Move FilePlayer and FileRecorder to Voice Engine

webrtc/voice_engine/voice_engine.gyp

 # Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 {
   'includes': [
@@ skipping 11 matching lines @@
         '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
         '<(webrtc_root)/modules/modules.gyp:audio_device',
         '<(webrtc_root)/modules/modules.gyp:audio_processing',
         '<(webrtc_root)/modules/modules.gyp:bitrate_controller',
         '<(webrtc_root)/modules/modules.gyp:media_file',
         '<(webrtc_root)/modules/modules.gyp:paced_sender',
         '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
         '<(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': [
         '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
       ],
       'sources': [
         'include/voe_audio_processing.h',
         'include/voe_base.h',
         'include/voe_codec.h',
         'include/voe_errors.h',
@@ skipping 46 matching lines @@
         'voe_video_sync_impl.cc',
         'voe_video_sync_impl.h',
         'voe_volume_control_impl.cc',
         'voe_volume_control_impl.h',
         'voice_engine_defines.h',
         'voice_engine_impl.cc',
         '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',
       'dependencies': [
         '<(webrtc_root)/base/base.gyp:rtc_base_approved',
         '<(webrtc_root)/common.gyp:webrtc_common',
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
       ],
       'sources': [
         'level_indicator.cc',
         'level_indicator.h',
       ]
     }
   ],
   'conditions': [
     ['OS=="win"', {
       'defines': ['WEBRTC_DRIFT_COMPENSATION_SUPPORTED',],
     }],
     ['include_tests==1', {
       'targets': [
         {
           'target_name': 'voice_engine_unittests',
           'type': '<(gtest_target_type)',
           'dependencies': [
             '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',
             '<(webrtc_root)/modules/modules.gyp:audio_device',
             '<(webrtc_root)/modules/modules.gyp:audio_processing',
             '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
             '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
             '<(webrtc_root)/modules/modules.gyp:media_file',
             '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
             '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
             '<(webrtc_root)/test/test.gyp:test_support_main',
           ],
           'sources': [
             'channel_unittest.cc',
+            'file_player_unittests.cc',
             'network_predictor_unittest.cc',
             'transmit_mixer_unittest.cc',
             'utility_unittest.cc',
             'voe_audio_processing_unittest.cc',
             'voe_base_unittest.cc',
             'voe_codec_unittest.cc',
             'voe_network_unittest.cc',
             'voice_engine_fixture.cc',
             'voice_engine_fixture.h',
           ],
           'conditions': [
             ['OS=="android"', {
               'dependencies': [
                 '<(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',
+              ],
+            }],
           ],
         },
         {
           # command line test that should work on linux/mac/win
           'target_name': 'voe_cmd_test',
           'type': 'executable',
           'dependencies': [
             'voice_engine',
             '<(DEPTH)/testing/gtest.gyp:gtest',
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
@@ skipping 151 matching lines @@
               'sources': [
                 'voe_auto_test.isolate',
               ],
             },
           ],
         }],
       ],  # conditions
     }], # include_tests==1
   ], # conditions
 }