
Removes a lot of code that prevents refactoring VideoCodingModule. Tests covering the module should be TEST_Fs, and this looks like like fairly unused code in general. Adds a 'rtp_player' binary which performs a small subset. BUG=4391 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/44559004 Cr-Commit-Position: refs/heads/master@{#8787} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8787 4adac7df-926f-26a2-2b94-8c16560cd09d
36 lines
1.2 KiB
Python
36 lines
1.2 KiB
Python
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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{
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'targets': [
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{
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'target_name': 'rtp_player',
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'type': 'executable',
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'dependencies': [
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'rtp_rtcp',
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'webrtc_video_coding',
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'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
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'<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
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],
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'sources': [
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# headers
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'main/test/receiver_tests.h',
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'main/test/rtp_player.h',
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'main/test/vcm_payload_sink_factory.h',
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# sources
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'main/test/rtp_player.cc',
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'main/test/test_util.cc',
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'main/test/tester_main.cc',
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'main/test/vcm_payload_sink_factory.cc',
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'main/test/video_rtp_play.cc',
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], # sources
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},
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],
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}
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