2017-06-30 04:23:22 -07:00
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# 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("audio_encoder_opus_config") {
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2018-01-10 15:54:53 +00:00
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visibility = [ "*" ]
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2017-06-30 04:23:22 -07:00
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sources = [
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"audio_encoder_opus_config.cc",
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"audio_encoder_opus_config.h",
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]
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deps = [
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2017-07-19 10:40:47 -07:00
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"../../../rtc_base:rtc_base_approved",
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2018-06-21 13:32:56 +02:00
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"//third_party/abseil-cpp/absl/types:optional",
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2017-06-30 04:23:22 -07:00
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]
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defines = []
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if (rtc_opus_variable_complexity) {
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defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
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} else {
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defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ]
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}
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}
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rtc_source_set("audio_encoder_opus") {
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2018-01-10 15:54:53 +00:00
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visibility = [ "*" ]
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2018-04-22 19:55:00 +02:00
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poisonous = [ "audio_codecs" ]
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2017-10-25 09:57:40 +02:00
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public = [
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2017-06-30 04:23:22 -07:00
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"audio_encoder_opus.h",
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]
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2017-10-25 09:57:40 +02:00
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sources = [
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"audio_encoder_opus.cc",
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]
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2017-06-30 04:23:22 -07:00
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deps = [
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":audio_encoder_opus_config",
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"..:audio_codecs_api",
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"../../../modules/audio_coding:webrtc_opus",
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2017-07-19 10:40:47 -07:00
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"../../../rtc_base:rtc_base_approved",
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2018-06-21 13:32:56 +02:00
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"//third_party/abseil-cpp/absl/types:optional",
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2017-06-30 04:23:22 -07:00
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]
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}
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2017-06-30 05:24:56 -07:00
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rtc_static_library("audio_decoder_opus") {
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2018-01-10 15:54:53 +00:00
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visibility = [ "*" ]
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2018-04-22 19:55:00 +02:00
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poisonous = [ "audio_codecs" ]
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2017-06-30 05:24:56 -07:00
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sources = [
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"audio_decoder_opus.cc",
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"audio_decoder_opus.h",
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]
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deps = [
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"..:audio_codecs_api",
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"../../..:webrtc_common",
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"../../../modules/audio_coding:webrtc_opus",
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2017-07-19 10:40:47 -07:00
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"../../../rtc_base:rtc_base_approved",
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Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
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"//third_party/abseil-cpp/absl/memory",
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2018-06-21 13:32:56 +02:00
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"//third_party/abseil-cpp/absl/types:optional",
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2017-06-30 05:24:56 -07:00
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]
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}
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