Extension and refactoring of the audioproc_f tool to be a fully fledged tool for audio processing m…

webrtc/modules/audio_processing/test/wav_based_simulator.cc

+/*
+ *  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 "webrtc/modules/audio_processing/test/wav_based_simulator.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/test/testsupport/trace_to_stderr.h"
+
+namespace webrtc {
+namespace test {
+
+std::vector<WavBasedSimulator::SimulationEventType>
+WavBasedSimulator::GetDefaultEventChain() const {
+  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));
+  }
+
+  call_chain_ = 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);
+    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