gecko/content/media/webaudio/WebAudioUtils.cpp

115 lines
4.1 KiB
C++

/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim:set ts=2 sw=2 sts=2 et cindent: */
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this
* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
#include "WebAudioUtils.h"
#include "AudioNodeStream.h"
#include "AudioParamTimeline.h"
#include "blink/HRTFDatabaseLoader.h"
#include "speex/speex_resampler.h"
namespace mozilla {
namespace dom {
// 32 is the minimum required by the spec and matches what is used by blink.
// The limit protects against large memory allocations.
const uint32_t WebAudioUtils::MaxChannelCount = 32;
struct ConvertTimeToTickHelper
{
AudioNodeStream* mSourceStream;
AudioNodeStream* mDestinationStream;
static int64_t Convert(double aTime, void* aClosure)
{
ConvertTimeToTickHelper* This = static_cast<ConvertTimeToTickHelper*> (aClosure);
MOZ_ASSERT(This->mSourceStream->SampleRate() == This->mDestinationStream->SampleRate());
return This->mSourceStream->
TicksFromDestinationTime(This->mDestinationStream, aTime);
}
};
void
WebAudioUtils::ConvertAudioParamToTicks(AudioParamTimeline& aParam,
AudioNodeStream* aSource,
AudioNodeStream* aDest)
{
MOZ_ASSERT(!aSource || aSource->SampleRate() == aDest->SampleRate());
ConvertTimeToTickHelper ctth;
ctth.mSourceStream = aSource;
ctth.mDestinationStream = aDest;
aParam.ConvertEventTimesToTicks(ConvertTimeToTickHelper::Convert, &ctth, aDest->SampleRate());
}
void
WebAudioUtils::Shutdown()
{
WebCore::HRTFDatabaseLoader::shutdown();
}
int
WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
uint32_t aChannel,
const float* aIn, uint32_t* aInLen,
float* aOut, uint32_t* aOutLen)
{
#ifdef MOZ_SAMPLE_TYPE_S16
nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp1;
nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp2;
tmp1.SetLength(*aInLen);
tmp2.SetLength(*aOutLen);
ConvertAudioSamples(aIn, tmp1.Elements(), *aInLen);
int result = speex_resampler_process_int(aResampler, aChannel, tmp1.Elements(), aInLen, tmp2.Elements(), aOutLen);
ConvertAudioSamples(tmp2.Elements(), aOut, *aOutLen);
return result;
#else
return speex_resampler_process_float(aResampler, aChannel, aIn, aInLen, aOut, aOutLen);
#endif
}
int
WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
uint32_t aChannel,
const int16_t* aIn, uint32_t* aInLen,
float* aOut, uint32_t* aOutLen)
{
nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp;
#ifdef MOZ_SAMPLE_TYPE_S16
tmp.SetLength(*aOutLen);
int result = speex_resampler_process_int(aResampler, aChannel, aIn, aInLen, tmp.Elements(), aOutLen);
ConvertAudioSamples(tmp.Elements(), aOut, *aOutLen);
return result;
#else
tmp.SetLength(*aInLen);
ConvertAudioSamples(aIn, tmp.Elements(), *aInLen);
int result = speex_resampler_process_float(aResampler, aChannel, tmp.Elements(), aInLen, aOut, aOutLen);
return result;
#endif
}
int
WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
uint32_t aChannel,
const int16_t* aIn, uint32_t* aInLen,
int16_t* aOut, uint32_t* aOutLen)
{
#ifdef MOZ_SAMPLE_TYPE_S16
return speex_resampler_process_int(aResampler, aChannel, aIn, aInLen, aOut, aOutLen);
#else
nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp1;
nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp2;
tmp1.SetLength(*aInLen);
tmp2.SetLength(*aOutLen);
ConvertAudioSamples(aIn, tmp1.Elements(), *aInLen);
int result = speex_resampler_process_float(aResampler, aChannel, tmp1.Elements(), aInLen, tmp2.Elements(), aOutLen);
ConvertAudioSamples(tmp2.Elements(), aOut, *aOutLen);
return result;
#endif
}
}
}