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webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
Sam Zackrisson 05a7004442 Revert "Remove APM-internal usage of EchoControlMobile" 
This reverts commit 2fbb83b16b.

Reason for revert: Speculative revert over failing Chromium bot:
https://ci.chromium.org/p/chromium/builders/luci.chromium.webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28M%20Nexus5X%29/117

Original change's description:
> Remove APM-internal usage of EchoControlMobile
> 
> This is a sibling CL to a similar one for EchoCancellation:
> https://webrtc-review.googlesource.com/c/src/+/97603
> 
>  - EchoControlMobileImpl will no longer inherit EchoControlMobile.
>  - Removes usage of AudioProcessing::echo_control_mobile() inside most of
>    the audio processing module and unit tests.
> 
> The CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (comfort noise, routing mode), but prints an
> error message when unsupported settings are encountered.
> 
> Tested: audioproc_f with .wav and aecdump inputs.
> Bug: webrtc:9535
> Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
> Reviewed-on: https://webrtc-review.googlesource.com/101621
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24888}

TBR=saza@webrtc.org,aleloi@webrtc.org

Change-Id: I1f8a27ac291f2cdc16c8daa32e399b74d489dbb9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/102642
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24895}
2018-09-28 13:39:19 +00:00

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/*
* Copyright (c) 2016 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 <iostream>
#include "modules/audio_processing/test/aec_dump_based_simulator.h"
#include "modules/audio_processing/test/runtime_setting_util.h"
#include "modules/audio_processing/test/protobuf_utils.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
namespace webrtc {
namespace test {
namespace {
// Verify output bitexactness for the fixed interface.
// TODO(peah): Check whether it would make sense to add a threshold
// to use for checking the bitexactness in a soft manner.
bool VerifyFixedBitExactness(const webrtc::audioproc::Stream& msg,
const AudioFrame& frame) {
if ((sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_) !=
msg.output_data().size()) {
return false;
} else {
const int16_t* frame_data = frame.data();
for (size_t k = 0; k < frame.num_channels_ * frame.samples_per_channel_;
++k) {
if (msg.output_data().data()[k] != frame_data[k]) {
return false;
}
}
}
return true;
}
// Verify output bitexactness for the float interface.
bool VerifyFloatBitExactness(const webrtc::audioproc::Stream& msg,
const StreamConfig& out_config,
const ChannelBuffer<float>& out_buf) {
if (static_cast<size_t>(msg.output_channel_size()) !=
out_config.num_channels() ||
msg.output_channel(0).size() != out_config.num_frames()) {
return false;
} else {
for (int ch = 0; ch < msg.output_channel_size(); ++ch) {
for (size_t sample = 0; sample < out_config.num_frames(); ++sample) {
if (msg.output_channel(ch).data()[sample] !=
out_buf.channels()[ch][sample]) {
return false;
}
}
}
}
return true;
}
} // namespace
AecDumpBasedSimulator::AecDumpBasedSimulator(
const SimulationSettings& settings,
std::unique_ptr<AudioProcessingBuilder> ap_builder)
: AudioProcessingSimulator(settings, std::move(ap_builder)) {}
AecDumpBasedSimulator::~AecDumpBasedSimulator() = default;
void AecDumpBasedSimulator::PrepareProcessStreamCall(
const webrtc::audioproc::Stream& msg) {
if (msg.has_input_data()) {
// Fixed interface processing.
// Verify interface invariance.
RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface ||
interface_used_ == InterfaceType::kNotSpecified);
interface_used_ = InterfaceType::kFixedInterface;
// Populate input buffer.
RTC_CHECK_EQ(sizeof(*fwd_frame_.data()) * fwd_frame_.samples_per_channel_ *
fwd_frame_.num_channels_,
msg.input_data().size());
memcpy(fwd_frame_.mutable_data(), msg.input_data().data(),
msg.input_data().size());
} else {
// Float interface processing.
// Verify interface invariance.
RTC_CHECK(interface_used_ == InterfaceType::kFloatInterface ||
interface_used_ == InterfaceType::kNotSpecified);
interface_used_ = InterfaceType::kFloatInterface;
RTC_CHECK_EQ(in_buf_->num_channels(),
static_cast<size_t>(msg.input_channel_size()));
// Populate input buffer.
for (size_t i = 0; i < in_buf_->num_channels(); ++i) {
RTC_CHECK_EQ(in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
msg.input_channel(i).size());
std::memcpy(in_buf_->channels()[i], msg.input_channel(i).data(),
msg.input_channel(i).size());
}
}
if (artificial_nearend_buffer_reader_) {
if (artificial_nearend_buffer_reader_->Read(
artificial_nearend_buf_.get())) {
if (msg.has_input_data()) {
int16_t* fwd_frame_data = fwd_frame_.mutable_data();
for (size_t k = 0; k < in_buf_->num_frames(); ++k) {
fwd_frame_data[k] = rtc::saturated_cast<int16_t>(
fwd_frame_data[k] +
static_cast<int16_t>(32767 *
artificial_nearend_buf_->channels()[0][k]));
}
} else {
for (int i = 0; i < msg.input_channel_size(); ++i) {
for (size_t k = 0; k < in_buf_->num_frames(); ++k) {
in_buf_->channels()[i][k] +=
artificial_nearend_buf_->channels()[0][k];
in_buf_->channels()[i][k] = std::min(
32767.f, std::max(-32768.f, in_buf_->channels()[i][k]));
}
}
}
} else {
if (!artificial_nearend_eof_reported_) {
std::cout << "The artificial nearend file ended before the recording.";
artificial_nearend_eof_reported_ = true;
}
}
}
if (!settings_.use_stream_delay || *settings_.use_stream_delay) {
if (!settings_.stream_delay) {
if (msg.has_delay()) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->set_stream_delay_ms(msg.delay()));
}
} else {
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->set_stream_delay_ms(*settings_.stream_delay));
}
}
if (!settings_.use_ts) {
if (msg.has_keypress()) {
ap_->set_stream_key_pressed(msg.keypress());
}
} else {
ap_->set_stream_key_pressed(*settings_.use_ts);
}
// Level is always logged in AEC dumps.
RTC_CHECK(msg.has_level());
aec_dump_mic_level_ = msg.level();
}
void AecDumpBasedSimulator::VerifyProcessStreamBitExactness(
const webrtc::audioproc::Stream& msg) {
if (bitexact_output_) {
if (interface_used_ == InterfaceType::kFixedInterface) {
bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_);
} else {
bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_);
}
}
}
void AecDumpBasedSimulator::PrepareReverseProcessStreamCall(
const webrtc::audioproc::ReverseStream& msg) {
if (msg.has_data()) {
// Fixed interface processing.
// Verify interface invariance.
RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface ||
interface_used_ == InterfaceType::kNotSpecified);
interface_used_ = InterfaceType::kFixedInterface;
// Populate input buffer.
RTC_CHECK_EQ(sizeof(int16_t) * rev_frame_.samples_per_channel_ *
rev_frame_.num_channels_,
msg.data().size());
memcpy(rev_frame_.mutable_data(), msg.data().data(), msg.data().size());
} else {
// Float interface processing.
// Verify interface invariance.
RTC_CHECK(interface_used_ == InterfaceType::kFloatInterface ||
interface_used_ == InterfaceType::kNotSpecified);
interface_used_ = InterfaceType::kFloatInterface;
RTC_CHECK_EQ(reverse_in_buf_->num_channels(),
static_cast<size_t>(msg.channel_size()));
// Populate input buffer.
for (int i = 0; i < msg.channel_size(); ++i) {
RTC_CHECK_EQ(reverse_in_buf_->num_frames() *
sizeof(*reverse_in_buf_->channels()[i]),
msg.channel(i).size());
std::memcpy(reverse_in_buf_->channels()[i], msg.channel(i).data(),
msg.channel(i).size());
}
}
}
void AecDumpBasedSimulator::Process() {
CreateAudioProcessor();
dump_input_file_ = OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
if (settings_.artificial_nearend_filename) {
std::unique_ptr<WavReader> artificial_nearend_file(
new WavReader(settings_.artificial_nearend_filename->c_str()));
RTC_CHECK_EQ(1, artificial_nearend_file->num_channels())
<< "Only mono files for the artificial nearend are supported, "
"reverted to not using the artificial nearend file";
const int sample_rate_hz = artificial_nearend_file->sample_rate();
artificial_nearend_buffer_reader_.reset(
new ChannelBufferWavReader(std::move(artificial_nearend_file)));
artificial_nearend_buf_.reset(new ChannelBuffer<float>(
rtc::CheckedDivExact(sample_rate_hz, kChunksPerSecond), 1));
}
webrtc::audioproc::Event event_msg;
int num_forward_chunks_processed = 0;
while (ReadMessageFromFile(dump_input_file_, &event_msg)) {
switch (event_msg.type()) {
case webrtc::audioproc::Event::INIT:
RTC_CHECK(event_msg.has_init());
HandleMessage(event_msg.init());
break;
case webrtc::audioproc::Event::STREAM:
RTC_CHECK(event_msg.has_stream());
HandleMessage(event_msg.stream());
++num_forward_chunks_processed;
break;
case webrtc::audioproc::Event::REVERSE_STREAM:
RTC_CHECK(event_msg.has_reverse_stream());
HandleMessage(event_msg.reverse_stream());
break;
case webrtc::audioproc::Event::CONFIG:
RTC_CHECK(event_msg.has_config());
HandleMessage(event_msg.config());
break;
case webrtc::audioproc::Event::RUNTIME_SETTING:
HandleMessage(event_msg.runtime_setting());
break;
case webrtc::audioproc::Event::UNKNOWN_EVENT:
RTC_CHECK(false);
break;
}
}
fclose(dump_input_file_);
DestroyAudioProcessor();
}
void AecDumpBasedSimulator::HandleMessage(
const webrtc::audioproc::Config& msg) {
if (settings_.use_verbose_logging) {
std::cout << "Config 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;
}
if (!settings_.discard_all_settings_in_aecdump) {
if (settings_.use_verbose_logging) {
std::cout << "Setting used in config:" << std::endl;
}
Config config;
AudioProcessing::Config apm_config = ap_->GetConfig();
if (msg.has_aec_enabled() || settings_.use_aec) {
bool enable = settings_.use_aec ? *settings_.use_aec : msg.aec_enabled();
apm_config.echo_canceller.enabled = enable;
if (settings_.use_verbose_logging) {
std::cout << " aec_enabled: " << (enable ? "true" : "false")
<< std::endl;
}
}
if (msg.has_aec_delay_agnostic_enabled() || settings_.use_delay_agnostic) {
bool enable = settings_.use_delay_agnostic
? *settings_.use_delay_agnostic
: msg.aec_delay_agnostic_enabled();
config.Set<DelayAgnostic>(new DelayAgnostic(enable));
if (settings_.use_verbose_logging) {
std::cout << " aec_delay_agnostic_enabled: "
<< (enable ? "true" : "false") << std::endl;
}
}
if (msg.has_aec_drift_compensation_enabled() ||
settings_.use_drift_compensation) {
if (settings_.use_drift_compensation
? *settings_.use_drift_compensation
: msg.aec_drift_compensation_enabled()) {
RTC_LOG(LS_ERROR)
<< "Ignoring deprecated setting: AEC2 drift compensation";
}
}
if (msg.has_aec_extended_filter_enabled() ||
settings_.use_extended_filter) {
bool enable = settings_.use_extended_filter
? *settings_.use_extended_filter
: msg.aec_extended_filter_enabled();
config.Set<ExtendedFilter>(new ExtendedFilter(enable));
if (settings_.use_verbose_logging) {
std::cout << " aec_extended_filter_enabled: "
<< (enable ? "true" : "false") << std::endl;
}
}
if (msg.has_aec_suppression_level() || settings_.aec_suppression_level) {
auto level = static_cast<webrtc::EchoCancellation::SuppressionLevel>(
settings_.aec_suppression_level ? *settings_.aec_suppression_level
: msg.aec_suppression_level());
if (level ==
webrtc::EchoCancellation::SuppressionLevel::kLowSuppression) {
RTC_LOG(LS_ERROR)
<< "Ignoring deprecated setting: AEC2 low suppression";
} else {
apm_config.echo_canceller.legacy_moderate_suppression_level =
(level ==
webrtc::EchoCancellation::SuppressionLevel::kModerateSuppression);
if (settings_.use_verbose_logging) {
std::cout << " aec_suppression_level: " << level << std::endl;
}
}
}
if (msg.has_aecm_enabled() || settings_.use_aecm) {
bool enable =
settings_.use_aecm ? *settings_.use_aecm : msg.aecm_enabled();
apm_config.echo_canceller.enabled |= enable;
apm_config.echo_canceller.mobile_mode = enable;
if (settings_.use_verbose_logging) {
std::cout << " aecm_enabled: " << (enable ? "true" : "false")
<< std::endl;
}
}
if (msg.has_aecm_comfort_noise_enabled() ||
settings_.use_aecm_comfort_noise) {
bool enable = settings_.use_aecm_comfort_noise
? *settings_.use_aecm_comfort_noise
: msg.aecm_comfort_noise_enabled();
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->echo_control_mobile()->enable_comfort_noise(enable));
if (settings_.use_verbose_logging) {
std::cout << " aecm_comfort_noise_enabled: "
<< (enable ? "true" : "false") << std::endl;
}
}
if (msg.has_aecm_routing_mode() || settings_.aecm_routing_mode) {
int routing_mode = settings_.aecm_routing_mode
? *settings_.aecm_routing_mode
: msg.aecm_routing_mode();
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->echo_control_mobile()->set_routing_mode(
static_cast<webrtc::EchoControlMobile::RoutingMode>(
routing_mode)));
if (settings_.use_verbose_logging) {
std::cout << " aecm_routing_mode: " << routing_mode << std::endl;
}
}
if (msg.has_agc_enabled() || settings_.use_agc) {
bool enable = settings_.use_agc ? *settings_.use_agc : msg.agc_enabled();
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->gain_control()->Enable(enable));
if (settings_.use_verbose_logging) {
std::cout << " agc_enabled: " << (enable ? "true" : "false")
<< std::endl;
}
}
if (msg.has_agc_mode() || settings_.agc_mode) {
int mode = settings_.agc_mode ? *settings_.agc_mode : msg.agc_mode();
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->gain_control()->set_mode(
static_cast<webrtc::GainControl::Mode>(mode)));
if (settings_.use_verbose_logging) {
std::cout << " agc_mode: " << mode << std::endl;
}
}
if (msg.has_agc_limiter_enabled() || settings_.use_agc_limiter) {
bool enable = settings_.use_agc_limiter ? *settings_.use_agc_limiter
: msg.agc_limiter_enabled();
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->gain_control()->enable_limiter(enable));
if (settings_.use_verbose_logging) {
std::cout << " agc_limiter_enabled: " << (enable ? "true" : "false")
<< std::endl;
}
}
// TODO(peah): Add support for controlling the Experimental AGC from the
// command line.
if (msg.has_noise_robust_agc_enabled()) {
config.Set<ExperimentalAgc>(
new ExperimentalAgc(msg.noise_robust_agc_enabled()));
if (settings_.use_verbose_logging) {
std::cout << " noise_robust_agc_enabled: "
<< (msg.noise_robust_agc_enabled() ? "true" : "false")
<< std::endl;
}
}
if (msg.has_transient_suppression_enabled() || settings_.use_ts) {
bool enable = settings_.use_ts ? *settings_.use_ts
: msg.transient_suppression_enabled();
config.Set<ExperimentalNs>(new ExperimentalNs(enable));
if (settings_.use_verbose_logging) {
std::cout << " transient_suppression_enabled: "
<< (enable ? "true" : "false") << std::endl;
}
}
if (msg.has_hpf_enabled() || settings_.use_hpf) {
bool enable = settings_.use_hpf ? *settings_.use_hpf : msg.hpf_enabled();
apm_config.high_pass_filter.enabled = enable;
if (settings_.use_verbose_logging) {
std::cout << " hpf_enabled: " << (enable ? "true" : "false")
<< std::endl;
}
}
if (msg.has_ns_enabled() || settings_.use_ns) {
bool enable = settings_.use_ns ? *settings_.use_ns : msg.ns_enabled();
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->noise_suppression()->Enable(enable));
if (settings_.use_verbose_logging) {
std::cout << " ns_enabled: " << (enable ? "true" : "false")
<< std::endl;
}
}
if (msg.has_ns_level() || settings_.ns_level) {
int level = settings_.ns_level ? *settings_.ns_level : msg.ns_level();
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->noise_suppression()->set_level(
static_cast<NoiseSuppression::Level>(level)));
if (settings_.use_verbose_logging) {
std::cout << " ns_level: " << level << std::endl;
}
}
if (msg.has_pre_amplifier_enabled() || settings_.use_pre_amplifier) {
const bool enable = settings_.use_pre_amplifier
? *settings_.use_pre_amplifier
: msg.pre_amplifier_enabled();
apm_config.pre_amplifier.enabled = enable;
apm_config.pre_amplifier.fixed_gain_factor =
settings_.pre_amplifier_gain_factor;
}
if (settings_.use_verbose_logging && msg.has_experiments_description() &&
!msg.experiments_description().empty()) {
std::cout << " experiments not included by default in the simulation: "
<< msg.experiments_description() << std::endl;
}
if (settings_.use_refined_adaptive_filter) {
config.Set<RefinedAdaptiveFilter>(
new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter));
}
if (settings_.use_ed) {
apm_config.residual_echo_detector.enabled = *settings_.use_ed;
}
ap_->ApplyConfig(apm_config);
ap_->SetExtraOptions(config);
}
}
void AecDumpBasedSimulator::HandleMessage(const webrtc::audioproc::Init& msg) {
RTC_CHECK(msg.has_sample_rate());
RTC_CHECK(msg.has_num_input_channels());
RTC_CHECK(msg.has_num_reverse_channels());
RTC_CHECK(msg.has_reverse_sample_rate());
if (settings_.use_verbose_logging) {
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) {
PrepareProcessStreamCall(msg);
ProcessStream(interface_used_ == InterfaceType::kFixedInterface);
VerifyProcessStreamBitExactness(msg);
}
void AecDumpBasedSimulator::HandleMessage(
const webrtc::audioproc::ReverseStream& msg) {
PrepareReverseProcessStreamCall(msg);
ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface);
}
void AecDumpBasedSimulator::HandleMessage(
const webrtc::audioproc::RuntimeSetting& msg) {
RTC_CHECK(ap_.get());
ReplayRuntimeSetting(ap_.get(), msg);
}
} // namespace test
} // namespace webrtc
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