Merge the revert to M40.

Revert of  Reland 597923002: Fix the way how we create webrtc::AudioProcessing in Chrome (patchset #4 id:60001 of https://codereview.chromium.org/663413002/)

Reason for revert:
This CL broke the configuration of AudioProcessing, we have to revert it to fix the echo issues it introduces.

Original issue's description:
> Reland 597923002: Fix the way how we create webrtc::AudioProcessing in Chrome.
>
> The original review thread is in https://codereview.chromium.org/588523002/
>
> Fix the way how we create webrtc::AudioProcessing in Chrome.
>
> TBR=tommi@chromium.org,maruel@chromium.org
>
> BUG=415935
> TEST=all webrtc tests in all bots + manual test to verify the agc loggings exist.
>
> Committed: https://crrev.com/79ef9085fbdbc8e09ac989ea4d5f4c28e516bba9
> Cr-Commit-Position: refs/heads/master@{#300509}

TBR=maruel@chromium.org,tommi@chromium.org
NOTREECHECKS=true
NOTRY=true
BUG=415935

Review URL: https://codereview.chromium.org/717203002

Cr-Commit-Position: refs/heads/master@{#303834}
(cherry picked from commit 515c2beb237eae38347a86f7011b897674496ea2)

Review URL: https://codereview.chromium.org/727383002

Cr-Commit-Position: refs/branch-heads/2214@{#60}
Cr-Branched-From: 03655fd3f6d72165dc3c9bd2c89807305316fe6c-refs/heads/master@{#303346}

5 files changed, 4 insertions, 39 deletions

diff --git a/third_party/libjingle/BUILD.gn b/third_party/libjingle/BUILD.gn
index 38f0a02..f56e81e 100644
--- a/third_party/libjingle/BUILD.gn
+++ b/third_party/libjingle/BUILD.gn
@@ -552,7 +552,6 @@ 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",
     ]
diff --git a/third_party/libjingle/libjingle.gyp b/third_party/libjingle/libjingle.gyp
index cc48f24..f062841 100644
--- a/third_party/libjingle/libjingle.gyp
+++ b/third_party/libjingle/libjingle.gyp
@@ -598,7 +598,6 @@
             '<(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',
diff --git a/third_party/libjingle/overrides/init_webrtc.cc b/third_party/libjingle/overrides/init_webrtc.cc
index b160cbb..041fb20 100644
--- a/third_party/libjingle/overrides/init_webrtc.cc
+++ b/third_party/libjingle/overrides/init_webrtc.cc
@@ -12,8 +12,6 @@
 #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"
 
@@ -82,13 +80,6 @@ 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
@@ -98,7 +89,6 @@ webrtc::AudioProcessing* CreateWebRtcAudioProcessing(
 // 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.
@@ -175,8 +165,8 @@ bool InitializeWebRtcModule() {
       &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;
@@ -200,12 +190,4 @@ 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
diff --git a/third_party/libjingle/overrides/init_webrtc.h b/third_party/libjingle/overrides/init_webrtc.h
index c29bd71..714f9c6 100644
--- a/third_party/libjingle/overrides/init_webrtc.h
+++ b/third_party/libjingle/overrides/init_webrtc.h
@@ -23,8 +23,6 @@ class WebRtcVideoEncoderFactory;
 
 namespace webrtc {
 class AudioDeviceModule;
-class AudioProcessing;
-class Config;
 namespace metrics {
 class Histogram;
 }  // namespace metrics
@@ -53,9 +51,6 @@ 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.
@@ -77,8 +72,7 @@ typedef bool (*InitializeModuleFunction)(
     webrtc::AddTraceEventPtr trace_add_trace_event,
     CreateWebRtcMediaEngineFunction* create_media_engine,
     DestroyWebRtcMediaEngineFunction* destroy_media_engine,
-    InitDiagnosticLoggingDelegateFunctionFunction* init_diagnostic_logging,
-    CreateWebRtcAudioProcessingFunction* create_audio_processing);
+    InitDiagnosticLoggingDelegateFunctionFunction* init_diagnostic_logging);
 
 #if !defined(LIBPEERCONNECTION_IMPLEMENTATION)
 // Load and initialize the shared WebRTC module (libpeerconnection).
@@ -87,11 +81,6 @@ 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_
diff --git a/third_party/libjingle/overrides/initialize_module.cc b/third_party/libjingle/overrides/initialize_module.cc
index 1250cfb..09afbc2 100644
--- a/third_party/libjingle/overrides/initialize_module.cc
+++ b/third_party/libjingle/overrides/initialize_module.cc
@@ -8,7 +8,6 @@
 #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"
 
@@ -99,9 +98,7 @@ bool InitializeModule(const CommandLine& command_line,
                       CreateWebRtcMediaEngineFunction* create_media_engine,
                       DestroyWebRtcMediaEngineFunction* destroy_media_engine,
                       InitDiagnosticLoggingDelegateFunctionFunction*
-                          init_diagnostic_logging,
-                      CreateWebRtcAudioProcessingFunction*
-                          create_audio_processing) {
+                          init_diagnostic_logging) {
 #if !defined(OS_MACOSX) && !defined(OS_ANDROID)
   g_alloc = alloc;
   g_dealloc = dealloc;
@@ -115,7 +112,6 @@ 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)