diff --git a/PRESUBMIT.py b/PRESUBMIT.py index c8386f6a4..320b07a24 100755 --- a/PRESUBMIT.py +++ b/PRESUBMIT.py @@ -127,7 +127,7 @@ def _CheckNoSourcesAboveGyp(input_api, gyp_files, output_api): # Disallow referencing source files with paths above the GYP file location. source_pattern = input_api.re.compile(r'sources.*?\[(.*?)\]', re.MULTILINE | re.DOTALL) - file_pattern = input_api.re.compile(r"'(\.\./.*?)'") + file_pattern = input_api.re.compile(r"'((\.\./.*?)|(<\(webrtc_root\).*?))'") violating_gyp_files = set() violating_source_entries = [] for gyp_file in gyp_files: diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.isolate b/webrtc/modules/audio_coding/audio_codec_speed_tests.isolate similarity index 100% rename from webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.isolate rename to webrtc/modules/audio_coding/audio_codec_speed_tests.isolate diff --git a/webrtc/modules/audio_coding/audio_coding.gypi b/webrtc/modules/audio_coding/audio_coding.gypi new file mode 100644 index 000000000..0a500d947 --- /dev/null +++ b/webrtc/modules/audio_coding/audio_coding.gypi @@ -0,0 +1,29 @@ +# Copyright (c) 2015 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. + +{ + 'includes': [ + '../../build/common.gypi', + 'codecs/interfaces.gypi', + 'codecs/cng/cng.gypi', + 'codecs/g711/g711.gypi', + 'codecs/g722/g722.gypi', + 'codecs/ilbc/ilbc.gypi', + 'codecs/isac/main/source/isac.gypi', + 'codecs/isac/fix/source/isacfix.gypi', + 'codecs/pcm16b/pcm16b.gypi', + 'codecs/red/red.gypi', + 'main/acm2/audio_coding_module.gypi', + 'neteq/neteq.gypi', + ], + 'conditions': [ + ['include_opus==1', { + 'includes': ['codecs/opus/opus.gypi',], + }], + ], +} diff --git a/webrtc/modules/audio_coding/audio_coding_tests.gypi b/webrtc/modules/audio_coding/audio_coding_tests.gypi new file mode 100644 index 000000000..86a92c595 --- /dev/null +++ b/webrtc/modules/audio_coding/audio_coding_tests.gypi @@ -0,0 +1,72 @@ +# Copyright (c) 2015 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. + +{ + 'includes': [ + '../../build/common.gypi', + 'codecs/isac/isac_test.gypi', + 'codecs/isac/isacfix_test.gypi', + ], + 'targets': [ + { + 'target_name': 'audio_codec_speed_tests', + 'type': '<(gtest_target_type)', + 'dependencies': [ + 'audio_processing', + 'iSACFix', + 'webrtc_opus', + '<(DEPTH)/testing/gtest.gyp:gtest', + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', + '<(webrtc_root)/test/test.gyp:test_support_main', + ], + 'sources': [ + 'codecs/isac/fix/test/isac_speed_test.cc', + 'codecs/opus/opus_speed_test.cc', + 'codecs/tools/audio_codec_speed_test.h', + 'codecs/tools/audio_codec_speed_test.cc', + ], + 'conditions': [ + ['OS=="android"', { + 'dependencies': [ + '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code', + ], + }], + ], + }, + ], + 'conditions': [ + ['OS=="android"', { + 'targets': [ + { + 'target_name': 'audio_codec_speed_tests_apk_target', + 'type': 'none', + 'dependencies': [ + '<(apk_tests_path):audio_codec_speed_tests_apk', + ], + }, + ], + }], + ['test_isolation_mode != "noop"', { + 'targets': [ + { + 'target_name': 'audio_codec_speed_tests_run', + 'type': 'none', + 'dependencies': [ + 'audio_codec_speed_tests', + ], + 'includes': [ + '../../build/isolate.gypi', + ], + 'sources': [ + 'audio_codec_speed_tests.isolate', + ], + }, + ], + }], + ], +} diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.gypi b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.gypi deleted file mode 100644 index e29663482..000000000 --- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.gypi +++ /dev/null @@ -1,66 +0,0 @@ -# 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. - -{ - 'targets': [ - { - 'target_name': 'audio_codec_speed_tests', - 'type': '<(gtest_target_type)', - 'dependencies': [ - 'audio_processing', - 'iSACFix', - 'webrtc_opus', - '<(DEPTH)/testing/gtest.gyp:gtest', - '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', - '<(webrtc_root)/test/test.gyp:test_support_main', - ], - 'sources': [ - 'audio_codec_speed_test.h', - 'audio_codec_speed_test.cc', - '<(webrtc_root)/modules/audio_coding/codecs/opus/opus_speed_test.cc', - '<(webrtc_root)/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc', - ], - 'conditions': [ - ['OS=="android"', { - 'dependencies': [ - '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code', - ], - }], - ], - }], - 'conditions': [ - ['OS=="android"', { - 'targets': [ - { - 'target_name': 'audio_codec_speed_tests_apk_target', - 'type': 'none', - 'dependencies': [ - '<(apk_tests_path):audio_codec_speed_tests_apk', - ], - }, - ], - }], - ['test_isolation_mode != "noop"', { - 'targets': [ - { - 'target_name': 'audio_codec_speed_tests_run', - 'type': 'none', - 'dependencies': [ - 'audio_codec_speed_tests', - ], - 'includes': [ - '../../../../build/isolate.gypi', - ], - 'sources': [ - 'audio_codec_speed_tests.isolate', - ], - }, - ], - }], - ], -} diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp index 691f308b4..23ee29f43 100644 --- a/webrtc/modules/modules.gyp +++ b/webrtc/modules/modules.gyp @@ -9,17 +9,7 @@ { 'includes': [ '../build/common.gypi', - 'audio_coding/codecs/interfaces.gypi', - 'audio_coding/codecs/cng/cng.gypi', - 'audio_coding/codecs/g711/g711.gypi', - 'audio_coding/codecs/g722/g722.gypi', - 'audio_coding/codecs/ilbc/ilbc.gypi', - 'audio_coding/codecs/isac/main/source/isac.gypi', - 'audio_coding/codecs/isac/fix/source/isacfix.gypi', - 'audio_coding/codecs/pcm16b/pcm16b.gypi', - 'audio_coding/codecs/red/red.gypi', - 'audio_coding/main/acm2/audio_coding_module.gypi', - 'audio_coding/neteq/neteq.gypi', + 'audio_coding/audio_coding.gypi', 'audio_conference_mixer/source/audio_conference_mixer.gypi', 'audio_device/audio_device.gypi', 'audio_processing/audio_processing.gypi', @@ -37,14 +27,9 @@ 'video_render/video_render.gypi', ], 'conditions': [ - ['include_opus==1', { - 'includes': ['audio_coding/codecs/opus/opus.gypi',], - }], ['include_tests==1', { 'includes': [ - 'audio_coding/codecs/isac/isac_test.gypi', - 'audio_coding/codecs/isac/isacfix_test.gypi', - 'audio_coding/codecs/tools/audio_codec_speed_tests.gypi', + 'audio_coding/audio_coding_tests.gypi', 'audio_processing/audio_processing_tests.gypi', 'rtp_rtcp/test/testFec/test_fec.gypi', 'video_coding/main/source/video_coding_test.gypi', @@ -101,6 +86,7 @@ '<(webrtc_root)/test/test.gyp:frame_generator', '<(webrtc_root)/test/test.gyp:rtp_test_utils', '<(webrtc_root)/test/test.gyp:test_support_main', + '<(webrtc_root)/tools/tools.gyp:agc_test_utils', ], 'sources': [ 'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc', @@ -175,7 +161,6 @@ 'audio_processing/agc/pitch_internal_unittest.cc', 'audio_processing/agc/pole_zero_filter_unittest.cc', 'audio_processing/agc/standalone_vad_unittest.cc', - 'audio_processing/agc/test/test_utils.cc', 'audio_processing/beamformer/complex_matrix_unittest.cc', 'audio_processing/beamformer/covariance_matrix_generator_unittest.cc', 'audio_processing/beamformer/matrix_unittest.cc', diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi index 6580281c6..c3853f635 100644 --- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi +++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi @@ -40,63 +40,67 @@ 'test/bwe_test_logging.h', ], # source }, - { - 'target_name': 'bwe_tools_util', - 'type': 'static_library', - 'dependencies': [ - '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', - 'rtp_rtcp', - ], - 'sources': [ - 'tools/bwe_rtp.cc', - 'tools/bwe_rtp.h', - ], - }, - { - 'target_name': 'bwe_rtp_to_text', - 'type': 'executable', - 'includes': [ - '../rtp_rtcp/source/rtp_rtcp.gypi', - ], - 'dependencies': [ - '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', - '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default', - 'bwe_tools_util', - 'rtp_rtcp', - ], - 'direct_dependent_settings': { - 'include_dirs': [ - 'include', - ], - }, - 'sources': [ - 'tools/rtp_to_text.cc', - '<(webrtc_root)/test/rtp_file_reader.cc', - '<(webrtc_root)/test/rtp_file_reader.h', - ], # source - }, - { - 'target_name': 'bwe_rtp_play', - 'type': 'executable', - 'includes': [ - '../rtp_rtcp/source/rtp_rtcp.gypi', - ], - 'dependencies': [ - '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', - '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default', - 'bwe_tools_util', - 'rtp_rtcp', - ], - 'direct_dependent_settings': { - 'include_dirs': [ - 'include', - ], - }, - 'sources': [ - 'tools/bwe_rtp_play.cc', - '<(webrtc_root)/test/rtp_file_reader.cc', - '<(webrtc_root)/test/rtp_file_reader.h', - ], # source - }, ], # targets + 'conditions': [ + ['include_tests==1', { + 'targets': [ + { + 'target_name': 'bwe_tools_util', + 'type': 'static_library', + 'dependencies': [ + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', + 'rtp_rtcp', + ], + 'sources': [ + 'tools/bwe_rtp.cc', + 'tools/bwe_rtp.h', + ], + }, + { + 'target_name': 'bwe_rtp_to_text', + 'type': 'executable', + 'includes': [ + '../rtp_rtcp/source/rtp_rtcp.gypi', + ], + 'dependencies': [ + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default', + '<(webrtc_root)/test/test.gyp:rtp_test_utils', + 'bwe_tools_util', + 'rtp_rtcp', + ], + 'direct_dependent_settings': { + 'include_dirs': [ + 'include', + ], + }, + 'sources': [ + 'tools/rtp_to_text.cc', + ], # source + }, + { + 'target_name': 'bwe_rtp_play', + 'type': 'executable', + 'includes': [ + '../rtp_rtcp/source/rtp_rtcp.gypi', + ], + 'dependencies': [ + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default', + '<(webrtc_root)/test/test.gyp:rtp_test_utils', + 'bwe_tools_util', + 'rtp_rtcp', + ], + 'direct_dependent_settings': { + 'include_dirs': [ + 'include', + ], + }, + 'sources': [ + 'tools/bwe_rtp_play.cc', + ], # source + }, + ], + }], # include_tests==1 + ], } diff --git a/webrtc/modules/audio_processing/agc/test/activity_metric.cc b/webrtc/tools/agc/activity_metric.cc similarity index 100% rename from webrtc/modules/audio_processing/agc/test/activity_metric.cc rename to webrtc/tools/agc/activity_metric.cc diff --git a/webrtc/modules/audio_processing/agc/test/agc_harness.cc b/webrtc/tools/agc/agc_harness.cc similarity index 99% rename from webrtc/modules/audio_processing/agc/test/agc_harness.cc rename to webrtc/tools/agc/agc_harness.cc index d7c32b08d..02d0f655d 100644 --- a/webrtc/modules/audio_processing/agc/test/agc_harness.cc +++ b/webrtc/tools/agc/agc_harness.cc @@ -12,12 +12,12 @@ #include "gflags/gflags.h" #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/audio_processing/agc/test/agc_manager.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/system_wrappers/interface/sleep.h" #include "webrtc/system_wrappers/interface/trace.h" #include "webrtc/test/channel_transport/include/channel_transport.h" #include "webrtc/test/testsupport/trace_to_stderr.h" +#include "webrtc/tools/agc/agc_manager.h" #include "webrtc/voice_engine/include/voe_audio_processing.h" #include "webrtc/voice_engine/include/voe_base.h" #include "webrtc/voice_engine/include/voe_codec.h" diff --git a/webrtc/modules/audio_processing/agc/test/agc_manager.cc b/webrtc/tools/agc/agc_manager.cc similarity index 99% rename from webrtc/modules/audio_processing/agc/test/agc_manager.cc rename to webrtc/tools/agc/agc_manager.cc index a741e6408..83c0d0075 100644 --- a/webrtc/modules/audio_processing/agc/test/agc_manager.cc +++ b/webrtc/tools/agc/agc_manager.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_processing/agc/test/agc_manager.h" +#include "webrtc/tools/agc/agc_manager.h" #include diff --git a/webrtc/modules/audio_processing/agc/test/agc_manager.h b/webrtc/tools/agc/agc_manager.h similarity index 93% rename from webrtc/modules/audio_processing/agc/test/agc_manager.h rename to webrtc/tools/agc/agc_manager.h index ec8161cdb..6b3e91d2e 100644 --- a/webrtc/modules/audio_processing/agc/test/agc_manager.h +++ b/webrtc/tools/agc/agc_manager.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_AGC_MANAGER_H_ -#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_AGC_MANAGER_H_ +#ifndef WEBRTC_TOOLS_AGC_AGC_MANAGER_H_ +#define WEBRTC_TOOLS_AGC_AGC_MANAGER_H_ #include "webrtc/modules/audio_processing/agc/agc_manager_direct.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" @@ -78,4 +78,4 @@ class AgcManager { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_AGC_MANAGER_H_ +#endif // WEBRTC_TOOLS_AGC_AGC_MANAGER_H_ diff --git a/webrtc/modules/audio_processing/agc/test/agc_manager_integrationtest.cc b/webrtc/tools/agc/agc_manager_integrationtest.cc similarity index 98% rename from webrtc/modules/audio_processing/agc/test/agc_manager_integrationtest.cc rename to webrtc/tools/agc/agc_manager_integrationtest.cc index 9dbbc2224..d4b36322b 100644 --- a/webrtc/modules/audio_processing/agc/test/agc_manager_integrationtest.cc +++ b/webrtc/tools/agc/agc_manager_integrationtest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_processing/agc/test/agc_manager.h" +#include "webrtc/tools/agc/agc_manager.h" #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" diff --git a/webrtc/modules/audio_processing/agc/test/agc_manager_unittest.cc b/webrtc/tools/agc/agc_manager_unittest.cc similarity index 99% rename from webrtc/modules/audio_processing/agc/test/agc_manager_unittest.cc rename to webrtc/tools/agc/agc_manager_unittest.cc index 92464ef84..fca8decc8 100644 --- a/webrtc/modules/audio_processing/agc/test/agc_manager_unittest.cc +++ b/webrtc/tools/agc/agc_manager_unittest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_processing/agc/test/agc_manager.h" +#include "webrtc/tools/agc/agc_manager.h" #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" diff --git a/webrtc/modules/audio_processing/agc/test/agc_test.cc b/webrtc/tools/agc/agc_test.cc similarity index 97% rename from webrtc/modules/audio_processing/agc/test/agc_test.cc rename to webrtc/tools/agc/agc_test.cc index 413b3b0c6..29769488c 100644 --- a/webrtc/modules/audio_processing/agc/test/agc_test.cc +++ b/webrtc/tools/agc/agc_test.cc @@ -16,13 +16,13 @@ #include "gflags/gflags.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/modules/audio_processing/agc/agc.h" -#include "webrtc/modules/audio_processing/agc/test/agc_manager.h" -#include "webrtc/modules/audio_processing/agc/test/test_utils.h" #include "webrtc/modules/audio_processing/agc/utility.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/modules/interface/module_common_types.h" #include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/test/testsupport/trace_to_stderr.h" +#include "webrtc/tools/agc/agc_manager.h" +#include "webrtc/tools/agc/test_utils.h" #include "webrtc/voice_engine/include/mock/fake_voe_external_media.h" #include "webrtc/voice_engine/include/mock/mock_voe_volume_control.h" diff --git a/webrtc/modules/audio_processing/agc/test/fake_agc.h b/webrtc/tools/agc/fake_agc.h similarity index 84% rename from webrtc/modules/audio_processing/agc/test/fake_agc.h rename to webrtc/tools/agc/fake_agc.h index e2aabd81f..6b39cd7c3 100644 --- a/webrtc/modules/audio_processing/agc/test/fake_agc.h +++ b/webrtc/tools/agc/fake_agc.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_FAKE_AGC_H_ -#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_FAKE_AGC_H_ +#ifndef WEBRTC_TOOLS_AGC_FAKE_AGC_H_ +#define WEBRTC_TOOLS_AGC_FAKE_AGC_H_ #include "webrtc/modules/audio_processing/agc/agc.h" @@ -43,4 +43,4 @@ class FakeAgc : public Agc { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_FAKE_AGC_H_ +#endif // WEBRTC_TOOLS_AGC_FAKE_AGC_H_ diff --git a/webrtc/modules/audio_processing/agc/test/test_utils.cc b/webrtc/tools/agc/test_utils.cc similarity index 96% rename from webrtc/modules/audio_processing/agc/test/test_utils.cc rename to webrtc/tools/agc/test_utils.cc index e7c884baf..3a26cb9ac 100644 --- a/webrtc/modules/audio_processing/agc/test/test_utils.cc +++ b/webrtc/tools/agc/test_utils.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_processing/agc/test/test_utils.h" +#include "webrtc/tools/agc/test_utils.h" #include diff --git a/webrtc/modules/audio_processing/agc/test/test_utils.h b/webrtc/tools/agc/test_utils.h similarity index 83% rename from webrtc/modules/audio_processing/agc/test/test_utils.h rename to webrtc/tools/agc/test_utils.h index 25dc496e4..2aca9996f 100644 --- a/webrtc/modules/audio_processing/agc/test/test_utils.h +++ b/webrtc/tools/agc/test_utils.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_ -#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_ +#ifndef WEBRTC_TOOLS_AGC_TEST_UTILS_H_ +#define WEBRTC_TOOLS_AGC_TEST_UTILS_H_ namespace webrtc { class AudioFrame; @@ -25,4 +25,4 @@ void SimulateMic(int gain_map[255], int mic_level, int last_mic_level, } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_ +#endif // WEBRTC_TOOLS_AGC_TEST_UTILS_H_ diff --git a/webrtc/tools/tools.gyp b/webrtc/tools/tools.gyp index 0a3d5313d..e2a54212e 100644 --- a/webrtc/tools/tools.gyp +++ b/webrtc/tools/tools.gyp @@ -110,8 +110,16 @@ '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine', ], 'sources': [ - '<(webrtc_root)/modules/audio_processing/agc/test/agc_manager.cc', - '<(webrtc_root)/modules/audio_processing/agc/test/agc_manager.h', + 'agc/agc_manager.cc', + 'agc/agc_manager.h', + ], + }, + { + 'target_name': 'agc_test_utils', + 'type': 'static_library', + 'sources': [ + 'agc/test_utils.cc', + 'agc/test_utils.h', ], }, { @@ -126,7 +134,7 @@ 'agc_manager', ], 'sources': [ - '<(webrtc_root)/modules/audio_processing/agc/test/agc_harness.cc', + 'agc/agc_harness.cc', ], }, # agc_harness { @@ -139,10 +147,10 @@ '<(webrtc_root)/test/test.gyp:test_support', '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default', 'agc_manager', + 'agc_test_utils', ], 'sources': [ - '<(webrtc_root)/modules/audio_processing/agc/test/agc_test.cc', - '<(webrtc_root)/modules/audio_processing/agc/test/test_utils.cc', + 'agc/agc_test.cc', ], }, # agc_proc { @@ -154,7 +162,7 @@ 'agc_manager', ], 'sources': [ - '<(webrtc_root)/modules/audio_processing/agc/test/activity_metric.cc', + 'agc/activity_metric.cc', ], }, # activity_metric { diff --git a/webrtc/webrtc_tests.gypi b/webrtc/webrtc_tests.gypi index 3f772eb4b..543ffc443 100644 --- a/webrtc/webrtc_tests.gypi +++ b/webrtc/webrtc_tests.gypi @@ -96,13 +96,13 @@ 'target_name': 'video_engine_tests', 'type': '<(gtest_target_type)', 'sources': [ - 'modules/audio_processing/agc/test/agc_manager_unittest.cc', + 'test/common_unittest.cc', + 'test/testsupport/metrics/video_metrics_unittest.cc', + 'tools/agc/agc_manager_unittest.cc', 'video/bitrate_estimator_tests.cc', 'video/end_to_end_tests.cc', 'video/send_statistics_proxy_unittest.cc', 'video/video_send_stream_tests.cc', - 'test/common_unittest.cc', - 'test/testsupport/metrics/video_metrics_unittest.cc', ], 'dependencies': [ '<(DEPTH)/testing/gmock.gyp:gmock', @@ -131,7 +131,7 @@ 'type': '<(gtest_target_type)', 'sources': [ 'modules/audio_coding/neteq/test/neteq_performance_unittest.cc', - 'modules/audio_processing/agc/test/agc_manager_integrationtest.cc', + 'tools/agc/agc_manager_integrationtest.cc', 'video/call_perf_tests.cc', 'video/full_stack.cc', 'video/rampup_tests.cc',