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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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54 } | 54 } |
55 | 55 |
56 deps = [ | 56 deps = [ |
57 "..:webrtc_common", | 57 "..:webrtc_common", |
58 "../api:optional", | 58 "../api:optional", |
59 "../api:transport_api", | 59 "../api:transport_api", |
60 "../api/video_codecs:video_codecs_api", | 60 "../api/video_codecs:video_codecs_api", |
61 "../call:call_interfaces", | 61 "../call:call_interfaces", |
62 "../call:rtp_interfaces", | 62 "../call:rtp_interfaces", |
63 "../call:video_stream_api", | 63 "../call:video_stream_api", |
64 | |
65 # For RtxReceiveStream. | |
66 "../call:rtp_receiver", | |
67 "../common_video", | 64 "../common_video", |
68 "../logging:rtc_event_log_api", | 65 "../logging:rtc_event_log_api", |
69 "../media:rtc_media_base", | 66 "../media:rtc_media_base", |
70 "../modules:module_api", | 67 "../modules:module_api", |
71 "../modules/bitrate_controller", | 68 "../modules/bitrate_controller", |
72 "../modules/congestion_controller", | 69 "../modules/congestion_controller", |
73 "../modules/pacing", | 70 "../modules/pacing", |
74 "../modules/remote_bitrate_estimator", | 71 "../modules/remote_bitrate_estimator", |
75 "../modules/rtp_rtcp", | 72 "../modules/rtp_rtcp", |
76 "../modules/utility", | 73 "../modules/utility", |
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311 ] | 308 ] |
312 if (!build_with_chromium && is_clang) { | 309 if (!build_with_chromium && is_clang) { |
313 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 310 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
314 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 311 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
315 } | 312 } |
316 if (rtc_use_h264) { | 313 if (rtc_use_h264) { |
317 defines += [ "WEBRTC_USE_H264" ] | 314 defines += [ "WEBRTC_USE_H264" ] |
318 } | 315 } |
319 } | 316 } |
320 } | 317 } |
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