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

webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc

Index: webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
diff --git a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
new file mode 100644
index 0000000000000000000000000000000000000000..c783ace927f33ce318f412575738fe21bd43294a
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -0,0 +1,464 @@
+/*
+ *  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/aec_dump_based_simulator.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
+#include "webrtc/test/testsupport/trace_to_stderr.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+// Verify output bitexactness for the fixed interface.
+bool VerifyFixedBitExactness(const webrtc::audioproc::Stream& msg,
+                             const AudioFrame& frame) {
+  RTC_CHECK_EQ(
+      sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_,
+      msg.output_data().size());
+  for (size_t k = 0; k < frame.num_channels_ * frame.samples_per_channel_;
+       ++k) {
+    if (msg.output_data().data()[k] != frame.data_[k]) {
+      return false;
+    }
+  }
+  return true;
+}
+
+// Verify output bitexactness for the float interface.
+bool VerifyFloatBitExactness(const webrtc::audioproc::Stream& msg,
+                             const StreamConfig& out_config,
+                             const ChannelBuffer<float>& out_buf) {
+  if (static_cast<size_t>(msg.output_channel_size()) !=
+          out_config.num_channels() ||
+      msg.output_channel(0).size() != out_config.num_frames()) {
+    return false;
+  } else {
+    for (int ch = 0; ch < msg.output_channel_size(); ++ch) {
+      for (size_t sample = 0; sample < out_config.num_frames(); ++sample) {
+        if (msg.output_channel(ch).data()[sample] !=
+            out_buf.channels()[ch][sample]) {
+          return false;
+        }
+      }
+    }
+  }
+  return true;
+}
+
+}  // namespace
+
+void AecDumpBasedSimulator::PrepareProcessStreamCall(
+    const webrtc::audioproc::Stream& msg) {
+  if (msg.has_input_data()) {
+    // Fixed interface processing.
+    // Verify interface invariance.
+    RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface ||
+              interface_used_ == InterfaceType::kNotSpecified);
+    interface_used_ = InterfaceType::kFixedInterface;
+
+    // ProcessStream could have changed this for the output frame.
+    fwd_frame_.num_channels_ = ap_->num_input_channels();
+
+    // Populate input buffer.
+    RTC_CHECK_EQ(sizeof(int16_t) * fwd_frame_.samples_per_channel_ *
+                     fwd_frame_.num_channels_,
+                 msg.input_data().size());
+    memcpy(fwd_frame_.data_, msg.input_data().data(), msg.input_data().size());
+  } else {
+    // Float interface processing.
+    // Verify interface invariance.
+    RTC_CHECK(interface_used_ == InterfaceType::kFloatInterface ||
+              interface_used_ == InterfaceType::kNotSpecified);
+    interface_used_ = InterfaceType::kFloatInterface;
+
+    RTC_CHECK_EQ(in_buf_->num_channels(),
+                 static_cast<size_t>(msg.input_channel_size()));
+
+    // Populate input buffer.
+    for (int i = 0; i < msg.input_channel_size(); ++i) {
+      RTC_CHECK_EQ(in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
+                   msg.input_channel(i).size());
+      std::memcpy(in_buf_->channels()[i], msg.input_channel(i).data(),
+                  msg.input_channel(i).size());
+    }
+  }
+
+  if (!settings_.stream_delay) {
+    if (msg.has_delay()) {
+      RTC_CHECK_EQ(AudioProcessing::kNoError,
+                   ap_->set_stream_delay_ms(msg.delay()));
+    }
+  } else {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->set_stream_delay_ms(*settings_.stream_delay));
+  }
+
+  if (!settings_.stream_drift_samples) {
+    if (msg.has_drift()) {
+      ap_->echo_cancellation()->set_stream_drift_samples(msg.drift());
+    }
+  } else {
+    ap_->echo_cancellation()->set_stream_drift_samples(
+        *settings_.stream_drift_samples);
+  }
+
+  if (msg.has_keypress()) {
+    ap_->set_stream_key_pressed(msg.keypress());
+  }
+}
+
+void AecDumpBasedSimulator::VerifyProcessStreamBitExactness(
+    const webrtc::audioproc::Stream& msg) {
+  if (bitexact_output_) {
+    if (interface_used_ == InterfaceType::kFixedInterface) {
+      bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_);
+    } else {
+      bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_);
+    }
+  }
+}
+
+void AecDumpBasedSimulator::PrepareReverseProcessStreamCall(
+    const webrtc::audioproc::ReverseStream& msg) {
+  if (msg.has_data()) {
+    // Fixed interface processing.
+    // Verify interface invariance.
+    RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface ||
+              interface_used_ == InterfaceType::kNotSpecified);
+    interface_used_ = InterfaceType::kFixedInterface;
+
+    // Populate input buffer.
+    RTC_CHECK_EQ(sizeof(int16_t) * rev_frame_.samples_per_channel_ *
+                     rev_frame_.num_channels_,
+                 msg.data().size());
+    memcpy(rev_frame_.data_, msg.data().data(), msg.data().size());
+  } else {
+    // Float interface processing.
+    // Verify interface invariance.
+    RTC_CHECK(interface_used_ == InterfaceType::kFloatInterface ||
+              interface_used_ == InterfaceType::kNotSpecified);
+    interface_used_ = InterfaceType::kFloatInterface;
+
+    RTC_CHECK_EQ(reverse_in_buf_->num_channels(),
+                 static_cast<size_t>(msg.channel_size()));
+
+    // Populate input buffer.
+    for (int i = 0; i < msg.channel_size(); ++i) {
+      RTC_CHECK_EQ(
+          reverse_in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
+          msg.channel(i).size());
+      std::memcpy(reverse_in_buf_->channels()[i], msg.channel(i).data(),
+                  msg.channel(i).size());
+    }
+  }
+}
+
+void AecDumpBasedSimulator::Process() {
+  std::unique_ptr<test::TraceToStderr> trace_to_stderr;
+  if (settings_.use_verbose_logging) {
+    trace_to_stderr.reset(new test::TraceToStderr(true));
+  }
+
+  CreateAudioProcessor();
+  dump_input_file_ = OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
+
+  webrtc::audioproc::Event event_msg;
+  int num_forward_chunks_processed = 0;
+  while (ReadMessageFromFile(dump_input_file_, &event_msg)) {
+    switch (event_msg.type()) {
+      case webrtc::audioproc::Event::INIT:
+        RTC_CHECK(event_msg.has_init());
+        HandleMessage(event_msg.init());
+        break;
+      case webrtc::audioproc::Event::STREAM:
+        RTC_CHECK(event_msg.has_stream());
+        HandleMessage(event_msg.stream());
+        ++num_forward_chunks_processed;
+        break;
+      case webrtc::audioproc::Event::REVERSE_STREAM:
+        RTC_CHECK(event_msg.has_reverse_stream());
+        HandleMessage(event_msg.reverse_stream());
+        break;
+      case webrtc::audioproc::Event::CONFIG:
+        RTC_CHECK(event_msg.has_config());
+        HandleMessage(event_msg.config());
+        break;
+      default:
+        RTC_CHECK(false);
+    }
+    if (trace_to_stderr) {
+      trace_to_stderr->SetTimeSeconds(
+          num_forward_chunks_processed * 1.f /
+          AudioProcessingSimulator::kChunksPerSecond);
+    }
+  }
+
+  fclose(dump_input_file_);
+
+  DestroyAudioProcessor();
+}
+
+void AecDumpBasedSimulator::HandleMessage(
+    const webrtc::audioproc::Config& msg) {
+  if (settings_.use_verbose_logging) {
+    printf("Config at frame: ");
+    printf(" Forward: %zu\n", get_num_process_stream_calls());
+    printf(" Reverse: %zu\n", get_num_reverse_process_stream_calls());
+  }
+
+  if (!settings_.discard_all_settings_in_aecdump) {
+    if (settings_.use_verbose_logging) {
+      printf("Setting used in config:\n");
+    }
+    Config config;
+
+    if (msg.has_aec_enabled()) {
+      bool enable = settings_.use_aec ? *settings_.use_aec : msg.aec_enabled();
+      RTC_CHECK_EQ(AudioProcessing::kNoError,
+                   ap_->echo_cancellation()->Enable(enable));
+      if (settings_.use_verbose_logging) {
+        printf(" aec_enabled: %s\n", enable ? "true" : "false");
+      }
+    }
+
+    if (msg.has_aec_delay_agnostic_enabled()) {
+      bool enable = settings_.use_delay_agnostic
+                        ? *settings_.use_delay_agnostic
+                        : msg.aec_delay_agnostic_enabled();
+      config.Set<DelayAgnostic>(new DelayAgnostic(enable));
+      if (settings_.use_verbose_logging) {
+        printf(" aec_delay_agnostic_enabled: %s\n", enable ? "true" : "false");
+      }
+    }
+
+    if (msg.has_aec_drift_compensation_enabled()) {
+      bool enable = settings_.use_drift_compensation
+                        ? *settings_.use_drift_compensation
+                        : msg.aec_drift_compensation_enabled();
+      RTC_CHECK_EQ(AudioProcessing::kNoError,
+                   ap_->echo_cancellation()->enable_drift_compensation(enable));
+      if (settings_.use_verbose_logging) {
+        printf(" aec_drift_compensation_enabled: %s\n",
+               enable ? "true" : "false");
+      }
+    }
+
+    if (msg.has_aec_extended_filter_enabled()) {
+      bool enable = settings_.use_extended_filter
+                        ? *settings_.use_extended_filter
+                        : msg.aec_extended_filter_enabled();
+      config.Set<ExtendedFilter>(new ExtendedFilter(enable));
+      if (settings_.use_verbose_logging) {
+        printf(" aec_extended_filter_enabled: %s\n", enable ? "true" : "false");
+      }
+    }
+
+    if (msg.has_aec_suppression_level()) {
+      int level = settings_.aec_suppression_level
+                      ? *settings_.aec_suppression_level
+                      : msg.aec_suppression_level();
+      RTC_CHECK_EQ(
+          AudioProcessing::kNoError,
+          ap_->echo_cancellation()->set_suppression_level(
+              static_cast<webrtc::EchoCancellation::SuppressionLevel>(level)));
+      if (settings_.use_verbose_logging) {
+        printf(" aec_suppression_level: %d\n", level);
+      }
+    }
+
+    if (msg.has_aecm_enabled()) {
+      bool enable =
+          settings_.use_aecm ? *settings_.use_aecm : msg.aecm_enabled();
+      RTC_CHECK_EQ(AudioProcessing::kNoError,
+                   ap_->echo_control_mobile()->Enable(enable));
+      if (settings_.use_verbose_logging) {
+        printf(" aecm_enabled: %s\n", enable ? "true" : "false");
+      }
+    }
+
+    if (msg.has_aecm_comfort_noise_enabled()) {
+      bool enable = settings_.use_aecm_comfort_noise
+                        ? *settings_.use_aecm_comfort_noise
+                        : msg.aecm_comfort_noise_enabled();
+      RTC_CHECK_EQ(AudioProcessing::kNoError,
+                   ap_->echo_control_mobile()->enable_comfort_noise(enable));
+      if (settings_.use_verbose_logging) {
+        printf(" aecm_comfort_noise_enabled: %s\n", enable ? "true" : "false");
+      }
+    }
+
+    if (msg.has_aecm_routing_mode()) {
+      int routing_mode = settings_.aecm_routing_mode
+                             ? *settings_.aecm_routing_mode
+                             : msg.aecm_routing_mode();
+      RTC_CHECK_EQ(AudioProcessing::kNoError,
+                   ap_->echo_control_mobile()->set_routing_mode(
+                       static_cast<webrtc::EchoControlMobile::RoutingMode>(
+                           routing_mode)));
+      if (settings_.use_verbose_logging) {
+        printf(" aecm_routing_mode: %d\n", routing_mode);
+      }
+    }
+
+    if (msg.has_agc_enabled()) {
+      bool enable = settings_.use_agc ? *settings_.use_agc : msg.agc_enabled();
+      RTC_CHECK_EQ(AudioProcessing::kNoError,
+                   ap_->gain_control()->Enable(enable));
+      if (settings_.use_verbose_logging) {
+        printf(" agc_enabled: %s\n", enable ? "true" : "false");
+      }
+    }
+
+    if (msg.has_agc_mode()) {
+      int mode = settings_.agc_mode ? *settings_.agc_mode : msg.agc_mode();
+      RTC_CHECK_EQ(AudioProcessing::kNoError,
+                   ap_->gain_control()->set_mode(
+                       static_cast<webrtc::GainControl::Mode>(mode)));
+      if (settings_.use_verbose_logging) {
+        printf(" agc_mode: %d\n", mode);
+      }
+    }
+
+    if (msg.has_agc_limiter_enabled()) {
+      bool enable = settings_.use_agc_limiter ? *settings_.use_agc_limiter
+                                              : msg.agc_limiter_enabled();
+      RTC_CHECK_EQ(AudioProcessing::kNoError,
+                   ap_->gain_control()->enable_limiter(enable));
+      if (settings_.use_verbose_logging) {
+        printf(" agc_limiter_enabled: %s\n", enable ? "true" : "false");
+      }
+    }
+
+    if (msg.has_noise_robust_agc_enabled()) {
+      bool enable =
+          settings_.use_ts ? *settings_.use_ts : msg.noise_robust_agc_enabled();
+      config.Set<ExperimentalNs>(new ExperimentalNs(enable));
+      if (settings_.use_verbose_logging) {
+        printf(" noise_robust_agc_enabled: %s\n", enable ? "true" : "false");
+      }
+    }
+
+    if (msg.has_hpf_enabled()) {
+      bool enable = settings_.use_hpf ? *settings_.use_hpf : msg.hpf_enabled();
+      RTC_CHECK_EQ(AudioProcessing::kNoError,
+                   ap_->high_pass_filter()->Enable(enable));
+      if (settings_.use_verbose_logging) {
+        printf(" hpf_enabled: %s\n", enable ? "true" : "false");
+      }
+    }
+
+    if (msg.has_ns_enabled()) {
+      bool enable = settings_.use_ns ? *settings_.use_ns : msg.ns_enabled();
+      RTC_CHECK_EQ(AudioProcessing::kNoError,
+                   ap_->noise_suppression()->Enable(enable));
+      if (settings_.use_verbose_logging) {
+        printf(" ns_enabled: %s\n", enable ? "true" : "false");
+      }
+    }
+
+    if (msg.has_ns_level()) {
+      int level = settings_.ns_level ? *settings_.ns_level : msg.ns_level();
+      RTC_CHECK_EQ(AudioProcessing::kNoError,
+                   ap_->noise_suppression()->set_level(
+                       static_cast<NoiseSuppression::Level>(level)));
+      if (settings_.use_verbose_logging) {
+        printf(" ns_level: %d\n", level);
+      }
+    }
+
+    if (settings_.use_verbose_logging && msg.has_experiments_description() &&
+        msg.experiments_description().size() > 0) {
+      printf(" experiments not included by default in the simulation: %s\n",
+             msg.experiments_description().c_str());
+    }
+
+    if (settings_.use_refined_adaptive_filter) {
+      config.Set<RefinedAdaptiveFilter>(
+          new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter));
+    }
+
+    if (settings_.use_aec3) {
+      config.Set<EchoCanceller3>(new EchoCanceller3(*settings_.use_aec3));
+    }
+
+    ap_->SetExtraOptions(config);
+  }
+}
+
+void AecDumpBasedSimulator::HandleMessage(const webrtc::audioproc::Init& msg) {
+  RTC_CHECK(msg.has_sample_rate());
+  RTC_CHECK(msg.has_num_input_channels());
+  RTC_CHECK(msg.has_num_reverse_channels());
+  RTC_CHECK(msg.has_reverse_sample_rate());
+
+  if (settings_.use_verbose_logging) {
+    printf("Init at frame:\n");
+    printf(" Forward: %zu\n", get_num_process_stream_calls());
+    printf(" Reverse: %zu\n", get_num_reverse_process_stream_calls());
+  }
+
+  int num_output_channels;
+  if (settings_.output_num_channels) {
+    num_output_channels = *settings_.output_num_channels;
+  } else {
+    num_output_channels = msg.has_num_output_channels()
+                              ? msg.num_output_channels()
+                              : msg.num_input_channels();
+  }
+
+  int output_sample_rate;
+  if (settings_.output_sample_rate_hz) {
+    output_sample_rate = *settings_.output_sample_rate_hz;
+  } else {
+    output_sample_rate = msg.has_output_sample_rate() ? msg.output_sample_rate()
+                                                      : msg.sample_rate();
+  }
+
+  int num_reverse_output_channels;
+  if (settings_.reverse_output_num_channels) {
+    num_reverse_output_channels = *settings_.reverse_output_num_channels;
+  } else {
+    num_reverse_output_channels = msg.has_num_reverse_output_channels()
+                                      ? msg.num_reverse_output_channels()
+                                      : msg.num_reverse_channels();
+  }
+
+  int reverse_output_sample_rate;
+  if (settings_.reverse_output_sample_rate_hz) {
+    reverse_output_sample_rate = *settings_.reverse_output_sample_rate_hz;
+  } else {
+    reverse_output_sample_rate = msg.has_reverse_output_sample_rate()
+                                     ? msg.reverse_output_sample_rate()
+                                     : msg.reverse_sample_rate();
+  }
+
+  SetupBuffersConfigsOutputs(
+      msg.sample_rate(), output_sample_rate, msg.reverse_sample_rate(),
+      reverse_output_sample_rate, msg.num_input_channels(), num_output_channels,
+      msg.num_reverse_channels(), num_reverse_output_channels);
+}
+
+void AecDumpBasedSimulator::HandleMessage(
+    const webrtc::audioproc::Stream& msg) {
+  PrepareProcessStreamCall(msg);
+  ProcessStream(interface_used_ == InterfaceType::kFixedInterface);
+  VerifyProcessStreamBitExactness(msg);
+}
+
+void AecDumpBasedSimulator::HandleMessage(
+    const webrtc::audioproc::ReverseStream& msg) {
+  PrepareReverseProcessStreamCall(msg);
+  ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface);
+}
+
+}  // namespace test
+}  // namespace webrtc