Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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modules/audio_processing/test/debug_dump_test.cc
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600
modules/audio_processing/test/debug_dump_test.cc
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/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <stddef.h> // size_t
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#include <memory>
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#include <string>
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#include <vector>
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#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
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#include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
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#include "webrtc/modules/audio_processing/test/debug_dump_replayer.h"
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#include "webrtc/modules/audio_processing/test/test_utils.h"
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#include "webrtc/rtc_base/task_queue.h"
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#include "webrtc/test/gtest.h"
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#include "webrtc/test/testsupport/fileutils.h"
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namespace webrtc {
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namespace test {
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namespace {
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void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer,
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const StreamConfig& config) {
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auto& buffer_ref = *buffer;
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if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
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buffer_ref->num_channels() != config.num_channels()) {
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buffer_ref.reset(new ChannelBuffer<float>(config.num_frames(),
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config.num_channels()));
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}
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}
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class DebugDumpGenerator {
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public:
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DebugDumpGenerator(const std::string& input_file_name,
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int input_rate_hz,
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int input_channels,
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const std::string& reverse_file_name,
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int reverse_rate_hz,
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int reverse_channels,
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const Config& config,
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const std::string& dump_file_name);
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// Constructor that uses default input files.
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explicit DebugDumpGenerator(const Config& config,
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const AudioProcessing::Config& apm_config);
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~DebugDumpGenerator();
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// Changes the sample rate of the input audio to the APM.
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void SetInputRate(int rate_hz);
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// Sets if converts stereo input signal to mono by discarding other channels.
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void ForceInputMono(bool mono);
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// Changes the sample rate of the reverse audio to the APM.
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void SetReverseRate(int rate_hz);
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// Sets if converts stereo reverse signal to mono by discarding other
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// channels.
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void ForceReverseMono(bool mono);
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// Sets the required sample rate of the APM output.
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void SetOutputRate(int rate_hz);
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// Sets the required channels of the APM output.
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void SetOutputChannels(int channels);
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std::string dump_file_name() const { return dump_file_name_; }
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void StartRecording();
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void Process(size_t num_blocks);
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void StopRecording();
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AudioProcessing* apm() const { return apm_.get(); }
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private:
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static void ReadAndDeinterleave(ResampleInputAudioFile* audio, int channels,
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const StreamConfig& config,
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float* const* buffer);
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// APM input/output settings.
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StreamConfig input_config_;
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StreamConfig reverse_config_;
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StreamConfig output_config_;
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// Input file format.
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const std::string input_file_name_;
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ResampleInputAudioFile input_audio_;
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const int input_file_channels_;
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// Reverse file format.
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const std::string reverse_file_name_;
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ResampleInputAudioFile reverse_audio_;
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const int reverse_file_channels_;
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// Buffer for APM input/output.
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std::unique_ptr<ChannelBuffer<float>> input_;
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std::unique_ptr<ChannelBuffer<float>> reverse_;
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std::unique_ptr<ChannelBuffer<float>> output_;
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rtc::TaskQueue worker_queue_;
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std::unique_ptr<AudioProcessing> apm_;
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const std::string dump_file_name_;
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};
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DebugDumpGenerator::DebugDumpGenerator(const std::string& input_file_name,
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int input_rate_hz,
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int input_channels,
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const std::string& reverse_file_name,
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int reverse_rate_hz,
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int reverse_channels,
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const Config& config,
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const std::string& dump_file_name)
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: input_config_(input_rate_hz, input_channels),
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reverse_config_(reverse_rate_hz, reverse_channels),
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output_config_(input_rate_hz, input_channels),
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input_audio_(input_file_name, input_rate_hz, input_rate_hz),
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input_file_channels_(input_channels),
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reverse_audio_(reverse_file_name, reverse_rate_hz, reverse_rate_hz),
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reverse_file_channels_(reverse_channels),
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input_(new ChannelBuffer<float>(input_config_.num_frames(),
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input_config_.num_channels())),
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reverse_(new ChannelBuffer<float>(reverse_config_.num_frames(),
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reverse_config_.num_channels())),
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output_(new ChannelBuffer<float>(output_config_.num_frames(),
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output_config_.num_channels())),
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worker_queue_("debug_dump_generator_worker_queue"),
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apm_(AudioProcessing::Create(config)),
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dump_file_name_(dump_file_name) {}
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DebugDumpGenerator::DebugDumpGenerator(
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const Config& config,
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const AudioProcessing::Config& apm_config)
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: DebugDumpGenerator(ResourcePath("near32_stereo", "pcm"),
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32000,
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2,
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ResourcePath("far32_stereo", "pcm"),
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32000,
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2,
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config,
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TempFilename(OutputPath(), "debug_aec")) {
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apm_->ApplyConfig(apm_config);
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}
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DebugDumpGenerator::~DebugDumpGenerator() {
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remove(dump_file_name_.c_str());
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}
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void DebugDumpGenerator::SetInputRate(int rate_hz) {
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input_audio_.set_output_rate_hz(rate_hz);
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input_config_.set_sample_rate_hz(rate_hz);
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MaybeResetBuffer(&input_, input_config_);
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}
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void DebugDumpGenerator::ForceInputMono(bool mono) {
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const int channels = mono ? 1 : input_file_channels_;
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input_config_.set_num_channels(channels);
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MaybeResetBuffer(&input_, input_config_);
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}
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void DebugDumpGenerator::SetReverseRate(int rate_hz) {
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reverse_audio_.set_output_rate_hz(rate_hz);
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reverse_config_.set_sample_rate_hz(rate_hz);
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MaybeResetBuffer(&reverse_, reverse_config_);
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}
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void DebugDumpGenerator::ForceReverseMono(bool mono) {
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const int channels = mono ? 1 : reverse_file_channels_;
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reverse_config_.set_num_channels(channels);
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MaybeResetBuffer(&reverse_, reverse_config_);
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}
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void DebugDumpGenerator::SetOutputRate(int rate_hz) {
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output_config_.set_sample_rate_hz(rate_hz);
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MaybeResetBuffer(&output_, output_config_);
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}
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void DebugDumpGenerator::SetOutputChannels(int channels) {
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output_config_.set_num_channels(channels);
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MaybeResetBuffer(&output_, output_config_);
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}
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void DebugDumpGenerator::StartRecording() {
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apm_->AttachAecDump(
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AecDumpFactory::Create(dump_file_name_.c_str(), -1, &worker_queue_));
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}
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void DebugDumpGenerator::Process(size_t num_blocks) {
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for (size_t i = 0; i < num_blocks; ++i) {
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ReadAndDeinterleave(&reverse_audio_, reverse_file_channels_,
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reverse_config_, reverse_->channels());
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ReadAndDeinterleave(&input_audio_, input_file_channels_, input_config_,
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input_->channels());
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RTC_CHECK_EQ(AudioProcessing::kNoError, apm_->set_stream_delay_ms(100));
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apm_->set_stream_key_pressed(i % 10 == 9);
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RTC_CHECK_EQ(AudioProcessing::kNoError,
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apm_->ProcessStream(input_->channels(), input_config_,
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output_config_, output_->channels()));
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RTC_CHECK_EQ(AudioProcessing::kNoError,
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apm_->ProcessReverseStream(reverse_->channels(),
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reverse_config_,
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reverse_config_,
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reverse_->channels()));
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}
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}
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void DebugDumpGenerator::StopRecording() {
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apm_->DetachAecDump();
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}
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void DebugDumpGenerator::ReadAndDeinterleave(ResampleInputAudioFile* audio,
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int channels,
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const StreamConfig& config,
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float* const* buffer) {
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const size_t num_frames = config.num_frames();
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const int out_channels = config.num_channels();
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std::vector<int16_t> signal(channels * num_frames);
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audio->Read(num_frames * channels, &signal[0]);
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// We only allow reducing number of channels by discarding some channels.
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RTC_CHECK_LE(out_channels, channels);
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for (int channel = 0; channel < out_channels; ++channel) {
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for (size_t i = 0; i < num_frames; ++i) {
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buffer[channel][i] = S16ToFloat(signal[i * channels + channel]);
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}
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}
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}
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} // namespace
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class DebugDumpTest : public ::testing::Test {
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public:
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// VerifyDebugDump replays a debug dump using APM and verifies that the result
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// is bit-exact-identical to the output channel in the dump. This is only
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// guaranteed if the debug dump is started on the first frame.
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void VerifyDebugDump(const std::string& in_filename);
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private:
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DebugDumpReplayer debug_dump_replayer_;
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};
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void DebugDumpTest::VerifyDebugDump(const std::string& in_filename) {
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ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(in_filename));
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while (const rtc::Optional<audioproc::Event> event =
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debug_dump_replayer_.GetNextEvent()) {
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debug_dump_replayer_.RunNextEvent();
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if (event->type() == audioproc::Event::STREAM) {
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const audioproc::Stream* msg = &event->stream();
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const StreamConfig output_config = debug_dump_replayer_.GetOutputConfig();
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const ChannelBuffer<float>* output = debug_dump_replayer_.GetOutput();
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// Check that output of APM is bit-exact to the output in the dump.
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ASSERT_EQ(output_config.num_channels(),
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static_cast<size_t>(msg->output_channel_size()));
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ASSERT_EQ(output_config.num_frames() * sizeof(float),
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msg->output_channel(0).size());
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for (int i = 0; i < msg->output_channel_size(); ++i) {
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ASSERT_EQ(0, memcmp(output->channels()[i],
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msg->output_channel(i).data(),
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msg->output_channel(i).size()));
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}
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}
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}
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}
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TEST_F(DebugDumpTest, SimpleCase) {
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Config config;
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DebugDumpGenerator generator(config, AudioProcessing::Config());
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generator.StartRecording();
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generator.Process(100);
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generator.StopRecording();
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VerifyDebugDump(generator.dump_file_name());
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}
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TEST_F(DebugDumpTest, ChangeInputFormat) {
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Config config;
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DebugDumpGenerator generator(config, AudioProcessing::Config());
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generator.StartRecording();
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generator.Process(100);
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generator.SetInputRate(48000);
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generator.ForceInputMono(true);
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// Number of output channel should not be larger than that of input. APM will
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// fail otherwise.
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generator.SetOutputChannels(1);
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generator.Process(100);
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generator.StopRecording();
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VerifyDebugDump(generator.dump_file_name());
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}
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TEST_F(DebugDumpTest, ChangeReverseFormat) {
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Config config;
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DebugDumpGenerator generator(config, AudioProcessing::Config());
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generator.StartRecording();
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generator.Process(100);
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generator.SetReverseRate(48000);
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generator.ForceReverseMono(true);
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generator.Process(100);
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generator.StopRecording();
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VerifyDebugDump(generator.dump_file_name());
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}
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TEST_F(DebugDumpTest, ChangeOutputFormat) {
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Config config;
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DebugDumpGenerator generator(config, AudioProcessing::Config());
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generator.StartRecording();
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generator.Process(100);
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generator.SetOutputRate(48000);
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generator.SetOutputChannels(1);
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generator.Process(100);
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generator.StopRecording();
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VerifyDebugDump(generator.dump_file_name());
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}
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TEST_F(DebugDumpTest, ToggleAec) {
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Config config;
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DebugDumpGenerator generator(config, AudioProcessing::Config());
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generator.StartRecording();
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generator.Process(100);
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EchoCancellation* aec = generator.apm()->echo_cancellation();
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EXPECT_EQ(AudioProcessing::kNoError, aec->Enable(!aec->is_enabled()));
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generator.Process(100);
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generator.StopRecording();
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VerifyDebugDump(generator.dump_file_name());
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}
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TEST_F(DebugDumpTest, ToggleDelayAgnosticAec) {
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Config config;
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config.Set<DelayAgnostic>(new DelayAgnostic(true));
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DebugDumpGenerator generator(config, AudioProcessing::Config());
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generator.StartRecording();
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generator.Process(100);
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EchoCancellation* aec = generator.apm()->echo_cancellation();
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EXPECT_EQ(AudioProcessing::kNoError, aec->Enable(!aec->is_enabled()));
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generator.Process(100);
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generator.StopRecording();
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VerifyDebugDump(generator.dump_file_name());
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}
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TEST_F(DebugDumpTest, VerifyRefinedAdaptiveFilterExperimentalString) {
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Config config;
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config.Set<RefinedAdaptiveFilter>(new RefinedAdaptiveFilter(true));
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DebugDumpGenerator generator(config, AudioProcessing::Config());
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generator.StartRecording();
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generator.Process(100);
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generator.StopRecording();
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DebugDumpReplayer debug_dump_replayer_;
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ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
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while (const rtc::Optional<audioproc::Event> event =
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debug_dump_replayer_.GetNextEvent()) {
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debug_dump_replayer_.RunNextEvent();
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if (event->type() == audioproc::Event::CONFIG) {
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const audioproc::Config* msg = &event->config();
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ASSERT_TRUE(msg->has_experiments_description());
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EXPECT_PRED_FORMAT2(testing::IsSubstring, "RefinedAdaptiveFilter",
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msg->experiments_description().c_str());
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}
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}
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}
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TEST_F(DebugDumpTest, VerifyCombinedExperimentalStringInclusive) {
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Config config;
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AudioProcessing::Config apm_config;
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config.Set<RefinedAdaptiveFilter>(new RefinedAdaptiveFilter(true));
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// Arbitrarily set clipping gain to 17, which will never be the default.
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config.Set<ExperimentalAgc>(new ExperimentalAgc(true, 0, 17));
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apm_config.echo_canceller3.enabled = true;
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DebugDumpGenerator generator(config, apm_config);
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generator.StartRecording();
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generator.Process(100);
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generator.StopRecording();
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DebugDumpReplayer debug_dump_replayer_;
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ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
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while (const rtc::Optional<audioproc::Event> event =
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debug_dump_replayer_.GetNextEvent()) {
|
||||
debug_dump_replayer_.RunNextEvent();
|
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if (event->type() == audioproc::Event::CONFIG) {
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const audioproc::Config* msg = &event->config();
|
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ASSERT_TRUE(msg->has_experiments_description());
|
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EXPECT_PRED_FORMAT2(testing::IsSubstring, "RefinedAdaptiveFilter",
|
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msg->experiments_description().c_str());
|
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EXPECT_PRED_FORMAT2(testing::IsSubstring, "EchoCanceller3",
|
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msg->experiments_description().c_str());
|
||||
EXPECT_PRED_FORMAT2(testing::IsSubstring, "AgcClippingLevelExperiment",
|
||||
msg->experiments_description().c_str());
|
||||
}
|
||||
}
|
||||
}
|
||||
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TEST_F(DebugDumpTest, VerifyCombinedExperimentalStringExclusive) {
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Config config;
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config.Set<RefinedAdaptiveFilter>(new RefinedAdaptiveFilter(true));
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DebugDumpGenerator generator(config, AudioProcessing::Config());
|
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generator.StartRecording();
|
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generator.Process(100);
|
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generator.StopRecording();
|
||||
|
||||
DebugDumpReplayer debug_dump_replayer_;
|
||||
|
||||
ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
|
||||
|
||||
while (const rtc::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, "RefinedAdaptiveFilter",
|
||||
msg->experiments_description().c_str());
|
||||
EXPECT_PRED_FORMAT2(testing::IsNotSubstring, "AEC3",
|
||||
msg->experiments_description().c_str());
|
||||
EXPECT_PRED_FORMAT2(testing::IsNotSubstring, "AgcClippingLevelExperiment",
|
||||
msg->experiments_description().c_str());
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
TEST_F(DebugDumpTest, VerifyAec3ExperimentalString) {
|
||||
Config config;
|
||||
AudioProcessing::Config apm_config;
|
||||
apm_config.echo_canceller3.enabled = true;
|
||||
DebugDumpGenerator generator(config, 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 rtc::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, "EchoCanceller3",
|
||||
msg->experiments_description().c_str());
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
TEST_F(DebugDumpTest, VerifyLevelControllerExperimentalString) {
|
||||
Config config;
|
||||
AudioProcessing::Config apm_config;
|
||||
apm_config.level_controller.enabled = true;
|
||||
DebugDumpGenerator generator(config, 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 rtc::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, "LevelController",
|
||||
msg->experiments_description().c_str());
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
TEST_F(DebugDumpTest, VerifyAgcClippingLevelExperimentalString) {
|
||||
Config config;
|
||||
// Arbitrarily set clipping gain to 17, which will never be the default.
|
||||
config.Set<ExperimentalAgc>(new ExperimentalAgc(true, 0, 17));
|
||||
DebugDumpGenerator generator(config, AudioProcessing::Config());
|
||||
generator.StartRecording();
|
||||
generator.Process(100);
|
||||
generator.StopRecording();
|
||||
|
||||
DebugDumpReplayer debug_dump_replayer_;
|
||||
|
||||
ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
|
||||
|
||||
while (const rtc::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, "AgcClippingLevelExperiment",
|
||||
msg->experiments_description().c_str());
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
TEST_F(DebugDumpTest, VerifyEmptyExperimentalString) {
|
||||
Config config;
|
||||
DebugDumpGenerator generator(config, AudioProcessing::Config());
|
||||
generator.StartRecording();
|
||||
generator.Process(100);
|
||||
generator.StopRecording();
|
||||
|
||||
DebugDumpReplayer debug_dump_replayer_;
|
||||
|
||||
ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
|
||||
|
||||
while (const rtc::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());
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
TEST_F(DebugDumpTest, ToggleAecLevel) {
|
||||
Config config;
|
||||
DebugDumpGenerator generator(config, AudioProcessing::Config());
|
||||
EchoCancellation* aec = generator.apm()->echo_cancellation();
|
||||
EXPECT_EQ(AudioProcessing::kNoError, aec->Enable(true));
|
||||
EXPECT_EQ(AudioProcessing::kNoError,
|
||||
aec->set_suppression_level(EchoCancellation::kLowSuppression));
|
||||
generator.StartRecording();
|
||||
generator.Process(100);
|
||||
|
||||
EXPECT_EQ(AudioProcessing::kNoError,
|
||||
aec->set_suppression_level(EchoCancellation::kHighSuppression));
|
||||
generator.Process(100);
|
||||
generator.StopRecording();
|
||||
VerifyDebugDump(generator.dump_file_name());
|
||||
}
|
||||
|
||||
// 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) {
|
||||
Config config;
|
||||
DebugDumpGenerator generator(config, AudioProcessing::Config());
|
||||
generator.StartRecording();
|
||||
generator.Process(100);
|
||||
|
||||
GainControl* agc = generator.apm()->gain_control();
|
||||
EXPECT_EQ(AudioProcessing::kNoError, agc->Enable(!agc->is_enabled()));
|
||||
|
||||
generator.Process(100);
|
||||
generator.StopRecording();
|
||||
VerifyDebugDump(generator.dump_file_name());
|
||||
}
|
||||
|
||||
TEST_F(DebugDumpTest, ToggleNs) {
|
||||
Config config;
|
||||
DebugDumpGenerator generator(config, AudioProcessing::Config());
|
||||
generator.StartRecording();
|
||||
generator.Process(100);
|
||||
|
||||
NoiseSuppression* ns = generator.apm()->noise_suppression();
|
||||
EXPECT_EQ(AudioProcessing::kNoError, ns->Enable(!ns->is_enabled()));
|
||||
|
||||
generator.Process(100);
|
||||
generator.StopRecording();
|
||||
VerifyDebugDump(generator.dump_file_name());
|
||||
}
|
||||
|
||||
TEST_F(DebugDumpTest, TransientSuppressionOn) {
|
||||
Config config;
|
||||
config.Set<ExperimentalNs>(new ExperimentalNs(true));
|
||||
DebugDumpGenerator generator(config, AudioProcessing::Config());
|
||||
generator.StartRecording();
|
||||
generator.Process(100);
|
||||
generator.StopRecording();
|
||||
VerifyDebugDump(generator.dump_file_name());
|
||||
}
|
||||
|
||||
} // namespace test
|
||||
} // namespace webrtc
|
||||
Reference in New Issue
Block a user