Adding APM configuration in AEC dump.

webrtc/modules/audio_processing/audio_processing_impl.cc

Index: webrtc/modules/audio_processing/audio_processing_impl.cc
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 4ef4e6da6a307e6533e8ac9ddf39e888c2c52713..9d0632649f49c9b99cfe9897271e271a809ea9bf 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -42,15 +42,6 @@ extern "C" {
 #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);       \
@@ -59,6 +50,15 @@ extern "C" {
     }                       \
   } while (0)
 
+#define UPDATE_CONFIG(field_name, new_value) \
+  { \
+    const auto value = new_value; \
+    if (!config_->has_##field_name() || config_->field_name() != value) { \
+      config_->set_##field_name(value); \
+      changed = true; \
+    } \
+  }
+
 namespace webrtc {
 namespace {
 
@@ -194,6 +194,7 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config,
 #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},
@@ -250,6 +251,10 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config,
   gain_control_for_new_agc_.reset(new GainControlForNewAgc(gain_control_));
 
   SetExtraOptions(config);
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  UpdateCurrentConfig();
+#endif
 }
 
 AudioProcessingImpl::~AudioProcessingImpl() {
@@ -560,6 +565,12 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
          api_format_.input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  if (debug_file_->Open() && UpdateCurrentConfig()) {
+    RETURN_ON_ERR(WriteConfigMessage());
+  }
+#endif
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_file_->Open()) {
     event_msg_->set_type(audioproc::Event::STREAM);
     audioproc::Stream* msg = event_msg_->mutable_stream();
@@ -946,7 +957,13 @@ int AudioProcessingImpl::StartDebugRecording(
     return kFileError;
   }
 
-  int err = WriteInitMessage();
+  UpdateCurrentConfig();
+  int err = WriteConfigMessage();
+  if (err != kNoError) {
+    return err;
+  }
+
+  err = WriteInitMessage();
   if (err != kNoError) {
     return err;
   }
@@ -975,7 +992,13 @@ int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
     return kFileError;
   }
 
-  int err = WriteInitMessage();
+  UpdateCurrentConfig();
+  int err = WriteConfigMessage();
+  if (err != kNoError) {
+    return err;
+  }
+
+  err = WriteInitMessage();
   if (err != kNoError) {
     return err;
   }
@@ -1255,6 +1278,52 @@ int AudioProcessingImpl::WriteInitMessage() {
 
   return kNoError;
 }
+
+bool AudioProcessingImpl::UpdateCurrentConfig() {
+  bool changed = false;
+
+  // Acoustic echo canceler.
+  UPDATE_CONFIG(aec_enabled, echo_cancellation_->is_enabled());
+  UPDATE_CONFIG(aec_delay_agnostic,
+                echo_cancellation_->is_delay_agnostic_enabled());
+  UPDATE_CONFIG(aec_drift_compensation,
+                echo_cancellation_->is_drift_compensation_enabled());
+  UPDATE_CONFIG(aec_extended_filter,
+                echo_cancellation_->is_extended_filter_enabled());
+  UPDATE_CONFIG(aec_suppression_level,
+                static_cast<int>(echo_cancellation_->suppression_level()));
+  // Mobile AEC.
+  UPDATE_CONFIG(aecm_enabled, echo_control_mobile_->is_enabled());
+  UPDATE_CONFIG(aecm_comfort_noise,
+                echo_control_mobile_->is_comfort_noise_enabled());
+  UPDATE_CONFIG(aecm_routing_mode,
+                static_cast<int>(echo_control_mobile_->routing_mode()));
+  // Automatic gain controller.
+  UPDATE_CONFIG(agc_enabled, gain_control_->is_enabled());
+  UPDATE_CONFIG(agc_experiment, use_new_agc_);
+  UPDATE_CONFIG(agc_mode, static_cast<int>(gain_control_->mode()));
+  UPDATE_CONFIG(agc_limiter, gain_control_->is_limiter_enabled());
+  // High pass filter.
+  UPDATE_CONFIG(hpf_enabled, high_pass_filter_->is_enabled());
+  // Noise suppression.
+  UPDATE_CONFIG(ns_enabled, noise_suppression_->is_enabled());
+  UPDATE_CONFIG(ns_experiment, transient_suppressor_enabled_);
+  UPDATE_CONFIG(ns_level, static_cast<int>(noise_suppression_->level()));
+
+  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