blob: 14c5c225a331711cca352eb12b2d21342eb860b2 [file] [log] [blame]
# Copyright 2018 The Chromium Authors. All rights reserved.
# Use of this source code is governed by a BSD-style license that can be
# found in the LICENSE file.
import("//media/media_options.gni")
import("//media/webrtc/audio_processing.gni")
import("//third_party/webrtc/webrtc.gni")
config("audio_processing_build_flag") {
if (audio_processing_in_audio_service_supported) {
defines = [ "AUDIO_PROCESSING_IN_AUDIO_SERVICE" ]
}
}
component("webrtc") {
output_name = "media_webrtc"
sources = [
"audio_delay_stats_reporter.cc",
"audio_delay_stats_reporter.h",
"helpers.cc",
"helpers.h",
"webrtc_switches.cc",
"webrtc_switches.h",
]
defines = [ "IS_MEDIA_WEBRTC_IMPL" ]
deps = [
"//base",
"//media:shared_memory_support",
"//third_party/webrtc/modules/audio_processing",
"//third_party/webrtc/modules/audio_processing:api",
"//third_party/webrtc/modules/audio_processing:audio_processing_statistics",
"//third_party/webrtc_overrides:init_webrtc",
]
if (audio_processing_in_audio_service_supported) {
# Only build this on platforms where it's supported and used.
sources += [
"audio_processor.cc",
"audio_processor.h",
"audio_processor_controls.h",
]
deps += [
"//media",
"//third_party/webrtc/api:libjingle_peerconnection_api",
"//third_party/webrtc/api:media_stream_interface",
"//third_party/webrtc/api/audio:aec3_factory",
"//third_party/webrtc/modules/audio_processing/aec_dump:aec_dump",
"//third_party/webrtc/rtc_base:rtc_task_queue",
"//third_party/webrtc_overrides:task_queue_factory",
]
public_configs = [ ":audio_processing_build_flag" ]
}
}
source_set("unit_tests") {
testonly = true
if (audio_processing_in_audio_service_supported) {
deps = [
"//base",
"//base/test:test_support",
"//media:test_support",
"//media/webrtc",
"//testing/gmock",
"//testing/gtest",
"//third_party/webrtc/modules/audio_processing",
"//third_party/webrtc_overrides:init_webrtc",
]
sources = [
"audio_processor_unittest.cc",
]
}
}