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  /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;

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