зеркало из https://github.com/mozilla/gecko-dev.git
110 строки
3.9 KiB
C++
110 строки
3.9 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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#include "WebAudioUtils.h"
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#include "AudioNodeStream.h"
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#include "AudioParamTimeline.h"
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#include "blink/HRTFDatabaseLoader.h"
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namespace mozilla {
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namespace dom {
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struct ConvertTimeToTickHelper
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{
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AudioNodeStream* mSourceStream;
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AudioNodeStream* mDestinationStream;
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static int64_t Convert(double aTime, void* aClosure)
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{
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ConvertTimeToTickHelper* This = static_cast<ConvertTimeToTickHelper*> (aClosure);
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MOZ_ASSERT(This->mSourceStream->SampleRate() == This->mDestinationStream->SampleRate());
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return This->mSourceStream->
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TicksFromDestinationTime(This->mDestinationStream, aTime);
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}
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};
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void
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WebAudioUtils::ConvertAudioParamToTicks(AudioParamTimeline& aParam,
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AudioNodeStream* aSource,
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AudioNodeStream* aDest)
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{
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MOZ_ASSERT(!aSource || aSource->SampleRate() == aDest->SampleRate());
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ConvertTimeToTickHelper ctth;
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ctth.mSourceStream = aSource;
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ctth.mDestinationStream = aDest;
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aParam.ConvertEventTimesToTicks(ConvertTimeToTickHelper::Convert, &ctth, aDest->SampleRate());
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}
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void
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WebAudioUtils::Shutdown()
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{
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WebCore::HRTFDatabaseLoader::shutdown();
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}
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int
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WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
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uint32_t aChannel,
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const float* aIn, uint32_t* aInLen,
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float* aOut, uint32_t* aOutLen)
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{
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#ifdef MOZ_SAMPLE_TYPE_S16
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nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp1;
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nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp2;
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tmp1.SetLength(*aInLen);
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tmp2.SetLength(*aOutLen);
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ConvertAudioSamples(aIn, tmp1.Elements(), *aInLen);
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int result = speex_resampler_process_int(aResampler, aChannel, tmp1.Elements(), aInLen, tmp2.Elements(), aOutLen);
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ConvertAudioSamples(tmp2.Elements(), aOut, *aOutLen);
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return result;
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#else
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return speex_resampler_process_float(aResampler, aChannel, aIn, aInLen, aOut, aOutLen);
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#endif
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}
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int
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WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
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uint32_t aChannel,
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const int16_t* aIn, uint32_t* aInLen,
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float* aOut, uint32_t* aOutLen)
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{
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nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp;
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#ifdef MOZ_SAMPLE_TYPE_S16
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tmp.SetLength(*aOutLen);
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int result = speex_resampler_process_int(aResampler, aChannel, aIn, aInLen, tmp.Elements(), aOutLen);
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ConvertAudioSamples(tmp.Elements(), aOut, *aOutLen);
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return result;
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#else
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tmp.SetLength(*aInLen);
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ConvertAudioSamples(aIn, tmp.Elements(), *aInLen);
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int result = speex_resampler_process_float(aResampler, aChannel, tmp.Elements(), aInLen, aOut, aOutLen);
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return result;
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#endif
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}
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int
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WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
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uint32_t aChannel,
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const int16_t* aIn, uint32_t* aInLen,
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int16_t* aOut, uint32_t* aOutLen)
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{
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#ifdef MOZ_SAMPLE_TYPE_S16
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return speex_resampler_process_int(aResampler, aChannel, aIn, aInLen, aOut, aOutLen);
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#else
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nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp1;
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nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp2;
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tmp1.SetLength(*aInLen);
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tmp2.SetLength(*aOutLen);
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ConvertAudioSamples(aIn, tmp1.Elements(), *aInLen);
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int result = speex_resampler_process_float(aResampler, aChannel, tmp1.Elements(), aInLen, tmp2.Elements(), aOutLen);
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ConvertAudioSamples(tmp2.Elements(), aOut, *aOutLen);
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return result;
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#endif
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}
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}
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}
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