| /cts/apps/CtsVerifier/src/com/android/cts/verifier/audioquality/experiments/ |
| BiasExperiment.java | 45 short[] pcm = Utils.byteToShortArray(record); local
46 float[] results = mNative.measureRms(pcm, AudioQualityVerifierActivity.SAMPLE_RATE, ONSET_THRESH);
|
| OverflowExperiment.java | 61 short[] pcm = Utils.byteToShortArray(record); local
62 float[] ret = mNative.overflowCheck(pcm, AudioQualityVerifierActivity.SAMPLE_RATE);
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| SoundLevelExperiment.java | 62 short[] pcm = Utils.byteToShortArray(record); local
63 float[] results = mNative.measureRms(pcm, AudioQualityVerifierActivity.SAMPLE_RATE, ONSET_THRESH);
|
| SpectrumShapeExperiment.java | 51 short[] pcm = Utils.byteToShortArray(record); local
53 float[] ret = mNative.compareSpectra(pcm, refPcm, AudioQualityVerifierActivity.SAMPLE_RATE);
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| GlitchExperiment.java | 75 short[] pcm = Utils.byteToShortArray(record); local
77 ONSET_THRESH, SNR_THRESH, pcm);
|
| /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...] |
| /cts/apps/CtsVerifier/jni/audioquality/ |
| Wrapper.cpp | 105 short *pcm = new short[numSamples]; local
106 env->GetShortArrayRegion(jpcm, 0, numSamples, pcm);
108 measureRms(pcm, numSamples, sampleRate, onsetThresh, ret, ret + 1,
133 short *pcm = new short[numSamples]; local
134 env->GetShortArrayRegion(jpcm, 0, numSamples, pcm);
140 int success = gt.checkToneSnr(pcm, numSamples, &duration, &badFrames);
164 short *pcm = new short[numSamples]; local
165 env->GetShortArrayRegion(jpcm, 0, numSamples, pcm);
170 int success = overflowCheck(pcm, numSamples, sampleRate,
195 short *pcm = new short[numSamples] local
[all...] |
| /external/tinyalsa/ |
| tinycap.c | 147 struct pcm *pcm; local
164 pcm = pcm_open(0, device, PCM_IN, &config);
165 if (!pcm || !pcm_is_ready(pcm)) {
166 fprintf(stderr, "Unable to open PCM device (%s)\n",
167 pcm_get_error(pcm));
171 size = pcm_get_buffer_size(pcm);
176 pcm_close(pcm);
182 while (capturing && !pcm_read(pcm, buffer, size))
[all...] |
| tinyplay.c | 94 fprintf(stderr, "Error: '%s' is not a PCM riff/wave file\n", argv[1]);
111 struct pcm *pcm; local
128 pcm = pcm_open(0, device, PCM_OUT, &config);
129 if (!pcm || !pcm_is_ready(pcm)) {
130 fprintf(stderr, "Unable to open PCM device %u (%s)\n",
131 device, pcm_get_error(pcm));
135 size = pcm_get_buffer_size(pcm);
140 pcm_close(pcm);
[all...] |
| pcm.c | 0 /* pcm.c
145 struct pcm { struct
162 unsigned int pcm_get_buffer_size(struct pcm *pcm)
164 return pcm->buffer_size;
167 const char* pcm_get_error(struct pcm *pcm)
169 return pcm->error;
172 static int oops(struct pcm *pcm, int e, const char *fmt, ...
461 struct pcm *pcm; local
[all...] |
| /prebuilt/linux-x86/toolchain/i686-linux-glibc2.7-4.4.3/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
|
| pcm_ioplug.h | 11 * ALSA external PCM plugin SDK
37 * See the \ref pcm page for more details.
61 #define SND_PCM_IOPLUG_FLAG_LISTED (1<<0) /* list up this PCM */
100 * PCM handle filled by #snd_pcm_extplug_create()
102 snd_pcm_t *pcm; member in struct:snd_pcm_ioplug
105 snd_pcm_state_t state; /**< current PCM state; read-only */
111 snd_pcm_format_t format; /**< PCM format; filled after hw_params is called */
121 * start the PCM; required
125 * stop the PCM; required
140 * close the PCM; optiona
[all...] |
| /system/media/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 | 294 SLDataFormat_PCM pcm; local
301 pcm.formatType = SL_DATAFORMAT_PCM;
302 pcm.numChannels = channels;
303 pcm.samplesPerSec = sampleRate * 1000;
304 pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
305 pcm.containerSize = 16;
306 pcm.channelMask = channels == 1 ? SL_SPEAKER_FRONT_CENTER :
308 pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
310 audiosrc.pFormat = &pcm;
350 audiosnk.pFormat = &pcm;
[all...] |
| slesTestRecBuffQueue.cpp | 29 Signed 16-bit PCM
167 SLDataFormat_PCM pcm; local
204 pcm.formatType = SL_DATAFORMAT_PCM;
205 pcm.numChannels = 1;
206 pcm.samplesPerSec = SL_SAMPLINGRATE_22_05;
207 pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
208 pcm.containerSize = 16;
209 pcm.channelMask = SL_SPEAKER_FRONT_LEFT;
210 pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
213 recDest.pFormat = (void * ) &pcm;
[all...] |
| slesTestDecodeToBuffQueue.cpp | 29 Signed 16-bit PCM
69 /* metadata key index for the PCM format information we want to retrieve */
72 /* size of the struct to retrieve the PCM format metadata values: the values we're interested in
79 /* we only want to query / display the PCM format once */
242 /* Example: query of the decoded PCM format */
251 // but the call was successful for the PCM format keys, so those conditions are implied
296 SLDataFormat_PCM pcm; local
311 /* allocate memory to receive the PCM format metadata */
346 pcm.formatType = SL_DATAFORMAT_PCM;
348 pcm.numChannels = 1
[all...] |
| /cts/tests/tests/nativemedia/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
202 /* Test case for creating an AudioPlayer object that plays from a PCM BufferQueue */
204 // source: PCM BufferQueue
208 SLDataFormat_PCM pcm; local
209 pcm.formatType = SL_DATAFORMAT_PCM;
210 pcm.numChannels = 2
[all...] |
| /system/media/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...] |
| /system/media/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;
|
| /external/qemu/audio/ |
| ossaudio.c | 398 void *pcm = advance (oss->pcm_buf, oss->wpos << hw->info.shift); local
400 bytes_written = write (oss->fd, pcm, bytes_to_write);
|
| /external/quake/quake/src/WinQuake/ |
| snd_android.cpp | 97 SLDataFormat_PCM pcm; variable
156 pcm.formatType = SL_DATAFORMAT_PCM;
157 pcm.numChannels = 2;
158 pcm.samplesPerSec = SL_SAMPLINGRATE_11_025;
159 pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
160 pcm.containerSize = 16;
161 pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
162 pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
164 audioSource.pFormat = (void *)&pcm;
228 /* Play the PCM samples using a buffer queue *
[all...] |
| /external/bluetooth/bluez/audio/ |
| unix.c | 254 pcm_capabilities_t *pcm; local
259 codec->length = sizeof(*pcm);
261 pcm = (void *) codec;
262 pcm->sampling_rate = 8000;
265 pcm->flags |= BT_PCM_FLAG_NREC;
267 pcm->flags |= BT_PCM_FLAG_PCM_ROUTING;
271 pcm->flags |= BT_PCM_FLAG_NREC;
[all...] |
| /frameworks/ex/variablespeed/jni/ |
| variablespeed.cc | 610 SLDataFormat_PCM pcm = {SL_DATAFORMAT_PCM, 1, SL_SAMPLINGRATE_44_1, local
613 SLDataSink decDest = { &decBuffQueue, &pcm };
|
| /device/samsung/tuna/audio/ |
| audio_hw.c | 488 struct pcm *pcm_modem_dl;
489 struct pcm *pcm_modem_ul;
518 struct pcm *pcm[PCM_TOTAL]; member in struct:tuna_stream_out
535 struct pcm *pcm; member in struct:tuna_stream_in
623 /* Open modem PCM channels */
627 LOGE("cannot open PCM modem DL stream: %s", pcm_get_error(adev->pcm_modem_dl));
635 LOGE("cannot open PCM modem UL stream: %s", pcm_get_error(adev->pcm_modem_ul));
[all...] |
