/* * Copyright (c) 2013 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_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_ #define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_ #include "webrtc/typedefs.h" namespace webrtc { // Clamp the floating |value| to the range representable by an int16_t. static inline float ClampInt16(float value) { const float kMaxInt16 = 32767.f; const float kMinInt16 = -32768.f; return value < kMinInt16 ? kMinInt16 : (value > kMaxInt16 ? kMaxInt16 : value); } // Return a rounded int16_t of the floating |value|. Doesn't handle overflow; // use ClampInt16 if necessary. static inline int16_t RoundToInt16(float value) { return static_cast(value < 0.f ? value - 0.5f : value + 0.5f); } // Deinterleave audio from |interleaved| to the channel buffers pointed to // by |deinterleaved|. There must be sufficient space allocated in the // |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel| // per buffer). void Deinterleave(const int16_t* interleaved, int samples_per_channel, int num_channels, int16_t** deinterleaved); // Interleave audio from the channel buffers pointed to by |deinterleaved| to // |interleaved|. There must be sufficient space allocated in |interleaved| // (|samples_per_channel| * |num_channels|). void Interleave(const int16_t* const* deinterleaved, int samples_per_channel, int num_channels, int16_t* interleaved); } // namespace webrtc #endif // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_