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.
27 lines
936 B
C
Executable File
27 lines
936 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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// This files is a forwarding header for other headers containing various
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// portability macros and functions.
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// This file is used for both C and C++!
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#ifndef ABSL_BASE_PORT_H_
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#define ABSL_BASE_PORT_H_
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#include "absl/base/attributes.h"
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#include "absl/base/config.h"
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#include "absl/base/optimization.h"
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#endif // ABSL_BASE_PORT_H_
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