Searched refs:int16_t (Results 426 - 450 of 1591) sorted by relevance

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/external/chromium_org/ppapi/generators/test_thunk/
H A Dbasic_test_types.idl20 int16_t;
/external/chromium_org/third_party/libvpx/source/libvpx/vp9/encoder/arm/neon/
H A Dvp9_dct_neon.c18 void vp9_fdct8x8_1_neon(const int16_t *input, int16_t *output, int stride) {
35 void vp9_fdct8x8_neon(const int16_t *input, int16_t *final_output, int stride) {
66 int32x4_t v_t2_lo = vmull_n_s16(vget_low_s16(v_x2), (int16_t)cospi_24_64);
67 int32x4_t v_t2_hi = vmull_n_s16(vget_high_s16(v_x2), (int16_t)cospi_24_64);
68 int32x4_t v_t3_lo = vmull_n_s16(vget_low_s16(v_x3), (int16_t)cospi_24_64);
69 int32x4_t v_t3_hi = vmull_n_s16(vget_high_s16(v_x3), (int16_t)cospi_24_64);
70 v_t2_lo = vmlal_n_s16(v_t2_lo, vget_low_s16(v_x3), (int16_t)cospi_8_64);
71 v_t2_hi = vmlal_n_s16(v_t2_hi, vget_high_s16(v_x3), (int16_t)cospi_8_6
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/external/chromium_org/third_party/ots/src/
H A Dvdmx.h23 int16_t y_max;
24 int16_t y_min;
/external/chromium_org/third_party/skia/src/animator/
H A DSkBoundable.h22 bool hasBounds() { return fBounds.fLeft != (int16_t)0x8000U; }
25 void clearBounds() { fBounds.fLeft = (int16_t) SkToU16(0x8000); }; // mark bounds as unset
/external/chromium_org/third_party/webrtc/common_audio/signal_processing/
H A Ddot_product_with_scale.c13 int32_t WebRtcSpl_DotProductWithScale(const int16_t* vector1,
14 const int16_t* vector2,
H A Dresample_48khz.c27 void WebRtcSpl_Resample48khzTo16khz(const int16_t* in, int16_t* out,
31 // int16_t in[480]
47 // int16_t out[160]
65 void WebRtcSpl_Resample16khzTo48khz(const int16_t* in, int16_t* out,
69 // int16_t in[160]
85 // int16_t out[480]
103 void WebRtcSpl_Resample48khzTo8khz(const int16_t* in, int16_t* ou
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/g711/test/
H A DtestG711.cc27 int readframe(int16_t* data, FILE* inp, int length) {
31 rlen = (short) fread(data, sizeof(int16_t), length, inp);
48 int16_t framelength = 80;
57 int16_t stream_len = 0;
58 int16_t shortdata[480];
59 int16_t decoded[480];
60 int16_t streamdata[500];
61 int16_t speechType[1];
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/g722/test/
H A DtestG722.cc32 int readframe(int16_t *data, FILE *inp, int length)
36 rlen = (short)fread(data, sizeof(int16_t), length, inp);
52 int16_t framelength = 160;
62 int16_t stream_len = 0;
63 int16_t shortdata[960];
64 int16_t decoded[960];
65 int16_t streamdata[80*3];
66 int16_t speechType[1];
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/ilbc/
H A Ddecode.c40 int16_t *decblock, /* (o) decoded signal block */
44 int16_t mode /* (i) 0: bad packet, PLC,
48 int16_t order_plus_one;
50 int16_t last_bit;
51 int16_t *data;
53 int16_t decresidual[BLOCKL_MAX];
54 int16_t PLCresidual[BLOCKL_MAX + LPC_FILTERORDER];
55 int16_t syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
56 int16_t PLClpc[LPC_FILTERORDER + 1];
90 int16_t lsfde
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H A Ddecode_residual.c40 int16_t *decresidual, /* (o) decoded residual frame */
41 int16_t *syntdenum /* (i) the decoded synthesis filter
44 int16_t meml_gotten, Nfor, Nback, diff, start_pos;
45 int16_t subcount, subframe;
46 int16_t *reverseDecresidual = iLBCdec_inst->enh_buf; /* Reversed decoded data, used for decoding backwards in time (reuse memory in state) */
47 int16_t *memVec = iLBCdec_inst->prevResidual; /* Memory for codebook and filter state (reuse memory in state) */
48 int16_t *mem = &memVec[CB_HALFFILTERLEN]; /* Memory for codebook */
69 WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-iLBCdec_inst->state_short_len));
79 ST_MEM_L_TBL, (int16_t)diff
90 WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEM
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H A Dencode.c47 const int16_t *block, /* (i) speech vector to encode */
52 int16_t diff, start_pos;
55 int16_t start_count, end_count;
56 int16_t *residual;
58 int16_t scale, max;
59 int16_t *syntdenum;
60 int16_t *decresidual;
61 int16_t *reverseResidual;
62 int16_t *reverseDecresidual;
64 int16_t weightdenu
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/
H A Dfilterbanks_mips.c16 int16_t* data_ch1, // Input and output in channel 1, in Q0.
17 int16_t* data_ch2, // Input and output in channel 2, in Q0.
18 const int16_t* factor_ch1, // Scaling factor for channel 1, in Q15.
19 const int16_t* factor_ch2, // Scaling factor for channel 2, in Q15.
106 void WebRtcIsacfix_HighpassFilterFixDec32MIPS(int16_t* io,
107 int16_t len,
108 const int16_t* coefficient,
216 io[k] = (int16_t)a1;
H A Dbandwidth_estimator.c27 static const int16_t kQRateTable[12] = {
46 static const int16_t kRecHeaderRate[2] = {
144 const int16_t frameSize,
147 const int16_t pksize,
161 int16_t immediateSet = 0;
177 int16_t errCode;
203 recRtpRate = (int16_t)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(kBitsByteSec,
246 frameSizeSampl = WEBRTC_SPL_MUL_16_16((int16_t)SAMPLES_PER_MSEC, frameSize);
360 (int16_t)bweStr->countUpdates);
378 (int16_t)(pksiz
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H A Dfilterbanks_neon.S22 @ int16_t *data_ch1, // Input and output in channel 1, in Q0
23 @ int16_t *data_ch2, // Input and output in channel 2, in Q0
24 @ const int16_t *factor_ch1, // Scaling factor for channel 1, in Q15
25 @ const int16_t *factor_ch2, // Scaling factor for channel 2, in Q15
123 @ int16_t *data_ch1, // Input and output in channel 1, in Q0
124 @ int16_t *data_ch2, // Input and output in channel 2, in Q0
125 @ const int16_t *factor_ch1, // Scaling factor for channel 1, in Q15
126 @ const int16_t *factor_ch2, // Scaling factor for channel 2, in Q15
133 @ int16_t sample0_ch1 = 0, sample0_ch2 = 0;
134 @ int16_t sample1_ch
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/mock/
H A Dmock_audio_decoder.h25 MOCK_METHOD4(Decode, int(const uint8_t*, size_t, int16_t*,
28 MOCK_METHOD2(DecodePlc, int(int, int16_t*));
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/
H A Dnormal.h53 int Process(const int16_t* input, size_t length,
55 int16_t* external_mute_factor_array,
/external/libogg/macos/compat/sys/
H A Dtypes.h8 #ifndef __SYS_TYPES_H__ #define __SYS_TYPES_H__ 1 #include <MacTypes.h> #include <alloca.h> #include <string.h> typedef short int16_t; typedef long int32_t; typedef long long int64_t; #define vorbis_size32_t long #if defined(__c (…)
/external/libvorbis/macos/compat/sys/
H A Dtypes.h8 #ifndef __SYS_TYPES_H__ #define __SYS_TYPES_H__ 1 #include <MacTypes.h> #include <alloca.h> #include <string.h> typedef short int16_t; typedef long int32_t; typedef long long int64_t; #define vorbis_size32_t l (…)
/external/libvpx/libvpx/vp9/common/
H A Dvp9_mv.h23 int16_t row;
24 int16_t col;
H A Dvp9_convolve.c31 const int16_t *const x_filter = x_filters[x_q4 & SUBPEL_MASK];
53 const int16_t *const x_filter = x_filters[x_q4 & SUBPEL_MASK];
77 const int16_t *const y_filter = y_filters[y_q4 & SUBPEL_MASK];
100 const int16_t *const y_filter = y_filters[y_q4 & SUBPEL_MASK];
141 static const InterpKernel *get_filter_base(const int16_t *filter) {
147 static int get_filter_offset(const int16_t *f, const InterpKernel *base) {
153 const int16_t *filter_x, int x_step_q4,
154 const int16_t *filter_y, int y_step_q4,
165 const int16_t *filter_x, int x_step_q4,
166 const int16_t *filter_
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/external/skia/src/animator/
H A DSkBoundable.h22 bool hasBounds() { return fBounds.fLeft != (int16_t)0x8000U; }
25 void clearBounds() { fBounds.fLeft = (int16_t) SkToU16(0x8000); }; // mark bounds as unset
/external/webrtc/src/common_audio/vad/
H A Dvad_unittest.cc27 const int16_t kModes[] = { 0, 1, 2, 3 };
31 const int16_t kRates[] = { 8000, 12000, 16000, 24000, 32000 };
34 const int16_t kMaxFrameLength = 960;
35 const int16_t kFrameLengths[] = { 80, 120, 160, 240, 320, 480, 640,
40 bool ValidRatesAndFrameLengths(int16_t rate, int16_t frame_length) {
83 int16_t zeros[kMaxFrameLength] = { 0 };
87 int16_t speech[kMaxFrameLength];
88 for (int16_t i = 0; i < kMaxFrameLength; i++) {
164 int16_t delt
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/external/webrtc/src/modules/audio_processing/
H A Daudio_buffer.cc24 void StereoToMono(const int16_t* left, const int16_t* right,
25 int16_t* out, int samples_per_channel) {
41 int16_t data[kSamplesPer32kHzChannel];
54 int16_t low_pass_data[kSamplesPer16kHzChannel];
55 int16_t high_pass_data[kSamplesPer16kHzChannel];
97 int16_t* AudioBuffer::data(int channel) const {
106 int16_t* AudioBuffer::low_pass_split_data(int channel) const {
115 int16_t* AudioBuffer::high_pass_split_data(int channel) const {
124 int16_t* AudioBuffe
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/
H A DPCMFile.cc49 int16_t PCMFile::ChooseFile(std::string* file_name, int16_t max_len,
55 int16_t n = 0;
67 n = (int16_t)(strlen(tmp_name) - 1);
77 int16_t len = (int16_t) strlen(tmp_name);
152 int16_t* stereo_audio = new int16_t[2 * audio_frame.samples_per_channel_];
158 if (fwrite(stereo_audio, sizeof(int16_t),
166 if (fwrite(audio_frame.data_, sizeof(int16_t),
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/opus/
H A Dopus_unittest.cc42 int16_t speech_data_[kOpusMaxFrameSamples];
43 int16_t output_data_[kOpusMaxFrameSamples];
63 static_cast<int32_t>(fread(speech_data_, sizeof(int16_t),
130 int16_t encoded_bytes;
131 int16_t audio_type;
132 int16_t output_data_decode_new[kOpusMaxFrameSamples];
133 int16_t output_data_decode[kOpusMaxFrameSamples];
134 int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
173 int16_t encoded_byte
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