/frameworks/wilhelm/src/desktop/ |
SDL.c | 52 fmt.samples = SndFile_BUFSIZE; 54 fmt.samples = SndFile_BUFSIZE / sizeof(short);
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/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 [all...] |
/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.
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/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 */ [all...] |
/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 */ [all...] |
/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;
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/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
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/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
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/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++ [all...] |
/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);
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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) [all...] |
/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;
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/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]);
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/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
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/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" [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/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 [all...] |
/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 [all...] |
/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 [all...] |
/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(); [all...] |
/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 */ [all...] |
/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...] |
/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 [all...] |
/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" [all...] |