Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #5 id:80001 of https://codereview.chromium.org/2037623002/ )

Reason for revert:
voice_engine_unittests: FilePlayerTest.PlayWavPcm16File and FilePlayerTest.PlayWavPcmuFile fail on 32-bit android (android_rel and android-dbg try bots, Android32 Tests (L Nexus5) and Android32 Tests (L Nexus7.2) build bots).

Not sure why this would happen, since I just moved the test without modifying it. Some test filtering that no longer manages to disable them? Anyway, reverting until I know how to fix.

This was actually caught by the try bots, but I missed it because I was manually ignoring them because of an error with the bots. :-(

Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> R=perkj@webrtc.org, solenberg@webrtc.org
>
> Committed: 65874b163e

TBR=perkj@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2092633002
Cr-Commit-Position: refs/heads/master@{#13267}

webrtc/modules/BUILD.gn

@@ -223,6 +223,7 @@ if (rtc_include_tests) {
       "rtp_rtcp/test/testAPI/test_api_rtcp.cc",
       "rtp_rtcp/test/testAPI/test_api_video.cc",
       "utility/source/audio_frame_operations_unittest.cc",
+      "utility/source/file_player_unittests.cc",
       "utility/source/process_thread_impl_unittest.cc",
       "video_coding/codecs/test/packet_manipulator_unittest.cc",
       "video_coding/codecs/test/stats_unittest.cc",

webrtc/modules/modules.gyp

@@ -351,6 +351,7 @@
             'rtp_rtcp/test/testAPI/test_api_rtcp.cc',
             'rtp_rtcp/test/testAPI/test_api_video.cc',
             'utility/source/audio_frame_operations_unittest.cc',
+            'utility/source/file_player_unittests.cc',
             'utility/source/process_thread_impl_unittest.cc',
             'video_coding/codecs/test/packet_manipulator_unittest.cc',
             'video_coding/codecs/test/stats_unittest.cc',

webrtc/modules/utility/BUILD.gn

@@ -11,10 +11,18 @@ import("../../build/webrtc.gni")
 source_set("utility") {
   sources = [
     "include/audio_frame_operations.h",
+    "include/file_player.h",
+    "include/file_recorder.h",
     "include/helpers_android.h",
     "include/jvm_android.h",
     "include/process_thread.h",
     "source/audio_frame_operations.cc",
+    "source/coder.cc",
+    "source/coder.h",
+    "source/file_player_impl.cc",
+    "source/file_player_impl.h",
+    "source/file_recorder_impl.cc",
+    "source/file_recorder_impl.h",
     "source/helpers_android.cc",
     "source/helpers_ios.mm",
     "source/jvm_android.cc",

webrtc/modules/utility/include/file_player.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
-#define WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
 
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
@@ -83,5 +83,4 @@ protected:
 
 };
 }  // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_

webrtc/modules/utility/include/file_recorder.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_
-#define WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
 
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
@@ -61,5 +61,4 @@ protected:
 
 };
 }  // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_

webrtc/modules/utility/source/coder.cc

@@ -8,11 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/voice_engine/coder.h"
-
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/source/coder.h"
 
 namespace webrtc {
 namespace {

webrtc/modules/utility/source/coder.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VOICE_ENGINE_CODER_H_
-#define WEBRTC_VOICE_ENGINE_CODER_H_
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
 
 #include <memory>
 
@@ -65,4 +65,4 @@ class AudioCoder : public AudioPacketizationCallback {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_VOICE_ENGINE_CODER_H_
+#endif  // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_

webrtc/modules/utility/source/file_player_impl.cc

@@ -8,8 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/voice_engine/file_player_impl.h"
-
+#include "webrtc/modules/utility/source/file_player_impl.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {

webrtc/modules/utility/source/file_player_impl.h

@@ -8,18 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_H_
-#define WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_H_
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
 
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/modules/media_file/media_file_defines.h"
+#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/modules/utility/source/coder.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
-#include "webrtc/voice_engine/coder.h"
-#include "webrtc/voice_engine/file_player.h"
 
 namespace webrtc {
 class FilePlayerImpl : public FilePlayer
@@ -75,5 +75,4 @@ private:
     float _scaling;
 };
 }  // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_H_
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_

webrtc/modules/utility/source/file_player_unittests.cc

@@ -10,6 +10,8 @@
 
 // Unit tests for FilePlayer.
 
+#include "webrtc/modules/utility/include/file_player.h"
+
 #include <stdio.h>
 #include <string>
 
@@ -18,7 +20,6 @@
 #include "webrtc/base/md5digest.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/voice_engine/file_player.h"
 
 DEFINE_bool(file_player_output, false, "Generate reference files.");
 

webrtc/modules/utility/source/file_recorder_impl.cc

@@ -8,10 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/voice_engine/file_recorder_impl.h"
-
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/media_file/media_file.h"
+#include "webrtc/modules/utility/source/file_recorder_impl.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {

webrtc/modules/utility/source/file_recorder_impl.h

@@ -12,8 +12,8 @@
 // multiple file formats. The unencoded input data is written to file in the
 // encoded format specified.
 
-#ifndef WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_H_
-#define WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_H_
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
 
 #include <list>
 
@@ -24,10 +24,10 @@
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/modules/media_file/media_file_defines.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
+#include "webrtc/modules/utility/source/coder.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/typedefs.h"
-#include "webrtc/voice_engine/coder.h"
-#include "webrtc/voice_engine/file_recorder.h"
 
 namespace webrtc {
 // The largest decoded frame size in samples (60ms with 32kHz sample rate).
@@ -76,5 +76,4 @@ private:
     Resampler _audioResampler;
 };
 }  // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_H_
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_

webrtc/modules/utility/utility.gypi

@@ -20,11 +20,19 @@
       ],
       'sources': [
         'include/audio_frame_operations.h',
+        'include/file_player.h',
+        'include/file_recorder.h',
         'include/helpers_android.h',
         'include/helpers_ios.h',
         'include/jvm_android.h',
         'include/process_thread.h',
         'source/audio_frame_operations.cc',
+        'source/coder.cc',
+        'source/coder.h',
+        'source/file_player_impl.cc',
+        'source/file_player_impl.h',
+        'source/file_recorder_impl.cc',
+        'source/file_recorder_impl.h',
         'source/helpers_android.cc',
         'source/helpers_ios.mm',
         'source/jvm_android.cc',

webrtc/voice_engine/BUILD.gn

@@ -16,14 +16,6 @@ source_set("voice_engine") {
     "channel_manager.h",
     "channel_proxy.cc",
     "channel_proxy.h",
-    "coder.cc",
-    "coder.h",
-    "file_player.h",
-    "file_player_impl.cc",
-    "file_player_impl.h",
-    "file_recorder.h",
-    "file_recorder_impl.cc",
-    "file_recorder_impl.h",
     "include/voe_audio_processing.h",
     "include/voe_base.h",
     "include/voe_codec.h",

webrtc/voice_engine/channel.h

@@ -26,8 +26,8 @@
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/voice_engine/file_player.h"
-#include "webrtc/voice_engine/file_recorder.h"
+#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/level_indicator.h"

webrtc/voice_engine/output_mixer.h

@@ -16,7 +16,7 @@
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
-#include "webrtc/voice_engine/file_recorder.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 

webrtc/voice_engine/transmit_mixer.h

@@ -16,8 +16,8 @@
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_processing/typing_detection.h"
 #include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/voice_engine/file_player.h"
-#include "webrtc/voice_engine/file_recorder.h"
+#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine/monitor_module.h"

webrtc/voice_engine/voice_engine.gyp

@@ -52,14 +52,6 @@
         'channel_manager.h',
         'channel_proxy.cc',
         'channel_proxy.h',
-        'coder.cc',
-        'coder.h',
-        'file_player.h',
-        'file_player_impl.cc',
-        'file_player_impl.h',
-        'file_recorder.h',
-        'file_recorder_impl.cc',
-        'file_recorder_impl.h',
         'level_indicator.cc',
         'level_indicator.h',
         'monitor_module.cc',
@@ -117,7 +109,6 @@
             'voice_engine',
             '<(DEPTH)/testing/gmock.gyp:gmock',
             '<(DEPTH)/testing/gtest.gyp:gtest',
-            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
             # The rest are to satisfy the unittests' include chain.
             # This would be unnecessary if we used qualified includes.
             '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
@@ -133,7 +124,6 @@
           ],
           'sources': [
             'channel_unittest.cc',
-            'file_player_unittests.cc',
             'network_predictor_unittest.cc',
             'transmit_mixer_unittest.cc',
             'utility_unittest.cc',