mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 21:30:45 +01:00
e5f4c6b8d2
This is a reland of 69241a93fb
Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33001}
675 lines
20 KiB
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675 lines
20 KiB
Text
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("//build/config/linux/pkg_config.gni")
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import("../webrtc.gni")
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group("media") {
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deps = []
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if (!build_with_mozilla) {
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deps += [
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":rtc_media",
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":rtc_media_base",
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]
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}
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}
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config("rtc_media_defines_config") {
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defines = [ "HAVE_WEBRTC_VIDEO" ]
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}
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rtc_library("rtc_h264_profile_id") {
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visibility = [ "*" ]
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sources = [
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"base/h264_profile_level_id.cc",
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"base/h264_profile_level_id.h",
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]
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deps = [
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"../rtc_base",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/system:rtc_export",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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}
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rtc_source_set("rtc_media_config") {
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visibility = [ "*" ]
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sources = [ "base/media_config.h" ]
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}
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rtc_library("rtc_vp9_profile") {
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visibility = [ "*" ]
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sources = [
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"base/vp9_profile.cc",
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"base/vp9_profile.h",
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]
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deps = [
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"../api/video_codecs:video_codecs_api",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/system:rtc_export",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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}
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rtc_library("rtc_sdp_fmtp_utils") {
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visibility = [ "*" ]
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sources = [
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"base/sdp_fmtp_utils.cc",
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"base/sdp_fmtp_utils.h",
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]
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deps = [
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"../api/video_codecs:video_codecs_api",
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"../rtc_base:stringutils",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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}
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rtc_library("rtc_media_base") {
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visibility = [ "*" ]
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defines = []
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libs = []
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deps = [
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":rtc_h264_profile_id",
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":rtc_media_config",
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":rtc_vp9_profile",
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"../api:array_view",
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"../api:audio_options_api",
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"../api:frame_transformer_interface",
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"../api:media_stream_interface",
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"../api:rtc_error",
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"../api:rtp_parameters",
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"../api:scoped_refptr",
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"../api/audio:audio_frame_processor",
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"../api/audio_codecs:audio_codecs_api",
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"../api/crypto:frame_decryptor_interface",
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"../api/crypto:frame_encryptor_interface",
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"../api/crypto:options",
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"../api/transport:stun_types",
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"../api/transport:webrtc_key_value_config",
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"../api/transport/rtp:rtp_source",
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"../api/video:video_bitrate_allocation",
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"../api/video:video_bitrate_allocator_factory",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../api/video_codecs:video_codecs_api",
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"../call:call_interfaces",
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"../call:video_stream_api",
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"../common_video",
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"../modules/async_audio_processing",
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"../modules/audio_processing:audio_processing_statistics",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_task_queue",
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"../rtc_base:sanitizer",
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"../rtc_base:socket",
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"../rtc_base:stringutils",
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"../rtc_base/synchronization:mutex",
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"../rtc_base/synchronization:sequence_checker",
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"../rtc_base/system:file_wrapper",
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"../rtc_base/system:rtc_export",
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"../rtc_base/third_party/sigslot",
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"../system_wrappers:field_trial",
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]
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absl_deps = [
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"//third_party/abseil-cpp/absl/algorithm:container",
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"//third_party/abseil-cpp/absl/strings",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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sources = [
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"base/adapted_video_track_source.cc",
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"base/adapted_video_track_source.h",
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"base/audio_source.h",
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"base/codec.cc",
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"base/codec.h",
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"base/delayable.h",
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"base/media_channel.cc",
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"base/media_channel.h",
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"base/media_constants.cc",
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"base/media_constants.h",
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"base/media_engine.cc",
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"base/media_engine.h",
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"base/rid_description.cc",
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"base/rid_description.h",
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"base/rtp_data_engine.cc",
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"base/rtp_data_engine.h",
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"base/rtp_utils.cc",
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"base/rtp_utils.h",
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"base/stream_params.cc",
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"base/stream_params.h",
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"base/turn_utils.cc",
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"base/turn_utils.h",
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"base/video_adapter.cc",
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"base/video_adapter.h",
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"base/video_broadcaster.cc",
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"base/video_broadcaster.h",
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"base/video_common.cc",
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"base/video_common.h",
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"base/video_source_base.cc",
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"base/video_source_base.h",
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]
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}
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rtc_library("rtc_constants") {
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defines = []
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libs = []
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deps = []
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sources = [
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"engine/constants.cc",
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"engine/constants.h",
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]
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}
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rtc_library("rtc_simulcast_encoder_adapter") {
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visibility = [ "*" ]
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defines = []
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libs = []
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sources = [
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"engine/simulcast_encoder_adapter.cc",
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"engine/simulcast_encoder_adapter.h",
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]
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deps = [
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":rtc_media_base",
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"../api:fec_controller_api",
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"../api:scoped_refptr",
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"../api/video:video_codec_constants",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../api/video_codecs:rtc_software_fallback_wrappers",
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"../api/video_codecs:video_codecs_api",
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"../call:video_stream_api",
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"../modules/video_coding:video_codec_interface",
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"../modules/video_coding:video_coding_utility",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/experiments:rate_control_settings",
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"../rtc_base/synchronization:sequence_checker",
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"../rtc_base/system:no_unique_address",
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"../rtc_base/system:rtc_export",
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"../system_wrappers",
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"../system_wrappers:field_trial",
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]
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absl_deps = [
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"//third_party/abseil-cpp/absl/algorithm:container",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_library("rtc_encoder_simulcast_proxy") {
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visibility = [ "*" ]
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defines = []
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libs = []
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sources = [
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"engine/encoder_simulcast_proxy.cc",
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"engine/encoder_simulcast_proxy.h",
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]
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deps = [
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":rtc_simulcast_encoder_adapter",
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"../api/video:video_bitrate_allocation",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../api/video_codecs:video_codecs_api",
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"../modules/video_coding:video_codec_interface",
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"../rtc_base/system:rtc_export",
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]
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}
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rtc_library("rtc_internal_video_codecs") {
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visibility = [ "*" ]
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allow_poison = [ "software_video_codecs" ]
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defines = []
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libs = []
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deps = [
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":rtc_constants",
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":rtc_encoder_simulcast_proxy",
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":rtc_h264_profile_id",
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":rtc_media_base",
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":rtc_simulcast_encoder_adapter",
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"../api/video:encoded_image",
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"../api/video:video_bitrate_allocation",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../api/video_codecs:rtc_software_fallback_wrappers",
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"../api/video_codecs:video_codecs_api",
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"../call:call_interfaces",
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"../call:video_stream_api",
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"../modules/video_coding:video_codec_interface",
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"../modules/video_coding:webrtc_h264",
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"../modules/video_coding:webrtc_multiplex",
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"../modules/video_coding:webrtc_vp8",
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"../modules/video_coding:webrtc_vp9",
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"../modules/video_coding/codecs/av1:libaom_av1_decoder",
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"../modules/video_coding/codecs/av1:libaom_av1_encoder",
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"../rtc_base:checks",
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"../rtc_base:deprecation",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/system:rtc_export",
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"../test:fake_video_codecs",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
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sources = [
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"engine/fake_video_codec_factory.cc",
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"engine/fake_video_codec_factory.h",
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"engine/internal_decoder_factory.cc",
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"engine/internal_decoder_factory.h",
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"engine/internal_encoder_factory.cc",
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"engine/internal_encoder_factory.h",
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"engine/multiplex_codec_factory.cc",
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"engine/multiplex_codec_factory.h",
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# TODO(bugs.webrtc.org/7925): stop exporting this header once downstream
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# targets depend on :rtc_encoder_simulcast_proxy directly.
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"engine/encoder_simulcast_proxy.h",
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]
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}
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rtc_library("rtc_audio_video") {
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visibility = [ "*" ]
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allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
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defines = []
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libs = []
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deps = [
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":rtc_constants",
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":rtc_media_base",
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"../api:call_api",
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"../api:libjingle_peerconnection_api",
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"../api:media_stream_interface",
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"../api:rtp_parameters",
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"../api:scoped_refptr",
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"../api:transport_api",
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"../api/audio:audio_frame_processor",
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"../api/audio:audio_mixer_api",
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"../api/audio_codecs:audio_codecs_api",
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"../api/task_queue",
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"../api/transport:bitrate_settings",
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"../api/transport:field_trial_based_config",
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"../api/transport:webrtc_key_value_config",
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"../api/transport/rtp:rtp_source",
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"../api/units:data_rate",
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"../api/video:video_bitrate_allocation",
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"../api/video:video_bitrate_allocator_factory",
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"../api/video:video_codec_constants",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../api/video_codecs:rtc_software_fallback_wrappers",
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"../api/video_codecs:video_codecs_api",
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"../call",
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"../call:call_interfaces",
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"../call:video_stream_api",
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"../common_video",
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"../modules/async_audio_processing:async_audio_processing",
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"../modules/audio_device",
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"../modules/audio_device:audio_device_impl",
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/audio_processing:api",
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"../modules/audio_processing/aec_dump",
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"../modules/audio_processing/agc:gain_control_interface",
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"../modules/video_coding",
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"../modules/video_coding:video_codec_interface",
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"../modules/video_coding:video_coding_utility",
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"../rtc_base",
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"../rtc_base:audio_format_to_string",
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"../rtc_base:checks",
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"../rtc_base:ignore_wundef",
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"../rtc_base:rtc_task_queue",
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"../rtc_base:stringutils",
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"../rtc_base:threading",
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"../rtc_base/experiments:field_trial_parser",
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"../rtc_base/experiments:min_video_bitrate_experiment",
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"../rtc_base/experiments:normalize_simulcast_size_experiment",
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"../rtc_base/experiments:rate_control_settings",
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"../rtc_base/synchronization:mutex",
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"../rtc_base/system:rtc_export",
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"../rtc_base/third_party/base64",
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"../system_wrappers",
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"../system_wrappers:metrics",
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]
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absl_deps = [
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"//third_party/abseil-cpp/absl/algorithm:container",
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"//third_party/abseil-cpp/absl/strings",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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sources = [
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"engine/adm_helpers.cc",
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"engine/adm_helpers.h",
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"engine/null_webrtc_video_engine.h",
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"engine/payload_type_mapper.cc",
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"engine/payload_type_mapper.h",
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"engine/simulcast.cc",
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"engine/simulcast.h",
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"engine/unhandled_packets_buffer.cc",
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"engine/unhandled_packets_buffer.h",
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"engine/webrtc_media_engine.cc",
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"engine/webrtc_media_engine.h",
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"engine/webrtc_video_engine.cc",
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"engine/webrtc_video_engine.h",
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"engine/webrtc_voice_engine.cc",
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"engine/webrtc_voice_engine.h",
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]
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public_configs = []
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if (!build_with_chromium) {
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public_configs += [ ":rtc_media_defines_config" ]
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deps += [ "../modules/video_capture:video_capture_internal_impl" ]
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}
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if (rtc_enable_protobuf) {
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deps += [
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"../modules/audio_coding:ana_config_proto",
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"../modules/audio_processing/aec_dump:aec_dump_impl",
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]
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} else {
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deps += [ "../modules/audio_processing/aec_dump:null_aec_dump_factory" ]
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}
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}
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# Heavy but optional helper for unittests and webrtc users who prefer to use
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# defaults factories or do not worry about extra dependencies and binary size.
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rtc_library("rtc_media_engine_defaults") {
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visibility = [ "*" ]
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allow_poison = [
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"audio_codecs",
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"default_task_queue",
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"software_video_codecs",
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]
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sources = [
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"engine/webrtc_media_engine_defaults.cc",
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"engine/webrtc_media_engine_defaults.h",
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]
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deps = [
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":rtc_audio_video",
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"../api/audio_codecs:builtin_audio_decoder_factory",
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"../api/audio_codecs:builtin_audio_encoder_factory",
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"../api/task_queue:default_task_queue_factory",
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"../api/video:builtin_video_bitrate_allocator_factory",
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"../api/video_codecs:builtin_video_decoder_factory",
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"../api/video_codecs:builtin_video_encoder_factory",
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"../modules/audio_processing:api",
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"../rtc_base:checks",
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"../rtc_base/system:rtc_export",
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]
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}
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rtc_library("rtc_data") {
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defines = [
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# "SCTP_DEBUG" # Uncomment for SCTP debugging.
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]
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deps = [
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":rtc_media_base",
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"../api:call_api",
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"../api:transport_api",
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"../p2p:rtc_p2p",
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"../rtc_base",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:threading",
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"../rtc_base/synchronization:mutex",
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"../rtc_base/task_utils:pending_task_safety_flag",
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"../rtc_base/task_utils:to_queued_task",
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"../rtc_base/third_party/sigslot",
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"../system_wrappers",
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]
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absl_deps = [
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"//third_party/abseil-cpp/absl/algorithm:container",
|
|
"//third_party/abseil-cpp/absl/base:core_headers",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
|
|
if (rtc_enable_sctp) {
|
|
sources = [
|
|
"sctp/sctp_transport.cc",
|
|
"sctp/sctp_transport.h",
|
|
"sctp/sctp_transport_internal.h",
|
|
]
|
|
} else {
|
|
# libtool on mac does not like empty targets.
|
|
sources = [ "sctp/noop.cc" ]
|
|
}
|
|
|
|
if (rtc_enable_sctp && rtc_build_usrsctp) {
|
|
deps += [
|
|
"../api/transport:sctp_transport_factory_interface",
|
|
"//third_party/usrsctp",
|
|
]
|
|
}
|
|
}
|
|
|
|
rtc_source_set("rtc_media") {
|
|
visibility = [ "*" ]
|
|
allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
|
|
deps = [
|
|
":rtc_audio_video",
|
|
":rtc_data",
|
|
]
|
|
}
|
|
|
|
if (rtc_include_tests) {
|
|
rtc_library("rtc_media_tests_utils") {
|
|
testonly = true
|
|
|
|
defines = []
|
|
deps = [
|
|
":rtc_audio_video",
|
|
":rtc_internal_video_codecs",
|
|
":rtc_media",
|
|
":rtc_media_base",
|
|
":rtc_simulcast_encoder_adapter",
|
|
"../api:call_api",
|
|
"../api:fec_controller_api",
|
|
"../api:scoped_refptr",
|
|
"../api/transport:field_trial_based_config",
|
|
"../api/video:encoded_image",
|
|
"../api/video:video_bitrate_allocation",
|
|
"../api/video:video_frame",
|
|
"../api/video:video_rtp_headers",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../call:call_interfaces",
|
|
"../call:mock_rtp_interfaces",
|
|
"../call:video_stream_api",
|
|
"../common_video",
|
|
"../modules/audio_processing",
|
|
"../modules/audio_processing:api",
|
|
"../modules/rtp_rtcp:rtp_rtcp_format",
|
|
"../modules/video_coding:video_codec_interface",
|
|
"../modules/video_coding:video_coding_utility",
|
|
"../p2p:rtc_p2p",
|
|
"../rtc_base",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:gunit_helpers",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../rtc_base:stringutils",
|
|
"../rtc_base:threading",
|
|
"../rtc_base/synchronization:mutex",
|
|
"../rtc_base/third_party/sigslot",
|
|
"../test:test_support",
|
|
"//testing/gtest",
|
|
]
|
|
absl_deps = [
|
|
"//third_party/abseil-cpp/absl/algorithm:container",
|
|
"//third_party/abseil-cpp/absl/strings",
|
|
]
|
|
sources = [
|
|
"base/fake_frame_source.cc",
|
|
"base/fake_frame_source.h",
|
|
"base/fake_media_engine.cc",
|
|
"base/fake_media_engine.h",
|
|
"base/fake_network_interface.h",
|
|
"base/fake_rtp.cc",
|
|
"base/fake_rtp.h",
|
|
"base/fake_video_renderer.cc",
|
|
"base/fake_video_renderer.h",
|
|
"base/test_utils.cc",
|
|
"base/test_utils.h",
|
|
"engine/fake_webrtc_call.cc",
|
|
"engine/fake_webrtc_call.h",
|
|
"engine/fake_webrtc_video_engine.cc",
|
|
"engine/fake_webrtc_video_engine.h",
|
|
]
|
|
}
|
|
|
|
rtc_media_unittests_resources = [
|
|
"../resources/media/captured-320x240-2s-48.frames",
|
|
"../resources/media/faces.1280x720_P420.yuv",
|
|
"../resources/media/faces_I400.jpg",
|
|
"../resources/media/faces_I411.jpg",
|
|
"../resources/media/faces_I420.jpg",
|
|
"../resources/media/faces_I422.jpg",
|
|
"../resources/media/faces_I444.jpg",
|
|
]
|
|
|
|
if (is_ios) {
|
|
bundle_data("rtc_media_unittests_bundle_data") {
|
|
testonly = true
|
|
sources = rtc_media_unittests_resources
|
|
outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
|
|
}
|
|
}
|
|
|
|
rtc_test("rtc_media_unittests") {
|
|
testonly = true
|
|
|
|
defines = []
|
|
deps = [
|
|
":rtc_audio_video",
|
|
":rtc_constants",
|
|
":rtc_data",
|
|
":rtc_encoder_simulcast_proxy",
|
|
":rtc_internal_video_codecs",
|
|
":rtc_media",
|
|
":rtc_media_base",
|
|
":rtc_media_engine_defaults",
|
|
":rtc_media_tests_utils",
|
|
":rtc_sdp_fmtp_utils",
|
|
":rtc_simulcast_encoder_adapter",
|
|
":rtc_vp9_profile",
|
|
"../api:create_simulcast_test_fixture_api",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:mock_video_bitrate_allocator",
|
|
"../api:mock_video_bitrate_allocator_factory",
|
|
"../api:mock_video_codec_factory",
|
|
"../api:mock_video_encoder",
|
|
"../api:rtp_parameters",
|
|
"../api:scoped_refptr",
|
|
"../api:simulcast_test_fixture_api",
|
|
"../api/audio_codecs:builtin_audio_decoder_factory",
|
|
"../api/audio_codecs:builtin_audio_encoder_factory",
|
|
"../api/rtc_event_log",
|
|
"../api/task_queue",
|
|
"../api/task_queue:default_task_queue_factory",
|
|
"../api/test/video:function_video_factory",
|
|
"../api/transport:field_trial_based_config",
|
|
"../api/units:time_delta",
|
|
"../api/video:builtin_video_bitrate_allocator_factory",
|
|
"../api/video:video_bitrate_allocation",
|
|
"../api/video:video_frame",
|
|
"../api/video:video_rtp_headers",
|
|
"../api/video_codecs:builtin_video_decoder_factory",
|
|
"../api/video_codecs:builtin_video_encoder_factory",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../audio",
|
|
"../call:call_interfaces",
|
|
"../common_video",
|
|
"../media:rtc_h264_profile_id",
|
|
"../modules/audio_device:mock_audio_device",
|
|
"../modules/audio_processing",
|
|
"../modules/audio_processing:api",
|
|
"../modules/audio_processing:mocks",
|
|
"../modules/rtp_rtcp",
|
|
"../modules/video_coding:simulcast_test_fixture_impl",
|
|
"../modules/video_coding:video_codec_interface",
|
|
"../modules/video_coding:webrtc_h264",
|
|
"../modules/video_coding:webrtc_vp8",
|
|
"../modules/video_coding/codecs/av1:libaom_av1_decoder",
|
|
"../p2p:p2p_test_utils",
|
|
"../rtc_base",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:gunit_helpers",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../rtc_base:stringutils",
|
|
"../rtc_base:threading",
|
|
"../rtc_base/experiments:min_video_bitrate_experiment",
|
|
"../rtc_base/synchronization:mutex",
|
|
"../rtc_base/third_party/sigslot",
|
|
"../test:audio_codec_mocks",
|
|
"../test:fake_video_codecs",
|
|
"../test:field_trial",
|
|
"../test:rtp_test_utils",
|
|
"../test:test_main",
|
|
"../test:test_support",
|
|
"../test:video_test_common",
|
|
"//third_party/abseil-cpp/absl/algorithm:container",
|
|
"//third_party/abseil-cpp/absl/memory",
|
|
"//third_party/abseil-cpp/absl/strings",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
sources = [
|
|
"base/codec_unittest.cc",
|
|
"base/media_engine_unittest.cc",
|
|
"base/rtp_data_engine_unittest.cc",
|
|
"base/rtp_utils_unittest.cc",
|
|
"base/sdp_fmtp_utils_unittest.cc",
|
|
"base/stream_params_unittest.cc",
|
|
"base/turn_utils_unittest.cc",
|
|
"base/video_adapter_unittest.cc",
|
|
"base/video_broadcaster_unittest.cc",
|
|
"base/video_common_unittest.cc",
|
|
"engine/encoder_simulcast_proxy_unittest.cc",
|
|
"engine/internal_decoder_factory_unittest.cc",
|
|
"engine/multiplex_codec_factory_unittest.cc",
|
|
"engine/null_webrtc_video_engine_unittest.cc",
|
|
"engine/payload_type_mapper_unittest.cc",
|
|
"engine/simulcast_encoder_adapter_unittest.cc",
|
|
"engine/simulcast_unittest.cc",
|
|
"engine/unhandled_packets_buffer_unittest.cc",
|
|
"engine/webrtc_media_engine_unittest.cc",
|
|
"engine/webrtc_video_engine_unittest.cc",
|
|
]
|
|
|
|
# TODO(kthelgason): Reenable this test on iOS.
|
|
# See bugs.webrtc.org/5569
|
|
if (!is_ios) {
|
|
sources += [ "engine/webrtc_voice_engine_unittest.cc" ]
|
|
}
|
|
|
|
if (rtc_enable_sctp) {
|
|
sources += [
|
|
"sctp/sctp_transport_reliability_unittest.cc",
|
|
"sctp/sctp_transport_unittest.cc",
|
|
]
|
|
}
|
|
|
|
if (rtc_opus_support_120ms_ptime) {
|
|
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
|
|
} else {
|
|
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
|
|
}
|
|
|
|
data = rtc_media_unittests_resources
|
|
|
|
if (is_android) {
|
|
deps += [ "//testing/android/native_test:native_test_support" ]
|
|
shard_timeout = 900
|
|
}
|
|
|
|
if (is_ios) {
|
|
deps += [ ":rtc_media_unittests_bundle_data" ]
|
|
}
|
|
|
|
if (rtc_enable_sctp && rtc_build_usrsctp) {
|
|
deps += [ "//third_party/usrsctp" ]
|
|
}
|
|
}
|
|
}
|