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.
34 lines
940 B
C++
Executable File
34 lines
940 B
C++
Executable File
// Copyright 2017 The Abseil Authors.
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//
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// Licensed under the Apache License, Version 2.0 (the "License");
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// you may not use this file except in compliance with the License.
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// You may obtain a copy of the License at
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//
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// http://www.apache.org/licenses/LICENSE-2.0
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//
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// Unless required by applicable law or agreed to in writing, software
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// distributed under the License is distributed on an "AS IS" BASIS,
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// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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// See the License for the specific language governing permissions and
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// limitations under the License.
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//
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#ifndef ABSL_BASE_INTERNAL_IDENTITY_H_
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#define ABSL_BASE_INTERNAL_IDENTITY_H_
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namespace absl {
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namespace internal {
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template <typename T>
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struct identity {
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typedef T type;
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};
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template <typename T>
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using identity_t = typename identity<T>::type;
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} // namespace internal
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} // namespace absl
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#endif // ABSL_BASE_INTERNAL_IDENTITY_H_
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