| /external/libopus/silk/ |
| main.h | 52 opus_int fs_kHz, /* I Sample rate (kHz) */
62 opus_int fs_kHz, /* I Samples rate (kHz) */
131 /* Control internal sampling rate */
217 opus_int32 *rate_dist_Q14, /* O best weighted quant error + mu * rate */
224 const opus_int mu_Q9, /* I tradeoff betw. weighted error and rate */
246 const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */
265 const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */
308 const opus_int NLSF_mu_Q20, /* I Rate weight for the RD optimization */
360 /* Set decoder sampling rate */
|
| /external/libvorbis/lib/ |
| vorbisenc.c | 272 g->coupling_pointlimit[0][i]=kHz*1000./vi->rate*ci->blocksizes[0];
273 g->coupling_pointlimit[1][i]=kHz*1000./vi->rate*ci->blocksizes[1];
277 g->sliding_lowpass[0][i]=kHz*1000./vi->rate*ci->blocksizes[0];
278 g->sliding_lowpass[1][i]=kHz*1000./vi->rate*ci->blocksizes[1];
284 g->coupling_pointlimit[0][i]=kHz*1000./vi->rate*ci->blocksizes[0];
285 g->coupling_pointlimit[1][i]=kHz*1000./vi->rate*ci->blocksizes[1];
291 g->sliding_lowpass[0][i]=kHz*1000./vi->rate*ci->blocksizes[0];
292 g->sliding_lowpass[1][i]=kHz*1000./vi->rate*ci->blocksizes[1];
521 double nyq=vi->rate/2.;
863 long rate){
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| neteq_quality_test.cc | 51 "Filename for input audio (specify sample rate with --input_sample_rate ,"
57 // Define switch for sample rate.
61 printf("Invalid sample rate should be 8000, 16000, 32000 or 48000 Hz.");
65 DEFINE_int32(input_sample_rate, 16000, "Sample rate of input file in Hz.");
99 // Define switch for packet loss rate.
168 // to achieve the target packet loss rate |loss_rate|, when a packet is not
302 // |unit_loss_rate| is the packet loss rate for each unit time interval
321 // To guarantee the overall packet loss rate, transition probabilities
430 Log() << "Average bit rate was "
|
| /external/webrtc/webrtc/voice_engine/ |
| voice_engine_defines.h | 30 // rate voice engine supports (10 ms of audio at 192 kHz).
77 // Min init target rate for iSAC-wb
79 // Max init target rate for iSAC-wb
81 // Min init target rate for iSAC-swb
83 // Max init target rate for iSAC-swb
85 // Lowest max rate for iSAC-wb
87 // Highest max rate for iSAC-wb
89 // Lowest max rate for iSAC-swb
91 // Highest max rate for iSAC-swb
|
| /frameworks/base/docs/html/ndk/guides/audio/ |
| output-latency.jd | 91 sample rate and buffer size. For more information, see
95 <h3>Obtain the optimal sample rate</h3>
97 <p>In Java, you can obtain the optimal sample rate from AudioManager as shown in the following
108 <strong>Note:</strong> The sample rate refers to the rate of each stream. If your source audio
114 <h3>Use the optimal sample rate when creating your audio player</h3>
116 <p>Once you have the optimal sample output rate, you can supply it when creating your player
134 <strong>Note:</strong> {@code samplesPerSec} refers to the <em>sample rate per channel in
272 <p>You can obtain the optimal buffer size in a similar way to the optimal frame rate, using the
|
| /frameworks/base/media/java/android/media/ |
| AudioFormat.java | 44 * <li><a href="#sampleRate">sample rate</a>
52 * <h4 id="sampleRate">Sample rate</h4>
53 * <p>Expressed in Hz, the sample rate in an <code>AudioFormat</code> instance expresses the number
55 * not the sample rate
56 * at which content is rendered or produced. For instance a sound at a media sample rate of 8000Hz
57 * can be played on a device operating at a sample rate of 48000Hz; the sample rate conversion is
62 * for <code>AudioRecord</code> and <code>AudioTrack</code>, with sample rate conversion
64 * To improve efficiency and avoid lossy conversions, it is recommended to match the sample rate
66 * sample rate, and limit the sample rate to no more than 48kHz unless there are specia
[all...] |
| /hardware/intel/common/omx-components/videocodec/libvpx_internal/libvpx/vp9/encoder/ |
| vp9_aq_cyclicrefresh.c | 42 // Projected rate and distortion for the current superblock.
45 // Thresholds applied to projected rate/distortion of the superblock.
93 // mode, and rate/distortion.
98 // If projected rate is below the thresh_rate (well below target,
115 // Rate/distortion not used for update.
201 // Rate target ratio to set q delta.
209 // Set rate threshold to some fraction of target (and scaled by 256).
216 // as rate/distortion are derived from model based on prediction residual.
247 // TODO(marpan): Incorporate the actual-vs-target rate over/undershoot from
|
| /hardware/ti/omap3/omx/video/src/openmax_il/video_encode/inc/ |
| OMX_VideoEnc_DSP.h | 216 OMX_U32 refFrameRate; /* Reference or input frame rate*1000 */
217 OMX_U32 targetFrameRate; /* Target frame rate * 1000 */
218 OMX_U32 targetBitRate; /* Target bit rate in bits per second */
255 OMX_U32 perceptualQuant ; /* Enable Perceptual Quantization a.k.a. Perceptual Rate Control*/
271 OMX_U32 sliceGroupChangeRate ; /* Slice Group Change Rate: Only valid when sliceGroupMapType
405 * IVIDEO_DEFAULT => Default rate control of encoder
406 * IVIDEO_LOW_DELAY => CBR rate control for video conferencing
407 * IVIDEO_STORAGE => VBR rate control for local storage (DVD) recording
408 * IVIDEO_TWOPASS => two pass rate control for non real time applications
|
| /prebuilts/gcc/linux-x86/host/x86_64-w64-mingw32-4.8/x86_64-w64-mingw32/include/ |
| tuner.h | 228 STDMETHOD_(HRESULT,get_SymbolRate)(THIS_ __LONG32 *Rate) PURE;
235 STDMETHOD_(HRESULT,put_SymbolRate)(THIS_ __LONG32 Rate) PURE;
254 #define ILocator_get_SymbolRate(This,Rate) (This)->lpVtbl->get_SymbolRate(This,Rate)
261 #define ILocator_put_SymbolRate(This,Rate) (This)->lpVtbl->put_SymbolRate(This,Rate)
294 STDMETHOD_(HRESULT,get_SymbolRate)(THIS_ __LONG32 *Rate) PURE;
301 STDMETHOD_(HRESULT,put_SymbolRate)(THIS_ __LONG32 Rate) PURE;
322 #define IDigitalLocator_get_SymbolRate(This,Rate) (This)->lpVtbl->get_SymbolRate(This,Rate)
[all...] |
| /system/extras/simpleperf/ |
| cmd_stat.cpp | 332 return android::base::StringPrintf("%lf%% miss rate", miss_rate * 100);
336 double rate = summary.count / duration_in_sec; local
337 if (rate > 1e9) {
338 return android::base::StringPrintf("%.3lf G/sec", rate / 1e9);
340 if (rate > 1e6) {
341 return android::base::StringPrintf("%.3lf M/sec", rate / 1e6);
343 if (rate > 1e3) {
344 return android::base::StringPrintf("%.3lf K/sec", rate / 1e3);
346 return android::base::StringPrintf("%.3lf /sec", rate);
|
| /external/autotest/client/cros/chameleon/ |
| audio_widget.py | 66 rate: sampling rate.
200 """Gets sampling rate of recorded data."""
201 return self._rec_format['rate']
387 rate: sampling rate.
519 rate=48000) variable in class:ChameleonLineOutOutputWidgetHandler
676 rate=48000) variable in class:CrosInputWidgetHandler
694 rate: sampling rate
735 rate=48000) variable in class:CrosUSBInputWidgetHandler
751 rate=48000) variable in class:CrosOutputWidgetHandler
[all...] |
| /hardware/qcom/audio/msm8909/hal/audio_extn/ |
| usb.c | 85 .rate = USB_DEFAULT_OUTPUT_SAMPLING_RATE,
259 ALOGE("%s: error unable to allocate memory to hold sample rate strings",
266 ALOGE("%s: error unable to allocate memory to hold sample rate string",
279 ALOGE("%s: error could not get rate size, returning", __func__);
294 ALOGE("%s: error could not get first rate val", __func__);
315 /* Sample Rate should be one of the proxy supported rates only
359 pcm_config_usbmod.rate = usbmod->sample_rate_playback;
362 pcm_config_usbmod.rate);
382 pcm_config_usbmod.rate = usbmod->sample_rate_playback;
388 pcm_config_usbmod.rate);
[all...] |
| /external/chromium-trace/catapult/telemetry/third_party/pyserial/serial/ |
| serialposix.py | 59 raise ValueError('Failed to set custom baud rate (%s): %s' % (baudrate, e))
101 raise ValueError("sorry don't know how to handle non standard baud rate on this platform")
123 raise ValueError("sorry don't know how to handle non standard baud rate on this platform")
134 raise ValueError("sorry don't know how to handle non standard baud rate on this platform")
151 raise ValueError("baud rate not supported")
165 raise ValueError("sorry don't know how to handle non standard baud rate on this platform")
175 raise ValueError("sorry don't know how to handle non standard baud rate on this platform")
185 raise ValueError("sorry don't know how to handle non standard baud rate on this platform")
195 raise ValueError("sorry don't know how to handle non standard baud rate on this platform")
205 raise ValueError("sorry don't know how to handle non standard baud rate on this platform"
[all...] |
| /hardware/ti/omap3/omx/system/src/openmax_il/perf/src/ |
| perf_rt.c | 76 /* allocate rate tracking structures */
146 /* VP_T - display rate */
151 /* VD_T - decode rate */
156 /* VE_T - encode rate */
161 /* CAMT - capture rate */
320 /* get the temporal rate */
334 /* calculate the average of the temporal rate */
405 /* for each rate, reset skip count as well as tn0 */
423 /* complete any temporary rate measurements */
459 /* see if we care about this buffer in the rate calculation *
470 int rate = (me->detailed == 2) || local
[all...] |
| /cts/tests/tests/media/src/android/media/cts/ |
| SoundPoolTest.java | 150 float rate = 1f; local
151 int streamID = mSoundPool.play(sampleID, leftVolume, rightVolume, priority, loop, rate);
154 rate = 1.4f;
155 mSoundPool.setRate(streamID, rate);
165 streamID = mSoundPool.play(sampleID, leftVolume, rightVolume, priority, loop, rate);
307 -1 /* loop (infinite) */, 1.0f /* rate */);
|
| /development/ndk/platforms/android-13/include/android/ |
| tts.h | 35 #define ANDROID_TTS_ENGINE_PROPERTY_RATE "rate"
107 // @param [in] uint32_t - Track sampling rate in Hz
225 // @param[inout] rate in: the desired audio sample rate
226 // out: the sample rate used by the TTS engine
290 // o rate,
|
| /external/svox/pico/compat/jni/ |
| tts.h | 33 #define ANDROID_TTS_ENGINE_PROPERTY_RATE "rate"
108 // @param [in] uint32_t - Track sampling rate in Hz
226 // @param[inout] rate in: the desired audio sample rate
227 // out: the sample rate used by the TTS engine
291 // o rate,
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/include/ |
| neteq.h | 33 uint16_t packet_loss_rate; // Loss rate (network + late) in Q14.
34 uint16_t packet_discard_rate; // Late loss rate in Q14.
147 // the same tick rate as the RTP timestamp of the current payload.
188 // produce samples at the rate |sample_rate_hz|. Returns kOK on success, kFail
261 // Returns the sample rate in Hz of the audio produced in the last GetAudio
262 // call. If GetAudio has not been called yet, the configured sample rate
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_receiver_impl.cc | 100 const uint32_t rate) {
108 payload_name, payload_type, frequency, channels, rate,
256 uint32_t rate = 0; local
287 rate = payload->typeSpecific.Audio.rate;
305 rtp_header.payload_type_frequency, channels, rate)) {
|
| /external/webrtc/webrtc/modules/video_coding/ |
| video_sender.cc | 70 // Force an encoder parameters update, so that incoming frame rate is
171 // Get encode frame rate
204 // No frame rate estimate available, use default.
221 // average frame rate and bit rate.
230 // requested FEC rate and NACK status (on/off).
|
| /frameworks/av/media/libmedia/ |
| Visualizer.cpp | 98 uint32_t rate, bool force)
100 if (rate > CAPTURE_RATE_MAX) {
119 mCaptureRate = rate;
122 mCaptureThread = new CaptureThread(*this, rate, ((flags & CAPTURE_CALL_JAVA) != 0));
124 ALOGV("setCaptureCallBack() rate: %d thread %p flags 0x%08x",
125 rate, mCaptureThread.get(), mCaptureFlags);
|
| /hardware/qcom/audio/legacy/alsa_sound/ |
| ALSAStreamOps.cpp | 96 uint32_t *rate,
134 if (rate && *rate > 0) {
135 if (mHandle->sampleRate != *rate)
137 } else if (rate) {
138 *rate = mHandle->sampleRate;
|
| /prebuilts/ndk/current/platforms/android-13/arch-arm/usr/include/android/ |
| tts.h | 35 #define ANDROID_TTS_ENGINE_PROPERTY_RATE "rate"
107 // @param [in] uint32_t - Track sampling rate in Hz
225 // @param[inout] rate in: the desired audio sample rate
226 // out: the sample rate used by the TTS engine
290 // o rate,
|
| /prebuilts/ndk/current/platforms/android-13/arch-mips/usr/include/android/ |
| tts.h | 35 #define ANDROID_TTS_ENGINE_PROPERTY_RATE "rate"
107 // @param [in] uint32_t - Track sampling rate in Hz
225 // @param[inout] rate in: the desired audio sample rate
226 // out: the sample rate used by the TTS engine
290 // o rate,
|
| /prebuilts/ndk/current/platforms/android-13/arch-x86/usr/include/android/ |
| tts.h | 35 #define ANDROID_TTS_ENGINE_PROPERTY_RATE "rate"
107 // @param [in] uint32_t - Track sampling rate in Hz
225 // @param[inout] rate in: the desired audio sample rate
226 // out: the sample rate used by the TTS engine
290 // o rate,
|
