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  /frameworks/wilhelm/src/desktop/
SDL.c 52 fmt.samples = SndFile_BUFSIZE;
54 fmt.samples = SndFile_BUFSIZE / sizeof(short);
  /ndk/build/tools/
gen-platforms.sh 110 --samples)
153 echo " --minimal Ignore samples, symlinks and generated shared libs."
155 echo " --samples Also generate samples directories."
280 # $3: description of directory contents (e.g. "sysroot" or "samples")
631 # Copy platform sysroot and samples into your destination
803 # $SRC/android-$PLATFORM/samples --> $DST/samples
806 # Copy platform samples and generic samples into your destinatio
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  /system/media/audio_utils/include/audio_utils/
channels.h 26 * in_buff points to the buffer of samples
28 * out_buff points to the buffer to receive converted samples.
  /external/chromium_org/third_party/webrtc/modules/audio_processing/agc/
analog_agc.c 114 int16_t 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 */
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  /external/webrtc/src/modules/audio_processing/agc/
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 */
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  /external/chromium_org/base/metrics/
histogram_base.h 54 typedef int32_t Sample; // Used for samples.
55 typedef subtle::Atomic32 AtomicCount; // Used to count samples.
123 virtual void AddSamples(const HistogramSamples& samples) = 0;
128 // does not serialize the samples.
131 // Try to find out data corruption from histogram and the samples.
133 virtual int FindCorruption(const HistogramSamples& samples) const;
  /external/chromium_org/native_client_sdk/src/examples/api/media_stream_audio/
media_stream_audio.cc 32 // This example demonstrates receiving audio samples from a
148 // Draw our samples.
170 uint32_t samples = buffer.GetNumberOfSamples() / channels; local
172 if (channel_count_ != channels || sample_count_ != samples) {
174 sample_count_ = samples;
177 // Try (+ 5) to ensure that we pick up a new set of samples between each
  /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface/
ilbc.h 129 * - len : Samples in speechIn (160, 240, 320 or 480)
168 * will be a multiple of 160 or 240 samples ((160 or 240)*frames/packet).
180 * Return value : >0 - Samples in decoded vector
204 * will be a multiple of 160 or 240 samples.
214 * Return value : >0 - Samples in decoded PLC vector
236 * Return value : >0 - Samples in decoded PLC vector
  /external/deqp/modules/gles3/functional/
es3fFlushFinishTests.cpp 142 void analyzeResults (const std::vector<Sample>& samples, const CalibrationParams& calibrationParams);
403 std::vector<Vec2> getPointsFromSamples (const std::vector<FlushFinishCase::Sample>& samples, const deUint64 FlushFinishCase::Sample::*field)
405 vector<Vec2> points(samples.size());
407 for (size_t ndx = 0; ndx < samples.size(); ndx++)
408 points[ndx] = Vec2(float(samples[ndx].numDrawCalls), float(samples[ndx].*field));
414 T getMaximumValue (const std::vector<FlushFinishCase::Sample>& samples, const T FlushFinishCase::Sample::*field)
416 DE_ASSERT(!samples.empty());
418 T maxVal = samples[0].*field;
420 for (size_t ndx = 1; ndx < samples.size(); ndx++
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  /external/guava/guava-testlib/src/com/google/common/collect/testing/testers/
ListSetTester.java 42 doTestSet(samples.e3);
61 doTestSet(samples.e3);
78 getList().set(-1, samples.e3);
89 getList().set(index, samples.e3);
100 getList().set(aValidIndex(), samples.e3);
111 getList().set(0, samples.e3);
ListSubListTester.java 101 subList.add(samples.e3);
102 expectAdded(0, samples.e3);
109 subList.set(0, samples.e3);
111 expected.set(0, samples.e3);
119 getList().set(0, samples.e3);
122 Collections.singletonList(samples.e3), subList);
129 subList.remove(samples.e2);
139 subList.add(0, samples.e3);
140 expectAdded(2, samples.e3);
147 subList.set(0, samples.e3)
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  /external/qemu/distrib/sdl-1.2.15/docs/man3/
SDL_OpenAudio.3 23 .IP "\fBdesired\fR->\fBfreq\fR" 10The desired audio frequency in samples-per-second\&.
25 .IP "\fBdesired\fR->\fBsamples\fR" 10The desired size of the audio buffer in samples\&. This number should be a power of two, and may be adjusted by the audio driver to a value more suitable for the hardware\&. Good values seem to range between 512 and 8192 inclusive, depending on the application and CPU speed\&. Smaller values yield faster response time, but can lead to underflow if the application is doing heavy processing and cannot fill the audio buffer in time\&. A stereo sample consists of both right and left channels in LR ordering\&. Note that the number of samples is directly related to time by the following formula: ms = (samples*1000)/freq
68 desired->samples=8192;
  /external/skia/tools/
bench_record.cpp 24 DEFINE_int32(samples, 10, "Number of times to re-record each SKP.");
34 "2: print all samples");
94 SkAutoTMalloc<double> samples(FLAGS_samples);
101 samples[i] = timer.fWall * scale / loops;
104 Stats stats(samples.get(), FLAGS_samples);
114 printf("\t%g", samples[i]);
  /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/qemu/audio/
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"
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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)
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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);
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  /external/chromium_org/third_party/webrtc/common_audio/resampler/
resampler.cc 430 // Synchronous resampling, all output samples are written to samplesOut
491 // Containers for temp samples
513 // We can only handle blocks of 160 samples
549 // We can only handle blocks of 80 samples
580 // We can only handle blocks of 40 samples
600 // WebRtcSpl_Resample16khzTo48khz() takes a block of 160 samples
601 // as input and outputs a resampled block of 480 samples. The
620 // We can only handle blocks of 160 samples
643 // We can only handle blocks of 80 samples
672 // We can only handle blocks of 80 samples
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  /external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/builder/
FragmentedMp4Builder.java 82 * @param intersectionMap a map from tracks to their fragments' first samples.
199 final List<ByteBuffer> samples = ByteBufferHelper.mergeAdjacentBuffers(bbs);
211 for (int i = 0; i < Math.ceil((double) samples.size() / STEPSIZE); i++) {
212 List<ByteBuffer> sublist = samples.subList(
214 (i + 1) * STEPSIZE < samples.size() ? (i + 1) * STEPSIZE : samples.size()); // end
222 for (ByteBuffer sample : samples) {
266 * Gets the all samples starting with <code>startSample</code> (one based -> one is the first) and
271 * @param track source of the samples
272 * @param sequenceNumber the fragment index of the requested list of samples
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  /external/webrtc/src/common_audio/resampler/
resampler.cc 430 // Synchronous resampling, all output samples are written to samplesOut
491 // Containers for temp samples
513 // We can only handle blocks of 160 samples
549 // We can only handle blocks of 80 samples
580 // We can only handle blocks of 40 samples
600 // WebRtcSpl_Resample16khzTo48khz() takes a block of 160 samples
601 // as input and outputs a resampled block of 480 samples. The
620 // We can only handle blocks of 160 samples
643 // We can only handle blocks of 80 samples
672 // We can only handle blocks of 80 samples
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  /external/guava/guava-testlib/src/com/google/common/collect/testing/
NavigableMapTestSuiteBuilder.java 122 // derive values for inclusive filtering from the input samples
123 SampleElements<Entry<K, V>> samples = delegate.samples(); local
126 samples.e0, samples.e1, samples.e2, samples.e3, samples.e4);
253 public SampleElements<Entry<K, V>> samples() {
254 return delegate.samples();
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  /external/libhevc/common/
ihevc_intra_pred_filters.c 96 * Reference substitution process for samples unavailable for prediction
155 /* Else fill the corresponding samples */
182 /* Else fill the corresponding samples */
231 /* If no neighbor flags are present, fill the neighbor samples with DC value */
250 /* Else fill the corresponding samples */
361 /* Else fill the corresponding samples */
472 /* Else fill the corresponding samples */
585 * Reference DC filtering for neighboring samples dependent on TU size and
658 /* Strong filtering of reference samples */
672 /* Perform bilinear filtering of Reference Samples */
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  /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 *
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  /external/bluetooth/bluedroid/bta/av/
bta_av_sbc.c 38 UINT32 src_sps; /* samples per second (source audio data) */
39 UINT32 dst_sps; /* samples per second (converted audio data) */
56 ** src_sps: samples per second (source audio data)
57 ** dst_sps: samples per second (converted audio data)
107 ** source speed (src_sps, samples per second),
112 ** src_samples: The number of source samples (number of bytes)
152 ** source speed (src_sps, samples per second),
157 ** src_samples: The number of source samples (in uint of 4 bytes)
215 ** source speed (src_sps, samples per second),
220 ** src_samples: The number of source samples (in uint of 2 bytes
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  /external/chromium_org/chrome/common/extensions/docs/server2/
known_broken_links.json 47 "extensions/samples.html#debugger",
53 "extensions/samples.html#privacy",
65 "extensions/samples.html#contextMenus",
71 "extensions/samples.html#webrequest",
77 "extensions/samples.html#browsingData",
83 "extensions/samples.html#chrome_url_overrides",
107 "extensions/samples.html#devtools.audits",
191 "apps/samples.html#contextMenus",
257 "extensions/samples.html#experimental",
262 "extensions/samples.html"
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