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09ceed2165
API to injecting a heavy audio processing operation into WebRTC audio capture pipeline Bug: webrtc:12003 Change-Id: I9f6f58f468bd84efd0a9d53d703db6229a03959e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165788 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32291}
44 lines
1.2 KiB
Text
44 lines
1.2 KiB
Text
# Copyright (c) 2020 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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rtc_library("async_audio_processing") {
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sources = [
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"async_audio_processing.cc",
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"async_audio_processing.h",
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]
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public = [ "async_audio_processing.h" ]
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deps = [
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"../../api:scoped_refptr",
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"../../api/audio:audio_frame_api",
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"../../api/audio:audio_frame_processor",
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"../../api/task_queue:task_queue",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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"../../rtc_base:rtc_task_queue",
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"../../rtc_base/synchronization:sequence_checker",
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]
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}
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if (rtc_include_tests) {
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rtc_library("async_audio_processing_test") {
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testonly = true
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sources = []
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deps = [
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":async_audio_processing",
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"../../api/audio:audio_frame_api",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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]
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}
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}
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