Fix the way how we create webrtc::AudioProcessing in Chrome

third_party/libjingle/overrides/initialize_module.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "allocator_shim/allocator_stub.h"
 #include "base/command_line.h"
 #include "base/files/file_path.h"
 #include "base/logging.h"
 #include "init_webrtc.h"
 #include "talk/media/webrtc/webrtcmediaengine.h"
+#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/logging.h"
 
 #if !defined(LIBPEERCONNECTION_IMPLEMENTATION) || defined(LIBPEERCONNECTION_LIB)
 #error "Only compile the allocator proxy with the shared_library implementation"
 #endif
 
 #if defined(OS_WIN)
 #define ALLOC_EXPORT __declspec(dllexport)
 #else
@@ skipping 43 matching lines @@
                       AllocateFunction alloc,
                       DellocateFunction dealloc,
 #endif
                       FieldTrialFindFullName field_trial_find,
                       logging::LogMessageHandlerFunction log_handler,
                       webrtc::GetCategoryEnabledPtr trace_get_category_enabled,
                       webrtc::AddTraceEventPtr trace_add_trace_event,
                       CreateWebRtcMediaEngineFunction* create_media_engine,
                       DestroyWebRtcMediaEngineFunction* destroy_media_engine,
                       InitDiagnosticLoggingDelegateFunctionFunction*
-                          init_diagnostic_logging) {
+                          init_diagnostic_logging,
+                      CreateWebRtcAudioProcessingFunction*
+                          create_audio_processing) {
 #if !defined(OS_MACOSX) && !defined(OS_ANDROID)
   g_alloc = alloc;
   g_dealloc = dealloc;
 #endif
 
   g_field_trial_find_ = field_trial_find;
 
   *create_media_engine = &CreateWebRtcMediaEngine;
   *destroy_media_engine = &DestroyWebRtcMediaEngine;
   *init_diagnostic_logging = &rtc::InitDiagnosticLoggingDelegateFunction;
+  *create_audio_processing = &webrtc::AudioProcessing::Create;
 
   if (CommandLine::Init(0, NULL)) {
 #if !defined(OS_WIN)
     // This is not needed on Windows since CommandLine::Init has already
     // done the equivalent thing via the GetCommandLine() API.
     CommandLine::ForCurrentProcess()->AppendArguments(command_line, true);
 #endif
     logging::LoggingSettings settings;
     settings.logging_dest = logging::LOG_TO_SYSTEM_DEBUG_LOG;
     logging::InitLogging(settings);
 
     // Override the log message handler to forward logs to chrome's handler.
     logging::SetLogMessageHandler(log_handler);
     webrtc::SetupEventTracer(trace_get_category_enabled,
                              trace_add_trace_event);
   }
 
   return true;
 }
 }  // extern "C"