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 /* * 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 COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_ #define COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_ #include #include #include #include #include #include "rtc_base/checks.h" namespace webrtc { typedef std::numeric_limits 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] // Dbfs: float [-20.0*log(10, 32768), 0] = [-90.3, 0] // The ratio conversion functions use this naming convention: // Ratio: float (0, +inf) // Db: float (-inf, +inf) static inline int16_t FloatToS16(float v) { if (v > 0) return v >= 1 ? limits_int16::max() : static_cast(v * limits_int16::max() + 0.5f); return v <= -1 ? limits_int16::min() : static_cast(-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(v + 0.5f); return v <= kMinRound ? limits_int16::min() : static_cast(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); inline float DbToRatio(float v) { return std::pow(10.0f, v / 20.0f); } inline float DbfsToFloatS16(float v) { static constexpr float kMaximumAbsFloatS16 = -limits_int16::min(); return DbToRatio(v) * kMaximumAbsFloatS16; } inline float FloatS16ToDbfs(float v) { RTC_DCHECK_GE(v, 0); // kMinDbfs is equal to -20.0 * log10(-limits_int16::min()) static constexpr float kMinDbfs = -90.30899869919436f; if (v <= 1.0f) { return kMinDbfs; } // Equal to 20 * log10(v / (-limits_int16::min())) return 20.0f * std::log10(v) + kMinDbfs; } // 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 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 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 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 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 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 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, 0); 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 void DownmixInterleavedToMono(const T* interleaved, size_t num_frames, int num_channels, T* deinterleaved); template <> void DownmixInterleavedToMono(const int16_t* interleaved, size_t num_frames, int num_channels, int16_t* deinterleaved); } // namespace webrtc #endif // COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_