5b88317820
Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
99 lines
2.9 KiB
Python
99 lines
2.9 KiB
Python
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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{
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'conditions': [
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['include_tests==1', {
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'includes': [
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'libjingle/xmllite/xmllite_tests.gypi',
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'libjingle/xmpp/xmpp_tests.gypi',
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'p2p/p2p_tests.gypi',
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'sound/sound_tests.gypi',
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'webrtc_tests.gypi',
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],
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}],
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],
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'includes': [
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'build/common.gypi',
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'video/webrtc_video.gypi',
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],
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'variables': {
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'webrtc_all_dependencies': [
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'base/base.gyp:*',
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'sound/sound.gyp:*',
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'common.gyp:*',
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'common_audio/common_audio.gyp:*',
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'common_video/common_video.gyp:*',
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'libjingle/xmllite/xmllite.gyp:*',
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'libjingle/xmpp/xmpp.gyp:*',
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'modules/modules.gyp:*',
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'p2p/p2p.gyp:*',
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'system_wrappers/source/system_wrappers.gyp:*',
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'video_engine/video_engine.gyp:*',
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'voice_engine/voice_engine.gyp:*',
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'<(webrtc_vp8_dir)/vp8.gyp:*',
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'<(webrtc_vp9_dir)/vp9.gyp:*',
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],
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},
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'targets': [
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{
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'target_name': 'webrtc_all',
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'type': 'none',
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'dependencies': [
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'<@(webrtc_all_dependencies)',
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'webrtc',
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],
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'conditions': [
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['include_tests==1', {
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'dependencies': [
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'common_video/common_video_unittests.gyp:*',
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'system_wrappers/source/system_wrappers_tests.gyp:*',
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'test/metrics.gyp:*',
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'test/test.gyp:*',
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'test/webrtc_test_common.gyp:webrtc_test_common_unittests',
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'tools/tools.gyp:*',
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'webrtc_tests',
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'rtc_unittests',
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],
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}],
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],
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},
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{
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# TODO(pbos): This is intended to contain audio parts as well as soon as
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# VoiceEngine moves to the same new API format.
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'target_name': 'webrtc',
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'type': 'static_library',
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'sources': [
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'call.h',
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'config.h',
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'experiments.h',
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'frame_callback.h',
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'transport.h',
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'video_receive_stream.h',
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'video_renderer.h',
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'video_send_stream.h',
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'<@(webrtc_video_sources)',
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],
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'dependencies': [
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'common.gyp:*',
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'<@(webrtc_video_dependencies)',
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],
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'conditions': [
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# TODO(andresp): Chromium libpeerconnection should link directly with
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# this and no if conditions should be needed on webrtc build files.
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['build_with_chromium==1', {
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'dependencies': [
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'<(webrtc_root)/modules/modules.gyp:video_capture_module_impl',
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'<(webrtc_root)/modules/modules.gyp:video_render_module_impl',
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],
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}],
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],
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},
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],
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}
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