17e40641b3
This is mainly to support the native audio format in Chrome. Although this implementation just moves the float->int conversion under the hood, we will transition AudioProcessing towards supporting this format throughout. - Add a test which verifies we get identical output with the float and int interfaces. - The float and int wrappers are tasked with conversion to the AudioBuffer format. A new shared Process/Analyze method does most of the work. - Add a new field to the debug.proto to hold deinterleaved data. - Add helpers to audio_utils.cc, and start using numeric_limits. - Note that there was no performance difference between numeric_limits and a literal value when measured on Linux using gcc or clang. BUG=2894 R=aluebs@webrtc.org, bjornv@webrtc.org, henrikg@webrtc.org, tommi@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5641 4adac7df-926f-26a2-2b94-8c16560cd09d
95 lines
3.4 KiB
C++
95 lines
3.4 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
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#define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
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#include <limits>
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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typedef std::numeric_limits<int16_t> limits_int16;
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static inline int16_t RoundToInt16(float v) {
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const float kMaxRound = limits_int16::max() - 0.5f;
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const float kMinRound = limits_int16::min() + 0.5f;
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if (v > 0)
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return v >= kMaxRound ? limits_int16::max() :
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static_cast<int16_t>(v + 0.5f);
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return v <= kMinRound ? limits_int16::min() :
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static_cast<int16_t>(v - 0.5f);
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}
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// Scale (from [-1, 1]) and round to full-range int16 with clamping.
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static inline int16_t ScaleAndRoundToInt16(float v) {
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if (v > 0)
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return v >= 1 ? limits_int16::max() :
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static_cast<int16_t>(v * limits_int16::max() + 0.5f);
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return v <= -1 ? limits_int16::min() :
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static_cast<int16_t>(-v * limits_int16::min() - 0.5f);
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}
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// Scale to float [-1, 1].
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static inline float ScaleToFloat(int16_t v) {
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const float kMaxInt16Inverse = 1.f / limits_int16::max();
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const float kMinInt16Inverse = 1.f / limits_int16::min();
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return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse);
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}
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// Round |size| elements of |src| to int16 with clamping and write to |dest|.
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void RoundToInt16(const float* src, int size, int16_t* dest);
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// Scale (from [-1, 1]) and round |size| elements of |src| to full-range int16
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// with clamping and write to |dest|.
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void ScaleAndRoundToInt16(const float* src, int size, int16_t* dest);
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// Scale |size| elements of |src| to float [-1, 1] and write to |dest|.
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void ScaleToFloat(const int16_t* src, int size, float* dest);
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// Deinterleave audio from |interleaved| to the channel buffers pointed to
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// by |deinterleaved|. There must be sufficient space allocated in the
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// |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
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// per buffer).
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template <typename T>
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void Deinterleave(const T* interleaved, int samples_per_channel,
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int num_channels, T** deinterleaved) {
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for (int i = 0; i < num_channels; ++i) {
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T* channel = deinterleaved[i];
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int interleaved_idx = i;
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for (int j = 0; j < samples_per_channel; ++j) {
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channel[j] = interleaved[interleaved_idx];
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interleaved_idx += num_channels;
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}
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}
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}
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// Interleave audio from the channel buffers pointed to by |deinterleaved| to
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// |interleaved|. There must be sufficient space allocated in |interleaved|
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// (|samples_per_channel| * |num_channels|).
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template <typename T>
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void Interleave(const T* const* deinterleaved, int samples_per_channel,
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int num_channels, T* interleaved) {
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for (int i = 0; i < num_channels; ++i) {
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const T* channel = deinterleaved[i];
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int interleaved_idx = i;
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for (int j = 0; j < samples_per_channel; ++j) {
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interleaved[interleaved_idx] = channel[j];
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interleaved_idx += num_channels;
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}
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}
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}
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} // namespace webrtc
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#endif // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
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