Bug 1847818 - restore aec logging from about:webrtc. r=ng,webrtc-reviewers

Differential Revision: https://phabricator.services.mozilla.com/D185709

third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.h

@@ -26,6 +26,7 @@
 #if WEBRTC_APM_DEBUG_DUMP == 1
 #include "common_audio/wav_file.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
 #include "rtc_base/string_utils.h"
 #endif
 
@@ -395,7 +396,7 @@ class ApmDataDumper {
       // Cheat and use aec_near as a stand-in for "size of the largest file"
       // in the dump.  We're looking to limit the total time, and that's a
       // reasonable stand-in.
-      if (strcmp(name, "aec_near") == 0) {
+      if (strcmp(name.data(), "aec_near") == 0) {
        updateDebugWritten(v_length * sizeof(float));
       }
     }
@@ -439,7 +440,7 @@ class ApmDataDumper {
 
   void updateDebugWritten(uint32_t amount) {
     debug_written_ += amount;
-    if (debug_written_ >= webrtc::Trace::aec_debug_size()) {
+    if (debug_written_ >= rtc::LogMessage::aec_debug_size()) {
       SetActivated(false);
     }
   }

third_party/libwebrtc/rtc_base/BUILD.gn

@@ -425,6 +425,12 @@ rtc_library("logging") {
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 
+  # Added to allow including apm_data_dumper.h in rtc_base/logging.cc
+  # for Mozilla controlling aec logging from about:webrtc.
+  if (build_with_mozilla) {
+    configs += [ "../modules/audio_processing:apm_debug_dump" ]
+  }
+
   if (build_with_chromium) {
     # Dependency on chromium's logging (in //base).
     deps += [ "//base" ]

third_party/libwebrtc/rtc_base/logging.cc

@@ -44,6 +44,7 @@ static const int kMaxLogLineSize = 1024 - 60;
 #include "absl/base/attributes.h"
 #include "absl/strings/string_view.h"
 #include "api/units/timestamp.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/platform_thread_types.h"
 #include "rtc_base/string_encode.h"
@@ -59,6 +60,11 @@ bool LogMessage::aec_debug_ = false;
 uint32_t LogMessage::aec_debug_size_ = 4*1024*1024;
 std::string LogMessage::aec_filename_base_;
 
+void LogMessage::set_aec_debug(bool enable) {
+  aec_debug_ = enable;
+  webrtc::ApmDataDumper::SetActivated(aec_debug_);
+}
+
 std::string LogMessage::aec_debug_filename() {
   return aec_filename_base_;
 }

third_party/libwebrtc/rtc_base/logging.h

@@ -590,7 +590,7 @@ class LogMessage {
 #endif  // RTC_LOG_ENABLED()
 
   // Enable dumping of AEC inputs and outputs.  Can be changed in mid-call
-  static void set_aec_debug(bool enable) { aec_debug_ = enable; }
+  static void set_aec_debug(bool enable);
   static void set_aec_debug_size(uint32_t size) { aec_debug_size_ = size; }
   static bool aec_debug() { return aec_debug_; }
   static uint32_t aec_debug_size() { return aec_debug_size_; }

third_party/libwebrtc/webrtc.gni

@@ -132,7 +132,7 @@ declare_args() {
 
   # Selects whether debug dumps for the audio processing module
   # should be generated.
-  apm_debug_dump = false
+  apm_debug_dump = build_with_mozilla
 
   # Selects whether the audio processing module should be excluded.
   rtc_exclude_audio_processing_module = false