Revert "Merge 663413002 to M39:"

This reverts commit 255ee24.

[email protected]

 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.
>
> [email protected],[email protected]
>
> BUG=415935
> TEST=all webrtc tests in all bots + manual test to verify the agc loggings exist.

Review URL: https://codereview.chromium.org/730563002

Cr-Commit-Position: refs/branch-heads/2171@{#419}
Cr-Branched-From: 267aeeb-refs/heads/master@{#297060}

Author: xians

build/android/pylib/gtest/setup.py

@@ -118,7 +118,6 @@ def _GenerateDepsDirUsingIsolate(suite_name, isolate_file_path=None):
       '--config-variable', 'component', 'static_library',
       '--config-variable', 'fastbuild', '0',
       '--config-variable', 'icu_use_data_file_flag', '1',
-      '--config-variable', 'libpeer_target_type', 'static_library',
       # TODO(maruel): This may not be always true.
       '--config-variable', 'target_arch', 'arm',
       '--config-variable', 'use_openssl', '0',

content/content_tests.gypi

@@ -838,21 +838,6 @@
             '../third_party/webrtc/modules/modules.gyp:desktop_capture',
           ],
         }],
-        ['enable_webrtc==1 and OS=="mac"', {
-          'variables': {
-            'libpeer_target_type%': 'static_library',
-          },
-          'conditions': [
-            ['libpeer_target_type!="static_library"', {
-              'copies': [{
-                'destination': '<(PRODUCT_DIR)/Libraries',
-                'files': [
-                  '<(PRODUCT_DIR)/libpeerconnection.so',
-                ],
-              }],
-            }],
-          ],
-        }],
         ['enable_webrtc==1 and chromeos==1', {
           'sources': [
             'browser/media/capture/desktop_capture_device_aura_unittest.cc',

content/content_unittests.isolate

@@ -56,13 +56,6 @@
         ],
       },
     }],
-    ['OS=="linux" and libpeer_target_type=="loadable_module"', {
-      'variables': {
-        'isolate_dependency_tracked': [
-          '<(PRODUCT_DIR)/lib/libpeerconnection.so',
-        ],
-      },
-    }],
     ['OS=="mac"', {
       'variables': {
         'command': [
@@ -98,13 +91,6 @@
         ],
       },
     }],
-    ['OS=="win" and libpeer_target_type=="loadable_module"', {
-      'variables': {
-        'isolate_dependency_tracked': [
-          '<(PRODUCT_DIR)/libpeerconnection.dll',
-        ],
-      },
-    }],
   ],
   'includes': [
     '../base/base.isolate',

content/renderer/media/media_stream_audio_processor.cc

@@ -19,7 +19,6 @@
 #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"
 
@@ -424,7 +423,7 @@ void MediaStreamAudioProcessor::InitializeAudioProcessingModule(
 #endif
 
   // Create and configure the webrtc::AudioProcessing.
-  audio_processing_.reset(CreateWebRtcAudioProcessing(config));
+  audio_processing_.reset(webrtc::AudioProcessing::Create(config));
 
   // Enable the audio processing components.
   if (echo_cancellation) {

content/renderer/media/media_stream_audio_processor_options.cc

@@ -251,10 +251,15 @@ void EnableTypingDetection(AudioProcessing* audio_processing,
 void StartEchoCancellationDump(AudioProcessing* audio_processing,
                                base::File aec_dump_file) {
   DCHECK(aec_dump_file.IsValid());
-  if (audio_processing->StartDebugRecordingForPlatformFile(
-      aec_dump_file.TakePlatformFile())) {
-    DLOG(ERROR) << "Fail to start AEC debug recording";
+
+  FILE* stream = base::FileToFILE(aec_dump_file.Pass(), "w");
+  if (!stream) {
+    LOG(ERROR) << "Failed to open AEC dump file";
+    return;
   }
+
+  if (audio_processing->StartDebugRecording(stream))
+    DLOG(ERROR) << "Fail to start AEC debug recording";
 }
 
 void StopEchoCancellationDump(AudioProcessing* audio_processing) {

third_party/libjingle/BUILD.gn

@@ -548,7 +548,6 @@ 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",
     ]

third_party/libjingle/libjingle.gyp

@@ -589,7 +589,6 @@
             '<(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',

third_party/libjingle/overrides/init_webrtc.cc

@@ -11,8 +11,6 @@
 #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"
 
@@ -55,13 +53,6 @@ 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
@@ -71,7 +62,6 @@ 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.
@@ -145,8 +135,8 @@ 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;
@@ -170,12 +160,4 @@ 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

third_party/libjingle/overrides/init_webrtc.h

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

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"
 
@@ -72,9 +71,7 @@ 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;
@@ -85,7 +82,6 @@ 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)