Update to current webrtc library

This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions

View File

@ -8,13 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
#define MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
#include <cstdlib>
#include "webrtc/typedefs.h"
// Provides a set of static methods to perform dyadic decimations.
namespace webrtc {
@ -44,7 +42,7 @@ inline size_t GetOutLengthToDyadicDecimate(size_t in_length,
// GetOutLengthToDyadicDecimate().
// Must be previously allocated.
// Returns the number of output samples, -1 on error.
template<typename T>
template <typename T>
static size_t DyadicDecimate(const T* in,
size_t in_length,
bool odd_sequence,
@ -67,4 +65,4 @@ static size_t DyadicDecimate(const T* in,
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
#endif // MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_