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Side by Side Diff: webrtc/video/BUILD.gn

Issue 2886993005: Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface. (Closed)
Patch Set: Return DELIVERY_UNKNOWN_SSRC, not DELIVERY_PACKET_ERROR, when receive_rtp_config_ lookup fails. Created 3 years, 6 months ago
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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
(...skipping 239 matching lines...) Expand 10 before | Expand all | Expand 10 after
250 "vie_encoder_unittest.cc", 250 "vie_encoder_unittest.cc",
251 ] 251 ]
252 deps = [ 252 deps = [
253 ":video", 253 ":video",
254 "..:video_stream_api", 254 "..:video_stream_api",
255 "../api:video_frame_api", 255 "../api:video_frame_api",
256 "../api/video_codecs:video_codecs_api", 256 "../api/video_codecs:video_codecs_api",
257 "../base:rtc_base_approved", 257 "../base:rtc_base_approved",
258 "../base:rtc_base_tests_utils", 258 "../base:rtc_base_tests_utils",
259 "../call:call_interfaces", 259 "../call:call_interfaces",
260 "../call:rtp_receiver",
260 "../common_video", 261 "../common_video",
261 "../logging:rtc_event_log_api", 262 "../logging:rtc_event_log_api",
262 "../media:rtc_media", 263 "../media:rtc_media",
263 "../media:rtc_media_base", 264 "../media:rtc_media_base",
264 "../media:rtc_media_tests_utils", 265 "../media:rtc_media_tests_utils",
265 "../modules:module_api", 266 "../modules:module_api",
266 "../modules/pacing", 267 "../modules/pacing",
267 "../modules/rtp_rtcp", 268 "../modules/rtp_rtcp",
268 "../modules/rtp_rtcp:mock_rtp_rtcp", 269 "../modules/rtp_rtcp:mock_rtp_rtcp",
269 "../modules/utility", 270 "../modules/utility",
(...skipping 17 matching lines...) Expand all
287 ] 288 ]
288 if (!build_with_chromium && is_clang) { 289 if (!build_with_chromium && is_clang) {
289 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 290 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
290 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 291 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
291 } 292 }
292 if (rtc_use_h264) { 293 if (rtc_use_h264) {
293 defines += [ "WEBRTC_USE_H264" ] 294 defines += [ "WEBRTC_USE_H264" ]
294 } 295 }
295 } 296 }
296 } 297 }
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