Update to current webrtc library
This is from the upstream library commit id 3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium 88.0.4290.1.
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@ -8,13 +8,11 @@
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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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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
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#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
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#define MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
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#include <cstdlib>
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#include "webrtc/typedefs.h"
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// Provides a set of static methods to perform dyadic decimations.
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namespace webrtc {
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@ -44,7 +42,7 @@ inline size_t GetOutLengthToDyadicDecimate(size_t in_length,
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// GetOutLengthToDyadicDecimate().
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// Must be previously allocated.
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// Returns the number of output samples, -1 on error.
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template<typename T>
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template <typename T>
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static size_t DyadicDecimate(const T* in,
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size_t in_length,
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bool odd_sequence,
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@ -67,4 +65,4 @@ static size_t DyadicDecimate(const T* in,
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
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#endif // MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
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