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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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253 "vie_encoder_unittest.cc", | 253 "vie_encoder_unittest.cc", |
254 ] | 254 ] |
255 deps = [ | 255 deps = [ |
256 ":video", | 256 ":video", |
257 "..:video_stream_api", | 257 "..:video_stream_api", |
258 "../api:video_frame_api", | 258 "../api:video_frame_api", |
259 "../api/video_codecs:video_codecs_api", | 259 "../api/video_codecs:video_codecs_api", |
260 "../base:rtc_base_approved", | 260 "../base:rtc_base_approved", |
261 "../base:rtc_base_tests_utils", | 261 "../base:rtc_base_tests_utils", |
262 "../call:call_interfaces", | 262 "../call:call_interfaces", |
| 263 "../call:rtp_receiver", |
263 "../common_video", | 264 "../common_video", |
264 "../logging:rtc_event_log_api", | 265 "../logging:rtc_event_log_api", |
265 "../media:rtc_media", | 266 "../media:rtc_media", |
266 "../media:rtc_media_base", | 267 "../media:rtc_media_base", |
267 "../media:rtc_media_tests_utils", | 268 "../media:rtc_media_tests_utils", |
268 "../modules:module_api", | 269 "../modules:module_api", |
269 "../modules/pacing", | 270 "../modules/pacing", |
270 "../modules/rtp_rtcp", | 271 "../modules/rtp_rtcp", |
271 "../modules/rtp_rtcp:mock_rtp_rtcp", | 272 "../modules/rtp_rtcp:mock_rtp_rtcp", |
272 "../modules/utility", | 273 "../modules/utility", |
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290 ] | 291 ] |
291 if (!build_with_chromium && is_clang) { | 292 if (!build_with_chromium && is_clang) { |
292 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 293 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
293 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 294 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
294 } | 295 } |
295 if (rtc_use_h264) { | 296 if (rtc_use_h264) { |
296 defines += [ "WEBRTC_USE_H264" ] | 297 defines += [ "WEBRTC_USE_H264" ] |
297 } | 298 } |
298 } | 299 } |
299 } | 300 } |
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