Adding APM configuration in AEC dump.

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 24 matching lines @@
 #include "webrtc/modules/audio_processing/noise_suppression_impl.h"
 #include "webrtc/modules/audio_processing/processing_component.h"
 #include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
 #include "webrtc/modules/audio_processing/voice_detection_impl.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/file_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/metrics.h"
 
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-// Files generated at build-time by the protobuf compiler.
-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
-#else
-#include "webrtc/audio_processing/debug.pb.h"
-#endif
-#endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
-
 #define RETURN_ON_ERR(expr) \
   do {                      \
     int err = (expr);       \
     if (err != kNoError) {  \
       return err;           \
     }                       \
   } while (0)
 
 namespace webrtc {
 namespace {
@@ skipping 123 matching lines @@
       echo_control_mobile_(NULL),
       gain_control_(NULL),
       high_pass_filter_(NULL),
       level_estimator_(NULL),
       noise_suppression_(NULL),
       voice_detection_(NULL),
       crit_(CriticalSectionWrapper::CreateCriticalSection()),
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
       debug_file_(FileWrapper::Create()),
       event_msg_(new audioproc::Event()),
+      config_(new audioproc::Config()),
 #endif
       api_format_({{{kSampleRate16kHz, 1, false},
                     {kSampleRate16kHz, 1, false},
                     {kSampleRate16kHz, 1, false},
                     {kSampleRate16kHz, 1, false}}}),
       fwd_proc_format_(kSampleRate16kHz),
       rev_proc_format_(kSampleRate16kHz, 1),
       split_rate_(kSampleRate16kHz),
       stream_delay_ms_(0),
       delay_offset_ms_(0),
@@ skipping 36 matching lines @@
 
   noise_suppression_ = new NoiseSuppressionImpl(this, crit_);
   component_list_.push_back(noise_suppression_);
 
   voice_detection_ = new VoiceDetectionImpl(this, crit_);
   component_list_.push_back(voice_detection_);
 
   gain_control_for_new_agc_.reset(new GainControlForNewAgc(gain_control_));
 
   SetExtraOptions(config);
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  UpdateCurrentConfig();
+#endif
 }
 
 AudioProcessingImpl::~AudioProcessingImpl() {
   {
     CriticalSectionScoped crit_scoped(crit_);
     // Depends on gain_control_ and gain_control_for_new_agc_.
     agc_manager_.reset();
     // Depends on gain_control_.
     gain_control_for_new_agc_.reset();
     while (!component_list_.empty()) {
@@ skipping 290 matching lines @@
 
   ProcessingConfig processing_config = api_format_;
   processing_config.input_stream() = input_config;
   processing_config.output_stream() = output_config;
 
   RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
   assert(processing_config.input_stream().num_frames() ==
          api_format_.input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  if (debug_file_->Open() && UpdateCurrentConfig()) {
+    RETURN_ON_ERR(WriteConfigMessage());
+  }
+
   if (debug_file_->Open()) {
     event_msg_->set_type(audioproc::Event::STREAM);
     audioproc::Stream* msg = event_msg_->mutable_stream();
     const size_t channel_size =
         sizeof(float) * api_format_.input_stream().num_frames();
     for (int i = 0; i < api_format_.input_stream().num_channels(); ++i)
       msg->add_input_channel(src[i], channel_size);
   }
 #endif
 
@@ skipping 366 matching lines @@
     if (debug_file_->CloseFile() == -1) {
       return kFileError;
     }
   }
 
   if (debug_file_->OpenFile(filename, false) == -1) {
     debug_file_->CloseFile();
     return kFileError;
   }
 
-  int err = WriteInitMessage();
+  UpdateCurrentConfig();
+  int err = WriteConfigMessage();
   if (err != kNoError) {
     return err;
   }
+
+  err = WriteInitMessage();
+  if (err != kNoError) {
+    return err;
+  }
   return kNoError;
 #else
   return kUnsupportedFunctionError;
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
 int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
   CriticalSectionScoped crit_scoped(crit_);
 
   if (handle == NULL) {
     return kNullPointerError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // Stop any ongoing recording.
   if (debug_file_->Open()) {
     if (debug_file_->CloseFile() == -1) {
       return kFileError;
     }
   }
 
   if (debug_file_->OpenFromFileHandle(handle, true, false) == -1) {
     return kFileError;
   }
 
-  int err = WriteInitMessage();
+  UpdateCurrentConfig();
+  int err = WriteConfigMessage();
   if (err != kNoError) {
     return err;
   }
+
+  err = WriteInitMessage();
+  if (err != kNoError) {
+    return err;
+  }
   return kNoError;
 #else
   return kUnsupportedFunctionError;
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
 int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
     rtc::PlatformFile handle) {
   FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
   return StartDebugRecording(stream);
@@ skipping 256 matching lines @@
   msg->set_output_sample_rate(api_format_.output_stream().sample_rate_hz());
   // TODO(ekmeyerson): Add reverse output fields to event_msg_.
 
   int err = WriteMessageToDebugFile();
   if (err != kNoError) {
     return err;
   }
 
   return kNoError;
 }
+
+bool AudioProcessingImpl::UpdateCurrentConfig() {

[peah-webrtc, 2015/10/02 08:34:27]

This function became very long (for good reasons) but I don't think the scope
limiting and all the if-statements is as they should. I'd rather implement this
as something like:

bool AudioProcessingImpl::UpdateCurrentConfig() {
  const bool aec_drift_compensation =
echo_cancellation_->is_drift_compensation_enabled();
  const bool aec_extended_filter =
echo_cancellation_->is_extended_filter_enabled();
  const int aec_suppression_level =
static_cast<int>(echo_cancellation_->suppression_level());
  ...

  bool changed = false;
  changed |= aec_drift_compensation != config_->aec_drift_compensation();
  changed |= aec_extended_filter != config_->aec_extended_filter();
  changed |= aec_suppression_level != config_->aec_suppression_level();
  ...


  if (changed) {
    config_->set_aec_drift_compensation(aec_drift_compensation);
    config_->set_aec_extended_filter(aec_extended_filter);
    config_->set_aec_suppression_level(aec_suppression_level);
    ...
  }

  return changed;
}

I think that is more readable and more compact (and hence less error prone) and
also more efficient as the fewer if-statements makes it more pipeline-able and
more efficient. It uses more stack space but that should not be a problem.

[minyue-webrtc, 2015/10/02 09:01:42]

also goes away per Andrew's suggestion

+  bool changed = false;
+
+  // Acoustic echo canceler.
+  {
+    const bool value = echo_cancellation_->is_enabled();
+    if (!config_->has_aec_enabled() || value != config_->aec_enabled()) {
+      config_->set_aec_enabled(value);
+      changed = true;
+    }
+  }
+  {
+    const bool value = echo_cancellation_->is_delay_agnostic_enabled();
+    if (!config_->has_aec_delay_agnostic() ||
+        value != config_->aec_delay_agnostic()) {
+      config_->set_aec_delay_agnostic(value);
+      changed = true;
+    }
+  }
+  {
+    const bool value = echo_cancellation_->is_drift_compensation_enabled();
+    if (!config_->has_aec_drift_compensation() ||
+        value != config_->aec_drift_compensation()) {
+      config_->set_aec_drift_compensation(value);
+      changed = true;
+    }
+  }
+  {
+    const bool value = echo_cancellation_->is_extended_filter_enabled();
+    if (!config_->has_aec_extended_filter() ||
+        value != config_->aec_extended_filter()) {
+      config_->set_aec_extended_filter(value);
+      changed = true;
+    }
+  }
+  {
+    const int value = static_cast<int>(echo_cancellation_->suppression_level());
+    if (!config_->has_aec_suppression_level() ||
+        value != config_->aec_suppression_level()) {
+      config_->set_aec_suppression_level(value);
+      changed = true;
+    }
+  }
+
+  // Mobile AEC.
+  {
+    const bool value = echo_control_mobile_->is_enabled();
+    if (!config_->has_aecm_enabled() || value != config_->aecm_enabled()) {
+      config_->set_aecm_enabled(value);
+      changed = true;
+    }
+  }
+  {
+    const bool value = echo_control_mobile_->is_comfort_noise_enabled();
+    if (!config_->has_aecm_comfort_noise() ||
+        value != config_->aecm_comfort_noise()) {
+      config_->set_aecm_comfort_noise(value);
+      changed = true;
+    }
+  }
+  {
+    const int value = static_cast<int>(echo_control_mobile_->routing_mode());
+    if (!config_->has_aecm_routing_mode() ||
+        value != config_->aecm_routing_mode()) {
+      config_->set_aecm_routing_mode(value);
+      changed = true;
+    }
+  }
+
+  // Automatic gain controller.
+  {
+    const bool value = gain_control_->is_enabled();
+    if (!config_->has_agc_enabled() || value != config_->agc_enabled()) {
+      config_->set_agc_enabled(value);
+      changed = true;
+    }
+  }
+  {
+    const bool value = use_new_agc_;
+    if (!config_->has_agc_experiment() || value != config_->agc_experiment()) {
+      config_->set_agc_experiment(value);
+      changed = true;
+    }
+  }
+  {
+    const int value = static_cast<int>(gain_control_->mode());
+    if (!config_->has_agc_mode() || value != config_->agc_mode()) {
+      config_->set_agc_mode(value);
+      changed = true;
+    }
+  }
+  {
+    const bool value = gain_control_->is_limiter_enabled();
+    if (!config_->has_agc_limiter() || value != config_->agc_limiter()) {
+      config_->set_agc_limiter(value);
+      changed = true;
+    }
+  }
+
+  // High pass filter.
+  {
+    const bool value = high_pass_filter_->is_enabled();
+    if (!config_->has_hpf_enabled() || value != config_->hpf_enabled()) {
+      config_->set_hpf_enabled(value);
+      changed = true;
+    }
+  }
+
+  // Noise suppression.
+  {
+    const bool value = noise_suppression_->is_enabled();
+    if (!config_->has_ns_enabled() || value != config_->ns_enabled()) {
+      config_->set_ns_enabled(value);
+      changed = true;
+    }
+  }
+  {
+    const bool value = transient_suppressor_enabled_;
+    if (!config_->has_ns_experiment() || value != config_->ns_experiment()) {
+      config_->set_ns_experiment(value);
+      changed = true;
+    }
+  }
+  {
+    const int value = static_cast<int>(noise_suppression_->level());
+    if (!config_->has_ns_level() || value != config_->ns_level()) {
+      config_->set_ns_level(value);
+      changed = true;
+    }
+  }
+
+  return changed;
+}
+
+int AudioProcessingImpl::WriteConfigMessage() {
+  event_msg_->set_type(audioproc::Event::CONFIG);
+  event_msg_->mutable_config()->CopyFrom(*config_);
+
+  int err = WriteMessageToDebugFile();
+  if (err != kNoError) {
+    return err;
+  }
+
+  return kNoError;
+}
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc