| # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
| # |
| # Use of this source code is governed by a BSD-style license |
| # that can be found in the LICENSE file in the root of the source |
| # tree. An additional intellectual property rights grant can be found |
| # in the file PATENTS. All contributing project authors may |
| # be found in the AUTHORS file in the root of the source tree. |
| |
| import("../build/webrtc.gni") |
| import("//testing/test.gni") |
| |
| source_set("video") { |
| sources = [ |
| "call_stats.cc", |
| "call_stats.h", |
| "encoder_state_feedback.cc", |
| "encoder_state_feedback.h", |
| "overuse_frame_detector.cc", |
| "overuse_frame_detector.h", |
| "payload_router.cc", |
| "payload_router.h", |
| "receive_statistics_proxy.cc", |
| "receive_statistics_proxy.h", |
| "report_block_stats.cc", |
| "report_block_stats.h", |
| "rtp_stream_receiver.cc", |
| "rtp_stream_receiver.h", |
| "rtp_streams_synchronizer.cc", |
| "rtp_streams_synchronizer.h", |
| "send_delay_stats.cc", |
| "send_delay_stats.h", |
| "send_statistics_proxy.cc", |
| "send_statistics_proxy.h", |
| "stats_counter.cc", |
| "stats_counter.h", |
| "stream_synchronization.cc", |
| "stream_synchronization.h", |
| "video_capture_input.cc", |
| "video_capture_input.h", |
| "video_decoder.cc", |
| "video_encoder.cc", |
| "video_receive_stream.cc", |
| "video_receive_stream.h", |
| "video_send_stream.cc", |
| "video_send_stream.h", |
| "video_stream_decoder.cc", |
| "video_stream_decoder.h", |
| "vie_encoder.cc", |
| "vie_encoder.h", |
| "vie_remb.cc", |
| "vie_remb.h", |
| ] |
| |
| configs += [ "..:common_config" ] |
| public_configs = [ "..:common_inherited_config" ] |
| |
| if (is_clang) { |
| # Suppress warnings from Chrome's Clang plugins. |
| # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. |
| configs -= [ "//build/config/clang:find_bad_constructs" ] |
| } |
| |
| deps = [ |
| "..:rtc_event_log", |
| "..:webrtc_common", |
| "../base:rtc_base_approved", |
| "../common_video", |
| "../modules/bitrate_controller", |
| "../modules/congestion_controller", |
| "../modules/pacing", |
| "../modules/remote_bitrate_estimator", |
| "../modules/rtp_rtcp", |
| "../modules/utility", |
| "../modules/video_capture:video_capture_module", |
| "../modules/video_coding", |
| "../modules/video_processing", |
| "../system_wrappers", |
| "../voice_engine", |
| ] |
| } |
| |
| if (rtc_include_tests) { |
| # TODO(pbos): Rename test suite. |
| source_set("video_tests") { |
| testonly = true |
| sources = [ |
| "call_stats_unittest.cc", |
| "encoder_state_feedback_unittest.cc", |
| "end_to_end_tests.cc", |
| "overuse_frame_detector_unittest.cc", |
| "payload_router_unittest.cc", |
| "report_block_stats_unittest.cc", |
| "send_delay_stats_unittest.cc", |
| "send_statistics_proxy_unittest.cc", |
| "stats_counter_unittest.cc", |
| "stream_synchronization_unittest.cc", |
| "video_capture_input_unittest.cc", |
| "video_decoder_unittest.cc", |
| "video_encoder_unittest.cc", |
| "video_send_stream_tests.cc", |
| "vie_remb_unittest.cc", |
| ] |
| configs += [ "..:common_config" ] |
| deps = [ |
| ":video", |
| "//testing/gmock", |
| "//testing/gtest", |
| ] |
| if (is_clang) { |
| # Suppress warnings from the Chromium Clang plugin. |
| # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. |
| configs -= [ "//build/config/clang:find_bad_constructs" ] |
| } |
| } |
| } |