Searched refs:int16_t (Results 326 - 350 of 1591) sorted by relevance

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/external/chromium_org/third_party/webrtc/modules/audio_processing/
H A Daudio_buffer.h61 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
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H A Dhigh_pass_filter_impl.cc23 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
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/external/libvpx/libvpx/vp9/common/mips/dspr2/
H A Dvp9_common_dspr2.h88 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_
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/external/chromium_org/third_party/webrtc/modules/audio_processing/aecm/
H A Daecm_core_c.c28 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
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H A Daecm_core_mips.c18 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
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/
H A Dfft.c21 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
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H A Dlpc_masking_model.c27 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 *
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H A Dpitch_estimator_c.c22 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
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H A Ddecode_bwe.c34 int16_t index;
35 int16_t frame_samples;
59 (int16_t) packet_size, /* in bytes */
H A Ddecode.c30 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
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/
H A Ddsp_helper.cc23 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
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H A Dmerge.cc27 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
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H A Ddtmf_tone_generator.h42 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.
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/
H A Dacm_isac.cc62 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
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H A Dacm_g722.cc26 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
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/
H A Dinput_audio_file_unittest.cc23 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++);
/external/webrtc/src/modules/audio_processing/
H A Daudio_buffer.h32 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_;
/external/chromium_org/third_party/webrtc/common_audio/signal_processing/
H A Ddownsample_fast_mips.c14 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];
H A Ddownsample_fast.c15 int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
17 int16_t* data_out,
19 const int16_t* __restrict coefficients,
/external/android-clat/
H A Dconfig.h28 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);
/external/chromium_org/third_party/webrtc/common_audio/
H A Daudio_util.cc17 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) {
/external/chromium_org/third_party/webrtc/common_audio/resampler/
H A Dpush_sinc_resampler.h36 int Resample(const int16_t* source, int source_frames,
37 int16_t* destination, int destination_capacity);
55 const int16_t* source_ptr_int_;
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/
H A Dspectrum_ar_model_tables.h46 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];
H A Dstructs.h225 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
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/
H A DTestRedFec.h34 int16_t RegisterSendCodec(char side, char* codecName,
37 void OpenOutFile(int16_t testNumber);
46 int16_t _testCntr;

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