/external/drrickorang/LoopbackApp/app/src/main/java/org/drrickorang/loopback/ |
D | GlitchesStringBuilder.java | 39 int[] glitchesData, int samplingRate, in getGlitchString() argument 44 double newSamplesInMs = ((double) newSamplesPerFFT / samplingRate) * in getGlitchString() 49 double allSamplesInMs = ((double) fftsamplingsize / samplingRate) * in getGlitchString() 71 int[] glitchesData, int samplingRate) { in getGlitchStringForFile() argument 75 double newSamplesInMs = ((double) newSamplesPerFFT / samplingRate) * in getGlitchStringForFile() 92 int[] glitchesData, int samplingRate) { in getGlitchMilliseconds() argument 97 double newSamplesInMs = ((double) newSamplesPerFFT / samplingRate) * in getGlitchMilliseconds()
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D | TestSettings.java | 22 public TestSettings(int samplingRate, int playerBufferSizeInBytes, in TestSettings() argument 24 mSamplingRate = samplingRate; in TestSettings() 41 public void setSamplingRate(int samplingRate) { in setSamplingRate() argument 42 mSamplingRate = Utilities.clamp(samplingRate, in setSamplingRate()
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D | LoopbackAudioThread.java | 84 int samplingRate = AudioTrack.getNativeOutputSampleRate(AudioManager.STREAM_MUSIC); in computeDefaultSettings() local 85 int minPlayerBufferSizeInBytes = AudioTrack.getMinBufferSize(samplingRate, in computeDefaultSettings() 87 int minRecorderBufferSizeInBytes = AudioRecord.getMinBufferSize(samplingRate, in computeDefaultSettings() 89 return new TestSettings(samplingRate, minPlayerBufferSizeInBytes, in computeDefaultSettings() 93 public LoopbackAudioThread(int samplingRate, int playerBufferInBytes, int recorderBufferInBytes, in LoopbackAudioThread() argument 99 mSamplingRate = samplingRate; in LoopbackAudioThread()
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D | ToneGeneration.java | 30 public ToneGeneration(int samplingRate) { in ToneGeneration() argument 31 mSamplingRate = samplingRate; in ToneGeneration()
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D | SoundLevelCalibration.java | 55 SoundLevelCalibration(int threadType, int samplingRate, int playerBufferSizeInBytes, in SoundLevelCalibration() argument 60 samplingRate); in SoundLevelCalibration() 62 mNativeAudioThread = new NativeAudioThread(threadType, samplingRate, in SoundLevelCalibration()
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D | Correlation.java | 52 public void init(int blockSize, int samplingRate) { in init() argument 54 mSamplingRate = samplingRate; in init() 57 public void computeCorrelation(double [] data, int samplingRate) { in computeCorrelation() argument 59 mSamplingRate = samplingRate; in computeCorrelation()
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D | RampedSineTone.java | 26 public RampedSineTone(int samplingRate, double frequency) { in RampedSineTone() argument 27 super(samplingRate, frequency); in RampedSineTone()
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D | SineWaveTone.java | 33 public SineWaveTone(int samplingRate, double frequency) { in SineWaveTone() argument 34 super(samplingRate); in SineWaveTone()
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D | NativeAudioThread.java | 106 int samplingRate = AudioTrack.getNativeOutputSampleRate(AudioManager.STREAM_MUSIC); in computeDefaultSettings() local 114 return new TestSettings(samplingRate, minBufferSizeInBytes, minBufferSizeInBytes); in computeDefaultSettings() 117 public NativeAudioThread(int threadType, int samplingRate, int playerBufferInBytes, in NativeAudioThread() argument 123 mSamplingRate = samplingRate; in NativeAudioThread() 167 int samplingRate, int frameCount, int micSource, in nativeInit() argument
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D | SettingsActivity.java | 100 int samplingRate = getApp().getSamplingRate(); in onCreate() local 110 String currentValue = String.valueOf(samplingRate); in onCreate() 331 int samplingRate = getApp().getSamplingRate(); in refresh() local 332 String currentValue = String.valueOf(samplingRate); in refresh() 361 int samplingRate = Integer.parseInt(stringValue); in onItemSelected() local 362 getApp().setSamplingRate(samplingRate); in onItemSelected()
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D | AudioFileOutput.java | 44 public AudioFileOutput(Context context, Uri uri, int samplingRate) { in AudioFileOutput() argument 47 mSamplingRate = samplingRate; in AudioFileOutput()
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/external/v8/tools/ |
D | profile_view.js | 35 function ViewBuilder(samplingRate) { argument 36 this.samplingRate = samplingRate; 50 var samplingRate = this.samplingRate; 53 var totalWeight = node.totalWeight * samplingRate; 54 var selfWeight = node.selfWeight * samplingRate;
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/external/aac/libAACdec/src/ |
D | channelinfo.cpp | 227 UINT samplingRate) { in getSamplingRateInfo() argument 235 UINT i, samplingRateSearch = samplingRate; in getSamplingRateInfo() 238 samplingRateSearch = (samplingRate * 4) / 3; in getSamplingRateInfo() 251 t->samplingRate = samplingRate; in getSamplingRateInfo() 285 t->samplingRate = 0; in getSamplingRateInfo()
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/external/drrickorang/LoopbackApp/app/src/main/cpp/lb2/ |
D | loopback2.cpp | 36 int lb2ComputeDefaultSettings(int performanceMode, int *samplingRate, in lb2ComputeDefaultSettings() argument 40 samplingRate, playerBufferFrameCount, recorderBufferFrameCount) ? in lb2ComputeDefaultSettings() 44 int lb2Init(void **ppLbData, int samplingRate, int frameCount, int /*micSource*/, in lb2Init() argument 54 CHANNEL_COUNT, samplingRate, ignoreFirstFrames, loopbackTone)); in lb2Init() 64 CHANNEL_COUNT, samplingRate, frequency1, std::move(byteBuffer))); in lb2Init()
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D | sound_system_aaudio.cpp | 213 PerformanceMode performanceMode, int *samplingRate, int *playerBufferFrameCount); 258 PerformanceMode performanceMode, int *samplingRate, int *playerBufferFrameCount) { in probeDefaults() argument 263 *samplingRate = mStream->getSamplingRateHz(); in probeDefaults() 265 mStream->getFramesPerBurst(), MINIMUM_STREAM_BUFFER_SIZE_MS, *samplingRate); in probeDefaults() 328 PerformanceMode performanceMode, int *samplingRate, int *recorderBufferFrameCount); 361 PerformanceMode performanceMode, int *samplingRate, int *recorderBufferFrameCount) { in probeDefaults() argument 366 *samplingRate = mStream->getSamplingRateHz(); in probeDefaults() 368 mStream->getFramesPerBurst(), MINIMUM_STREAM_BUFFER_SIZE_MS, *samplingRate); in probeDefaults() 450 bool SoundSystemAAudio::probeDefaultSettings(PerformanceMode performanceMode, int *samplingRate, in probeDefaultSettings() argument 454 performanceMode, samplingRate, recorderBufferFrameCount) in probeDefaultSettings() [all …]
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D | loopback2.h | 26 int lb2ComputeDefaultSettings(int performanceMode, int *samplingRate, 28 int lb2Init(void ** ppCtx, int samplingRate, int frameCount, int micSource,
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/external/libchrome/base/sampling_heap_profiler/ |
D | benchmark-octane.js | 11 async function runOctane(samplingRate) { argument 13 if (samplingRate) 14 args.push(`--sampling-heap-profiler=${samplingRate}`);
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/external/drrickorang/LoopbackApp/app/src/main/cpp/ |
D | jni_native.c | 42 int samplingRate, playerBufferFrameCount, recorderBufferFrameCount; in Java_org_drrickorang_loopback_NativeAudioThread_nativeComputeDefaultSettings() local 43 if (sEngines[engine].computeDefaultSettings(performanceMode, &samplingRate, in Java_org_drrickorang_loopback_NativeAudioThread_nativeComputeDefaultSettings() 48 samplingRate, in Java_org_drrickorang_loopback_NativeAudioThread_nativeComputeDefaultSettings() 58 (JNIEnv *env, jobject obj __unused, jint threadType, jint samplingRate, jint frameCount, in Java_org_drrickorang_loopback_NativeAudioThread_nativeInit() argument 80 if (sEngines[engine].init(&pContext, samplingRate, frameCount, micSource, in Java_org_drrickorang_loopback_NativeAudioThread_nativeInit()
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D | loopback_sles.h | 26 int slesComputeDefaultSettings(int performanceMode, int *samplingRate, 28 int slesInit(void ** ppCtx, int samplingRate, int frameCount, int micSource,
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D | loopback.h | 57 int (*computeDefaultSettings)(int performanceMode, int *samplingRate, 59 int (*init)(void **ppCtx, int samplingRate, int frameCount, int micSource,
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/external/aac/libAACenc/src/ |
D | aacenc.cpp | 123 const INT samplingRate) { in FDKaacEnc_CalcBitsPerFrame() argument 126 (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate) { in FDKaacEnc_CalcBitsPerFrame() 130 return (bitRate * (frameLength >> shift)) / (samplingRate >> shift); in FDKaacEnc_CalcBitsPerFrame() 134 const INT samplingRate) { in FDKaacEnc_CalcBitrate() argument 137 (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate) { in FDKaacEnc_CalcBitrate() 141 return (bitsPerFrame * (samplingRate >> shift)) / (frameLength >> shift); in FDKaacEnc_CalcBitrate()
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D | aacenc.h | 278 const INT samplingRate); 291 const INT samplingRate);
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/external/aac/libMpegTPEnc/include/ |
D | tp_data.h | 370 const INT samplingRate, const INT frameSize, 419 static inline int getSamplingRateIndex(UINT samplingRate, UINT nBits) { in getSamplingRateIndex() argument 424 if (SamplingRateTable[sf_index] == samplingRate) break; in getSamplingRateIndex()
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/external/aac/libMpegTPDec/include/ |
D | tp_data.h | 370 const INT samplingRate, const INT frameSize, 419 static inline int getSamplingRateIndex(UINT samplingRate, UINT nBits) { in getSamplingRateIndex() argument 424 if (SamplingRateTable[sf_index] == samplingRate) break; in getSamplingRateIndex()
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/external/robolectric-shadows/shadows/framework/src/main/java/org/robolectric/shadows/ |
D | ShadowMediaRecorder.java | 72 protected void setAudioSamplingRate(int samplingRate) { in setAudioSamplingRate() argument 73 audioSamplingRate = samplingRate; in setAudioSamplingRate()
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