/external/libvpx/libvpx/test/ |
H A D | fdct4x4_test.cc | 26 void vp9_idct4x4_16_add_c(const int16_t *input, uint8_t *output, int pitch); 33 typedef void (*fdct_t)(const int16_t *in, int16_t *out, int stride); 34 typedef void (*idct_t)(const int16_t *in, uint8_t *out, int stride); 35 typedef void (*fht_t) (const int16_t *in, int16_t *out, int stride, 37 typedef void (*iht_t) (const int16_t *in, uint8_t *out, int stride, 43 void fdct4x4_ref(const int16_t *in, int16_t *out, int stride, int tx_type) { 47 void fht4x4_ref(const int16_t *i [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | acm_g7221.cc | 23 // int16_t WebRtcG7221_CreateEnc16(G722_1_16_encinst_t_** enc_inst); 24 // int16_t WebRtcG7221_CreateEnc24(G722_1_24_encinst_t_** enc_inst); 25 // int16_t WebRtcG7221_CreateEnc32(G722_1_32_encinst_t_** enc_inst); 26 // int16_t WebRtcG7221_CreateDec16(G722_1_16_decinst_t_** dec_inst); 27 // int16_t WebRtcG7221_CreateDec24(G722_1_24_decinst_t_** dec_inst); 28 // int16_t WebRtcG7221_CreateDec32(G722_1_32_decinst_t_** dec_inst); 30 // int16_t WebRtcG7221_FreeEnc16(G722_1_16_encinst_t_** enc_inst); 31 // int16_t WebRtcG7221_FreeEnc24(G722_1_24_encinst_t_** enc_inst); 32 // int16_t WebRtcG7221_FreeEnc32(G722_1_32_encinst_t_** enc_inst); 33 // int16_t WebRtcG7221_FreeDec1 [all...] |
H A D | acm_g7221c.cc | 23 // int16_t WebRtcG7221C_CreateEnc24(G722_1C_24_encinst_t_** enc_inst); 24 // int16_t WebRtcG7221C_CreateEnc32(G722_1C_32_encinst_t_** enc_inst); 25 // int16_t WebRtcG7221C_CreateEnc48(G722_1C_48_encinst_t_** enc_inst); 26 // int16_t WebRtcG7221C_CreateDec24(G722_1C_24_decinst_t_** dec_inst); 27 // int16_t WebRtcG7221C_CreateDec32(G722_1C_32_decinst_t_** dec_inst); 28 // int16_t WebRtcG7221C_CreateDec48(G722_1C_48_decinst_t_** dec_inst); 30 // int16_t WebRtcG7221C_FreeEnc24(G722_1C_24_encinst_t_** enc_inst); 31 // int16_t WebRtcG7221C_FreeEnc32(G722_1C_32_encinst_t_** enc_inst); 32 // int16_t WebRtcG7221C_FreeEnc48(G722_1C_48_encinst_t_** enc_inst); 33 // int16_t WebRtcG7221C_FreeDec2 [all...] |
H A D | acm_generic_codec.cc | 99 const int16_t* data, 107 const int16_t* data, 148 int16_t missed_samples = in_audio_ix_write_ + length_smpl * audio_channel - 154 sizeof(int16_t)); 158 data, length_smpl * audio_channel * sizeof(int16_t)); 161 int16_t missed_10ms_blocks =static_cast<int16_t>( 182 length_smpl * audio_channel * sizeof(int16_t)); 199 int16_t ACMGenericCodec::Encode(uint8_t* bitstream, 200 int16_t* bitstream_len_byt [all...] |
H A D | acm_opus.cc | 26 ACMOpus::ACMOpus(int16_t /* codec_id */) 39 int16_t ACMOpus::InternalEncode(uint8_t* /* bitstream */, 40 int16_t* /* bitstream_len_byte */) { 44 int16_t ACMOpus::InternalInitEncoder(WebRtcACMCodecParams* /* codec_params */) { 52 int16_t ACMOpus::InternalCreateEncoder() { 60 int16_t ACMOpus::SetBitRateSafe(const int32_t /*rate*/) { 66 ACMOpus::ACMOpus(int16_t codec_id) 94 int16_t ACMOpus::InternalEncode(uint8_t* bitstream, 95 int16_t* bitstream_len_byte) { 116 int16_t ACMOpu [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | time_stretch.cc | 23 const int16_t* input, 26 int16_t* length_change_samples) { 31 const int16_t* signal; 32 scoped_ptr<int16_t[]> signal_array; 42 signal_array.reset(new int16_t[signal_len]); 64 int16_t peak_value; 87 const int16_t* vec1 = &signal[fs_mult_120 - peak_index]; 89 const int16_t* vec2 = &signal[fs_mult_120]; 105 int16_t best_correlation; 122 // Scale energies to int16_t [all...] |
H A D | audio_vector.cc | 29 memcpy(copy_to->array_.get(), array_.get(), Size() * sizeof(int16_t)); 37 memmove(&array_[insert_length], &array_[0], Size() * sizeof(int16_t)); 38 memcpy(&array_[0], &prepend_this.array_[0], insert_length * sizeof(int16_t)); 42 void AudioVector::PushFront(const int16_t* prepend_this, size_t length) { 51 void AudioVector::PushBack(const int16_t* append_this, size_t length) { 53 memcpy(&array_[first_free_ix_], append_this, length * sizeof(int16_t)); 63 memmove(&array_[0], &array_[length], remaining_samples * sizeof(int16_t)); 76 memset(&array_[first_free_ix_], 0, extra_length * sizeof(int16_t)); 80 void AudioVector::InsertAt(const int16_t* insert_this, 87 int16_t* insert_position_pt [all...] |
H A D | normal.cc | 27 int Normal::Process(const int16_t* input, 30 int16_t* external_mute_factor_array, 46 int16_t* signal = &(*output)[0][0]; 69 external_mute_factor_array[channel_ix] = static_cast<int16_t>( 73 int16_t* signal = &(*output)[channel_ix][0]; 76 int16_t decoded_max = WebRtcSpl_MaxAbsValueW16( 100 int16_t energy_scaled = energy << scaling; 101 int16_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled); 111 int16_t increment = 64 / fs_mult; 144 int16_t cng_outpu [all...] |
H A D | expand.cc | 38 int16_t random_vector[kMaxSampleRate / 8000 * 120 + 30]; 39 int16_t scaled_random_vector[kMaxSampleRate / 8000 * 125]; 41 int16_t temp_data[kTempDataSize]; // TODO(hlundin) Remove this. 42 int16_t* voiced_vector_storage = temp_data; 43 int16_t* voiced_vector = &voiced_vector_storage[overlap_length_]; 45 int16_t unvoiced_array_memory[kNoiseLpcOrder + kMaxSampleRate / 8000 * 125]; 46 int16_t* unvoiced_vector = unvoiced_array_memory + kUnvoicedLpcOrder; 47 int16_t* noise_vector = unvoiced_array_memory + kNoiseLpcOrder; 58 int16_t rand_length = max_lag_; 84 sizeof(int16_t) * temp_lengt [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | isacfix.c | 46 int16_t WebRtcIsacfix_AssignSize(int *sizeinbytes) { 47 *sizeinbytes=sizeof(ISACFIX_SubStruct)*2/sizeof(int16_t); 60 int16_t WebRtcIsacfix_Assign(ISACFIX_MainStruct **inst, void *ISACFIX_inst_Addr) { 88 int16_t WebRtcIsacfix_Create(ISACFIX_MainStruct **ISAC_main_inst) 117 int16_t WebRtcIsacfix_CreateInternal(ISACFIX_MainStruct *ISAC_main_inst) 151 int16_t WebRtcIsacfix_Free(ISACFIX_MainStruct *ISAC_main_inst) 169 int16_t WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst) 248 int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst, 249 int16_t CodingMode) 252 int16_t statusIni [all...] |
H A D | lattice_mips.c | 16 void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0, // Input samples 17 int16_t* ar_f_Q0, // Input samples 18 int16_t* cth_Q15, // Filter coefficients 19 int16_t* sth_Q15, // Filter coefficients 20 int16_t order_coef) { // order of the filter 27 int16_t* tmp_cth; 28 int16_t* tmp_sth; 29 int16_t* tmp_arg; 125 void WebRtcIsacfix_FilterMaLoopMIPS(int16_t input0, // Filter coefficient 126 int16_t input [all...] |
H A D | pitch_estimator_mips.c | 17 void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8) { 18 int16_t scaling,n,k; 21 const int16_t* x; 22 const int16_t* inptr; 26 scaling = WebRtcSpl_GetScalingSquare((int16_t*)in, 33 const int16_t* tmp_x = x; 34 const int16_t* tmp_in = in; 104 const int16_t* tmp_in1 = &in[k - 1]; 105 const int16_t* tmp_in2 = &in[PITCH_CORR_LEN2 + k - 1]; 106 const int16_t* tmp_ [all...] |
H A D | decode_plc.c | 36 static int16_t plc_filterma_Fast( 37 int16_t *In, /* (i) Vector to be filtered. InOut[-orderCoef+1] 39 int16_t *Out, /* (o) Filtered vector */ 40 int16_t *B, /* (i) The filter coefficients (in Q0) */ 41 int16_t Blen, /* (i) Number of B coefficients */ 42 int16_t len, /* (i) Number of samples to be filtered */ 43 int16_t reduceDecay, 44 int16_t decay, 45 int16_t rshift ) 55 const int16_t *b_pt [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_processing/agc/ |
H A D | digital_agc.c | 59 static const int16_t kAvgDecayTime = 250; // frames; < 3000 62 int16_t digCompGaindB, // Q0 63 int16_t targetLevelDbfs,// Q0 65 int16_t analogTarget) // Q0 76 const int16_t kCompRatio = 3; 77 const int16_t kSoftLimiterLeft = 1; 78 int16_t limiterOffset = 0; // Limiter offset 79 int16_t limiterIdx, limiterLvlX; 80 int16_t constLinApprox, zeroGainLvl, maxGain, diffGain; 81 int16_t [all...] |
/external/chromium_org/third_party/webrtc/common_audio/signal_processing/ |
H A D | spl_sqrt.c | 25 int16_t x_half, t16; 40 x_half = (int16_t)WEBRTC_SPL_RSHIFT_W32(B, 16);// x_half = x/2 = (in-1)/2 50 t16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(A, 16); 54 t16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(A, 16); 56 t16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(A, 16); 60 t16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(x2, 16); 135 int16_t x_norm, nshift, t16, sh; 138 int16_t k_sqrt_2 = 23170; // 1/sqrt2 (==5a82) 155 x_norm = (int16_t)WEBRTC_SPL_RSHIFT_W32(A, 16); // x_norm = AH 167 t16 = (int16_t)WEBRTC_SPL_RSHIFT_W3 [all...] |
H A D | min_max_operations.c | 36 int16_t WebRtcSpl_MaxAbsValueW16C(const int16_t* vector, int length) { 56 return (int16_t)maximum; 84 int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, int length) { 85 int16_t maximum = WEBRTC_SPL_WORD16_MIN; 116 int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, int length) { 117 int16_t minimum = WEBRTC_SPL_WORD16_MAX; 148 int WebRtcSpl_MaxAbsIndexW16(const int16_t* vecto [all...] |
/external/chromium_org/third_party/skia/tests/ |
H A D | WArrayTest.cpp | 17 static const int16_t data1[] = {-1, 0, -3, 4, 5, 6, 7, 0, 0, 0, 8, 0, 0, 0, 0}; 22 static const int16_t data2[] = {0, 0, 0, 100, 100, 100, 100, 100, 100, 100, 0, 0}; 27 static const int16_t data3[] = {1, 2, 0, 0, 0, 3, 4, 0, 0, 0, 0, 5}; 32 static const int16_t data4[] = {1, 0, 0, 0, 1, 2, 2, 2, 3, 0, 0, 0, 0, 3, 4}; 37 static const int16_t data5[] = {1, 1, 2, 3, 0, 0, 0, 0, 5, 5, 6, 7, 0, 0, 0, 0, 8, 0}; 42 static const int16_t data6[] = {1, 0, 0, 0, 0, 1, 2, 3, 3, 4, 5, 5, 5, 6}; 47 static const int16_t data7[] = {1, 2, 10, 11, 2, 3}; 54 static const int16_t data8[] = {1, 2, 10, 11, 12, 2, 3}; 61 static const int16_t data9[] = {1, 1, 10, 2, 3}; 68 static const int16_t data1 [all...] |
/external/skia/tests/ |
H A D | WArrayTest.cpp | 17 static const int16_t data1[] = {-1, 0, -3, 4, 5, 6, 7, 0, 0, 0, 8, 0, 0, 0, 0}; 22 static const int16_t data2[] = {0, 0, 0, 100, 100, 100, 100, 100, 100, 100, 0, 0}; 27 static const int16_t data3[] = {1, 2, 0, 0, 0, 3, 4, 0, 0, 0, 0, 5}; 32 static const int16_t data4[] = {1, 0, 0, 0, 1, 2, 2, 2, 3, 0, 0, 0, 0, 3, 4}; 37 static const int16_t data5[] = {1, 1, 2, 3, 0, 0, 0, 0, 5, 5, 6, 7, 0, 0, 0, 0, 8, 0}; 42 static const int16_t data6[] = {1, 0, 0, 0, 0, 1, 2, 3, 3, 4, 5, 5, 5, 6}; 47 static const int16_t data7[] = {1, 2, 10, 11, 2, 3}; 54 static const int16_t data8[] = {1, 2, 10, 11, 12, 2, 3}; 61 static const int16_t data9[] = {1, 1, 10, 2, 3}; 68 static const int16_t data1 [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_processing/ns/ |
H A D | nsx_core_neon.c | 17 const int16_t WebRtcNsx_kLogTable[9] = { 21 const int16_t WebRtcNsx_kCounterDiv[201] = { 38 const int16_t WebRtcNsx_kLogTableFrac[256] = { 60 const int16_t kExp2Const = 11819; // Q13 61 int16_t* ptr_noiseEstLogQuantile = NULL; 62 int16_t* ptr_noiseEstQuantile = NULL; 68 int16_t tmp16 = WebRtcSpl_MaxValueW16(inst->noiseEstLogQuantile + offset, 92 // tmp16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21); 98 // tmp16 += (int16_t) inst->qNoise; 124 tmp16 = (int16_t) WEBRTC_SPL_RSHIFT_W3 [all...] |
/external/chromium_org/third_party/webrtc/common_audio/vad/ |
H A D | vad_filterbank_unittest.cc | 28 static const int16_t kReference[kNumValidFrameLengths] = { 48, 11, 11 }; 29 static const int16_t kFeatures[kNumValidFrameLengths * kNumChannels] = { 34 static const int16_t kOffsetVector[kNumChannels] = { 36 int16_t features[kNumChannels]; 40 int16_t speech[kMaxFrameLength]; 41 for (int16_t i = 0; i < kMaxFrameLength; ++i) { 76 for (int16_t i = 0; i < kMaxFrameLength; ++i) {
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | Channel.h | 30 int16_t maxPayloadLen; 41 int16_t payloadType; 42 int16_t lastPayloadLenByte; 50 Channel(int16_t chID = -1); 63 int16_t Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats); 106 int16_t _lastPayloadType; 113 int16_t _packetLoss;
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/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/ |
H A D | mixing_test.cc | 21 const int16_t kLimiterHeadroom = 29204; // == -1 dbFS 22 const int16_t kInt16Max = 0x7fff; 59 int16_t input_value, 60 int16_t max_output_value, 61 int16_t min_output_value, 112 void GenerateInputFile(int16_t input_value) { 121 void VerifyMixedOutput(int16_t max_output_value, int16_t min_output_value) { 125 int16_t output_value = 0; 234 const int16_t kInputValu [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_processing/aecm/ |
H A D | aecm_core.c | 30 const int16_t WebRtcAecm_kCosTable[] = { 73 const int16_t WebRtcAecm_kSinTable[] = { 122 static const int16_t kChannelStored8kHz[PART_LEN1] = { 135 static const int16_t kChannelStored16kHz[PART_LEN1] = { 220 sizeof(int16_t)); 229 sizeof(int16_t)); 238 sizeof(int16_t)); 247 sizeof(int16_t)); 282 aecm->xBuf = (int16_t*) (((uintptr_t)aecm->xBuf_buf + 31) & ~ 31); 283 aecm->dBufClean = (int16_t*) (((uintptr_ [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/ilbc/test/ |
H A D | iLBC_test.c | 47 int16_t data[BLOCKL_MAX]; 48 int16_t encoded_data[ILBCNOOFWORDS_MAX], decoded_data[BLOCKL_MAX]; 54 int16_t speechType; 158 while (((int16_t)fread(data,sizeof(int16_t),frameLen,ifileid))== 166 len=WebRtcIlbcfix_Encode(Enc_Inst, data, (int16_t)frameLen, encoded_data); 171 if (fwrite(encoded_data, sizeof(int16_t), 172 ((len+1)/sizeof(int16_t)), efileid) != 173 (size_t)(((len+1)/sizeof(int16_t)))) { 179 if (fread(&pli, sizeof(int16_t), [all...] |
/external/webrtc/src/modules/audio_processing/ns/ |
H A D | nsx_core_neon.c | 19 const int16_t kExp2Const = 11819; // Q13 20 int16_t* ptr_noiseEstLogQuantile = NULL; 21 int16_t* ptr_noiseEstQuantile = NULL; 27 int16_t tmp16 = WebRtcSpl_MaxValueW16(inst->noiseEstLogQuantile + offset, 51 // tmp16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21); 57 // tmp16 += (int16_t) inst->qNoise; 83 tmp16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21); 85 tmp16 += (int16_t) inst->qNoise; //shift to get result in Q(qNoise) 98 int16_t* q_noise) { 99 int16_t lmag [all...] |