APM: Add ability to turn on/off dumping of internal data

This CL modifies the internal data logging and the audioproc_f tool
to allow controlling that via the command line, rather than solely via a
build flag. The logging of internal data is by default off.

Bug: webrtc:5298
Change-Id: I96d1b4f990582938527b9039d6c2ecbb6f76e9ca
Reviewed-on: https://webrtc-review.googlesource.com/c/107713
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25352}

modules/audio_processing/BUILD.gn

@@ -514,6 +514,7 @@ if (rtc_include_tests) {
   if (rtc_enable_protobuf) {
     rtc_source_set("audioproc_f_impl") {
       testonly = true
+      configs += [ ":apm_debug_dump" ]
       sources = [
         "test/aec_dump_based_simulator.cc",
         "test/aec_dump_based_simulator.h",
@@ -527,6 +528,7 @@ if (rtc_include_tests) {
 
       deps = [
         ":analog_mic_simulation",
+        ":apm_logging",
         ":audio_processing",
         ":audioproc_debug_proto",
         ":audioproc_protobuf_utils",

modules/audio_processing/logging/apm_data_dumper.cc

@@ -46,6 +46,8 @@ ApmDataDumper::ApmDataDumper(int instance_index) {}
 ApmDataDumper::~ApmDataDumper() {}
 
 #if WEBRTC_APM_DEBUG_DUMP == 1
+bool ApmDataDumper::recording_activated_ = false;
+;
 FILE* ApmDataDumper::GetRawFile(const char* name) {
   std::string filename =
       FormFileName(name, instance_index_, recording_set_index_, ".dat");

modules/audio_processing/logging/apm_data_dumper.h

@@ -50,6 +50,13 @@ class ApmDataDumper {
 
   ~ApmDataDumper();
 
+  // Activates or deactivate the dumping functionality.
+  static void SetActivated(bool activated) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    recording_activated_ = activated;
+#endif
+  }
+
   // Reinitializes the data dumping such that new versions
   // of all files being dumped to are created.
   void InitiateNewSetOfRecordings() {
@@ -62,117 +69,151 @@ class ApmDataDumper {
   // various formats.
   void DumpRaw(const char* name, double v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    FILE* file = GetRawFile(name);
+    if (recording_activated_) {
-    fwrite(&v, sizeof(v), 1, file);
+      FILE* file = GetRawFile(name);
+      fwrite(&v, sizeof(v), 1, file);
+    }
 #endif
   }
 
   void DumpRaw(const char* name, size_t v_length, const double* v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    FILE* file = GetRawFile(name);
+    if (recording_activated_) {
-    fwrite(v, sizeof(v[0]), v_length, file);
+      FILE* file = GetRawFile(name);
+      fwrite(v, sizeof(v[0]), v_length, file);
+    }
 #endif
   }
 
   void DumpRaw(const char* name, rtc::ArrayView<const double> v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpRaw(name, v.size(), v.data());
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
 #endif
   }
 
   void DumpRaw(const char* name, float v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    FILE* file = GetRawFile(name);
+    if (recording_activated_) {
-    fwrite(&v, sizeof(v), 1, file);
+      FILE* file = GetRawFile(name);
+      fwrite(&v, sizeof(v), 1, file);
+    }
 #endif
   }
 
   void DumpRaw(const char* name, size_t v_length, const float* v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    FILE* file = GetRawFile(name);
+    if (recording_activated_) {
-    fwrite(v, sizeof(v[0]), v_length, file);
+      FILE* file = GetRawFile(name);
+      fwrite(v, sizeof(v[0]), v_length, file);
+    }
 #endif
   }
 
   void DumpRaw(const char* name, rtc::ArrayView<const float> v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpRaw(name, v.size(), v.data());
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
 #endif
   }
 
   void DumpRaw(const char* name, bool v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpRaw(name, static_cast<int16_t>(v));
+    if (recording_activated_) {
+      DumpRaw(name, static_cast<int16_t>(v));
+    }
 #endif
   }
 
   void DumpRaw(const char* name, size_t v_length, const bool* v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    FILE* file = GetRawFile(name);
+    if (recording_activated_) {
-    for (size_t k = 0; k < v_length; ++k) {
+      FILE* file = GetRawFile(name);
-      int16_t value = static_cast<int16_t>(v[k]);
+      for (size_t k = 0; k < v_length; ++k) {
-      fwrite(&value, sizeof(value), 1, file);
+        int16_t value = static_cast<int16_t>(v[k]);
+        fwrite(&value, sizeof(value), 1, file);
+      }
     }
 #endif
   }
 
   void DumpRaw(const char* name, rtc::ArrayView<const bool> v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpRaw(name, v.size(), v.data());
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
 #endif
   }
 
   void DumpRaw(const char* name, int16_t v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    FILE* file = GetRawFile(name);
+    if (recording_activated_) {
-    fwrite(&v, sizeof(v), 1, file);
+      FILE* file = GetRawFile(name);
+      fwrite(&v, sizeof(v), 1, file);
+    }
 #endif
   }
 
   void DumpRaw(const char* name, size_t v_length, const int16_t* v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    FILE* file = GetRawFile(name);
+    if (recording_activated_) {
-    fwrite(v, sizeof(v[0]), v_length, file);
+      FILE* file = GetRawFile(name);
+      fwrite(v, sizeof(v[0]), v_length, file);
+    }
 #endif
   }
 
   void DumpRaw(const char* name, rtc::ArrayView<const int16_t> v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpRaw(name, v.size(), v.data());
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
 #endif
   }
 
   void DumpRaw(const char* name, int32_t v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    FILE* file = GetRawFile(name);
+    if (recording_activated_) {
-    fwrite(&v, sizeof(v), 1, file);
+      FILE* file = GetRawFile(name);
+      fwrite(&v, sizeof(v), 1, file);
+    }
 #endif
   }
 
   void DumpRaw(const char* name, size_t v_length, const int32_t* v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    FILE* file = GetRawFile(name);
+    if (recording_activated_) {
-    fwrite(v, sizeof(v[0]), v_length, file);
+      FILE* file = GetRawFile(name);
+      fwrite(v, sizeof(v[0]), v_length, file);
+    }
 #endif
   }
 
   void DumpRaw(const char* name, size_t v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    FILE* file = GetRawFile(name);
+    if (recording_activated_) {
-    fwrite(&v, sizeof(v), 1, file);
+      FILE* file = GetRawFile(name);
+      fwrite(&v, sizeof(v), 1, file);
+    }
 #endif
   }
 
   void DumpRaw(const char* name, size_t v_length, const size_t* v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    FILE* file = GetRawFile(name);
+    if (recording_activated_) {
-    fwrite(v, sizeof(v[0]), v_length, file);
+      FILE* file = GetRawFile(name);
+      fwrite(v, sizeof(v[0]), v_length, file);
+    }
 #endif
   }
 
   void DumpRaw(const char* name, rtc::ArrayView<const int32_t> v) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpRaw(name, v.size(), v.data());
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
 #endif
   }
 
@@ -182,8 +223,10 @@ class ApmDataDumper {
                int sample_rate_hz,
                int num_channels) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels);
+    if (recording_activated_) {
-    file->WriteSamples(v, v_length);
+      WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels);
+      file->WriteSamples(v, v_length);
+    }
 #endif
   }
 
@@ -192,12 +235,15 @@ class ApmDataDumper {
                int sample_rate_hz,
                int num_channels) {
 #if WEBRTC_APM_DEBUG_DUMP == 1
-    DumpWav(name, v.size(), v.data(), sample_rate_hz, num_channels);
+    if (recording_activated_) {
+      DumpWav(name, v.size(), v.data(), sample_rate_hz, num_channels);
+    }
 #endif
   }
 
  private:
 #if WEBRTC_APM_DEBUG_DUMP == 1
+  static bool recording_activated_;
   const int instance_index_;
   int recording_set_index_ = 0;
   std::unordered_map<std::string, std::unique_ptr<FILE, RawFileCloseFunctor>>

modules/audio_processing/test/audio_processing_simulator.cc

@@ -25,6 +25,7 @@
 #include "modules/audio_processing/echo_cancellation_impl.h"
 #include "modules/audio_processing/echo_control_mobile_impl.h"
 #include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
 #include "modules/audio_processing/test/fake_recording_device.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -114,6 +115,9 @@ AudioProcessingSimulator::AudioProcessingSimulator(
           settings.initial_mic_level,
           settings_.simulate_mic_gain ? *settings.simulated_mic_kind : 0),
       worker_queue_("file_writer_task_queue") {
+  RTC_CHECK(!settings_.dump_internal_data || WEBRTC_APM_DEBUG_DUMP == 1);
+  ApmDataDumper::SetActivated(settings_.dump_internal_data);
+
   if (settings_.ed_graph_output_filename &&
       !settings_.ed_graph_output_filename->empty()) {
     residual_echo_likelihood_graph_writer_.open(

modules/audio_processing/test/audio_processing_simulator.h

@@ -92,6 +92,7 @@ struct SimulationSettings {
   bool fixed_interface = false;
   bool store_intermediate_output = false;
   bool print_aec3_parameter_values = false;
+  bool dump_internal_data = false;
   absl::optional<std::string> custom_call_order_filename;
   absl::optional<std::string> aec3_settings_filename;
 };

modules/audio_processing/test/audioproc_float_impl.cc

@@ -200,6 +200,9 @@ WEBRTC_DEFINE_bool(print_aec3_parameter_values,
 WEBRTC_DEFINE_string(aec3_settings,
                      "",
                      "File in JSON-format with custom AEC3 settings");
+WEBRTC_DEFINE_bool(dump_data,
+                   false,
+                   "Dump internal data during the call (requires build flag)");
 WEBRTC_DEFINE_bool(help, false, "Print this message");
 
 void SetSettingIfSpecified(const std::string& value,
@@ -311,6 +314,7 @@ SimulationSettings CreateSettings() {
   settings.fixed_interface = FLAG_fixed_interface;
   settings.store_intermediate_output = FLAG_store_intermediate_output;
   settings.print_aec3_parameter_values = FLAG_print_aec3_parameter_values;
+  settings.dump_internal_data = FLAG_dump_data;
 
   return settings;
 }
@@ -461,6 +465,10 @@ void PerformBasicParameterSanityChecks(const SimulationSettings& settings) {
       settings.artificial_nearend_filename &&
           !valid_wav_name(*settings.artificial_nearend_filename),
       "Error: --artifical_nearend must be a valid .wav file name.\n");
+
+  ReportConditionalErrorAndExit(
+      WEBRTC_APM_DEBUG_DUMP == 0 && settings.dump_internal_data,
+      "Error: --dump_data cannot be set without proper build support.\n");
 }
 
 }  // namespace