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

BUG=415935
TEST=all webrtc tests in all bots + manual test to verify the agc loggings exist.

Review URL: https://codereview.chromium.org/588523002

Cr-Commit-Position: refs/heads/master@{#295990}

content/renderer/media/media_stream_audio_processor.cc

@@ -19,6 +19,7 @@
 #include "media/base/audio_fifo.h"
 #include "media/base/channel_layout.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
+#include "third_party/libjingle/overrides/init_webrtc.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
 #include "third_party/webrtc/modules/audio_processing/typing_detection.h"
 
@@ -423,7 +424,7 @@ void MediaStreamAudioProcessor::InitializeAudioProcessingModule(
 #endif
 
   // Create and configure the webrtc::AudioProcessing.
-  audio_processing_.reset(webrtc::AudioProcessing::Create(config));
+  audio_processing_.reset(CreateWebRtcAudioProcessing(config));
 
   // Enable the audio processing components.
   if (echo_cancellation) {

third_party/libjingle/BUILD.gn

@@ -549,6 +549,7 @@ if (enable_webrtc) {
     deps = [
       ":libjingle_webrtc_common",
       "//third_party/webrtc",
+      "//third_party/webrtc/modules/audio_processing",
       "//third_party/webrtc/system_wrappers",
       "//third_party/webrtc/voice_engine",
     ]

third_party/libjingle/libjingle.gyp

@@ -589,6 +589,7 @@
             '<(libjingle_source)/talk/media/webrtc/webrtcvoiceengine.h',
           ],
           'dependencies': [
+            '<(DEPTH)/third_party/webrtc/modules/modules.gyp:audio_processing',
             '<(DEPTH)/third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers',
             '<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine',
             '<(DEPTH)/third_party/webrtc/webrtc.gyp:webrtc',

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

third_party/libjingle/overrides/init_webrtc.h

@@ -23,6 +23,8 @@ class WebRtcVideoEncoderFactory;
 
 namespace webrtc {
 class AudioDeviceModule;
+class AudioProcessing;
+class Config;
 }  // namespace webrtc
 
 typedef std::string (*FieldTrialFindFullName)(const std::string& trial_name);
@@ -39,6 +41,9 @@ typedef void (*DestroyWebRtcMediaEngineFunction)(
 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.
@@ -56,7 +61,8 @@ typedef bool (*InitializeModuleFunction)(
     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).
@@ -65,6 +71,11 @@ typedef bool (*InitializeModuleFunction)(
 // 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_

third_party/libjingle/overrides/initialize_module.cc

@@ -8,6 +8,7 @@
 #include "base/logging.h"
 #include "init_webrtc.h"
 #include "talk/media/webrtc/webrtcmediaengine.h"
+#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/logging.h"
 
@@ -71,7 +72,9 @@ bool InitializeModule(const CommandLine& command_line,
                       CreateWebRtcMediaEngineFunction* create_media_engine,
                       DestroyWebRtcMediaEngineFunction* destroy_media_engine,
                       InitDiagnosticLoggingDelegateFunctionFunction*
-                          init_diagnostic_logging) {
+                          init_diagnostic_logging,
+                      CreateWebRtcAudioProcessingFunction*
+                          create_audio_processing) {
 #if !defined(OS_MACOSX) && !defined(OS_ANDROID)
   g_alloc = alloc;
   g_dealloc = dealloc;
@@ -82,6 +85,7 @@ bool InitializeModule(const CommandLine& command_line,
   *create_media_engine = &CreateWebRtcMediaEngine;
   *destroy_media_engine = &DestroyWebRtcMediaEngine;
   *init_diagnostic_logging = &rtc::InitDiagnosticLoggingDelegateFunction;
+  *create_audio_processing = &webrtc::AudioProcessing::Create;
 
   if (CommandLine::Init(0, NULL)) {
 #if !defined(OS_WIN)