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4dcad1fa98
Replace nsAudioStream::Format with an AUDIO_OUTPUT_FORMAT enum value so we can use it as a template parameter. Introduce AudioSampleTraits<AudioSampleFormat> to give us access to the C++ type corresponding to an enum value. Move SampleToFloat/FloatToSample to AudioSampleFormat.h. Introduce ConvertAudioSamples and ConvertAudioSamplesWithScale functions and use them from various places. Moves AudioDataValue to AudioSampleFormat.h. The name isn't great, but it'll do.
136 lines
3.7 KiB
C++
136 lines
3.7 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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/* vim:set ts=2 sw=2 sts=2 et cindent: */
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this
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* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
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#ifndef MOZILLA_AUDIOSAMPLEFORMAT_H_
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#define MOZILLA_AUDIOSAMPLEFORMAT_H_
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#include "nsAlgorithm.h"
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namespace mozilla {
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/**
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* Audio formats supported in MediaStreams and media elements.
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*
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* Only one of these is supported by nsAudioStream, and that is determined
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* at compile time (roughly, FLOAT32 on desktops, S16 on mobile). Media decoders
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* produce that format only; queued AudioData always uses that format.
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*/
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enum AudioSampleFormat
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{
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// Native-endian signed 16-bit audio samples
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AUDIO_FORMAT_S16,
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// Signed 32-bit float samples
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AUDIO_FORMAT_FLOAT32,
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// The format used for output by nsAudioStream.
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#ifdef MOZ_SAMPLE_TYPE_S16
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AUDIO_OUTPUT_FORMAT = AUDIO_FORMAT_S16
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#else
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AUDIO_OUTPUT_FORMAT = AUDIO_FORMAT_FLOAT32
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#endif
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};
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template <AudioSampleFormat Format> class AudioSampleTraits;
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template <> class AudioSampleTraits<AUDIO_FORMAT_FLOAT32> {
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public:
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typedef float Type;
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};
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template <> class AudioSampleTraits<AUDIO_FORMAT_S16> {
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public:
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typedef int16_t Type;
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};
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typedef AudioSampleTraits<AUDIO_OUTPUT_FORMAT>::Type AudioDataValue;
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// Single-sample conversion
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/*
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* Use "2^N" conversion since it's simple, fast, "bit transparent", used by
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* many other libraries and apparently behaves reasonably.
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* http://blog.bjornroche.com/2009/12/int-float-int-its-jungle-out-there.html
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* http://blog.bjornroche.com/2009/12/linearity-and-dynamic-range-in-int.html
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*/
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inline float
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AudioSampleToFloat(float aValue)
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{
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return aValue;
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}
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inline float
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AudioSampleToFloat(int16_t aValue)
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{
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return aValue/32768.0f;
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}
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template <typename T> T FloatToAudioSample(float aValue);
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template <> inline float
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FloatToAudioSample<float>(float aValue)
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{
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return aValue;
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}
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template <> inline int16_t
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FloatToAudioSample<int16_t>(float aValue)
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{
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float v = aValue*32768.0f;
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float clamped = NS_MAX(-32768.0f, NS_MIN(32767.0f, v));
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return int16_t(clamped);
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}
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// Sample buffer conversion
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template <typename From, typename To> inline void
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ConvertAudioSamples(const From* aFrom, To* aTo, int aCount)
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{
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for (int i = 0; i < aCount; ++i) {
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aTo[i] = FloatToAudioSample<To>(AudioSampleToFloat(aFrom[i]));
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}
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}
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inline void
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ConvertAudioSamples(const int16_t* aFrom, int16_t* aTo, int aCount)
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{
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memcpy(aTo, aFrom, sizeof(*aTo)*aCount);
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}
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inline void
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ConvertAudioSamples(const float* aFrom, float* aTo, int aCount)
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{
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memcpy(aTo, aFrom, sizeof(*aTo)*aCount);
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}
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// Sample buffer conversion with scale
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template <typename From, typename To> inline void
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ConvertAudioSamplesWithScale(const From* aFrom, To* aTo, int aCount, float aScale)
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{
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if (aScale == 1.0f) {
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ConvertAudioSamples(aFrom, aTo, aCount);
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return;
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}
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for (int i = 0; i < aCount; ++i) {
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aTo[i] = FloatToAudioSample<To>(AudioSampleToFloat(aFrom[i])*aScale);
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}
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}
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inline void
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ConvertAudioSamplesWithScale(const int16_t* aFrom, int16_t* aTo, int aCount, float aScale)
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{
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if (aScale == 1.0f) {
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ConvertAudioSamples(aFrom, aTo, aCount);
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return;
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}
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if (0.0f <= aScale && aScale < 1.0f) {
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int32_t scale = int32_t((1 << 16) * aScale);
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for (int i = 0; i < aCount; ++i) {
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aTo[i] = int16_t((int32_t(aFrom[i]) * scale) >> 16);
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}
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return;
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}
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for (int i = 0; i < aCount; ++i) {
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aTo[i] = FloatToAudioSample<int16_t>(AudioSampleToFloat(aFrom[i])*aScale);
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}
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}
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}
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#endif /* MOZILLA_AUDIOSAMPLEFORMAT_H_ */
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