Refactor neteq_rtpplay

webrtc/modules/audio_coding/neteq/neteq.gypi

 # Copyright (c) 2012 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.
 
 {
   'variables': {
@@ skipping 158 matching lines @@
             }],
             ['OS=="ios"', {
               'mac_bundle_resources': [
                 '<(DEPTH)/resources/audio_coding/testfile32kHz.pcm',
               ],
             }],
           ],
         }, # audio_decoder_unittests
 
         {
+          'target_name': 'rtc_event_log_source',
+          'type': 'static_library',
+          'dependencies': [
+            '<(webrtc_root)/webrtc.gyp:rtc_event_log_parser',
+            '<(webrtc_root)/webrtc.gyp:rtc_event_log_proto',
+          ],
+          'export_dependent_settings': [
+            '<(webrtc_root)/webrtc.gyp:rtc_event_log_parser',
+          ],
+          'sources': [
+            'tools/rtc_event_log_source.h',
+            'tools/rtc_event_log_source.cc',
+          ],
+        },
+
+        {
+          'target_name': 'neteq_unittest_proto',
+          'type': 'static_library',
+          'sources': [
+            'neteq_unittest.proto',
+          ],
+          'variables': {
+            'proto_in_dir': '.',
+            # Workaround to protect against gyp's pathname relativization when
+            # this file is included by modules.gyp.
+            'proto_out_protected': 'webrtc/audio_coding/neteq',
+            'proto_out_dir': '<(proto_out_protected)',
+          },
+          'includes': ['../../../build/protoc.gypi',],
+        },
+
+        {
           'target_name': 'neteq_unittest_tools',
           'type': 'static_library',
           'dependencies': [
+            'neteq',
             'rtp_rtcp',
+            'rtc_event_log_source',
             '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
             '<(webrtc_root)/test/test.gyp:rtp_test_utils',
           ],
           'direct_dependent_settings': {
             'include_dirs': [
               'tools',
             ],
           },
           'include_dirs': [
             'tools',
           ],
           'sources': [
             'tools/audio_checksum.h',
             'tools/audio_loop.cc',
             'tools/audio_loop.h',
             'tools/audio_sink.h',
             'tools/constant_pcm_packet_source.cc',
             'tools/constant_pcm_packet_source.h',
+            'tools/fake_decode_from_file.cc',
+            'tools/fake_decode_from_file.h',
             'tools/input_audio_file.cc',
             'tools/input_audio_file.h',
+            'tools/neteq_input.h',
+            'tools/neteq_packet_source_input.cc',
+            'tools/neteq_packet_source_input.h',
+            'tools/neteq_replacement_input.cc',
+            'tools/neteq_replacement_input.h',
+            'tools/neteq_test.cc',
+            'tools/neteq_test.h',
             'tools/output_audio_file.h',
             'tools/output_wav_file.h',
             'tools/packet.cc',
             'tools/packet.h',
             'tools/packet_source.h',
             'tools/resample_input_audio_file.cc',
             'tools/resample_input_audio_file.h',
             'tools/rtp_file_source.cc',
             'tools/rtp_file_source.h',
             'tools/rtp_generator.cc',
             'tools/rtp_generator.h',
           ],
         }, # neteq_unittest_tools
       ], # targets
     }], # include_tests
   ], # conditions
 }