mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
37 lines
944 B
C++
37 lines
944 B
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 <string>
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#ifndef RTC_BASE_PROTOBUF_UTILS_H_
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#define RTC_BASE_PROTOBUF_UTILS_H_
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namespace webrtc {
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using ProtoString = std::string;
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} // namespace webrtc
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#if WEBRTC_ENABLE_PROTOBUF
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#include "third_party/protobuf/src/google/protobuf/message_lite.h"
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#include "third_party/protobuf/src/google/protobuf/repeated_field.h"
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namespace webrtc {
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using google::protobuf::MessageLite;
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using google::protobuf::RepeatedPtrField;
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} // namespace webrtc
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#endif // WEBRTC_ENABLE_PROTOBUF
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#endif // RTC_BASE_PROTOBUF_UTILS_H_
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