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

third_party/libjingle/overrides/init_webrtc.h

 // 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.
 
 #ifndef THIRD_PARTY_LIBJINGLE_OVERRIDES_INIT_WEBRTC_H_
 #define THIRD_PARTY_LIBJINGLE_OVERRIDES_INIT_WEBRTC_H_
 
 #include <string>
 
 #include "allocator_shim/allocator_stub.h"
 #include "base/logging.h"
 #include "third_party/webrtc/system_wrappers/interface/event_tracer.h"
 
 namespace base {
 class CommandLine;
 }
 
 namespace cricket {
 class MediaEngineInterface;
 class WebRtcVideoDecoderFactory;
 class WebRtcVideoEncoderFactory;
 }  // namespace cricket
 
 namespace webrtc {
 class AudioDeviceModule;
+class AudioProcessing;
+class Config;
 }  // namespace webrtc
 
 typedef std::string (*FieldTrialFindFullName)(const std::string& trial_name);
 
 typedef cricket::MediaEngineInterface* (*CreateWebRtcMediaEngineFunction)(
     webrtc::AudioDeviceModule* adm,
     webrtc::AudioDeviceModule* adm_sc,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory);
 
 typedef void (*DestroyWebRtcMediaEngineFunction)(
     cricket::MediaEngineInterface* media_engine);
 
 typedef void (*InitDiagnosticLoggingDelegateFunctionFunction)(
     void (*DelegateFunction)(const std::string&));
 
+typedef webrtc::AudioProcessing* (*CreateWebRtcAudioProcessingFunction)(
+    const webrtc::Config& config);
+
 // A typedef for the main initialize function in libpeerconnection.
 // This will initialize logging in the module with the proper arguments
 // as well as provide pointers back to a couple webrtc factory functions.
 // The reason we get pointers to these functions this way is to avoid having
 // to go through GetProcAddress et al and rely on specific name mangling.
 typedef bool (*InitializeModuleFunction)(
     const base::CommandLine& command_line,
 #if !defined(OS_MACOSX) && !defined(OS_ANDROID)
     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);
+    InitDiagnosticLoggingDelegateFunctionFunction* init_diagnostic_logging,
+    CreateWebRtcAudioProcessingFunction* create_audio_processing);
 
 #if !defined(LIBPEERCONNECTION_IMPLEMENTATION)
 // Load and initialize the shared WebRTC module (libpeerconnection).
 // Call this explicitly to load and initialize the WebRTC module (e.g. before
 // initializing the sandbox in Chrome).
 // If not called explicitly, this function will still be called from the main
 // CreateWebRtcMediaEngine factory function the first time it is called.
 bool InitializeWebRtcModule();
+
+// Return a webrtc::AudioProcessing object.
+webrtc::AudioProcessing* CreateWebRtcAudioProcessing(
+    const webrtc::Config& config);
+
 #endif
 
 #endif // THIRD_PARTY_LIBJINGLE_OVERRIDES_INIT_WEBRTC_H_