Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #6 id:100001 of https://codereview.webrtc.org/2240163002/ )

Reason for revert:
Breaks downstream code, so revert again. Yay.

Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> (This is a re-land of https://codereview.webrtc.org/2037623002, which
> had to be reverted.)
>
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/dc65ea29b3270ad418050658ad962ddd33ee70c1
> Cr-Commit-Position: refs/heads/master@{#13757}

TBR=perkj@webrtc.org,kjellander@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/2245153002
Cr-Commit-Position: refs/heads/master@{#13758}

.gn

@@ -19,12 +19,7 @@ secondary_source = "//build/secondary/"
 # their includes checked for proper dependencies when you run either
 # "gn check" or "gn gen --check".
 # TODO(kjellander): Keep adding paths to this list as work in webrtc:5589 is done.
-check_targets = [
+check_targets = [ "//webrtc/voice_engine:level_indicator" ]
-  "//webrtc/voice_engine:audio_coder",
-  "//webrtc/voice_engine:file_player",
-  "//webrtc/voice_engine:file_recorder",
-  "//webrtc/voice_engine:level_indicator",
-]
 
 # These are the list of GN files that run exec_script. This whitelist exists
 # to force additional review for new uses of exec_script, which is strongly

webrtc/modules/BUILD.gn

@@ -306,6 +306,7 @@ 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",
@@ -595,6 +596,8 @@ if (rtc_include_tests) {
         "//resources/synthetic-trace.rx",
         "//resources/tmobile-downlink.rx",
         "//resources/tmobile-uplink.rx",
+        "//resources/utility/encapsulated_pcm16b_8khz.wav",
+        "//resources/utility/encapsulated_pcmu_8khz.wav",
         "//resources/verizon3g-downlink.rx",
         "//resources/verizon3g-uplink.rx",
         "//resources/verizon4g-downlink.rx",

webrtc/modules/audio_mixer/audio_mixer.h

@@ -16,7 +16,7 @@
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_mixer/new_audio_conference_mixer.h"
 #include "webrtc/modules/audio_mixer/audio_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/modules/modules.gyp

@@ -358,6 +358,7 @@
             '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',
@@ -598,6 +599,8 @@
                 '<(DEPTH)/resources/synthetic-trace.rx',
                 '<(DEPTH)/resources/tmobile-downlink.rx',
                 '<(DEPTH)/resources/tmobile-uplink.rx',
+                '<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav',
+                '<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav',
                 '<(DEPTH)/resources/verizon3g-downlink.rx',
                 '<(DEPTH)/resources/verizon3g-uplink.rx',
                 '<(DEPTH)/resources/verizon4g-downlink.rx',

webrtc/modules/modules_unittests.isolate

@@ -110,6 +110,8 @@
           '<(DEPTH)/resources/synthetic-trace.rx',
           '<(DEPTH)/resources/tmobile-downlink.rx',
           '<(DEPTH)/resources/tmobile-uplink.rx',
+          '<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav',
+          '<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav',
           '<(DEPTH)/resources/verizon3g-downlink.rx',
           '<(DEPTH)/resources/verizon3g-uplink.rx',
           '<(DEPTH)/resources/verizon4g-downlink.rx',

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_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
-#define WEBRTC_VOICE_ENGINE_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_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
-#endif // WEBRTC_VOICE_ENGINE_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_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
-#define WEBRTC_VOICE_ENGINE_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_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
-#endif // WEBRTC_VOICE_ENGINE_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_
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
-#define WEBRTC_VOICE_ENGINE_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_
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
-#define WEBRTC_VOICE_ENGINE_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_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
-#endif // WEBRTC_VOICE_ENGINE_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_
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
-#define WEBRTC_VOICE_ENGINE_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_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
-#endif // WEBRTC_VOICE_ENGINE_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

@@ -9,74 +9,6 @@
 import("../build/webrtc.gni")
 import("//testing/test.gni")
 
-source_set("audio_coder") {
-  sources = [
-    "coder.cc",
-    "coder.h",
-  ]
-  configs += [ "..:common_config" ]
-  public_configs = [ "..:common_inherited_config" ]
-  deps = [
-    "../modules/audio_coding:audio_coding",
-    "../modules/audio_coding:builtin_audio_decoder_factory",
-    "../modules/audio_coding:rent_a_codec",
-    "..:webrtc_common",
-  ]
-
-  if (is_clang) {
-    # Suppress warnings from Chrome's Clang plugins.
-    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-    configs -= [ "//build/config/clang:find_bad_constructs" ]
-  }
-}
-
-source_set("file_player") {
-  sources = [
-    "file_player.h",
-    "file_player_impl.cc",
-    "file_player_impl.h",
-  ]
-  configs += [ "..:common_config" ]
-  public_configs = [ "..:common_inherited_config" ]
-  deps = [
-    "../common_audio:common_audio",
-    "../modules/media_file:media_file",
-    "../system_wrappers:system_wrappers",
-    "..:webrtc_common",
-    ":audio_coder",
-  ]
-
-  if (is_clang) {
-    # Suppress warnings from Chrome's Clang plugins.
-    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-    configs -= [ "//build/config/clang:find_bad_constructs" ]
-  }
-}
-
-source_set("file_recorder") {
-  sources = [
-    "file_recorder.h",
-    "file_recorder_impl.cc",
-    "file_recorder_impl.h",
-  ]
-  configs += [ "..:common_config" ]
-  public_configs = [ "..:common_inherited_config" ]
-  deps = [
-    "../base:rtc_base_approved",
-    "../common_audio:common_audio",
-    "../modules/media_file:media_file",
-    "../system_wrappers:system_wrappers",
-    "..:webrtc_common",
-    ":audio_coder",
-  ]
-
-  if (is_clang) {
-    # Suppress warnings from Chrome's Clang plugins.
-    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-    configs -= [ "//build/config/clang:find_bad_constructs" ]
-  }
-}
-
 source_set("voice_engine") {
   sources = [
     "channel.cc",
@@ -157,8 +89,6 @@ source_set("voice_engine") {
   }
 
   deps = [
-    ":file_player",
-    ":file_recorder",
     ":level_indicator",
     "..:rtc_event_log",
     "..:webrtc_common",
@@ -199,7 +129,6 @@ if (rtc_include_tests) {
       ":voice_engine",
       "//testing/gmock",
       "//testing/gtest",
-      "//third_party/gflags",
       "//webrtc/common_audio",
       "//webrtc/modules/audio_coding",
       "//webrtc/modules/audio_conference_mixer",
@@ -215,15 +144,10 @@ if (rtc_include_tests) {
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_native_code" ]
       shard_timeout = 900
-      data = [
-        "//resources/utility/encapsulated_pcm16b_8khz.wav",
-        "//resources/utility/encapsulated_pcmu_8khz.wav",
-      ]
     }
 
     sources = [
       "channel_unittest.cc",
-      "file_player_unittests.cc",
       "network_predictor_unittest.cc",
       "transmit_mixer_unittest.cc",
       "utility_unittest.cc",

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/modules/utility/include/file_player.h"
-#include "webrtc/voice_engine/file_recorder.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/modules/utility/include/file_player.h"
-#include "webrtc/voice_engine/file_recorder.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

@@ -29,8 +29,6 @@
         '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
         '<(webrtc_root)/webrtc.gyp:rtc_event_log',
-        'file_player',
-        'file_recorder',
         'level_indicator',
       ],
       'export_dependent_settings': [
@@ -96,38 +94,6 @@
         'voice_engine_impl.h',
       ],
     },
-    {
-      'target_name': 'audio_coder',
-      'type': 'static_library',
-      'sources': [
-        'coder.cc',
-        'coder.h',
-      ],
-    },
-    {
-      'target_name': 'file_player',
-      'type': 'static_library',
-      'sources': [
-        'file_player.h',
-        'file_player_impl.cc',
-        'file_player_impl.h',
-      ],
-      'dependencies': [
-        'audio_coder',
-      ],
-    },
-    {
-      'target_name': 'file_recorder',
-      'type': 'static_library',
-      'sources': [
-        'file_recorder.h',
-        'file_recorder_impl.cc',
-        'file_recorder_impl.h',
-      ],
-      'dependencies': [
-        'audio_coder',
-      ],
-    },
     {
       'target_name': 'level_indicator',
       'type': 'static_library',
@@ -155,7 +121,6 @@
             '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',
@@ -171,7 +136,6 @@
           ],
           'sources': [
             'channel_unittest.cc',
-            'file_player_unittests.cc',
             'network_predictor_unittest.cc',
             'transmit_mixer_unittest.cc',
             'utility_unittest.cc',
@@ -188,12 +152,6 @@
                 '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
               ],
             }],
-            ['OS=="ios"', {
-              'mac_bundle_resources': [
-                '<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav',
-                '<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav',
-              ],
-            }],
           ],
         },
         {

webrtc/voice_engine/voice_engine_unittests.isolate

@@ -19,13 +19,5 @@
         ],
       },
     }],
-    ['OS=="linux" or OS=="mac" or OS=="win" or OS=="android"', {
-      'variables': {
-        'files': [
-          '<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav',
-          '<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav',
-        ],
-      },
-    }],
   ],
 }