blob: a80ad4d46004499ec50d54be6a4221ac8e79f4e2 [file] [log] [blame]
/*
* 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/wav_based_simulator.h"
#include <stdio.h>
#include <iostream>
#include "modules/audio_processing/test/test_utils.h"
#include "rtc_base/checks.h"
#include "test/testsupport/trace_to_stderr.h"
namespace webrtc {
namespace test {
std::vector<WavBasedSimulator::SimulationEventType>
WavBasedSimulator::GetCustomEventChain(const std::string& filename) {
std::vector<WavBasedSimulator::SimulationEventType> call_chain;
FILE* stream = OpenFile(filename.c_str(), "r");
RTC_CHECK(stream) << "Could not open the custom call order file, reverting "
"to using the default call order";
char c;
size_t num_read = fread(&c, sizeof(char), 1, stream);
while (num_read > 0) {
switch (c) {
case 'r':
call_chain.push_back(SimulationEventType::kProcessReverseStream);
break;
case 'c':
call_chain.push_back(SimulationEventType::kProcessStream);
break;
case '\n':
break;
default:
FATAL() << "Incorrect custom call order file, reverting to using the "
"default call order";
fclose(stream);
return WavBasedSimulator::GetDefaultEventChain();
}
num_read = fread(&c, sizeof(char), 1, stream);
}
fclose(stream);
return call_chain;
}
WavBasedSimulator::WavBasedSimulator(const SimulationSettings& settings)
: AudioProcessingSimulator(settings) {}
WavBasedSimulator::~WavBasedSimulator() = default;
std::vector<WavBasedSimulator::SimulationEventType>
WavBasedSimulator::GetDefaultEventChain() {
std::vector<WavBasedSimulator::SimulationEventType> call_chain(2);
call_chain[0] = SimulationEventType::kProcessStream;
call_chain[1] = SimulationEventType::kProcessReverseStream;
return call_chain;
}
void WavBasedSimulator::PrepareProcessStreamCall() {
if (settings_.fixed_interface) {
CopyToAudioFrame(*in_buf_, &fwd_frame_);
}
ap_->set_stream_key_pressed(settings_.use_ts && (*settings_.use_ts));
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->set_stream_delay_ms(
settings_.stream_delay ? *settings_.stream_delay : 0));
ap_->echo_cancellation()->set_stream_drift_samples(
settings_.stream_drift_samples ? *settings_.stream_drift_samples : 0);
RTC_CHECK_EQ(AudioProcessing::kNoError,
ap_->gain_control()->set_stream_analog_level(
last_specified_microphone_level_));
}
void WavBasedSimulator::PrepareReverseProcessStreamCall() {
if (settings_.fixed_interface) {
CopyToAudioFrame(*reverse_in_buf_, &rev_frame_);
}
}
void WavBasedSimulator::Process() {
std::unique_ptr<test::TraceToStderr> trace_to_stderr;
if (settings_.use_verbose_logging) {
trace_to_stderr.reset(new test::TraceToStderr(true));
}
if (settings_.custom_call_order_filename) {
call_chain_ = WavBasedSimulator::GetCustomEventChain(
*settings_.custom_call_order_filename);
} else {
call_chain_ = WavBasedSimulator::GetDefaultEventChain();
}
CreateAudioProcessor();
Initialize();
bool samples_left_to_process = true;
int call_chain_index = 0;
int num_forward_chunks_processed = 0;
const int kOneBykChunksPerSecond =
1.f / AudioProcessingSimulator::kChunksPerSecond;
while (samples_left_to_process) {
switch (call_chain_[call_chain_index]) {
case SimulationEventType::kProcessStream:
samples_left_to_process = HandleProcessStreamCall();
++num_forward_chunks_processed;
break;
case SimulationEventType::kProcessReverseStream:
if (settings_.reverse_input_filename) {
samples_left_to_process = HandleProcessReverseStreamCall();
}
break;
default:
RTC_CHECK(false);
}
call_chain_index = (call_chain_index + 1) % call_chain_.size();
if (trace_to_stderr) {
trace_to_stderr->SetTimeSeconds(num_forward_chunks_processed *
kOneBykChunksPerSecond);
}
}
DestroyAudioProcessor();
}
bool WavBasedSimulator::HandleProcessStreamCall() {
bool samples_left_to_process = buffer_reader_->Read(in_buf_.get());
if (samples_left_to_process) {
PrepareProcessStreamCall();
ProcessStream(settings_.fixed_interface);
// Call stream analog level to ensure that any side-effects are triggered.
(void)ap_->gain_control()->stream_analog_level();
last_specified_microphone_level_ =
ap_->gain_control()->stream_analog_level();
}
return samples_left_to_process;
}
bool WavBasedSimulator::HandleProcessReverseStreamCall() {
bool samples_left_to_process =
reverse_buffer_reader_->Read(reverse_in_buf_.get());
if (samples_left_to_process) {
PrepareReverseProcessStreamCall();
ProcessReverseStream(settings_.fixed_interface);
}
return samples_left_to_process;
}
void WavBasedSimulator::Initialize() {
std::unique_ptr<WavReader> in_file(
new WavReader(settings_.input_filename->c_str()));
int input_sample_rate_hz = in_file->sample_rate();
int input_num_channels = in_file->num_channels();
buffer_reader_.reset(new ChannelBufferWavReader(std::move(in_file)));
int output_sample_rate_hz = settings_.output_sample_rate_hz
? *settings_.output_sample_rate_hz
: input_sample_rate_hz;
int output_num_channels = settings_.output_num_channels
? *settings_.output_num_channels
: input_num_channels;
int reverse_sample_rate_hz = 48000;
int reverse_num_channels = 1;
int reverse_output_sample_rate_hz = 48000;
int reverse_output_num_channels = 1;
if (settings_.reverse_input_filename) {
std::unique_ptr<WavReader> reverse_in_file(
new WavReader(settings_.reverse_input_filename->c_str()));
reverse_sample_rate_hz = reverse_in_file->sample_rate();
reverse_num_channels = reverse_in_file->num_channels();
reverse_buffer_reader_.reset(
new ChannelBufferWavReader(std::move(reverse_in_file)));
reverse_output_sample_rate_hz =
settings_.reverse_output_sample_rate_hz
? *settings_.reverse_output_sample_rate_hz
: reverse_sample_rate_hz;
reverse_output_num_channels = settings_.reverse_output_num_channels
? *settings_.reverse_output_num_channels
: reverse_num_channels;
}
SetupBuffersConfigsOutputs(
input_sample_rate_hz, output_sample_rate_hz, reverse_sample_rate_hz,
reverse_output_sample_rate_hz, input_num_channels, output_num_channels,
reverse_num_channels, reverse_output_num_channels);
}
} // namespace test
} // namespace webrtc