs[i], 2, frame_, &float_cb_);
EXPECT_NOERR(ProcessStreamChooser(kIntFormat));
EXPECT_EQ(fs[i], apm_->input_sample_rate_hz());
}
}
TEST_F(ApmTest, EchoCancellation) {
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_drift_compensation(true));
EXPECT_TRUE(apm_->echo_cancellation()->is_drift_compensation_enabled());
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_drift_compensation(false));
EXPECT_FALSE(apm_->echo_cancellation()->is_drift_compensation_enabled());
EchoCancellation::SuppressionLevel level[] = {
EchoCancellation::kLowSuppression,
EchoCancellation::kModerateSuppression,
EchoCancellation::kHighSuppression,
};
for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_suppression_level(level[i]));
EXPECT_EQ(level[i],
apm_->echo_cancellation()->suppression_level());
}
EchoCancellation::Metrics metrics;
EXPECT_EQ(apm_->kNotEnabledError,
apm_->echo_cancellation()->GetMetrics(&metrics));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_metrics(true));
EXPECT_TRUE(apm_->echo_cancellation()->are_metrics_enabled());
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_metrics(false));
EXPECT_FALSE(apm_->echo_cancellation()->are_metrics_enabled());
int median = 0;
int std = 0;
float poor_fraction = 0;
EXPECT_EQ(apm_->kNotEnabledError,
apm_->echo_cancellation()->GetDelayMetrics(&median, &std,
&poor_fraction));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_delay_logging(true));
EXPECT_TRUE(apm_->echo_cancellation()->is_delay_logging_enabled());
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_delay_logging(false));
EXPECT_FALSE(apm_->echo_cancellation()->is_delay_logging_enabled());
EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
EXPECT_TRUE(apm_->echo_cancellation()->is_enabled());
EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(false));
EXPECT_FALSE(apm_->echo_cancellation()->is_enabled());
EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
EXPECT_TRUE(apm_->echo_cancellation()->is_enabled());
EXPECT_TRUE(apm_->echo_cancellation()->aec_core() != NULL);
EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(false));
EXPECT_FALSE(apm_->echo_cancellation()->is_enabled());
EXPECT_FALSE(apm_->echo_cancellation()->aec_core() != NULL);
}
TEST_F(ApmTest, DISABLED_EchoCancellationReportsCorrectDelays) {
// TODO(bjornv): Fix this test to work with DA-AEC.
// Enable AEC only.
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_drift_compensation(false));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_metrics(false));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_delay_logging(true));
EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
Config config;
config.Set<ReportedDelay>(new ReportedDelay(true));
apm_->SetExtraOptions(config);
// Internally in the AEC the amount of lookahead the delay estimation can
// handle is 15 blocks and the maximum delay is set to 60 blocks.
const int kLookaheadBlocks = 15;
const int kMaxDelayBlocks = 60;
// The AEC has a startup time before it actually starts to process. This
// procedure can flush the internal far-end buffer, which of course affects
// the delay estimation. Therefore, we set a system_delay high enough to
// avoid that. The smallest system_delay you can report without flushing the
// buffer is 66 ms in 8 kHz.
//
// It is known that for 16 kHz (and 32 kHz) sampling frequency there is an
// additional stuffing of 8 ms on the fly, but it seems to have no impact on
// delay estimation. This should be noted though. In case of test failure,
// this could be the cause.
const int kSystemDelayMs = 66;
// Test a couple of corner cases and verify that the estimated delay is
// within a valid region (set to +-1.5 blocks). Note that these cases are
// sampling frequency dependent.
for (size_t i = 0; i < kProcessSampleRatesSize; i++) {
Init(kProcessSampleRates[i],
kProcessSampleRates[i],
kProcessSampleRates[i],
2,
2,
2,
false);
// Sampling frequency dependent variables.
const int num_ms_per_block = std::max(4,
640 / frame_->samples_per_channel_);
const int delay_min_ms = -kLookaheadBlocks * num_ms_per_block;
const int delay_max_ms = (kMaxDelayBlocks - 1) * num_ms_per_block;
// 1) Verify correct delay estimate at lookahead boundary.
int delay_ms = TruncateToMultipleOf10(kSystemDelayMs + delay_min_ms);
ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
delay_max_ms);
// 2) A delay less than maximum lookahead should give an delay estimate at
// the boundary (= -kLookaheadBlocks * num_ms_per_block).
delay_ms -= 20;
ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
delay_max_ms);
// 3) Three values around zero delay. Note that we need to compensate for
// the fake system_delay.
delay_ms = TruncateToMultipleOf10(kSystemDelayMs - 10);
ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
delay_max_ms);
delay_ms = TruncateToMultipleOf10(kSystemDelayMs);
ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
delay_max_ms);
delay_ms = TruncateToMultipleOf10(kSystemDelayMs + 10);
ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
delay_max_ms);
// 4) Verify correct delay estimate at maximum delay boundary.
delay_ms = TruncateToMultipleOf10(kSystemDelayMs + delay_max_ms);
ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
delay_max_ms);
// 5) A delay above the maximum delay should give an estimate at the
// boundary (= (kMaxDelayBlocks - 1) * num_ms_per_block).
delay_ms += 20;
ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
delay_max_ms);
}
}
TEST_F(ApmTest, EchoControlMobile) {
// AECM won't use super-wideband.
SetFrameSampleRate(frame_, 32000);
EXPECT_NOERR(apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kBadSampleRateError,
apm_->echo_control_mobile()->Enable(true));
SetFrameSampleRate(frame_, 16000);
EXPECT_NOERR(apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError,
apm_->echo_control_mobile()->Enable(true));
SetFrameSampleRate(frame_, 32000);
EXPECT_EQ(apm_->kUnsupportedComponentError, apm_->ProcessStream(frame_));
// Turn AECM on (and AEC off)
Init(16000, 16000, 16000, 2, 2, 2, false);
EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
EXPECT_TRUE(apm_->echo_control_mobile()->is_enabled());
// Toggle routing modes
EchoControlMobile::RoutingMode mode[] = {
EchoControlMobile::kQuietEarpieceOrHeadset,
EchoControlMobile::kEarpiece,
EchoControlMobile::kLoudEarpiece,
EchoControlMobile::kSpeakerphone,
EchoControlMobile::kLoudSpeakerphone,
};
for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
EXPECT_EQ(apm_->kNoError,
apm_->echo_control_mobile()->set_routing_mode(mode[i]));
EXPECT_EQ(mode[i],
apm_->echo_control_mobile()->routing_mode());
}
// Turn comfort noise off/on
EXPECT_EQ(apm_->kNoError,
apm_->echo_control_mobile()->enable_comfort_noise(false));
EXPECT_FALSE(apm_->echo_control_mobile()->is_comfort_noise_enabled());
EXPECT_EQ(apm_->kNoError,
apm_->echo_control_mobile()->enable_comfort_noise(true));
EXPECT_TRUE(apm_->echo_control_mobile()->is_comfort_noise_enabled());
// Set and get echo path
const size_t echo_path_size =
apm_->echo_control_mobile()->echo_path_size_bytes();
rtc::scoped_ptr<char[]> echo_path_in(new char[echo_path_size]);
rtc::scoped_ptr<char[]> echo_path_out(new char[echo_path_size]);
EXPECT_EQ(apm_->kNullPointerError,
apm_->echo_control_mobile()->SetEchoPath(NULL, echo_path_size));
EXPECT_EQ(apm_->kNullPointerError,
apm_->echo_control_mobile()->GetEchoPath(NULL, echo_path_size));
EXPECT_EQ(apm_->kBadParameterError,
apm_->echo_control_mobile()->GetEchoPath(echo_path_out.get(), 1));
EXPECT_EQ(apm_->kNoError,
apm_->echo_control_mobile()->GetEchoPath(echo_path_out.get(),
echo_path_size));
for (size_t i = 0; i < echo_path_size; i++) {
echo_path_in[i] = echo_path_out[i] + 1;
}
EXPECT_EQ(apm_->kBadParameterError,
apm_->echo_control_mobile()->SetEchoPath(echo_path_in.get(), 1));
EXPECT_EQ(apm_->kNoError,
apm_->echo_control_mobile()->SetEchoPath(echo_path_in.get(),
echo_path_size));
EXPECT_EQ(apm_->kNoError,
apm_->echo_control_mobile()->GetEchoPath(echo_path_out.get(),
echo_path_size));
for (size_t i = 0; i < echo_path_size; i++) {
EXPECT_EQ(echo_path_in[i], echo_path_out[i]);
}
// Process a few frames with NS in the default disabled state. This exercises
// a different codepath than with it enabled.
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
// Turn AECM off
EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(false));
EXPECT_FALSE(apm_->echo_control_mobile()->is_enabled());
}
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 < sizeof(mode)/sizeof(*mode); i++) {
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_mode(mode[i]));
EXPECT_EQ(mode[i], apm_->gain_control()->mode());
}
// Testing invalid target levels
EXPECT_EQ(apm_->kBadParameterError,
apm_->gain_control()->set_target_level_dbfs(-3));
EXPECT_EQ(apm_->kBadParameterError,
apm_->gain_control()->set_target_level_dbfs(-40));
// Testing valid 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 < sizeof(level_dbfs)/sizeof(*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 invalid compression gains
EXPECT_EQ(apm_->kBadParameterError,
apm_->gain_control()->set_compression_gain_db(-1));
EXPECT_EQ(apm_->kBadParameterError,
apm_->gain_control()->set_compression_gain_db(100));
// Testing valid 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 < sizeof(gain_db)/sizeof(*gain_db); i++) {
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_compression_gain_db(gain_db[i]));
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 invalid level limits
EXPECT_EQ(apm_->kBadParameterError,
apm_->gain_control()->set_analog_level_limits(-1, 512));
EXPECT_EQ(apm_->kBadParameterError,
apm_->gain_control()->set_analog_level_limits(100000, 512));
EXPECT_EQ(apm_->kBadParameterError,
apm_->gain_control()->set_analog_level_limits(512, -1));
EXPECT_EQ(apm_->kBadParameterError,
apm_->gain_control()->set_analog_level_limits(512, 100000));
EXPECT_EQ(apm_->kBadParameterError,
apm_->gain_control()->set_analog_level_limits(512, 255));
// Testing valid 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 < sizeof(min_level)/sizeof(*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 < sizeof(min_level)/sizeof(*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());
}
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 < kSampleRatesSize; ++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 < kSampleRatesSize; ++i) {
RunManualVolumeChangeIsPossibleTest(kSampleRates[i]);
}
}
#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
const int kSampleRateHz = 16000;
const int kSamplesPerChannel =
AudioProcessing::kChunkSizeMs * kSampleRateHz / 1000;
const int kNumInputChannels = 2;
const int kNumOutputChannels = 1;
const int kNumChunks = 700;
const float kScaleFactor = 0.25f;
Config config;
std::vector<webrtc::Point> geometry;
geometry.push_back(webrtc::Point(0.f, 0.f, 0.f));
geometry.push_back(webrtc::Point(0.05f, 0.f, 0.f));
config.Set<Beamforming>(new Beamforming(true, geometry));
testing::NiceMock<MockNonlinearBeamformer>* beamformer =
new testing::NiceMock<MockNonlinearBeamformer>(geometry);
rtc::scoped_ptr<AudioProcessing> apm(
AudioProcessing::Create(config, beamformer));
EXPECT_EQ(kNoErr, apm->gain_control()->Enable(true));
ChannelBuffer<float> src_buf(kSamplesPerChannel, kNumInputChannels);
ChannelBuffer<float> dest_buf(kSamplesPerChannel, kNumOutputChannels);
const int max_length = kSamplesPerChannel * std::max(kNumInputChannels,
kNumOutputChannels);
rtc::scoped_ptr<int16_t[]> int_data(new int16_t[max_length]);
rtc::scoped_ptr<float[]> float_data(new float[max_length]);
std::string filename = ResourceFilePath("far", kSampleRateHz);
FILE* far_file = fopen(filename.c_str(), "rb");
ASSERT_TRUE(far_file != NULL) << "Could not open file " << filename << "\n";
const int kDefaultVolume = apm->gain_control()->stream_analog_level();
const int kDefaultCompressionGain =
apm->gain_control()->compression_gain_db();
bool is_target = false;
EXPECT_CALL(*beamformer, is_target_present())
.WillRepeatedly(testing::ReturnPointee(&is_target));
for (int i = 0; i < kNumChunks; ++i) {
ASSERT_TRUE(ReadChunk(far_file,
int_data.get(),
float_data.get(),
&src_buf));
for (int j = 0; j < kNumInputChannels; ++j) {
for (int k = 0; k < kSamplesPerChannel; ++k) {
src_buf.channels()[j][k] *= kScaleFactor;
}
}
EXPECT_EQ(kNoErr,
apm->ProcessStream(src_buf.channels(),
src_buf.num_frames(),
kSampleRateHz,
LayoutFromChannels(src_buf.num_channels()),
kSampleRateHz,
LayoutFromChannels(dest_buf.num_channels()),
dest_buf.channels()));
}
EXPECT_EQ(kDefaultVolume,
apm->gain_control()->stream_analog_level());
EXPECT_EQ(kDefaultCompressionGain,
apm->gain_control()->compression_gain_db());
rewind(far_file);
is_target = true;
for (int i = 0; i < kNumChunks; ++i) {
ASSERT_TRUE(ReadChunk(far_file,
int_data.get(),
float_data.get(),
&src_buf));
for (int j = 0; j < kNumInputChannels; ++j) {
for (int k = 0; k < kSamplesPerChannel; ++k) {
src_buf.channels()[j][k] *= kScaleFactor;
}
}
EXPECT_EQ(kNoErr,
apm->ProcessStream(src_buf.channels(),
src_buf.num_frames(),
kSampleRateHz,
LayoutFromChannels(src_buf.num_channels()),
kSampleRateHz,
LayoutFromChannels(dest_buf.num_channels()),
dest_buf.channels()));
}
EXPECT_LT(kDefaultVolume,
apm->gain_control()->stream_analog_level());
EXPECT_LT(kDefaultCompressionGain,
apm->gain_control()->compression_gain_db());
ASSERT_EQ(0, fclose(far_file));
}
#endif
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 < sizeof(level)/sizeof(*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
EXPECT_EQ(apm_->kNoError, apm_->high_pass_filter()->Enable(true));
EXPECT_TRUE(apm_->high_pass_filter()->is_enabled());
EXPECT_EQ(apm_->kNoError, apm_->high_pass_filter()->Enable(false));
EXPECT_FALSE(apm_->high_pass_filter()->is_enabled());
}
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 < sizeof(likelihood)/sizeof(*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 < sizeof(activity)/sizeof(*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) {
EXPECT_FALSE(apm_->echo_cancellation()->is_enabled());
EXPECT_FALSE(apm_->echo_control_mobile()->is_enabled());
EXPECT_FALSE(apm_->gain_control()->is_enabled());
EXPECT_FALSE(apm_->high_pass_filter()->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 < kSampleRatesSize; 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));
}
}
}
TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabledFloat) {
// Test that ProcessStream copies input to output even with no processing.
const size_t kSamples = 80;
const int sample_rate = 8000;
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(AudioProcessing::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]);
}
}
TEST_F(ApmTest, IdenticalInputChannelsResultInIdenticalOutputChannels) {
EnableAllComponents();
for (size_t i = 0; i < kProcessSampleRatesSize; 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_->data_, revframe_->samples_per_channel_);
ASSERT_EQ(kNoErr, apm_->AnalyzeReverseStream(revframe_));
CopyLeftToRightChannel(frame_->data_, frame_->samples_per_channel_);
frame_->vad_activity_ = AudioFrame::kVadUnknown;
ASSERT_EQ(kNoErr, apm_->set_stream_delay_ms(0));
apm_->echo_cancellation()->set_stream_drift_samples(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. Both VAD 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));
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));
// 5. Not using super-wb.
frame_->samples_per_channel_ = 160;
frame_->num_channels_ = 2;
frame_->sample_rate_hz_ = 16000;
// Enable AEC, which would require the filter in super-wb. We rely on the
// first few frames of data being unaffected by the AEC.
// TODO(andrew): This test, and the one below, rely rather tenuously on the
// behavior of the AEC. Think of something more robust.
EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
// Make sure we have extended filter enabled. This makes sure nothing is
// touched until we have a farend frame.
Config config;
config.Set<DelayCorrection>(new DelayCorrection(true));
apm_->SetExtraOptions(config);
SetFrameTo(frame_, 1000);
frame_copy.CopyFrom(*frame_);
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
apm_->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
// Check the test is valid. We should have distortion from the filter
// when AEC is enabled (which won't affect the audio).
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));
apm_->echo_cancellation()->set_stream_drift_samples(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) {
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) {
// StartDebugRecording() writes an additional init message. Don't start
// recording until after the first init to avoid the extra message.
EXPECT_NOERR(apm_->StartDebugRecording(out_filename.c_str()));
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_, 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_->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()));
apm_->echo_cancellation()->set_stream_drift_samples(msg.drift());
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_, 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_->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);
}
}
EXPECT_NOERR(apm_->StopDebugRecording());
fclose(in_file);
}
void ApmTest::VerifyDebugDumpTest(Format format) {
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");
EnableAllComponents();
ProcessDebugDump(in_filename, ref_filename, format);
ProcessDebugDump(ref_filename, out_filename, format);
FILE* ref_file = fopen(ref_filename.c_str(), "rb");
FILE* out_file = fopen(out_filename.c_str(), "rb");
ASSERT_TRUE(ref_file != NULL);
ASSERT_TRUE(out_file != NULL);
rtc::scoped_ptr<uint8_t[]> ref_bytes;
rtc::scoped_ptr<uint8_t[]> out_bytes;
size_t ref_size = ReadMessageBytesFromFile(ref_file, &ref_bytes);
size_t out_size = ReadMessageBytesFromFile(out_file, &out_bytes);
size_t bytes_read = 0;
while (ref_size > 0 && out_size > 0) {
bytes_read += ref_size;
EXPECT_EQ(ref_size, out_size);
EXPECT_EQ(0, memcmp(ref_bytes.get(), out_bytes.get(), ref_size));
ref_size = ReadMessageBytesFromFile(ref_file, &ref_bytes);
out_size = ReadMessageBytesFromFile(out_file, &out_bytes);
}
EXPECT_GT(bytes_read, 0u);
EXPECT_NE(0, feof(ref_file));
EXPECT_NE(0, feof(out_file));
ASSERT_EQ(0, fclose(ref_file));
ASSERT_EQ(0, fclose(out_file));
remove(ref_filename.c_str());
remove(out_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) {
const std::string filename =
test::TempFilename(test::OutputPath(), "debug_aec");
EXPECT_EQ(apm_->kNullPointerError,
apm_->StartDebugRecording(static_cast<const char*>(NULL)));
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// Stopping without having started should be OK.
EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(filename.c_str()));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
// 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
EXPECT_EQ(apm_->kUnsupportedFunctionError,
apm_->StartDebugRecording(filename.c_str()));
EXPECT_EQ(apm_->kUnsupportedFunctionError, apm_->StopDebugRecording());
// 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) {
FILE* fid = NULL;
EXPECT_EQ(apm_->kNullPointerError, apm_->StartDebugRecording(fid));
const std::string filename =
test::TempFilename(test::OutputPath(), "debug_aec");
fid = fopen(filename.c_str(), "w");
ASSERT_TRUE(fid);
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// Stopping without having started should be OK.
EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(fid));
EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
// Verify the file has been written.
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
EXPECT_EQ(apm_->kUnsupportedFunctionError,
apm_->StartDebugRecording(fid));
EXPECT_EQ(apm_->kUnsupportedFunctionError, apm_->StopDebugRecording());
ASSERT_EQ(0, fclose(fid));
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
audioproc::OutputData ref_data;
OpenFileAndReadMessage(ref_filename_, &ref_data);
Config config;
config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
rtc::scoped_ptr<AudioProcessing> fapm(AudioProcessing::Create(config));
EnableAllComponents();
EnableAllAPComponents(fapm.get());
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): Restore downmixing test cases.
if (test->num_input_channels() != test->num_output_channels())
continue;
const int num_render_channels = test->num_reverse_channels();
const int num_input_channels = test->num_input_channels();
const int num_output_channels = test->num_output_channels();
const int samples_per_channel = test->sample_rate() *
AudioProcessing::kChunkSizeMs / 1000;
Init(test->sample_rate(), test->sample_rate(), test->sample_rate(),
num_input_channels, num_output_channels, num_render_channels, true);
Init(fapm.get());
ChannelBuffer<int16_t> output_cb(samples_per_channel, num_input_channels);
ChannelBuffer<int16_t> output_int16(samples_per_channel,
num_input_channels);
int analog_level = 127;
while (ReadFrame(far_file_, revframe_, revfloat_cb_.get()) &&
ReadFrame(near_file_, frame_, float_cb_.get())) {
frame_->vad_activity_ = AudioFrame::kVadUnknown;
EXPECT_NOERR(apm_->AnalyzeReverseStream(revframe_));
EXPECT_NOERR(fapm->AnalyzeReverseStream(
revfloat_cb_->channels(),
samples_per_channel,
test->sample_rate(),
LayoutFromChannels(num_render_channels)));
EXPECT_NOERR(apm_->set_stream_delay_ms(0));
EXPECT_NOERR(fapm->set_stream_delay_ms(0));
apm_->echo_cancellation()->set_stream_drift_samples(0);
fapm->echo_cancellation()->set_stream_drift_samples(0);
EXPECT_NOERR(apm_->gain_control()->set_stream_analog_level(analog_level));
EXPECT_NOERR(fapm->gain_control()->set_stream_analog_level(analog_level));
EXPECT_NOERR(apm_->ProcessStream(frame_));
Deinterleave(frame_->data_, samples_per_channel, num_output_channels,
output_int16.channels());
EXPECT_NOERR(fapm->ProcessStream(
float_cb_->channels(),
samples_per_channel,
test->sample_rate(),
LayoutFromChannels(num_input_channels),
test->sample_rate(),
LayoutFromChannels(num_output_channels),
float_cb_->channels()));
for (int j = 0; j < num_output_channels; ++j) {
FloatToS16(float_cb_->channels()[j],
samples_per_channel,
output_cb.channels()[j]);
float variance = 0;
float snr = ComputeSNR(output_int16.channels()[j],
output_cb.channels()[j],
samples_per_channel, &variance);
#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
// There are a few chunks in the fixed-point profile that give low SNR.
// Listening confirmed the difference is acceptable.
const float kVarianceThreshold = 150;
const float kSNRThreshold = 10;
#else
const float kVarianceThreshold = 20;
const float kSNRThreshold = 20;
#endif
// Skip frames with low energy.
if (sqrt(variance) > kVarianceThreshold) {
EXPECT_LT(kSNRThreshold, snr);
}
}
analog_level = fapm->gain_control()->stream_analog_level();
EXPECT_EQ(apm_->gain_control()->stream_analog_level(),
fapm->gain_control()->stream_analog_level());
EXPECT_EQ(apm_->echo_cancellation()->stream_has_echo(),
fapm->echo_cancellation()->stream_has_echo());
EXPECT_NEAR(apm_->noise_suppression()->speech_probability(),
fapm->noise_suppression()->speech_probability(),
0.0005);
// Reset in case of downmixing.
frame_->num_channels_ = test->num_input_channels();
}
rewind(far_file_);
rewind(near_file_);
}
}
// 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 (!write_ref_data) {
OpenFileAndReadMessage(ref_filename_, &ref_data);
} else {
// Write the desired tests to the protobuf reference file.
for (size_t i = 0; i < kChannelsSize; i++) {
for (size_t j = 0; j < kChannelsSize; j++) {
for (size_t l = 0; l < kProcessSampleRatesSize; 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<DelayCorrection>(
new DelayCorrection(test->use_aec_extended_filter()));
apm_.reset(AudioProcessing::Create(config));
EnableAllComponents();
Init(test->sample_rate(),
test->sample_rate(),
test->sample_rate(),
test->num_input_channels(),
test->num_output_channels(),
test->num_reverse_channels(),
true);
int frame_count = 0;
int has_echo_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;
while (ReadFrame(far_file_, revframe_) && ReadFrame(near_file_, frame_)) {
EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
frame_->vad_activity_ = AudioFrame::kVadUnknown;
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
apm_->echo_cancellation()->set_stream_drift_samples(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(test->num_output_channels(), frame_->num_channels_);
max_output_average += MaxAudioFrame(*frame_);
if (apm_->echo_cancellation()->stream_has_echo()) {
has_echo_count++;
}
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();
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_ = test->num_input_channels();
frame_count++;
}
max_output_average /= frame_count;
analog_level_average /= frame_count;
ns_speech_prob_average /= frame_count;
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
EchoCancellation::Metrics echo_metrics;
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->GetMetrics(&echo_metrics));
int median = 0;
int std = 0;
float fraction_poor_delays = 0;
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->GetDelayMetrics(
&median, &std, &fraction_poor_delays));
int rms_level = apm_->level_estimator()->RMS();
EXPECT_LE(0, rms_level);
EXPECT_GE(127, rms_level);
#endif
if (!write_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)
const int kHasVoiceCountOffset = 3;
const int kHasVoiceCountNear = 3;
const int kMaxOutputAverageOffset = 9;
const int kMaxOutputAverageNear = 9;
#else
const int kHasVoiceCountOffset = 0;
const int kHasVoiceCountNear = kIntNear;
const int kMaxOutputAverageOffset = 0;
const int kMaxOutputAverageNear = kIntNear;
#endif
EXPECT_NEAR(test->has_echo_count(), has_echo_count, kIntNear);
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)
audioproc::Test::EchoMetrics reference = test->echo_metrics();
TestStats(echo_metrics.residual_echo_return_loss,
reference.residual_echo_return_loss());
TestStats(echo_metrics.echo_return_loss,
reference.echo_return_loss());
TestStats(echo_metrics.echo_return_loss_enhancement,
reference.echo_return_loss_enhancement());
TestStats(echo_metrics.a_nlp,
reference.a_nlp());
const double kFloatNear = 0.0005;
audioproc::Test::DelayMetrics reference_delay = test->delay_metrics();
EXPECT_NEAR(reference_delay.median(), median, kIntNear);
EXPECT_NEAR(reference_delay.std(), std, kIntNear);
EXPECT_NEAR(reference_delay.fraction_poor_delays(), fraction_poor_delays,
kFloatNear);
EXPECT_NEAR(test->rms_level(), rms_level, kIntNear);
EXPECT_NEAR(test->ns_speech_probability_average(),
ns_speech_prob_average,
kFloatNear);
#endif
} else {
test->set_has_echo_count(has_echo_count);
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)
audioproc::Test::EchoMetrics* message = test->mutable_echo_metrics();
WriteStatsMessage(echo_metrics.residual_echo_return_loss,
message->mutable_residual_echo_return_loss());
WriteStatsMessage(echo_metrics.echo_return_loss,
message->mutable_echo_return_loss());
WriteStatsMessage(echo_metrics.echo_return_loss_enhancement,
message->mutable_echo_return_loss_enhancement());
WriteStatsMessage(echo_metrics.a_nlp,
message->mutable_a_nlp());
audioproc::Test::DelayMetrics* message_delay =
test->mutable_delay_metrics();
message_delay->set_median(median);
message_delay->set_std(std);
message_delay->set_fraction_poor_delays(fraction_poor_delays);
test->set_rms_level(rms_level);
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);
#endif
}
rewind(far_file_);
rewind(near_file_);
}
if (write_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},
};
size_t channel_format_size = sizeof(cf) / sizeof(*cf);
rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create());
// Enable one component just to ensure some processing takes place.
ap->noise_suppression()->Enable(true);
for (size_t i = 0; i < channel_format_size; ++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,
int 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 (int 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::tr1::tuple<int, int, int, double> AudioProcessingTestData;
class AudioProcessingTest
: public testing::TestWithParam<AudioProcessingTestData> {
public:
AudioProcessingTest()
: input_rate_(std::tr1::get<0>(GetParam())),
output_rate_(std::tr1::get<1>(GetParam())),
reverse_rate_(std::tr1::get<2>(GetParam())),
expected_snr_(std::tr1::get<3>(GetParam())) {}
virtual ~AudioProcessingTest() {}
static void SetUpTestCase() {
// Create all needed output reference files.
const int kNativeRates[] = {8000, 16000, 32000};
const size_t kNativeRatesSize =
sizeof(kNativeRates) / sizeof(*kNativeRates);
const int kNumChannels[] = {1, 2};
const size_t kNumChannelsSize =
sizeof(kNumChannels) / sizeof(*kNumChannels);
for (size_t i = 0; i < kNativeRatesSize; ++i) {
for (size_t j = 0; j < kNumChannelsSize; ++j) {
for (size_t k = 0; k < kNumChannelsSize; ++k) {
// The reference files always have matching input and output channels.
ProcessFormat(kNativeRates[i],
kNativeRates[i],
kNativeRates[i],
kNumChannels[j],
kNumChannels[j],
kNumChannels[k],
"ref");
}
}
}
}
static void TearDownTestCase() {
ClearTempFiles();
}
// Runs a process pass on files with the given parameters and dumps the output
// to a file specified with |output_file_prefix|.
static void ProcessFormat(int input_rate,
int output_rate,
int reverse_rate,
int num_input_channels,
int num_output_channels,
int num_reverse_channels,
std::string output_file_prefix) {
Config config;
config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(config));
EnableAllAPComponents(ap.get());
ap->Initialize(input_rate,
output_rate,
reverse_rate,
LayoutFromChannels(num_input_channels),
LayoutFromChannels(num_output_channels),
LayoutFromChannels(num_reverse_channels));
FILE* far_file = fopen(ResourceFilePath("far", reverse_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_rate,
num_input_channels,
num_output_channels,
num_reverse_channels).c_str(), "wb");
ASSERT_TRUE(far_file != NULL);
ASSERT_TRUE(near_file != NULL);
ASSERT_TRUE(out_file != NULL);
ChannelBuffer<float> fwd_cb(SamplesFromRate(input_rate),
num_input_channels);
ChannelBuffer<float> rev_cb(SamplesFromRate(reverse_rate),
num_reverse_channels);
ChannelBuffer<float> out_cb(SamplesFromRate(output_rate),
num_output_channels);
// Temporary buffers.
const int max_length =
2 * std::max(out_cb.num_frames(),
std::max(fwd_cb.num_frames(),
rev_cb.num_frames()));
rtc::scoped_ptr<float[]> float_data(new float[max_length]);
rtc::scoped_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->AnalyzeReverseStream(
rev_cb.channels(),
rev_cb.num_frames(),
reverse_rate,
LayoutFromChannels(num_reverse_channels)));
EXPECT_NOERR(ap->set_stream_delay_ms(0));
ap->echo_cancellation()->set_stream_drift_samples(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()));
Interleave(out_cb.channels(),
out_cb.num_frames(),
out_cb.num_channels(),
float_data.get());
// Dump output to file.
int out_length = out_cb.num_channels() * out_cb.num_frames();
ASSERT_EQ(static_cast<size_t>(out_length),
fwrite(float_data.get(), sizeof(float_data[0]),
out_length, out_file));
analog_level = ap->gain_control()->stream_analog_level();
}
fclose(far_file);
fclose(near_file);
fclose(out_file);
}
protected:
int input_rate_;
int output_rate_;
int reverse_rate_;
double expected_snr_;
};
TEST_P(AudioProcessingTest, Formats) {
struct ChannelFormat {
int num_input;
int num_output;
int num_reverse;
};
ChannelFormat cf[] = {
{1, 1, 1},
{1, 1, 2},
{2, 1, 1},
{2, 1, 2},
{2, 2, 1},
{2, 2, 2},
};
size_t channel_format_size = sizeof(cf) / sizeof(*cf);
for (size_t i = 0; i < channel_format_size; ++i) {
ProcessFormat(input_rate_,
output_rate_,
reverse_rate_,
cf[i].num_input,
cf[i].num_output,
cf[i].num_reverse,
"out");
int min_ref_rate = std::min(input_rate_, output_rate_);
int ref_rate;
if (min_ref_rate > 16000) {
ref_rate = 32000;
} else if (min_ref_rate > 8000) {
ref_rate = 16000;
} else {
ref_rate = 8000;
}
#ifdef WEBRTC_AUDIOPROC_FIXED_PROFILE
ref_rate = std::min(ref_rate, 16000);
#endif
FILE* out_file = fopen(OutputFilePath("out",
input_rate_,
output_rate_,
reverse_rate_,
cf[i].num_input,
cf[i].num_output,
cf[i].num_reverse).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,
cf[i].num_output,
cf[i].num_output,
cf[i].num_reverse).c_str(), "rb");
ASSERT_TRUE(out_file != NULL);
ASSERT_TRUE(ref_file != NULL);
const int ref_length = SamplesFromRate(ref_rate) * cf[i].num_output;
const int out_length = SamplesFromRate(output_rate_) * cf[i].num_output;
// Data from the reference file.
rtc::scoped_ptr<float[]> ref_data(new float[ref_length]);
// Data from the output file.
rtc::scoped_ptr<float[]> out_data(new float[out_length]);
// Data from the resampled output, in case the reference and output rates
// don't match.
rtc::scoped_ptr<float[]> cmp_data(new float[ref_length]);
PushResampler<float> resampler;
resampler.InitializeIfNeeded(output_rate_, ref_rate, cf[i].num_output);
// 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 (input_rate_ != ref_rate) {
// Input resampling delay.
expected_delay_sec +=
PushSincResampler::AlgorithmicDelaySeconds(input_rate_);
}
if (output_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(output_rate_);
}
int expected_delay = floor(expected_delay_sec * ref_rate + 0.5f) *
cf[i].num_output;
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 (output_rate_ != ref_rate) {
// Resample the output back to its internal processing rate if necssary.
ASSERT_EQ(ref_length, 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.