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2507f8cdc9
This CL replaces all remaining usage of AudioFrame within APM, with the exception of the AudioProcessing interface. The main changes are within the unittests. Bug: webrtc:5298 Change-Id: I219cdd08f81a8679b28d9dd1359a56837945f3d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170362 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30831}
558 lines
20 KiB
C++
558 lines
20 KiB
C++
/*
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* Copyright (c) 2016 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 "modules/audio_processing/test/aec_dump_based_simulator.h"
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#include <iostream>
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#include <memory>
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#include "modules/audio_processing/echo_control_mobile_impl.h"
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#include "modules/audio_processing/test/protobuf_utils.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/numerics/safe_conversions.h"
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namespace webrtc {
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namespace test {
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namespace {
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// Verify output bitexactness for the fixed interface.
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// TODO(peah): Check whether it would make sense to add a threshold
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// to use for checking the bitexactness in a soft manner.
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bool VerifyFixedBitExactness(const webrtc::audioproc::Stream& msg,
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const Int16Frame& frame) {
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if (sizeof(frame.data[0]) * frame.data.size() != msg.output_data().size()) {
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return false;
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} else {
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const int16_t* frame_data = frame.data.data();
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for (int k = 0; k < frame.num_channels * frame.samples_per_channel; ++k) {
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if (msg.output_data().data()[k] != frame_data[k]) {
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return false;
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}
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}
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}
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return true;
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}
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// Verify output bitexactness for the float interface.
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bool VerifyFloatBitExactness(const webrtc::audioproc::Stream& msg,
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const StreamConfig& out_config,
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const ChannelBuffer<float>& out_buf) {
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if (static_cast<size_t>(msg.output_channel_size()) !=
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out_config.num_channels() ||
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msg.output_channel(0).size() != out_config.num_frames()) {
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return false;
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} else {
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for (int ch = 0; ch < msg.output_channel_size(); ++ch) {
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for (size_t sample = 0; sample < out_config.num_frames(); ++sample) {
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if (msg.output_channel(ch).data()[sample] !=
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out_buf.channels()[ch][sample]) {
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return false;
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}
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}
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}
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}
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return true;
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}
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} // namespace
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AecDumpBasedSimulator::AecDumpBasedSimulator(
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const SimulationSettings& settings,
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std::unique_ptr<AudioProcessingBuilder> ap_builder)
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: AudioProcessingSimulator(settings, std::move(ap_builder)) {
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MaybeOpenCallOrderFile();
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}
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AecDumpBasedSimulator::~AecDumpBasedSimulator() = default;
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void AecDumpBasedSimulator::PrepareProcessStreamCall(
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const webrtc::audioproc::Stream& msg) {
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if (msg.has_input_data()) {
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// Fixed interface processing.
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// Verify interface invariance.
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RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface ||
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interface_used_ == InterfaceType::kNotSpecified);
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interface_used_ = InterfaceType::kFixedInterface;
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// Populate input buffer.
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RTC_CHECK_EQ(sizeof(fwd_frame_.data[0]) * fwd_frame_.data.size(),
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msg.input_data().size());
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memcpy(fwd_frame_.data.data(), msg.input_data().data(),
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msg.input_data().size());
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} else {
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// Float interface processing.
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// Verify interface invariance.
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RTC_CHECK(interface_used_ == InterfaceType::kFloatInterface ||
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interface_used_ == InterfaceType::kNotSpecified);
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interface_used_ = InterfaceType::kFloatInterface;
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RTC_CHECK_EQ(in_buf_->num_channels(),
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static_cast<size_t>(msg.input_channel_size()));
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// Populate input buffer.
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for (size_t i = 0; i < in_buf_->num_channels(); ++i) {
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RTC_CHECK_EQ(in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
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msg.input_channel(i).size());
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std::memcpy(in_buf_->channels()[i], msg.input_channel(i).data(),
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msg.input_channel(i).size());
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}
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}
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if (artificial_nearend_buffer_reader_) {
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if (artificial_nearend_buffer_reader_->Read(
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artificial_nearend_buf_.get())) {
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if (msg.has_input_data()) {
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int16_t* fwd_frame_data = fwd_frame_.data.data();
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for (size_t k = 0; k < in_buf_->num_frames(); ++k) {
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fwd_frame_data[k] = rtc::saturated_cast<int16_t>(
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fwd_frame_data[k] +
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static_cast<int16_t>(32767 *
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artificial_nearend_buf_->channels()[0][k]));
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}
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} else {
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for (int i = 0; i < msg.input_channel_size(); ++i) {
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for (size_t k = 0; k < in_buf_->num_frames(); ++k) {
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in_buf_->channels()[i][k] +=
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artificial_nearend_buf_->channels()[0][k];
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in_buf_->channels()[i][k] = std::min(
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32767.f, std::max(-32768.f, in_buf_->channels()[i][k]));
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}
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}
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}
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} else {
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if (!artificial_nearend_eof_reported_) {
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std::cout << "The artificial nearend file ended before the recording.";
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artificial_nearend_eof_reported_ = true;
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}
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}
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}
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if (!settings_.use_stream_delay || *settings_.use_stream_delay) {
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if (!settings_.stream_delay) {
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if (msg.has_delay()) {
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RTC_CHECK_EQ(AudioProcessing::kNoError,
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ap_->set_stream_delay_ms(msg.delay()));
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}
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} else {
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RTC_CHECK_EQ(AudioProcessing::kNoError,
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ap_->set_stream_delay_ms(*settings_.stream_delay));
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}
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}
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if (!settings_.use_ts) {
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if (msg.has_keypress()) {
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ap_->set_stream_key_pressed(msg.keypress());
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}
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} else {
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ap_->set_stream_key_pressed(*settings_.use_ts);
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}
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// Level is always logged in AEC dumps.
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RTC_CHECK(msg.has_level());
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aec_dump_mic_level_ = msg.level();
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}
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void AecDumpBasedSimulator::VerifyProcessStreamBitExactness(
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const webrtc::audioproc::Stream& msg) {
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if (bitexact_output_) {
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if (interface_used_ == InterfaceType::kFixedInterface) {
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bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_);
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} else {
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bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_);
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}
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}
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}
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void AecDumpBasedSimulator::PrepareReverseProcessStreamCall(
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const webrtc::audioproc::ReverseStream& msg) {
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if (msg.has_data()) {
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// Fixed interface processing.
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// Verify interface invariance.
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RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface ||
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interface_used_ == InterfaceType::kNotSpecified);
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interface_used_ = InterfaceType::kFixedInterface;
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// Populate input buffer.
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RTC_CHECK_EQ(sizeof(rev_frame_.data[0]) * rev_frame_.data.size(),
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msg.data().size());
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memcpy(rev_frame_.data.data(), msg.data().data(), msg.data().size());
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} else {
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// Float interface processing.
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// Verify interface invariance.
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RTC_CHECK(interface_used_ == InterfaceType::kFloatInterface ||
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interface_used_ == InterfaceType::kNotSpecified);
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interface_used_ = InterfaceType::kFloatInterface;
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RTC_CHECK_EQ(reverse_in_buf_->num_channels(),
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static_cast<size_t>(msg.channel_size()));
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// Populate input buffer.
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for (int i = 0; i < msg.channel_size(); ++i) {
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RTC_CHECK_EQ(reverse_in_buf_->num_frames() *
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sizeof(*reverse_in_buf_->channels()[i]),
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msg.channel(i).size());
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std::memcpy(reverse_in_buf_->channels()[i], msg.channel(i).data(),
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msg.channel(i).size());
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}
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}
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}
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void AecDumpBasedSimulator::Process() {
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CreateAudioProcessor();
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if (settings_.artificial_nearend_filename) {
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std::unique_ptr<WavReader> artificial_nearend_file(
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new WavReader(settings_.artificial_nearend_filename->c_str()));
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RTC_CHECK_EQ(1, artificial_nearend_file->num_channels())
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<< "Only mono files for the artificial nearend are supported, "
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"reverted to not using the artificial nearend file";
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const int sample_rate_hz = artificial_nearend_file->sample_rate();
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artificial_nearend_buffer_reader_.reset(
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new ChannelBufferWavReader(std::move(artificial_nearend_file)));
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artificial_nearend_buf_.reset(new ChannelBuffer<float>(
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rtc::CheckedDivExact(sample_rate_hz, kChunksPerSecond), 1));
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}
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webrtc::audioproc::Event event_msg;
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int num_forward_chunks_processed = 0;
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if (settings_.aec_dump_input_string.has_value()) {
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std::stringstream input;
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input << settings_.aec_dump_input_string.value();
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while (ReadMessageFromString(&input, &event_msg))
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HandleEvent(event_msg, &num_forward_chunks_processed);
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} else {
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dump_input_file_ =
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OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
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while (ReadMessageFromFile(dump_input_file_, &event_msg))
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HandleEvent(event_msg, &num_forward_chunks_processed);
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fclose(dump_input_file_);
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}
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DestroyAudioProcessor();
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}
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void AecDumpBasedSimulator::HandleEvent(
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const webrtc::audioproc::Event& event_msg,
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int* num_forward_chunks_processed) {
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switch (event_msg.type()) {
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case webrtc::audioproc::Event::INIT:
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RTC_CHECK(event_msg.has_init());
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HandleMessage(event_msg.init());
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break;
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case webrtc::audioproc::Event::STREAM:
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RTC_CHECK(event_msg.has_stream());
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HandleMessage(event_msg.stream());
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++num_forward_chunks_processed;
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break;
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case webrtc::audioproc::Event::REVERSE_STREAM:
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RTC_CHECK(event_msg.has_reverse_stream());
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HandleMessage(event_msg.reverse_stream());
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break;
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case webrtc::audioproc::Event::CONFIG:
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RTC_CHECK(event_msg.has_config());
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HandleMessage(event_msg.config());
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break;
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case webrtc::audioproc::Event::RUNTIME_SETTING:
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HandleMessage(event_msg.runtime_setting());
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break;
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case webrtc::audioproc::Event::UNKNOWN_EVENT:
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RTC_CHECK(false);
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break;
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}
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}
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void AecDumpBasedSimulator::HandleMessage(
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const webrtc::audioproc::Config& msg) {
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if (settings_.use_verbose_logging) {
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std::cout << "Config at frame:" << std::endl;
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std::cout << " Forward: " << get_num_process_stream_calls() << std::endl;
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std::cout << " Reverse: " << get_num_reverse_process_stream_calls()
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<< std::endl;
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}
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if (!settings_.discard_all_settings_in_aecdump) {
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if (settings_.use_verbose_logging) {
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std::cout << "Setting used in config:" << std::endl;
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}
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Config config;
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AudioProcessing::Config apm_config = ap_->GetConfig();
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if (msg.has_aec_enabled() || settings_.use_aec) {
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bool enable = settings_.use_aec ? *settings_.use_aec : msg.aec_enabled();
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apm_config.echo_canceller.enabled = enable;
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if (settings_.use_verbose_logging) {
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std::cout << " aec_enabled: " << (enable ? "true" : "false")
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<< std::endl;
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}
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}
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if (msg.has_aecm_enabled() || settings_.use_aecm) {
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bool enable =
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settings_.use_aecm ? *settings_.use_aecm : msg.aecm_enabled();
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apm_config.echo_canceller.enabled |= enable;
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apm_config.echo_canceller.mobile_mode = enable;
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if (settings_.use_verbose_logging) {
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std::cout << " aecm_enabled: " << (enable ? "true" : "false")
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<< std::endl;
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}
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}
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if (msg.has_aecm_comfort_noise_enabled() &&
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msg.aecm_comfort_noise_enabled()) {
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RTC_LOG(LS_ERROR) << "Ignoring deprecated setting: AECM comfort noise";
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}
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if (msg.has_aecm_routing_mode() &&
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static_cast<webrtc::EchoControlMobileImpl::RoutingMode>(
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msg.aecm_routing_mode()) != EchoControlMobileImpl::kSpeakerphone) {
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RTC_LOG(LS_ERROR) << "Ignoring deprecated setting: AECM routing mode: "
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<< msg.aecm_routing_mode();
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}
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if (msg.has_agc_enabled() || settings_.use_agc) {
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bool enable = settings_.use_agc ? *settings_.use_agc : msg.agc_enabled();
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apm_config.gain_controller1.enabled = enable;
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if (settings_.use_verbose_logging) {
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std::cout << " agc_enabled: " << (enable ? "true" : "false")
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<< std::endl;
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}
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}
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if (msg.has_agc_mode() || settings_.agc_mode) {
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int mode = settings_.agc_mode ? *settings_.agc_mode : msg.agc_mode();
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apm_config.gain_controller1.mode =
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static_cast<webrtc::AudioProcessing::Config::GainController1::Mode>(
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mode);
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if (settings_.use_verbose_logging) {
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std::cout << " agc_mode: " << mode << std::endl;
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}
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}
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if (msg.has_agc_limiter_enabled() || settings_.use_agc_limiter) {
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bool enable = settings_.use_agc_limiter ? *settings_.use_agc_limiter
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: msg.agc_limiter_enabled();
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apm_config.gain_controller1.enable_limiter = enable;
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if (settings_.use_verbose_logging) {
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std::cout << " agc_limiter_enabled: " << (enable ? "true" : "false")
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<< std::endl;
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}
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}
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if (settings_.use_agc2) {
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bool enable = *settings_.use_agc2;
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apm_config.gain_controller2.enabled = enable;
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if (settings_.agc2_fixed_gain_db) {
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apm_config.gain_controller2.fixed_digital.gain_db =
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*settings_.agc2_fixed_gain_db;
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}
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if (settings_.use_verbose_logging) {
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std::cout << " agc2_enabled: " << (enable ? "true" : "false")
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<< std::endl;
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}
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}
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if (msg.has_noise_robust_agc_enabled()) {
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apm_config.gain_controller1.analog_gain_controller.enabled =
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settings_.use_analog_agc ? *settings_.use_analog_agc
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: msg.noise_robust_agc_enabled();
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if (settings_.use_verbose_logging) {
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std::cout << " noise_robust_agc_enabled: "
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<< (msg.noise_robust_agc_enabled() ? "true" : "false")
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<< std::endl;
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}
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}
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if (msg.has_transient_suppression_enabled() || settings_.use_ts) {
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bool enable = settings_.use_ts ? *settings_.use_ts
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: msg.transient_suppression_enabled();
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apm_config.transient_suppression.enabled = enable;
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if (settings_.use_verbose_logging) {
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std::cout << " transient_suppression_enabled: "
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<< (enable ? "true" : "false") << std::endl;
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}
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}
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if (msg.has_hpf_enabled() || settings_.use_hpf) {
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bool enable = settings_.use_hpf ? *settings_.use_hpf : msg.hpf_enabled();
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apm_config.high_pass_filter.enabled = enable;
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if (settings_.use_verbose_logging) {
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std::cout << " hpf_enabled: " << (enable ? "true" : "false")
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<< std::endl;
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}
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}
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if (msg.has_ns_enabled() || settings_.use_ns) {
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bool enable = settings_.use_ns ? *settings_.use_ns : msg.ns_enabled();
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apm_config.noise_suppression.enabled = enable;
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if (settings_.use_verbose_logging) {
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std::cout << " ns_enabled: " << (enable ? "true" : "false")
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<< std::endl;
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}
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}
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if (msg.has_ns_level() || settings_.ns_level) {
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int level = settings_.ns_level ? *settings_.ns_level : msg.ns_level();
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apm_config.noise_suppression.level =
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static_cast<AudioProcessing::Config::NoiseSuppression::Level>(level);
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if (settings_.use_verbose_logging) {
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std::cout << " ns_level: " << level << std::endl;
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}
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}
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if (msg.has_pre_amplifier_enabled() || settings_.use_pre_amplifier) {
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const bool enable = settings_.use_pre_amplifier
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? *settings_.use_pre_amplifier
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: msg.pre_amplifier_enabled();
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apm_config.pre_amplifier.enabled = enable;
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}
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if (msg.has_pre_amplifier_fixed_gain_factor() ||
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settings_.pre_amplifier_gain_factor) {
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const float gain = settings_.pre_amplifier_gain_factor
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? *settings_.pre_amplifier_gain_factor
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: msg.pre_amplifier_fixed_gain_factor();
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apm_config.pre_amplifier.fixed_gain_factor = gain;
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}
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if (settings_.use_verbose_logging && msg.has_experiments_description() &&
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!msg.experiments_description().empty()) {
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std::cout << " experiments not included by default in the simulation: "
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<< msg.experiments_description() << std::endl;
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}
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if (settings_.use_ed) {
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apm_config.residual_echo_detector.enabled = *settings_.use_ed;
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}
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ap_->ApplyConfig(apm_config);
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ap_->SetExtraOptions(config);
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}
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}
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void AecDumpBasedSimulator::HandleMessage(const webrtc::audioproc::Init& msg) {
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RTC_CHECK(msg.has_sample_rate());
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RTC_CHECK(msg.has_num_input_channels());
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RTC_CHECK(msg.has_num_reverse_channels());
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RTC_CHECK(msg.has_reverse_sample_rate());
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MaybeOpenCallOrderFile();
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if (settings_.use_verbose_logging) {
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|
std::cout << "Init at frame:" << std::endl;
|
|
std::cout << " Forward: " << get_num_process_stream_calls() << std::endl;
|
|
std::cout << " Reverse: " << get_num_reverse_process_stream_calls()
|
|
<< std::endl;
|
|
}
|
|
|
|
int num_output_channels;
|
|
if (settings_.output_num_channels) {
|
|
num_output_channels = *settings_.output_num_channels;
|
|
} else {
|
|
num_output_channels = msg.has_num_output_channels()
|
|
? msg.num_output_channels()
|
|
: msg.num_input_channels();
|
|
}
|
|
|
|
int output_sample_rate;
|
|
if (settings_.output_sample_rate_hz) {
|
|
output_sample_rate = *settings_.output_sample_rate_hz;
|
|
} else {
|
|
output_sample_rate = msg.has_output_sample_rate() ? msg.output_sample_rate()
|
|
: msg.sample_rate();
|
|
}
|
|
|
|
int num_reverse_output_channels;
|
|
if (settings_.reverse_output_num_channels) {
|
|
num_reverse_output_channels = *settings_.reverse_output_num_channels;
|
|
} else {
|
|
num_reverse_output_channels = msg.has_num_reverse_output_channels()
|
|
? msg.num_reverse_output_channels()
|
|
: msg.num_reverse_channels();
|
|
}
|
|
|
|
int reverse_output_sample_rate;
|
|
if (settings_.reverse_output_sample_rate_hz) {
|
|
reverse_output_sample_rate = *settings_.reverse_output_sample_rate_hz;
|
|
} else {
|
|
reverse_output_sample_rate = msg.has_reverse_output_sample_rate()
|
|
? msg.reverse_output_sample_rate()
|
|
: msg.reverse_sample_rate();
|
|
}
|
|
|
|
SetupBuffersConfigsOutputs(
|
|
msg.sample_rate(), output_sample_rate, msg.reverse_sample_rate(),
|
|
reverse_output_sample_rate, msg.num_input_channels(), num_output_channels,
|
|
msg.num_reverse_channels(), num_reverse_output_channels);
|
|
}
|
|
|
|
void AecDumpBasedSimulator::HandleMessage(
|
|
const webrtc::audioproc::Stream& msg) {
|
|
if (call_order_output_file_) {
|
|
*call_order_output_file_ << "c";
|
|
}
|
|
PrepareProcessStreamCall(msg);
|
|
ProcessStream(interface_used_ == InterfaceType::kFixedInterface);
|
|
VerifyProcessStreamBitExactness(msg);
|
|
}
|
|
|
|
void AecDumpBasedSimulator::HandleMessage(
|
|
const webrtc::audioproc::ReverseStream& msg) {
|
|
if (call_order_output_file_) {
|
|
*call_order_output_file_ << "r";
|
|
}
|
|
PrepareReverseProcessStreamCall(msg);
|
|
ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface);
|
|
}
|
|
|
|
void AecDumpBasedSimulator::HandleMessage(
|
|
const webrtc::audioproc::RuntimeSetting& msg) {
|
|
RTC_CHECK(ap_.get());
|
|
if (msg.has_capture_pre_gain()) {
|
|
// Handle capture pre-gain runtime setting only if not overridden.
|
|
if ((!settings_.use_pre_amplifier || *settings_.use_pre_amplifier) &&
|
|
!settings_.pre_amplifier_gain_factor) {
|
|
ap_->SetRuntimeSetting(
|
|
AudioProcessing::RuntimeSetting::CreateCapturePreGain(
|
|
msg.capture_pre_gain()));
|
|
}
|
|
} else if (msg.has_capture_fixed_post_gain()) {
|
|
// Handle capture fixed-post-gain runtime setting only if not overridden.
|
|
if ((!settings_.use_agc2 || *settings_.use_agc2) &&
|
|
!settings_.agc2_fixed_gain_db) {
|
|
ap_->SetRuntimeSetting(
|
|
AudioProcessing::RuntimeSetting::CreateCaptureFixedPostGain(
|
|
msg.capture_fixed_post_gain()));
|
|
}
|
|
} else if (msg.has_playout_volume_change()) {
|
|
ap_->SetRuntimeSetting(
|
|
AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(
|
|
msg.playout_volume_change()));
|
|
} else if (msg.has_playout_audio_device_change()) {
|
|
ap_->SetRuntimeSetting(
|
|
AudioProcessing::RuntimeSetting::CreatePlayoutAudioDeviceChange(
|
|
{msg.playout_audio_device_change().id(),
|
|
msg.playout_audio_device_change().max_volume()}));
|
|
}
|
|
}
|
|
|
|
void AecDumpBasedSimulator::MaybeOpenCallOrderFile() {
|
|
if (settings_.call_order_output_filename.has_value()) {
|
|
const std::string filename = settings_.store_intermediate_output
|
|
? *settings_.call_order_output_filename +
|
|
"_" +
|
|
std::to_string(output_reset_counter_)
|
|
: *settings_.call_order_output_filename;
|
|
call_order_output_file_ = std::make_unique<std::ofstream>(filename);
|
|
}
|
|
}
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|