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/*
* Copyright (c) 2015 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 <stddef.h> // size_t
#include <memory>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/audio/echo_canceller3_factory.h"
#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
#include "modules/audio_processing/test/debug_dump_replayer.h"
#include "modules/audio_processing/test/test_utils.h"
#include "rtc_base/task_queue_for_test.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
namespace test {
namespace {
void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer,
const StreamConfig& config) {
auto& buffer_ref = *buffer;
if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
buffer_ref->num_channels() != config.num_channels()) {
buffer_ref.reset(
new ChannelBuffer<float>(config.num_frames(), config.num_channels()));
}
}
class DebugDumpGenerator {
public:
DebugDumpGenerator(absl::string_view input_file_name,
int input_rate_hz,
int input_channels,
absl::string_view reverse_file_name,
int reverse_rate_hz,
int reverse_channels,
absl::string_view dump_file_name,
bool enable_pre_amplifier);
// Constructor that uses default input files.
explicit DebugDumpGenerator(const AudioProcessing::Config& apm_config);
~DebugDumpGenerator();
// Changes the sample rate of the input audio to the APM.
void SetInputRate(int rate_hz);
// Sets if converts stereo input signal to mono by discarding other channels.
void ForceInputMono(bool mono);
// Changes the sample rate of the reverse audio to the APM.
void SetReverseRate(int rate_hz);
// Sets if converts stereo reverse signal to mono by discarding other
// channels.
void ForceReverseMono(bool mono);
// Sets the required sample rate of the APM output.
void SetOutputRate(int rate_hz);
// Sets the required channels of the APM output.
void SetOutputChannels(int channels);
std::string dump_file_name() const { return dump_file_name_; }
void StartRecording();
void Process(size_t num_blocks);
void StopRecording();
AudioProcessing* apm() const { return apm_.get(); }
private:
static void ReadAndDeinterleave(ResampleInputAudioFile* audio,
int channels,
const StreamConfig& config,
float* const* buffer);
// APM input/output settings.
StreamConfig input_config_;
StreamConfig reverse_config_;
StreamConfig output_config_;
// Input file format.
const std::string input_file_name_;
ResampleInputAudioFile input_audio_;
const int input_file_channels_;
// Reverse file format.
const std::string reverse_file_name_;
ResampleInputAudioFile reverse_audio_;
const int reverse_file_channels_;
// Buffer for APM input/output.
std::unique_ptr<ChannelBuffer<float>> input_;
std::unique_ptr<ChannelBuffer<float>> reverse_;
std::unique_ptr<ChannelBuffer<float>> output_;
bool enable_pre_amplifier_;
TaskQueueForTest worker_queue_;
rtc::scoped_refptr<AudioProcessing> apm_;
const std::string dump_file_name_;
};
DebugDumpGenerator::DebugDumpGenerator(absl::string_view input_file_name,
int input_rate_hz,
int input_channels,
absl::string_view reverse_file_name,
int reverse_rate_hz,
int reverse_channels,
absl::string_view dump_file_name,
bool enable_pre_amplifier)
: input_config_(input_rate_hz, input_channels),
reverse_config_(reverse_rate_hz, reverse_channels),
output_config_(input_rate_hz, input_channels),
input_audio_(input_file_name, input_rate_hz, input_rate_hz),
input_file_channels_(input_channels),
reverse_audio_(reverse_file_name, reverse_rate_hz, reverse_rate_hz),
reverse_file_channels_(reverse_channels),
input_(new ChannelBuffer<float>(input_config_.num_frames(),
input_config_.num_channels())),
reverse_(new ChannelBuffer<float>(reverse_config_.num_frames(),
reverse_config_.num_channels())),
output_(new ChannelBuffer<float>(output_config_.num_frames(),
output_config_.num_channels())),
enable_pre_amplifier_(enable_pre_amplifier),
worker_queue_("debug_dump_generator_worker_queue"),
dump_file_name_(dump_file_name) {
AudioProcessingBuilderForTesting apm_builder;
apm_ = apm_builder.Create();
}
DebugDumpGenerator::DebugDumpGenerator(
const AudioProcessing::Config& apm_config)
: DebugDumpGenerator(ResourcePath("near32_stereo", "pcm"),
32000,
2,
ResourcePath("far32_stereo", "pcm"),
32000,
2,
TempFilename(OutputPath(), "debug_aec"),
apm_config.pre_amplifier.enabled) {
apm_->ApplyConfig(apm_config);
}
DebugDumpGenerator::~DebugDumpGenerator() {
remove(dump_file_name_.c_str());
}
void DebugDumpGenerator::SetInputRate(int rate_hz) {
input_audio_.set_output_rate_hz(rate_hz);
input_config_.set_sample_rate_hz(rate_hz);
MaybeResetBuffer(&input_, input_config_);
}
void DebugDumpGenerator::ForceInputMono(bool mono) {
const int channels = mono ? 1 : input_file_channels_;
input_config_.set_num_channels(channels);
MaybeResetBuffer(&input_, input_config_);
}
void DebugDumpGenerator::SetReverseRate(int rate_hz) {
reverse_audio_.set_output_rate_hz(rate_hz);
reverse_config_.set_sample_rate_hz(rate_hz);
MaybeResetBuffer(&reverse_, reverse_config_);
}
void DebugDumpGenerator::ForceReverseMono(bool mono) {
const int channels = mono ? 1 : reverse_file_channels_;
reverse_config_.set_num_channels(channels);
MaybeResetBuffer(&reverse_, reverse_config_);
}
void DebugDumpGenerator::SetOutputRate(int rate_hz) {
output_config_.set_sample_rate_hz(rate_hz);
MaybeResetBuffer(&output_, output_config_);
}
void DebugDumpGenerator::SetOutputChannels(int channels) {
output_config_.set_num_channels(channels);
MaybeResetBuffer(&output_, output_config_);
}
void DebugDumpGenerator::StartRecording() {
apm_->AttachAecDump(
AecDumpFactory::Create(dump_file_name_.c_str(), -1, worker_queue_.Get()));
}
void DebugDumpGenerator::Process(size_t num_blocks) {
for (size_t i = 0; i < num_blocks; ++i) {
ReadAndDeinterleave(&reverse_audio_, reverse_file_channels_,
reverse_config_, reverse_->channels());
ReadAndDeinterleave(&input_audio_, input_file_channels_, input_config_,
input_->channels());
RTC_CHECK_EQ(AudioProcessing::kNoError, apm_->set_stream_delay_ms(100));
apm_->set_stream_analog_level(100);
if (enable_pre_amplifier_) {
apm_->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCapturePreGain(1 + i % 10));
}
apm_->set_stream_key_pressed(i % 10 == 9);
RTC_CHECK_EQ(AudioProcessing::kNoError,
apm_->ProcessStream(input_->channels(), input_config_,
output_config_, output_->channels()));
RTC_CHECK_EQ(
AudioProcessing::kNoError,
apm_->ProcessReverseStream(reverse_->channels(), reverse_config_,
reverse_config_, reverse_->channels()));
}
}
void DebugDumpGenerator::StopRecording() {
apm_->DetachAecDump();
}
void DebugDumpGenerator::ReadAndDeinterleave(ResampleInputAudioFile* audio,
int channels,
const StreamConfig& config,
float* const* buffer) {
const size_t num_frames = config.num_frames();
const int out_channels = config.num_channels();
std::vector<int16_t> signal(channels * num_frames);
audio->Read(num_frames * channels, &signal[0]);
// We only allow reducing number of channels by discarding some channels.
RTC_CHECK_LE(out_channels, channels);
for (int channel = 0; channel < out_channels; ++channel) {
for (size_t i = 0; i < num_frames; ++i) {
buffer[channel][i] = S16ToFloat(signal[i * channels + channel]);
}
}
}
} // namespace
class DebugDumpTest : public ::testing::Test {
public:
// VerifyDebugDump replays a debug dump using APM and verifies that the result
// is bit-exact-identical to the output channel in the dump. This is only
// guaranteed if the debug dump is started on the first frame.
void VerifyDebugDump(absl::string_view in_filename);
private:
DebugDumpReplayer debug_dump_replayer_;
};
void DebugDumpTest::VerifyDebugDump(absl::string_view in_filename) {
ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(in_filename));
while (const std::optional<audioproc::Event> event =
debug_dump_replayer_.GetNextEvent()) {
debug_dump_replayer_.RunNextEvent();
if (event->type() == audioproc::Event::STREAM) {
const audioproc::Stream* msg = &event->stream();
const StreamConfig output_config = debug_dump_replayer_.GetOutputConfig();
const ChannelBuffer<float>* output = debug_dump_replayer_.GetOutput();
// Check that output of APM is bit-exact to the output in the dump.
ASSERT_EQ(output_config.num_channels(),
static_cast<size_t>(msg->output_channel_size()));
ASSERT_EQ(output_config.num_frames() * sizeof(float),
msg->output_channel(0).size());
for (int i = 0; i < msg->output_channel_size(); ++i) {
ASSERT_EQ(0,
memcmp(output->channels()[i], msg->output_channel(i).data(),
msg->output_channel(i).size()));
}
}
}
}
TEST_F(DebugDumpTest, SimpleCase) {
DebugDumpGenerator generator(/*apm_config=*/{});
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
VerifyDebugDump(generator.dump_file_name());
}
// TODO(bugs.webrtc.org/345674542): Fix/enable.
#if defined(__has_feature) && __has_feature(undefined_behavior_sanitizer)
TEST_F(DebugDumpTest, DISABLED_ChangeInputFormat) {
#else
TEST_F(DebugDumpTest, ChangeInputFormat) {
#endif
DebugDumpGenerator generator(/*apm_config=*/{});
generator.StartRecording();
generator.Process(100);
generator.SetInputRate(48000);
generator.ForceInputMono(true);
// Number of output channel should not be larger than that of input. APM will
// fail otherwise.
generator.SetOutputChannels(1);
generator.Process(100);
generator.StopRecording();
VerifyDebugDump(generator.dump_file_name());
}
// TODO(bugs.webrtc.org/345674542): Fix/enable.
#if defined(__has_feature) && __has_feature(undefined_behavior_sanitizer)
TEST_F(DebugDumpTest, DISABLED_ChangeReverseFormat) {
#else
TEST_F(DebugDumpTest, ChangeReverseFormat) {
#endif
DebugDumpGenerator generator(/*apm_config=*/{});
generator.StartRecording();
generator.Process(100);
generator.SetReverseRate(48000);
generator.ForceReverseMono(true);
generator.Process(100);
generator.StopRecording();
VerifyDebugDump(generator.dump_file_name());
}
TEST_F(DebugDumpTest, ChangeOutputFormat) {
DebugDumpGenerator generator(/*apm_config=*/{});
generator.StartRecording();
generator.Process(100);
generator.SetOutputRate(48000);
generator.SetOutputChannels(1);
generator.Process(100);
generator.StopRecording();
VerifyDebugDump(generator.dump_file_name());
}
TEST_F(DebugDumpTest, ToggleAec) {
AudioProcessing::Config apm_config;
apm_config.echo_canceller.enabled = true;
DebugDumpGenerator generator(apm_config);
generator.StartRecording();
generator.Process(100);
apm_config.echo_canceller.enabled = false;
generator.apm()->ApplyConfig(apm_config);
generator.Process(100);
generator.StopRecording();
VerifyDebugDump(generator.dump_file_name());
}
TEST_F(DebugDumpTest, VerifyCombinedExperimentalStringInclusive) {
AudioProcessing::Config apm_config;
apm_config.echo_canceller.enabled = true;
apm_config.gain_controller1.analog_gain_controller.enabled = true;
apm_config.gain_controller1.analog_gain_controller.startup_min_volume = 0;
DebugDumpGenerator generator(apm_config);
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
DebugDumpReplayer debug_dump_replayer_;
ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
while (const std::optional<audioproc::Event> event =
debug_dump_replayer_.GetNextEvent()) {
debug_dump_replayer_.RunNextEvent();
if (event->type() == audioproc::Event::CONFIG) {
const audioproc::Config* msg = &event->config();
ASSERT_TRUE(msg->has_experiments_description());
EXPECT_PRED_FORMAT2(::testing::IsSubstring, "EchoController",
msg->experiments_description().c_str());
}
}
}
TEST_F(DebugDumpTest, VerifyCombinedExperimentalStringExclusive) {
AudioProcessing::Config apm_config;
apm_config.echo_canceller.enabled = true;
DebugDumpGenerator generator(apm_config);
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
DebugDumpReplayer debug_dump_replayer_;
ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
while (const std::optional<audioproc::Event> event =
debug_dump_replayer_.GetNextEvent()) {
debug_dump_replayer_.RunNextEvent();
if (event->type() == audioproc::Event::CONFIG) {
const audioproc::Config* msg = &event->config();
ASSERT_TRUE(msg->has_experiments_description());
EXPECT_PRED_FORMAT2(::testing::IsNotSubstring,
"AgcClippingLevelExperiment",
msg->experiments_description().c_str());
}
}
}
TEST_F(DebugDumpTest, VerifyAec3ExperimentalString) {
AudioProcessing::Config apm_config;
apm_config.echo_canceller.enabled = true;
DebugDumpGenerator generator(apm_config);
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
DebugDumpReplayer debug_dump_replayer_;
ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
while (const std::optional<audioproc::Event> event =
debug_dump_replayer_.GetNextEvent()) {
debug_dump_replayer_.RunNextEvent();
if (event->type() == audioproc::Event::CONFIG) {
const audioproc::Config* msg = &event->config();
ASSERT_TRUE(msg->has_experiments_description());
EXPECT_PRED_FORMAT2(::testing::IsSubstring, "EchoController",
msg->experiments_description().c_str());
}
}
}
TEST_F(DebugDumpTest, VerifyEmptyExperimentalString) {
DebugDumpGenerator generator(/*apm_config=*/{});
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
DebugDumpReplayer debug_dump_replayer_;
ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
while (const std::optional<audioproc::Event> event =
debug_dump_replayer_.GetNextEvent()) {
debug_dump_replayer_.RunNextEvent();
if (event->type() == audioproc::Event::CONFIG) {
const audioproc::Config* msg = &event->config();
ASSERT_TRUE(msg->has_experiments_description());
EXPECT_EQ(0u, msg->experiments_description().size());
}
}
}
// AGC is not supported on Android or iOS.
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
#define MAYBE_ToggleAgc DISABLED_ToggleAgc
#else
#define MAYBE_ToggleAgc ToggleAgc
#endif
TEST_F(DebugDumpTest, MAYBE_ToggleAgc) {
DebugDumpGenerator generator(/*apm_config=*/{});
generator.StartRecording();
generator.Process(100);
AudioProcessing::Config apm_config = generator.apm()->GetConfig();
apm_config.gain_controller1.enabled = !apm_config.gain_controller1.enabled;
generator.apm()->ApplyConfig(apm_config);
generator.Process(100);
generator.StopRecording();
VerifyDebugDump(generator.dump_file_name());
}
TEST_F(DebugDumpTest, ToggleNs) {
DebugDumpGenerator generator(/*apm_config=*/{});
generator.StartRecording();
generator.Process(100);
AudioProcessing::Config apm_config = generator.apm()->GetConfig();
apm_config.noise_suppression.enabled = !apm_config.noise_suppression.enabled;
generator.apm()->ApplyConfig(apm_config);
generator.Process(100);
generator.StopRecording();
VerifyDebugDump(generator.dump_file_name());
}
TEST_F(DebugDumpTest, TransientSuppressionOn) {
DebugDumpGenerator generator(/*apm_config=*/{});
AudioProcessing::Config apm_config = generator.apm()->GetConfig();
apm_config.transient_suppression.enabled = true;
generator.apm()->ApplyConfig(apm_config);
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
VerifyDebugDump(generator.dump_file_name());
}
TEST_F(DebugDumpTest, PreAmplifierIsOn) {
AudioProcessing::Config apm_config;
apm_config.pre_amplifier.enabled = true;
DebugDumpGenerator generator(apm_config);
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
VerifyDebugDump(generator.dump_file_name());
}
} // namespace test
} // namespace webrtc