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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc | 504 |
1 files changed, 504 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc new file mode 100644 index 0000000000..cded5de217 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc @@ -0,0 +1,504 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <stddef.h> // size_t + +#include <memory> +#include <string> +#include <vector> + +#include "absl/strings/string_view.h" +#include "api/audio/echo_canceller3_factory.h" +#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h" +#include "modules/audio_processing/aec_dump/aec_dump_factory.h" +#include "modules/audio_processing/test/audio_processing_builder_for_testing.h" +#include "modules/audio_processing/test/debug_dump_replayer.h" +#include "modules/audio_processing/test/test_utils.h" +#include "rtc_base/task_queue_for_test.h" +#include "test/gtest.h" +#include "test/testsupport/file_utils.h" + +namespace webrtc { +namespace test { + +namespace { + +void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer, + const StreamConfig& config) { + auto& buffer_ref = *buffer; + if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() || + buffer_ref->num_channels() != config.num_channels()) { + buffer_ref.reset( + new ChannelBuffer<float>(config.num_frames(), config.num_channels())); + } +} + +class DebugDumpGenerator { + public: + DebugDumpGenerator(absl::string_view input_file_name, + int input_rate_hz, + int input_channels, + absl::string_view reverse_file_name, + int reverse_rate_hz, + int reverse_channels, + absl::string_view dump_file_name, + bool enable_pre_amplifier); + + // Constructor that uses default input files. + explicit DebugDumpGenerator(const AudioProcessing::Config& apm_config); + + ~DebugDumpGenerator(); + + // Changes the sample rate of the input audio to the APM. + void SetInputRate(int rate_hz); + + // Sets if converts stereo input signal to mono by discarding other channels. + void ForceInputMono(bool mono); + + // Changes the sample rate of the reverse audio to the APM. + void SetReverseRate(int rate_hz); + + // Sets if converts stereo reverse signal to mono by discarding other + // channels. + void ForceReverseMono(bool mono); + + // Sets the required sample rate of the APM output. + void SetOutputRate(int rate_hz); + + // Sets the required channels of the APM output. + void SetOutputChannels(int channels); + + std::string dump_file_name() const { return dump_file_name_; } + + void StartRecording(); + void Process(size_t num_blocks); + void StopRecording(); + AudioProcessing* apm() const { return apm_.get(); } + + private: + static void ReadAndDeinterleave(ResampleInputAudioFile* audio, + int channels, + const StreamConfig& config, + float* const* buffer); + + // APM input/output settings. + StreamConfig input_config_; + StreamConfig reverse_config_; + StreamConfig output_config_; + + // Input file format. + const std::string input_file_name_; + ResampleInputAudioFile input_audio_; + const int input_file_channels_; + + // Reverse file format. + const std::string reverse_file_name_; + ResampleInputAudioFile reverse_audio_; + const int reverse_file_channels_; + + // Buffer for APM input/output. + std::unique_ptr<ChannelBuffer<float>> input_; + std::unique_ptr<ChannelBuffer<float>> reverse_; + std::unique_ptr<ChannelBuffer<float>> output_; + + bool enable_pre_amplifier_; + + TaskQueueForTest worker_queue_; + rtc::scoped_refptr<AudioProcessing> apm_; + + const std::string dump_file_name_; +}; + +DebugDumpGenerator::DebugDumpGenerator(absl::string_view input_file_name, + int input_rate_hz, + int input_channels, + absl::string_view reverse_file_name, + int reverse_rate_hz, + int reverse_channels, + absl::string_view dump_file_name, + bool enable_pre_amplifier) + : input_config_(input_rate_hz, input_channels), + reverse_config_(reverse_rate_hz, reverse_channels), + output_config_(input_rate_hz, input_channels), + input_audio_(input_file_name, input_rate_hz, input_rate_hz), + input_file_channels_(input_channels), + reverse_audio_(reverse_file_name, reverse_rate_hz, reverse_rate_hz), + reverse_file_channels_(reverse_channels), + input_(new ChannelBuffer<float>(input_config_.num_frames(), + input_config_.num_channels())), + reverse_(new ChannelBuffer<float>(reverse_config_.num_frames(), + reverse_config_.num_channels())), + output_(new ChannelBuffer<float>(output_config_.num_frames(), + output_config_.num_channels())), + enable_pre_amplifier_(enable_pre_amplifier), + worker_queue_("debug_dump_generator_worker_queue"), + dump_file_name_(dump_file_name) { + AudioProcessingBuilderForTesting apm_builder; + apm_ = apm_builder.Create(); +} + +DebugDumpGenerator::DebugDumpGenerator( + const AudioProcessing::Config& apm_config) + : DebugDumpGenerator(ResourcePath("near32_stereo", "pcm"), + 32000, + 2, + ResourcePath("far32_stereo", "pcm"), + 32000, + 2, + TempFilename(OutputPath(), "debug_aec"), + apm_config.pre_amplifier.enabled) { + apm_->ApplyConfig(apm_config); +} + +DebugDumpGenerator::~DebugDumpGenerator() { + remove(dump_file_name_.c_str()); +} + +void DebugDumpGenerator::SetInputRate(int rate_hz) { + input_audio_.set_output_rate_hz(rate_hz); + input_config_.set_sample_rate_hz(rate_hz); + MaybeResetBuffer(&input_, input_config_); +} + +void DebugDumpGenerator::ForceInputMono(bool mono) { + const int channels = mono ? 1 : input_file_channels_; + input_config_.set_num_channels(channels); + MaybeResetBuffer(&input_, input_config_); +} + +void DebugDumpGenerator::SetReverseRate(int rate_hz) { + reverse_audio_.set_output_rate_hz(rate_hz); + reverse_config_.set_sample_rate_hz(rate_hz); + MaybeResetBuffer(&reverse_, reverse_config_); +} + +void DebugDumpGenerator::ForceReverseMono(bool mono) { + const int channels = mono ? 1 : reverse_file_channels_; + reverse_config_.set_num_channels(channels); + MaybeResetBuffer(&reverse_, reverse_config_); +} + +void DebugDumpGenerator::SetOutputRate(int rate_hz) { + output_config_.set_sample_rate_hz(rate_hz); + MaybeResetBuffer(&output_, output_config_); +} + +void DebugDumpGenerator::SetOutputChannels(int channels) { + output_config_.set_num_channels(channels); + MaybeResetBuffer(&output_, output_config_); +} + +void DebugDumpGenerator::StartRecording() { + apm_->AttachAecDump( + AecDumpFactory::Create(dump_file_name_.c_str(), -1, &worker_queue_)); +} + +void DebugDumpGenerator::Process(size_t num_blocks) { + for (size_t i = 0; i < num_blocks; ++i) { + ReadAndDeinterleave(&reverse_audio_, reverse_file_channels_, + reverse_config_, reverse_->channels()); + ReadAndDeinterleave(&input_audio_, input_file_channels_, input_config_, + input_->channels()); + RTC_CHECK_EQ(AudioProcessing::kNoError, apm_->set_stream_delay_ms(100)); + apm_->set_stream_analog_level(100); + if (enable_pre_amplifier_) { + apm_->SetRuntimeSetting( + AudioProcessing::RuntimeSetting::CreateCapturePreGain(1 + i % 10)); + } + apm_->set_stream_key_pressed(i % 10 == 9); + RTC_CHECK_EQ(AudioProcessing::kNoError, + apm_->ProcessStream(input_->channels(), input_config_, + output_config_, output_->channels())); + + RTC_CHECK_EQ( + AudioProcessing::kNoError, + apm_->ProcessReverseStream(reverse_->channels(), reverse_config_, + reverse_config_, reverse_->channels())); + } +} + +void DebugDumpGenerator::StopRecording() { + apm_->DetachAecDump(); +} + +void DebugDumpGenerator::ReadAndDeinterleave(ResampleInputAudioFile* audio, + int channels, + const StreamConfig& config, + float* const* buffer) { + const size_t num_frames = config.num_frames(); + const int out_channels = config.num_channels(); + + std::vector<int16_t> signal(channels * num_frames); + + audio->Read(num_frames * channels, &signal[0]); + + // We only allow reducing number of channels by discarding some channels. + RTC_CHECK_LE(out_channels, channels); + for (int channel = 0; channel < out_channels; ++channel) { + for (size_t i = 0; i < num_frames; ++i) { + buffer[channel][i] = S16ToFloat(signal[i * channels + channel]); + } + } +} + +} // namespace + +class DebugDumpTest : public ::testing::Test { + public: + // VerifyDebugDump replays a debug dump using APM and verifies that the result + // is bit-exact-identical to the output channel in the dump. This is only + // guaranteed if the debug dump is started on the first frame. + void VerifyDebugDump(absl::string_view in_filename); + + private: + DebugDumpReplayer debug_dump_replayer_; +}; + +void DebugDumpTest::VerifyDebugDump(absl::string_view in_filename) { + ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(in_filename)); + + while (const absl::optional<audioproc::Event> event = + debug_dump_replayer_.GetNextEvent()) { + debug_dump_replayer_.RunNextEvent(); + if (event->type() == audioproc::Event::STREAM) { + const audioproc::Stream* msg = &event->stream(); + const StreamConfig output_config = debug_dump_replayer_.GetOutputConfig(); + const ChannelBuffer<float>* output = debug_dump_replayer_.GetOutput(); + // Check that output of APM is bit-exact to the output in the dump. + ASSERT_EQ(output_config.num_channels(), + static_cast<size_t>(msg->output_channel_size())); + ASSERT_EQ(output_config.num_frames() * sizeof(float), + msg->output_channel(0).size()); + for (int i = 0; i < msg->output_channel_size(); ++i) { + ASSERT_EQ(0, + memcmp(output->channels()[i], msg->output_channel(i).data(), + msg->output_channel(i).size())); + } + } + } +} + +TEST_F(DebugDumpTest, SimpleCase) { + DebugDumpGenerator generator(/*apm_config=*/{}); + generator.StartRecording(); + generator.Process(100); + generator.StopRecording(); + VerifyDebugDump(generator.dump_file_name()); +} + +TEST_F(DebugDumpTest, ChangeInputFormat) { + DebugDumpGenerator generator(/*apm_config=*/{}); + + generator.StartRecording(); + generator.Process(100); + generator.SetInputRate(48000); + + generator.ForceInputMono(true); + // Number of output channel should not be larger than that of input. APM will + // fail otherwise. + generator.SetOutputChannels(1); + + generator.Process(100); + generator.StopRecording(); + VerifyDebugDump(generator.dump_file_name()); +} + +TEST_F(DebugDumpTest, ChangeReverseFormat) { + DebugDumpGenerator generator(/*apm_config=*/{}); + generator.StartRecording(); + generator.Process(100); + generator.SetReverseRate(48000); + generator.ForceReverseMono(true); + generator.Process(100); + generator.StopRecording(); + VerifyDebugDump(generator.dump_file_name()); +} + +TEST_F(DebugDumpTest, ChangeOutputFormat) { + DebugDumpGenerator generator(/*apm_config=*/{}); + generator.StartRecording(); + generator.Process(100); + generator.SetOutputRate(48000); + generator.SetOutputChannels(1); + generator.Process(100); + generator.StopRecording(); + VerifyDebugDump(generator.dump_file_name()); +} + +TEST_F(DebugDumpTest, ToggleAec) { + AudioProcessing::Config apm_config; + apm_config.echo_canceller.enabled = true; + DebugDumpGenerator generator(apm_config); + generator.StartRecording(); + generator.Process(100); + + apm_config.echo_canceller.enabled = false; + generator.apm()->ApplyConfig(apm_config); + + generator.Process(100); + generator.StopRecording(); + VerifyDebugDump(generator.dump_file_name()); +} + +TEST_F(DebugDumpTest, VerifyCombinedExperimentalStringInclusive) { + AudioProcessing::Config apm_config; + apm_config.echo_canceller.enabled = true; + apm_config.gain_controller1.analog_gain_controller.enabled = true; + apm_config.gain_controller1.analog_gain_controller.startup_min_volume = 0; + DebugDumpGenerator generator(apm_config); + generator.StartRecording(); + generator.Process(100); + generator.StopRecording(); + + DebugDumpReplayer debug_dump_replayer_; + + ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name())); + + while (const absl::optional<audioproc::Event> event = + debug_dump_replayer_.GetNextEvent()) { + debug_dump_replayer_.RunNextEvent(); + if (event->type() == audioproc::Event::CONFIG) { + const audioproc::Config* msg = &event->config(); + ASSERT_TRUE(msg->has_experiments_description()); + EXPECT_PRED_FORMAT2(::testing::IsSubstring, "EchoController", + msg->experiments_description().c_str()); + } + } +} + +TEST_F(DebugDumpTest, VerifyCombinedExperimentalStringExclusive) { + AudioProcessing::Config apm_config; + apm_config.echo_canceller.enabled = true; + DebugDumpGenerator generator(apm_config); + generator.StartRecording(); + generator.Process(100); + generator.StopRecording(); + + DebugDumpReplayer debug_dump_replayer_; + + ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name())); + + while (const absl::optional<audioproc::Event> event = + debug_dump_replayer_.GetNextEvent()) { + debug_dump_replayer_.RunNextEvent(); + if (event->type() == audioproc::Event::CONFIG) { + const audioproc::Config* msg = &event->config(); + ASSERT_TRUE(msg->has_experiments_description()); + EXPECT_PRED_FORMAT2(::testing::IsNotSubstring, + "AgcClippingLevelExperiment", + msg->experiments_description().c_str()); + } + } +} + +TEST_F(DebugDumpTest, VerifyAec3ExperimentalString) { + AudioProcessing::Config apm_config; + apm_config.echo_canceller.enabled = true; + DebugDumpGenerator generator(apm_config); + generator.StartRecording(); + generator.Process(100); + generator.StopRecording(); + + DebugDumpReplayer debug_dump_replayer_; + + ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name())); + + while (const absl::optional<audioproc::Event> event = + debug_dump_replayer_.GetNextEvent()) { + debug_dump_replayer_.RunNextEvent(); + if (event->type() == audioproc::Event::CONFIG) { + const audioproc::Config* msg = &event->config(); + ASSERT_TRUE(msg->has_experiments_description()); + EXPECT_PRED_FORMAT2(::testing::IsSubstring, "EchoController", + msg->experiments_description().c_str()); + } + } +} + +TEST_F(DebugDumpTest, VerifyEmptyExperimentalString) { + DebugDumpGenerator generator(/*apm_config=*/{}); + generator.StartRecording(); + generator.Process(100); + generator.StopRecording(); + + DebugDumpReplayer debug_dump_replayer_; + + ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name())); + + while (const absl::optional<audioproc::Event> event = + debug_dump_replayer_.GetNextEvent()) { + debug_dump_replayer_.RunNextEvent(); + if (event->type() == audioproc::Event::CONFIG) { + const audioproc::Config* msg = &event->config(); + ASSERT_TRUE(msg->has_experiments_description()); + EXPECT_EQ(0u, msg->experiments_description().size()); + } + } +} + +// AGC is not supported on Android or iOS. +#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) +#define MAYBE_ToggleAgc DISABLED_ToggleAgc +#else +#define MAYBE_ToggleAgc ToggleAgc +#endif +TEST_F(DebugDumpTest, MAYBE_ToggleAgc) { + DebugDumpGenerator generator(/*apm_config=*/{}); + generator.StartRecording(); + generator.Process(100); + + AudioProcessing::Config apm_config = generator.apm()->GetConfig(); + apm_config.gain_controller1.enabled = !apm_config.gain_controller1.enabled; + generator.apm()->ApplyConfig(apm_config); + + generator.Process(100); + generator.StopRecording(); + VerifyDebugDump(generator.dump_file_name()); +} + +TEST_F(DebugDumpTest, ToggleNs) { + DebugDumpGenerator generator(/*apm_config=*/{}); + generator.StartRecording(); + generator.Process(100); + + AudioProcessing::Config apm_config = generator.apm()->GetConfig(); + apm_config.noise_suppression.enabled = !apm_config.noise_suppression.enabled; + generator.apm()->ApplyConfig(apm_config); + + generator.Process(100); + generator.StopRecording(); + VerifyDebugDump(generator.dump_file_name()); +} + +TEST_F(DebugDumpTest, TransientSuppressionOn) { + DebugDumpGenerator generator(/*apm_config=*/{}); + + AudioProcessing::Config apm_config = generator.apm()->GetConfig(); + apm_config.transient_suppression.enabled = true; + generator.apm()->ApplyConfig(apm_config); + + generator.StartRecording(); + generator.Process(100); + generator.StopRecording(); + VerifyDebugDump(generator.dump_file_name()); +} + +TEST_F(DebugDumpTest, PreAmplifierIsOn) { + AudioProcessing::Config apm_config; + apm_config.pre_amplifier.enabled = true; + DebugDumpGenerator generator(apm_config); + generator.StartRecording(); + generator.Process(100); + generator.StopRecording(); + VerifyDebugDump(generator.dump_file_name()); +} + +} // namespace test +} // namespace webrtc |