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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}
72 lines
2.1 KiB
C++
72 lines
2.1 KiB
C++
/*
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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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*/
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#include "api/audio_codecs/builtin_audio_encoder_factory.h"
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#include <memory>
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#include <vector>
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#include "api/audio_codecs/L16/audio_encoder_L16.h"
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#include "api/audio_codecs/audio_encoder_factory_template.h"
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#include "api/audio_codecs/g711/audio_encoder_g711.h"
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#include "api/audio_codecs/g722/audio_encoder_g722.h"
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#if WEBRTC_USE_BUILTIN_ILBC
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#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h" // nogncheck
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#endif
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#include "api/audio_codecs/isac/audio_encoder_isac.h"
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#if WEBRTC_USE_BUILTIN_OPUS
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#include "api/audio_codecs/opus/audio_encoder_opus.h" // nogncheck
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#endif
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namespace webrtc {
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namespace {
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// Modify an audio encoder to not advertise support for anything.
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template <typename T>
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struct NotAdvertised {
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using Config = typename T::Config;
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static absl::optional<Config> SdpToConfig(
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const SdpAudioFormat& audio_format) {
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return T::SdpToConfig(audio_format);
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}
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static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
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// Don't advertise support for anything.
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}
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static AudioCodecInfo QueryAudioEncoder(const Config& config) {
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return T::QueryAudioEncoder(config);
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}
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static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
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const Config& config,
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int payload_type,
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absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) {
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return T::MakeAudioEncoder(config, payload_type, codec_pair_id);
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}
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};
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} // namespace
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rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory() {
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return CreateAudioEncoderFactory<
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#if WEBRTC_USE_BUILTIN_OPUS
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AudioEncoderOpus,
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#endif
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AudioEncoderIsac, AudioEncoderG722,
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#if WEBRTC_USE_BUILTIN_ILBC
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AudioEncoderIlbc,
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#endif
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AudioEncoderG711, NotAdvertised<AudioEncoderL16>>();
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}
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} // namespace webrtc
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