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

webrtc/modules/audio_processing/test/audio_processing_simulator.cc

Index: webrtc/modules/audio_processing/test/audio_processing_simulator.cc
diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
new file mode 100644
index 0000000000000000000000000000000000000000..6cb614db32f45240099a7484c5c8d2b6d21bc1f6
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
@@ -0,0 +1,334 @@
+/*
+ *  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/audio_processing_simulator.h"
+
+#include <algorithm>
+#include <iostream>
+#include <sstream>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/stringutils.h"
+#include "webrtc/common_audio/include/audio_util.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) {
+  RTC_CHECK_EQ(src.num_channels_, dest->num_channels());
+  RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames());
+  // Copy the data from the input buffer.
+  std::vector<float> tmp(src.samples_per_channel_ * src.num_channels_);
+  S16ToFloat(src.data_, tmp.size(), tmp.data());
+  Deinterleave(tmp.data(), src.samples_per_channel_, src.num_channels_,
+               dest->channels());
+}
+
+std::string GetIndexedOutputWavFilename(const std::string& wav_name,
+                                        int counter) {
+  std::stringstream ss;
+  ss << wav_name.substr(0, wav_name.size() - 4) << "_" << counter
+     << wav_name.substr(wav_name.size() - 4);
+  return ss.str();
+}
+
+}  // namespace
+
+void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
+  RTC_CHECK_EQ(src.num_channels(), dest->num_channels_);
+  RTC_CHECK_EQ(src.num_frames(), dest->samples_per_channel_);
+  for (size_t ch = 0; ch < dest->num_channels_; ++ch) {
+    for (size_t sample = 0; sample < dest->samples_per_channel_; ++sample) {
+      dest->data_[sample * dest->num_channels_ + ch] =
+          src.channels()[ch][sample] * 32767;
+    }
+  }
+}
+
+AudioProcessingSimulator::ScopedTimer::~ScopedTimer() {
+  int64_t interval = rtc::TimeNanos() - start_time_;
+  proc_time_->sum += interval;
+  proc_time_->max = std::max(proc_time_->max, interval);
+  proc_time_->min = std::min(proc_time_->min, interval);
+}
+
+void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
+  if (fixed_interface) {
+    {
+      const auto st = ScopedTimer(mutable_proc_time());
+      RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->ProcessStream(&fwd_frame_));
+    }
+    CopyFromAudioFrame(fwd_frame_, out_buf_.get());
+  } else {
+    const auto st = ScopedTimer(mutable_proc_time());
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->ProcessStream(in_buf_->channels(), in_config_,
+                                    out_config_, out_buf_->channels()));
+  }
+
+  if (buffer_writer_) {
+    buffer_writer_->Write(*out_buf_);
+  }
+
+  ++num_process_stream_calls_;
+}
+
+void AudioProcessingSimulator::ProcessReverseStream(bool fixed_interface) {
+  if (fixed_interface) {
+    const auto st = ScopedTimer(mutable_proc_time());
+    RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->ProcessStream(&rev_frame_));
+    CopyFromAudioFrame(rev_frame_, reverse_out_buf_.get());
+
+  } else {
+    const auto st = ScopedTimer(mutable_proc_time());
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->ProcessReverseStream(
+                     reverse_in_buf_->channels(), reverse_in_config_,
+                     reverse_out_config_, reverse_out_buf_->channels()));
+  }
+
+  if (reverse_buffer_writer_) {
+    reverse_buffer_writer_->Write(*reverse_out_buf_);
+  }
+
+  ++num_reverse_process_stream_calls_;
+}
+
+void AudioProcessingSimulator::SetupBuffersConfigsOutputs(
+    int input_sample_rate_hz,
+    int output_sample_rate_hz,
+    int reverse_input_sample_rate_hz,
+    int reverse_output_sample_rate_hz,
+    int input_num_channels,
+    int output_num_channels,
+    int reverse_input_num_channels,
+    int reverse_output_num_channels) {
+  in_config_ = StreamConfig(input_sample_rate_hz, input_num_channels);
+  in_buf_.reset(new ChannelBuffer<float>(
+      rtc::CheckedDivExact(input_sample_rate_hz, kChunksPerSecond),
+      input_num_channels));
+
+  reverse_in_config_ =
+      StreamConfig(reverse_input_sample_rate_hz, reverse_input_num_channels);
+  reverse_in_buf_.reset(new ChannelBuffer<float>(
+      rtc::CheckedDivExact(reverse_input_sample_rate_hz, kChunksPerSecond),
+      reverse_input_num_channels));
+
+  out_config_ = StreamConfig(output_sample_rate_hz, output_num_channels);
+  out_buf_.reset(new ChannelBuffer<float>(
+      rtc::CheckedDivExact(output_sample_rate_hz, kChunksPerSecond),
+      output_num_channels));
+
+  reverse_out_config_ =
+      StreamConfig(reverse_output_sample_rate_hz, reverse_output_num_channels);
+  reverse_out_buf_.reset(new ChannelBuffer<float>(
+      rtc::CheckedDivExact(reverse_output_sample_rate_hz, kChunksPerSecond),
+      reverse_output_num_channels));
+
+  fwd_frame_.sample_rate_hz_ = input_sample_rate_hz;
+  fwd_frame_.samples_per_channel_ =
+      rtc::CheckedDivExact(fwd_frame_.sample_rate_hz_, kChunksPerSecond);
+  fwd_frame_.num_channels_ = input_num_channels;
+
+  rev_frame_.sample_rate_hz_ = reverse_input_sample_rate_hz;
+  rev_frame_.samples_per_channel_ =
+      rtc::CheckedDivExact(rev_frame_.sample_rate_hz_, kChunksPerSecond);
+  rev_frame_.num_channels_ = reverse_input_num_channels;
+
+  if (settings_.use_verbose_logging) {
+    std::cout << "Sample rates:" << std::endl;
+    std::cout << " Forward input: " << input_sample_rate_hz << std::endl;
+    std::cout << " Forward output: " << output_sample_rate_hz << std::endl;
+    std::cout << " Reverse input: " << reverse_input_sample_rate_hz
+              << std::endl;
+    std::cout << " Reverse output: " << reverse_output_sample_rate_hz
+              << std::endl;
+    std::cout << "Number of channels: " << std::endl;
+    std::cout << " Forward input: " << input_num_channels << std::endl;
+    std::cout << " Forward output: " << output_num_channels << std::endl;
+    std::cout << " Reverse input: " << reverse_input_num_channels << std::endl;
+    std::cout << " Reverse output: " << reverse_output_num_channels
+              << std::endl;
+  }
+
+  SetupOutput();
+}
+
+void AudioProcessingSimulator::SetupOutput() {
+  if (settings_.output_filename) {
+    std::string filename;
+    if (settings_.store_intermediate_output) {
+      filename = GetIndexedOutputWavFilename(*settings_.output_filename,
+                                             output_reset_counter_);
+    } else {
+      filename = *settings_.output_filename;
+    }
+
+    std::unique_ptr<WavWriter> out_file(
+        new WavWriter(filename, out_config_.sample_rate_hz(),
+                      static_cast<size_t>(out_config_.num_channels())));
+    buffer_writer_.reset(new ChannelBufferWavWriter(std::move(out_file)));
+  }
+
+  if (settings_.reverse_output_filename) {
+    std::string filename;
+    if (settings_.store_intermediate_output) {
+      filename = GetIndexedOutputWavFilename(*settings_.reverse_output_filename,
+                                             output_reset_counter_);
+    } else {
+      filename = *settings_.reverse_output_filename;
+    }
+
+    std::unique_ptr<WavWriter> reverse_out_file(
+        new WavWriter(filename, reverse_out_config_.sample_rate_hz(),
+                      static_cast<size_t>(reverse_out_config_.num_channels())));
+    reverse_buffer_writer_.reset(
+        new ChannelBufferWavWriter(std::move(reverse_out_file)));
+  }
+
+  ++output_reset_counter_;
+}
+
+void AudioProcessingSimulator::DestroyAudioProcessor() {
+  if (settings_.aec_dump_output_filename) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->StopDebugRecording());
+  }
+}
+
+void AudioProcessingSimulator::CreateAudioProcessor() {
+  Config config;
+  if (settings_.use_bf && *settings_.use_bf) {
+    config.Set<Beamforming>(new Beamforming(
+        true, ParseArrayGeometry(*settings_.microphone_positions),
+        SphericalPointf(DegreesToRadians(settings_.target_angle_degrees), 0.f,
+                        1.f)));
+  }
+  if (settings_.use_ts) {
+    config.Set<ExperimentalNs>(new ExperimentalNs(*settings_.use_ts));
+  }
+  if (settings_.use_ie) {
+    config.Set<Intelligibility>(new Intelligibility(*settings_.use_ie));
+  }
+  if (settings_.use_aec3) {
+    config.Set<EchoCanceller3>(new EchoCanceller3(*settings_.use_aec3));
+  }
+  if (settings_.use_refined_adaptive_filter) {
+    config.Set<RefinedAdaptiveFilter>(
+        new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter));
+  }
+  config.Set<ExtendedFilter>(new ExtendedFilter(
+      !settings_.use_extended_filter || *settings_.use_extended_filter));
+  config.Set<DelayAgnostic>(new DelayAgnostic(!settings_.use_delay_agnostic ||
+                                              *settings_.use_delay_agnostic));
+
+  ap_.reset(AudioProcessing::Create(config));
+
+  if (settings_.use_aec) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->echo_cancellation()->Enable(*settings_.use_aec));
+  }
+  if (settings_.use_aecm) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->echo_control_mobile()->Enable(*settings_.use_aecm));
+  }
+  if (settings_.use_agc) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->gain_control()->Enable(*settings_.use_agc));
+  }
+  if (settings_.use_hpf) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->high_pass_filter()->Enable(*settings_.use_hpf));
+  }
+  if (settings_.use_ns) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->noise_suppression()->Enable(*settings_.use_ns));
+  }
+  if (settings_.use_le) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->level_estimator()->Enable(*settings_.use_le));
+  }
+  if (settings_.use_vad) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->voice_detection()->Enable(*settings_.use_vad));
+  }
+  if (settings_.use_agc_limiter) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->gain_control()->enable_limiter(
+                                                *settings_.use_agc_limiter));
+  }
+  if (settings_.agc_target_level) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->gain_control()->set_target_level_dbfs(
+                     *settings_.agc_target_level));
+  }
+
+  if (settings_.agc_mode) {
+    RTC_CHECK_EQ(
+        AudioProcessing::kNoError,
+        ap_->gain_control()->set_mode(
+            static_cast<webrtc::GainControl::Mode>(*settings_.agc_mode)));
+  }
+
+  if (settings_.use_drift_compensation) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->echo_cancellation()->enable_drift_compensation(
+                     *settings_.use_drift_compensation));
+  }
+
+  if (settings_.aec_suppression_level) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->echo_cancellation()->set_suppression_level(
+                     static_cast<webrtc::EchoCancellation::SuppressionLevel>(
+                         *settings_.aec_suppression_level)));
+  }
+
+  if (settings_.aecm_routing_mode) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->echo_control_mobile()->set_routing_mode(
+                     static_cast<webrtc::EchoControlMobile::RoutingMode>(
+                         *settings_.aecm_routing_mode)));
+  }
+
+  if (settings_.use_aecm_comfort_noise) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->echo_control_mobile()->enable_comfort_noise(
+                     *settings_.use_aecm_comfort_noise));
+  }
+
+  if (settings_.vad_likelihood) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->voice_detection()->set_likelihood(
+                     static_cast<webrtc::VoiceDetection::Likelihood>(
+                         *settings_.vad_likelihood)));
+  }
+  if (settings_.ns_level) {
+    RTC_CHECK_EQ(
+        AudioProcessing::kNoError,
+        ap_->noise_suppression()->set_level(
+            static_cast<NoiseSuppression::Level>(*settings_.ns_level)));
+  }
+
+  if (settings_.use_ts) {
+    ap_->set_stream_key_pressed(*settings_.use_ts);
+  }
+
+  if (settings_.aec_dump_output_filename) {
+    size_t kMaxFilenameSize = AudioProcessing::kMaxFilenameSize;
+    RTC_CHECK_LE(settings_.aec_dump_output_filename->size(), kMaxFilenameSize);
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->StartDebugRecording(
+                     settings_.aec_dump_output_filename->c_str(), -1));
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc