/external/sonivox/arm-hybrid-22k/lib_src/ |
H A D | eas_fmengine.c | 195 * Assign the left and right gain values corresponding to the given pan value. 283 EAS_I32 gain; local 291 /* establish local gain variable */ 292 gain = (EAS_I32) p->gain << 16; 294 /* calculate gain increment */ 296 gainInc = ((EAS_I32) gainTarget - (EAS_I32) p->gain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS); 337 /* internal gain for modulation effects */ 338 temp = FMUL_15x15(temp, (gain >> 16)); 340 /* output gain calculatio 393 EAS_I32 gain; local [all...] |
H A D | eas_mixer.c | 137 EAS_U16 gain; local 141 /* calculate the gain multiplier */ 149 gain = 32767; 151 gain = (EAS_U16) temp; 154 gain = (EAS_U16) pEASData->masterGain; 156 gain = (EAS_U16) pEASData->masterGain; 159 /* Not using all the gain bits for now 164 gain = gain >> 5; 166 gain [all...] |
H A D | eas_fmengine.h | 56 /* LFO modulation to gain control */ 67 EAS_U16 gain; /* current internal gain */ member in struct:__anon30908 68 EAS_U16 outputGain; /* current output gain */ 78 EAS_U16 gainLeft; /* left gain multiplier */ 79 EAS_U16 gainRight; /* right gain multiplier */ 87 EAS_U16 gain[4]; /* initial operator gain value */ member in struct:__anon30910 88 EAS_U16 outputGain[4]; /* initial operator output gain value */ 89 EAS_U16 voiceGain; /* initial voice gain */ 99 EAS_U16 gain[4]; /* new operator gain value */ member in struct:__anon30911 [all...] |
H A D | ARM-E_voice_gain_gnu.s | 59 gain .req r8
label 137 LDR gain, [pWTFrame, #m_prevGain]
138 MOV gain, gain, LSL #(NUM_MIXER_GUARD_BITS + 4)
141 SUB gainIncrement, gainIncrement, gain
149 ADD gain, gain, gainIncrement @ gain step to eliminate zipper noise
150 SMULWB tmp0, gain, tmp0 @ sample * local gain
[all...] |
/external/sonivox/arm-fm-22k/lib_src/ |
H A D | eas_mixer.c | 137 EAS_U16 gain; local 141 /* calculate the gain multiplier */ 149 gain = 32767; 151 gain = (EAS_U16) temp; 154 gain = (EAS_U16) pEASData->masterGain; 156 gain = (EAS_U16) pEASData->masterGain; 159 /* Not using all the gain bits for now 164 gain = gain >> 5; 166 gain [all...] |
H A D | eas_fmengine.h | 56 /* LFO modulation to gain control */ 67 EAS_U16 gain; /* current internal gain */ member in struct:__anon30858 68 EAS_U16 outputGain; /* current output gain */ 78 EAS_U16 gainLeft; /* left gain multiplier */ 79 EAS_U16 gainRight; /* right gain multiplier */ 87 EAS_U16 gain[4]; /* initial operator gain value */ member in struct:__anon30860 88 EAS_U16 outputGain[4]; /* initial operator output gain value */ 89 EAS_U16 voiceGain; /* initial voice gain */ 99 EAS_U16 gain[4]; /* new operator gain value */ member in struct:__anon30861 [all...] |
/external/sonivox/arm-wt-22k/lib_src/ |
H A D | eas_mixer.c | 137 EAS_U16 gain; local 141 /* calculate the gain multiplier */ 149 gain = 32767; 151 gain = (EAS_U16) temp; 154 gain = (EAS_U16) pEASData->masterGain; 156 gain = (EAS_U16) pEASData->masterGain; 159 /* Not using all the gain bits for now 164 gain = gain >> 5; 166 gain [all...] |
H A D | ARM-E_voice_gain_gnu.s | 59 gain .req r8
label 137 LDR gain, [pWTFrame, #m_prevGain]
138 MOV gain, gain, LSL #(NUM_MIXER_GUARD_BITS + 4)
141 SUB gainIncrement, gainIncrement, gain
149 ADD gain, gain, gainIncrement @ gain step to eliminate zipper noise
150 SMULWB tmp0, gain, tmp0 @ sample * local gain
[all...] |
/external/chromium_org/chrome/browser/extensions/api/audio/ |
H A D | audio_service.cc | 24 int gain) OVERRIDE; 51 int gain) { 48 SetDeviceProperties(const std::string& device_id, bool muted, int volume, int gain) argument
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/external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
H A D | BiquadFilterNode.idl | 44 readonly attribute AudioParam gain; // in Decibels
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H A D | GainNode.cpp | 54 // FIXME: for some cases there is a nice optimization to avoid processing here, and let the gain change 66 if (gain()->hasSampleAccurateValues()) { 67 // Apply sample-accurate gain scaling for precise envelopes, grain windows, etc. 71 gain()->calculateSampleAccurateValues(gainValues, framesToProcess); 75 // Apply the gain with de-zippering into the output bus. 76 outputBus->copyWithGainFrom(*inputBus, &m_lastGain, gain()->value()); 81 // FIXME: this can go away when we do mixing with gain directly in summing junction of AudioNodeInput
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H A D | GainNode.h | 37 // GainNode is an AudioNode with one input and one output which applies a gain (volume) change to the audio signal. 38 // De-zippering (smoothing) is applied when the gain value is changed dynamically. 55 AudioParam* gain() { return m_gain.get(); } function in class:blink::FINAL
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/external/chromium_org/third_party/WebKit/Source/platform/audio/ |
H A D | Cone.h | 37 // Cone gain is defined according to the OpenAL specification 43 // Returns scalar gain for the given source/listener positions/orientations 44 double gain(FloatPoint3D sourcePosition, FloatPoint3D sourceOrientation, FloatPoint3D listenerPosition);
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/external/chromium_org/third_party/webrtc/common_audio/signal_processing/ |
H A D | ilbc_specific_functions.c | 70 int16_t gain, int32_t add_constant, 81 (*outPtr++) += (int16_t)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain) 87 int16_t gain, int32_t add_constant, 98 (*outPtr++) = (int16_t)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain) 69 WebRtcSpl_AddAffineVectorToVector(int16_t *out, int16_t *in, int16_t gain, int32_t add_constant, int16_t right_shifts, int vector_length) argument 86 WebRtcSpl_AffineTransformVector(int16_t *out, int16_t *in, int16_t gain, int32_t add_constant, int16_t right_shifts, int vector_length) argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | lpc_analysis.h | 38 double* gain,
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/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
H A D | lpc_analysis.h | 38 double* gain,
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/external/aac/libAACenc/src/ |
H A D | quantize.cpp | 100 input: global gain, number of lines to process, spectral data 104 static void FDKaacEnc_quantizeLines(INT gain, argument 111 FIXP_QTD quantizer = FDKaacEnc_quantTableQ[(-gain)&3]; 112 INT quantizershift = ((-gain)>>2)+1; 156 mdctSpectrum = iquaSpectrum^4/3 *2^(0.25*gain) 157 input: global gain, number of lines to process,quantized spectrum 161 static void FDKaacEnc_invQuantizeLines(INT gain, argument 171 iquantizermod = gain&3; 172 iquantizershift = gain>>2; 292 input: gain, numbe 350 FDKaacEnc_calcSfbQuantEnergyAndDist(FIXP_DBL *mdctSpectrum, SHORT *quantSpectrum, INT noOfLines, INT gain, FIXP_DBL *en, FIXP_DBL *dist) argument [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | pitch_filter_mips.c | 14 int16_t gain, 36 // Load coefficients outside the loop and sign-extend gain and sign 49 "seh %[gain32], %[gain] \n\t" 55 : [coefficient] "r" (coefficient), [gain] "r" (gain), 13 WebRtcIsacfix_PitchFilterCore(int loopNumber, int16_t gain, int index, int16_t sign, int16_t* inputState, int16_t* outputBuf2, const int16_t* coefficient, int16_t* inputBuf, int16_t* outputBuf, int* index2) argument
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/external/skia/include/effects/ |
H A D | SkMatrixConvolutionImageFilter.h | 37 @param gain A scale factor applied to each pixel after 56 SkScalar gain, 63 return SkNEW_ARGS(SkMatrixConvolutionImageFilter, (kernelSize, kernel, gain, bias, 73 SkScalar gain, 54 Create(const SkISize& kernelSize, const SkScalar* kernel, SkScalar gain, SkScalar bias, const SkIPoint& kernelOffset, TileMode tileMode, bool convolveAlpha, SkImageFilter* input = NULL, const CropRect* cropRect = NULL) argument
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/external/aac/libSBRenc/src/ |
H A D | resampler.h | 110 FIXP_DBL gain; /*! overall gain factor */ member in struct:__anon263
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/external/chromium_org/content/browser/speech/endpointer/ |
H A D | endpointer_unittest.cc | 36 float gain = 0.0; local 42 gain = 2000.0; 44 gain = 1.0; 50 samples[i] = static_cast<int16>(gain * randNum);
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/external/chromium_org/third_party/speex/libspeex/ |
H A D | ltp.c | 173 void open_loop_nbest_pitch(spx_word16_t *sw, int start, int end, int len, int *pitch, spx_word16_t *gain, int N, char *stack) argument 260 /* Search for the best pitch prediction gain */ 291 /* Compute open-loop gain if necessary */ 292 if (gain) 302 gain[j]=g; 376 spx_word16_t gain[3]; local 487 gain[0] = ADD16(32,(spx_word16_t)gain_cdbk[best_cdbk*4]); 488 gain[1] = ADD16(32,(spx_word16_t)gain_cdbk[best_cdbk*4+1]); 489 gain[2] = ADD16(32,(spx_word16_t)gain_cdbk[best_cdbk*4+2]); 490 /*printf ("%d %d %d %d\n",gain[ 676 spx_word16_t gain[3]; local [all...] |
/external/speex/libspeex/ |
H A D | ltp.c | 173 void open_loop_nbest_pitch(spx_word16_t *sw, int start, int end, int len, int *pitch, spx_word16_t *gain, int N, char *stack) argument 260 /* Search for the best pitch prediction gain */ 291 /* Compute open-loop gain if necessary */ 292 if (gain) 302 gain[j]=g; 376 spx_word16_t gain[3]; local 487 gain[0] = ADD16(32,(spx_word16_t)gain_cdbk[best_cdbk*4]); 488 gain[1] = ADD16(32,(spx_word16_t)gain_cdbk[best_cdbk*4+1]); 489 gain[2] = ADD16(32,(spx_word16_t)gain_cdbk[best_cdbk*4+2]); 490 /*printf ("%d %d %d %d\n",gain[ 676 spx_word16_t gain[3]; local [all...] |
/external/webrtc/src/common_audio/signal_processing/ |
H A D | ilbc_specific_functions.c | 89 WebRtc_Word16 gain, WebRtc_Word32 add_constant, 100 (*outPtr++) += (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain) 106 WebRtc_Word16 gain, WebRtc_Word32 add_constant, 117 (*outPtr++) = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain) 88 WebRtcSpl_AddAffineVectorToVector(WebRtc_Word16 *out, WebRtc_Word16 *in, WebRtc_Word16 gain, WebRtc_Word32 add_constant, WebRtc_Word16 right_shifts, int vector_length) argument 105 WebRtcSpl_AffineTransformVector(WebRtc_Word16 *out, WebRtc_Word16 *in, WebRtc_Word16 gain, WebRtc_Word32 add_constant, WebRtc_Word16 right_shifts, int vector_length) argument
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H A D | vector_scaling_operations.c | 95 WebRtc_Word16 gain, WebRtc_Word16 in_vector_length, 98 // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts 108 (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts); 113 WebRtc_Word16 gain, WebRtc_Word16 in_vector_length, 116 // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts 127 tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts); 94 WebRtcSpl_ScaleVector(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word16 *out_vector, WebRtc_Word16 gain, WebRtc_Word16 in_vector_length, WebRtc_Word16 right_shifts) argument 112 WebRtcSpl_ScaleVectorWithSat(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word16 *out_vector, WebRtc_Word16 gain, WebRtc_Word16 in_vector_length, WebRtc_Word16 right_shifts) argument
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