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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 "modules/audio_processing/test/debug_dump_replayer.h"
#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
#include "modules/audio_processing/test/protobuf_utils.h"
#include "modules/audio_processing/test/runtime_setting_util.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
namespace {
void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer,
const StreamConfig& config) {
auto& buffer_ref = *buffer;
if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
buffer_ref->num_channels() != config.num_channels()) {
buffer_ref.reset(
new ChannelBuffer<float>(config.num_frames(), config.num_channels()));
}
}
} // namespace
DebugDumpReplayer::DebugDumpReplayer()
: input_(nullptr), // will be created upon usage.
reverse_(nullptr),
output_(nullptr),
apm_(nullptr),
debug_file_(nullptr) {}
DebugDumpReplayer::~DebugDumpReplayer() {
if (debug_file_)
fclose(debug_file_);
}
bool DebugDumpReplayer::SetDumpFile(const std::string& filename) {
debug_file_ = fopen(filename.c_str(), "rb");
LoadNextMessage();
return debug_file_;
}
// Get next event that has not run.
absl::optional<audioproc::Event> DebugDumpReplayer::GetNextEvent() const {
if (!has_next_event_)
return absl::nullopt;
else
return next_event_;
}
// Run the next event. Returns the event type.
bool DebugDumpReplayer::RunNextEvent() {
if (!has_next_event_)
return false;
switch (next_event_.type()) {
case audioproc::Event::INIT:
OnInitEvent(next_event_.init());
break;
case audioproc::Event::STREAM:
OnStreamEvent(next_event_.stream());
break;
case audioproc::Event::REVERSE_STREAM:
OnReverseStreamEvent(next_event_.reverse_stream());
break;
case audioproc::Event::CONFIG:
OnConfigEvent(next_event_.config());
break;
case audioproc::Event::RUNTIME_SETTING:
OnRuntimeSettingEvent(next_event_.runtime_setting());
break;
case audioproc::Event::UNKNOWN_EVENT:
// We do not expect to receive UNKNOWN event.
RTC_CHECK(false);
return false;
}
LoadNextMessage();
return true;
}
const ChannelBuffer<float>* DebugDumpReplayer::GetOutput() const {
return output_.get();
}
StreamConfig DebugDumpReplayer::GetOutputConfig() const {
return output_config_;
}
// OnInitEvent reset the input/output/reserve channel format.
void DebugDumpReplayer::OnInitEvent(const audioproc::Init& msg) {
RTC_CHECK(msg.has_num_input_channels());
RTC_CHECK(msg.has_output_sample_rate());
RTC_CHECK(msg.has_num_output_channels());
RTC_CHECK(msg.has_reverse_sample_rate());
RTC_CHECK(msg.has_num_reverse_channels());
input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
output_config_ =
StreamConfig(msg.output_sample_rate(), msg.num_output_channels());
reverse_config_ =
StreamConfig(msg.reverse_sample_rate(), msg.num_reverse_channels());
MaybeResetBuffer(&input_, input_config_);
MaybeResetBuffer(&output_, output_config_);
MaybeResetBuffer(&reverse_, reverse_config_);
}
// OnStreamEvent replays an input signal and verifies the output.
void DebugDumpReplayer::OnStreamEvent(const audioproc::Stream& msg) {
// APM should have been created.
RTC_CHECK(apm_.get());
apm_->set_stream_analog_level(msg.level());
RTC_CHECK_EQ(AudioProcessing::kNoError,
apm_->set_stream_delay_ms(msg.delay()));
if (msg.has_keypress()) {
apm_->set_stream_key_pressed(msg.keypress());
} else {
apm_->set_stream_key_pressed(true);
}
RTC_CHECK_EQ(input_config_.num_channels(),
static_cast<size_t>(msg.input_channel_size()));
RTC_CHECK_EQ(input_config_.num_frames() * sizeof(float),
msg.input_channel(0).size());
for (int i = 0; i < msg.input_channel_size(); ++i) {
memcpy(input_->channels()[i], msg.input_channel(i).data(),
msg.input_channel(i).size());
}
RTC_CHECK_EQ(AudioProcessing::kNoError,
apm_->ProcessStream(input_->channels(), input_config_,
output_config_, output_->channels()));
}
void DebugDumpReplayer::OnReverseStreamEvent(
const audioproc::ReverseStream& msg) {
// APM should have been created.
RTC_CHECK(apm_.get());
RTC_CHECK_GT(msg.channel_size(), 0);
RTC_CHECK_EQ(reverse_config_.num_channels(),
static_cast<size_t>(msg.channel_size()));
RTC_CHECK_EQ(reverse_config_.num_frames() * sizeof(float),
msg.channel(0).size());
for (int i = 0; i < msg.channel_size(); ++i) {
memcpy(reverse_->channels()[i], msg.channel(i).data(),
msg.channel(i).size());
}
RTC_CHECK_EQ(
AudioProcessing::kNoError,
apm_->ProcessReverseStream(reverse_->channels(), reverse_config_,
reverse_config_, reverse_->channels()));
}
void DebugDumpReplayer::OnConfigEvent(const audioproc::Config& msg) {
MaybeRecreateApm(msg);
ConfigureApm(msg);
}
void DebugDumpReplayer::OnRuntimeSettingEvent(
const audioproc::RuntimeSetting& msg) {
RTC_CHECK(apm_.get());
ReplayRuntimeSetting(apm_.get(), msg);
}
void DebugDumpReplayer::MaybeRecreateApm(const audioproc::Config& msg) {
// These configurations cannot be changed on the fly.
Config config;
RTC_CHECK(msg.has_aec_delay_agnostic_enabled());
RTC_CHECK(msg.has_aec_extended_filter_enabled());
// We only create APM once, since changes on these fields should not
// happen in current implementation.
if (!apm_.get()) {
apm_.reset(AudioProcessingBuilderForTesting().Create(config));
}
}
void DebugDumpReplayer::ConfigureApm(const audioproc::Config& msg) {
AudioProcessing::Config apm_config;
// AEC2/AECM configs.
RTC_CHECK(msg.has_aec_enabled());
RTC_CHECK(msg.has_aecm_enabled());
apm_config.echo_canceller.enabled = msg.aec_enabled() || msg.aecm_enabled();
apm_config.echo_canceller.mobile_mode = msg.aecm_enabled();
// HPF configs.
RTC_CHECK(msg.has_hpf_enabled());
apm_config.high_pass_filter.enabled = msg.hpf_enabled();
// Preamp configs.
RTC_CHECK(msg.has_pre_amplifier_enabled());
apm_config.pre_amplifier.enabled = msg.pre_amplifier_enabled();
apm_config.pre_amplifier.fixed_gain_factor =
msg.pre_amplifier_fixed_gain_factor();
// NS configs.
RTC_CHECK(msg.has_ns_enabled());
RTC_CHECK(msg.has_ns_level());
apm_config.noise_suppression.enabled = msg.ns_enabled();
apm_config.noise_suppression.level =
static_cast<AudioProcessing::Config::NoiseSuppression::Level>(
msg.ns_level());
// TS configs.
RTC_CHECK(msg.has_transient_suppression_enabled());
apm_config.transient_suppression.enabled =
msg.transient_suppression_enabled();
// AGC configs.
RTC_CHECK(msg.has_agc_enabled());
RTC_CHECK(msg.has_agc_mode());
RTC_CHECK(msg.has_agc_limiter_enabled());
apm_config.gain_controller1.enabled = msg.agc_enabled();
apm_config.gain_controller1.mode =
static_cast<AudioProcessing::Config::GainController1::Mode>(
msg.agc_mode());
apm_config.gain_controller1.enable_limiter = msg.agc_limiter_enabled();
RTC_CHECK(msg.has_noise_robust_agc_enabled());
apm_config.gain_controller1.analog_gain_controller.enabled =
msg.noise_robust_agc_enabled();
apm_->ApplyConfig(apm_config);
}
void DebugDumpReplayer::LoadNextMessage() {
has_next_event_ =
debug_file_ && ReadMessageFromFile(debug_file_, &next_event_);
}
} // namespace test
} // namespace webrtc
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