/external/webrtc/src/common_audio/signal_processing/ |
H A D | copy_set_operations.c | 67 WebRtc_Word16 samples, 70 // Copy the last <samples> of the input vector to vector_out 71 WEBRTC_SPL_MEMCPY_W16(vector_out, &vector_in[length - samples], samples); 73 return samples; 65 WebRtcSpl_CopyFromEndW16(G_CONST WebRtc_Word16 *vector_in, WebRtc_Word16 length, WebRtc_Word16 samples, WebRtc_Word16 *vector_out) argument
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/external/webrtc/src/modules/audio_processing/agc/ |
H A D | analog_agc.c | 114 WebRtc_Word16 samples) 130 if (samples == 80) 135 } else if (samples == 160) 144 "AGC->add_mic, frame %d: Invalid number of samples\n\n", 151 if (samples == 160) 156 } else if (samples == 320) 165 "AGC->add_mic, frame %d: Invalid number of samples\n\n", 173 if (samples == 160) 228 for (i = 0; i < samples; i++) 277 /* iterate over samples */ 113 WebRtcAgc_AddMic(void *state, WebRtc_Word16 *in_mic, WebRtc_Word16 *in_mic_H, WebRtc_Word16 samples) argument 330 WebRtcAgc_AddFarend(void *state, const WebRtc_Word16 *in_far, WebRtc_Word16 samples) argument 396 WebRtcAgc_VirtualMic(void *agcInst, WebRtc_Word16 *in_near, WebRtc_Word16 *in_near_H, WebRtc_Word16 samples, WebRtc_Word32 micLevelIn, WebRtc_Word32 *micLevelOut) argument 1245 WebRtcAgc_Process(void *agcInst, const WebRtc_Word16 *in_near, const WebRtc_Word16 *in_near_H, WebRtc_Word16 samples, WebRtc_Word16 *out, WebRtc_Word16 *out_H, WebRtc_Word32 inMicLevel, WebRtc_Word32 *outMicLevel, WebRtc_Word16 echo, WebRtc_UWord8 *saturationWarning) argument [all...] |
/external/tinycompress/ |
H A D | compress.c | 336 unsigned long *samples, unsigned int *sampling_rate) 346 *samples = ktstamp.pcm_io_frames; 335 compress_get_tstamp(struct compress *compress, unsigned long *samples, unsigned int *sampling_rate) argument
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/external/tremolo/Tremolo/ |
H A D | dsp.c | 143 /* pcm==0 indicates we just want the pending samples, no more */ 144 int vorbis_dsp_pcmout(vorbis_dsp_state *v,ogg_int16_t *pcm,int samples){ argument 151 if(n>samples)n=samples; 315 /* partial last frame. Strip the extra samples off */
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H A D | vorbisfile.c | 447 /* less than zero? This is a stream with samples trimmed off 592 int i,samples; local 614 samples=vorbis_dsp_pcmout(vf->vd,NULL,0); 616 granulepos-=samples; 954 /* returns: total PCM length (samples) of content if i==-1 PCM length 955 (samples) of that logical bitstream for i==0 to n 1407 /* discard samples until we reach the desired position. Crossing a 1411 long samples=vorbis_dsp_pcmout(vf->vd,NULL,0); local 1413 if(samples>target)samples 1582 long samples; local [all...] |
/external/tcpdump/ |
H A D | print-sflow.c | 59 * | num samples in datagram | 71 u_int8_t samples[4]; member in struct:sflow_datagram_t 849 nsamples=EXTRACT_32BITS(sflow_datagram->samples); 850 printf("sFlowv%u, %s agent %s, agent-id %u, seqnum %u, uptime %u, samples %u, length %u",
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/external/svox/pico/lib/ |
H A D | picoos.c | 1432 nr of samples in file 'nrSamples', header size 'hdrSize', 1500 /* warn "inconsistent number of samples" */ 1501 PICODBG_WARN(("inconsistent number of samples in wav file: %d vs. %d",nrFileSamples,(*nrSamples))); 1603 picoos_int16 samples[]) 1634 /* set n=min(rem,buffer_length) and try loading next n samples */ 1639 samples[j] = sdFile->buf[i]; 1824 extern picoos_bool picoos_sdfPutSamples (picoos_SDFile sdFile, picoos_uint32 nrSamples, picoos_int16 samples[]) argument 1833 s = samples[i]; 1599 picoos_sdfGetSamples( picoos_SDFile sdFile, picoos_uint32 start, picoos_uint32 * nrSamples, picoos_int16 samples[]) argument
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/external/srec/srec/EventLog/src/ |
H A D | riff.c | 50 * - fills in *length with the number of samples converted 51 * - allocates memory for *samples 61 char *cb, short **samples, int *length, int doSwap); 64 char *cb, short **samples, int *length, int doSwap); 66 char *cb, short **samples, int *length, int doSwap); 68 char *cb, short **samples, int *length, int doSwap); 219 char *cb, short **samples, int *length, int doSwap) 235 *samples = MALLOC(*length * sizeof(short), MTAG); 238 memcpy(*samples, cb, *length*sizeof(short)); 240 for (i = 0;i < *length;i++) swapShort(*samples 218 readPCMWave(WaveFormat *wf, ChunkContext *data, char *cb, short **samples, int *length, int doSwap) argument 251 readMulawWave(WaveFormat *wf, ChunkContext *data, char *cb, short **samples, int *length, int doSwap) argument 276 readAlawWave(WaveFormat *wf, ChunkContext *data, char *cb, short **samples, int *length, int doSwap) argument 556 riffReadWave2L16(FILE *f, double from, double to, short **samples, int *rate, int *length, SwiRiffStruct *swichunk) argument [all...] |
/external/srec/srec_jni/ |
H A D | android_speech_srec_Recognizer.cpp | 244 size_t samples = length / sizeof(asr_int16_t); local 245 length = samples * sizeof(asr_int16_t); 250 // put the samples into the recognizer 252 (asr_int16_t*)buffer, &samples, isLast ? ESR_TRUE : ESR_FALSE)); 253 if (samples != length / sizeof(asr_int16_t)) { 258 return samples * sizeof(asr_int16_t);
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/external/skia/samplecode/ |
H A D | SamplePatch.cpp | 69 static void eval_patch_edge(const SkPoint cubic[], SkPoint samples[], int segs) { argument 73 samples[0] = cubic[0]; 76 SkEvalCubicAt(cubic, t, &samples[i], NULL, NULL);
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/external/skia/src/gpu/gl/ |
H A D | GrGLNoOpInterface.cpp | 405 GrGLsizei samples, 404 noOpGLRenderbufferStorageMultisample(GrGLenum target, GrGLsizei samples, GrGLenum internalformat, GrGLsizei width, GrGLsizei height) argument
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H A D | GrGpuGL.cpp | 1175 int samples = rt->numSamples(); local 1195 if (samples > 0) { 1197 samples, 1216 samples, format)));
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/external/skia/src/gpu/gl/unix/ |
H A D | SkNativeGLContext_unix.cpp | 97 // Pick the FB config/visual with the most samples per pixel 105 int samp_buf, samples; local 107 glXGetFBConfigAttrib(fDisplay, fbc[i], GLX_SAMPLES, &samples); 111 // i, (unsigned int)vi->visualid, samp_buf, samples); 113 if (best_fbc < 0 || (samp_buf && samples > best_num_samp)) 114 best_fbc = i, best_num_samp = samples; 143 int samp_buf, samples; local 146 glXGetConfig(fDisplay, &visReturn[i], GLX_SAMPLES, &samples); 148 if (best < 0 || (samp_buf && samples > best_num_samp)) 149 best = i, best_num_samp = samples; [all...] |
/external/skia/tools/ |
H A D | Stats.h | 5 Stats(const double samples[], int n) { argument 6 min = samples[0]; 7 max = samples[0]; 9 if (samples[i] < min) { min = samples[i]; } 10 if (samples[i] > max) { max = samples[i]; } 15 sum += samples[i]; 21 err += (samples[i] - mean) * (samples[ [all...] |
/external/skia/dm/ |
H A D | DMGpuSupport.h | 23 int samples) { 24 return SkSurface::NewRenderTarget(grFactory->get(type), info, samples); 20 NewGpuSurface(GrContextFactory* grFactory, GrContextFactory::GLContextType type, SkImageInfo info, int samples) argument
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/external/skia/experimental/Intersection/ |
H A D | CubicToQuadratics_Test.cpp | 189 Cubic samples[arrayMax][sampleMax]; local 213 memcpy(samples[count][sCount], cubic, sizeof(Cubic)); 233 const Cubic& cubic = samples[x][y];
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/external/skia/gm/ |
H A D | morphology.cpp | 63 } samples[] = { local 74 for (unsigned i = 0; i < SK_ARRAY_COUNT(samples); ++i) { 78 samples[i].fRadiusX, 79 samples[i].fRadiusY, 84 samples[i].fRadiusX, 85 samples[i].fRadiusY,
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/external/qemu/audio/ |
H A D | alsaaudio.c | 170 snd_pcm_uframes_t samples; member in struct:alsa_params_obt 497 dolog ("obtained: samples %ld\n", obt->samples); 738 obt->samples = obt_buffer_size; 787 int left_till_end_samples = hw->samples - alsa->wpos; 838 alsa->wpos = (alsa->wpos + written) % hw->samples; 910 hw->samples = obt.samples; 912 alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift); 914 dolog ("Could not allocate DAC buffer (%d samples, eac [all...] |
H A D | audio.c | 803 int samples, struct mixeng_volume *vol) 807 (void) samples; 956 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) { 957 dolog ("live=%d hw->samples=%d\n", live, hw->samples); 966 int left = hw->samples - pending; 973 int samples_till_end_of_buf = hw->samples - hw->rpos; 978 hw->rpos = (hw->rpos + samples_to_clip) % hw->samples; 994 if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) { 995 dolog ("live=%d hw->samples 802 noop_conv(struct st_sample *dst, const void *src, int samples, struct mixeng_volume *vol) argument 1011 int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0; local 1108 int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck; local 1441 audio_capture_mix_and_clear(HWVoiceOut *hw, int rpos, int samples) argument [all...] |
H A D | audio_int.h | 84 int samples; member in struct:HWVoiceOut 104 int samples; member in struct:HWVoiceIn
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H A D | audio_template.h | 83 HWBUF = audio_calloc (AUDIO_FUNC, hw->samples, sizeof (struct st_sample)); 85 dolog ("Could not allocate " NAME " buffer (%d samples)\n", 86 hw->samples); 109 int samples; local 112 samples = sw->hw->samples; 114 samples = ((int64_t) sw->hw->samples << 32) / sw->ratio; 117 sw->buf = audio_calloc (AUDIO_FUNC, samples, sizeof (struct st_sample)); 119 dolog ("Could not allocate buffer for `%s' (%d samples)\ [all...] |
H A D | esdaudio.c | 93 int samples; member in struct:__anon29236 98 .samples = 1024, 120 threshold = conf.divisor ? hw->samples / conf.divisor : 0; 152 int chunk = audio_MIN (to_mix, hw->samples - rpos); 176 rpos = (rpos + wsamples) % hw->samples; 180 rpos = (rpos + chunk) % hw->samples; 241 dolog ("Will use 16 instead of 32 bit samples\n"); 259 hw->samples = conf.samples; 260 esd->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, [all...] |
H A D | mixeng_template.h | 112 (struct st_sample *dst, const void *src, int samples, struct mixeng_volume *vol) 118 mixeng_clear (dst, samples); 124 while (samples--) { 132 (struct st_sample *dst, const void *src, int samples, struct mixeng_volume *vol) 138 mixeng_clear (dst, samples); 144 while (samples--) { 153 (void *dst, const struct st_sample *src, int samples) 157 while (samples--) { 165 (void *dst, const struct st_sample *src, int samples) 169 while (samples 111 _to_stereo(struct st_sample *dst, const void *src, int samples, struct mixeng_volume *vol) argument 131 _to_mono(struct st_sample *dst, const void *src, int samples, struct mixeng_volume *vol) argument 152 _from_stereo(void *dst, const struct st_sample *src, int samples) argument 164 _from_mono(void *dst, const struct st_sample *src, int samples) argument [all...] |
H A D | noaudio.c | 44 int decr, samples; local 53 samples = bytes >> hw->info.shift; 56 decr = audio_MIN (live, samples); 57 hw->rpos = (hw->rpos + decr) % hw->samples; 69 hw->samples = 1024; 88 hw->samples = 1024; 101 int dead = hw->samples - live; 102 int samples = 0; local 112 samples = bytes >> hw->info.shift; 113 samples 120 int samples = size >> sw->info.shift; local [all...] |
H A D | paaudio.c | 66 int samples; member in struct:__anon29243 72 .samples = 1024, 93 threshold = conf.divisor ? hw->samples / conf.divisor : 0; 125 int chunk = audio_MIN (to_mix, hw->samples - rpos); 136 rpos = (rpos + chunk) % hw->samples; 188 threshold = conf.divisor ? hw->samples / conf.divisor : 0; 220 int chunk = audio_MIN (to_grab, hw->samples - wpos); 230 wpos = (wpos + chunk) % hw->samples; 258 dead = hw->samples - live; 355 hw->samples [all...] |