| /cts/tests/tests/media/src/android/media/cts/ |
| SoundPoolTest.java | 147 float rate = 1f; local
148 int streamID = mSoundPool.play(sampleID, leftVolume, rightVolume, priority, loop, rate);
151 rate = 1.4f;
152 mSoundPool.setRate(streamID, rate);
162 streamID = mSoundPool.play(sampleID, leftVolume, rightVolume, priority, loop, rate);
|
| /external/chromium_org/chromeos/ime/ |
| xkeyboard.cc | 345 bool XKeyboard::SetAutoRepeatRate(const AutoRepeatRate& rate) {
346 DVLOG(1) << "Set auto-repeat rate to: "
347 << rate.initial_delay_in_ms << " ms delay, "
348 << rate.repeat_interval_in_ms << " ms interval";
350 rate.initial_delay_in_ms,
351 rate.repeat_interval_in_ms) != True) {
352 DVLOG(1) << "Failed to set auto-repeat rate";
|
| /external/iproute2/tc/ |
| q_red.c | 42 unsigned rate = 0; local
89 if (get_rate(&rate, *argv)) {
112 if (rate == 0)
113 get_rate(&rate, "10Mbit");
141 if ((wlog = tc_red_eval_idle_damping(opt.Wlog, avpkt, rate, sbuf)) < 0) {
|
| tc_util.h | 64 extern int get_rate(unsigned *rate, const char *str);
70 extern void print_rate(char *buf, int len, __u32 rate);
75 extern char * sprint_rate(__u32 rate, char *buf);
|
| q_hfsc.c | 54 "SC := [ [ umax BYTE ] dmax SEC ] rate BPS\n"
58 " rate : rate\n"
340 unsigned int umax = 0, dmax = 0, rate = 0; local
360 if (matches(*argv, "rate") == 0) {
362 if (get_rate(&rate, *argv) < 0) {
363 explain1("rate");
374 if (dmax != 0 && ceil(1.0 * umax * TIME_UNITS_PER_SEC / dmax) > rate) {
381 sc->m2 = rate;
385 * is at dmax - umax / rate
[all...] |
| q_cbq.c | 29 fprintf(stderr, "Usage: ... cbq bandwidth BPS rate BPS maxburst PKTS [ avpkt BYTES ]\n");
31 fprintf(stderr, " [ allot BYTES ] [ mpu BYTES ] [ weight RATE ]\n");
67 matches(*argv, "rate") == 0) {
69 if (get_rate(&r.rate, *argv)) {
141 if (r.rate == 0) {
155 fprintf(stderr, "CBQ: failed to calculate rate table.\n");
162 lss.maxidle = tc_calc_xmittime(r.rate, avpkt);
206 if (matches(*argv, "rate") == 0) {
208 if (get_rate(&r.rate, *argv)) {
209 explain1("rate");
[all...] |
| q_choke.c | 41 unsigned rate = 0; local
59 if (get_rate(&rate, *argv)) {
106 if (!rate || !opt.limit) {
148 wlog = tc_red_eval_idle_damping(opt.Wlog, avpkt, rate, sbuf);
|
| /external/tinycompress/ |
| cplay.c | 209 unsigned int channels, rate, bits; local
221 if (parse_mp3_header(&header, &channels, &rate, &bits) == -1) {
229 codec.sample_rate = compress_get_alsa_rate(rate);
231 fprintf(stderr, "invalid sample rate %d\n", rate);
285 printf("Format %u Channels %u, %u Hz, Bit Rate %d\n",
286 SND_AUDIOCODEC_MP3, channels, rate, bits);
|
| /frameworks/av/media/libmedia/ |
| SoundPool.cpp | 238 int priority, int loop, float rate)
240 ALOGV("play sampleID=%d, leftVolume=%f, rightVolume=%f, priority=%d, loop=%d, rate=%f",
241 sampleID, leftVolume, rightVolume, priority, loop, rate);
272 channel->play(sample, channelID, leftVolume, rightVolume, priority, loop, rate);
399 void SoundPool::setRate(int channelID, float rate)
401 ALOGV("setRate(%d, %f)", channelID, rate);
405 channel->setRate(rate);
515 ALOGE("Sample rate (%u) out of range", sampleRate);
546 float rightVolume, int priority, int loop, float rate)
556 " priority=%d, loop=%d, rate=%f"
658 float rate; local
[all...] |
| /hardware/invensense/60xx/mlsdk/mllite/ |
| mlBiasNoMotion.c | 158 long rate; local
171 // We always run the accel low pass filter at the highest sample rate possible
178 rate = inv_get_fifo_rate() * 5 + 5;
179 if (rate > 200)
180 rate = 200;
182 gain = inv_obj->accel_lpf_gain * 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...] |
| backends.h | 43 long rate; member in struct:__anon23832
|
| /external/chromium_org/third_party/opus/src/celt/ |
| opus_custom_demo.c | 51 opus_int32 frame_size, channels, rate; local
64 fprintf (stderr, "Usage: test_opus_custom <rate> <channels> <frame size> "
65 " <bytes per packet> [<complexity> [packet loss rate]] "
70 rate = (opus_int32)atol(argv[1]);
73 mode = opus_custom_mode_create(rate, frame_size, NULL);
|
| /external/libvpx/libvpx/vp8/encoder/ |
| encodemb.c | 204 int rate; member in struct:vp8_token_state
283 tokens[eob][0].rate = 0;
305 rate0 = tokens[next][0].rate;
306 rate1 = tokens[next][1].rate;
330 tokens[i][0].rate = base_bits + (best ? rate1 : rate0);
337 rate0 = tokens[next][0].rate;
338 rate1 = tokens[next][1].rate;
400 tokens[i][1].rate = base_bits + (best ? rate1 : rate0);
420 tokens[next][0].rate += mb->token_costs[type][band][0][t0];
425 tokens[next][1].rate += mb->token_costs[type][band][0][t1]
[all...] |
| /device/asus/grouper/audio/ |
| audio_hw.c | 72 .rate = OUT_SAMPLING_RATE,
81 .rate = IN_SAMPLING_RATE,
91 .rate = SCO_SAMPLING_RATE,
262 * Due to the lack of sample rate converters in the SoC,
281 * the most common rate, but group 2 is required for SCO.
286 if (((out->pcm_config->rate % 8000 == 0) &&
287 (in->pcm_config->rate % 8000) != 0) ||
288 ((out->pcm_config->rate % 11025 == 0) &&
289 (in->pcm_config->rate % 11025) != 0))
303 * If the stream rate differs from the PCM rate, we need t
[all...] |
| /external/iptables/extensions/ |
| libxt_rateest.c | 18 " --rateest1 name Rate estimator name\n"
19 " --rateest2 name Rate estimator name\n"
20 " --rateest-delta Compare difference(s) to given rate(s)\n"
25 " [!] --rateest-lt Match if rate is less than given rate/estimator\n"
26 " [!] --rateest-gt Match if rate is greater than given rate/estimator\n"
27 " [!] --rateest-eq Match if rate is equal to given rate/estimator\n");
87 rateest_get_rate(uint32_t *rate, const char *str
[all...] |
| /external/chromium/chrome/browser/extensions/ |
| extension_tts_api.h | 24 // will be empty strings, and rate, pitch, and volume will be -1.0.
34 double rate,
86 double rate() { return rate_; } function in class:Utterance
|
| /external/chromium_org/chrome/common/extensions/docs/examples/extensions/ttsdemo/ |
| ttsdemo.js | 56 var rateValue = Number(rate.value);
58 options.rate = rateValue;
|
| /external/chromium_org/media/audio/alsa/ |
| alsa_wrapper.h | 44 unsigned int rate, int soft_resample,
79 unsigned int channels, unsigned int rate,
|
| /external/chromium_org/media/audio/sounds/ |
| sounds_manager.cc | 78 const int64 rate = wav_audio.byte_rate(); local
79 return base::TimeDelta::FromMicroseconds(size * 1000000 / rate);
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/main/interface/ |
| isac.h | 35 * - samplingRate : sampling rate of the input/output audio.
56 * - samplingRate : sampling rate of the input/output audio.
109 * - CodingMode : 0 -> Bit rate and frame length are
114 * rate which is taken as the maximum
115 * short-term average bit rate.
131 * rate, or 320 if operating at 32 kHz sampling rate. The encoder buffers the
203 * This function decodes an ISAC frame. At 16 kHz sampling rate, the length
205 * 30 or 60 ms respectively. At 32 kHz sampling rate, the length of the
256 * This function sets the limit on the short-term average bit-rate and th
[all...] |
| /frameworks/base/core/java/android/view/ |
| OrientationEventListener.java | 63 * @param rate at which sensor events are processed (see also
68 public OrientationEventListener(Context context, int rate) {
70 mRate = rate;
|
| /cts/suite/audio_quality/lib/src/audio/ |
| AudioRecordingLocal.cpp | 49 config.rate = samplingRate;
|
| /external/chromium/chrome/browser/chromeos/input_method/ |
| xkeyboard.h | 57 // Sets the auto-repeat rate of the keyboard, initial delay in ms, and repeat
59 bool SetAutoRepeatRate(const AutoRepeatRate& rate);
|
| /external/libvorbis/examples/ |
| vorbisfile_example.c | 61 fprintf(stderr,"\nBitstream is %d channel, %ldHz\n",vi->channels,vi->rate);
81 /* we don't bother dealing with sample rate changes, etc, but
|
