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0bc58cf876
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'api'
Then undo changes to optional target itself and optional_unittests
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I44093da213369d6a502e33792c694f620f53b779
Reviewed-on: https://webrtc-review.googlesource.com/84621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23707}
101 lines
2.7 KiB
Text
101 lines
2.7 KiB
Text
# 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_source_set("audio_codecs_api") {
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visibility = [ "*" ]
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sources = [
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"audio_codec_pair_id.cc",
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"audio_codec_pair_id.h",
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"audio_decoder.cc",
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"audio_decoder.h",
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"audio_decoder_factory.h",
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"audio_decoder_factory_template.h",
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"audio_encoder.cc",
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"audio_encoder.h",
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"audio_encoder_factory.h",
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"audio_encoder_factory_template.h",
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"audio_format.cc",
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"audio_format.h",
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]
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deps = [
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"..:array_view",
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"../..:webrtc_common",
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"../../:typedefs",
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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:sanitizer",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_static_library("builtin_audio_decoder_factory") {
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visibility = [ "*" ]
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allow_poison = [ "audio_codecs" ]
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sources = [
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"builtin_audio_decoder_factory.cc",
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"builtin_audio_decoder_factory.h",
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]
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deps = [
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":audio_codecs_api",
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"../../rtc_base:rtc_base_approved",
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"L16:audio_decoder_L16",
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"g711:audio_decoder_g711",
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"g722:audio_decoder_g722",
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"isac:audio_decoder_isac",
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]
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defines = []
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if (rtc_include_ilbc) {
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deps += [ "ilbc:audio_decoder_ilbc" ]
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defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
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} else {
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defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
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}
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if (rtc_include_opus) {
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deps += [ "opus:audio_decoder_opus" ]
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defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
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} else {
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defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
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}
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}
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rtc_static_library("builtin_audio_encoder_factory") {
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visibility = [ "*" ]
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allow_poison = [ "audio_codecs" ]
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sources = [
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"builtin_audio_encoder_factory.cc",
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"builtin_audio_encoder_factory.h",
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]
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deps = [
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":audio_codecs_api",
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"../../rtc_base:rtc_base_approved",
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"L16:audio_encoder_L16",
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"g711:audio_encoder_g711",
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"g722:audio_encoder_g722",
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"isac:audio_encoder_isac",
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]
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defines = []
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if (rtc_include_ilbc) {
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deps += [ "ilbc:audio_encoder_ilbc" ]
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defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
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} else {
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defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
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}
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if (rtc_include_opus) {
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deps += [ "opus:audio_encoder_opus" ]
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defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
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} else {
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defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
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}
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}
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