Revert of Reland 597923002: 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/metrics/histogram.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 48 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 56 matching lines @@
 #endif
       &webrtc::field_trial::FindFullName,
       &webrtc::metrics::HistogramFactoryGetCounts,
       &webrtc::metrics::HistogramFactoryGetEnumeration,
       &webrtc::metrics::HistogramAdd,
       logging::GetLogMessageHandler(),
       &GetCategoryGroupEnabled,
       &AddTraceEvent,
       &g_create_webrtc_media_engine,
       &g_destroy_webrtc_media_engine,
-      &init_diagnostic_logging,
-      &g_create_webrtc_audio_processing);
+      &init_diagnostic_logging);
+
   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