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1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
2 # | 2 # |
3 # Use of this source code is governed by a BSD-style license | 3 # Use of this source code is governed by a BSD-style license |
4 # that can be found in the LICENSE file in the root of the source | 4 # that can be found in the LICENSE file in the root of the source |
5 # tree. An additional intellectual property rights grant can be found | 5 # tree. An additional intellectual property rights grant can be found |
6 # in the file PATENTS. All contributing project authors may | 6 # in the file PATENTS. All contributing project authors may |
7 # be found in the AUTHORS file in the root of the source tree. | 7 # be found in the AUTHORS file in the root of the source tree. |
8 | 8 |
9 import("../webrtc.gni") | 9 import("../webrtc.gni") |
10 | 10 |
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260 "video_stream_encoder_unittest.cc", | 260 "video_stream_encoder_unittest.cc", |
261 ] | 261 ] |
262 deps = [ | 262 deps = [ |
263 ":video", | 263 ":video", |
264 "..:video_stream_api", | 264 "..:video_stream_api", |
265 "../api:video_frame_api", | 265 "../api:video_frame_api", |
266 "../api/video_codecs:video_codecs_api", | 266 "../api/video_codecs:video_codecs_api", |
267 "../call:call_interfaces", | 267 "../call:call_interfaces", |
268 "../call:mock_rtp_interfaces", | 268 "../call:mock_rtp_interfaces", |
269 "../call:rtp_receiver", | 269 "../call:rtp_receiver", |
| 270 "../call:rtp_sender", |
270 "../common_video", | 271 "../common_video", |
271 "../logging:rtc_event_log_api", | 272 "../logging:rtc_event_log_api", |
272 "../media:rtc_media", | 273 "../media:rtc_media", |
273 "../media:rtc_media_base", | 274 "../media:rtc_media_base", |
274 "../media:rtc_media_tests_utils", | 275 "../media:rtc_media_tests_utils", |
275 "../modules:module_api", | 276 "../modules:module_api", |
276 "../modules/pacing", | 277 "../modules/pacing", |
277 "../modules/rtp_rtcp", | 278 "../modules/rtp_rtcp", |
278 "../modules/rtp_rtcp:mock_rtp_rtcp", | 279 "../modules/rtp_rtcp:mock_rtp_rtcp", |
279 "../modules/utility", | 280 "../modules/utility", |
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300 ] | 301 ] |
301 if (!build_with_chromium && is_clang) { | 302 if (!build_with_chromium && is_clang) { |
302 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 303 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
303 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 304 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
304 } | 305 } |
305 if (rtc_use_h264) { | 306 if (rtc_use_h264) { |
306 defines += [ "WEBRTC_USE_H264" ] | 307 defines += [ "WEBRTC_USE_H264" ] |
307 } | 308 } |
308 } | 309 } |
309 } | 310 } |
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