webrtc_m130/webrtc/webrtc_tests.gypi

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# Copyright (c) 2013 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.
{
'targets': [
{
'target_name': 'rtc_unittests',
'type': '<(gtest_target_type)',
'dependencies': [
'base/base.gyp:rtc_base',
'base/base_tests.gyp:rtc_base_tests_utils',
'base/base_tests.gyp:rtc_base_tests',
'libjingle/xmllite/xmllite.gyp:rtc_xmllite',
'libjingle/xmpp/xmpp.gyp:rtc_xmpp',
'p2p/p2p.gyp:rtc_p2p',
'p2p/p2p.gyp:libstunprober',
'rtc_p2p_unittest',
'rtc_sound_tests',
'rtc_xmllite_unittest',
'rtc_xmpp_unittest',
'sound/sound.gyp:rtc_sound',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/testing/gmock.gyp:gmock',
],
'conditions': [
['OS=="android"', {
'dependencies': [
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
],
}],
['OS=="ios"', {
'dependencies': [
'api/api_tests.gyp:rtc_api_objc_tests',
]
}]
],
},
{
'target_name': 'webrtc_tests',
'type': 'none',
'dependencies': [
'video_engine_tests',
'video_loopback',
'video_replay',
'webrtc_perf_tests',
'webrtc_nonparallel_tests',
],
},
{
Refactoring full stack and loopback tests Refactoring full stack, video and screenshare tests to use the same code basis for parametrization and initialization. This patch is done on top of recently commited full stack graphs CL https://codereview.webrtc.org/1289933003/, but virtually no changes have been made to full_stack_plot.py nor to the VideoAnalyzer in full stack, except moving it to video_quality_test.cc. Also, full_stack_samples.cc (build target) was removed and replaced with -output_filename and -duration cmdline arguments in video_loopback and screenshare_loopback. The important things to review: - video_quality_test.h Is the structure of Params good? (examples of usage can be found in full_stack.cc, video_loopback.cc and screenshare_loopback.cc) - video_quality_test.cc Is the initialization correct? The case for using Analyzer and using local renderer are different, can they be further merged? - webrtc_tests.gypi Reproducing the different bitrate settings the full stack and loopback tests had was a little bit tricky. To support both simultaneously, I added BitrateConfig to the Params struct, as well as separate start_bitrate and target_bitrate flags for loopback tests. Note: Side-by-side diff for video_quality_test.cc compares that file directly with the old full_stack.cc, so changes to VideoAnalyzer are clearly visible. Note: Recent CL I've committed added -num_temporal_layers and -sl_discard_threshold args to loopback tests. This was removed here. Support for streams and SVC will be added in a CL following this one. Review URL: https://codereview.webrtc.org/1308403003 Cr-Commit-Position: refs/heads/master@{#9969}
2015-09-17 05:30:24 -07:00
'target_name': 'video_quality_test',
'type': 'static_library',
'sources': [
Refactoring full stack and loopback tests Refactoring full stack, video and screenshare tests to use the same code basis for parametrization and initialization. This patch is done on top of recently commited full stack graphs CL https://codereview.webrtc.org/1289933003/, but virtually no changes have been made to full_stack_plot.py nor to the VideoAnalyzer in full stack, except moving it to video_quality_test.cc. Also, full_stack_samples.cc (build target) was removed and replaced with -output_filename and -duration cmdline arguments in video_loopback and screenshare_loopback. The important things to review: - video_quality_test.h Is the structure of Params good? (examples of usage can be found in full_stack.cc, video_loopback.cc and screenshare_loopback.cc) - video_quality_test.cc Is the initialization correct? The case for using Analyzer and using local renderer are different, can they be further merged? - webrtc_tests.gypi Reproducing the different bitrate settings the full stack and loopback tests had was a little bit tricky. To support both simultaneously, I added BitrateConfig to the Params struct, as well as separate start_bitrate and target_bitrate flags for loopback tests. Note: Side-by-side diff for video_quality_test.cc compares that file directly with the old full_stack.cc, so changes to VideoAnalyzer are clearly visible. Note: Recent CL I've committed added -num_temporal_layers and -sl_discard_threshold args to loopback tests. This was removed here. Support for streams and SVC will be added in a CL following this one. Review URL: https://codereview.webrtc.org/1308403003 Cr-Commit-Position: refs/heads/master@{#9969}
2015-09-17 05:30:24 -07:00
'video/video_quality_test.cc',
'video/video_quality_test.h',
],
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/modules/modules.gyp:video_render',
Refactoring full stack and loopback tests Refactoring full stack, video and screenshare tests to use the same code basis for parametrization and initialization. This patch is done on top of recently commited full stack graphs CL https://codereview.webrtc.org/1289933003/, but virtually no changes have been made to full_stack_plot.py nor to the VideoAnalyzer in full stack, except moving it to video_quality_test.cc. Also, full_stack_samples.cc (build target) was removed and replaced with -output_filename and -duration cmdline arguments in video_loopback and screenshare_loopback. The important things to review: - video_quality_test.h Is the structure of Params good? (examples of usage can be found in full_stack.cc, video_loopback.cc and screenshare_loopback.cc) - video_quality_test.cc Is the initialization correct? The case for using Analyzer and using local renderer are different, can they be further merged? - webrtc_tests.gypi Reproducing the different bitrate settings the full stack and loopback tests had was a little bit tricky. To support both simultaneously, I added BitrateConfig to the Params struct, as well as separate start_bitrate and target_bitrate flags for loopback tests. Note: Side-by-side diff for video_quality_test.cc compares that file directly with the old full_stack.cc, so changes to VideoAnalyzer are clearly visible. Note: Recent CL I've committed added -num_temporal_layers and -sl_discard_threshold args to loopback tests. This was removed here. Support for streams and SVC will be added in a CL following this one. Review URL: https://codereview.webrtc.org/1308403003 Cr-Commit-Position: refs/heads/master@{#9969}
2015-09-17 05:30:24 -07:00
'<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'webrtc',
],
Refactoring full stack and loopback tests Refactoring full stack, video and screenshare tests to use the same code basis for parametrization and initialization. This patch is done on top of recently commited full stack graphs CL https://codereview.webrtc.org/1289933003/, but virtually no changes have been made to full_stack_plot.py nor to the VideoAnalyzer in full stack, except moving it to video_quality_test.cc. Also, full_stack_samples.cc (build target) was removed and replaced with -output_filename and -duration cmdline arguments in video_loopback and screenshare_loopback. The important things to review: - video_quality_test.h Is the structure of Params good? (examples of usage can be found in full_stack.cc, video_loopback.cc and screenshare_loopback.cc) - video_quality_test.cc Is the initialization correct? The case for using Analyzer and using local renderer are different, can they be further merged? - webrtc_tests.gypi Reproducing the different bitrate settings the full stack and loopback tests had was a little bit tricky. To support both simultaneously, I added BitrateConfig to the Params struct, as well as separate start_bitrate and target_bitrate flags for loopback tests. Note: Side-by-side diff for video_quality_test.cc compares that file directly with the old full_stack.cc, so changes to VideoAnalyzer are clearly visible. Note: Recent CL I've committed added -num_temporal_layers and -sl_discard_threshold args to loopback tests. This was removed here. Support for streams and SVC will be added in a CL following this one. Review URL: https://codereview.webrtc.org/1308403003 Cr-Commit-Position: refs/heads/master@{#9969}
2015-09-17 05:30:24 -07:00
'conditions': [
['OS=="android"', {
'dependencies!': [
'<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
],
}],
],
},
{
'target_name': 'video_loopback',
'type': 'executable',
'sources': [
'test/mac/run_test.mm',
'test/run_test.cc',
'test/run_test.h',
'video/video_loopback.cc',
],
'conditions': [
['OS=="mac"', {
'sources!': [
'test/run_test.cc',
],
}],
],
'dependencies': [
Refactoring full stack and loopback tests Refactoring full stack, video and screenshare tests to use the same code basis for parametrization and initialization. This patch is done on top of recently commited full stack graphs CL https://codereview.webrtc.org/1289933003/, but virtually no changes have been made to full_stack_plot.py nor to the VideoAnalyzer in full stack, except moving it to video_quality_test.cc. Also, full_stack_samples.cc (build target) was removed and replaced with -output_filename and -duration cmdline arguments in video_loopback and screenshare_loopback. The important things to review: - video_quality_test.h Is the structure of Params good? (examples of usage can be found in full_stack.cc, video_loopback.cc and screenshare_loopback.cc) - video_quality_test.cc Is the initialization correct? The case for using Analyzer and using local renderer are different, can they be further merged? - webrtc_tests.gypi Reproducing the different bitrate settings the full stack and loopback tests had was a little bit tricky. To support both simultaneously, I added BitrateConfig to the Params struct, as well as separate start_bitrate and target_bitrate flags for loopback tests. Note: Side-by-side diff for video_quality_test.cc compares that file directly with the old full_stack.cc, so changes to VideoAnalyzer are clearly visible. Note: Recent CL I've committed added -num_temporal_layers and -sl_discard_threshold args to loopback tests. This was removed here. Support for streams and SVC will be added in a CL following this one. Review URL: https://codereview.webrtc.org/1308403003 Cr-Commit-Position: refs/heads/master@{#9969}
2015-09-17 05:30:24 -07:00
'video_quality_test',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'test/webrtc_test_common.gyp:webrtc_test_common',
'test/webrtc_test_common.gyp:webrtc_test_renderer',
'test/test.gyp:test_main',
'webrtc',
],
},
{
'target_name': 'screenshare_loopback',
'type': 'executable',
'sources': [
'test/mac/run_test.mm',
'test/run_test.cc',
'test/run_test.h',
'video/screenshare_loopback.cc',
],
'conditions': [
['OS=="mac"', {
'sources!': [
'test/run_test.cc',
],
}],
],
'dependencies': [
Refactoring full stack and loopback tests Refactoring full stack, video and screenshare tests to use the same code basis for parametrization and initialization. This patch is done on top of recently commited full stack graphs CL https://codereview.webrtc.org/1289933003/, but virtually no changes have been made to full_stack_plot.py nor to the VideoAnalyzer in full stack, except moving it to video_quality_test.cc. Also, full_stack_samples.cc (build target) was removed and replaced with -output_filename and -duration cmdline arguments in video_loopback and screenshare_loopback. The important things to review: - video_quality_test.h Is the structure of Params good? (examples of usage can be found in full_stack.cc, video_loopback.cc and screenshare_loopback.cc) - video_quality_test.cc Is the initialization correct? The case for using Analyzer and using local renderer are different, can they be further merged? - webrtc_tests.gypi Reproducing the different bitrate settings the full stack and loopback tests had was a little bit tricky. To support both simultaneously, I added BitrateConfig to the Params struct, as well as separate start_bitrate and target_bitrate flags for loopback tests. Note: Side-by-side diff for video_quality_test.cc compares that file directly with the old full_stack.cc, so changes to VideoAnalyzer are clearly visible. Note: Recent CL I've committed added -num_temporal_layers and -sl_discard_threshold args to loopback tests. This was removed here. Support for streams and SVC will be added in a CL following this one. Review URL: https://codereview.webrtc.org/1308403003 Cr-Commit-Position: refs/heads/master@{#9969}
2015-09-17 05:30:24 -07:00
'video_quality_test',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'test/webrtc_test_common.gyp:webrtc_test_common',
'test/webrtc_test_common.gyp:webrtc_test_renderer',
'test/test.gyp:test_main',
'webrtc',
],
},
{
'target_name': 'video_replay',
'type': 'executable',
'sources': [
'test/mac/run_test.mm',
'test/run_test.cc',
'test/run_test.h',
'video/replay.cc',
],
'conditions': [
['OS=="mac"', {
'sources!': [
'test/run_test.cc',
],
}],
],
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'test/webrtc_test_common.gyp:webrtc_test_common',
'test/webrtc_test_common.gyp:webrtc_test_renderer',
'<(webrtc_root)/modules/modules.gyp:video_capture',
'<(webrtc_root)/modules/modules.gyp:video_render',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'webrtc',
],
},
{
# TODO(solenberg): Rename to webrtc_call_tests.
'target_name': 'video_engine_tests',
'type': '<(gtest_target_type)',
'sources': [
'audio/audio_receive_stream_unittest.cc',
'audio/audio_send_stream_unittest.cc',
'audio/audio_state_unittest.cc',
'call/bitrate_allocator_unittest.cc',
'call/bitrate_estimator_tests.cc',
'call/call_unittest.cc',
'call/packet_injection_tests.cc',
'test/common_unittest.cc',
'test/testsupport/metrics/video_metrics_unittest.cc',
'video/call_stats_unittest.cc',
'video/encoder_state_feedback_unittest.cc',
'video/end_to_end_tests.cc',
'video/overuse_frame_detector_unittest.cc',
'video/payload_router_unittest.cc',
'video/report_block_stats_unittest.cc',
'video/send_statistics_proxy_unittest.cc',
'video/stream_synchronization_unittest.cc',
'video/video_capture_input_unittest.cc',
'video/video_decoder_unittest.cc',
'video/video_encoder_unittest.cc',
'video/video_send_stream_tests.cc',
'video/vie_remb_unittest.cc',
],
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
'<(webrtc_root)/modules/modules.gyp:video_capture',
'<(webrtc_root)/modules/modules.gyp:video_render',
'<(webrtc_root)/test/test.gyp:channel_transport',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
'test/metrics.gyp:metrics',
'test/test.gyp:test_main',
'test/webrtc_test_common.gyp:webrtc_test_common',
'webrtc',
],
'conditions': [
['rtc_use_h264==1', {
'defines': [
'WEBRTC_END_TO_END_H264_TESTS',
],
}],
['OS=="android"', {
'dependencies': [
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
],
}],
['enable_protobuf==1', {
'defines': [
'ENABLE_RTC_EVENT_LOG',
],
'dependencies': [
'webrtc.gyp:rtc_event_log',
'webrtc.gyp:rtc_event_log_proto',
],
'sources': [
'call/rtc_event_log_unittest.cc',
],
}],
],
},
{
'target_name': 'webrtc_perf_tests',
'type': '<(gtest_target_type)',
'sources': [
'call/call_perf_tests.cc',
'call/rampup_tests.cc',
'call/rampup_tests.h',
'modules/audio_coding/neteq/test/neteq_performance_unittest.cc',
'modules/audio_processing/audio_processing_performance_unittest.cc',
'modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc',
'video/full_stack.cc',
],
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/modules/modules.gyp:audio_processing',
'<(webrtc_root)/modules/modules.gyp:audioproc_test_utils',
'<(webrtc_root)/modules/modules.gyp:video_capture',
'<(webrtc_root)/test/test.gyp:channel_transport',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
Refactoring full stack and loopback tests Refactoring full stack, video and screenshare tests to use the same code basis for parametrization and initialization. This patch is done on top of recently commited full stack graphs CL https://codereview.webrtc.org/1289933003/, but virtually no changes have been made to full_stack_plot.py nor to the VideoAnalyzer in full stack, except moving it to video_quality_test.cc. Also, full_stack_samples.cc (build target) was removed and replaced with -output_filename and -duration cmdline arguments in video_loopback and screenshare_loopback. The important things to review: - video_quality_test.h Is the structure of Params good? (examples of usage can be found in full_stack.cc, video_loopback.cc and screenshare_loopback.cc) - video_quality_test.cc Is the initialization correct? The case for using Analyzer and using local renderer are different, can they be further merged? - webrtc_tests.gypi Reproducing the different bitrate settings the full stack and loopback tests had was a little bit tricky. To support both simultaneously, I added BitrateConfig to the Params struct, as well as separate start_bitrate and target_bitrate flags for loopback tests. Note: Side-by-side diff for video_quality_test.cc compares that file directly with the old full_stack.cc, so changes to VideoAnalyzer are clearly visible. Note: Recent CL I've committed added -num_temporal_layers and -sl_discard_threshold args to loopback tests. This was removed here. Support for streams and SVC will be added in a CL following this one. Review URL: https://codereview.webrtc.org/1308403003 Cr-Commit-Position: refs/heads/master@{#9969}
2015-09-17 05:30:24 -07:00
'video_quality_test',
'modules/modules.gyp:neteq_test_support',
'modules/modules.gyp:bwe_simulator',
'modules/modules.gyp:rtp_rtcp',
'test/test.gyp:test_main',
'test/webrtc_test_common.gyp:webrtc_test_common',
Refactoring full stack and loopback tests Refactoring full stack, video and screenshare tests to use the same code basis for parametrization and initialization. This patch is done on top of recently commited full stack graphs CL https://codereview.webrtc.org/1289933003/, but virtually no changes have been made to full_stack_plot.py nor to the VideoAnalyzer in full stack, except moving it to video_quality_test.cc. Also, full_stack_samples.cc (build target) was removed and replaced with -output_filename and -duration cmdline arguments in video_loopback and screenshare_loopback. The important things to review: - video_quality_test.h Is the structure of Params good? (examples of usage can be found in full_stack.cc, video_loopback.cc and screenshare_loopback.cc) - video_quality_test.cc Is the initialization correct? The case for using Analyzer and using local renderer are different, can they be further merged? - webrtc_tests.gypi Reproducing the different bitrate settings the full stack and loopback tests had was a little bit tricky. To support both simultaneously, I added BitrateConfig to the Params struct, as well as separate start_bitrate and target_bitrate flags for loopback tests. Note: Side-by-side diff for video_quality_test.cc compares that file directly with the old full_stack.cc, so changes to VideoAnalyzer are clearly visible. Note: Recent CL I've committed added -num_temporal_layers and -sl_discard_threshold args to loopback tests. This was removed here. Support for streams and SVC will be added in a CL following this one. Review URL: https://codereview.webrtc.org/1308403003 Cr-Commit-Position: refs/heads/master@{#9969}
2015-09-17 05:30:24 -07:00
'test/webrtc_test_common.gyp:webrtc_test_renderer',
'webrtc',
],
'conditions': [
['OS=="android"', {
'dependencies': [
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
],
}],
],
},
{
'target_name': 'webrtc_nonparallel_tests',
'type': '<(gtest_target_type)',
'sources': [
'base/nullsocketserver_unittest.cc',
'base/physicalsocketserver_unittest.cc',
'base/socket_unittest.cc',
'base/socket_unittest.h',
'base/socketaddress_unittest.cc',
'base/virtualsocket_unittest.cc',
],
'defines': [
'GTEST_RELATIVE_PATH',
],
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'base/base.gyp:rtc_base',
'test/test.gyp:test_main',
],
'conditions': [
['OS=="android"', {
'dependencies': [
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
],
}],
['OS=="win"', {
'sources': [
'base/win32socketserver_unittest.cc',
],
'sources!': [
# TODO(ronghuawu): Fix TestUdpReadyToSendIPv6 on windows bot
# then reenable these tests.
# TODO(pbos): Move test disabling to ifdefs within the test files
# instead of here.
'base/physicalsocketserver_unittest.cc',
'base/socket_unittest.cc',
'base/win32socketserver_unittest.cc',
],
}],
['OS=="mac"', {
'sources': [
'base/macsocketserver_unittest.cc',
],
}],
['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
'defines': [
'CARBON_DEPRECATED=YES',
],
}],
],
},
],
'conditions': [
['OS=="android"', {
'targets': [
{
'target_name': 'rtc_unittests_apk_target',
'type': 'none',
'dependencies': [
'<(apk_tests_path):rtc_unittests_apk',
],
},
{
'target_name': 'video_engine_tests_apk_target',
'type': 'none',
'dependencies': [
'<(apk_tests_path):video_engine_tests_apk',
],
},
{
'target_name': 'webrtc_perf_tests_apk_target',
'type': 'none',
'dependencies': [
'<(apk_tests_path):webrtc_perf_tests_apk',
],
},
{
'target_name': 'webrtc_nonparallel_tests_apk_target',
'type': 'none',
'dependencies': [
'<(apk_tests_path):webrtc_nonparallel_tests_apk',
],
},
],
}],
['test_isolation_mode != "noop"', {
'targets': [
{
'target_name': 'rtc_unittests_run',
'type': 'none',
'dependencies': [
'rtc_unittests',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'rtc_unittests.isolate',
],
Move talk/media to webrtc/media I removed the 'libjingle' target in talk/libjingle.gyp and replaced all users of it with base/base.gyp:rtc_base. It seems the jsoncpp and expat dependencies were not used by it's previous references. The files in talk/media/testdata were uploaded to Google Storage and added .sha1 files in resources/media instead of simply moving them. The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL in order to not break Git history. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc. * Unused GYP reference to libjingle_tests_additional_deps was removed. * Removed duplicated GYP entries of webrtc/base/testutils.cc webrtc/base/testutils.h The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media, so they were moved to the media.gyp. I also checked that none of EXPAT_RELATIVE_PATH, FEATURE_ENABLE_VOICEMAIL, GTEST_RELATIVE_PATH, JSONCPP_RELATIVE_PATH, LOGGING=1, SRTP_RELATIVE_PATH, FEATURE_ENABLE_SSL, FEATURE_ENABLE_VOICEMAIL, FEATURE_ENABLE_PSTN, HAVE_SCTP, HAVE_SRTP, are used by the talk/media code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/ BUG=webrtc:5420 NOPRESUBMIT=True TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1587193006 Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
},
{
'target_name': 'rtc_media_unittests_run',
Move talk/media to webrtc/media I removed the 'libjingle' target in talk/libjingle.gyp and replaced all users of it with base/base.gyp:rtc_base. It seems the jsoncpp and expat dependencies were not used by it's previous references. The files in talk/media/testdata were uploaded to Google Storage and added .sha1 files in resources/media instead of simply moving them. The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL in order to not break Git history. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc. * Unused GYP reference to libjingle_tests_additional_deps was removed. * Removed duplicated GYP entries of webrtc/base/testutils.cc webrtc/base/testutils.h The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media, so they were moved to the media.gyp. I also checked that none of EXPAT_RELATIVE_PATH, FEATURE_ENABLE_VOICEMAIL, GTEST_RELATIVE_PATH, JSONCPP_RELATIVE_PATH, LOGGING=1, SRTP_RELATIVE_PATH, FEATURE_ENABLE_SSL, FEATURE_ENABLE_VOICEMAIL, FEATURE_ENABLE_PSTN, HAVE_SCTP, HAVE_SRTP, are used by the talk/media code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/ BUG=webrtc:5420 NOPRESUBMIT=True TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1587193006 Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
'type': 'none',
'dependencies': [
'rtc_media_unittests',
Move talk/media to webrtc/media I removed the 'libjingle' target in talk/libjingle.gyp and replaced all users of it with base/base.gyp:rtc_base. It seems the jsoncpp and expat dependencies were not used by it's previous references. The files in talk/media/testdata were uploaded to Google Storage and added .sha1 files in resources/media instead of simply moving them. The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL in order to not break Git history. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc. * Unused GYP reference to libjingle_tests_additional_deps was removed. * Removed duplicated GYP entries of webrtc/base/testutils.cc webrtc/base/testutils.h The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media, so they were moved to the media.gyp. I also checked that none of EXPAT_RELATIVE_PATH, FEATURE_ENABLE_VOICEMAIL, GTEST_RELATIVE_PATH, JSONCPP_RELATIVE_PATH, LOGGING=1, SRTP_RELATIVE_PATH, FEATURE_ENABLE_SSL, FEATURE_ENABLE_VOICEMAIL, FEATURE_ENABLE_PSTN, HAVE_SCTP, HAVE_SRTP, are used by the talk/media code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/ BUG=webrtc:5420 NOPRESUBMIT=True TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1587193006 Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'rtc_media_unittests.isolate',
Move talk/media to webrtc/media I removed the 'libjingle' target in talk/libjingle.gyp and replaced all users of it with base/base.gyp:rtc_base. It seems the jsoncpp and expat dependencies were not used by it's previous references. The files in talk/media/testdata were uploaded to Google Storage and added .sha1 files in resources/media instead of simply moving them. The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL in order to not break Git history. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc. * Unused GYP reference to libjingle_tests_additional_deps was removed. * Removed duplicated GYP entries of webrtc/base/testutils.cc webrtc/base/testutils.h The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media, so they were moved to the media.gyp. I also checked that none of EXPAT_RELATIVE_PATH, FEATURE_ENABLE_VOICEMAIL, GTEST_RELATIVE_PATH, JSONCPP_RELATIVE_PATH, LOGGING=1, SRTP_RELATIVE_PATH, FEATURE_ENABLE_SSL, FEATURE_ENABLE_VOICEMAIL, FEATURE_ENABLE_PSTN, HAVE_SCTP, HAVE_SRTP, are used by the talk/media code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/ BUG=webrtc:5420 NOPRESUBMIT=True TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1587193006 Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
],
},
{
'target_name': 'rtc_pc_unittests_run',
'type': 'none',
'dependencies': [
'rtc_pc_unittests',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'rtc_pc_unittests.isolate',
],
},
{
'target_name': 'video_engine_tests_run',
'type': 'none',
'dependencies': [
'video_engine_tests',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'video_engine_tests.isolate',
],
},
{
'target_name': 'webrtc_nonparallel_tests_run',
'type': 'none',
'dependencies': [
'webrtc_nonparallel_tests',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'webrtc_nonparallel_tests.isolate',
],
},
{
'target_name': 'webrtc_perf_tests_run',
'type': 'none',
'dependencies': [
'webrtc_perf_tests',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'webrtc_perf_tests.isolate',
],
},
],
}],
],
}