mirror of
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124 lines
4.0 KiB
C++
124 lines
4.0 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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#include "AudioStream.h"
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namespace mozilla {
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template <class SrcT, class DestT>
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static void
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InterleaveAndConvertBuffer(const SrcT* aSource, int32_t aSourceLength,
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int32_t aLength,
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float aVolume,
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int32_t aChannels,
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DestT* aOutput)
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{
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DestT* output = aOutput;
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for (int32_t i = 0; i < aLength; ++i) {
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for (int32_t channel = 0; channel < aChannels; ++channel) {
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float v = AudioSampleToFloat(aSource[channel*aSourceLength + i])*aVolume;
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*output = FloatToAudioSample<DestT>(v);
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++output;
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}
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}
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}
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static inline void
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InterleaveAndConvertBuffer(const int16_t* aSource, int32_t aSourceLength,
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int32_t aLength,
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float aVolume,
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int32_t aChannels,
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int16_t* aOutput)
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{
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int16_t* output = aOutput;
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if (0.0f <= aVolume && aVolume <= 1.0f) {
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int32_t scale = int32_t((1 << 16) * aVolume);
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for (int32_t i = 0; i < aLength; ++i) {
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for (int32_t channel = 0; channel < aChannels; ++channel) {
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int16_t s = aSource[channel*aSourceLength + i];
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*output = int16_t((int32_t(s) * scale) >> 16);
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++output;
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}
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}
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return;
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}
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for (int32_t i = 0; i < aLength; ++i) {
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for (int32_t channel = 0; channel < aChannels; ++channel) {
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float v = AudioSampleToFloat(aSource[channel*aSourceLength + i])*aVolume;
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*output = FloatToAudioSample<int16_t>(v);
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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 void* aSource, AudioSampleFormat aSourceFormat,
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int32_t aSourceLength,
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int32_t aOffset, int32_t aLength,
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float aVolume,
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int32_t aChannels,
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AudioDataValue* aOutput)
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{
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switch (aSourceFormat) {
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case AUDIO_FORMAT_FLOAT32:
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InterleaveAndConvertBuffer(static_cast<const float*>(aSource) + aOffset,
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aSourceLength,
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aLength,
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aVolume,
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aChannels,
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aOutput);
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break;
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case AUDIO_FORMAT_S16:
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InterleaveAndConvertBuffer(static_cast<const int16_t*>(aSource) + aOffset,
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aSourceLength,
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aLength,
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aVolume,
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aChannels,
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aOutput);
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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_SAMPLES = 10000;
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void
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AudioSegment::WriteTo(AudioStream* aOutput)
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{
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NS_ASSERTION(mChannels == aOutput->GetChannels(), "Wrong number of channels");
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nsAutoTArray<AudioDataValue,STATIC_AUDIO_SAMPLES> buf;
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for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
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AudioChunk& c = *ci;
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if (uint64_t(mChannels)*c.mDuration > INT32_MAX) {
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NS_ERROR("Buffer overflow");
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return;
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}
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buf.SetLength(int32_t(mChannels*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, int32_t(c.mDuration),
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c.mVolume,
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aOutput->GetChannels(),
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buf.Elements());
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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()*sizeof(AudioDataValue));
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}
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aOutput->Write(buf.Elements(), int32_t(c.mDuration));
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}
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}
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}
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