/external/chromium_org/ppapi/generators/test_thunk/ |
H A D | basic_test_types.idl | 20 int16_t;
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/external/chromium_org/third_party/libvpx/source/libvpx/vp9/encoder/arm/neon/ |
H A D | vp9_dct_neon.c | 18 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 [all...] |
/external/chromium_org/third_party/ots/src/ |
H A D | vdmx.h | 23 int16_t y_max; 24 int16_t y_min;
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/external/chromium_org/third_party/skia/src/animator/ |
H A D | SkBoundable.h | 22 bool hasBounds() { return fBounds.fLeft != (int16_t)0x8000U; } 25 void clearBounds() { fBounds.fLeft = (int16_t) SkToU16(0x8000); }; // mark bounds as unset
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/external/chromium_org/third_party/webrtc/common_audio/signal_processing/ |
H A D | dot_product_with_scale.c | 13 int32_t WebRtcSpl_DotProductWithScale(const int16_t* vector1, 14 const int16_t* vector2,
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H A D | resample_48khz.c | 27 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 [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/g711/test/ |
H A D | testG711.cc | 27 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];
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/g722/test/ |
H A D | testG722.cc | 32 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];
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/ilbc/ |
H A D | decode.c | 40 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 [all...] |
H A D | decode_residual.c | 40 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 [all...] |
H A D | encode.c | 47 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 [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | filterbanks_mips.c | 16 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;
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H A D | bandwidth_estimator.c | 27 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 [all...] |
H A D | filterbanks_neon.S | 22 @ 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 [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/mock/ |
H A D | mock_audio_decoder.h | 25 MOCK_METHOD4(Decode, int(const uint8_t*, size_t, int16_t*, 28 MOCK_METHOD2(DecodePlc, int(int, int16_t*));
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | normal.h | 53 int Process(const int16_t* input, size_t length, 55 int16_t* external_mute_factor_array,
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/external/libogg/macos/compat/sys/ |
H A D | types.h | 8 #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 (…)
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/external/libvorbis/macos/compat/sys/ |
H A D | types.h | 8 #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 (…)
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/external/libvpx/libvpx/vp9/common/ |
H A D | vp9_mv.h | 23 int16_t row; 24 int16_t col;
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H A D | vp9_convolve.c | 31 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_ [all...] |
/external/skia/src/animator/ |
H A D | SkBoundable.h | 22 bool hasBounds() { return fBounds.fLeft != (int16_t)0x8000U; } 25 void clearBounds() { fBounds.fLeft = (int16_t) SkToU16(0x8000); }; // mark bounds as unset
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/external/webrtc/src/common_audio/vad/ |
H A D | vad_unittest.cc | 27 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 [all...] |
/external/webrtc/src/modules/audio_processing/ |
H A D | audio_buffer.cc | 24 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 [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | PCMFile.cc | 49 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), [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/opus/ |
H A D | opus_unittest.cc | 42 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 [all...] |