webrtc/webrtc/modules/video_coding/video_coding_test.gypi
pbos@webrtc.org d21406d333 Remove command-line tool 'video_coding_test'.
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
2015-03-19 08:19:44 +00:00

36 lines
1.2 KiB
Python

# 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.
{
'targets': [
{
'target_name': 'rtp_player',
'type': 'executable',
'dependencies': [
'rtp_rtcp',
'webrtc_video_coding',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
],
'sources': [
# headers
'main/test/receiver_tests.h',
'main/test/rtp_player.h',
'main/test/vcm_payload_sink_factory.h',
# sources
'main/test/rtp_player.cc',
'main/test/test_util.cc',
'main/test/tester_main.cc',
'main/test/vcm_payload_sink_factory.cc',
'main/test/video_rtp_play.cc',
], # sources
},
],
}