Arun Raghavan 753eada3aa Update audio_processing module
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Pull in third party license file

 * Replace .gypi files with BUILD.gn to keep track of what changes
   upstream

 * Bunch of new filse pulled in as dependencies

 * Won't build yet due to changes needed on top of these
2015-10-15 16:18:45 +05:30

28 lines
878 B
C++

/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_COMMON_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_COMMON_H_
namespace webrtc {
namespace ts {
static const float kPi = 3.14159265358979323846f;
static const int kChunkSizeMs = 10;
enum {
kSampleRate8kHz = 8000,
kSampleRate16kHz = 16000,
kSampleRate32kHz = 32000,
kSampleRate48kHz = 48000
};
} // namespace ts
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_COMMON_H_