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