Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(233)

Side by Side Diff: webrtc/modules/video_coding/video_coding_test.gypi

Issue 1417283007: modules/video_coding refactorings (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix the other copy of the mock include header Created 5 years, 1 month ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 # Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
2 # Use of this source code is governed by a BSD-style license 2 # Use of this source code is governed by a BSD-style license
3 # that can be found in the LICENSE file in the root of the source 3 # that can be found in the LICENSE file in the root of the source
4 # tree. An additional intellectual property rights grant can be found 4 # tree. An additional intellectual property rights grant can be found
5 # in the file PATENTS. All contributing project authors may 5 # in the file PATENTS. All contributing project authors may
6 # be found in the AUTHORS file in the root of the source tree. 6 # be found in the AUTHORS file in the root of the source tree.
7 7
8 { 8 {
9 'targets': [ 9 'targets': [
10 { 10 {
11 'target_name': 'rtp_player', 11 'target_name': 'rtp_player',
12 'type': 'executable', 12 'type': 'executable',
13 'dependencies': [ 13 'dependencies': [
14 'rtp_rtcp', 14 'rtp_rtcp',
15 'webrtc_video_coding', 15 'webrtc_video_coding',
16 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', 16 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
17 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_def ault', 17 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_def ault',
18 '<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common', 18 '<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
19 ], 19 ],
20 'sources': [ 20 'sources': [
21 # headers 21 # headers
22 'main/test/receiver_tests.h', 22 'test/receiver_tests.h',
23 'main/test/rtp_player.h', 23 'test/rtp_player.h',
24 'main/test/vcm_payload_sink_factory.h', 24 'test/vcm_payload_sink_factory.h',
25 25
26 # sources 26 # sources
27 'main/test/rtp_player.cc', 27 'test/rtp_player.cc',
28 'main/test/test_util.cc', 28 'test/test_util.cc',
29 'main/test/tester_main.cc', 29 'test/tester_main.cc',
30 'main/test/vcm_payload_sink_factory.cc', 30 'test/vcm_payload_sink_factory.cc',
31 'main/test/video_rtp_play.cc', 31 'test/video_rtp_play.cc',
32 ], # sources 32 ], # sources
33 }, 33 },
34 ], 34 ],
35 } 35 }
OLDNEW
« no previous file with comments | « webrtc/modules/video_coding/video_coding_robustness_unittest.cc ('k') | webrtc/modules/video_coding/video_receiver.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698