/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
H A D | audio_buffer.h | 61 int16_t* data(int channel); 62 const int16_t* data(int channel) const; 63 int16_t* low_pass_split_data(int channel); 64 const int16_t* low_pass_split_data(int channel) const; 65 int16_t* high_pass_split_data(int channel); 66 const int16_t* high_pass_split_data(int channel) const; 69 const int16_t* mixed_low_pass_data(); 70 const int16_t* low_pass_reference(int channel) const; 73 // as necessary. The range of the numbers are the same as for int16_t. 122 scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels [all...] |
H A D | high_pass_filter_impl.cc | 23 const int16_t kFilterCoefficients8kHz[5] = 26 const int16_t kFilterCoefficients[5] = 30 int16_t y[4]; 31 int16_t x[2]; 32 const int16_t* ba; 50 int Filter(FilterState* hpf, int16_t* data, int length) { 54 int16_t* y = hpf->y; 55 int16_t* x = hpf->x; 56 const int16_t* ba = hpf->ba; 84 y[0] = static_cast<int16_t>(tmp_int3 [all...] |
/external/libvpx/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...] |
/external/chromium_org/third_party/webrtc/modules/audio_processing/aecm/ |
H A D | aecm_core_c.c | 28 extern const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END; 30 static const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = { 57 static const int16_t kNoiseEstQDomain = 15; 58 static const int16_t kNoiseEstIncCount = 5; 63 const int16_t* lambda); 66 int16_t* fft, 67 const int16_t* time_signal, 76 fft[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( 80 fft[PART_LEN + i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( 88 WebRtcSpl_RealForwardFFT(aecm->real_fft, fft, (int16_t*)freq_signa [all...] |
H A D | aecm_core_mips.c | 18 static const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = { 29 static const int16_t kNoiseEstQDomain = 15; 30 static const int16_t kNoiseEstIncCount = 5; 32 static int16_t coefTable[] = { 51 static int16_t coefTable_ifft[] = { 73 const int16_t* lambda); 76 int16_t* fft, 77 const int16_t* time_signal, 82 int16_t* pfrfi; 84 int16_t f_coe [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | fft.c | 21 const int16_t kSortTabFft[240] = { 45 const int16_t kCosTabFfftQ14[240] = { 71 int16_t WebRtcIsacfix_FftRadix16Fastest(int16_t RexQx[], int16_t ImxQx[], int16_t iSign) { 73 int16_t dd, ee, ff, gg, hh, ii; 74 int16_t k0, k1, k2, k3, k4, kk; 75 int16_t tmp116, tmp216; 77 int16_t ccc1Q1 [all...] |
H A D | lpc_masking_model.c | 27 int16_t *a16, /* Q11 */ 28 int16_t useOrder, 29 int16_t *k16 /* Q15 */ 36 int16_t tmp_inv_denum16; 42 tmp_inv_denum16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp_inv_denum32, 15); // (1 - k^2) in Q15 52 a16[k] = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32[k], 1); //Q12>>1 => Q11 56 k16[m-1] = (int16_t) WEBRTC_SPL_LSHIFT_W32(tmp32[m], 3); //Q12<<3 => Q15 66 int16_t WebRtcSpl_LevinsonW32_JSK( 68 int16_t *A, /* (o) A[0..order] LPC coefficients (Q11) */ 69 int16_t * [all...] |
H A D | pitch_estimator_c.c | 22 void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8) { 23 int16_t scaling,n,k; 26 const int16_t* x; 27 const int16_t* inptr; 32 scaling = WebRtcSpl_GetScalingSquare((int16_t*)in, 39 ysum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)in[n], 40 (int16_t)in[n], 42 csum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)x[n], 43 (int16_t)in[n], 63 ysum32 -= WEBRTC_SPL_MUL_16_16_RSFT((int16_t)i [all...] |
H A D | decode_bwe.c | 34 int16_t index; 35 int16_t frame_samples; 59 (int16_t) packet_size, /* in bytes */
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H A D | decode.c | 30 int16_t WebRtcIsacfix_DecodeImpl(int16_t *signal_out16, 32 int16_t *current_framesamples) 36 int16_t BWno; 37 int16_t len = 0; 39 int16_t model; 42 int16_t Vector_Word16_1[FRAMESAMPLES/2]; 43 int16_t Vector_Word16_2[FRAMESAMPLES/2]; 48 int16_t lofilt_coefQ15[ORDERLO*SUBFRAMES]; //refl. coeffs 49 int16_t hifilt_coefQ1 [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | dsp_helper.cc | 23 const int16_t DspHelper::kParabolaCoefficients[17][3] = { 47 const int16_t DspHelper::kDownsample8kHzTbl[3] = { 1229, 1638, 1229 }; 50 const int16_t DspHelper::kDownsample16kHzTbl[5] = { 614, 819, 1229, 819, 614 }; 53 const int16_t DspHelper::kDownsample32kHzTbl[7] = { 57 const int16_t DspHelper::kDownsample48kHzTbl[7] = { 60 int DspHelper::RampSignal(const int16_t* input, 64 int16_t* output) { 76 int DspHelper::RampSignal(int16_t* signal, 102 void DspHelper::PeakDetection(int16_t* data, int data_length, 104 int* peak_index, int16_t* peak_valu [all...] |
H A D | merge.cc | 27 int Merge::Process(int16_t* input, size_t input_length, 28 int16_t* external_mute_factor_array, 46 int16_t best_correlation_index = 0; 50 int16_t* input_channel = &input_vector[channel][0]; 51 int16_t* expanded_channel = &expanded_[channel][0]; 52 int16_t expanded_max, input_max; 53 int16_t new_mute_factor = SignalScaling( 59 int16_t* external_mute_factor = &external_mute_factor_array[channel]; 66 static_cast<int16_t>(16384)); 83 int16_t temp_dat [all...] |
H A D | dtmf_tone_generator.h | 42 static const int16_t kAmpMultiplier = 23171; // 3 dB attenuation (in Q15). 48 int16_t sample_history1_[2]; // Last 2 samples for the 1st oscillator. 49 int16_t sample_history2_[2]; // Last 2 samples for the 2nd oscillator.
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | acm_isac.cc | 62 ACMISAC::ACMISAC(int16_t /* codec_id */) 79 int16_t ACMISAC::InternalEncode(uint8_t* /* bitstream */, 80 int16_t* /* bitstream_len_byte */) { 84 int16_t ACMISAC::InternalInitEncoder(WebRtcACMCodecParams* /* codec_params */) { 88 int16_t ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* /* codec_params */) { 92 int16_t ACMISAC::InternalCreateEncoder() { return -1; } 96 int16_t ACMISAC::Transcode(uint8_t* /* bitstream */, 97 int16_t* /* bitstream_len_byte */, 98 int16_t /* q_bwe */, 104 int16_t ACMISA [all...] |
H A D | acm_g722.cc | 26 ACMG722::ACMG722(int16_t /* codec_id */) 34 const int16_t* /* data */, 40 int16_t ACMG722::InternalEncode(uint8_t* /* bitstream */, 41 int16_t* /* bitstream_len_byte */) { 45 int16_t ACMG722::InternalInitEncoder(WebRtcACMCodecParams* /* codec_params */) { 51 int16_t ACMG722::InternalCreateEncoder() { return -1; } 67 ACMG722::ACMG722(int16_t codec_id) 96 const int16_t* data, 103 int16_t ACMG722::InternalEncode(uint8_t* bitstream, 104 int16_t* bitstream_len_byt [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
H A D | input_audio_file_unittest.cc | 23 int16_t input[kSamples]; 27 int16_t output[kSamples * kChannels]; 31 int16_t* output_ptr = output; 34 EXPECT_EQ(static_cast<int16_t>(i), *output_ptr++); 42 int16_t input[kSamples * kChannels]; 49 int16_t* output_ptr = input; 52 EXPECT_EQ(static_cast<int16_t>(i), *output_ptr++);
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/external/webrtc/src/modules/audio_processing/ |
H A D | audio_buffer.h | 32 int16_t* data(int channel) const; 33 int16_t* low_pass_split_data(int channel) const; 34 int16_t* high_pass_split_data(int channel) const; 35 int16_t* mixed_data(int channel) const; 36 int16_t* mixed_low_pass_data(int channel) const; 37 int16_t* low_pass_reference(int channel) const; 72 int16_t* data_;
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/external/chromium_org/third_party/webrtc/common_audio/signal_processing/ |
H A D | downsample_fast_mips.c | 14 int WebRtcSpl_DownsampleFast_mips(const int16_t* data_in, 16 int16_t* data_out, 18 const int16_t* __restrict coefficients, 29 int16_t* p_coefficients; 30 int16_t* p_data_in; 31 int16_t* p_data_in_0 = (int16_t*)&data_in[delay]; 32 int16_t* p_coefficients_0 = (int16_t*)&coefficients[0];
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H A D | downsample_fast.c | 15 int WebRtcSpl_DownsampleFastC(const int16_t* data_in, 17 int16_t* data_out, 19 const int16_t* __restrict coefficients,
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/external/android-clat/ |
H A D | config.h | 28 int16_t mtu, ipv4mtu; 32 int16_t ipv4_local_prefixlen; 44 in_addr_t config_select_ipv4_address(const struct in_addr *ip, int16_t prefixlen);
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/external/chromium_org/third_party/webrtc/common_audio/ |
H A D | audio_util.cc | 17 void RoundToInt16(const float* src, size_t size, int16_t* dest) { 22 void ScaleAndRoundToInt16(const float* src, size_t size, int16_t* dest) { 27 void ScaleToFloat(const int16_t* src, size_t size, float* dest) {
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/external/chromium_org/third_party/webrtc/common_audio/resampler/ |
H A D | push_sinc_resampler.h | 36 int Resample(const int16_t* source, int source_frames, 37 int16_t* destination, int destination_capacity); 55 const int16_t* source_ptr_int_;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | spectrum_ar_model_tables.h | 46 extern const int16_t WebRtcIsac_kQArBoundaryLevels[NUM_AR_RC_QUANT_BAUNDARY]; 55 extern const int16_t *WebRtcIsac_kQArRcLevelsPtr[AR_ORDER]; 76 extern const int16_t WebRtcIsac_kCos[6][60];
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H A D | structs.h | 225 int16_t inWaitLatePkts; 270 int16_t framelength; 286 int16_t fre[FRAMESAMPLES]; 287 int16_t fim[FRAMESAMPLES]; 288 int16_t AvgPitchGain[2]; 304 int16_t realFFT[FRAMESAMPLES_HALF]; 305 int16_t imagFFT[FRAMESAMPLES_HALF]; 321 int16_t current_framesamples; 327 int16_t new_framelength; 331 int16_t payloadLimitBytes3 [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | TestRedFec.h | 34 int16_t RegisterSendCodec(char side, char* codecName, 37 void OpenOutFile(int16_t testNumber); 46 int16_t _testCntr;
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