gecko/content/media/webaudio/WebAudioUtils.cpp
Karl Tomlinson df408748ac b=991533 move sample rate limits to WebAudioUtils r=padenot
Also change WebAudioUtils from a class to a namespace, so that constant
variables can be defined inline with internal linkage.

static class variables cannot be defined inline because this violates the one
definition rule, even though some compilers may not notice.

--HG--
extra : transplant_source : %9F4%2Ct%BA%D2%BD%8A1Xev%92%C0%A1%AD%88IH%BF
2014-05-16 08:44:17 +12:00

111 lines
3.9 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 {
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
}
}
}