Reland 597923002: 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 041fb20b05c1caa00fbf41aff20fc6dbeb96e51c..b160cbb1bf1ed37128f436982476b5ba425355c4 100644
--- a/third_party/libjingle/overrides/init_webrtc.cc
+++ b/third_party/libjingle/overrides/init_webrtc.cc
@@ -12,6 +12,8 @@
 #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"
 
@@ -80,6 +82,13 @@ 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
@@ -89,6 +98,7 @@ 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.
@@ -165,8 +175,8 @@ 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);
   return init_ok;
@@ -190,4 +200,12 @@ 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