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This CL extends the WebRTC testing API to allow audioproc_f -based testing using a pre-created AudioProcessing object. This is an important feature to allow testing any AudioProcessing objects that are injected into WebRTC. Beyond adding this, the CL also changes the simulation code to operate on a scoped_refptr<AudioProcessing> object instead of a std::unique<AudioProcessing> object Bug: webrtc:5298 Change-Id: I70179f19518fc583ad0101bd59c038478a3cc23d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175568 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31319}
191 lines
6.3 KiB
C++
191 lines
6.3 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/wav_based_simulator.h"
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#include <stdio.h>
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#include <iostream>
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#include "modules/audio_processing/test/test_utils.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/system/file_wrapper.h"
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namespace webrtc {
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namespace test {
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std::vector<WavBasedSimulator::SimulationEventType>
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WavBasedSimulator::GetCustomEventChain(const std::string& filename) {
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std::vector<WavBasedSimulator::SimulationEventType> call_chain;
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FileWrapper file_wrapper = FileWrapper::OpenReadOnly(filename.c_str());
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RTC_CHECK(file_wrapper.is_open())
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<< "Could not open the custom call order file, reverting "
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"to using the default call order";
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char c;
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size_t num_read = file_wrapper.Read(&c, sizeof(char));
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while (num_read > 0) {
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switch (c) {
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case 'r':
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call_chain.push_back(SimulationEventType::kProcessReverseStream);
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break;
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case 'c':
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call_chain.push_back(SimulationEventType::kProcessStream);
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break;
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case '\n':
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break;
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default:
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FATAL() << "Incorrect custom call order file, reverting to using the "
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"default call order";
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return WavBasedSimulator::GetDefaultEventChain();
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}
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num_read = file_wrapper.Read(&c, sizeof(char));
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}
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return call_chain;
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}
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WavBasedSimulator::WavBasedSimulator(
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const SimulationSettings& settings,
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rtc::scoped_refptr<AudioProcessing> audio_processing,
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std::unique_ptr<AudioProcessingBuilder> ap_builder)
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: AudioProcessingSimulator(settings,
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std::move(audio_processing),
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std::move(ap_builder)) {
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if (settings_.call_order_input_filename) {
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call_chain_ = WavBasedSimulator::GetCustomEventChain(
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*settings_.call_order_input_filename);
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} else {
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call_chain_ = WavBasedSimulator::GetDefaultEventChain();
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}
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}
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WavBasedSimulator::~WavBasedSimulator() = default;
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std::vector<WavBasedSimulator::SimulationEventType>
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WavBasedSimulator::GetDefaultEventChain() {
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std::vector<WavBasedSimulator::SimulationEventType> call_chain(2);
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call_chain[0] = SimulationEventType::kProcessStream;
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call_chain[1] = SimulationEventType::kProcessReverseStream;
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return call_chain;
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}
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void WavBasedSimulator::PrepareProcessStreamCall() {
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if (settings_.fixed_interface) {
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fwd_frame_.CopyFrom(*in_buf_);
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}
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ap_->set_stream_key_pressed(settings_.use_ts && (*settings_.use_ts));
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if (!settings_.use_stream_delay || *settings_.use_stream_delay) {
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RTC_CHECK_EQ(AudioProcessing::kNoError,
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ap_->set_stream_delay_ms(
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settings_.stream_delay ? *settings_.stream_delay : 0));
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}
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}
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void WavBasedSimulator::PrepareReverseProcessStreamCall() {
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if (settings_.fixed_interface) {
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rev_frame_.CopyFrom(*reverse_in_buf_);
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}
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}
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void WavBasedSimulator::Process() {
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ConfigureAudioProcessor();
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Initialize();
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bool samples_left_to_process = true;
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int call_chain_index = 0;
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int num_forward_chunks_processed = 0;
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while (samples_left_to_process) {
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switch (call_chain_[call_chain_index]) {
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case SimulationEventType::kProcessStream:
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samples_left_to_process = HandleProcessStreamCall();
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++num_forward_chunks_processed;
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break;
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case SimulationEventType::kProcessReverseStream:
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if (settings_.reverse_input_filename) {
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samples_left_to_process = HandleProcessReverseStreamCall();
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}
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break;
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default:
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RTC_CHECK(false);
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}
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call_chain_index = (call_chain_index + 1) % call_chain_.size();
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}
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DetachAecDump();
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}
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bool WavBasedSimulator::HandleProcessStreamCall() {
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bool samples_left_to_process = buffer_reader_->Read(in_buf_.get());
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if (samples_left_to_process) {
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PrepareProcessStreamCall();
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ProcessStream(settings_.fixed_interface);
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}
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return samples_left_to_process;
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}
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bool WavBasedSimulator::HandleProcessReverseStreamCall() {
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bool samples_left_to_process =
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reverse_buffer_reader_->Read(reverse_in_buf_.get());
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if (samples_left_to_process) {
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PrepareReverseProcessStreamCall();
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ProcessReverseStream(settings_.fixed_interface);
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}
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return samples_left_to_process;
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}
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void WavBasedSimulator::Initialize() {
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std::unique_ptr<WavReader> in_file(
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new WavReader(settings_.input_filename->c_str()));
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int input_sample_rate_hz = in_file->sample_rate();
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int input_num_channels = in_file->num_channels();
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buffer_reader_.reset(new ChannelBufferWavReader(std::move(in_file)));
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int output_sample_rate_hz = settings_.output_sample_rate_hz
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? *settings_.output_sample_rate_hz
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: input_sample_rate_hz;
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int output_num_channels = settings_.output_num_channels
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? *settings_.output_num_channels
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: input_num_channels;
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int reverse_sample_rate_hz = 48000;
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int reverse_num_channels = 1;
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int reverse_output_sample_rate_hz = 48000;
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int reverse_output_num_channels = 1;
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if (settings_.reverse_input_filename) {
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std::unique_ptr<WavReader> reverse_in_file(
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new WavReader(settings_.reverse_input_filename->c_str()));
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reverse_sample_rate_hz = reverse_in_file->sample_rate();
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reverse_num_channels = reverse_in_file->num_channels();
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reverse_buffer_reader_.reset(
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new ChannelBufferWavReader(std::move(reverse_in_file)));
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reverse_output_sample_rate_hz =
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settings_.reverse_output_sample_rate_hz
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? *settings_.reverse_output_sample_rate_hz
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: reverse_sample_rate_hz;
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reverse_output_num_channels = settings_.reverse_output_num_channels
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? *settings_.reverse_output_num_channels
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: reverse_num_channels;
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}
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SetupBuffersConfigsOutputs(
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input_sample_rate_hz, output_sample_rate_hz, reverse_sample_rate_hz,
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reverse_output_sample_rate_hz, input_num_channels, output_num_channels,
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reverse_num_channels, reverse_output_num_channels);
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}
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} // namespace test
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} // namespace webrtc
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