content/media/webaudio/WebAudioUtils.cpp

Tue, 06 Jan 2015 21:39:09 +0100

author
Michael Schloh von Bennewitz <michael@schloh.com>
date
Tue, 06 Jan 2015 21:39:09 +0100
branch
TOR_BUG_9701
changeset 8
97036ab72558
permissions
-rw-r--r--

Conditionally force memory storage according to privacy.thirdparty.isolate;
This solves Tor bug #9701, complying with disk avoidance documented in
https://www.torproject.org/projects/torbrowser/design/#disk-avoidance.

michael@0 1 /* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
michael@0 2 /* vim:set ts=2 sw=2 sts=2 et cindent: */
michael@0 3 /* This Source Code Form is subject to the terms of the Mozilla Public
michael@0 4 * License, v. 2.0. If a copy of the MPL was not distributed with this
michael@0 5 * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
michael@0 6
michael@0 7 #include "WebAudioUtils.h"
michael@0 8 #include "AudioNodeStream.h"
michael@0 9 #include "AudioParamTimeline.h"
michael@0 10 #include "blink/HRTFDatabaseLoader.h"
michael@0 11 #include "speex/speex_resampler.h"
michael@0 12
michael@0 13 namespace mozilla {
michael@0 14
michael@0 15 namespace dom {
michael@0 16
michael@0 17 struct ConvertTimeToTickHelper
michael@0 18 {
michael@0 19 AudioNodeStream* mSourceStream;
michael@0 20 AudioNodeStream* mDestinationStream;
michael@0 21
michael@0 22 static int64_t Convert(double aTime, void* aClosure)
michael@0 23 {
michael@0 24 ConvertTimeToTickHelper* This = static_cast<ConvertTimeToTickHelper*> (aClosure);
michael@0 25 MOZ_ASSERT(This->mSourceStream->SampleRate() == This->mDestinationStream->SampleRate());
michael@0 26 return This->mSourceStream->
michael@0 27 TicksFromDestinationTime(This->mDestinationStream, aTime);
michael@0 28 }
michael@0 29 };
michael@0 30
michael@0 31 void
michael@0 32 WebAudioUtils::ConvertAudioParamToTicks(AudioParamTimeline& aParam,
michael@0 33 AudioNodeStream* aSource,
michael@0 34 AudioNodeStream* aDest)
michael@0 35 {
michael@0 36 MOZ_ASSERT(!aSource || aSource->SampleRate() == aDest->SampleRate());
michael@0 37 ConvertTimeToTickHelper ctth;
michael@0 38 ctth.mSourceStream = aSource;
michael@0 39 ctth.mDestinationStream = aDest;
michael@0 40 aParam.ConvertEventTimesToTicks(ConvertTimeToTickHelper::Convert, &ctth, aDest->SampleRate());
michael@0 41 }
michael@0 42
michael@0 43 void
michael@0 44 WebAudioUtils::Shutdown()
michael@0 45 {
michael@0 46 WebCore::HRTFDatabaseLoader::shutdown();
michael@0 47 }
michael@0 48
michael@0 49 int
michael@0 50 WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
michael@0 51 uint32_t aChannel,
michael@0 52 const float* aIn, uint32_t* aInLen,
michael@0 53 float* aOut, uint32_t* aOutLen)
michael@0 54 {
michael@0 55 #ifdef MOZ_SAMPLE_TYPE_S16
michael@0 56 nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp1;
michael@0 57 nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp2;
michael@0 58 tmp1.SetLength(*aInLen);
michael@0 59 tmp2.SetLength(*aOutLen);
michael@0 60 ConvertAudioSamples(aIn, tmp1.Elements(), *aInLen);
michael@0 61 int result = speex_resampler_process_int(aResampler, aChannel, tmp1.Elements(), aInLen, tmp2.Elements(), aOutLen);
michael@0 62 ConvertAudioSamples(tmp2.Elements(), aOut, *aOutLen);
michael@0 63 return result;
michael@0 64 #else
michael@0 65 return speex_resampler_process_float(aResampler, aChannel, aIn, aInLen, aOut, aOutLen);
michael@0 66 #endif
michael@0 67 }
michael@0 68
michael@0 69 int
michael@0 70 WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
michael@0 71 uint32_t aChannel,
michael@0 72 const int16_t* aIn, uint32_t* aInLen,
michael@0 73 float* aOut, uint32_t* aOutLen)
michael@0 74 {
michael@0 75 nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp;
michael@0 76 #ifdef MOZ_SAMPLE_TYPE_S16
michael@0 77 tmp.SetLength(*aOutLen);
michael@0 78 int result = speex_resampler_process_int(aResampler, aChannel, aIn, aInLen, tmp.Elements(), aOutLen);
michael@0 79 ConvertAudioSamples(tmp.Elements(), aOut, *aOutLen);
michael@0 80 return result;
michael@0 81 #else
michael@0 82 tmp.SetLength(*aInLen);
michael@0 83 ConvertAudioSamples(aIn, tmp.Elements(), *aInLen);
michael@0 84 int result = speex_resampler_process_float(aResampler, aChannel, tmp.Elements(), aInLen, aOut, aOutLen);
michael@0 85 return result;
michael@0 86 #endif
michael@0 87 }
michael@0 88
michael@0 89 int
michael@0 90 WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
michael@0 91 uint32_t aChannel,
michael@0 92 const int16_t* aIn, uint32_t* aInLen,
michael@0 93 int16_t* aOut, uint32_t* aOutLen)
michael@0 94 {
michael@0 95 #ifdef MOZ_SAMPLE_TYPE_S16
michael@0 96 return speex_resampler_process_int(aResampler, aChannel, aIn, aInLen, aOut, aOutLen);
michael@0 97 #else
michael@0 98 nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp1;
michael@0 99 nsAutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp2;
michael@0 100 tmp1.SetLength(*aInLen);
michael@0 101 tmp2.SetLength(*aOutLen);
michael@0 102 ConvertAudioSamples(aIn, tmp1.Elements(), *aInLen);
michael@0 103 int result = speex_resampler_process_float(aResampler, aChannel, tmp1.Elements(), aInLen, tmp2.Elements(), aOutLen);
michael@0 104 ConvertAudioSamples(tmp2.Elements(), aOut, *aOutLen);
michael@0 105 return result;
michael@0 106 #endif
michael@0 107 }
michael@0 108
michael@0 109 }
michael@0 110 }

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