/hardware/libhardware/modules/usbaudio/ |
alsa_device_proxy.h | 29 struct pcm * pcm; member in struct:__anon41452
|
alsa_device_profile.c | 218 struct pcm * pcm = pcm_open(profile->card, profile->device, local 221 if (pcm != NULL) { 222 works = pcm_is_ready(pcm); 223 pcm_close(pcm);
|
/cts/suite/audio_quality/lib/src/audio/ |
AudioPlaybackLocal.cpp | 72 LOGE("Unable to open PCM device(%d) (%s)\n", mHwId, pcm_get_error(mPcmHandle)); 103 pcm_* pcm = (pcm_*)mPcmHandle; local 104 ioctl(pcm->fd, SNDRV_PCM_IOCTL_DRAIN);
|
/external/libvorbis/examples/ |
decoder_example.c | 18 /* Takes a vorbis bitstream from stdin and writes raw stereo PCM to 53 vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */ 54 vorbis_block vb; /* local working space for packet->PCM decode */ 200 packet->PCM decoder. */ 227 float **pcm; local 234 **pcm is a multichannel float vector. In stereo, for 235 example, pcm[0] is left, and pcm[1] is right. samples is 237 (-1.<=range<=1.) to whatever PCM format and write it out */ 239 while((samples=vorbis_synthesis_pcmout(&vd,&pcm))>0) [all...] |
/external/libvorbis/test/ |
write_read.c | 255 float **pcm; local 260 while ((samples = vorbis_synthesis_pcmout (&vd,&pcm)) > 0 && read_total < count) { 264 memcpy (data + read_total, pcm[0], bout * sizeof (float)) ;
|
/external/tremolo/Tremolo/ |
mapping0.c | 141 /* recover the spectral envelope; store it in the PCM vector for now */ 196 //_analysis_output("coupled",seq+j,vb->pcm[j],-8,n/2,0,0); 227 //_analysis_output("residue",seq+j,vb->pcm[j],-8,n/2,0,0); 231 ogg_int32_t *pcm=vd->work[i]; local 241 floor1_inverse2(vd,ci->floor_param[floorno],floormemo[i],pcm); 244 floor0_inverse2(vd,ci->floor_param[floorno],floormemo[i],pcm); 249 //_analysis_output("mdct",seq+j,vb->pcm[j],-24,n/2,0,1); 251 /* transform the PCM data; takes PCM vector, vb; modifies PCM vector * [all...] |
/frameworks/base/core/java/android/speech/srec/ |
UlawEncoderInputStream.java | 28 * InputStream which transforms 16 bit pcm data to ulaw data. 81 int pcm = (0xff & pcmBuf[pcmOffset++]) + (pcmBuf[pcmOffset++] << 8); local 82 pcm = (pcm * coef) >> SCALE_BITS; 85 if (pcm >= 0) { 86 ulaw = pcm <= 0 ? 0xff : 87 pcm <= 30 ? 0xf0 + (( 30 - pcm) >> 1) : 88 pcm <= 94 ? 0xe0 + (( 94 - pcm) >> 2) 123 int pcm = (0xff & pcmBuf[offset++]) + (pcmBuf[offset++] << 8); local [all...] |
/external/tinyalsa/ |
tinycap.c | 189 struct pcm *pcm; local 203 pcm = pcm_open(card, device, PCM_IN, &config); 204 if (!pcm || !pcm_is_ready(pcm)) { 205 fprintf(stderr, "Unable to open PCM device (%s)\n", 206 pcm_get_error(pcm)); 210 size = pcm_frames_to_bytes(pcm, pcm_get_buffer_size(pcm)); 215 pcm_close(pcm); [all...] |
tinyplay.c | 196 fprintf(stderr, "Unable to open PCM device %u.\n", device); 216 struct pcm *pcm; local 237 pcm = pcm_open(card, device, PCM_OUT, &config); 238 if (!pcm || !pcm_is_ready(pcm)) { 239 fprintf(stderr, "Unable to open PCM device %u (%s)\n", 240 device, pcm_get_error(pcm)); 244 size = pcm_frames_to_bytes(pcm, pcm_get_buffer_size(pcm)); [all...] |
/frameworks/base/libs/usb/tests/accessorytest/ |
audio.c | 125 struct pcm *pcm = NULL; local 138 while (!pcm) { 139 pcm = pcm_open(input_card, input_device, PCM_IN, &config); 140 if (pcm && !pcm_is_ready(pcm)) { 141 pcm_close(pcm); 142 pcm = NULL; 144 if (!pcm) 148 while (pcm) { 166 struct pcm *pcm = arg; local 193 struct pcm *pcm; local [all...] |
/frameworks/wilhelm/tests/examples/ |
slesTestSawtoothBufferQueue.cpp | 116 SLDataFormat_PCM pcm; local 169 pcm.formatType = SL_DATAFORMAT_PCM; 170 pcm.numChannels = 1;//2; 171 pcm.samplesPerSec = SL_SAMPLINGRATE_44_1; 172 pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; 173 pcm.containerSize = 16; 174 pcm.channelMask = SL_SPEAKER_FRONT_LEFT;// | SL_SPEAKER_FRONT_RIGHT; 175 pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; 177 audioSource.pFormat = (void *)&pcm; 244 /* Play the PCM samples using a buffer queue * [all...] |
slesTestFeedback.cpp | 358 SLDataFormat_PCM pcm; local 365 pcm.formatType = SL_DATAFORMAT_PCM; 366 pcm.numChannels = channels; 367 pcm.samplesPerSec = sampleRate * 1000; 368 pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; 369 pcm.containerSize = 16; 370 pcm.channelMask = channels == 1 ? SL_SPEAKER_FRONT_CENTER : 372 pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; 374 audiosrc.pFormat = &pcm; 438 audiosnk.pFormat = &pcm; [all...] |
slesTestRecBuffQueue.cpp | 29 Signed 16-bit PCM 170 SLDataFormat_PCM pcm; local 207 pcm.formatType = SL_DATAFORMAT_PCM; 208 pcm.numChannels = 1; 209 pcm.samplesPerSec = SL_SAMPLINGRATE_22_05; 210 pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; 211 pcm.containerSize = 16; 212 pcm.channelMask = SL_SPEAKER_FRONT_LEFT; 213 pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; 216 recDest.pFormat = (void * ) &pcm; [all...] |
/prebuilts/gcc/linux-x86/host/x86_64-linux-glibc2.11-4.6/sysroot/usr/include/alsa/ |
pcm_extplug.h | 11 * ALSA external PCM plugin SDK (draft version) 37 * See the \ref pcm page for more details. 86 * PCM handle filled by #snd_pcm_extplug_create() 88 snd_pcm_t *pcm; member in struct:snd_pcm_extplug 135 * close the PCM; optional 180 * set the parameter constraint for slave PCM with a single value
|
/prebuilts/gcc/linux-x86/host/x86_64-linux-glibc2.11-4.8/sysroot/usr/include/alsa/ |
pcm_extplug.h | 11 * ALSA external PCM plugin SDK (draft version) 37 * See the \ref pcm page for more details. 86 * PCM handle filled by #snd_pcm_extplug_create() 88 snd_pcm_t *pcm; member in struct:snd_pcm_extplug 135 * close the PCM; optional 180 * set the parameter constraint for slave PCM with a single value
|
/cts/tests/tests/nativemedia/sl/src/ |
SLObjectCreationTest.cpp | 26 * * source is BufferQueue of PCM buffers 28 * * source is URI, sink is BufferQueue of PCM buffers 29 * * source is FD, sink is BufferQueue of PCM buffers 30 * * source is AndroidBufferQueue of AAC ADTS buffers, sink is BufferQueue of PCM buffers 32 * * source is IO device, sink is BufferQueue of PCM buffers 206 /* Test case for creating an AudioPlayer object that plays from a PCM BufferQueue */ 208 // source: PCM BufferQueue 212 SLDataFormat_PCM pcm; local 213 pcm.formatType = SL_DATAFORMAT_PCM; 214 pcm.numChannels = 2 [all...] |
/device/asus/fugu/libaudio/ |
AudioStreamIn.cpp | 358 struct pcm* pcm = pcm_open(deviceInfo->pcmCard, deviceInfo->pcmDevice, local 361 if (!pcm_is_ready(pcm)) { 363 pcm_close(pcm); 372 mBufferSize = pcm_frames_to_bytes(pcm, mPcmConfig.period_size); 393 pcm_close(pcm); 398 mPcm = pcm;
|
/external/libvorbis/include/vorbis/ |
codec.h | 63 float **pcm; member in struct:vorbis_dsp_state 90 float **pcm; /* this is a pointer into local storage */ member in struct:vorbis_block 156 packet back into PCM audio. 212 extern int vorbis_synthesis_pcmout(vorbis_dsp_state *v,float ***pcm); 213 extern int vorbis_synthesis_lapout(vorbis_dsp_state *v,float ***pcm);
|
/external/libvorbis/lib/ |
block.c | 13 function: PCM data vector blocking, windowing and dis/reassembly 16 Handle windowing, overlap-add, etc of the PCM vectors. This is made 44 /* pcm accumulator examples (not exhaustive): 251 v->pcm=_ogg_malloc(vi->channels*sizeof(*v->pcm)); 256 v->pcm[i]=_ogg_calloc(v->pcm_storage,sizeof(*v->pcm[i])); 368 if(v->pcm){ 371 if(v->pcm[i])_ogg_free(v->pcm[i]) 774 float *pcm=v->pcm[j]+prevCenter; local 781 float *pcm=v->pcm[j]+prevCenter+n1\/2-n0\/2; local 790 float *pcm=v->pcm[j]+prevCenter; local 799 float *pcm=v->pcm[j]+prevCenter; local 808 float *pcm=v->pcm[j]+thisCenter; local [all...] |
envelope.c | 13 function: PCM data envelope analysis 227 /* make sure we have enough storage to match the PCM */ 241 float *pcm=v->pcm[i]+ve->searchstep*(j); local 242 ret|=_ve_amp(ve,gi,pcm,ve->band,ve->filter+i*VE_BANDS); 289 _analysis_output_always("pcmL",seq,v->pcm[0],v->pcm_current,0,0,totalshift); 290 _analysis_output_always("pcmR",seq,v->pcm[1],v->pcm_current,0,0,totalshift); 292 _analysis_output_always("markL",seq,v->pcm[0],j,0,0,totalshift); 293 _analysis_output_always("markR",seq,v->pcm[1],j,0,0,totalshift);
|
mapping0.c | 266 float *pcm =vb->pcm[i]; local 267 float *logfft =pcm; 290 _analysis_output("pcmL",seq,pcm,n,0,0,total-n/2); 292 _analysis_output("pcmR",seq,pcm,n,0,0,total-n/2); 294 _analysis_output("pcm",seq,pcm,n,0,0,total-n/2); 298 /* window the PCM data */ 299 _vorbis_apply_window(pcm,b->window,ci->blocksizes,vb->lW,vb->W,vb->nW); 304 _analysis_output("windowedL",seq,pcm,n,0,0,total-n/2) 791 float *pcm=vb->pcm[i]; local 801 float *pcm=vb->pcm[i]; local [all...] |
psytune.c | 242 float *pcm[2],*out[2],*window,*flr[2],*mask[2],*work[2]; local 276 pcm[0]=_ogg_malloc(framesize*sizeof(float)); 277 pcm[1]=_ogg_malloc(framesize*sizeof(float)); 314 pcm[0][i]=((buffer[i*4+1]<<8)| 316 pcm[1][i]=((buffer[i*4+3]<<8)| 330 float *mdct=pcm[i]; 333 analysis("pre",frameno+i,pcm[i],framesize,0,0); 337 fft[j]=pcm[i][j]*=window[j]; 351 mdct_forward(&m_look,pcm[i],mdct); 363 float *mdct=pcm[i] [all...] |
/frameworks/base/cmds/bootanimation/ |
AudioPlayer.cpp | 207 struct pcm *pcm = NULL; local 284 pcm = pcm_open(mCard, mDevice, PCM_OUT, &config); 285 if (!pcm || !pcm_is_ready(pcm)) { 286 ALOGE("Unable to open PCM device (%s)\n", pcm_get_error(pcm)); 290 bufferSize = pcm_frames_to_bytes(pcm, pcm_get_buffer_size(pcm)); 298 if (pcm_write(pcm, wavData, count)) [all...] |
/frameworks/wilhelm/tests/automated/ |
BufferQueue_test.cpp | 80 SLDataFormat_PCM pcm; member in class:TestBufferQueue 120 pcm.formatType = SL_DATAFORMAT_PCM; 121 pcm.numChannels = 2; 122 pcm.samplesPerSec = SL_SAMPLINGRATE_44_1; 123 pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; 124 pcm.containerSize = 16; 125 pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; 126 pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; 129 audiosrc.pFormat = &pcm; 140 int pcm = (int) (pcm_ * 32766.0 * gVolume) local [all...] |
/frameworks/wilhelm/tests/sandbox/ |
intbufq.c | 105 SLDataFormat_PCM pcm; local 112 pcm.formatType = SL_DATAFORMAT_PCM; 113 pcm.numChannels = 2; 114 pcm.samplesPerSec = SL_SAMPLINGRATE_44_1; 115 pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; 116 pcm.containerSize = 16; 117 pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; 118 pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; 120 audiosrc.pFormat = &pcm;
|