| /external/oprofile/libop/ |
| op_interface.h | 32 /** Data type to transfer samples counts from the module to the daemon */
52 size_t count; /**< number of samples in this buffer */
|
| /external/qemu/audio/ |
| mixeng.h | 37 int samples, struct mixeng_volume *vol);
38 typedef void (f_sample) (void *dst, const struct st_sample *src, int samples);
|
| 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)\n"
[all...] |
| ossaudio.c | 395 int samples_till_end = hw->samples - oss->wpos;
417 oss->wpos = (oss->wpos + samples_written) % hw->samples;
432 bufsize = hw->samples << hw->info.shift;
494 err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
497 oss->pcm_buf, hw->samples << hw->info.shift);
549 hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
555 hw->samples << hw->info.shift,
563 hw->samples << hw->info.shift);
585 err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
588 oss->pcm_buf, hw->samples << hw->info.shift)
[all...] |
| winaudio.c | 243 hw->samples = conf.nb_samples*2;
273 int hw_samples = audio_MIN(hw->samples - hw->rpos, live);
284 // s->write_pos, s->write_size, wav_samples, wav_bytes, live, hw->rpos, hw->samples);
287 if (hw->rpos >= hw->samples)
288 hw->rpos -= hw->samples;
494 hw->samples = conf.nb_samples*2;
504 /* report the number of captured samples to the audio subsystem */
511 int live = hw->samples - hw->total_samples_captured;
533 int hw_samples = audio_MIN(hw->samples - hw->wpos, live);
545 hw->wpos, hw->samples);
[all...] |
| /external/qemu/distrib/sdl-1.2.12/src/audio/sun/ |
| SDL_sunaudio.h | 40 Sint32 written; /* The number of samples written */
41 int fragsize; /* The audio fragment size in samples */
|
| /external/webkit/Source/ThirdParty/ANGLE/src/libGLESv2/ |
| Renderbuffer.cpp | 197 Colorbuffer::Colorbuffer(int width, int height, GLenum format, GLsizei samples)
203 int supportedSamples = getContext()->getNearestSupportedSamples(requestedFormat, samples);
322 DepthStencilbuffer::DepthStencilbuffer(int width, int height, GLsizei samples)
328 int supportedSamples = getContext()->getNearestSupportedSamples(D3DFMT_D24S8, samples);
417 Depthbuffer::Depthbuffer(int width, int height, GLsizei samples) : DepthStencilbuffer(width, height, samples)
455 Stencilbuffer::Stencilbuffer(int width, int height, GLsizei samples) : DepthStencilbuffer(width, height, samples)
|
| /external/webrtc/src/common_audio/resampler/main/interface/ |
| resampler.h | 73 // Synchronous resampling, all output samples are written to samplesOut
80 // Asynchronous resampling output, remaining samples are buffered
|
| /frameworks/base/core/java/android/service/textservice/ |
| package.html | 35 href="{@docRoot}resources/samples/SpellChecker/SampleSpellCheckerService/index.html">Spell
37 href="{@docRoot}resources/samples/SpellChecker/HelloSpellChecker/index.html">Spell
|
| /system/media/wilhelm/src/desktop/ |
| SDL.c | 52 fmt.samples = SndFile_BUFSIZE;
54 fmt.samples = SndFile_BUFSIZE / sizeof(short);
|
| /external/webrtc/src/modules/audio_processing/agc/main/source/ |
| 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)
224 for (i = 0; i < samples; i++)
273 /* iterate over samples */
[all...] |
| /external/chromium/chrome/common/extensions/docs/static/ |
| samples.html | 1 <link rel="stylesheet" href="css/samples.css" />
11 * Adds the samples manifest data + API modules.
13 pageData.samples = samples;
29 samples.forEach(function(sample) {
82 <div jsselect="samples" jsvalues="id:id" class="sample">
|
| /frameworks/base/tools/localize/ |
| localize_test.cpp | 36 if (all) err |= test_filename("//device/samples/NotePad/res/values/strings.xml", "zz_ZZ",
37 "//device/samples/NotePad/res/values-zz-rZZ/strings.xml");
39 if (all) err |= test_filename("//device/samples/NotePad/res/values/strings.xml", "zz",
40 "//device/samples/NotePad/res/values-zz/strings.xml");
42 if (all) err |= test_filename("//device/samples/NotePad/res/values/strings.xml", "",
43 "//device/samples/NotePad/res/values/strings.xml");
|
| /frameworks/ex/variablespeed/jni/ |
| sola_time_scaler.h | 54 // @param buffer1 pointer to interleaved input samples
55 // @param buffer2 pointer to interleaved input samples
57 // samples / number of channels)
106 // @param buffer pointer to interleaved float input samples
156 // Munges input samples to produce output.
157 // @returns true if any output samples were generated
|
| /external/srec/audio/AudioIn/UNIX/src/ |
| audioin.c | 92 /* size in samples */
103 and 2 samples if stereo. This should be distinguished from what the
105 a set of consecutive samples.
132 static audioinSample gSamplesBufferCircularFifo[SAMPLES_BUFFER_SIZE]; /* circular buffer that buffers the incoming samples */
134 static int gWriteIndexPointer = 0; /* write pointer in the circular FIFO samples buffer */
135 static int gReadIndexPointer = 0; /* read pointer in the circular FIFO samples buffer */
339 * samples from the codec and store them in a big circular *
407 /* buffer of 16 bits samples */
427 /* NOTE: here a frame is comprised of 1 sample if mono, 2 samples if stereo, etc */
428 int iReadSamples = 0; /* number of samples acquired by the codec *
[all...] |
| /frameworks/base/media/libeffects/lvm/lib/StereoWidening/src/ |
| LVCS_Process.c | 61 /* NumSamples Number of samples in the input buffer */
103 NumSamples); /* Number of samples to process */
111 NumSamples); /* Number of samples to process */
118 NumSamples); /* Number of samples to process */
127 NumSamples); /* Number of samples to process */
154 /* NumSamples Number of samples in the input buffer */
174 * Check the number of samples is not too large
|
| /ndk/tests/ |
| run-tests.sh | 28 # and that the samples will be under samples/ and platforms/android-N/samples/
46 TESTABLES="samples build device awk"
93 --only-samples)
94 TESTABLES=samples
136 echo " --only-samples Only rebuild samples"
357 ### REBUILD ALL SAMPLES FIRST
409 # Determine list of samples directories
[all...] |
| /prebuilt/linux-x86/toolchain/i686-linux-glibc2.7-4.4.3/i686-linux/include/c++/4.4.3/parallel/ |
| multiway_mergesort.h | 80 /** @brief Samples. */
81 value_type* samples; member in struct:__gnu_parallel::PMWMSSortingData
91 * @brief Select samples from a sequence.
93 * @c sd->samples.
94 * @param num_samples Number of samples to select.
113 ::new(&(sd->samples[iam * num_samples + i]))
206 __gnu_sequential::sort(sd->samples,
207 sd->samples + (num_samples * sd->num_threads),
220 sd->samples[num_samples * iam],
232 sd->samples[num_samples * (iam + 1)]
[all...] |
| /external/oprofile/daemon/liblegacy/ |
| opd_proc.c | 209 * or module samples files. Here we will catch also
292 /* There is a race for samples received between fork/exec sequence.
293 * These samples belong to the old mapping but we can not say if
294 * samples has been received before the exec or after. This explains
298 * samples, the error message can count only out of mapping samples but
299 * not samples between the race and inside the mapping of the exec'ed
303 * only manner to handle this is to flush the module samples hash table
385 * age dead proc in such way if a proc doesn't receive any samples
|
| /libcore/dalvik/src/test/java/dalvik/system/profiler/ |
| SamplingProfilerTest.java | 159 // should have no samples to start
181 Set<Sample> samples = hprofData.getSamples(); local
182 assertNotNull(samples);
183 assertNotSame(samples, hprofData.getSamples());
184 assertEquals(1, samples.size());
185 Sample sample = samples.iterator().next();
246 Set<Sample> samples = hprofData.getSamples(); local
247 assertNotNull(samples);
248 for (Sample sample : samples) {
|
| /cts/apps/CtsVerifier/src/com/android/cts/verifier/audioquality/experiments/ |
| LoopbackExperiment.java | 65 int samples = playbackData.length / 2; local
66 int duration = (samples * 1000) / AudioQualityVerifierActivity.SAMPLE_RATE; // In ms
68 int totalSamples = samples + 2 * padSamples;
120 Recorder(byte[] buffer, int samples) {
122 mSamples = samples;
|
| /external/speex/include/speex/ |
| speex_echo.h | 73 * @param frame_size Number of samples to process at one time (should correspond to 10-20 ms)
74 * @param filter_length Number of samples of echo to cancel (should generally correspond to 100-500 ms)
80 * @param frame_size Number of samples to process at one time (should correspond to 10-20 ms)
81 * @param filter_length Number of samples of echo to cancel (should generally correspond to 100-500 ms)
145 * @param frame_size Size of the frame to process at ones (counting samples *per* channel)
|
| /external/srec/audio/AudioIn/UNIX/include/ |
| audioin.h | 78 samples.*/
80 audio device. The MMSystem is not providing any more samples, or the
83 be fixed by decreasing the number of samples that you want to retrieve or by
84 waiting till samples are available. A time out period is set internally on
90 AUDIOIN_FIFOOVERRUN, /* @emem The buffer has been overfilled with audio samples. You can still retrieve
91 samples from the FIFO with the lhs_audioinGetSamples function, but the audio-in
94 AUDIOIN_FLUSHED, /* @emem The buffer has been overfilled with audio samples. You can no longer retrieve
95 samples from the FIFO, since you already emptied it with lhs_audioinGetSamples.
99 AUDIOIN_HWOVERRUN, /* @emem The buffer has been overfilled with audio samples inside the device component
102 recommended to initialize the audio-in component before retrieving samples
[all...] |
| /external/webrtc/src/common_audio/signal_processing_library/main/source/ |
| resample_by_2_internal.c | 39 // lower allpass filter (operates on even input samples)
69 // upper allpass filter (operates on odd input samples)
134 // lower allpass filter (operates on even input samples)
164 // upper allpass filter (operates on odd input samples)
206 // upper allpass filter (generates odd output samples)
236 // lower allpass filter (generates even output samples)
276 // upper allpass filter (generates odd output samples)
306 // lower allpass filter (generates even output samples)
346 // upper allpass filter (generates odd output samples)
381 // lower allpass filter (generates even output samples)
[all...] |
| /frameworks/base/media/libeffects/lvm/lib/Bundle/src/ |
| LVM_Private.h | 65 #define LVM_FIRSTLASTCALL 3 /* Single call for small number of samples */
147 LVM_INT16 InDelaySamples; /* Number of samples in the input delay buffer */
150 LVM_INT16 OutDelaySamples; /* Number of samples in the output delay buffer, left and right */
151 LVM_INT16 SamplesToOutput; /* Samples to write to the output */
183 LVM_INT16 SamplesToProcess; /* Input samples left to process */
|
