|  | /* | 
|  | *  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 <limits> | 
|  | #include <cstring> | 
|  |  | 
|  | #include "webrtc/base/checks.h" | 
|  | #include "webrtc/typedefs.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | typedef std::numeric_limits<int16_t> limits_int16; | 
|  |  | 
|  | // The conversion functions use the following naming convention: | 
|  | // S16:      int16_t [-32768, 32767] | 
|  | // Float:    float   [-1.0, 1.0] | 
|  | // FloatS16: float   [-32768.0, 32767.0] | 
|  | static inline int16_t FloatToS16(float v) { | 
|  | if (v > 0) | 
|  | return v >= 1 ? limits_int16::max() | 
|  | : static_cast<int16_t>(v * limits_int16::max() + 0.5f); | 
|  | return v <= -1 ? limits_int16::min() | 
|  | : static_cast<int16_t>(-v * limits_int16::min() - 0.5f); | 
|  | } | 
|  |  | 
|  | static inline float S16ToFloat(int16_t v) { | 
|  | static const float kMaxInt16Inverse = 1.f / limits_int16::max(); | 
|  | static const float kMinInt16Inverse = 1.f / limits_int16::min(); | 
|  | return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse); | 
|  | } | 
|  |  | 
|  | static inline int16_t FloatS16ToS16(float v) { | 
|  | static const float kMaxRound = limits_int16::max() - 0.5f; | 
|  | static const float kMinRound = limits_int16::min() + 0.5f; | 
|  | if (v > 0) | 
|  | return v >= kMaxRound ? limits_int16::max() | 
|  | : static_cast<int16_t>(v + 0.5f); | 
|  | return v <= kMinRound ? limits_int16::min() : static_cast<int16_t>(v - 0.5f); | 
|  | } | 
|  |  | 
|  | static inline float FloatToFloatS16(float v) { | 
|  | return v * (v > 0 ? limits_int16::max() : -limits_int16::min()); | 
|  | } | 
|  |  | 
|  | static inline float FloatS16ToFloat(float v) { | 
|  | static const float kMaxInt16Inverse = 1.f / limits_int16::max(); | 
|  | static const float kMinInt16Inverse = 1.f / limits_int16::min(); | 
|  | return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse); | 
|  | } | 
|  |  | 
|  | void FloatToS16(const float* src, size_t size, int16_t* dest); | 
|  | void S16ToFloat(const int16_t* src, size_t size, float* dest); | 
|  | void FloatS16ToS16(const float* src, size_t size, int16_t* dest); | 
|  | void FloatToFloatS16(const float* src, size_t size, float* dest); | 
|  | void FloatS16ToFloat(const float* src, size_t size, float* dest); | 
|  |  | 
|  | // Copy audio from |src| channels to |dest| channels unless |src| and |dest| | 
|  | // point to the same address. |src| and |dest| must have the same number of | 
|  | // channels, and there must be sufficient space allocated in |dest|. | 
|  | template <typename T> | 
|  | void CopyAudioIfNeeded(const T* const* src, | 
|  | int num_frames, | 
|  | int num_channels, | 
|  | T* const* dest) { | 
|  | for (int i = 0; i < num_channels; ++i) { | 
|  | if (src[i] != dest[i]) { | 
|  | std::copy(src[i], src[i] + num_frames, dest[i]); | 
|  | } | 
|  | } | 
|  | } | 
|  |  | 
|  | // 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). | 
|  | template <typename T> | 
|  | void Deinterleave(const T* interleaved, | 
|  | size_t samples_per_channel, | 
|  | size_t num_channels, | 
|  | T* const* deinterleaved) { | 
|  | for (size_t i = 0; i < num_channels; ++i) { | 
|  | T* channel = deinterleaved[i]; | 
|  | size_t interleaved_idx = i; | 
|  | for (size_t j = 0; j < samples_per_channel; ++j) { | 
|  | channel[j] = interleaved[interleaved_idx]; | 
|  | interleaved_idx += num_channels; | 
|  | } | 
|  | } | 
|  | } | 
|  |  | 
|  | // 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|). | 
|  | template <typename T> | 
|  | void Interleave(const T* const* deinterleaved, | 
|  | size_t samples_per_channel, | 
|  | size_t num_channels, | 
|  | T* interleaved) { | 
|  | for (size_t i = 0; i < num_channels; ++i) { | 
|  | const T* channel = deinterleaved[i]; | 
|  | size_t interleaved_idx = i; | 
|  | for (size_t j = 0; j < samples_per_channel; ++j) { | 
|  | interleaved[interleaved_idx] = channel[j]; | 
|  | interleaved_idx += num_channels; | 
|  | } | 
|  | } | 
|  | } | 
|  |  | 
|  | // Copies audio from a single channel buffer pointed to by |mono| to each | 
|  | // channel of |interleaved|. There must be sufficient space allocated in | 
|  | // |interleaved| (|samples_per_channel| * |num_channels|). | 
|  | template <typename T> | 
|  | void UpmixMonoToInterleaved(const T* mono, | 
|  | int num_frames, | 
|  | int num_channels, | 
|  | T* interleaved) { | 
|  | int interleaved_idx = 0; | 
|  | for (int i = 0; i < num_frames; ++i) { | 
|  | for (int j = 0; j < num_channels; ++j) { | 
|  | interleaved[interleaved_idx++] = mono[i]; | 
|  | } | 
|  | } | 
|  | } | 
|  |  | 
|  | template <typename T, typename Intermediate> | 
|  | void DownmixToMono(const T* const* input_channels, | 
|  | size_t num_frames, | 
|  | int num_channels, | 
|  | T* out) { | 
|  | for (size_t i = 0; i < num_frames; ++i) { | 
|  | Intermediate value = input_channels[0][i]; | 
|  | for (int j = 1; j < num_channels; ++j) { | 
|  | value += input_channels[j][i]; | 
|  | } | 
|  | out[i] = value / num_channels; | 
|  | } | 
|  | } | 
|  |  | 
|  | // Downmixes an interleaved multichannel signal to a single channel by averaging | 
|  | // all channels. | 
|  | template <typename T, typename Intermediate> | 
|  | void DownmixInterleavedToMonoImpl(const T* interleaved, | 
|  | size_t num_frames, | 
|  | int num_channels, | 
|  | T* deinterleaved) { | 
|  | RTC_DCHECK_GT(num_channels, 0); | 
|  | RTC_DCHECK_GT(num_frames, 0u); | 
|  |  | 
|  | const T* const end = interleaved + num_frames * num_channels; | 
|  |  | 
|  | while (interleaved < end) { | 
|  | const T* const frame_end = interleaved + num_channels; | 
|  |  | 
|  | Intermediate value = *interleaved++; | 
|  | while (interleaved < frame_end) { | 
|  | value += *interleaved++; | 
|  | } | 
|  |  | 
|  | *deinterleaved++ = value / num_channels; | 
|  | } | 
|  | } | 
|  |  | 
|  | template <typename T> | 
|  | void DownmixInterleavedToMono(const T* interleaved, | 
|  | size_t num_frames, | 
|  | int num_channels, | 
|  | T* deinterleaved); | 
|  |  | 
|  | template <> | 
|  | void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved, | 
|  | size_t num_frames, | 
|  | int num_channels, | 
|  | int16_t* deinterleaved); | 
|  |  | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_ |