| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| #include "modules/audio_processing/include/audio_processing.h" |
| |
| #include <math.h> |
| #include <stdio.h> |
| |
| #include <algorithm> |
| #include <cmath> |
| #include <limits> |
| #include <memory> |
| #include <numeric> |
| #include <queue> |
| #include <string> |
| |
| #include "absl/flags/flag.h" |
| #include "absl/strings/string_view.h" |
| #include "api/audio/echo_detector_creator.h" |
| #include "api/make_ref_counted.h" |
| #include "common_audio/include/audio_util.h" |
| #include "common_audio/resampler/include/push_resampler.h" |
| #include "common_audio/resampler/push_sinc_resampler.h" |
| #include "common_audio/signal_processing/include/signal_processing_library.h" |
| #include "modules/audio_processing/aec_dump/aec_dump_factory.h" |
| #include "modules/audio_processing/audio_processing_impl.h" |
| #include "modules/audio_processing/include/mock_audio_processing.h" |
| #include "modules/audio_processing/test/audio_processing_builder_for_testing.h" |
| #include "modules/audio_processing/test/protobuf_utils.h" |
| #include "modules/audio_processing/test/test_utils.h" |
| #include "rtc_base/arraysize.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/fake_clock.h" |
| #include "rtc_base/gtest_prod_util.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/numerics/safe_minmax.h" |
| #include "rtc_base/protobuf_utils.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "rtc_base/swap_queue.h" |
| #include "rtc_base/system/arch.h" |
| #include "rtc_base/task_queue_for_test.h" |
| #include "rtc_base/thread.h" |
| #include "system_wrappers/include/cpu_features_wrapper.h" |
| #include "test/gtest.h" |
| #include "test/testsupport/file_utils.h" |
| |
| #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" |
| #include "external/webrtc/webrtc/modules/audio_processing/test/unittest.pb.h" |
| #else |
| #include "modules/audio_processing/debug.pb.h" |
| #include "modules/audio_processing/test/unittest.pb.h" |
| #endif |
| |
| ABSL_FLAG(bool, |
| write_apm_ref_data, |
| false, |
| "Write ApmTest.Process results to file, instead of comparing results " |
| "to the existing reference data file."); |
| |
| namespace webrtc { |
| namespace { |
| |
| // All sample rates used by APM internally during processing. Other input / |
| // output rates are resampled to / from one of these. |
| const int kProcessSampleRates[] = {16000, 32000, 48000}; |
| |
| enum StreamDirection { kForward = 0, kReverse }; |
| |
| void ConvertToFloat(const int16_t* int_data, ChannelBuffer<float>* cb) { |
| ChannelBuffer<int16_t> cb_int(cb->num_frames(), cb->num_channels()); |
| Deinterleave(int_data, cb->num_frames(), cb->num_channels(), |
| cb_int.channels()); |
| for (size_t i = 0; i < cb->num_channels(); ++i) { |
| S16ToFloat(cb_int.channels()[i], cb->num_frames(), cb->channels()[i]); |
| } |
| } |
| |
| void ConvertToFloat(const Int16FrameData& frame, ChannelBuffer<float>* cb) { |
| ConvertToFloat(frame.data.data(), cb); |
| } |
| |
| void MixStereoToMono(const float* stereo, |
| float* mono, |
| size_t samples_per_channel) { |
| for (size_t i = 0; i < samples_per_channel; ++i) |
| mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) / 2; |
| } |
| |
| void MixStereoToMono(const int16_t* stereo, |
| int16_t* mono, |
| size_t samples_per_channel) { |
| for (size_t i = 0; i < samples_per_channel; ++i) |
| mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) >> 1; |
| } |
| |
| void CopyLeftToRightChannel(int16_t* stereo, size_t samples_per_channel) { |
| for (size_t i = 0; i < samples_per_channel; i++) { |
| stereo[i * 2 + 1] = stereo[i * 2]; |
| } |
| } |
| |
| void VerifyChannelsAreEqual(const int16_t* stereo, size_t samples_per_channel) { |
| for (size_t i = 0; i < samples_per_channel; i++) { |
| EXPECT_EQ(stereo[i * 2 + 1], stereo[i * 2]); |
| } |
| } |
| |
| void SetFrameTo(Int16FrameData* frame, int16_t value) { |
| for (size_t i = 0; i < frame->samples_per_channel * frame->num_channels; |
| ++i) { |
| frame->data[i] = value; |
| } |
| } |
| |
| void SetFrameTo(Int16FrameData* frame, int16_t left, int16_t right) { |
| ASSERT_EQ(2u, frame->num_channels); |
| for (size_t i = 0; i < frame->samples_per_channel * 2; i += 2) { |
| frame->data[i] = left; |
| frame->data[i + 1] = right; |
| } |
| } |
| |
| void ScaleFrame(Int16FrameData* frame, float scale) { |
| for (size_t i = 0; i < frame->samples_per_channel * frame->num_channels; |
| ++i) { |
| frame->data[i] = FloatS16ToS16(frame->data[i] * scale); |
| } |
| } |
| |
| bool FrameDataAreEqual(const Int16FrameData& frame1, |
| const Int16FrameData& 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.data(), frame2.data.data(), |
| frame1.samples_per_channel * frame1.num_channels * sizeof(int16_t))) { |
| return false; |
| } |
| return true; |
| } |
| |
| rtc::ArrayView<int16_t> GetMutableFrameData(Int16FrameData* frame) { |
| int16_t* ptr = frame->data.data(); |
| const size_t len = frame->samples_per_channel * frame->num_channels; |
| return rtc::ArrayView<int16_t>(ptr, len); |
| } |
| |
| rtc::ArrayView<const int16_t> GetFrameData(const Int16FrameData& frame) { |
| const int16_t* ptr = frame.data.data(); |
| const size_t len = frame.samples_per_channel * frame.num_channels; |
| return rtc::ArrayView<const int16_t>(ptr, len); |
| } |
| |
| void EnableAllAPComponents(AudioProcessing* ap) { |
| AudioProcessing::Config apm_config = ap->GetConfig(); |
| apm_config.echo_canceller.enabled = true; |
| #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE) |
| apm_config.echo_canceller.mobile_mode = true; |
| |
| apm_config.gain_controller1.enabled = true; |
| apm_config.gain_controller1.mode = |
| AudioProcessing::Config::GainController1::kAdaptiveDigital; |
| #elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) |
| apm_config.echo_canceller.mobile_mode = false; |
| |
| apm_config.gain_controller1.enabled = true; |
| apm_config.gain_controller1.mode = |
| AudioProcessing::Config::GainController1::kAdaptiveAnalog; |
| #endif |
| |
| apm_config.noise_suppression.enabled = true; |
| |
| apm_config.high_pass_filter.enabled = true; |
| apm_config.pipeline.maximum_internal_processing_rate = 48000; |
| ap->ApplyConfig(apm_config); |
| } |
| |
| // These functions are only used by ApmTest.Process. |
| template <class T> |
| T AbsValue(T a) { |
| return a > 0 ? a : -a; |
| } |
| |
| int16_t MaxAudioFrame(const Int16FrameData& frame) { |
| const size_t length = frame.samples_per_channel * frame.num_channels; |
| int16_t max_data = AbsValue(frame.data[0]); |
| for (size_t i = 1; i < length; i++) { |
| max_data = std::max(max_data, AbsValue(frame.data[i])); |
| } |
| |
| return max_data; |
| } |
| |
| void OpenFileAndWriteMessage(absl::string_view filename, |
| const MessageLite& msg) { |
| FILE* file = fopen(std::string(filename).c_str(), "wb"); |
| ASSERT_TRUE(file != NULL); |
| |
| int32_t size = rtc::checked_cast<int32_t>(msg.ByteSizeLong()); |
| ASSERT_GT(size, 0); |
| std::unique_ptr<uint8_t[]> array(new uint8_t[size]); |
| ASSERT_TRUE(msg.SerializeToArray(array.get(), size)); |
| |
| ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file)); |
| ASSERT_EQ(static_cast<size_t>(size), |
| fwrite(array.get(), sizeof(array[0]), size, file)); |
| fclose(file); |
| } |
| |
| std::string ResourceFilePath(absl::string_view name, int sample_rate_hz) { |
| rtc::StringBuilder ss; |
| // Resource files are all stereo. |
| ss << name << sample_rate_hz / 1000 << "_stereo"; |
| return test::ResourcePath(ss.str(), "pcm"); |
| } |
| |
| // Temporary filenames unique to this process. Used to be able to run these |
| // tests in parallel as each process needs to be running in isolation they can't |
| // have competing filenames. |
| std::map<std::string, std::string> temp_filenames; |
| |
| std::string OutputFilePath(absl::string_view name, |
| int input_rate, |
| int output_rate, |
| int reverse_input_rate, |
| int reverse_output_rate, |
| size_t num_input_channels, |
| size_t num_output_channels, |
| size_t num_reverse_input_channels, |
| size_t num_reverse_output_channels, |
| StreamDirection file_direction) { |
| rtc::StringBuilder ss; |
| ss << name << "_i" << num_input_channels << "_" << input_rate / 1000 << "_ir" |
| << num_reverse_input_channels << "_" << reverse_input_rate / 1000 << "_"; |
| if (num_output_channels == 1) { |
| ss << "mono"; |
| } else if (num_output_channels == 2) { |
| ss << "stereo"; |
| } else { |
| RTC_DCHECK_NOTREACHED(); |
| } |
| ss << output_rate / 1000; |
| if (num_reverse_output_channels == 1) { |
| ss << "_rmono"; |
| } else if (num_reverse_output_channels == 2) { |
| ss << "_rstereo"; |
| } else { |
| RTC_DCHECK_NOTREACHED(); |
| } |
| ss << reverse_output_rate / 1000; |
| ss << "_d" << file_direction << "_pcm"; |
| |
| std::string filename = ss.str(); |
| if (temp_filenames[filename].empty()) |
| temp_filenames[filename] = test::TempFilename(test::OutputPath(), filename); |
| return temp_filenames[filename]; |
| } |
| |
| void ClearTempFiles() { |
| for (auto& kv : temp_filenames) |
| remove(kv.second.c_str()); |
| } |
| |
| // Only remove "out" files. Keep "ref" files. |
| void ClearTempOutFiles() { |
| for (auto it = temp_filenames.begin(); it != temp_filenames.end();) { |
| const std::string& filename = it->first; |
| if (filename.substr(0, 3).compare("out") == 0) { |
| remove(it->second.c_str()); |
| temp_filenames.erase(it++); |
| } else { |
| it++; |
| } |
| } |
| } |
| |
| void OpenFileAndReadMessage(absl::string_view filename, MessageLite* msg) { |
| FILE* file = fopen(std::string(filename).c_str(), "rb"); |
| ASSERT_TRUE(file != NULL); |
| ReadMessageFromFile(file, msg); |
| fclose(file); |
| } |
| |
| // Reads a 10 ms chunk (actually AudioProcessing::GetFrameSize() samples per |
| // channel) of int16 interleaved audio from the given (assumed stereo) file, |
| // converts to deinterleaved float (optionally downmixing) and returns the |
| // result in `cb`. Returns false if the file ended (or on error) and true |
| // otherwise. |
| // |
| // `int_data` and `float_data` are just temporary space that must be |
| // sufficiently large to hold the 10 ms chunk. |
| bool ReadChunk(FILE* file, |
| int16_t* int_data, |
| float* float_data, |
| ChannelBuffer<float>* cb) { |
| // The files always contain stereo audio. |
| size_t frame_size = cb->num_frames() * 2; |
| size_t read_count = fread(int_data, sizeof(int16_t), frame_size, file); |
| if (read_count != frame_size) { |
| // Check that the file really ended. |
| RTC_DCHECK(feof(file)); |
| return false; // This is expected. |
| } |
| |
| S16ToFloat(int_data, frame_size, float_data); |
| if (cb->num_channels() == 1) { |
| MixStereoToMono(float_data, cb->channels()[0], cb->num_frames()); |
| } else { |
| Deinterleave(float_data, cb->num_frames(), 2, cb->channels()); |
| } |
| |
| return true; |
| } |
| |
| // Returns the reference file name that matches the current CPU |
| // architecture/optimizations. |
| std::string GetReferenceFilename() { |
| #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE) |
| return test::ResourcePath("audio_processing/output_data_fixed", "pb"); |
| #elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) |
| if (GetCPUInfo(kAVX2) != 0) { |
| return test::ResourcePath("audio_processing/output_data_float_avx2", "pb"); |
| } |
| return test::ResourcePath("audio_processing/output_data_float", "pb"); |
| #endif |
| } |
| |
| // Flag that can temporarily be enabled for local debugging to inspect |
| // `ApmTest.VerifyDebugDump(Int|Float)` failures. Do not upload code changes |
| // with this flag set to true. |
| constexpr bool kDumpWhenExpectMessageEqFails = false; |
| |
| // Checks the debug constants values used in this file so that no code change is |
| // submitted with values temporarily used for local debugging. |
| TEST(ApmUnitTests, CheckDebugConstants) { |
| ASSERT_FALSE(kDumpWhenExpectMessageEqFails); |
| } |
| |
| // Expects the equality of `actual` and `expected` by inspecting a hard-coded |
| // subset of `audioproc::Stream` fields. |
| void ExpectStreamFieldsEq(const audioproc::Stream& actual, |
| const audioproc::Stream& expected) { |
| EXPECT_EQ(actual.input_data(), expected.input_data()); |
| EXPECT_EQ(actual.output_data(), expected.output_data()); |
| EXPECT_EQ(actual.delay(), expected.delay()); |
| EXPECT_EQ(actual.drift(), expected.drift()); |
| EXPECT_EQ(actual.applied_input_volume(), expected.applied_input_volume()); |
| EXPECT_EQ(actual.keypress(), expected.keypress()); |
| } |
| |
| // Expects the equality of `actual` and `expected` by inspecting a hard-coded |
| // subset of `audioproc::Event` fields. |
| void ExpectEventFieldsEq(const audioproc::Event& actual, |
| const audioproc::Event& expected) { |
| EXPECT_EQ(actual.type(), expected.type()); |
| if (actual.type() != expected.type()) { |
| return; |
| } |
| switch (actual.type()) { |
| case audioproc::Event::STREAM: |
| ExpectStreamFieldsEq(actual.stream(), expected.stream()); |
| break; |
| default: |
| // Not implemented. |
| break; |
| } |
| } |
| |
| // Returns true if the `actual` and `expected` byte streams share the same size |
| // and contain the same data. If they differ and `kDumpWhenExpectMessageEqFails` |
| // is true, checks the equality of a subset of `audioproc::Event` (nested) |
| // fields. |
| bool ExpectMessageEq(rtc::ArrayView<const uint8_t> actual, |
| rtc::ArrayView<const uint8_t> expected) { |
| EXPECT_EQ(actual.size(), expected.size()); |
| if (actual.size() != expected.size()) { |
| return false; |
| } |
| if (memcmp(actual.data(), expected.data(), actual.size()) == 0) { |
| // Same message. No need to parse. |
| return true; |
| } |
| if (kDumpWhenExpectMessageEqFails) { |
| // Parse differing messages and expect equality to produce detailed error |
| // messages. |
| audioproc::Event event_actual, event_expected; |
| RTC_DCHECK(event_actual.ParseFromArray(actual.data(), actual.size())); |
| RTC_DCHECK(event_expected.ParseFromArray(expected.data(), expected.size())); |
| ExpectEventFieldsEq(event_actual, event_expected); |
| } |
| return false; |
| } |
| |
| class ApmTest : public ::testing::Test { |
| protected: |
| ApmTest(); |
| virtual void SetUp(); |
| virtual void TearDown(); |
| |
| static void SetUpTestSuite() {} |
| |
| static void TearDownTestSuite() { ClearTempFiles(); } |
| |
| // Used to select between int and float interface tests. |
| enum Format { kIntFormat, kFloatFormat }; |
| |
| void Init(int sample_rate_hz, |
| int output_sample_rate_hz, |
| int reverse_sample_rate_hz, |
| size_t num_input_channels, |
| size_t num_output_channels, |
| size_t num_reverse_channels, |
| bool open_output_file); |
| void Init(AudioProcessing* ap); |
| void EnableAllComponents(); |
| bool ReadFrame(FILE* file, Int16FrameData* frame); |
| bool ReadFrame(FILE* file, Int16FrameData* frame, ChannelBuffer<float>* cb); |
| void ReadFrameWithRewind(FILE* file, Int16FrameData* frame); |
| void ReadFrameWithRewind(FILE* file, |
| Int16FrameData* frame, |
| ChannelBuffer<float>* cb); |
| void ProcessDelayVerificationTest(int delay_ms, |
| int system_delay_ms, |
| int delay_min, |
| int delay_max); |
| void TestChangingChannelsInt16Interface( |
| size_t num_channels, |
| AudioProcessing::Error expected_return); |
| void TestChangingForwardChannels(size_t num_in_channels, |
| size_t num_out_channels, |
| AudioProcessing::Error expected_return); |
| void TestChangingReverseChannels(size_t num_rev_channels, |
| AudioProcessing::Error expected_return); |
| void RunQuantizedVolumeDoesNotGetStuckTest(int sample_rate); |
| void RunManualVolumeChangeIsPossibleTest(int sample_rate); |
| void StreamParametersTest(Format format); |
| int ProcessStreamChooser(Format format); |
| int AnalyzeReverseStreamChooser(Format format); |
| void ProcessDebugDump(absl::string_view in_filename, |
| absl::string_view out_filename, |
| Format format, |
| int max_size_bytes); |
| void VerifyDebugDumpTest(Format format); |
| |
| const std::string output_path_; |
| const std::string ref_filename_; |
| rtc::scoped_refptr<AudioProcessing> apm_; |
| Int16FrameData frame_; |
| Int16FrameData revframe_; |
| std::unique_ptr<ChannelBuffer<float>> float_cb_; |
| std::unique_ptr<ChannelBuffer<float>> revfloat_cb_; |
| int output_sample_rate_hz_; |
| size_t num_output_channels_; |
| FILE* far_file_; |
| FILE* near_file_; |
| FILE* out_file_; |
| }; |
| |
| ApmTest::ApmTest() |
| : output_path_(test::OutputPath()), |
| ref_filename_(GetReferenceFilename()), |
| output_sample_rate_hz_(0), |
| num_output_channels_(0), |
| far_file_(NULL), |
| near_file_(NULL), |
| out_file_(NULL) { |
| apm_ = AudioProcessingBuilderForTesting().Create(); |
| AudioProcessing::Config apm_config = apm_->GetConfig(); |
| apm_config.gain_controller1.analog_gain_controller.enabled = false; |
| apm_config.pipeline.maximum_internal_processing_rate = 48000; |
| apm_->ApplyConfig(apm_config); |
| } |
| |
| void ApmTest::SetUp() { |
| ASSERT_TRUE(apm_.get() != NULL); |
| |
| Init(32000, 32000, 32000, 2, 2, 2, false); |
| } |
| |
| void ApmTest::TearDown() { |
| if (far_file_) { |
| ASSERT_EQ(0, fclose(far_file_)); |
| } |
| far_file_ = NULL; |
| |
| if (near_file_) { |
| ASSERT_EQ(0, fclose(near_file_)); |
| } |
| near_file_ = NULL; |
| |
| if (out_file_) { |
| ASSERT_EQ(0, fclose(out_file_)); |
| } |
| out_file_ = NULL; |
| } |
| |
| void ApmTest::Init(AudioProcessing* ap) { |
| ASSERT_EQ( |
| kNoErr, |
| ap->Initialize({{{frame_.sample_rate_hz, frame_.num_channels}, |
| {output_sample_rate_hz_, num_output_channels_}, |
| {revframe_.sample_rate_hz, revframe_.num_channels}, |
| {revframe_.sample_rate_hz, revframe_.num_channels}}})); |
| } |
| |
| void ApmTest::Init(int sample_rate_hz, |
| int output_sample_rate_hz, |
| int reverse_sample_rate_hz, |
| size_t num_input_channels, |
| size_t num_output_channels, |
| size_t num_reverse_channels, |
| bool open_output_file) { |
| SetContainerFormat(sample_rate_hz, num_input_channels, &frame_, &float_cb_); |
| output_sample_rate_hz_ = output_sample_rate_hz; |
| num_output_channels_ = num_output_channels; |
| |
| SetContainerFormat(reverse_sample_rate_hz, num_reverse_channels, &revframe_, |
| &revfloat_cb_); |
| Init(apm_.get()); |
| |
| if (far_file_) { |
| ASSERT_EQ(0, fclose(far_file_)); |
| } |
| std::string filename = ResourceFilePath("far", sample_rate_hz); |
| far_file_ = fopen(filename.c_str(), "rb"); |
| ASSERT_TRUE(far_file_ != NULL) << "Could not open file " << filename << "\n"; |
| |
| if (near_file_) { |
| ASSERT_EQ(0, fclose(near_file_)); |
| } |
| filename = ResourceFilePath("near", sample_rate_hz); |
| near_file_ = fopen(filename.c_str(), "rb"); |
| ASSERT_TRUE(near_file_ != NULL) << "Could not open file " << filename << "\n"; |
| |
| if (open_output_file) { |
| if (out_file_) { |
| ASSERT_EQ(0, fclose(out_file_)); |
| } |
| filename = OutputFilePath( |
| "out", sample_rate_hz, output_sample_rate_hz, reverse_sample_rate_hz, |
| reverse_sample_rate_hz, num_input_channels, num_output_channels, |
| num_reverse_channels, num_reverse_channels, kForward); |
| out_file_ = fopen(filename.c_str(), "wb"); |
| ASSERT_TRUE(out_file_ != NULL) |
| << "Could not open file " << filename << "\n"; |
| } |
| } |
| |
| void ApmTest::EnableAllComponents() { |
| EnableAllAPComponents(apm_.get()); |
| } |
| |
| bool ApmTest::ReadFrame(FILE* file, |
| Int16FrameData* 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.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.data(), frame->data.data(), |
| frame->samples_per_channel); |
| } |
| |
| if (cb) { |
| ConvertToFloat(*frame, cb); |
| } |
| return true; |
| } |
| |
| bool ApmTest::ReadFrame(FILE* file, Int16FrameData* 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, |
| Int16FrameData* 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, Int16FrameData* frame) { |
| ReadFrameWithRewind(file, frame, NULL); |
| } |
| |
| int ApmTest::ProcessStreamChooser(Format format) { |
| if (format == kIntFormat) { |
| return apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data()); |
| } |
| return apm_->ProcessStream( |
| float_cb_->channels(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(output_sample_rate_hz_, num_output_channels_), |
| float_cb_->channels()); |
| } |
| |
| int ApmTest::AnalyzeReverseStreamChooser(Format format) { |
| if (format == kIntFormat) { |
| return apm_->ProcessReverseStream( |
| revframe_.data.data(), |
| StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels), |
| StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels), |
| revframe_.data.data()); |
| } |
| return apm_->AnalyzeReverseStream( |
| revfloat_cb_->channels(), |
| StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels)); |
| } |
| |
| void ApmTest::ProcessDelayVerificationTest(int delay_ms, |
| int system_delay_ms, |
| int delay_min, |
| int delay_max) { |
| // The `revframe_` and `frame_` should include the proper frame information, |
| // hence can be used for extracting information. |
| Int16FrameData tmp_frame; |
| std::queue<Int16FrameData*> 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) { |
| Int16FrameData* frame = new Int16FrameData(); |
| frame->CopyFrom(tmp_frame); |
| frame_queue.push(frame); |
| frame_delay++; |
| causal = false; |
| } |
| while (frame_delay > 0) { |
| Int16FrameData* frame = new Int16FrameData(); |
| 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) { |
| Int16FrameData* frame = new Int16FrameData(); |
| 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); |
| Int16FrameData* reverse_frame = frame; |
| Int16FrameData* process_frame = frame_queue.front(); |
| if (!causal) { |
| reverse_frame = frame_queue.front(); |
| // When we call ProcessStream() the frame is modified, so we can't use the |
| // pointer directly when things are non-causal. Use an intermediate frame |
| // and copy the data. |
| process_frame = &tmp_frame; |
| process_frame->CopyFrom(*frame); |
| } |
| EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream( |
| reverse_frame->data.data(), |
| StreamConfig(reverse_frame->sample_rate_hz, |
| reverse_frame->num_channels), |
| StreamConfig(reverse_frame->sample_rate_hz, |
| reverse_frame->num_channels), |
| reverse_frame->data.data())); |
| EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(system_delay_ms)); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessStream(process_frame->data.data(), |
| StreamConfig(process_frame->sample_rate_hz, |
| process_frame->num_channels), |
| StreamConfig(process_frame->sample_rate_hz, |
| process_frame->num_channels), |
| process_frame->data.data())); |
| frame = frame_queue.front(); |
| frame_queue.pop(); |
| delete frame; |
| |
| if (frame_count == 250) { |
| // Discard the first delay metrics to avoid convergence effects. |
| static_cast<void>(apm_->GetStatistics()); |
| } |
| } |
| |
| rewind(near_file_); |
| while (!frame_queue.empty()) { |
| Int16FrameData* frame = frame_queue.front(); |
| frame_queue.pop(); |
| delete frame; |
| } |
| // Calculate expected delay estimate and acceptable regions. Further, |
| // limit them w.r.t. AEC delay estimation support. |
| const size_t samples_per_ms = |
| rtc::SafeMin<size_t>(16u, frame_.samples_per_channel / 10); |
| const int expected_median = |
| rtc::SafeClamp<int>(delay_ms - system_delay_ms, delay_min, delay_max); |
| const int expected_median_high = rtc::SafeClamp<int>( |
| expected_median + rtc::dchecked_cast<int>(96 / samples_per_ms), delay_min, |
| delay_max); |
| const int expected_median_low = rtc::SafeClamp<int>( |
| expected_median - rtc::dchecked_cast<int>(96 / samples_per_ms), delay_min, |
| delay_max); |
| // Verify delay metrics. |
| AudioProcessingStats stats = apm_->GetStatistics(); |
| ASSERT_TRUE(stats.delay_median_ms.has_value()); |
| int32_t median = *stats.delay_median_ms; |
| EXPECT_GE(expected_median_high, median); |
| EXPECT_LE(expected_median_low, median); |
| } |
| |
| void ApmTest::StreamParametersTest(Format format) { |
| // No errors when the components are disabled. |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format)); |
| |
| // -- Missing AGC level -- |
| AudioProcessing::Config apm_config = apm_->GetConfig(); |
| apm_config.gain_controller1.enabled = true; |
| apm_->ApplyConfig(apm_config); |
| EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format)); |
| |
| // Resets after successful ProcessStream(). |
| apm_->set_stream_analog_level(127); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format)); |
| EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format)); |
| |
| // Other stream parameters set correctly. |
| apm_config.echo_canceller.enabled = true; |
| apm_config.echo_canceller.mobile_mode = false; |
| apm_->ApplyConfig(apm_config); |
| EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100)); |
| EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format)); |
| apm_config.gain_controller1.enabled = false; |
| apm_->ApplyConfig(apm_config); |
| |
| // -- Missing delay -- |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format)); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format)); |
| |
| // Resets after successful ProcessStream(). |
| EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100)); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format)); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format)); |
| |
| // Other stream parameters set correctly. |
| apm_config.gain_controller1.enabled = true; |
| apm_->ApplyConfig(apm_config); |
| apm_->set_stream_analog_level(127); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format)); |
| apm_config.gain_controller1.enabled = false; |
| apm_->ApplyConfig(apm_config); |
| |
| // -- No stream parameters -- |
| EXPECT_EQ(apm_->kNoError, AnalyzeReverseStreamChooser(format)); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format)); |
| |
| // -- All there -- |
| EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100)); |
| apm_->set_stream_analog_level(127); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format)); |
| } |
| |
| TEST_F(ApmTest, StreamParametersInt) { |
| StreamParametersTest(kIntFormat); |
| } |
| |
| TEST_F(ApmTest, StreamParametersFloat) { |
| StreamParametersTest(kFloatFormat); |
| } |
| |
| void ApmTest::TestChangingChannelsInt16Interface( |
| size_t num_channels, |
| AudioProcessing::Error expected_return) { |
| frame_.num_channels = num_channels; |
| |
| EXPECT_EQ(expected_return, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| EXPECT_EQ(expected_return, |
| apm_->ProcessReverseStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| } |
| |
| void ApmTest::TestChangingForwardChannels( |
| size_t num_in_channels, |
| size_t num_out_channels, |
| AudioProcessing::Error expected_return) { |
| const StreamConfig input_stream = {frame_.sample_rate_hz, num_in_channels}; |
| const StreamConfig output_stream = {output_sample_rate_hz_, num_out_channels}; |
| |
| EXPECT_EQ(expected_return, |
| apm_->ProcessStream(float_cb_->channels(), input_stream, |
| output_stream, float_cb_->channels())); |
| } |
| |
| void ApmTest::TestChangingReverseChannels( |
| size_t num_rev_channels, |
| AudioProcessing::Error expected_return) { |
| const ProcessingConfig processing_config = { |
| {{frame_.sample_rate_hz, apm_->num_input_channels()}, |
| {output_sample_rate_hz_, apm_->num_output_channels()}, |
| {frame_.sample_rate_hz, num_rev_channels}, |
| {frame_.sample_rate_hz, num_rev_channels}}}; |
| |
| EXPECT_EQ( |
| expected_return, |
| apm_->ProcessReverseStream( |
| float_cb_->channels(), processing_config.reverse_input_stream(), |
| processing_config.reverse_output_stream(), float_cb_->channels())); |
| } |
| |
| TEST_F(ApmTest, ChannelsInt16Interface) { |
| // Testing number of invalid and valid channels. |
| Init(16000, 16000, 16000, 4, 4, 4, false); |
| |
| TestChangingChannelsInt16Interface(0, apm_->kBadNumberChannelsError); |
| |
| for (size_t i = 1; i < 4; i++) { |
| TestChangingChannelsInt16Interface(i, kNoErr); |
| EXPECT_EQ(i, apm_->num_input_channels()); |
| } |
| } |
| |
| TEST_F(ApmTest, Channels) { |
| // Testing number of invalid and valid channels. |
| Init(16000, 16000, 16000, 4, 4, 4, false); |
| |
| TestChangingForwardChannels(0, 1, apm_->kBadNumberChannelsError); |
| TestChangingReverseChannels(0, apm_->kBadNumberChannelsError); |
| |
| for (size_t i = 1; i < 4; ++i) { |
| for (size_t j = 0; j < 1; ++j) { |
| // Output channels much be one or match input channels. |
| if (j == 1 || i == j) { |
| TestChangingForwardChannels(i, j, kNoErr); |
| TestChangingReverseChannels(i, kNoErr); |
| |
| EXPECT_EQ(i, apm_->num_input_channels()); |
| EXPECT_EQ(j, apm_->num_output_channels()); |
| // The number of reverse channels used for processing to is always 1. |
| EXPECT_EQ(1u, apm_->num_reverse_channels()); |
| } else { |
| TestChangingForwardChannels(i, j, |
| AudioProcessing::kBadNumberChannelsError); |
| } |
| } |
| } |
| } |
| |
| TEST_F(ApmTest, SampleRatesInt) { |
| // Testing some valid sample rates. |
| for (int sample_rate : {8000, 12000, 16000, 32000, 44100, 48000, 96000}) { |
| SetContainerFormat(sample_rate, 2, &frame_, &float_cb_); |
| EXPECT_NOERR(ProcessStreamChooser(kIntFormat)); |
| } |
| } |
| |
| // This test repeatedly reconfigures the pre-amplifier in APM, processes a |
| // number of frames, and checks that output signal has the right level. |
| TEST_F(ApmTest, PreAmplifier) { |
| // Fill the audio frame with a sawtooth pattern. |
| rtc::ArrayView<int16_t> frame_data = GetMutableFrameData(&frame_); |
| const size_t samples_per_channel = frame_.samples_per_channel; |
| for (size_t i = 0; i < samples_per_channel; i++) { |
| for (size_t ch = 0; ch < frame_.num_channels; ++ch) { |
| frame_data[i + ch * samples_per_channel] = 10000 * ((i % 3) - 1); |
| } |
| } |
| // Cache the frame in tmp_frame. |
| Int16FrameData tmp_frame; |
| tmp_frame.CopyFrom(frame_); |
| |
| auto compute_power = [](const Int16FrameData& frame) { |
| rtc::ArrayView<const int16_t> data = GetFrameData(frame); |
| return std::accumulate(data.begin(), data.end(), 0.0f, |
| [](float a, float b) { return a + b * b; }) / |
| data.size() / 32768 / 32768; |
| }; |
| |
| const float input_power = compute_power(tmp_frame); |
| // Double-check that the input data is large compared to the error kEpsilon. |
| constexpr float kEpsilon = 1e-4f; |
| RTC_DCHECK_GE(input_power, 10 * kEpsilon); |
| |
| // 1. Enable pre-amp with 0 dB gain. |
| AudioProcessing::Config config = apm_->GetConfig(); |
| config.pre_amplifier.enabled = true; |
| config.pre_amplifier.fixed_gain_factor = 1.0f; |
| apm_->ApplyConfig(config); |
| |
| for (int i = 0; i < 20; ++i) { |
| frame_.CopyFrom(tmp_frame); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat)); |
| } |
| float output_power = compute_power(frame_); |
| EXPECT_NEAR(output_power, input_power, kEpsilon); |
| config = apm_->GetConfig(); |
| EXPECT_EQ(config.pre_amplifier.fixed_gain_factor, 1.0f); |
| |
| // 2. Change pre-amp gain via ApplyConfig. |
| config.pre_amplifier.fixed_gain_factor = 2.0f; |
| apm_->ApplyConfig(config); |
| |
| for (int i = 0; i < 20; ++i) { |
| frame_.CopyFrom(tmp_frame); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat)); |
| } |
| output_power = compute_power(frame_); |
| EXPECT_NEAR(output_power, 4 * input_power, kEpsilon); |
| config = apm_->GetConfig(); |
| EXPECT_EQ(config.pre_amplifier.fixed_gain_factor, 2.0f); |
| |
| // 3. Change pre-amp gain via a RuntimeSetting. |
| apm_->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCapturePreGain(1.5f)); |
| |
| for (int i = 0; i < 20; ++i) { |
| frame_.CopyFrom(tmp_frame); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat)); |
| } |
| output_power = compute_power(frame_); |
| EXPECT_NEAR(output_power, 2.25 * input_power, kEpsilon); |
| config = apm_->GetConfig(); |
| EXPECT_EQ(config.pre_amplifier.fixed_gain_factor, 1.5f); |
| } |
| |
| // Ensures that the emulated analog mic gain functionality runs without |
| // crashing. |
| TEST_F(ApmTest, AnalogMicGainEmulation) { |
| // Fill the audio frame with a sawtooth pattern. |
| rtc::ArrayView<int16_t> frame_data = GetMutableFrameData(&frame_); |
| const size_t samples_per_channel = frame_.samples_per_channel; |
| for (size_t i = 0; i < samples_per_channel; i++) { |
| for (size_t ch = 0; ch < frame_.num_channels; ++ch) { |
| frame_data[i + ch * samples_per_channel] = 100 * ((i % 3) - 1); |
| } |
| } |
| // Cache the frame in tmp_frame. |
| Int16FrameData tmp_frame; |
| tmp_frame.CopyFrom(frame_); |
| |
| // Enable the analog gain emulation. |
| AudioProcessing::Config config = apm_->GetConfig(); |
| config.capture_level_adjustment.enabled = true; |
| config.capture_level_adjustment.analog_mic_gain_emulation.enabled = true; |
| config.capture_level_adjustment.analog_mic_gain_emulation.initial_level = 21; |
| config.gain_controller1.enabled = true; |
| config.gain_controller1.mode = |
| AudioProcessing::Config::GainController1::Mode::kAdaptiveAnalog; |
| config.gain_controller1.analog_gain_controller.enabled = true; |
| apm_->ApplyConfig(config); |
| |
| // Process a number of frames to ensure that the code runs without crashes. |
| for (int i = 0; i < 20; ++i) { |
| frame_.CopyFrom(tmp_frame); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat)); |
| } |
| } |
| |
| // This test repeatedly reconfigures the capture level adjustment functionality |
| // in APM, processes a number of frames, and checks that output signal has the |
| // right level. |
| TEST_F(ApmTest, CaptureLevelAdjustment) { |
| // Fill the audio frame with a sawtooth pattern. |
| rtc::ArrayView<int16_t> frame_data = GetMutableFrameData(&frame_); |
| const size_t samples_per_channel = frame_.samples_per_channel; |
| for (size_t i = 0; i < samples_per_channel; i++) { |
| for (size_t ch = 0; ch < frame_.num_channels; ++ch) { |
| frame_data[i + ch * samples_per_channel] = 100 * ((i % 3) - 1); |
| } |
| } |
| // Cache the frame in tmp_frame. |
| Int16FrameData tmp_frame; |
| tmp_frame.CopyFrom(frame_); |
| |
| auto compute_power = [](const Int16FrameData& frame) { |
| rtc::ArrayView<const int16_t> data = GetFrameData(frame); |
| return std::accumulate(data.begin(), data.end(), 0.0f, |
| [](float a, float b) { return a + b * b; }) / |
| data.size() / 32768 / 32768; |
| }; |
| |
| const float input_power = compute_power(tmp_frame); |
| // Double-check that the input data is large compared to the error kEpsilon. |
| constexpr float kEpsilon = 1e-20f; |
| RTC_DCHECK_GE(input_power, 10 * kEpsilon); |
| |
| // 1. Enable pre-amp with 0 dB gain. |
| AudioProcessing::Config config = apm_->GetConfig(); |
| config.capture_level_adjustment.enabled = true; |
| config.capture_level_adjustment.pre_gain_factor = 0.5f; |
| config.capture_level_adjustment.post_gain_factor = 4.f; |
| const float expected_output_power1 = |
| config.capture_level_adjustment.pre_gain_factor * |
| config.capture_level_adjustment.pre_gain_factor * |
| config.capture_level_adjustment.post_gain_factor * |
| config.capture_level_adjustment.post_gain_factor * input_power; |
| apm_->ApplyConfig(config); |
| |
| for (int i = 0; i < 20; ++i) { |
| frame_.CopyFrom(tmp_frame); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat)); |
| } |
| float output_power = compute_power(frame_); |
| EXPECT_NEAR(output_power, expected_output_power1, kEpsilon); |
| config = apm_->GetConfig(); |
| EXPECT_EQ(config.capture_level_adjustment.pre_gain_factor, 0.5f); |
| EXPECT_EQ(config.capture_level_adjustment.post_gain_factor, 4.f); |
| |
| // 2. Change pre-amp gain via ApplyConfig. |
| config.capture_level_adjustment.pre_gain_factor = 1.0f; |
| config.capture_level_adjustment.post_gain_factor = 2.f; |
| const float expected_output_power2 = |
| config.capture_level_adjustment.pre_gain_factor * |
| config.capture_level_adjustment.pre_gain_factor * |
| config.capture_level_adjustment.post_gain_factor * |
| config.capture_level_adjustment.post_gain_factor * input_power; |
| apm_->ApplyConfig(config); |
| |
| for (int i = 0; i < 20; ++i) { |
| frame_.CopyFrom(tmp_frame); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat)); |
| } |
| output_power = compute_power(frame_); |
| EXPECT_NEAR(output_power, expected_output_power2, kEpsilon); |
| config = apm_->GetConfig(); |
| EXPECT_EQ(config.capture_level_adjustment.pre_gain_factor, 1.0f); |
| EXPECT_EQ(config.capture_level_adjustment.post_gain_factor, 2.f); |
| |
| // 3. Change pre-amp gain via a RuntimeSetting. |
| constexpr float kPreGain3 = 0.5f; |
| constexpr float kPostGain3 = 3.f; |
| const float expected_output_power3 = |
| kPreGain3 * kPreGain3 * kPostGain3 * kPostGain3 * input_power; |
| |
| apm_->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCapturePreGain(kPreGain3)); |
| apm_->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCapturePostGain(kPostGain3)); |
| |
| for (int i = 0; i < 20; ++i) { |
| frame_.CopyFrom(tmp_frame); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat)); |
| } |
| output_power = compute_power(frame_); |
| EXPECT_NEAR(output_power, expected_output_power3, kEpsilon); |
| config = apm_->GetConfig(); |
| EXPECT_EQ(config.capture_level_adjustment.pre_gain_factor, 0.5f); |
| EXPECT_EQ(config.capture_level_adjustment.post_gain_factor, 3.f); |
| } |
| |
| TEST_F(ApmTest, GainControl) { |
| AudioProcessing::Config config = apm_->GetConfig(); |
| config.gain_controller1.enabled = false; |
| apm_->ApplyConfig(config); |
| config.gain_controller1.enabled = true; |
| apm_->ApplyConfig(config); |
| |
| // Testing gain modes |
| for (auto mode : |
| {AudioProcessing::Config::GainController1::kAdaptiveDigital, |
| AudioProcessing::Config::GainController1::kFixedDigital, |
| AudioProcessing::Config::GainController1::kAdaptiveAnalog}) { |
| config.gain_controller1.mode = mode; |
| apm_->ApplyConfig(config); |
| apm_->set_stream_analog_level(100); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat)); |
| } |
| |
| // Testing target levels |
| for (int target_level_dbfs : {0, 15, 31}) { |
| config.gain_controller1.target_level_dbfs = target_level_dbfs; |
| apm_->ApplyConfig(config); |
| apm_->set_stream_analog_level(100); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat)); |
| } |
| |
| // Testing compression gains |
| for (int compression_gain_db : {0, 10, 90}) { |
| config.gain_controller1.compression_gain_db = compression_gain_db; |
| apm_->ApplyConfig(config); |
| apm_->set_stream_analog_level(100); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat)); |
| } |
| |
| // Testing limiter off/on |
| for (bool enable : {false, true}) { |
| config.gain_controller1.enable_limiter = enable; |
| apm_->ApplyConfig(config); |
| apm_->set_stream_analog_level(100); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat)); |
| } |
| |
| // Testing level limits. |
| constexpr int kMinLevel = 0; |
| constexpr int kMaxLevel = 255; |
| apm_->set_stream_analog_level(kMinLevel); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat)); |
| apm_->set_stream_analog_level((kMinLevel + kMaxLevel) / 2); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat)); |
| apm_->set_stream_analog_level(kMaxLevel); |
| EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat)); |
| } |
| |
| #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) |
| using ApmDeathTest = ApmTest; |
| |
| TEST_F(ApmDeathTest, GainControlDiesOnTooLowTargetLevelDbfs) { |
| auto config = apm_->GetConfig(); |
| config.gain_controller1.enabled = true; |
| config.gain_controller1.target_level_dbfs = -1; |
| EXPECT_DEATH(apm_->ApplyConfig(config), ""); |
| } |
| |
| TEST_F(ApmDeathTest, GainControlDiesOnTooHighTargetLevelDbfs) { |
| auto config = apm_->GetConfig(); |
| config.gain_controller1.enabled = true; |
| config.gain_controller1.target_level_dbfs = 32; |
| EXPECT_DEATH(apm_->ApplyConfig(config), ""); |
| } |
| |
| TEST_F(ApmDeathTest, GainControlDiesOnTooLowCompressionGainDb) { |
| auto config = apm_->GetConfig(); |
| config.gain_controller1.enabled = true; |
| config.gain_controller1.compression_gain_db = -1; |
| EXPECT_DEATH(apm_->ApplyConfig(config), ""); |
| } |
| |
| TEST_F(ApmDeathTest, GainControlDiesOnTooHighCompressionGainDb) { |
| auto config = apm_->GetConfig(); |
| config.gain_controller1.enabled = true; |
| config.gain_controller1.compression_gain_db = 91; |
| EXPECT_DEATH(apm_->ApplyConfig(config), ""); |
| } |
| |
| TEST_F(ApmDeathTest, ApmDiesOnTooLowAnalogLevel) { |
| auto config = apm_->GetConfig(); |
| config.gain_controller1.enabled = true; |
| apm_->ApplyConfig(config); |
| EXPECT_DEATH(apm_->set_stream_analog_level(-1), ""); |
| } |
| |
| TEST_F(ApmDeathTest, ApmDiesOnTooHighAnalogLevel) { |
| auto config = apm_->GetConfig(); |
| config.gain_controller1.enabled = true; |
| apm_->ApplyConfig(config); |
| EXPECT_DEATH(apm_->set_stream_analog_level(256), ""); |
| } |
| #endif |
| |
| void ApmTest::RunQuantizedVolumeDoesNotGetStuckTest(int sample_rate) { |
| Init(sample_rate, sample_rate, sample_rate, 2, 2, 2, false); |
| auto config = apm_->GetConfig(); |
| config.gain_controller1.enabled = true; |
| config.gain_controller1.mode = |
| AudioProcessing::Config::GainController1::kAdaptiveAnalog; |
| apm_->ApplyConfig(config); |
| |
| int out_analog_level = 0; |
| for (int i = 0; i < 2000; ++i) { |
| ReadFrameWithRewind(near_file_, &frame_); |
| // Ensure the audio is at a low level, so the AGC will try to increase it. |
| ScaleFrame(&frame_, 0.25); |
| |
| // Always pass in the same volume. |
| apm_->set_stream_analog_level(100); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| out_analog_level = apm_->recommended_stream_analog_level(); |
| } |
| |
| // Ensure the AGC is still able to reach the maximum. |
| EXPECT_EQ(255, out_analog_level); |
| } |
| |
| // Verifies that despite volume slider quantization, the AGC can continue to |
| // increase its volume. |
| TEST_F(ApmTest, QuantizedVolumeDoesNotGetStuck) { |
| for (size_t sample_rate_hz : kProcessSampleRates) { |
| SCOPED_TRACE(::testing::Message() << "sample_rate_hz=" << sample_rate_hz); |
| RunQuantizedVolumeDoesNotGetStuckTest(sample_rate_hz); |
| } |
| } |
| |
| void ApmTest::RunManualVolumeChangeIsPossibleTest(int sample_rate) { |
| Init(sample_rate, sample_rate, sample_rate, 2, 2, 2, false); |
| auto config = apm_->GetConfig(); |
| config.gain_controller1.enabled = true; |
| config.gain_controller1.mode = |
| AudioProcessing::Config::GainController1::kAdaptiveAnalog; |
| apm_->ApplyConfig(config); |
| |
| int out_analog_level = 100; |
| for (int i = 0; i < 1000; ++i) { |
| ReadFrameWithRewind(near_file_, &frame_); |
| // Ensure the audio is at a low level, so the AGC will try to increase it. |
| ScaleFrame(&frame_, 0.25); |
| |
| apm_->set_stream_analog_level(out_analog_level); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| out_analog_level = apm_->recommended_stream_analog_level(); |
| } |
| |
| // Ensure the volume was raised. |
| EXPECT_GT(out_analog_level, 100); |
| int highest_level_reached = out_analog_level; |
| // Simulate a user manual volume change. |
| out_analog_level = 100; |
| |
| for (int i = 0; i < 300; ++i) { |
| ReadFrameWithRewind(near_file_, &frame_); |
| ScaleFrame(&frame_, 0.25); |
| |
| apm_->set_stream_analog_level(out_analog_level); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| out_analog_level = apm_->recommended_stream_analog_level(); |
| // Check that AGC respected the manually adjusted volume. |
| EXPECT_LT(out_analog_level, highest_level_reached); |
| } |
| // Check that the volume was still raised. |
| EXPECT_GT(out_analog_level, 100); |
| } |
| |
| TEST_F(ApmTest, ManualVolumeChangeIsPossible) { |
| for (size_t sample_rate_hz : kProcessSampleRates) { |
| SCOPED_TRACE(::testing::Message() << "sample_rate_hz=" << sample_rate_hz); |
| RunManualVolumeChangeIsPossibleTest(sample_rate_hz); |
| } |
| } |
| |
| TEST_F(ApmTest, HighPassFilter) { |
| // Turn HP filter on/off |
| AudioProcessing::Config apm_config; |
| apm_config.high_pass_filter.enabled = true; |
| apm_->ApplyConfig(apm_config); |
| apm_config.high_pass_filter.enabled = false; |
| apm_->ApplyConfig(apm_config); |
| } |
| |
| TEST_F(ApmTest, AllProcessingDisabledByDefault) { |
| AudioProcessing::Config config = apm_->GetConfig(); |
| EXPECT_FALSE(config.echo_canceller.enabled); |
| EXPECT_FALSE(config.high_pass_filter.enabled); |
| EXPECT_FALSE(config.gain_controller1.enabled); |
| EXPECT_FALSE(config.noise_suppression.enabled); |
| } |
| |
| TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabledInt) { |
| // Test that ProcessStream simply copies input to output when all components |
| // are disabled. |
| // Runs over all processing rates, and some particularly common or special |
| // rates. |
| // - 8000 Hz: lowest sample rate seen in Chrome metrics, |
| // - 22050 Hz: APM input/output frames are not exactly 10 ms, |
| // - 44100 Hz: very common desktop sample rate. |
| constexpr int kSampleRatesHz[] = {8000, 16000, 22050, 32000, 44100, 48000}; |
| for (size_t sample_rate_hz : kSampleRatesHz) { |
| SCOPED_TRACE(::testing::Message() << "sample_rate_hz=" << sample_rate_hz); |
| Init(sample_rate_hz, sample_rate_hz, sample_rate_hz, 2, 2, 2, false); |
| SetFrameTo(&frame_, 1000, 2000); |
| Int16FrameData frame_copy; |
| frame_copy.CopyFrom(frame_); |
| for (int j = 0; j < 1000; j++) { |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy)); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessReverseStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy)); |
| } |
| } |
| } |
| |
| TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabledFloat) { |
| // Test that ProcessStream simply copies input to output when all components |
| // are disabled. |
| const size_t kSamples = 160; |
| const int sample_rate = 16000; |
| const float src[kSamples] = {-1.0f, 0.0f, 1.0f}; |
| float dest[kSamples] = {}; |
| |
| auto src_channels = &src[0]; |
| auto dest_channels = &dest[0]; |
| |
| apm_ = AudioProcessingBuilderForTesting().Create(); |
| EXPECT_NOERR(apm_->ProcessStream(&src_channels, StreamConfig(sample_rate, 1), |
| StreamConfig(sample_rate, 1), |
| &dest_channels)); |
| |
| for (size_t i = 0; i < kSamples; ++i) { |
| EXPECT_EQ(src[i], dest[i]); |
| } |
| |
| // Same for ProcessReverseStream. |
| float rev_dest[kSamples] = {}; |
| auto rev_dest_channels = &rev_dest[0]; |
| |
| StreamConfig input_stream = {sample_rate, 1}; |
| StreamConfig output_stream = {sample_rate, 1}; |
| EXPECT_NOERR(apm_->ProcessReverseStream(&src_channels, input_stream, |
| output_stream, &rev_dest_channels)); |
| |
| for (size_t i = 0; i < kSamples; ++i) { |
| EXPECT_EQ(src[i], rev_dest[i]); |
| } |
| } |
| |
| TEST_F(ApmTest, IdenticalInputChannelsResultInIdenticalOutputChannels) { |
| EnableAllComponents(); |
| |
| for (size_t i = 0; i < arraysize(kProcessSampleRates); i++) { |
| Init(kProcessSampleRates[i], kProcessSampleRates[i], kProcessSampleRates[i], |
| 2, 2, 2, false); |
| int analog_level = 127; |
| ASSERT_EQ(0, feof(far_file_)); |
| ASSERT_EQ(0, feof(near_file_)); |
| while (ReadFrame(far_file_, &revframe_) && ReadFrame(near_file_, &frame_)) { |
| CopyLeftToRightChannel(revframe_.data.data(), |
| revframe_.samples_per_channel); |
| |
| ASSERT_EQ( |
| kNoErr, |
| apm_->ProcessReverseStream( |
| revframe_.data.data(), |
| StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels), |
| StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels), |
| revframe_.data.data())); |
| |
| CopyLeftToRightChannel(frame_.data.data(), frame_.samples_per_channel); |
| |
| ASSERT_EQ(kNoErr, apm_->set_stream_delay_ms(0)); |
| apm_->set_stream_analog_level(analog_level); |
| ASSERT_EQ(kNoErr, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| analog_level = apm_->recommended_stream_analog_level(); |
| |
| VerifyChannelsAreEqual(frame_.data.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); |
| Int16FrameData frame_copy; |
| frame_copy.CopyFrom(frame_); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy)); |
| |
| // 2. Only the level estimator is enabled... |
| auto apm_config = apm_->GetConfig(); |
| SetFrameTo(&frame_, 1000); |
| frame_copy.CopyFrom(frame_); |
| apm_->ApplyConfig(apm_config); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy)); |
| apm_->ApplyConfig(apm_config); |
| |
| // Check the test is valid. We should have distortion from the filter |
| // when AEC is enabled (which won't affect the audio). |
| apm_config.echo_canceller.enabled = true; |
| apm_config.echo_canceller.mobile_mode = false; |
| apm_->ApplyConfig(apm_config); |
| frame_.samples_per_channel = 320; |
| frame_.num_channels = 2; |
| frame_.sample_rate_hz = 32000; |
| SetFrameTo(&frame_, 1000); |
| frame_copy.CopyFrom(frame_); |
| EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0)); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| EXPECT_FALSE(FrameDataAreEqual(frame_, frame_copy)); |
| } |
| |
| #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP |
| void ApmTest::ProcessDebugDump(absl::string_view in_filename, |
| absl::string_view out_filename, |
| Format format, |
| int max_size_bytes) { |
| TaskQueueForTest worker_queue("ApmTest_worker_queue"); |
| FILE* in_file = fopen(std::string(in_filename).c_str(), "rb"); |
| ASSERT_TRUE(in_file != NULL); |
| audioproc::Event event_msg; |
| bool first_init = true; |
| |
| while (ReadMessageFromFile(in_file, &event_msg)) { |
| if (event_msg.type() == audioproc::Event::INIT) { |
| const audioproc::Init msg = event_msg.init(); |
| int reverse_sample_rate = msg.sample_rate(); |
| if (msg.has_reverse_sample_rate()) { |
| reverse_sample_rate = msg.reverse_sample_rate(); |
| } |
| int output_sample_rate = msg.sample_rate(); |
| if (msg.has_output_sample_rate()) { |
| output_sample_rate = msg.output_sample_rate(); |
| } |
| |
| Init(msg.sample_rate(), output_sample_rate, reverse_sample_rate, |
| msg.num_input_channels(), msg.num_output_channels(), |
| msg.num_reverse_channels(), false); |
| if (first_init) { |
| // AttachAecDump() writes an additional init message. Don't start |
| // recording until after the first init to avoid the extra message. |
| auto aec_dump = |
| AecDumpFactory::Create(out_filename, max_size_bytes, &worker_queue); |
| EXPECT_TRUE(aec_dump); |
| apm_->AttachAecDump(std::move(aec_dump)); |
| first_init = false; |
| } |
| |
| } else if (event_msg.type() == audioproc::Event::REVERSE_STREAM) { |
| const audioproc::ReverseStream msg = event_msg.reverse_stream(); |
| |
| if (msg.channel_size() > 0) { |
| ASSERT_EQ(revframe_.num_channels, |
| static_cast<size_t>(msg.channel_size())); |
| for (int i = 0; i < msg.channel_size(); ++i) { |
| memcpy(revfloat_cb_->channels()[i], msg.channel(i).data(), |
| msg.channel(i).size()); |
| } |
| } else { |
| memcpy(revframe_.data.data(), msg.data().data(), msg.data().size()); |
| if (format == kFloatFormat) { |
| // We're using an int16 input file; convert to float. |
| ConvertToFloat(revframe_, revfloat_cb_.get()); |
| } |
| } |
| AnalyzeReverseStreamChooser(format); |
| |
| } else if (event_msg.type() == audioproc::Event::STREAM) { |
| const audioproc::Stream msg = event_msg.stream(); |
| // ProcessStream could have changed this for the output frame. |
| frame_.num_channels = apm_->num_input_channels(); |
| |
| apm_->set_stream_analog_level(msg.applied_input_volume()); |
| EXPECT_NOERR(apm_->set_stream_delay_ms(msg.delay())); |
| if (msg.has_keypress()) { |
| apm_->set_stream_key_pressed(msg.keypress()); |
| } else { |
| apm_->set_stream_key_pressed(true); |
| } |
| |
| if (msg.input_channel_size() > 0) { |
| ASSERT_EQ(frame_.num_channels, |
| static_cast<size_t>(msg.input_channel_size())); |
| for (int i = 0; i < msg.input_channel_size(); ++i) { |
| memcpy(float_cb_->channels()[i], msg.input_channel(i).data(), |
| msg.input_channel(i).size()); |
| } |
| } else { |
| memcpy(frame_.data.data(), msg.input_data().data(), |
| msg.input_data().size()); |
| if (format == kFloatFormat) { |
| // We're using an int16 input file; convert to float. |
| ConvertToFloat(frame_, float_cb_.get()); |
| } |
| } |
| ProcessStreamChooser(format); |
| } |
| } |
| apm_->DetachAecDump(); |
| fclose(in_file); |
| } |
| |
| void ApmTest::VerifyDebugDumpTest(Format format) { |
| rtc::ScopedFakeClock fake_clock; |
| const std::string in_filename = test::ResourcePath("ref03", "aecdump"); |
| std::string format_string; |
| switch (format) { |
| case kIntFormat: |
| format_string = "_int"; |
| break; |
| case kFloatFormat: |
| format_string = "_float"; |
| break; |
| } |
| const std::string ref_filename = test::TempFilename( |
| test::OutputPath(), std::string("ref") + format_string + "_aecdump"); |
| const std::string out_filename = test::TempFilename( |
| test::OutputPath(), std::string("out") + format_string + "_aecdump"); |
| const std::string limited_filename = test::TempFilename( |
| test::OutputPath(), std::string("limited") + format_string + "_aecdump"); |
| const size_t logging_limit_bytes = 100000; |
| // We expect at least this many bytes in the created logfile. |
| const size_t logging_expected_bytes = 95000; |
| EnableAllComponents(); |
| ProcessDebugDump(in_filename, ref_filename, format, -1); |
| ProcessDebugDump(ref_filename, out_filename, format, -1); |
| ProcessDebugDump(ref_filename, limited_filename, format, logging_limit_bytes); |
| |
| FILE* ref_file = fopen(ref_filename.c_str(), "rb"); |
| FILE* out_file = fopen(out_filename.c_str(), "rb"); |
| FILE* limited_file = fopen(limited_filename.c_str(), "rb"); |
| ASSERT_TRUE(ref_file != NULL); |
| ASSERT_TRUE(out_file != NULL); |
| ASSERT_TRUE(limited_file != NULL); |
| std::unique_ptr<uint8_t[]> ref_bytes; |
| std::unique_ptr<uint8_t[]> out_bytes; |
| std::unique_ptr<uint8_t[]> limited_bytes; |
| |
| size_t ref_size = ReadMessageBytesFromFile(ref_file, &ref_bytes); |
| size_t out_size = ReadMessageBytesFromFile(out_file, &out_bytes); |
| size_t limited_size = ReadMessageBytesFromFile(limited_file, &limited_bytes); |
| size_t bytes_read = 0; |
| size_t bytes_read_limited = 0; |
| while (ref_size > 0 && out_size > 0) { |
| bytes_read += ref_size; |
| bytes_read_limited += limited_size; |
| EXPECT_EQ(ref_size, out_size); |
| EXPECT_GE(ref_size, limited_size); |
| EXPECT_TRUE(ExpectMessageEq(/*actual=*/{out_bytes.get(), out_size}, |
| /*expected=*/{ref_bytes.get(), ref_size})); |
| if (limited_size > 0) { |
| EXPECT_TRUE( |
| ExpectMessageEq(/*actual=*/{limited_bytes.get(), limited_size}, |
| /*expected=*/{ref_bytes.get(), ref_size})); |
| } |
| ref_size = ReadMessageBytesFromFile(ref_file, &ref_bytes); |
| out_size = ReadMessageBytesFromFile(out_file, &out_bytes); |
| limited_size = ReadMessageBytesFromFile(limited_file, &limited_bytes); |
| } |
| EXPECT_GT(bytes_read, 0u); |
| EXPECT_GT(bytes_read_limited, logging_expected_bytes); |
| EXPECT_LE(bytes_read_limited, logging_limit_bytes); |
| EXPECT_NE(0, feof(ref_file)); |
| EXPECT_NE(0, feof(out_file)); |
| EXPECT_NE(0, feof(limited_file)); |
| ASSERT_EQ(0, fclose(ref_file)); |
| ASSERT_EQ(0, fclose(out_file)); |
| ASSERT_EQ(0, fclose(limited_file)); |
| remove(ref_filename.c_str()); |
| remove(out_filename.c_str()); |
| remove(limited_filename.c_str()); |
| } |
| |
| TEST_F(ApmTest, VerifyDebugDumpInt) { |
| VerifyDebugDumpTest(kIntFormat); |
| } |
| |
| TEST_F(ApmTest, VerifyDebugDumpFloat) { |
| VerifyDebugDumpTest(kFloatFormat); |
| } |
| #endif |
| |
| // TODO(andrew): expand test to verify output. |
| TEST_F(ApmTest, DebugDump) { |
| TaskQueueForTest worker_queue("ApmTest_worker_queue"); |
| const std::string filename = |
| test::TempFilename(test::OutputPath(), "debug_aec"); |
| { |
| auto aec_dump = AecDumpFactory::Create("", -1, &worker_queue); |
| EXPECT_FALSE(aec_dump); |
| } |
| |
| #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP |
| // Stopping without having started should be OK. |
| apm_->DetachAecDump(); |
| |
| auto aec_dump = AecDumpFactory::Create(filename, -1, &worker_queue); |
| EXPECT_TRUE(aec_dump); |
| apm_->AttachAecDump(std::move(aec_dump)); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessReverseStream( |
| revframe_.data.data(), |
| StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels), |
| StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels), |
| revframe_.data.data())); |
| apm_->DetachAecDump(); |
| |
| // Verify the file has been written. |
| FILE* fid = fopen(filename.c_str(), "r"); |
| ASSERT_TRUE(fid != NULL); |
| |
| // Clean it up. |
| ASSERT_EQ(0, fclose(fid)); |
| ASSERT_EQ(0, remove(filename.c_str())); |
| #else |
| // Verify the file has NOT been written. |
| ASSERT_TRUE(fopen(filename.c_str(), "r") == NULL); |
| #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP |
| } |
| |
| // TODO(andrew): expand test to verify output. |
| TEST_F(ApmTest, DebugDumpFromFileHandle) { |
| TaskQueueForTest worker_queue("ApmTest_worker_queue"); |
| |
| const std::string filename = |
| test::TempFilename(test::OutputPath(), "debug_aec"); |
| FileWrapper f = FileWrapper::OpenWriteOnly(filename); |
| ASSERT_TRUE(f.is_open()); |
| |
| #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP |
| // Stopping without having started should be OK. |
| apm_->DetachAecDump(); |
| |
| auto aec_dump = AecDumpFactory::Create(std::move(f), -1, &worker_queue); |
| EXPECT_TRUE(aec_dump); |
| apm_->AttachAecDump(std::move(aec_dump)); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessReverseStream( |
| revframe_.data.data(), |
| StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels), |
| StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels), |
| revframe_.data.data())); |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| apm_->DetachAecDump(); |
| |
| // Verify the file has been written. |
| FILE* fid = fopen(filename.c_str(), "r"); |
| ASSERT_TRUE(fid != NULL); |
| |
| // Clean it up. |
| ASSERT_EQ(0, fclose(fid)); |
| ASSERT_EQ(0, remove(filename.c_str())); |
| #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP |
| } |
| |
| // TODO(andrew): Add a test to process a few frames with different combinations |
| // of enabled components. |
| |
| TEST_F(ApmTest, Process) { |
| GOOGLE_PROTOBUF_VERIFY_VERSION; |
| audioproc::OutputData ref_data; |
| |
| if (!absl::GetFlag(FLAGS_write_apm_ref_data)) { |
| OpenFileAndReadMessage(ref_filename_, &ref_data); |
| } else { |
| const int kChannels[] = {1, 2}; |
| // Write the desired tests to the protobuf reference file. |
| for (size_t i = 0; i < arraysize(kChannels); i++) { |
| for (size_t j = 0; j < arraysize(kChannels); j++) { |
| for (int sample_rate_hz : AudioProcessing::kNativeSampleRatesHz) { |
| 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(sample_rate_hz); |
| 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; |
| |
| apm_ = AudioProcessingBuilderForTesting() |
| .SetEchoDetector(CreateEchoDetector()) |
| .Create(); |
| AudioProcessing::Config apm_config = apm_->GetConfig(); |
| apm_config.gain_controller1.analog_gain_controller.enabled = false; |
| apm_->ApplyConfig(apm_config); |
| |
| EnableAllComponents(); |
| |
| Init(test->sample_rate(), test->sample_rate(), test->sample_rate(), |
| static_cast<size_t>(test->num_input_channels()), |
| static_cast<size_t>(test->num_output_channels()), |
| static_cast<size_t>(test->num_reverse_channels()), true); |
| |
| int frame_count = 0; |
| int analog_level = 127; |
| int analog_level_average = 0; |
| int max_output_average = 0; |
| #if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) |
| int stats_index = 0; |
| #endif |
| |
| while (ReadFrame(far_file_, &revframe_) && ReadFrame(near_file_, &frame_)) { |
| EXPECT_EQ( |
| apm_->kNoError, |
| apm_->ProcessReverseStream( |
| revframe_.data.data(), |
| StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels), |
| StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels), |
| revframe_.data.data())); |
| |
| EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0)); |
| apm_->set_stream_analog_level(analog_level); |
| |
| EXPECT_EQ(apm_->kNoError, |
| apm_->ProcessStream( |
| frame_.data.data(), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| StreamConfig(frame_.sample_rate_hz, frame_.num_channels), |
| frame_.data.data())); |
| |
| // Ensure the frame was downmixed properly. |
| EXPECT_EQ(static_cast<size_t>(test->num_output_channels()), |
| frame_.num_channels); |
| |
| max_output_average += MaxAudioFrame(frame_); |
| |
| analog_level = apm_->recommended_stream_analog_level(); |
| analog_level_average += analog_level; |
| AudioProcessingStats stats = apm_->GetStatistics(); |
| |
| size_t frame_size = frame_.samples_per_channel * frame_.num_channels; |
| size_t write_count = |
| fwrite(frame_.data.data(), sizeof(int16_t), frame_size, out_file_); |
| ASSERT_EQ(frame_size, write_count); |
| |
| // Reset in case of downmixing. |
| frame_.num_channels = static_cast<size_t>(test->num_input_channels()); |
| frame_count++; |
| |
| #if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) |
| const int kStatsAggregationFrameNum = 100; // 1 second. |
| if (frame_count % kStatsAggregationFrameNum == 0) { |
| // Get echo and delay metrics. |
| AudioProcessingStats stats2 = apm_->GetStatistics(); |
| |
| // Echo metrics. |
| const float echo_return_loss = stats2.echo_return_loss.value_or(-1.0f); |
| const float echo_return_loss_enhancement = |
| stats2.echo_return_loss_enhancement.value_or(-1.0f); |
| const float residual_echo_likelihood = |
| stats2.residual_echo_likelihood.value_or(-1.0f); |
| const float residual_echo_likelihood_recent_max = |
| stats2.residual_echo_likelihood_recent_max.value_or(-1.0f); |
| |
| if (!absl::GetFlag(FLAGS_write_apm_ref_data)) { |
| const audioproc::Test::EchoMetrics& reference = |
| test->echo_metrics(stats_index); |
| constexpr float kEpsilon = 0.01; |
| EXPECT_NEAR(echo_return_loss, reference.echo_return_loss(), kEpsilon); |
| EXPECT_NEAR(echo_return_loss_enhancement, |
| reference.echo_return_loss_enhancement(), kEpsilon); |
| EXPECT_NEAR(residual_echo_likelihood, |
| reference.residual_echo_likelihood(), kEpsilon); |
| EXPECT_NEAR(residual_echo_likelihood_recent_max, |
| reference.residual_echo_likelihood_recent_max(), |
| kEpsilon); |
| ++stats_index; |
| } else { |
| audioproc::Test::EchoMetrics* message_echo = test->add_echo_metrics(); |
| message_echo->set_echo_return_loss(echo_return_loss); |
| message_echo->set_echo_return_loss_enhancement( |
| echo_return_loss_enhancement); |
| message_echo->set_residual_echo_likelihood(residual_echo_likelihood); |
| message_echo->set_residual_echo_likelihood_recent_max( |
| residual_echo_likelihood_recent_max); |
| } |
| } |
| #endif // defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE). |
| } |
| max_output_average /= frame_count; |
| analog_level_average /= frame_count; |
| |
| if (!absl::GetFlag(FLAGS_write_apm_ref_data)) { |
| const int kIntNear = 1; |
| // All numbers being consistently higher on N7 compare to the reference |
| // data. |
| // TODO(bjornv): If we start getting more of these offsets on Android we |
| // should consider a different approach. Either using one slack for all, |
| // or generate a separate android reference. |
| #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) |
| const int kMaxOutputAverageOffset = 9; |
| const int kMaxOutputAverageNear = 26; |
| #else |
| const int kMaxOutputAverageOffset = 0; |
| const int kMaxOutputAverageNear = kIntNear; |
| #endif |
| EXPECT_NEAR(test->analog_level_average(), analog_level_average, kIntNear); |
| EXPECT_NEAR(test->max_output_average(), |
| max_output_average - kMaxOutputAverageOffset, |
| kMaxOutputAverageNear); |
| } else { |
| test->set_analog_level_average(analog_level_average); |
| test->set_max_output_average(max_output_average); |
| } |
| |
| rewind(far_file_); |
| rewind(near_file_); |
| } |
| |
| if (absl::GetFlag(FLAGS_write_apm_ref_data)) { |
| OpenFileAndWriteMessage(ref_filename_, ref_data); |
| } |
| } |
| |
| // Compares the reference and test arrays over a region around the expected |
| // delay. Finds the highest SNR in that region and adds the variance and squared |
| // error results to the supplied accumulators. |
| void UpdateBestSNR(const float* ref, |
| const float* test, |
| size_t length, |
| int expected_delay, |
| double* variance_acc, |
| double* sq_error_acc) { |
| RTC_CHECK_LT(expected_delay, length) |
| << "delay greater than signal length, cannot compute SNR"; |
| double best_snr = std::numeric_limits<double>::min(); |
| double best_variance = 0; |
| double best_sq_error = 0; |
| // Search over a region of nine samples around the expected delay. |
| for (int delay = std::max(expected_delay - 4, 0); delay <= expected_delay + 4; |
| ++delay) { |
| double sq_error = 0; |
| double variance = 0; |
| for (size_t i = 0; i < length - delay; ++i) { |
| double error = test[i + delay] - ref[i]; |
| sq_error += error * error; |
| variance += ref[i] * ref[i]; |
| } |
| |
| if (sq_error == 0) { |
| *variance_acc += variance; |
| return; |
| } |
| double snr = variance / sq_error; |
| if (snr > best_snr) { |
| best_snr = snr; |
| best_variance = variance; |
| best_sq_error = sq_error; |
| } |
| } |
| |
| *variance_acc += best_variance; |
| *sq_error_acc += best_sq_error; |
| } |
| |
| // Used to test a multitude of sample rate and channel combinations. It works |
| // by first producing a set of reference files (in SetUpTestCase) that are |
| // assumed to be correct, as the used parameters are verified by other tests |
| // in this collection. Primarily the reference files are all produced at |
| // "native" rates which do not involve any resampling. |
| |
| // Each test pass produces an output file with a particular format. The output |
| // is matched against the reference file closest to its internal processing |
| // format. If necessary the output is resampled back to its process format. |
| // Due to the resampling distortion, we don't expect identical results, but |
| // enforce SNR thresholds which vary depending on the format. 0 is a special |
| // case SNR which corresponds to inf, or zero error. |
| typedef std::tuple<int, int, int, int, double, double> AudioProcessingTestData; |
| class AudioProcessingTest |
| : public ::testing::TestWithParam<AudioProcessingTestData> { |
| public: |
| AudioProcessingTest() |
| : input_rate_(std::get<0>(GetParam())), |
| output_rate_(std::get<1>(GetParam())), |
| reverse_input_rate_(std::get<2>(GetParam())), |
| reverse_output_rate_(std::get<3>(GetParam())), |
| expected_snr_(std::get<4>(GetParam())), |
| expected_reverse_snr_(std::get<5>(GetParam())) {} |
| |
| virtual ~AudioProcessingTest() {} |
| |
| static void SetUpTestSuite() { |
| // Create all needed output reference files. |
| const size_t kNumChannels[] = {1, 2}; |
| for (size_t i = 0; i < arraysize(kProcessSampleRates); ++i) { |
| for (size_t j = 0; j < arraysize(kNumChannels); ++j) { |
| for (size_t k = 0; k < arraysize(kNumChannels); ++k) { |
| // The reference files always have matching input and output channels. |
| ProcessFormat(kProcessSampleRates[i], kProcessSampleRates[i], |
| kProcessSampleRates[i], kProcessSampleRates[i], |
| kNumChannels[j], kNumChannels[j], kNumChannels[k], |
| kNumChannels[k], "ref"); |
| } |
| } |
| } |
| } |
| |
| void TearDown() { |
| // Remove "out" files after each test. |
| ClearTempOutFiles(); |
| } |
| |
| static void TearDownTestSuite() { ClearTempFiles(); } |
| |
| // Runs a process pass on files with the given parameters and dumps the output |
| // to a file specified with `output_file_prefix`. Both forward and reverse |
| // output streams are dumped. |
| static void ProcessFormat(int input_rate, |
| int output_rate, |
| int reverse_input_rate, |
| int reverse_output_rate, |
| size_t num_input_channels, |
| size_t num_output_channels, |
| size_t num_reverse_input_channels, |
| size_t num_reverse_output_channels, |
| absl::string_view output_file_prefix) { |
| AudioProcessing::Config apm_config; |
| apm_config.gain_controller1.analog_gain_controller.enabled = false; |
| rtc::scoped_refptr<AudioProcessing> ap = |
| AudioProcessingBuilderForTesting().SetConfig(apm_config).Create(); |
| |
| EnableAllAPComponents(ap.get()); |
| |
| ProcessingConfig processing_config = { |
| {{input_rate, num_input_channels}, |
| {output_rate, num_output_channels}, |
| {reverse_input_rate, num_reverse_input_channels}, |
| {reverse_output_rate, num_reverse_output_channels}}}; |
| ap->Initialize(processing_config); |
| |
| FILE* far_file = |
| fopen(ResourceFilePath("far", reverse_input_rate).c_str(), "rb"); |
| FILE* near_file = fopen(ResourceFilePath("near", input_rate).c_str(), "rb"); |
| FILE* out_file = fopen( |
| OutputFilePath( |
| output_file_prefix, input_rate, output_rate, reverse_input_rate, |
| reverse_output_rate, num_input_channels, num_output_channels, |
| num_reverse_input_channels, num_reverse_output_channels, kForward) |
| .c_str(), |
| "wb"); |
| FILE* rev_out_file = fopen( |
| OutputFilePath( |
| output_file_prefix, input_rate, output_rate, reverse_input_rate, |
| reverse_output_rate, num_input_channels, num_output_channels, |
| num_reverse_input_channels, num_reverse_output_channels, kReverse) |
| .c_str(), |
| "wb"); |
| ASSERT_TRUE(far_file != NULL); |
| ASSERT_TRUE(near_file != NULL); |
| ASSERT_TRUE(out_file != NULL); |
| ASSERT_TRUE(rev_out_file != NULL); |
| |
| ChannelBuffer<float> fwd_cb(AudioProcessing::GetFrameSize(input_rate), |
| num_input_channels); |
| ChannelBuffer<float> rev_cb( |
| AudioProcessing::GetFrameSize(reverse_input_rate), |
| num_reverse_input_channels); |
| ChannelBuffer<float> out_cb(AudioProcessing::GetFrameSize(output_rate), |
| num_output_channels); |
| ChannelBuffer<float> rev_out_cb( |
| AudioProcessing::GetFrameSize(reverse_output_rate), |
| num_reverse_output_channels); |
| |
| // Temporary buffers. |
| const int max_length = |
| 2 * std::max(std::max(out_cb.num_frames(), rev_out_cb.num_frames()), |
| std::max(fwd_cb.num_frames(), rev_cb.num_frames())); |
| std::unique_ptr<float[]> float_data(new float[max_length]); |
| std::unique_ptr<int16_t[]> int_data(new int16_t[max_length]); |
| |
| int analog_level = 127; |
| while (ReadChunk(far_file, int_data.get(), float_data.get(), &rev_cb) && |
| ReadChunk(near_file, int_data.get(), float_data.get(), &fwd_cb)) { |
| EXPECT_NOERR(ap->ProcessReverseStream( |
| rev_cb.channels(), processing_config.reverse_input_stream(), |
| processing_config.reverse_output_stream(), rev_out_cb.channels())); |
| |
| EXPECT_NOERR(ap->set_stream_delay_ms(0)); |
| ap->set_stream_analog_level(analog_level); |
| |
| EXPECT_NOERR(ap->ProcessStream( |
| fwd_cb.channels(), StreamConfig(input_rate, num_input_channels), |
| StreamConfig(output_rate, num_output_channels), out_cb.channels())); |
| |
| // Dump forward output to file. |
| Interleave(out_cb.channels(), out_cb.num_frames(), out_cb.num_channels(), |
| float_data.get()); |
| size_t out_length = out_cb.num_channels() * out_cb.num_frames(); |
| |
| ASSERT_EQ(out_length, fwrite(float_data.get(), sizeof(float_data[0]), |
| out_length, out_file)); |
| |
| // Dump reverse output to file. |
| Interleave(rev_out_cb.channels(), rev_out_cb.num_frames(), |
| rev_out_cb.num_channels(), float_data.get()); |
| size_t rev_out_length = |
| rev_out_cb.num_channels() * rev_out_cb.num_frames(); |
| |
| ASSERT_EQ(rev_out_length, fwrite(float_data.get(), sizeof(float_data[0]), |
| rev_out_length, rev_out_file)); |
| |
| analog_level = ap->recommended_stream_analog_level(); |
| } |
| fclose(far_file); |
| fclose(near_file); |
| fclose(out_file); |
| fclose(rev_out_file); |
| } |
| |
| protected: |
| int input_rate_; |
| int output_rate_; |
| int reverse_input_rate_; |
| int reverse_output_rate_; |
| double expected_snr_; |
| double expected_reverse_snr_; |
| }; |
| |
| TEST_P(AudioProcessingTest, Formats) { |
| struct ChannelFormat { |
| int num_input; |
| int num_output; |
| int num_reverse_input; |
| int num_reverse_output; |
| }; |
| ChannelFormat cf[] = { |
| {1, 1, 1, 1}, {1, 1, 2, 1}, {2, 1, 1, 1}, |
| {2, 1, 2, 1}, {2, 2, 1, 1}, {2, 2, 2, 2}, |
| }; |
| |
| for (size_t i = 0; i < arraysize(cf); ++i) { |
| ProcessFormat(input_rate_, output_rate_, reverse_input_rate_, |
| reverse_output_rate_, cf[i].num_input, cf[i].num_output, |
| cf[i].num_reverse_input, cf[i].num_reverse_output, "out"); |
| |
| // Verify output for both directions. |
| std::vector<StreamDirection> stream_directions; |
| stream_directions.push_back(kForward); |
| stream_directions.push_back(kReverse); |
| for (StreamDirection file_direction : stream_directions) { |
| const int in_rate = file_direction ? reverse_input_rate_ : input_rate_; |
| const int out_rate = file_direction ? reverse_output_rate_ : output_rate_; |
| const int out_num = |
| file_direction ? cf[i].num_reverse_output : cf[i].num_output; |
| const double expected_snr = |
| file_direction ? expected_reverse_snr_ : expected_snr_; |
| |
| const int min_ref_rate = std::min(in_rate, out_rate); |
| int ref_rate; |
| if (min_ref_rate > 32000) { |
| ref_rate = 48000; |
| } else if (min_ref_rate > 16000) { |
| ref_rate = 32000; |
| } else { |
| ref_rate = 16000; |
| } |
| |
| FILE* out_file = fopen( |
| OutputFilePath("out", input_rate_, output_rate_, reverse_input_rate_, |
| reverse_output_rate_, cf[i].num_input, |
| cf[i].num_output, cf[i].num_reverse_input, |
| cf[i].num_reverse_output, file_direction) |
| .c_str(), |
| "rb"); |
| // The reference files always have matching input and output channels. |
| FILE* ref_file = |
| fopen(OutputFilePath("ref", ref_rate, ref_rate, ref_rate, ref_rate, |
| cf[i].num_output, cf[i].num_output, |
| cf[i].num_reverse_output, |
| cf[i].num_reverse_output, file_direction) |
| .c_str(), |
| "rb"); |
| ASSERT_TRUE(out_file != NULL); |
| ASSERT_TRUE(ref_file != NULL); |
| |
| const size_t ref_length = |
| AudioProcessing::GetFrameSize(ref_rate) * out_num; |
| const size_t out_length = |
| AudioProcessing::GetFrameSize(out_rate) * out_num; |
| // Data from the reference file. |
| std::unique_ptr<float[]> ref_data(new float[ref_length]); |
| // Data from the output file. |
| std::unique_ptr<float[]> out_data(new float[out_length]); |
| // Data from the resampled output, in case the reference and output rates |
| // don't match. |
| std::unique_ptr<float[]> cmp_data(new float[ref_length]); |
| |
| PushResampler<float> resampler; |
| resampler.InitializeIfNeeded(out_rate, ref_rate, out_num); |
| |
| // Compute the resampling delay of the output relative to the reference, |
| // to find the region over which we should search for the best SNR. |
| float expected_delay_sec = 0; |
| if (in_rate != ref_rate) { |
| // Input resampling delay. |
| expected_delay_sec += |
| PushSincResampler::AlgorithmicDelaySeconds(in_rate); |
| } |
| if (out_rate != ref_rate) { |
| // Output resampling delay. |
| expected_delay_sec += |
| PushSincResampler::AlgorithmicDelaySeconds(ref_rate); |
| // Delay of converting the output back to its processing rate for |
| // testing. |
| expected_delay_sec += |
| PushSincResampler::AlgorithmicDelaySeconds(out_rate); |
| } |
| // The delay is multiplied by the number of channels because |
| // UpdateBestSNR() computes the SNR over interleaved data without taking |
| // channels into account. |
| int expected_delay = |
| std::floor(expected_delay_sec * ref_rate + 0.5f) * out_num; |
| |
| double variance = 0; |
| double sq_error = 0; |
| while (fread(out_data.get(), sizeof(out_data[0]), out_length, out_file) && |
| fread(ref_data.get(), sizeof(ref_data[0]), ref_length, ref_file)) { |
| float* out_ptr = out_data.get(); |
| if (out_rate != ref_rate) { |
| // Resample the output back to its internal processing rate if |
| // necessary. |
| ASSERT_EQ(ref_length, |
| static_cast<size_t>(resampler.Resample( |
| out_ptr, out_length, cmp_data.get(), ref_length))); |
| out_ptr = cmp_data.get(); |
| } |
| |
| // Update the `sq_error` and `variance` accumulators with the highest |
| // SNR of reference vs output. |
| UpdateBestSNR(ref_data.get(), out_ptr, ref_length, expected_delay, |
| &variance, &sq_error); |
| } |
| |
| std::cout << "(" << input_rate_ << ", " << output_rate_ << ", " |
| << reverse_input_rate_ << ", " << reverse_output_rate_ << ", " |
| << cf[i].num_input << ", " << cf[i].num_output << ", " |
| << cf[i].num_reverse_input << ", " << cf[i].num_reverse_output |
| << ", " << file_direction << "): "; |
| if (sq_error > 0) { |
| double snr = 10 * log10(variance / sq_error); |
| EXPECT_GE(snr, expected_snr); |
| EXPECT_NE(0, expected_snr); |
| std::cout << "SNR=" << snr << " dB" << std::endl; |
| } else { |
| std::cout << "SNR=inf dB" << std::endl; |
| } |
| |
| fclose(out_file); |
| fclose(ref_file); |
| } |
| } |
| } |
| |
| #if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) |
| INSTANTIATE_TEST_SUITE_P( |
| CommonFormats, |
| AudioProcessingTest, |
| // Internal processing rates and the particularly common sample rate 44100 |
| // Hz are tested in a grid of combinations (capture in, render in, out). |
| ::testing::Values(std::make_tuple(48000, 48000, 48000, 48000, 0, 0), |
| std::make_tuple(48000, 48000, 32000, 48000, 40, 30), |
| std::make_tuple(48000, 48000, 16000, 48000, 40, 20), |
| std::make_tuple(48000, 44100, 48000, 44100, 20, 20), |
| std::make_tuple(48000, 44100, 32000, 44100, 20, 15), |
| std::make_tuple(48000, 44100, 16000, 44100, 20, 15), |
| std::make_tuple(48000, 32000, 48000, 32000, 30, 35), |
| std::make_tuple(48000, 32000, 32000, 32000, 30, 0), |
| std::make_tuple(48000, 32000, 16000, 32000, 30, 20), |
| std::make_tuple(48000, 16000, 48000, 16000, 25, 20), |
| std::make_tuple(48000, 16000, 32000, 16000, 25, 20), |
| std::make_tuple(48000, 16000, 16000, 16000, 25, 0), |
| |
| std::make_tuple(44100, 48000, 48000, 48000, 30, 0), |
| std::make_tuple(44100, 48000, 32000, 48000, 30, 30), |
| std::make_tuple(44100, 48000, 16000, 48000, 30, 20), |
| std::make_tuple(44100, 44100, 48000, 44100, 20, 20), |
| std::make_tuple(44100, 44100, 32000, 44100, 20, 15), |
| std::make_tuple(44100, 44100, 16000, 44100, 20, 15), |
| std::make_tuple(44100, 32000, 48000, 32000, 30, 35), |
| std::make_tuple(44100, 32000, 32000, 32000, 30, 0), |
| std::make_tuple(44100, 32000, 16000, 32000, 30, 20), |
| std::make_tuple(44100, 16000, 48000, 16000, 25, 20), |
| std::make_tuple(44100, 16000, 32000, 16000, 25, 20), |
| std::make_tuple(44100, 16000, 16000, 16000, 25, 0), |
| |
| std::make_tuple(32000, 48000, 48000, 48000, 15, 0), |
| std::make_tuple(32000, 48000, 32000, 48000, 15, 30), |
| std::make_tuple(32000, 48000, 16000, 48000, 15, 20), |
| std::make_tuple(32000, 44100, 48000, 44100, 19, 20), |
| std::make_tuple(32000, 44100, 32000, 44100, 19, 15), |
| std::make_tuple(32000, 44100, 16000, 44100, 19, 15), |
| std::make_tuple(32000, 32000, 48000, 32000, 40, 35), |
| std::make_tuple(32000, 32000, 32000, 32000, 0, 0), |
| std::make_tuple(32000, 32000, 16000, 32000, 39, 20), |
| std::make_tuple(32000, 16000, 48000, 16000, 25, 20), |
| std::make_tuple(32000, 16000, 32000, 16000, 25, 20), |
| std::make_tuple(32000, 16000, 16000, 16000, 25, 0), |
| |
| std::make_tuple(16000, 48000, 48000, 48000, 9, 0), |
| std::make_tuple(16000, 48000, 32000, 48000, 9, 30), |
| std::make_tuple(16000, 48000, 16000, 48000, 9, 20), |
| std::make_tuple(16000, 44100, 48000, 44100, 15, 20), |
| std::make_tuple(16000, 44100, 32000, 44100, 15, 15), |
| std::make_tuple(16000, 44100, 16000, 44100, 15, 15), |
| std::make_tuple(16000, 32000, 48000, 32000, 25, 35), |
| std::make_tuple(16000, 32000, 32000, 32000, 25, 0), |
| std::make_tuple(16000, 32000, 16000, 32000, 25, 20), |
| std::make_tuple(16000, 16000, 48000, 16000, 39, 20), |
| std::make_tuple(16000, 16000, 32000, 16000, 39, 20), |
| std::make_tuple(16000, 16000, 16000, 16000, 0, 0), |
| |
| // Other sample rates are not tested exhaustively, to keep |
| // the test runtime manageable. |
| // |
| // Testing most other sample rates logged by Chrome UMA: |
| // - WebRTC.AudioInputSampleRate |
| // - WebRTC.AudioOutputSampleRate |
| // ApmConfiguration.HandlingOfRateCombinations covers |
| // remaining sample rates. |
| std::make_tuple(192000, 192000, 48000, 192000, 20, 40), |
| std::make_tuple(176400, 176400, 48000, 176400, 20, 35), |
| std::make_tuple(96000, 96000, 48000, 96000, 20, 40), |
| std::make_tuple(88200, 88200, 48000, 88200, 20, 20), |
| std::make_tuple(44100, 44100, 48000, 44100, 20, 20))); |
| |
| #elif defined(WEBRTC_AUDIOPROC_FIXED_PROFILE) |
| INSTANTIATE_TEST_SUITE_P( |
| CommonFormats, |
| AudioProcessingTest, |
| ::testing::Values(std::make_tuple(48000, 48000, 48000, 48000, 19, 0), |
| std::make_tuple(48000, 48000, 32000, 48000, 19, 30), |
| std::make_tuple(48000, 48000, 16000, 48000, 19, 20), |
| std::make_tuple(48000, 44100, 48000, 44100, 15, 20), |
| std::make_tuple(48000, 44100, 32000, 44100, 15, 15), |
| std::make_tuple(48000, 44100, 16000, 44100, 15, 15), |
| std::make_tuple(48000, 32000, 48000, 32000, 19, 35), |
| std::make_tuple(48000, 32000, 32000, 32000, 19, 0), |
| std::make_tuple(48000, 32000, 16000, 32000, 19, 20), |
| std::make_tuple(48000, 16000, 48000, 16000, 20, 20), |
| std::make_tuple(48000, 16000, 32000, 16000, 20, 20), |
| std::make_tuple(48000, 16000, 16000, 16000, 20, 0), |
| |
| std::make_tuple(44100, 48000, 48000, 48000, 15, 0), |
| std::make_tuple(44100, 48000, 32000, 48000, 15, 30), |
| std::make_tuple(44100, 48000, 16000, 48000, 15, 20), |
| std::make_tuple(44100, 44100, 48000, 44100, 15, 20), |
| std::make_tuple(44100, 44100, 32000, 44100, 15, 15), |
| std::make_tuple(44100, 44100, 16000, 44100, 15, 15), |
| std::make_tuple(44100, 32000, 48000, 32000, 18, 35), |
| std::make_tuple(44100, 32000, 32000, 32000, 18, 0), |
| std::make_tuple(44100, 32000, 16000, 32000, 18, 20), |
| std::make_tuple(44100, 16000, 48000, 16000, 19, 20), |
| std::make_tuple(44100, 16000, 32000, 16000, 19, 20), |
| std::make_tuple(44100, 16000, 16000, 16000, 19, 0), |
| |
| std::make_tuple(32000, 48000, 48000, 48000, 17, 0), |
| std::make_tuple(32000, 48000, 32000, 48000, 17, 30), |
| std::make_tuple(32000, 48000, 16000, 48000, 17, 20), |
| std::make_tuple(32000, 44100, 48000, 44100, 20, 20), |
| std::make_tuple(32000, 44100, 32000, 44100, 20, 15), |
| std::make_tuple(32000, 44100, 16000, 44100, 20, 15), |
| std::make_tuple(32000, 32000, 48000, 32000, 27, 35), |
| std::make_tuple(32000, 32000, 32000, 32000, 0, 0), |
| std::make_tuple(32000, 32000, 16000, 32000, 30, 20), |
| std::make_tuple(32000, 16000, 48000, 16000, 20, 20), |
| std::make_tuple(32000, 16000, 32000, 16000, 20, 20), |
| std::make_tuple(32000, 16000, 16000, 16000, 20, 0), |
| |
| std::make_tuple(16000, 48000, 48000, 48000, 11, 0), |
| std::make_tuple(16000, 48000, 32000, 48000, 11, 30), |
| std::make_tuple(16000, 48000, 16000, 48000, 11, 20), |
| std::make_tuple(16000, 44100, 48000, 44100, 15, 20), |
| std::make_tuple(16000, 44100, 32000, 44100, 15, 15), |
| std::make_tuple(16000, 44100, 16000, 44100, 15, 15), |
| std::make_tuple(16000, 32000, 48000, 32000, 24, 35), |
| std::make_tuple(16000, 32000, 32000, 32000, 24, 0), |
| std::make_tuple(16000, 32000, 16000, 32000, 25, 20), |
| std::make_tuple(16000, 16000, 48000, 16000, 28, 20), |
| std::make_tuple(16000, 16000, 32000, 16000, 28, 20), |
| std::make_tuple(16000, 16000, 16000, 16000, 0, 0), |
| |
| std::make_tuple(192000, 192000, 48000, 192000, 20, 40), |
| std::make_tuple(176400, 176400, 48000, 176400, 20, 35), |
| std::make_tuple(96000, 96000, 48000, 96000, 20, 40), |
| std::make_tuple(88200, 88200, 48000, 88200, 20, 20), |
| std::make_tuple(44100, 44100, 48000, 44100, 20, 20))); |
| #endif |
| |
| // Produces a scoped trace debug output. |
| std::string ProduceDebugText(int render_input_sample_rate_hz, |
| int render_output_sample_rate_hz, |
| int capture_input_sample_rate_hz, |
| int capture_output_sample_rate_hz, |
| size_t render_input_num_channels, |
| size_t render_output_num_channels, |
| size_t capture_input_num_channels, |
| size_t capture_output_num_channels) { |
| rtc::StringBuilder ss; |
| ss << "Sample rates:" |
| "\n Render input: " |
| << render_input_sample_rate_hz |
| << " Hz" |
| "\n Render output: " |
| << render_output_sample_rate_hz |
| << " Hz" |
| "\n Capture input: " |
| << capture_input_sample_rate_hz |
| << " Hz" |
| "\n Capture output: " |
| << capture_output_sample_rate_hz |
| << " Hz" |
| "\nNumber of channels:" |
| "\n Render input: " |
| << render_input_num_channels |
| << "\n Render output: " << render_output_num_channels |
| << "\n Capture input: " << capture_input_num_channels |
| << "\n Capture output: " << capture_output_num_channels; |
| return ss.Release(); |
| } |
| |
| // Validates that running the audio processing module using various combinations |
| // of sample rates and number of channels works as intended. |
| void RunApmRateAndChannelTest( |
| rtc::ArrayView<const int> sample_rates_hz, |
| rtc::ArrayView<const int> render_channel_counts, |
| rtc::ArrayView<const int> capture_channel_counts) { |
| webrtc::AudioProcessing::Config apm_config; |
| apm_config.pipeline.multi_channel_render = true; |
| apm_config.pipeline.multi_channel_capture = true; |
| apm_config.echo_canceller.enabled = true; |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting().SetConfig(apm_config).Create(); |
| |
| StreamConfig render_input_stream_config; |
| StreamConfig render_output_stream_config; |
| StreamConfig capture_input_stream_config; |
| StreamConfig capture_output_stream_config; |
| |
| std::vector<float> render_input_frame_channels; |
| std::vector<float*> render_input_frame; |
| std::vector<float> render_output_frame_channels; |
| std::vector<float*> render_output_frame; |
| std::vector<float> capture_input_frame_channels; |
| std::vector<float*> capture_input_frame; |
| std::vector<float> capture_output_frame_channels; |
| std::vector<float*> capture_output_frame; |
| |
| for (auto render_input_sample_rate_hz : sample_rates_hz) { |
| for (auto render_output_sample_rate_hz : sample_rates_hz) { |
| for (auto capture_input_sample_rate_hz : sample_rates_hz) { |
| for (auto capture_output_sample_rate_hz : sample_rates_hz) { |
| for (size_t render_input_num_channels : render_channel_counts) { |
| for (size_t capture_input_num_channels : capture_channel_counts) { |
| size_t render_output_num_channels = render_input_num_channels; |
| size_t capture_output_num_channels = capture_input_num_channels; |
| auto populate_audio_frame = [](int sample_rate_hz, |
| size_t num_channels, |
| StreamConfig* cfg, |
| std::vector<float>* channels_data, |
| std::vector<float*>* frame_data) { |
| cfg->set_sample_rate_hz(sample_rate_hz); |
| cfg->set_num_channels(num_channels); |
| |
| size_t max_frame_size = |
| AudioProcessing::GetFrameSize(sample_rate_hz); |
| channels_data->resize(num_channels * max_frame_size); |
| std::fill(channels_data->begin(), channels_data->end(), 0.5f); |
| frame_data->resize(num_channels); |
| for (size_t channel = 0; channel < num_channels; ++channel) { |
| (*frame_data)[channel] = |
| &(*channels_data)[channel * max_frame_size]; |
| } |
| }; |
| |
| populate_audio_frame( |
| render_input_sample_rate_hz, render_input_num_channels, |
| &render_input_stream_config, &render_input_frame_channels, |
| &render_input_frame); |
| populate_audio_frame( |
| render_output_sample_rate_hz, render_output_num_channels, |
| &render_output_stream_config, &render_output_frame_channels, |
| &render_output_frame); |
| populate_audio_frame( |
| capture_input_sample_rate_hz, capture_input_num_channels, |
| &capture_input_stream_config, &capture_input_frame_channels, |
| &capture_input_frame); |
| populate_audio_frame( |
| capture_output_sample_rate_hz, capture_output_num_channels, |
| &capture_output_stream_config, &capture_output_frame_channels, |
| &capture_output_frame); |
| |
| for (size_t frame = 0; frame < 2; ++frame) { |
| SCOPED_TRACE(ProduceDebugText( |
| render_input_sample_rate_hz, render_output_sample_rate_hz, |
| capture_input_sample_rate_hz, capture_output_sample_rate_hz, |
| render_input_num_channels, render_output_num_channels, |
| render_input_num_channels, capture_output_num_channels)); |
| |
| int result = apm->ProcessReverseStream( |
| &render_input_frame[0], render_input_stream_config, |
| render_output_stream_config, &render_output_frame[0]); |
| EXPECT_EQ(result, AudioProcessing::kNoError); |
| result = apm->ProcessStream( |
| &capture_input_frame[0], capture_input_stream_config, |
| capture_output_stream_config, &capture_output_frame[0]); |
| EXPECT_EQ(result, AudioProcessing::kNoError); |
| } |
| } |
| } |
| } |
| } |
| } |
| } |
| } |
| |
| constexpr void Toggle(bool& b) { |
| b ^= true; |
| } |
| |
| } // namespace |
| |
| TEST(RuntimeSettingTest, TestDefaultCtor) { |
| auto s = AudioProcessing::RuntimeSetting(); |
| EXPECT_EQ(AudioProcessing::RuntimeSetting::Type::kNotSpecified, s.type()); |
| } |
| |
| TEST(RuntimeSettingTest, TestUsageWithSwapQueue) { |
| SwapQueue<AudioProcessing::RuntimeSetting> q(1); |
| auto s = AudioProcessing::RuntimeSetting(); |
| ASSERT_TRUE(q.Insert(&s)); |
| ASSERT_TRUE(q.Remove(&s)); |
| EXPECT_EQ(AudioProcessing::RuntimeSetting::Type::kNotSpecified, s.type()); |
| } |
| |
| TEST(ApmConfiguration, EnablePostProcessing) { |
| // Verify that apm uses a capture post processing module if one is provided. |
| auto mock_post_processor_ptr = |
| new ::testing::NiceMock<test::MockCustomProcessing>(); |
| auto mock_post_processor = |
| std::unique_ptr<CustomProcessing>(mock_post_processor_ptr); |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting() |
| .SetCapturePostProcessing(std::move(mock_post_processor)) |
| .Create(); |
| |
| Int16FrameData audio; |
| audio.num_channels = 1; |
| SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz); |
| |
| EXPECT_CALL(*mock_post_processor_ptr, Process(::testing::_)).Times(1); |
| apm->ProcessStream(audio.data.data(), |
| StreamConfig(audio.sample_rate_hz, audio.num_channels), |
| StreamConfig(audio.sample_rate_hz, audio.num_channels), |
| audio.data.data()); |
| } |
| |
| TEST(ApmConfiguration, EnablePreProcessing) { |
| // Verify that apm uses a capture post processing module if one is provided. |
| auto mock_pre_processor_ptr = |
| new ::testing::NiceMock<test::MockCustomProcessing>(); |
| auto mock_pre_processor = |
| std::unique_ptr<CustomProcessing>(mock_pre_processor_ptr); |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting() |
| .SetRenderPreProcessing(std::move(mock_pre_processor)) |
| .Create(); |
| |
| Int16FrameData audio; |
| audio.num_channels = 1; |
| SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz); |
| |
| EXPECT_CALL(*mock_pre_processor_ptr, Process(::testing::_)).Times(1); |
| apm->ProcessReverseStream( |
| audio.data.data(), StreamConfig(audio.sample_rate_hz, audio.num_channels), |
| StreamConfig(audio.sample_rate_hz, audio.num_channels), |
| audio.data.data()); |
| } |
| |
| TEST(ApmConfiguration, EnableCaptureAnalyzer) { |
| // Verify that apm uses a capture analyzer if one is provided. |
| auto mock_capture_analyzer_ptr = |
| new ::testing::NiceMock<test::MockCustomAudioAnalyzer>(); |
| auto mock_capture_analyzer = |
| std::unique_ptr<CustomAudioAnalyzer>(mock_capture_analyzer_ptr); |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting() |
| .SetCaptureAnalyzer(std::move(mock_capture_analyzer)) |
| .Create(); |
| |
| Int16FrameData audio; |
| audio.num_channels = 1; |
| SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz); |
| |
| EXPECT_CALL(*mock_capture_analyzer_ptr, Analyze(::testing::_)).Times(1); |
| apm->ProcessStream(audio.data.data(), |
| StreamConfig(audio.sample_rate_hz, audio.num_channels), |
| StreamConfig(audio.sample_rate_hz, audio.num_channels), |
| audio.data.data()); |
| } |
| |
| TEST(ApmConfiguration, PreProcessingReceivesRuntimeSettings) { |
| auto mock_pre_processor_ptr = |
| new ::testing::NiceMock<test::MockCustomProcessing>(); |
| auto mock_pre_processor = |
| std::unique_ptr<CustomProcessing>(mock_pre_processor_ptr); |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting() |
| .SetRenderPreProcessing(std::move(mock_pre_processor)) |
| .Create(); |
| apm->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCustomRenderSetting(0)); |
| |
| // RuntimeSettings forwarded during 'Process*Stream' calls. |
| // Therefore we have to make one such call. |
| Int16FrameData audio; |
| audio.num_channels = 1; |
| SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz); |
| |
| EXPECT_CALL(*mock_pre_processor_ptr, SetRuntimeSetting(::testing::_)) |
| .Times(1); |
| apm->ProcessReverseStream( |
| audio.data.data(), StreamConfig(audio.sample_rate_hz, audio.num_channels), |
| StreamConfig(audio.sample_rate_hz, audio.num_channels), |
| audio.data.data()); |
| } |
| |
| class MyEchoControlFactory : public EchoControlFactory { |
| public: |
| std::unique_ptr<EchoControl> Create(int sample_rate_hz) { |
| auto ec = new test::MockEchoControl(); |
| EXPECT_CALL(*ec, AnalyzeRender(::testing::_)).Times(1); |
| EXPECT_CALL(*ec, AnalyzeCapture(::testing::_)).Times(2); |
| EXPECT_CALL(*ec, ProcessCapture(::testing::_, ::testing::_, ::testing::_)) |
| .Times(2); |
| return std::unique_ptr<EchoControl>(ec); |
| } |
| |
| std::unique_ptr<EchoControl> Create(int sample_rate_hz, |
| int num_render_channels
|