/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/ilbc/ |
H A D | hp_input.c | 26 int16_t *signal, /* (i/o) signal vector */ 27 int16_t *ba, /* (i) B- and A-coefficients (2:nd order) 30 int16_t *y, /* (i/o) Filter state yhi[n-1] ylow[n-1] 32 int16_t *x, /* (i/o) Filter state x[n-1] x[n-2] */ 33 int16_t len) /* (i) Number of samples to filter */ 68 signal[i] = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmpW32b, 13); 83 y[0] = (int16_t)(tmpW32 >> 16); 84 y[1] = (int16_t)((tmpW32 - WEBRTC_SPL_LSHIFT_W32((int32_t)y[0], 16))>>1);
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H A D | hp_output.c | 26 int16_t *signal, /* (i/o) signal vector */ 27 int16_t *ba, /* (i) B- and A-coefficients (2:nd order) 30 int16_t *y, /* (i/o) Filter state yhi[n-1] ylow[n-1] 32 int16_t *x, /* (i/o) Filter state x[n-1] x[n-2] */ 33 int16_t len) /* (i) Number of samples to filter */ 68 signal[i] = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmpW32b, 11); 83 y[0] = (int16_t)(tmpW32 >> 16); 84 y[1] = (int16_t)((tmpW32 - WEBRTC_SPL_LSHIFT_W32((int32_t)y[0], 16))>>1);
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H A D | simple_lpc_analysis.c | 30 int16_t *lsf, /* (o) lsf coefficients */ 31 int16_t *data, /* (i) new block of speech */ 37 int16_t is; 38 int16_t stability; 40 int16_t A[LPC_FILTERORDER + 1]; 42 int16_t windowedData[BLOCKL_MAX]; 43 int16_t rc[LPC_FILTERORDER]; 83 WebRtcIlbcfix_BwExpand(A, A, (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, LPC_FILTERORDER+1);
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H A D | get_lsp_poly.c | 41 int16_t *lsp, /* (i) LSP in Q15 */ 46 int16_t high, low; 47 int16_t *lspPtr; 67 high = (int16_t)WEBRTC_SPL_RSHIFT_W32(fPtr[-1], 16); 68 low = (int16_t)WEBRTC_SPL_RSHIFT_W32(fPtr[-1]-WEBRTC_SPL_LSHIFT_W32(((int32_t)high),16), 1);
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H A D | lsf_check.c | 27 int16_t *lsf, /* LSF parameters */ 32 const int16_t eps=319; /* 0.039 in Q13 (50 Hz)*/ 33 const int16_t eps2=160; /* eps/2.0 in Q13;*/ 34 const int16_t maxlsf=25723; /* 3.14; (4000 Hz)*/ 35 const int16_t minlsf=82; /* 0.01; (0 Hz)*/
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | time_stretch.h | 59 ReturnCodes Process(const int16_t* input, 62 int16_t* length_change_samples); 69 int16_t* best_correlation, 76 const int16_t* input, size_t input_length, size_t peak_index, 77 int16_t best_correlation, bool active_speech, 92 int16_t max_input_value_; 93 int16_t downsampled_input_[kDownsampledLen]; 96 int16_t auto_correlation_[kCorrelationLen + 1];
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H A D | accelerate.h | 44 ReturnCodes Process(const int16_t* input, 47 int16_t* length_change_samples); 53 int16_t* best_correlation, 59 const int16_t* input, size_t input_length, size_t peak_index, 60 int16_t best_correlation, bool active_speech,
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H A D | preemptive_expand.h | 48 ReturnCodes Process(const int16_t *pw16_decoded, 52 int16_t* length_change_samples); 58 int16_t* w16_bestCorr, 64 const int16_t *pw16_decoded, size_t len, size_t w16_bestIndex, 65 int16_t w16_bestCorr, bool w16_VAD,
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H A D | audio_vector.h | 26 : array_(new int16_t[kDefaultInitialSize]), 32 : array_(new int16_t[initial_size]), 35 memset(array_.get(), 0, initial_size * sizeof(int16_t)); 54 virtual void PushFront(const int16_t* prepend_this, size_t length); 60 virtual void PushBack(const int16_t* append_this, size_t length); 76 virtual void InsertAt(const int16_t* insert_this, size_t length, 87 virtual void OverwriteAt(const int16_t* insert_this, 103 const int16_t& operator[](size_t index) const; 104 int16_t& operator[](size_t index); 111 scoped_ptr<int16_t[]> array [all...] |
/external/compiler-rt/test/ubsan/TestCases/Integer/ |
H A D | add-overflow.cpp | 11 (void)(int16_t(0x3fff) + int16_t(0x4000));
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H A D | sub-overflow.cpp | 11 (void)(int16_t(-2) - int16_t(0x7fff));
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/external/libvpx/libvpx/vp9/common/ |
H A D | vp9_convolve.h | 22 const int16_t *filter_x, int x_step_q4, 23 const int16_t *filter_y, int y_step_q4,
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H A D | vp9_quant_common.c | 16 static const int16_t dc_qlookup[QINDEX_RANGE] = { 51 static const int16_t ac_qlookup[QINDEX_RANGE] = { 88 static int16_t dc_qlookup[QINDEX_RANGE]; 89 static int16_t ac_qlookup[QINDEX_RANGE]; 117 ac_qlookup[i] = (int16_t)ac_val; 118 dc_qlookup[i] = (int16_t)MAX(ACDC_MIN, poly3(0.000000305, -0.00065, 0.9, 124 int16_t vp9_dc_quant(int qindex, int delta) { 128 int16_t vp9_ac_quant(int qindex, int delta) {
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H A D | vp9_quant_common.h | 27 int16_t vp9_dc_quant(int qindex, int delta); 28 int16_t vp9_ac_quant(int qindex, int delta);
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/external/chromium_org/third_party/icu/source/i18n/ |
H A D | olsontz.h | 298 int16_t transitionCount() const; 300 int64_t transitionTimeInSeconds(int16_t transIdx) const; 301 double transitionTime(int16_t transIdx) const; 307 int32_t zoneOffsetAt(int16_t transIdx) const; 308 int32_t rawOffsetAt(int16_t transIdx) const; 309 int32_t dstOffsetAt(int16_t transIdx) const; 320 int16_t transitionCountPre32; 321 int16_t transitionCount32; 322 int16_t transitionCountPost32; 347 int16_t typeCoun [all...] |
/external/chromium_org/third_party/webrtc/common_audio/signal_processing/ |
H A D | resample_by_2_internal.h | 26 void WebRtcSpl_DownBy2IntToShort(int32_t *in, int32_t len, int16_t *out, 29 void WebRtcSpl_DownBy2ShortToInt(const int16_t *in, int32_t len, 32 void WebRtcSpl_UpBy2ShortToInt(const int16_t *in, int32_t len, 39 int16_t *out, int32_t *state); 41 void WebRtcSpl_LPBy2ShortToInt(const int16_t* in, int32_t len,
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | acm_g7221.h | 31 explicit ACMG722_1(int16_t codec_id); 37 int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte); 39 int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params); 44 int16_t InternalCreateEncoder();
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H A D | acm_g7221c.h | 31 explicit ACMG722_1C(int16_t codec_id); 37 int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte); 39 int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params); 44 int16_t InternalCreateEncoder();
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | TestVADDTX.h | 31 int32_t InFrameType(int16_t frameType); 57 int16_t RegisterSendCodec(char side, 62 void OpenOutFile(int16_t testNumber); 65 void SetVAD(bool statusDTX, bool statusVAD, int16_t vadMode); 67 int16_t VerifyTest();
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
H A D | audio_loop.cc | 26 audio_array_.reset(new int16_t[max_loop_length_samples + 28 size_t samples_read = fread(audio_array_.get(), sizeof(int16_t), 39 block_length_samples * sizeof(int16_t)); 46 const int16_t* AudioLoop::GetNextBlock() { 50 const int16_t* output_ptr = &audio_array_[next_index_];
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H A D | input_audio_file.cc | 22 bool InputAudioFile::Read(size_t samples, int16_t* destination) { 26 size_t samples_read = fread(destination, sizeof(int16_t), samples, fp_); 31 if (fread(destination, sizeof(int16_t), missing_samples, fp_) < 40 void InputAudioFile::DuplicateInterleaved(const int16_t* source, size_t samples, 42 int16_t* destination) {
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/external/icu/icu4c/source/i18n/ |
H A D | olsontz.h | 298 int16_t transitionCount() const; 300 int64_t transitionTimeInSeconds(int16_t transIdx) const; 301 double transitionTime(int16_t transIdx) const; 307 int32_t zoneOffsetAt(int16_t transIdx) const; 308 int32_t rawOffsetAt(int16_t transIdx) const; 309 int32_t dstOffsetAt(int16_t transIdx) const; 320 int16_t transitionCountPre32; 321 int16_t transitionCount32; 322 int16_t transitionCountPost32; 347 int16_t typeCoun [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | pitch_estimator.c | 21 static const int16_t kLogLagWinQ8[3] = { 26 static const int16_t kACoefQ12[3] = { 32 int16_t frac; 35 frac=(int16_t)WEBRTC_SPL_RSHIFT_W32(((uint32_t)(WEBRTC_SPL_LSHIFT_W32(x, zeros))&0x7FFFFFFF), 23); 43 static __inline int16_t Exp2Q10(int16_t x) { // Both in and out in Q10 45 int16_t tmp16_1, tmp16_2; 47 tmp16_2=(int16_t)(0x0400|(x&0x03FF)); 48 tmp16_1=-(int16_t)WEBRTC_SPL_RSHIFT_W16(x,10); 50 return (int16_t) WEBRTC_SPL_RSHIFT_W1 [all...] |
H A D | filterbanks.c | 33 int16_t *data_ch1, // Input and output in channel 1, in Q0 34 int16_t *data_ch2, // Input and output in channel 2, in Q0 35 const int16_t *factor_ch1, // Scaling factor for channel 1, in Q15 36 const int16_t *factor_ch2, // Scaling factor for channel 2, in Q15 43 int16_t in_out = 0; 55 a = WEBRTC_SPL_MUL_16_16(-factor_ch1[0], (int16_t) (b >> 16)); // Q15 57 in_out = (int16_t) (b >> 16); // Save as Q0 62 a = WEBRTC_SPL_MUL_16_16(-factor_ch1[1], (int16_t) (b >> 16)); // Q15 64 data_ch1[n] = (int16_t) (b >> 16); // Save as Q0 71 a = WEBRTC_SPL_MUL_16_16(-factor_ch2[0], (int16_t) ( [all...] |
/external/chromium_org/third_party/libvpx/source/libvpx/vp9/common/mips/dspr2/ |
H A D | vp9_common_dspr2.h | 88 void vp9_idct32_cols_add_blk_dspr2(int16_t *input, uint8_t *dest, 93 const int16_t *filter_x, int x_step_q4, 94 const int16_t *filter_y, int y_step_q4, 99 const int16_t *filter_x, int x_step_q4, 100 const int16_t *filter_y, int y_step_q4, 105 const int16_t *filter_x, int x_step_q4, 106 const int16_t *filter_y, int y_step_q4, 111 const int16_t *filter, 116 const int16_t *filter_x, int x_step_q4, 117 const int16_t *filter_ [all...] |