зеркало из https://github.com/mozilla/pjs.git
194 строки
6.3 KiB
C++
194 строки
6.3 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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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 file,
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* You can obtain one at http://mozilla.org/MPL/2.0/. */
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#include "AudioSegment.h"
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namespace mozilla {
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static PRUint16
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FlipByteOrderIfBigEndian(PRUint16 aValue)
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{
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PRUint16 s = aValue;
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#if defined(IS_BIG_ENDIAN)
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s = (s << 8) | (s >> 8);
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#endif
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return s;
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}
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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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static float
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SampleToFloat(float aValue)
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{
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return aValue;
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}
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static float
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SampleToFloat(PRUint8 aValue)
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{
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return (aValue - 128)/128.0f;
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}
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static float
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SampleToFloat(PRInt16 aValue)
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{
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return PRInt16(FlipByteOrderIfBigEndian(aValue))/32768.0f;
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}
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static void
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FloatToSample(float aValue, float* aOut)
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{
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*aOut = aValue;
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}
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static void
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FloatToSample(float aValue, PRUint8* aOut)
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{
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float v = aValue*128 + 128;
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float clamped = NS_MAX(0.0f, NS_MIN(255.0f, v));
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*aOut = PRUint8(clamped);
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}
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static void
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FloatToSample(float aValue, PRInt16* aOut)
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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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*aOut = PRInt16(FlipByteOrderIfBigEndian(PRInt16(clamped)));
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}
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template <class SrcT, class DestT>
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static void
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InterleaveAndConvertBuffer(const SrcT* aSource, PRInt32 aSourceLength,
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PRInt32 aLength,
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float aVolume,
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PRInt32 aChannels,
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DestT* aOutput)
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{
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DestT* output = aOutput;
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for (PRInt32 i = 0; i < aLength; ++i) {
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for (PRInt32 channel = 0; channel < aChannels; ++channel) {
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float v = SampleToFloat(aSource[channel*aSourceLength + i])*aVolume;
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FloatToSample(v, output);
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++output;
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}
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}
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}
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static void
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InterleaveAndConvertBuffer(const PRInt16* aSource, PRInt32 aSourceLength,
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PRInt32 aLength,
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float aVolume,
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PRInt32 aChannels,
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PRInt16* aOutput)
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{
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PRInt16* output = aOutput;
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float v = NS_MAX(NS_MIN(aVolume, 1.0f), -1.0f);
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PRInt32 volume = PRInt32((1 << 16) * v);
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for (PRInt32 i = 0; i < aLength; ++i) {
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for (PRInt32 channel = 0; channel < aChannels; ++channel) {
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PRInt16 s = FlipByteOrderIfBigEndian(aSource[channel*aSourceLength + i]);
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*output = FlipByteOrderIfBigEndian(PRInt16((PRInt32(s) * volume) >> 16));
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++output;
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}
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}
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}
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template <class SrcT>
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static void
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InterleaveAndConvertBuffer(const SrcT* aSource, PRInt32 aSourceLength,
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PRInt32 aLength,
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float aVolume,
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PRInt32 aChannels,
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void* aOutput, nsAudioStream::SampleFormat aOutputFormat)
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{
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switch (aOutputFormat) {
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case nsAudioStream::FORMAT_FLOAT32:
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InterleaveAndConvertBuffer(aSource, aSourceLength, aLength, aVolume,
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aChannels, static_cast<float*>(aOutput));
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break;
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case nsAudioStream::FORMAT_S16_LE:
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InterleaveAndConvertBuffer(aSource, aSourceLength, aLength, aVolume,
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aChannels, static_cast<PRInt16*>(aOutput));
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break;
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case nsAudioStream::FORMAT_U8:
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InterleaveAndConvertBuffer(aSource, aSourceLength, aLength, aVolume,
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aChannels, static_cast<PRUint8*>(aOutput));
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break;
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}
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}
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static void
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InterleaveAndConvertBuffer(const void* aSource, nsAudioStream::SampleFormat aSourceFormat,
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PRInt32 aSourceLength,
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PRInt32 aOffset, PRInt32 aLength,
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float aVolume,
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PRInt32 aChannels,
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void* aOutput, nsAudioStream::SampleFormat aOutputFormat)
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{
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switch (aSourceFormat) {
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case nsAudioStream::FORMAT_FLOAT32:
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InterleaveAndConvertBuffer(static_cast<const float*>(aSource) + aOffset, aSourceLength,
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aLength,
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aVolume,
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aChannels,
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aOutput, aOutputFormat);
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break;
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case nsAudioStream::FORMAT_S16_LE:
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InterleaveAndConvertBuffer(static_cast<const PRInt16*>(aSource) + aOffset, aSourceLength,
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aLength,
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aVolume,
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aChannels,
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aOutput, aOutputFormat);
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break;
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case nsAudioStream::FORMAT_U8:
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InterleaveAndConvertBuffer(static_cast<const PRUint8*>(aSource) + aOffset, aSourceLength,
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aLength,
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aVolume,
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aChannels,
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aOutput, aOutputFormat);
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break;
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}
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}
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void
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AudioSegment::ApplyVolume(float aVolume)
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{
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for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
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ci->mVolume *= aVolume;
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}
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}
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static const int STATIC_AUDIO_BUFFER_BYTES = 50000;
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void
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AudioSegment::WriteTo(nsAudioStream* aOutput)
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{
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NS_ASSERTION(mChannels == aOutput->GetChannels(), "Wrong number of channels");
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nsAutoTArray<PRUint8,STATIC_AUDIO_BUFFER_BYTES> buf;
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PRUint32 frameSize = GetSampleSize(aOutput->GetFormat())*mChannels;
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for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
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AudioChunk& c = *ci;
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if (frameSize*c.mDuration > PR_UINT32_MAX) {
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NS_ERROR("Buffer overflow");
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return;
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}
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buf.SetLength(PRInt32(frameSize*c.mDuration));
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if (c.mBuffer) {
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InterleaveAndConvertBuffer(c.mBuffer->Data(), c.mBufferFormat, c.mBufferLength,
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c.mOffset, PRInt32(c.mDuration),
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c.mVolume,
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aOutput->GetChannels(),
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buf.Elements(), aOutput->GetFormat());
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} else {
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// Assumes that a bit pattern of zeroes == 0.0f
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memset(buf.Elements(), 0, buf.Length());
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}
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aOutput->Write(buf.Elements(), PRInt32(c.mDuration));
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}
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}
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}
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