kjellander@webrtc.org 4298f73031 Revert 4547 "Isolate GYP target and .isolate files for tests"
As this breaks the FYI bots in 
http://build.chromium.org/p/chromium.webrtc.fyi/waterfall
due to different path to isolate.gypi (which cannot easily
be resolved due to limitations in GYP)

> Isolate GYP target and .isolate files for tests
> 
> Implemented according to the instructions at
> http://www.chromium.org/developers/testing/isolated-testing
> 
> Workflow has been like this:
> 1. create _run GYP target
> 2. create a stripped down .isolate file
> 3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
> 4. runhooks
> 5. compile
> 6. test if the test would run (i.e. find it's dependencies) without
>    actually executing it:
>    tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
> 7. If failing, run the fix_test_cases.py script like this:
>    tools/swarm_client/fix_test_cases.py --isolated out/Release/testname.isolated
> 
> All tests that run on the bots for WebRTC has got _run target
> and .isolate file created.
> 
> "Normal tests" that run fine on any machine:
> * audio_decoder_unittests
> * common_audio_unittests
> * common_video_unittests
> * metrics_unittests
> * modules_integrationtests
> * modules_unittests
> * neteq_unittests
> * system_wrappers_unittests
> * test_support_unittests
> * tools_unittests
> * video_engine_core_unittests
> * voice_engine_unittests
> 
> Tests that requires bare-metal and audio/video devices:
> * audio_device_integrationtests
> * video_capture_integrationtests
> 
> I also added the isolate boilerplate code for the following
> tests that are not yet pure gtest binaries (which means they
> cannot run isolated yet):
> * video_render_integrationtests
> * vie_auto_test
> * voe_auto_test
> 
> TEST=running isolate.py as described above.
> BUG=1916
> R=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1673004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2040004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4548 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 11:29:58 +00:00

211 lines
6.2 KiB
Python

# Copyright (c) 2012 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.
{
'variables': {
'neteq_dependencies': [
'G711',
'G722',
'PCM16B',
'iLBC',
'iSAC',
'iSACFix',
'CNG',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
],
'neteq_defines': [],
'conditions': [
['include_opus==1', {
'neteq_dependencies': ['webrtc_opus',],
'neteq_defines': ['WEBRTC_CODEC_OPUS',],
}],
],
},
'targets': [
{
'target_name': 'NetEq4',
'type': 'static_library',
'dependencies': [
'<@(neteq_dependencies)',
],
'defines': [
'<@(neteq_defines)',
],
'include_dirs': [
'interface',
],
'direct_dependent_settings': {
'include_dirs': [
'interface',
],
},
'sources': [
'interface/audio_decoder.h',
'interface/neteq.h',
'accelerate.cc',
'accelerate.h',
'audio_decoder_impl.cc',
'audio_decoder_impl.h',
'audio_decoder.cc',
'audio_multi_vector.cc',
'audio_multi_vector.h',
'audio_vector.cc',
'audio_vector.h',
'background_noise.cc',
'background_noise.h',
'buffer_level_filter.cc',
'buffer_level_filter.h',
'comfort_noise.cc',
'comfort_noise.h',
'decision_logic.cc',
'decision_logic.h',
'decision_logic_fax.cc',
'decision_logic_fax.h',
'decision_logic_normal.cc',
'decision_logic_normal.h',
'decoder_database.cc',
'decoder_database.h',
'defines.h',
'delay_manager.cc',
'delay_manager.h',
'delay_peak_detector.cc',
'delay_peak_detector.h',
'dsp_helper.cc',
'dsp_helper.h',
'dtmf_buffer.cc',
'dtmf_buffer.h',
'dtmf_tone_generator.cc',
'dtmf_tone_generator.h',
'expand.cc',
'expand.h',
'merge.cc',
'merge.h',
'neteq_impl.cc',
'neteq_impl.h',
'neteq.cc',
'statistics_calculator.cc',
'statistics_calculator.h',
'normal.cc',
'normal.h',
'packet_buffer.cc',
'packet_buffer.h',
'payload_splitter.cc',
'payload_splitter.h',
'post_decode_vad.cc',
'post_decode_vad.h',
'preemptive_expand.cc',
'preemptive_expand.h',
'random_vector.cc',
'random_vector.h',
'rtcp.cc',
'rtcp.h',
'sync_buffer.cc',
'sync_buffer.h',
'timestamp_scaler.cc',
'timestamp_scaler.h',
'time_stretch.cc',
'time_stretch.h',
],
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
4267, # size_t to int truncation.
],
},
], # targets
'conditions': [
['include_tests==1', {
'includes': ['neteq_tests.gypi',],
'targets': [
{
'target_name': 'audio_decoder_unittests',
'type': '<(gtest_target_type)',
'dependencies': [
'<@(neteq_dependencies)',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/test/test.gyp:test_support_main',
],
'defines': [
'AUDIO_DECODER_UNITTEST',
'WEBRTC_CODEC_G722',
'WEBRTC_CODEC_ILBC',
'WEBRTC_CODEC_ISACFX',
'WEBRTC_CODEC_ISAC',
'WEBRTC_CODEC_PCM16',
'<@(neteq_defines)',
],
'sources': [
'audio_decoder_impl.cc',
'audio_decoder_impl.h',
'audio_decoder_unittest.cc',
'audio_decoder.cc',
'interface/audio_decoder.h',
],
'conditions': [
# TODO(henrike): remove build_with_chromium==1 when the bots are
# using Chromium's buildbots.
['build_with_chromium==1 and OS=="android" and gtest_target_type=="shared_library"', {
'dependencies': [
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
],
}],
],
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
4267, # size_t to int truncation.
],
}, # audio_decoder_unittests
{
'target_name': 'neteq_unittest_tools',
'type': 'static_library',
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/test/test.gyp:test_support_main',
],
'direct_dependent_settings': {
'include_dirs': [
'tools',
],
},
'include_dirs': [
'tools',
],
'sources': [
'tools/input_audio_file.cc',
'tools/input_audio_file.h',
'tools/rtp_generator.cc',
'tools/rtp_generator.h',
],
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
4267, # size_t to int truncation.
],
}, # neteq_unittest_tools
], # targets
'conditions': [
# TODO(henrike): remove build_with_chromium==1 when the bots are using
# Chromium's buildbots.
['build_with_chromium==1 and OS=="android" and gtest_target_type=="shared_library"', {
'targets': [
{
'target_name': 'audio_decoder_unittests_apk_target',
'type': 'none',
'dependencies': [
'<(apk_tests_path):audio_decoder_unittests_apk',
],
},
],
}],
],
}], # include_tests
], # conditions
}