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webrtc_m130/webrtc/modules/BUILD.gn

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GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
# Copyright (c) 2016 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("audio_coding/audio_coding.gni")
declare_args() {
# Desktop capturer is supported only on Windows, OSX and Linux.
Remove ScreenCapturer and WindowCapturer This change removes ScreenCapturer and WindowCapturer from WebRTC. BUG=webrtc:6513 Review-Url: https://codereview.webrtc.org/2490063002 Cr-Commit-Position: refs/heads/master@{#15033}
2016-11-10 21:57:10 -08:00
rtc_desktop_capture_supported = is_win || is_mac || is_linux
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
group("modules") {
GN: Change group deps to public_deps. During GN vs GYP auditing it was discovered that some GN targets that had public_configs were not exposing them to dependents where the dependent depended on a group, which in turn included that target as a dependency. Instead of changing those public_configs to all_dependent_configs (which would be a change from GYP), it's better to just change those group targets to use public_deps instead. BUG=webrtc:6323 NOTRY=True TESTED=Generated GYP and GN project files on Mac and ran the tools/gyp_flag_compare.py script before and after this patch was applied. The file in question used for inspection was the webrtc/api/webrtcsessiondescriptionfactory.cc which is a part of the libjingle_peerconnection target. Review-Url: https://codereview.webrtc.org/2344623002 Cr-Commit-Position: refs/heads/master@{#14222}
2016-09-15 00:53:26 -07:00
public_deps = [
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_coding",
"audio_conference_mixer",
"audio_device",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
"audio_mixer",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_processing",
"bitrate_controller",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
"congestion_controller",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"desktop_capture",
"media_file",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
"pacing",
"remote_bitrate_estimator",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"rtp_rtcp",
"utility",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
"video_capture",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"video_coding",
"video_processing",
]
}
if (rtc_include_tests) {
GN: Declare resources for targets. Declare resources for GN targets so that they can be isolated NOTRY=True BUG=chromium:497757 Review-Url: https://codereview.webrtc.org/2340753002 Cr-Commit-Position: refs/heads/master@{#14210}
2016-09-14 05:10:01 -07:00
modules_tests_resources = [
"//resources/audio_coding/testfile32kHz.pcm",
"//resources/audio_coding/teststereo32kHz.pcm",
"//resources/foreman_cif.yuv",
"//resources/paris_qcif.yuv",
]
GN: Fix resource files for iOS test target In order to get resource files to be properly packaged into the .app for a unit test on iOS, the resource files needs to be listed as sources in a bundle_data target. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2292853002 Cr-Commit-Position: refs/heads/master@{#13968}
2016-08-30 02:53:49 -07:00
if (is_ios) {
bundle_data("modules_tests_bundle_data") {
testonly = true
sources = modules_tests_resources
outputs = [
"{{bundle_resources_dir}}/{{source_file_part}}",
]
}
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_test("modules_tests") {
GN: Add target for modules_tests. Additional changes I needed to make it work: - Modified a header in RTPFile.cc. Every other file is using "webrtc/engine_configurations.h" instead. - Disabled flag 4373 for msvs because it was disabled in build/common.gypi. BUG=webrtc:6038 TBR=kwiberg@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2187563005 Cr-Commit-Position: refs/heads/master@{#13628}
2016-08-03 10:46:47 -07:00
testonly = true
videoprocessor_defines = []
if (rtc_use_h264) {
videoprocessor_defines += [ "WEBRTC_VIDEOPROCESSOR_H264_TESTS" ]
}
defines = audio_coding_defines + videoprocessor_defines
deps = [
"..:webrtc_common",
"../common_video",
Pass selected cricket::VideoCodec down to internal H264 encoder Pass the selected cricket::VideoCodec to H264EncoderImpl::H264EncoderImpl. The cricket::VideoCodec contains relevant information for H264 about selected profile and packetization mode. BUG=chromium:600254,webrtc:6402, webrtc:6337 Review-Url: https://codereview.webrtc.org/2474993002 Cr-Commit-Position: refs/heads/master@{#15270}
2016-11-28 07:20:21 -08:00
"../media:rtc_media_base",
GN: Add target for modules_tests. Additional changes I needed to make it work: - Modified a header in RTPFile.cc. Every other file is using "webrtc/engine_configurations.h" instead. - Disabled flag 4373 for msvs because it was disabled in build/common.gypi. BUG=webrtc:6038 TBR=kwiberg@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2187563005 Cr-Commit-Position: refs/heads/master@{#13628}
2016-08-03 10:46:47 -07:00
"../modules/audio_coding",
Stop using old AudioCodingModule::RegisterReceiveCodec overloads BUG=webrtc:5801 Review-Url: https://codereview.webrtc.org/2388153004 Cr-Commit-Position: refs/heads/master@{#14753}
2016-10-24 13:47:09 -07:00
"../modules/audio_coding:audio_format_conversion",
GN: Add target for modules_tests. Additional changes I needed to make it work: - Modified a header in RTPFile.cc. Every other file is using "webrtc/engine_configurations.h" instead. - Disabled flag 4373 for msvs because it was disabled in build/common.gypi. BUG=webrtc:6038 TBR=kwiberg@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2187563005 Cr-Commit-Position: refs/heads/master@{#13628}
2016-08-03 10:46:47 -07:00
"../modules/rtp_rtcp",
"../modules/utility",
"../modules/video_coding",
"../modules/video_coding:video_codecs_test_framework",
"../system_wrappers",
"../test:test_support",
"../test:test_support_main",
Move ScreenCapturer 'real' tests out of screen_capturer_unittest.cc. This is a trivial change, I just cutted and pasted part of the code in screen_capturer_unittest.cc to screen_capturer_integration_test.cc, removed DISABLED_ prefixes, and updated build file. BUG=webrtc:6366 Review-Url: https://codereview.webrtc.org/2444583002 Cr-Commit-Position: refs/heads/master@{#14806}
2016-10-27 16:50:35 -07:00
"//testing/gmock",
GN: Add target for modules_tests. Additional changes I needed to make it work: - Modified a header in RTPFile.cc. Every other file is using "webrtc/engine_configurations.h" instead. - Disabled flag 4373 for msvs because it was disabled in build/common.gypi. BUG=webrtc:6038 TBR=kwiberg@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2187563005 Cr-Commit-Position: refs/heads/master@{#13628}
2016-08-03 10:46:47 -07:00
"//testing/gtest",
]
sources = [
"audio_coding/test/APITest.cc",
"audio_coding/test/Channel.cc",
"audio_coding/test/EncodeDecodeTest.cc",
"audio_coding/test/PCMFile.cc",
"audio_coding/test/PacketLossTest.cc",
"audio_coding/test/RTPFile.cc",
"audio_coding/test/TestAllCodecs.cc",
"audio_coding/test/TestRedFec.cc",
"audio_coding/test/TestStereo.cc",
"audio_coding/test/TestVADDTX.cc",
"audio_coding/test/Tester.cc",
"audio_coding/test/TwoWayCommunication.cc",
"audio_coding/test/iSACTest.cc",
"audio_coding/test/opus_test.cc",
"audio_coding/test/target_delay_unittest.cc",
"audio_coding/test/utility.cc",
"rtp_rtcp/test/testFec/test_fec.cc",
"video_coding/codecs/test/videoprocessor_integrationtest.cc",
"video_coding/codecs/vp8/test/vp8_impl_unittest.cc",
]
Move ScreenCapturer 'real' tests out of screen_capturer_unittest.cc. This is a trivial change, I just cutted and pasted part of the code in screen_capturer_unittest.cc to screen_capturer_integration_test.cc, removed DISABLED_ prefixes, and updated build file. BUG=webrtc:6366 Review-Url: https://codereview.webrtc.org/2444583002 Cr-Commit-Position: refs/heads/master@{#14806}
2016-10-27 16:50:35 -07:00
if (rtc_desktop_capture_supported) {
Remove screen_capturer_mock_objects.h This is a trivial change to remove MockScreenCapturerCallback, and use MockDesktopCapturerCallback to replace it. BUG=webrtc:6513 Review-Url: https://codereview.webrtc.org/2494013003 Cr-Commit-Position: refs/heads/master@{#15047}
2016-11-11 17:18:34 -08:00
deps += [
"desktop_capture:desktop_capture_mock",
"desktop_capture:screen_drawer",
]
Add CreateWindowCapturer() and CreateScreenCapturer() in DesktopCapturer This change copies ScreenCapturerDifferWrapper to a new DesktopCapturerDifferWrapper, and adds DesktopCapturer::CreateWindowCapturer and DesktopCapturer::CreateScreenCapturer functions to replace WindowCapturer::Create and ScreenCapturer::Create. BUG=webrtc:6513 Committed: https://crrev.com/b763e39beba92b45baa09542f949daabbe6258a3 Review-Url: https://codereview.webrtc.org/2468753002 Cr-Original-Commit-Position: refs/heads/master@{#14880} Cr-Commit-Position: refs/heads/master@{#14903}
2016-11-02 14:49:35 -07:00
sources += [
"desktop_capture/screen_capturer_integration_test.cc",
"desktop_capture/screen_drawer_unittest.cc",
]
Move ScreenCapturer 'real' tests out of screen_capturer_unittest.cc. This is a trivial change, I just cutted and pasted part of the code in screen_capturer_unittest.cc to screen_capturer_integration_test.cc, removed DISABLED_ prefixes, and updated build file. BUG=webrtc:6366 Review-Url: https://codereview.webrtc.org/2444583002 Cr-Commit-Position: refs/heads/master@{#14806}
2016-10-27 16:50:35 -07:00
}
GN: Declare resources for targets. Declare resources for GN targets so that they can be isolated NOTRY=True BUG=chromium:497757 Review-Url: https://codereview.webrtc.org/2340753002 Cr-Commit-Position: refs/heads/master@{#14210}
2016-09-14 05:10:01 -07:00
data = modules_tests_resources
GN: Add target for modules_tests. Additional changes I needed to make it work: - Modified a header in RTPFile.cc. Every other file is using "webrtc/engine_configurations.h" instead. - Disabled flag 4373 for msvs because it was disabled in build/common.gypi. BUG=webrtc:6038 TBR=kwiberg@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2187563005 Cr-Commit-Position: refs/heads/master@{#13628}
2016-08-03 10:46:47 -07:00
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
GN: Update tests to have the correct shard timeout value on Android. TBR=mflodman@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2231413002 Cr-Commit-Position: refs/heads/master@{#13741}
2016-08-15 02:29:11 -07:00
shard_timeout = 900
GN: Add target for modules_tests. Additional changes I needed to make it work: - Modified a header in RTPFile.cc. Every other file is using "webrtc/engine_configurations.h" instead. - Disabled flag 4373 for msvs because it was disabled in build/common.gypi. BUG=webrtc:6038 TBR=kwiberg@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2187563005 Cr-Commit-Position: refs/heads/master@{#13628}
2016-08-03 10:46:47 -07:00
}
GN: Fix resource files for iOS test target In order to get resource files to be properly packaged into the .app for a unit test on iOS, the resource files needs to be listed as sources in a bundle_data target. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2292853002 Cr-Commit-Position: refs/heads/master@{#13968}
2016-08-30 02:53:49 -07:00
if (is_ios) {
deps += [ ":modules_tests_bundle_data" ]
GN: Add target for modules_tests. Additional changes I needed to make it work: - Modified a header in RTPFile.cc. Every other file is using "webrtc/engine_configurations.h" instead. - Disabled flag 4373 for msvs because it was disabled in build/common.gypi. BUG=webrtc:6038 TBR=kwiberg@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2187563005 Cr-Commit-Position: refs/heads/master@{#13628}
2016-08-03 10:46:47 -07:00
}
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
GN: Add target for modules_tests. Additional changes I needed to make it work: - Modified a header in RTPFile.cc. Every other file is using "webrtc/engine_configurations.h" instead. - Disabled flag 4373 for msvs because it was disabled in build/common.gypi. BUG=webrtc:6038 TBR=kwiberg@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2187563005 Cr-Commit-Position: refs/heads/master@{#13628}
2016-08-03 10:46:47 -07:00
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Add target for modules_tests. Additional changes I needed to make it work: - Modified a header in RTPFile.cc. Every other file is using "webrtc/engine_configurations.h" instead. - Disabled flag 4373 for msvs because it was disabled in build/common.gypi. BUG=webrtc:6038 TBR=kwiberg@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2187563005 Cr-Commit-Position: refs/heads/master@{#13628}
2016-08-03 10:46:47 -07:00
}
if (is_win) {
cflags = [
# TODO(phoglund): get rid of 4373 supression when
# http://code.google.com/p/webrtc/issues/detail?id=261 is solved.
# legacy warning for ignoring const / volatile in signatures.
"/wd4373",
]
}
}
GN: Declare resources for targets. Declare resources for GN targets so that they can be isolated NOTRY=True BUG=chromium:497757 Review-Url: https://codereview.webrtc.org/2340753002 Cr-Commit-Position: refs/heads/master@{#14210}
2016-09-14 05:10:01 -07:00
modules_unittests_resources = [
"//resources/att-downlink.rx",
"//resources/att-uplink.rx",
"//resources/audio_coding/neteq_opus.rtp",
"//resources/audio_coding/neteq_universal_new.rtp",
"//resources/audio_coding/speech_mono_16kHz.pcm",
"//resources/audio_coding/speech_mono_32_48kHz.pcm",
"//resources/audio_coding/testfile32kHz.pcm",
"//resources/audio_coding/teststereo32kHz.pcm",
"//resources/audio_device/audio_short16.pcm",
"//resources/audio_device/audio_short44.pcm",
"//resources/audio_device/audio_short48.pcm",
"//resources/audio_processing/agc/agc_audio.pcm",
"//resources/audio_processing/agc/agc_no_circular_buffer.dat",
"//resources/audio_processing/agc/agc_pitch_gain.dat",
"//resources/audio_processing/agc/agc_pitch_lag.dat",
"//resources/audio_processing/agc/agc_spectral_peak.dat",
"//resources/audio_processing/agc/agc_vad.dat",
"//resources/audio_processing/agc/agc_voicing_prob.dat",
"//resources/audio_processing/agc/agc_with_circular_buffer.dat",
Modify audio_processing_unittest to use ResourcePath instead of ProjectRootPath. Move the resources to //resources and upload them to Google Storage. BUG=webrtc:6727 Review-Url: https://codereview.webrtc.org/2508943004 Cr-Commit-Position: refs/heads/master@{#15152}
2016-11-18 04:52:22 -08:00
"//resources/audio_processing/output_data_fixed.pb",
"//resources/audio_processing/output_data_float.pb",
"//resources/audio_processing/output_data_mac.pb",
GN: Declare resources for targets. Declare resources for GN targets so that they can be isolated NOTRY=True BUG=chromium:497757 Review-Url: https://codereview.webrtc.org/2340753002 Cr-Commit-Position: refs/heads/master@{#14210}
2016-09-14 05:10:01 -07:00
"//resources/audio_processing/transient/ajm-macbook-1-spke16m.pcm",
"//resources/audio_processing/transient/audio16kHz.pcm",
"//resources/audio_processing/transient/audio32kHz.pcm",
"//resources/audio_processing/transient/audio48kHz.pcm",
"//resources/audio_processing/transient/audio8kHz.pcm",
"//resources/audio_processing/transient/detect16kHz.dat",
"//resources/audio_processing/transient/detect32kHz.dat",
"//resources/audio_processing/transient/detect48kHz.dat",
"//resources/audio_processing/transient/detect8kHz.dat",
"//resources/audio_processing/transient/double-utils.dat",
"//resources/audio_processing/transient/float-utils.dat",
"//resources/audio_processing/transient/suppressed16kHz.pcm",
"//resources/audio_processing/transient/suppressed32kHz.pcm",
"//resources/audio_processing/transient/suppressed8kHz.pcm",
"//resources/audio_processing/transient/wpd0.dat",
"//resources/audio_processing/transient/wpd1.dat",
"//resources/audio_processing/transient/wpd2.dat",
"//resources/audio_processing/transient/wpd3.dat",
"//resources/audio_processing/transient/wpd4.dat",
"//resources/audio_processing/transient/wpd5.dat",
"//resources/audio_processing/transient/wpd6.dat",
"//resources/audio_processing/transient/wpd7.dat",
"//resources/deflicker_before_cif_short.yuv",
"//resources/far16_stereo.pcm",
"//resources/far32_stereo.pcm",
"//resources/far44_stereo.pcm",
"//resources/far48_stereo.pcm",
"//resources/far8_stereo.pcm",
"//resources/foremanColorEnhanced_cif_short.yuv",
"//resources/foreman_cif.yuv",
"//resources/foreman_cif_short.yuv",
"//resources/near16_stereo.pcm",
"//resources/near32_stereo.pcm",
"//resources/near44_stereo.pcm",
"//resources/near48_stereo.pcm",
"//resources/near8_stereo.pcm",
"//resources/ref03.aecdump",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_0_AST.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_0_TOF.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_1_AST.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_1_TOF.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_0_AST.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_0_TOF.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_1_AST.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_1_TOF.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingDelay1_0_AST.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingDelay1_0_TOF.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingLoss1_0_AST.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingLoss1_0_TOF.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_Multi1_1_AST.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_Multi1_1_TOF.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_0_AST.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_0_TOF.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_1_AST.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_1_TOF.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyDelay_0_AST.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyDelay_0_TOF.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyLoss_0_AST.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyLoss_0_TOF.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_UnlimitedSpeed_0_AST.bin",
"//resources/remote_bitrate_estimator/VideoSendersTest_BweTest_UnlimitedSpeed_0_TOF.bin",
"//resources/short_mixed_mono_48.dat",
"//resources/short_mixed_mono_48_arm.dat",
"//resources/short_mixed_mono_48.pcm",
"//resources/short_mixed_stereo_48.dat",
"//resources/short_mixed_stereo_48.pcm",
"//resources/sprint-downlink.rx",
"//resources/sprint-uplink.rx",
"//resources/synthetic-trace.rx",
"//resources/tmobile-downlink.rx",
"//resources/tmobile-uplink.rx",
"//resources/utility/encapsulated_pcm16b_8khz.wav",
"//resources/utility/encapsulated_pcmu_8khz.wav",
"//resources/verizon3g-downlink.rx",
"//resources/verizon3g-uplink.rx",
"//resources/verizon4g-downlink.rx",
"//resources/verizon4g-uplink.rx",
WebRTC: Replace ProjectRootPath by ResourcePath BUG=webrtc:6727 NOTRY=True Review-Url: https://codereview.webrtc.org/2513363004 Cr-Commit-Position: refs/heads/master@{#15201}
2016-11-22 10:42:59 -08:00
"//resources/voice_engine/audio_tiny48.wav",
GN: Declare resources for targets. Declare resources for GN targets so that they can be isolated NOTRY=True BUG=chromium:497757 Review-Url: https://codereview.webrtc.org/2340753002 Cr-Commit-Position: refs/heads/master@{#14210}
2016-09-14 05:10:01 -07:00
]
GN: Fix resource files for iOS test target In order to get resource files to be properly packaged into the .app for a unit test on iOS, the resource files needs to be listed as sources in a bundle_data target. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2292853002 Cr-Commit-Position: refs/heads/master@{#13968}
2016-08-30 02:53:49 -07:00
if (is_ios) {
bundle_data("modules_unittests_bundle_data") {
testonly = true
sources = modules_unittests_resources
outputs = [
"{{bundle_resources_dir}}/{{source_file_part}}",
]
}
}
Relanding of "Adding BitrateController to audio network adaptor." Adding BitrateController to audio network adaptor was first landed in https://codereview.webrtc.org/2334613002/ but reverted in https://codereview.webrtc.org/2352223002/ due to ODR violation. This CL tries to use namespace trick to solve the ODR problem. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2353293002 Cr-Commit-Position: refs/heads/master@{#14343}
2016-09-21 23:01:26 -07:00
rtc_source_set("audio_network_adaptor_unittests") {
# Put sources for unittests of audio network adaptor in a separate
# rtc_source_set to solve name collision on bitrate_controller_unittest.cc.
testonly = true
sources = [
"audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc",
"audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc",
"audio_coding/audio_network_adaptor/channel_controller_unittest.cc",
"audio_coding/audio_network_adaptor/controller_manager_unittest.cc",
"audio_coding/audio_network_adaptor/dtx_controller_unittest.cc",
Adding FecController to audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2337103006 Cr-Commit-Position: refs/heads/master@{#14351}
2016-09-22 06:20:50 -07:00
"audio_coding/audio_network_adaptor/fec_controller_unittest.cc",
Relanding of "Adding BitrateController to audio network adaptor." Adding BitrateController to audio network adaptor was first landed in https://codereview.webrtc.org/2334613002/ but reverted in https://codereview.webrtc.org/2352223002/ due to ODR violation. This CL tries to use namespace trick to solve the ODR problem. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2353293002 Cr-Commit-Position: refs/heads/master@{#14343}
2016-09-21 23:01:26 -07:00
"audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc",
"audio_coding/audio_network_adaptor/mock/mock_controller.h",
"audio_coding/audio_network_adaptor/mock/mock_controller_manager.h",
]
deps = [
"audio_coding:audio_network_adaptor",
"//testing/gmock",
"//testing/gtest",
]
Creating controller manager from config string in audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2364403004 Cr-Commit-Position: refs/heads/master@{#14466}
2016-10-02 14:53:37 -07:00
if (rtc_enable_protobuf) {
deps += [ "audio_coding:ana_config_proto" ]
defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ]
}
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
Relanding of "Adding BitrateController to audio network adaptor." Adding BitrateController to audio network adaptor was first landed in https://codereview.webrtc.org/2334613002/ but reverted in https://codereview.webrtc.org/2352223002/ due to ODR violation. This CL tries to use namespace trick to solve the ODR problem. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2353293002 Cr-Commit-Position: refs/heads/master@{#14343}
2016-09-21 23:01:26 -07:00
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_test("modules_unittests") {
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
testonly = true
defines = audio_coding_defines
Moved legacy_encoded_audio_frame into audio_decoder_interface. audio_decoder.cc depends on LegacyEncodedAudioFrame and LegacyEncodedAudioFrame depends on AudioDecoder::EncodedAudioFrame, so there's no clear way to separate them as of now. This error is also hodling up builds downstream. I expect we'll revisit these dependencies as part of the upcoming larger restructuring effort. NOTRY=true BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2359763002 Cr-Commit-Position: refs/heads/master@{#14329}
2016-09-21 05:50:37 -07:00
deps = []
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
sources = [
Drop _oldapi from ACM test file names Updating GN files, include paths, and include guards BUG=None NOTRY=True NOPRESUBMIT=true Review-Url: https://codereview.webrtc.org/2387113005 Cr-Commit-Position: refs/heads/master@{#14542}
2016-10-06 01:31:32 -07:00
"audio_coding/acm2/acm_receiver_unittest.cc",
"audio_coding/acm2/audio_coding_module_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_coding/acm2/call_statistics_unittest.cc",
"audio_coding/acm2/codec_manager_unittest.cc",
"audio_coding/acm2/rent_a_codec_unittest.cc",
"audio_coding/codecs/audio_decoder_factory_unittest.cc",
"audio_coding/codecs/cng/audio_encoder_cng_unittest.cc",
"audio_coding/codecs/cng/cng_unittest.cc",
iLBC: Handle a case of bad input data We detect an unreasonable state (caused by a bad encoded stream) before it can lead to problems, and handle it by resetting the decoder. NOPRESUBMIT=true BUG=chromium:617124 Review-Url: https://codereview.webrtc.org/2255203002 Cr-Commit-Position: refs/heads/master@{#13888}
2016-08-24 02:46:44 -07:00
"audio_coding/codecs/ilbc/ilbc_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc",
"audio_coding/codecs/isac/fix/source/filters_unittest.cc",
"audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc",
"audio_coding/codecs/isac/fix/source/transform_unittest.cc",
"audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc",
"audio_coding/codecs/isac/main/source/isac_unittest.cc",
"audio_coding/codecs/isac/unittest.cc",
Moved codec-specific audio packet splitting into decoders. There's still some code run specifically for Opus w/ FEC. It will be addressed in a separate CL. BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2326003002 Cr-Commit-Position: refs/heads/master@{#14319}
2016-09-21 01:57:31 -07:00
"audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_coding/codecs/mock/mock_audio_encoder.cc",
"audio_coding/codecs/opus/audio_encoder_opus_unittest.cc",
"audio_coding/codecs/opus/opus_unittest.cc",
"audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc",
"audio_coding/neteq/audio_classifier_unittest.cc",
"audio_coding/neteq/audio_multi_vector_unittest.cc",
"audio_coding/neteq/audio_vector_unittest.cc",
"audio_coding/neteq/background_noise_unittest.cc",
"audio_coding/neteq/buffer_level_filter_unittest.cc",
"audio_coding/neteq/comfort_noise_unittest.cc",
"audio_coding/neteq/decision_logic_unittest.cc",
"audio_coding/neteq/decoder_database_unittest.cc",
"audio_coding/neteq/delay_manager_unittest.cc",
"audio_coding/neteq/delay_peak_detector_unittest.cc",
"audio_coding/neteq/dsp_helper_unittest.cc",
"audio_coding/neteq/dtmf_buffer_unittest.cc",
"audio_coding/neteq/dtmf_tone_generator_unittest.cc",
"audio_coding/neteq/expand_unittest.cc",
"audio_coding/neteq/merge_unittest.cc",
"audio_coding/neteq/mock/mock_audio_decoder.h",
"audio_coding/neteq/mock/mock_buffer_level_filter.h",
"audio_coding/neteq/mock/mock_decoder_database.h",
"audio_coding/neteq/mock/mock_delay_manager.h",
"audio_coding/neteq/mock/mock_delay_peak_detector.h",
"audio_coding/neteq/mock/mock_dtmf_buffer.h",
"audio_coding/neteq/mock/mock_dtmf_tone_generator.h",
"audio_coding/neteq/mock/mock_expand.h",
"audio_coding/neteq/mock/mock_external_decoder_pcm16b.h",
"audio_coding/neteq/mock/mock_packet_buffer.h",
Moved Opus-specific payload splitting into AudioDecoderOpus. The biggest change to NetEq is the move from a primary flag, to a Priority with two separate levels: one set by RED splitting and one set by the codec itself. This allows us to unambigously prioritize "fallback" packets from these two sources. I've chosen what I believe is the sensible ordering: packets that the codec prioritizes are chosen first, regardless of if they are secondary RED packets or not. So if we were to use Opus w/ FEC in RED, we'd only do Opus FEC decoding if there was no RED packet that could cover the time slot. With this change, PayloadSplitter now only deals with RED packets. Maybe it should be renamed RedPayloadSplitter? BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2342443005 Cr-Commit-Position: refs/heads/master@{#14347}
2016-09-22 02:06:28 -07:00
"audio_coding/neteq/mock/mock_red_payload_splitter.h",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_coding/neteq/nack_tracker_unittest.cc",
"audio_coding/neteq/neteq_external_decoder_unittest.cc",
"audio_coding/neteq/neteq_impl_unittest.cc",
"audio_coding/neteq/neteq_network_stats_unittest.cc",
"audio_coding/neteq/neteq_stereo_unittest.cc",
"audio_coding/neteq/neteq_unittest.cc",
"audio_coding/neteq/normal_unittest.cc",
"audio_coding/neteq/packet_buffer_unittest.cc",
"audio_coding/neteq/post_decode_vad_unittest.cc",
"audio_coding/neteq/random_vector_unittest.cc",
Moved Opus-specific payload splitting into AudioDecoderOpus. The biggest change to NetEq is the move from a primary flag, to a Priority with two separate levels: one set by RED splitting and one set by the codec itself. This allows us to unambigously prioritize "fallback" packets from these two sources. I've chosen what I believe is the sensible ordering: packets that the codec prioritizes are chosen first, regardless of if they are secondary RED packets or not. So if we were to use Opus w/ FEC in RED, we'd only do Opus FEC decoding if there was no RED packet that could cover the time slot. With this change, PayloadSplitter now only deals with RED packets. Maybe it should be renamed RedPayloadSplitter? BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2342443005 Cr-Commit-Position: refs/heads/master@{#14347}
2016-09-22 02:06:28 -07:00
"audio_coding/neteq/red_payload_splitter_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_coding/neteq/sync_buffer_unittest.cc",
"audio_coding/neteq/tick_timer_unittest.cc",
"audio_coding/neteq/time_stretch_unittest.cc",
"audio_coding/neteq/timestamp_scaler_unittest.cc",
"audio_coding/neteq/tools/input_audio_file_unittest.cc",
"audio_coding/neteq/tools/packet_unittest.cc",
"audio_conference_mixer/test/audio_conference_mixer_unittest.cc",
"audio_device/fine_audio_buffer_unittest.cc",
Changed ramping functionality of the AudioMixer. BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2398083005 Cr-Commit-Position: refs/heads/master@{#14607}
2016-10-12 02:14:59 -07:00
"audio_mixer/audio_frame_manipulator_unittest.cc",
Implemented further mixer interface change suggestions from https://codereview.webrtc.org/2386383003/ Changed mixability status into AddSource/RemoveSource. Added 'ssrc()' method to the MixerSource interface. Removed unnecessary member 'num_audio_sources_' and made the mixer be refcounted. BUG=webrtc:6346 NOTRY=True Review-Url: https://codereview.webrtc.org/2408683002 Cr-Commit-Position: refs/heads/master@{#14612}
2016-10-12 06:07:07 -07:00
"audio_mixer/audio_mixer_impl_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_processing/aec/echo_cancellation_unittest.cc",
"audio_processing/aec/system_delay_unittest.cc",
"audio_processing/agc/agc_manager_direct_unittest.cc",
"audio_processing/agc/loudness_histogram_unittest.cc",
"audio_processing/agc/mock_agc.h",
Keep track of the user-facing number of channels in a ChannelBuffer Before this change the ChannelBuffer had a fixed number of channels. This meant for example that when the Beamformer would reduce the number of channels to one, the merging filter bank was still merging all the channels, which was unnecessary since they were not processed and just discarded later. This change doesn't change the signal at all. It just reflects the number of channels in the ChannelBuffer, reducing the complexity. R=henrik.lundin@webrtc.org, peah@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/2053773002 . Cr-Commit-Position: refs/heads/master@{#13352}
2016-06-30 15:33:37 -07:00
"audio_processing/audio_buffer_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_processing/beamformer/array_util_unittest.cc",
"audio_processing/beamformer/complex_matrix_unittest.cc",
"audio_processing/beamformer/covariance_matrix_generator_unittest.cc",
"audio_processing/beamformer/matrix_unittest.cc",
"audio_processing/beamformer/mock_nonlinear_beamformer.h",
"audio_processing/beamformer/nonlinear_beamformer_unittest.cc",
Moving/renaming webrtc/common.h. This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code. - For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel(). - Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state. BUG=webrtc:5879 Review-Url: https://codereview.webrtc.org/2307533004 Cr-Commit-Position: refs/heads/master@{#14109}
2016-09-07 07:34:41 -07:00
"audio_processing/config_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_processing/echo_cancellation_impl_unittest.cc",
"audio_processing/splitting_filter_unittest.cc",
"audio_processing/transient/dyadic_decimator_unittest.cc",
"audio_processing/transient/file_utils.cc",
"audio_processing/transient/file_utils.h",
"audio_processing/transient/file_utils_unittest.cc",
"audio_processing/transient/moving_moments_unittest.cc",
"audio_processing/transient/transient_detector_unittest.cc",
"audio_processing/transient/transient_suppressor_unittest.cc",
"audio_processing/transient/wpd_node_unittest.cc",
"audio_processing/transient/wpd_tree_unittest.cc",
"audio_processing/utility/block_mean_calculator_unittest.cc",
"audio_processing/utility/delay_estimator_unittest.cc",
"audio_processing/vad/gmm_unittest.cc",
"audio_processing/vad/pitch_based_vad_unittest.cc",
"audio_processing/vad/pitch_internal_unittest.cc",
"audio_processing/vad/pole_zero_filter_unittest.cc",
"audio_processing/vad/standalone_vad_unittest.cc",
"audio_processing/vad/vad_audio_proc_unittest.cc",
"audio_processing/vad/vad_circular_buffer_unittest.cc",
"audio_processing/vad/voice_activity_detector_unittest.cc",
"bitrate_controller/bitrate_controller_unittest.cc",
"bitrate_controller/send_side_bandwidth_estimation_unittest.cc",
"congestion_controller/congestion_controller_unittest.cc",
Reset InterArrival if arrival time clock makes a jump. Also adds a copy of the BWE test suite to the new DelayBasedBwe class. BUG=webrtc:6079 Review-Url: https://codereview.webrtc.org/2126793002 Cr-Commit-Position: refs/heads/master@{#13428}
2016-07-11 01:44:02 -07:00
"congestion_controller/delay_based_bwe_unittest.cc",
"congestion_controller/delay_based_bwe_unittest_helper.cc",
"congestion_controller/delay_based_bwe_unittest_helper.h",
Probe bitrate estimator correction. Since the interval between the timestamps does not include the send/receive time of the last/first packet we correct the interval by adding the average of the interval between probing packets. BUG=webrtc:5859 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2224173003 . Cr-Commit-Position: refs/heads/master@{#13719}
2016-08-10 19:00:39 +02:00
"congestion_controller/probe_bitrate_estimator_unittest.cc",
Probing: Add support for exponential startup probing Adds support for exponentially probing the bandwidth at start-up to allow ramp-up to real capacity of the network. BUG=webrtc:6332 R=philipel@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2235373004 . Cr-Commit-Position: refs/heads/master@{#14189}
2016-09-12 12:28:54 -07:00
"congestion_controller/probe_controller_unittest.cc",
Add interval estimator to remote bitrate estimator. To be able to smooth the bandwidth estimation according to the probing interval. BUG=webrtc:6443 Review-Url: https://codereview.webrtc.org/2380883003 Cr-Commit-Position: refs/heads/master@{#15123}
2016-11-17 01:18:43 -08:00
"congestion_controller/probing_interval_estimator_unittest.cc",
Reland: Fix race / crash in OnNetworkRouteChanged(). To achieve this some refactoring was done to make it possible to synchronize access to the DelayBasedBwe in TransportFeedbackAdapter: - The callback was removed from DelayBasedBwe, it now instead returns its result. - TransportFeedbackAdapter was moved to modules/congestion_controller to avoid unnecessary dependencies. Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds. BUG=webrtc:6427, webrtc:6422 R=terelius@webrtc.org Review URL: https://codereview.webrtc.org/2378103005 . Cr-Commit-Position: refs/heads/master@{#14452}
2016-09-30 10:06:51 +02:00
"congestion_controller/transport_feedback_adapter_unittest.cc",
Add a new overuse estimator for the delay based BWE behind experiment. Parse the estimation parameters from the field trial string. BUG=webrtc:6690 Review-Url: https://codereview.webrtc.org/2489323002 Cr-Commit-Position: refs/heads/master@{#15126}
2016-11-17 03:48:18 -08:00
"congestion_controller/trendline_estimator_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"media_file/media_file_unittest.cc",
"module_common_types_unittest.cc",
Add AlrDetector This is a simple application limited region detector that is quite conservative at the moment. We detect as being application-limited if we see sending rate as less than 30% of the estimated bandwidth over 500 ms. The moment we detect a single burst above 30% over a 100 ms period, we consider ourselves network limited. This class is currently not used. A follow up CL will leverage this to enable probing. BUG=webrtc:6332 Review-Url: https://codereview.webrtc.org/2340763004 Cr-Commit-Position: refs/heads/master@{#14505}
2016-10-04 08:43:09 -07:00
"pacing/alr_detector_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"pacing/bitrate_prober_unittest.cc",
"pacing/paced_sender_unittest.cc",
"pacing/packet_router_unittest.cc",
Add interval estimator to remote bitrate estimator. To be able to smooth the bandwidth estimation according to the probing interval. BUG=webrtc:6443 Review-Url: https://codereview.webrtc.org/2380883003 Cr-Commit-Position: refs/heads/master@{#15123}
2016-11-17 01:18:43 -08:00
"remote_bitrate_estimator/aimd_rate_control_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h",
"remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h",
"remote_bitrate_estimator/inter_arrival_unittest.cc",
"remote_bitrate_estimator/overuse_detector_unittest.cc",
"remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc",
"remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc",
"remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc",
"remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h",
"remote_bitrate_estimator/remote_estimator_proxy_unittest.cc",
"remote_bitrate_estimator/send_time_history_unittest.cc",
"remote_bitrate_estimator/test/bwe_test_framework_unittest.cc",
"remote_bitrate_estimator/test/bwe_unittest.cc",
"remote_bitrate_estimator/test/estimators/nada_unittest.cc",
"remote_bitrate_estimator/test/metric_recorder_unittest.cc",
"rtp_rtcp/source/byte_io_unittest.cc",
"rtp_rtcp/source/fec_test_helper.cc",
"rtp_rtcp/source/fec_test_helper.h",
Add FlexFEC header formatters. Add classes that can read and finalize FlexFEC headers. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2269903002 Cr-Commit-Position: refs/heads/master@{#14469}
2016-10-03 00:43:25 -07:00
"rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc",
Add a FlexfecReceiver class. This class is split in interface/implementation classes, since it will be referenced from the Call level. Its purpose is to interface the erasure code decoder with a new class FlexfecReceiveStream (for received packets), as well as with the main RTP pipeline (for recovered packets). BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2392663006 Cr-Commit-Position: refs/heads/master@{#14594}
2016-10-10 16:44:57 -07:00
"rtp_rtcp/source/flexfec_receiver_unittest.cc",
Add FlexfecSender. This class will interface RTPSenderVideo with the underlying erasure code. It is functionally similar to ProducerFec (to be renamed UlpfecGenerator). In fact, the FlexfecSender is a friend of ProducerFec, and reuses most of its implementation. Besides the fact that FlexfecSender outputs FlexFEC packets, the main difference with ProducerFec is that FlexfecSender allocates RTP sequence numbers, whereas ProducerFec does not do this for the RED-encapsulated ULPFEC packets. This class is split as interface/implementation, since it will be owned by VideoSendStream initially. Further along, it may be owned by PacedSender. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2441613002 Cr-Commit-Position: refs/heads/master@{#14922}
2016-11-03 09:22:33 -07:00
"rtp_rtcp/source/flexfec_sender_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"rtp_rtcp/source/nack_rtx_unittest.cc",
"rtp_rtcp/source/packet_loss_stats_unittest.cc",
Add playout_delay_oracle_unittest as gn target BUG= Review-Url: https://codereview.webrtc.org/2256743002 Cr-Commit-Position: refs/heads/master@{#13821}
2016-08-18 07:53:36 -07:00
"rtp_rtcp/source/playout_delay_oracle_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"rtp_rtcp/source/receive_statistics_unittest.cc",
"rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc",
"rtp_rtcp/source/rtcp_packet/app_unittest.cc",
"rtp_rtcp/source/rtcp_packet/bye_unittest.cc",
"rtp_rtcp/source/rtcp_packet/common_header_unittest.cc",
"rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc",
"rtp_rtcp/source/rtcp_packet/dlrr_unittest.cc",
"rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc",
"rtp_rtcp/source/rtcp_packet/extended_reports_unittest.cc",
"rtp_rtcp/source/rtcp_packet/fir_unittest.cc",
"rtp_rtcp/source/rtcp_packet/nack_unittest.cc",
"rtp_rtcp/source/rtcp_packet/pli_unittest.cc",
"rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc",
"rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc",
"rtp_rtcp/source/rtcp_packet/remb_unittest.cc",
"rtp_rtcp/source/rtcp_packet/report_block_unittest.cc",
"rtp_rtcp/source/rtcp_packet/rpsi_unittest.cc",
"rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc",
"rtp_rtcp/source/rtcp_packet/sdes_unittest.cc",
"rtp_rtcp/source/rtcp_packet/sender_report_unittest.cc",
"rtp_rtcp/source/rtcp_packet/sli_unittest.cc",
Add RTCP packet class for signaling encoder target bitrate. This is a proposal for a new RTCP message. Feel free to comment on the message structure, selected type ids etc, as well as code for serialization/deserialization. Once we agree on this, I'll continue with wiring it up in the actual rtcp sender and receiver. BUG=webrtc:6301 Review-Url: https://codereview.webrtc.org/2306873003 Cr-Commit-Position: refs/heads/master@{#14867}
2016-11-01 02:50:12 -07:00
"rtp_rtcp/source/rtcp_packet/target_bitrate_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc",
"rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc",
"rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc",
"rtp_rtcp/source/rtcp_packet/voip_metric_unittest.cc",
"rtp_rtcp/source/rtcp_packet_unittest.cc",
"rtp_rtcp/source/rtcp_receiver_unittest.cc",
"rtp_rtcp/source/rtcp_sender_unittest.cc",
"rtp_rtcp/source/rtcp_utility_unittest.cc",
"rtp_rtcp/source/rtp_fec_unittest.cc",
"rtp_rtcp/source/rtp_format_h264_unittest.cc",
"rtp_rtcp/source/rtp_format_vp8_test_helper.cc",
"rtp_rtcp/source/rtp_format_vp8_test_helper.h",
"rtp_rtcp/source/rtp_format_vp8_unittest.cc",
"rtp_rtcp/source/rtp_format_vp9_unittest.cc",
"rtp_rtcp/source/rtp_header_extension_unittest.cc",
"rtp_rtcp/source/rtp_packet_history_unittest.cc",
"rtp_rtcp/source/rtp_packet_unittest.cc",
"rtp_rtcp/source/rtp_payload_registry_unittest.cc",
"rtp_rtcp/source/rtp_rtcp_impl_unittest.cc",
"rtp_rtcp/source/rtp_sender_unittest.cc",
Fix parsing padding byte in rtp header extension BUG=chromium:664598 Review-Url: https://codereview.webrtc.org/2498903003 Cr-Commit-Position: refs/heads/master@{#15230}
2016-11-24 09:06:05 -08:00
"rtp_rtcp/source/rtp_utility_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"rtp_rtcp/source/time_util_unittest.cc",
Rename ProducerFec to UlpfecGenerator. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2449783002 Cr-Commit-Position: refs/heads/master@{#14847}
2016-10-31 05:27:07 -07:00
"rtp_rtcp/source/ulpfec_generator_unittest.cc",
Generalize FEC header formatting. - Split out reading/writing of FEC headers to classes separate from ForwardErrorCorrection. This makes ForwardErrorCorrection oblivious to what FEC header scheme is used, and lets it focus on encoding/decoding the FEC payloads. - Add unit tests for FEC header readers/writers. - Split ForwardErrorCorrection::XorPackets into XorHeaders and XorPayloads and reuse these functions for both encoding and decoding. - Rename AttemptRecover -> AttemptRecovery in ForwardErrorCorrection. BUG=webrtc:5654 R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2260803002 . Cr-Commit-Position: refs/heads/master@{#14316}
2016-09-21 09:19:34 +02:00
"rtp_rtcp/source/ulpfec_header_reader_writer_unittest.cc",
Rename FecReceiver to UlpfecReceiver. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2451643002 Cr-Commit-Position: refs/heads/master@{#14846}
2016-10-31 04:51:33 -07:00
"rtp_rtcp/source/ulpfec_receiver_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"rtp_rtcp/source/vp8_partition_aggregator_unittest.cc",
"rtp_rtcp/test/testAPI/test_api.cc",
"rtp_rtcp/test/testAPI/test_api.h",
"rtp_rtcp/test/testAPI/test_api_audio.cc",
"rtp_rtcp/test/testAPI/test_api_rtcp.cc",
"rtp_rtcp/test/testAPI/test_api_video.cc",
"utility/source/audio_frame_operations_unittest.cc",
Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #3 id:40001 of https://codereview.webrtc.org/2247033003/ ) Reason for revert: Reverting, because it turns out that third-party code was using webrtc::FilePlayer. I'm not at all sure that this is something WebRTC ought to be exporting, but since we did export it, we have to live with it for now. Original issue's description: > Move FilePlayer and FileRecorder to Voice Engine > > Because Voice Engine was the only user. > > (This has been landed twice before, as > https://codereview.webrtc.org/2037623002 and > https://codereview.webrtc.org/2240163002. Third time's a charm!) > > NOPRESUBMIT=True > TBR=kjellander@webrtc.org > > Committed: https://crrev.com/427ce3d86f6328dc994f84a15c28bb7bfbaa46ef > Cr-Commit-Position: refs/heads/master@{#13777} TBR= # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review-Url: https://codereview.webrtc.org/2245413002 Cr-Commit-Position: refs/heads/master@{#13779}
2016-08-16 04:08:30 -07:00
"utility/source/file_player_unittests.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"utility/source/process_thread_impl_unittest.cc",
"video_coding/codecs/test/packet_manipulator_unittest.cc",
"video_coding/codecs/test/stats_unittest.cc",
"video_coding/codecs/test/videoprocessor_unittest.cc",
"video_coding/codecs/vp8/default_temporal_layers_unittest.cc",
"video_coding/codecs/vp8/reference_picture_selection_unittest.cc",
"video_coding/codecs/vp8/screenshare_layers_unittest.cc",
"video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc",
"video_coding/codecs/vp8/simulcast_unittest.cc",
"video_coding/codecs/vp8/simulcast_unittest.h",
"video_coding/decoding_state_unittest.cc",
"video_coding/frame_buffer2_unittest.cc",
H264SpsPpsTracker class which keep tracks of SPS/PPS. The H264SpsPpsTracker class: - Keeps track of all received SPS/PPS. - Decides whether a packet should be inserted into the PacketBuffer or not. - Don't insert if this packet only contains SPS and/or PPS. - Don't insert if this is the first packet of and IDR and we have not received the required SPS/PPS. - Insert start codes, and in the case of the first packet of an IDR prepend the bitstream with the given SPS/PPS for this IDR. BUG=webrtc:5514 Review-Url: https://codereview.webrtc.org/2466993003 Cr-Commit-Position: refs/heads/master@{#14906}
2016-11-03 04:03:01 -07:00
"video_coding/h264_sps_pps_tracker_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"video_coding/histogram_unittest.cc",
"video_coding/include/mock/mock_vcm_callbacks.h",
"video_coding/jitter_buffer_unittest.cc",
"video_coding/jitter_estimator_tests.cc",
"video_coding/nack_module_unittest.cc",
"video_coding/percentile_filter_unittest.cc",
"video_coding/protection_bitrate_calculator_unittest.cc",
"video_coding/receiver_unittest.cc",
PacketBuffer is now ref counted. Since all FrameObjects have a reference to its PacketBuffer and since the PacketBuffer can be thrown away at any moment the PacketBuffer has to be ref counted in order to avoid FrameObjects dereferencing a potentially destroyed object. BUG=webrtc:5514 R=danilchap@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2199133004 . Cr-Commit-Position: refs/heads/master@{#13725}
2016-08-11 15:09:26 +02:00
"video_coding/rtp_frame_reference_finder_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"video_coding/sequence_number_util_unittest.cc",
"video_coding/session_info_unittest.cc",
"video_coding/test/stream_generator.cc",
"video_coding/test/stream_generator.h",
"video_coding/timing_unittest.cc",
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
"video_coding/utility/default_video_bitrate_allocator_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"video_coding/utility/frame_dropper_unittest.cc",
"video_coding/utility/ivf_file_writer_unittest.cc",
Refactor QualityScaler and MovingAverage The MovingAverage class was very specific to the QualityScaler. This commit generalizes the MovingAverage class to be useful in other situations as well, and adapts the QualityScaler to use the new MovingAverage. BUG=webrtc:6304 Review-Url: https://codereview.webrtc.org/2310853002 Cr-Commit-Position: refs/heads/master@{#14207}
2016-09-14 02:14:58 -07:00
"video_coding/utility/moving_average_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"video_coding/utility/quality_scaler_unittest.cc",
Extract simulcast rate allocation outside of video encoder. This is a first step to refactor this code. I'm deprecating https://codereview.webrtc.org/1913073002 and implementing this in smaller more isolated steps. BUG=webrtc:5206 R=asapersson@webrtc.org, kjellander@webrtc.org, noahric@chromium.org Review URL: https://codereview.webrtc.org/2288223002 . Cr-Commit-Position: refs/heads/master@{#14186}
2016-09-12 16:04:43 +02:00
"video_coding/utility/simulcast_rate_allocator_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"video_coding/video_coding_robustness_unittest.cc",
"video_coding/video_packet_buffer_unittest.cc",
"video_coding/video_receiver_unittest.cc",
"video_coding/video_sender_unittest.cc",
"video_processing/test/denoiser_test.cc",
"video_processing/test/video_processing_unittest.cc",
"video_processing/test/video_processing_unittest.h",
]
In order to ensure that the same code is run in the tests as is otherwise run it is important that the same build flags are used in the code being tested. For the debugging functionality inside APM, that was not the case and this is corrected in this CL. This CL is chained to the CL https://codereview.webrtc.org/2300813004/ BUG=webrtc:5298 Review-Url: https://codereview.webrtc.org/2307563002 Cr-Commit-Position: refs/heads/master@{#14031}
2016-09-01 22:03:23 -07:00
if (apm_debug_dump) {
defines += [ "WEBRTC_APM_DEBUG_DUMP=1" ]
} else {
defines += [ "WEBRTC_APM_DEBUG_DUMP=0" ]
}
Deactivated the intelligibility enhancement functionality by default NOTRY=true BUG= Review-Url: https://codereview.webrtc.org/2272423003 Cr-Commit-Position: refs/heads/master@{#13937}
2016-08-26 07:16:04 -07:00
if (rtc_enable_intelligibility_enhancer) {
defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
sources += [
"audio_processing/intelligibility/intelligibility_enhancer_unittest.cc",
"audio_processing/intelligibility/intelligibility_utils_unittest.cc",
]
} else {
defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
if (rtc_libvpx_build_vp9) {
sources +=
[ "video_coding/codecs/vp9/vp9_screenshare_layers_unittest.cc" ]
}
if (rtc_desktop_capture_supported || is_android) {
deps += [ "desktop_capture" ]
sources += [
"desktop_capture/desktop_region_unittest.cc",
"desktop_capture/differ_block_unittest.cc",
]
}
if (rtc_desktop_capture_supported) {
Add CreateWindowCapturer() and CreateScreenCapturer() in DesktopCapturer This change copies ScreenCapturerDifferWrapper to a new DesktopCapturerDifferWrapper, and adds DesktopCapturer::CreateWindowCapturer and DesktopCapturer::CreateScreenCapturer functions to replace WindowCapturer::Create and ScreenCapturer::Create. BUG=webrtc:6513 Committed: https://crrev.com/b763e39beba92b45baa09542f949daabbe6258a3 Review-Url: https://codereview.webrtc.org/2468753002 Cr-Original-Commit-Position: refs/heads/master@{#14880} Cr-Commit-Position: refs/heads/master@{#14903}
2016-11-02 14:49:35 -07:00
deps += [ "desktop_capture:desktop_capture_mock" ]
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
sources += [
"desktop_capture/desktop_and_cursor_composer_unittest.cc",
Add CreateWindowCapturer() and CreateScreenCapturer() in DesktopCapturer This change copies ScreenCapturerDifferWrapper to a new DesktopCapturerDifferWrapper, and adds DesktopCapturer::CreateWindowCapturer and DesktopCapturer::CreateScreenCapturer functions to replace WindowCapturer::Create and ScreenCapturer::Create. BUG=webrtc:6513 Committed: https://crrev.com/b763e39beba92b45baa09542f949daabbe6258a3 Review-Url: https://codereview.webrtc.org/2468753002 Cr-Original-Commit-Position: refs/heads/master@{#14880} Cr-Commit-Position: refs/heads/master@{#14903}
2016-11-02 14:49:35 -07:00
"desktop_capture/desktop_capturer_differ_wrapper_unittest.cc",
Add DesktopFrame rotation functions This change adds RotateDesktopFrame(), RotateRect(), RotateSize(), ReverseRotate() functions, so an implementation can use these free functions to rotate and copy pixels from one DesktopFrame to another at the same time. This is the first part of the change to support rotation in DirectX capturer. In a coming change, these functions will be used in DxgiOutputDuplicator to do the rotation and copying. Background, DirectX APIs always return unrotated data buffer, so we need to rotate it to match the user-selected rotation. What worse is except for the data buffer, other variables return by these APIs are all rotated, e.g. output size, monitor position. So we will eventually not be able to capture the rotated monitors, because we cannot set their position and size correctly. Though DXGI_OUTDUPL_DESC provides a DXGI_MODE_ROTATION enumeration to indicate the output rotation, it does not provide a simple way to rotate an IDXGIResource, which is the only thing we can get from duplication APIs. A typical user case here is to use a matrix to transform the IDXGIResource and render it to a surface. But since we do not render the IDXGIResource at all, we need to manually rotate it. BUG=314516 Review-Url: https://codereview.webrtc.org/2500883004 Cr-Commit-Position: refs/heads/master@{#15205}
2016-11-22 17:17:09 -08:00
"desktop_capture/desktop_frame_rotation_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"desktop_capture/mouse_cursor_monitor_unittest.cc",
Use RgbaColor in DesktopFrameGenerator and add RgbaColorTest This change uses RgbaColor in DesktopFrameGenerator instead of raw uint32_t to avoid potential endian issues. BUG=633802 Review-Url: https://codereview.webrtc.org/2334853002 Cr-Commit-Position: refs/heads/master@{#14337}
2016-09-21 12:23:15 -07:00
"desktop_capture/rgba_color_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"desktop_capture/screen_capturer_helper_unittest.cc",
"desktop_capture/screen_capturer_mac_unittest.cc",
"desktop_capture/screen_capturer_mock_objects.h",
"desktop_capture/screen_capturer_unittest.cc",
Add DesktopFrame rotation functions This change adds RotateDesktopFrame(), RotateRect(), RotateSize(), ReverseRotate() functions, so an implementation can use these free functions to rotate and copy pixels from one DesktopFrame to another at the same time. This is the first part of the change to support rotation in DirectX capturer. In a coming change, these functions will be used in DxgiOutputDuplicator to do the rotation and copying. Background, DirectX APIs always return unrotated data buffer, so we need to rotate it to match the user-selected rotation. What worse is except for the data buffer, other variables return by these APIs are all rotated, e.g. output size, monitor position. So we will eventually not be able to capture the rotated monitors, because we cannot set their position and size correctly. Though DXGI_OUTDUPL_DESC provides a DXGI_MODE_ROTATION enumeration to indicate the output rotation, it does not provide a simple way to rotate an IDXGIResource, which is the only thing we can get from duplication APIs. A typical user case here is to use a matrix to transform the IDXGIResource and render it to a surface. But since we do not render the IDXGIResource at all, we need to manually rotate it. BUG=314516 Review-Url: https://codereview.webrtc.org/2500883004 Cr-Commit-Position: refs/heads/master@{#15205}
2016-11-22 17:17:09 -08:00
"desktop_capture/test_utils.cc",
"desktop_capture/test_utils.h",
"desktop_capture/test_utils_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"desktop_capture/win/cursor_unittest.cc",
"desktop_capture/win/cursor_unittest_resources.h",
"desktop_capture/win/cursor_unittest_resources.rc",
"desktop_capture/window_capturer_unittest.cc",
]
}
if (rtc_prefer_fixed_point) {
defines += [ "WEBRTC_AUDIOPROC_FIXED_PROFILE" ]
} else {
defines += [ "WEBRTC_AUDIOPROC_FLOAT_PROFILE" ]
}
if (rtc_enable_protobuf) {
defines += [
"WEBRTC_AUDIOPROC_DEBUG_DUMP",
"WEBRTC_NETEQ_UNITTEST_BITEXACT",
]
deps += [
"audio_coding:neteq_unittest_proto",
"audio_processing:audioproc_protobuf_utils",
"audio_processing:audioproc_unittest_proto",
]
sources += [
"audio_processing/audio_processing_impl_locking_unittest.cc",
"audio_processing/audio_processing_impl_unittest.cc",
"audio_processing/audio_processing_unittest.cc",
"audio_processing/echo_cancellation_bit_exact_unittest.cc",
"audio_processing/echo_control_mobile_unittest.cc",
Add algorithm for Residual Echo Detector. This algorithm calculates an estimate of the Pearson product-moment correlation coefficient between the power of 10ms audio buffers taken from the render and capture sides, for various different delay values. BUG=webrtc:6525 Review-Url: https://codereview.webrtc.org/2419563003 Cr-Commit-Position: refs/heads/master@{#14824}
2016-10-28 07:04:03 -07:00
"audio_processing/echo_detector/circular_buffer_unittest.cc",
"audio_processing/echo_detector/mean_variance_estimator_unittest.cc",
"audio_processing/echo_detector/normalized_covariance_estimator_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_processing/gain_control_unittest.cc",
Added missing unittest to the modules/BUILD.gn build file NOTRY=True BUG= Review-Url: https://codereview.webrtc.org/2255093002 Cr-Commit-Position: refs/heads/master@{#13813}
2016-08-18 02:41:48 -07:00
"audio_processing/level_controller/level_controller_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_processing/level_estimator_unittest.cc",
This CL introduces the new functionality for setting the APM parameters to the high-pass filter. The introduction will be done in three steps: 1) This CL which introduces the new scheme and changes the code in webrtcvoiceengine.cc to use it. 2) Introduce the scheme into upstream code. 3) Remove the HighPassFilter interface in APM. BUG=webrtc::6220, webrtc::6296, webrtc::6297, webrtc::6181, webrtc::5298 Review-Url: https://codereview.webrtc.org/2415403002 Cr-Commit-Position: refs/heads/master@{#15197}
2016-11-22 07:24:52 -08:00
"audio_processing/low_cut_filter_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_processing/noise_suppression_unittest.cc",
Add algorithm for Residual Echo Detector. This algorithm calculates an estimate of the Pearson product-moment correlation coefficient between the power of 10ms audio buffers taken from the render and capture sides, for various different delay values. BUG=webrtc:6525 Review-Url: https://codereview.webrtc.org/2419563003 Cr-Commit-Position: refs/heads/master@{#14824}
2016-10-28 07:04:03 -07:00
"audio_processing/residual_echo_detector_unittest.cc",
Adding a unit test for RMSLevel BUG=webrtc:6622 Review-Url: https://codereview.webrtc.org/2524273003 Cr-Commit-Position: refs/heads/master@{#15242}
2016-11-25 05:30:47 -08:00
"audio_processing/rms_level_unittest.cc",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_processing/test/bitexactness_tools.cc",
"audio_processing/test/bitexactness_tools.h",
"audio_processing/test/debug_dump_replayer.cc",
"audio_processing/test/debug_dump_replayer.h",
"audio_processing/test/debug_dump_test.cc",
"audio_processing/test/test_utils.h",
"audio_processing/voice_detection_unittest.cc",
]
}
if (rtc_build_libvpx) {
deps += [ rtc_libvpx_dir ]
}
# TODO(jschuh): bugs.webrtc.org/1348: fix this warning.
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
if (is_win) {
cflags = [
# TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
"/wd4373", # virtual function override.
]
}
deps += [
Relanding of "Adding BitrateController to audio network adaptor." Adding BitrateController to audio network adaptor was first landed in https://codereview.webrtc.org/2334613002/ but reverted in https://codereview.webrtc.org/2352223002/ due to ODR violation. This CL tries to use namespace trick to solve the ODR problem. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2353293002 Cr-Commit-Position: refs/heads/master@{#14343}
2016-09-21 23:01:26 -07:00
":audio_network_adaptor_unittests",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"..:webrtc_common",
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. transport.h defines an interface for sending rtp and rtcp packets, which is used by MediaChannel in webrtc/media/engine, {Audio|Video}{Send|Receive}Stream and in a few other places. It was part of the build target //webrtc:webrtc, which is a monolithic target with all webrtc production code. This CL moves the header to its own target in webrtc/api and deprecates the old location. Targets in webrtc/api should in general only depend on other targets in webrtc/api. The target webrtc/api:call_api depends on transport.h. This change also makes webrtc/voice_engine pass GN's header include checker and is needed in order for webrtc/api:call_api to pass it. transport.h will be completely removed in a follow-up CL in a few weeks after clients have updated their includes. NOTRY=True BUG=webrtc:5589, webrtc:5878, webrtc:6785 Review-Url: https://codereview.webrtc.org/2426563003 Cr-Commit-Position: refs/heads/master@{#15267}
2016-11-28 07:02:13 -08:00
"../api:transport_api",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"../base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
"../common_audio",
"../common_video",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
"../system_wrappers",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"../test:rtp_test_utils",
"../test:test_common",
Split audio mixer into interface and implementation. The AudioMixer is now split in a mixer and audio source interface part, which has moved to webrtc/api, and a default implementation part, which lies in webrtc/modules. This change makes it possible to create other mixer implementations and is a first step to facilitate passing down a mixer from outside of WebRTC. It will also create less build dependencies when the new mixer has replaced the old one. NOTRY=True TBR=henrik.lundin@webrtc.org BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2411313003 Cr-Commit-Position: refs/heads/master@{#14705}
2016-10-20 05:06:39 -07:00
"../test:test_support",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"../test:test_support_main",
"../test:video_test_common",
"audio_coding",
"audio_coding:acm_receive_test",
"audio_coding:acm_send_test",
"audio_coding:builtin_audio_decoder_factory",
"audio_coding:cng",
"audio_coding:isac_fix",
"audio_coding:neteq",
"audio_coding:neteq_test_support",
"audio_coding:neteq_unittest_tools",
"audio_coding:pcm16b",
"audio_coding:red",
"audio_coding:webrtc_opus",
"audio_conference_mixer",
"audio_device",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
"audio_mixer",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"audio_processing",
"audio_processing:audioproc_test_utils",
"bitrate_controller",
"media_file",
"pacing",
"remote_bitrate_estimator",
Make bwe_simulator a separate test target that doesnt run on the bots but remove the #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE so that it always builds. BUG=webrtc:6497 Review-Url: https://codereview.webrtc.org/2398123002 Cr-Commit-Position: refs/heads/master@{#14564}
2016-10-07 00:39:56 -07:00
"remote_bitrate_estimator:bwe_simulator_lib",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"rtp_rtcp",
"utility",
"video_capture",
"video_coding",
"video_coding:video_codecs_test_framework",
"video_coding:webrtc_vp8",
"video_coding:webrtc_vp9",
"video_processing",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
]
GN: Declare resources for targets. Declare resources for GN targets so that they can be isolated NOTRY=True BUG=chromium:497757 Review-Url: https://codereview.webrtc.org/2340753002 Cr-Commit-Position: refs/heads/master@{#14210}
2016-09-14 05:10:01 -07:00
data = modules_unittests_resources
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
if (is_android) {
GN: Add dependency libjingle_peerconnection_java to modules_unittests. This change only affects the Android build. BUG=webrtc:6189 TBR=tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2233903002 Cr-Commit-Position: refs/heads/master@{#13722}
2016-08-11 00:59:15 -07:00
deps += [
"//testing/android/native_test:native_test_support",
"//webrtc/api:libjingle_peerconnection_java",
]
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
# Need to disable error due to the line in
# base/android/jni_android.h triggering it:
# const BASE_EXPORT jobject GetApplicationContext()
# error: type qualifiers ignored on function return type
cflags = [ "-Wno-ignored-qualifiers" ]
sources += [
"audio_device/android/audio_device_unittest.cc",
"audio_device/android/audio_manager_unittest.cc",
"audio_device/android/ensure_initialized.cc",
"audio_device/android/ensure_initialized.h",
]
GN: Synchronize resources between Android and iOS. iOS tests packaged into an .app uses the same way of defining resources (the data attribute). Some iOS simulator tests are failing due to missing resources, so let's sync them all. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2277753003 Cr-Commit-Position: refs/heads/master@{#13898}
2016-08-24 07:48:42 -07:00
shard_timeout = 900
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
if (is_ios) {
Add custom info.plist to modules_unittests This is to fix an issue introduced with iOS 10 where all applications that access the microphone have to include a string in the Info.plist file explaining why they need it. BUG=webrtc:6403 Review-Url: https://codereview.webrtc.org/2359863003 Cr-Commit-Position: refs/heads/master@{#14354}
2016-09-22 08:00:52 -07:00
info_plist = "//webrtc/test/ios/Info.plist"
GN: Fix resource files for iOS test target In order to get resource files to be properly packaged into the .app for a unit test on iOS, the resource files needs to be listed as sources in a bundle_data target. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2292853002 Cr-Commit-Position: refs/heads/master@{#13968}
2016-08-30 02:53:49 -07:00
deps += [ ":modules_unittests_bundle_data" ]
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
configs += [
"..:common_objc",
"//build/config/compiler:enable_arc",
]
Reland of Add a webrtc{en,de}coderfactory implementation for VideoToolbox (patchset #1 id:1 of https://codereview.webrtc.org/2483273002/ ) Reason for revert: Fix gyp build Original issue's description: > Revert of Add a webrtc{en,de}coderfactory implementation for VideoToolbox (patchset #2 id:20001 of https://codereview.webrtc.org/2463313002/ ) > > Reason for revert: > Broke dependent project because the .gn changes weren't accompanied by corresponding .gyp changes. > > Original issue's description: > > Add a webrtc{en,de}coderfactory implementation for VideoToolbox > > > > This CL removes the coupling of the VideoToolbox h264 implementation > > to the generic h264 code. The files have been moved into sdb/obj/Framework > > and all dependency on them has been removed from the rest of WebRTC. > > We now add it as an external encoder via a factory supplied to the > > CreatePeerConnectionFactory call. This also brings the iOS implementation > > closer to what we do on Android for MediaCodec. > > > > BUG=webrtc:6619 > > > > Committed: https://crrev.com/6a5047dad31f14f53dd9f8bc1ecde19e1dede2e4 > > Cr-Commit-Position: refs/heads/master@{#14953} > > TBR=magjed@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org > BUG=webrtc:6619 > > Committed: https://crrev.com/d69ad84420d9c0e1c11450c352f6c92e7c9583f1 > Cr-Commit-Position: refs/heads/master@{#14985} R=magjed@webrtc.org TBR=kwiberg@webrtc.org, magjed@webrtc.org, stefan@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6619 Review URL: https://codereview.webrtc.org/2487723004 . Cr-Commit-Position: refs/heads/master@{#14992}
2016-11-09 10:43:26 +01:00
sources += [ "audio_device/ios/objc/RTCAudioSessionTest.mm" ]
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
Fix modules_unittests on iOS. Some of the module tests were failing on iOS, causing them to be ignored on the trybots. Specifically this CL: - Skips some tests that should not run on the simulator - Fixes iOS file system test helpers - Fixes failing nalu parser unittest BUG=webrtc:4752 R=henrika@webrtc.org, peah@webrtc.org, sprang@webrtc.org TBR=tkchin@webrtc.org Review URL: https://codereview.webrtc.org/2352063002 . Cr-Commit-Position: refs/heads/master@{#14472}
2016-10-03 13:13:29 +02:00
if (target_cpu != "x64") {
sources += [ "audio_device/ios/audio_device_unittest_ios.cc" ]
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
ldflags = [ "-ObjC" ]
}
}
Make bwe_simulator a separate test target that doesnt run on the bots but remove the #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE so that it always builds. BUG=webrtc:6497 Review-Url: https://codereview.webrtc.org/2398123002 Cr-Commit-Position: refs/heads/master@{#14564}
2016-10-07 00:39:56 -07:00
rtc_test("bwe_simulator") {
testonly = true
deps = []
sources = [
"remote_bitrate_estimator/bwe_simulations.cc",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
Make bwe_simulator a separate test target that doesnt run on the bots but remove the #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE so that it always builds. BUG=webrtc:6497 Review-Url: https://codereview.webrtc.org/2398123002 Cr-Commit-Position: refs/heads/master@{#14564}
2016-10-07 00:39:56 -07:00
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
if (is_win) {
cflags = [
# TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
"/wd4373", # virtual function override.
]
}
deps += [
"..:webrtc_common",
"../base:rtc_base_approved",
"../test:test_common",
"../test:test_support_main",
"remote_bitrate_estimator:bwe_simulator_lib",
"video_capture",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
]
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
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