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

third_party/libjingle/overrides/init_webrtc.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 "init_webrtc.h"
 
 #include "base/command_line.h"
 #include "base/debug/trace_event.h"
 #include "base/files/file_path.h"
 #include "base/files/file_util.h"
 #include "base/metrics/field_trial.h"
 #include "base/native_library.h"
 #include "base/path_service.h"
+#include "third_party/webrtc/common.h"
+#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/logging.h"
 
 const unsigned char* GetCategoryGroupEnabled(const char* category_group) {
   return TRACE_EVENT_API_GET_CATEGORY_GROUP_ENABLED(category_group);
 }
 
 void AddTraceEvent(char phase,
                    const unsigned char* category_group_enabled,
                    const char* name,
@@ skipping 22 matching lines @@
 
 // libpeerconnection is being compiled as a static lib.  In this case
 // we don't need to do any initializing but to keep things simple we
 // provide an empty intialization routine so that this #ifdef doesn't
 // have to be in other places.
 bool InitializeWebRtcModule() {
   webrtc::SetupEventTracer(&GetCategoryGroupEnabled, &AddTraceEvent);
   return true;
 }
 
+webrtc::AudioProcessing* CreateWebRtcAudioProcessing(
+    const webrtc::Config& config) {
+  // libpeerconnection is being compiled as a static lib, use
+  // webrtc::AudioProcessing directly.
+  return webrtc::AudioProcessing::Create(config);
+}
+
 #else  // !LIBPEERCONNECTION_LIB
 
 // When being compiled as a shared library, we need to bridge the gap between
 // the current module and the libpeerconnection module, so things get a tad
 // more complicated.
 
 // Global function pointers to the factory functions in the shared library.
 CreateWebRtcMediaEngineFunction g_create_webrtc_media_engine = NULL;
 DestroyWebRtcMediaEngineFunction g_destroy_webrtc_media_engine = NULL;
+CreateWebRtcAudioProcessingFunction g_create_webrtc_audio_processing = NULL;
 
 // Returns the full or relative path to the libpeerconnection module depending
 // on what platform we're on.
 static base::FilePath GetLibPeerConnectionPath() {
   base::FilePath path;
   CHECK(PathService::Get(base::DIR_MODULE, &path));
 #if defined(OS_WIN)
   path = path.Append(FILE_PATH_LITERAL("libpeerconnection.dll"));
 #elif defined(OS_MACOSX)
   // Simulate '@loader_path/Libraries'.
@@ skipping 53 matching lines @@
 #if !defined(OS_MACOSX) && !defined(OS_ANDROID)
                                    &Allocate,
                                    &Dellocate,
 #endif
                                    &webrtc::field_trial::FindFullName,
                                    logging::GetLogMessageHandler(),
                                    &GetCategoryGroupEnabled,
                                    &AddTraceEvent,
                                    &g_create_webrtc_media_engine,
                                    &g_destroy_webrtc_media_engine,
-                                   &init_diagnostic_logging);
+                                   &init_diagnostic_logging,
+                                   &g_create_webrtc_audio_processing);
 
   if (init_ok)
     rtc::SetExtraLoggingInit(init_diagnostic_logging);
   return init_ok;
 }
 
 cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
     webrtc::AudioDeviceModule* adm,
     webrtc::AudioDeviceModule* adm_sc,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory) {
   // For convenience of tests etc, we call InitializeWebRtcModule here.
   // For Chrome however, InitializeWebRtcModule must be called
   // explicitly before the sandbox is initialized.  In that case, this call is
   // effectively a noop.
   InitializeWebRtcModule();
   return g_create_webrtc_media_engine(adm, adm_sc, encoder_factory,
       decoder_factory);
 }
 
 void DestroyWebRtcMediaEngine(cricket::MediaEngineInterface* media_engine) {
   g_destroy_webrtc_media_engine(media_engine);
 }
 
+webrtc::AudioProcessing* CreateWebRtcAudioProcessing(
+    const webrtc::Config& config) {
+  // The same as CreateWebRtcMediaEngine(), we call InitializeWebRtcModule here
+  // for convenience of tests.
+  InitializeWebRtcModule();
+  return g_create_webrtc_audio_processing(config);
+}
+
 #endif  // LIBPEERCONNECTION_LIB