a9cc40b7d2
[This CL is a rebase of an original CL by solenberg@: https://codereview.webrtc.org/2948763002/ which in turn was a rebase of an original CL by peah@: https://chromium-review.googlesource.com/c/527032/] Allow an external audio processing module to be used in WebRTC This CL adds support for optionally using an externally created audio processing module in a peerconnection. The ownership is shared between the peerconnection and the external creator of the module. As part of this the internal ownership of the audio processing module is moved from VoiceEngine to WebRtcVoiceEngine. BUG=webrtc:7775 Review-Url: https://codereview.webrtc.org/2961723004 Cr-Commit-Position: refs/heads/master@{#18837}
100 lines
2.9 KiB
Plaintext
100 lines
2.9 KiB
Plaintext
# Copyright (c) 2017 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("../webrtc.gni")
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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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rtc_static_library("ortc") {
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defines = []
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sources = [
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"ortcfactory.cc",
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"ortcfactory.h",
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"ortcrtpreceiveradapter.cc",
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"ortcrtpreceiveradapter.h",
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"ortcrtpsenderadapter.cc",
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"ortcrtpsenderadapter.h",
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"rtpparametersconversion.cc",
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"rtpparametersconversion.h",
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"rtptransportadapter.cc",
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"rtptransportadapter.h",
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"rtptransportcontrolleradapter.cc",
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"rtptransportcontrolleradapter.h",
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]
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# TODO(deadbeef): Create a separate target for the common things ORTC and
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# PeerConnection code shares, so that ortc can depend on that instead of
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# libjingle_peerconnection.
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deps = [
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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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"../base:rtc_base",
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"../base:rtc_base_approved",
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"../call:call_interfaces",
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"../logging:rtc_event_log_api",
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"../media:rtc_media",
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"../media:rtc_media_base",
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"../modules/audio_processing:audio_processing",
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"../p2p:rtc_p2p",
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"../pc:libjingle_peerconnection",
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"../pc:rtc_pc",
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]
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public_deps = [
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"../api:ortc_api",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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if (rtc_include_tests) {
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rtc_test("ortc_unittests") {
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testonly = true
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sources = [
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"ortcfactory_integrationtest.cc",
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"ortcfactory_unittest.cc",
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"ortcrtpreceiver_unittest.cc",
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"ortcrtpsender_unittest.cc",
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"rtpparametersconversion_unittest.cc",
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"rtptransport_unittest.cc",
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"rtptransportcontroller_unittest.cc",
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"srtptransport_unittest.cc",
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"testrtpparameters.cc",
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"testrtpparameters.h",
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]
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deps = [
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":ortc",
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"../base:rtc_base",
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"../base:rtc_base_approved",
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"../base:rtc_base_tests_main",
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"../base:rtc_base_tests_utils",
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"../media:rtc_media_tests_utils",
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"../p2p:p2p_test_utils",
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"../p2p:rtc_p2p",
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"../pc:pc_test_utils",
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"../system_wrappers:metrics_default",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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if (is_android) {
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deps += [ "//testing/android/native_test:native_test_support" ]
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}
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}
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}
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