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

third_party/libjingle/overrides/init_webrtc.cc

Index: third_party/libjingle/overrides/init_webrtc.cc
diff --git a/third_party/libjingle/overrides/init_webrtc.cc b/third_party/libjingle/overrides/init_webrtc.cc
index ab89d5884adf1ffca2d5b5fe929eb344a212165a..6db34f66fbab5c82b7f8e8376954ba31bf6f34fd 100644
--- a/third_party/libjingle/overrides/init_webrtc.cc
+++ b/third_party/libjingle/overrides/init_webrtc.cc
@@ -11,6 +11,8 @@
 #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"
 
@@ -53,6 +55,13 @@ bool InitializeWebRtcModule() {
   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
@@ -62,6 +71,7 @@ bool InitializeWebRtcModule() {
 // 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.
@@ -135,7 +145,8 @@ bool InitializeWebRtcModule() {
                                    &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);
@@ -160,4 +171,12 @@ 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