| /frameworks/wilhelm/src/itf/ |
| IPlaybackRate.c | 22 static SLresult IPlaybackRate_SetRate(SLPlaybackRateItf self, SLpermille rate)
28 if (!(thiz->mMinRate <= rate && rate <= thiz->mMaxRate)) {
36 result = android_audioPlayer_setPlaybackRateAndConstraints(ap, rate, thiz->mProperties);
44 thiz->mRate = rate;
62 SLpermille rate = thiz->mRate; local
64 *pRate = rate;
128 SLpermille rate, SLuint32 *pCapabilities)
138 if (!(thiz->mMinRate <= rate && rate <= thiz->mMaxRate))
[all...] |
| /external/chromium/chrome/browser/extensions/ |
| extension_tts_api_mac.mm | 23 double rate,
54 double rate,
60 if (rate >= 0.0) {
61 // The TTS api defines rate via words per minute.
63 setObject:[NSNumber numberWithInt:rate * 400]
|
| /frameworks/base/tests/RenderScriptTests/Fountain/src/com/example/android/rs/fountain/ |
| FountainRS.java | 60 int rate = (int)(pressure * pressure * 500.f); local
61 if (rate > 500) {
62 rate = 500;
64 if (rate > 0) {
65 mScript.invoke_addParticles(rate, x, y, id, !holdingColor[id]);
|
| /frameworks/base/tests/RenderScriptTests/Fountain_v11/src/com/android/fountain/ |
| FountainRS.java | 60 int rate = (int)(pressure * pressure * 500.f); local
61 if (rate > 500) {
62 rate = 500;
64 if (rate > 0) {
65 mScript.invoke_addParticles(rate, x, y, id, !holdingColor[id]);
|
| /external/iproute2/man/man8/ |
| tc-htb.8 | 21 .B ] htb rate
22 rate
24 rate
115 rate rate
116 Maximum rate this class and all its children are guaranteed. Mandatory.
119 ceil rate
120 Maximum rate at which a class can send, if its parent has bandwidth to spare.
121 Defaults to the configured rate, which implies no borrowing
128 .B rate.
[all...] |
| tc-tbf.8 | 5 .B tc qdisc ... tbf rate
6 rate
16 rate
30 itself, although packets are available, to ensure that the configured rate is not exceeded.
36 case, data is on average dequeued at the configured rate but may be sent much faster at millisecond
51 in one go. Tokens arrive at a steady rate, until the bucket is full.
77 bucket, the rate and possibly the peakrate (if set). These two parameters
84 if you want to reach your configured rate!
87 The minimum buffer size can be calculated by dividing the rate by HZ.
100 rate
[all...] |
| /frameworks/av/media/libeffects/lvm/lib/StereoWidening/src/ |
| LVCS_Headphone_Coeffs.h | 28 /* Stereo Enhancer coefficients for 8000 Hz sample rate, scaled with 0.161258 */
42 /* Stereo Enhancer coefficients for 11025Hz sample rate, scaled with 0.162943 */
56 /* Stereo Enhancer coefficients for 12000Hz sample rate, scaled with 0.162191 */
70 /* Stereo Enhancer coefficients for 16000Hz sample rate, scaled with 0.162371 */
84 /* Stereo Enhancer coefficients for 22050Hz sample rate, scaled with 0.160781 */
98 /* Stereo Enhancer coefficients for 24000Hz sample rate, scaled with 0.161882 */
112 /* Stereo Enhancer coefficients for 32000Hz sample rate, scaled with 0.160322 */
126 /* Stereo Enhancer coefficients for 44100Hz sample rate, scaled with 0.163834 */
140 /* Stereo Enhancer coefficients for 48000Hz sample rate, scaled with 0.164402 */
162 #define LVCS_STEREODELAY_CS_8KHZ 93 /* Sample rate 8kS/s *
[all...] |
| /prebuilts/python/darwin-x86/2.7.5/lib/python2.7/test/ |
| test_linuxaudiodev.py | 35 size, enc, rate, nchannels, extra = sunaudio.gethdr(fp)
52 self.dev.setparameters(rate, 16, nchannels, fmt)
59 rate = 8000
64 self.assertEqual(err.args[0], "expected rate >= 0, not -1")
66 self.dev.setparameters(rate, -2, nchannels, fmt)
70 self.dev.setparameters(rate, size, 3, fmt)
74 self.dev.setparameters(rate, size, nchannels, 177)
78 self.dev.setparameters(rate, size, nchannels, linuxaudiodev.AFMT_U16_LE)
83 self.dev.setparameters(rate, 16, nchannels, fmt)
|
| /prebuilts/python/linux-x86/2.7.5/lib/python2.7/test/ |
| test_linuxaudiodev.py | 35 size, enc, rate, nchannels, extra = sunaudio.gethdr(fp)
52 self.dev.setparameters(rate, 16, nchannels, fmt)
59 rate = 8000
64 self.assertEqual(err.args[0], "expected rate >= 0, not -1")
66 self.dev.setparameters(rate, -2, nchannels, fmt)
70 self.dev.setparameters(rate, size, 3, fmt)
74 self.dev.setparameters(rate, size, nchannels, 177)
78 self.dev.setparameters(rate, size, nchannels, linuxaudiodev.AFMT_U16_LE)
83 self.dev.setparameters(rate, 16, nchannels, fmt)
|
| /external/iproute2/misc/ |
| ifstat.c | 60 double rate[MAXS]; member in struct:ifstat_ent
110 memset(&n->rate, 0, sizeof(n->rate));
184 unsigned rate; local
195 if (sscanf(p, "%u", &rate) != 1)
197 n->rate[i] = rate;
221 double *rates = n->rate;
229 rates = h1->rate;
292 fprintf(fp, "%8s/%-6s ", "RX Pkts", "Rate");
[all...] |
| /external/chromium_org/chrome/common/extensions/docs/examples/api/ttsEngine/console_tts_engine/ |
| console_tts_engine.js | 12 var properties = ['voiceName', 'lang', 'gender', 'rate', 'pitch', 'volume'];
35 getTtsElement("rate").innerHTML = curOptions.rate;
84 // Fastest timeout == 1 ms (@ options.rate = 10.0)
85 milliseconds = 10 / curOptions.rate;
95 milliseconds = 10 / curOptions.rate;
|
| console_tts_engine.html | 33 <th>Rate</th>
41 <td id="rate"></td>
|
| /prebuilts/gcc/linux-x86/host/i686-linux-glibc2.7-4.4.3/sysroot/usr/include/alsa/ |
| pcm_rate.h | 3 * \brief External Rate-Converter-Plugin SDK
7 * External Rate-Converter-Plugin SDK
11 * ALSA external PCM rate-converter plugin SDK (draft version)
46 unsigned int rate; member in struct:snd_pcm_rate_side_info
58 /** Callback table of rate-converter */
108 * Define the object entry for external PCM rate-converter plugins
|
| /prebuilts/gcc/linux-x86/host/i686-linux-glibc2.7-4.6/sysroot/usr/include/alsa/ |
| pcm_rate.h | 3 * \brief External Rate-Converter-Plugin SDK
7 * External Rate-Converter-Plugin SDK
11 * ALSA external PCM rate-converter plugin SDK (draft version)
46 unsigned int rate; member in struct:snd_pcm_rate_side_info
58 /** Callback table of rate-converter */
108 * Define the object entry for external PCM rate-converter plugins
|
| /prebuilts/gcc/linux-x86/host/x86_64-linux-glibc2.7-4.6/sysroot/usr/include/alsa/ |
| pcm_rate.h | 3 * \brief External Rate-Converter-Plugin SDK
7 * External Rate-Converter-Plugin SDK
11 * ALSA external PCM rate-converter plugin SDK (draft version)
46 unsigned int rate; member in struct:snd_pcm_rate_side_info
58 /** Callback table of rate-converter */
108 * Define the object entry for external PCM rate-converter plugins
|
| /external/chromium/third_party/libjingle/source/talk/xmpp/ |
| ratelimitmanager.h | 43 // RateLimitManager imposes client-side rate limiting for xmpp tasks and
48 // to occur, it can check its rate limit with a call to VerifyRateLimit.
51 // appropriate rate limits. Else, it will return false.
66 // Checks if the event is under the defined rate limit and updates the
67 // rate limit if so. Returns true if it's under the rate limit.
71 // Checks if the event is under the defined rate limit and updates the
72 // rate limit if so *or* if always_update = true.
98 // Updates time and counter for rate limit
118 int per_x_seconds_; // interval size for rate limi
[all...] |
| /external/chromium_org/remoting/codec/ |
| audio_encoder_opus_unittest.cc | 27 // The sampling rate that OPUS uses internally and that we expect to get
55 AudioPacket::SamplingRate rate,
59 double angle = pos * 2 * M_PI * frequency_hz / rate +
68 AudioPacket::SamplingRate rate,
73 data[i * kChannels] = GetSampleValue(rate, frequency_hz, i + pos, 0);
74 data[i * kChannels + 1] = GetSampleValue(rate, frequency_hz, i + pos, 1);
81 packet->set_sampling_rate(rate);
107 AudioPacket::SamplingRate rate,
115 GetSampleValue(rate, frequency_hz, i - shift, 0);
118 GetSampleValue(rate, frequency_hz, i - shift, 1)
[all...] |
| /external/chromium_org/third_party/WebKit/Tools/Scripts/webkitpy/thirdparty/coverage/ |
| xmlreport.py | 10 def rate(hit, num): function
73 xpackage.setAttribute("line-rate", rate(lhits, lnum))
74 xpackage.setAttribute("branch-rate", rate(bhits, bnum))
82 xcoverage.setAttribute("line-rate", rate(lhits_tot, lnum_tot))
83 xcoverage.setAttribute("branch-rate", rate(bhits_tot, bnum_tot))
141 xclass.setAttribute("line-rate", rate(class_hits, class_lines)
[all...] |
| /external/qemu/audio/ |
| mixeng.c | 270 * Sound Tools rate change effect file.
289 struct rate { struct
301 struct rate *rate = audio_calloc (AUDIO_FUNC, 1, sizeof (*rate)); local
303 if (!rate) {
304 dolog ("Could not allocate resampler (%zu bytes)\n", sizeof (*rate));
308 rate->opos = 0;
311 rate->opos_inc = ((uint64_t) inrate << 32) / outrate;
313 rate->ipos = 0
[all...] |
| /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...] |
| /external/chromium_org/components/autofill/core/common/ |
| autofill_pref_names.cc | 18 // Double that indicates negative (for not matched forms) upload rate.
21 // Double that indicates positive (for matched forms) upload rate.
|
| /external/iproute2/tc/ |
| q_netem.c | 42 " [ rate RATE [PACKETOVERHEAD] [CELLSIZE] [CELLOVERHEAD]]\n");
179 struct tc_netem_rate rate; local
187 memset(&rate, 0, sizeof(rate));
386 } else if (matches(*argv, "rate") == 0) {
389 if (get_rate(&rate.rate, *argv)) {
390 explain1("rate");
395 if (get_s32(&rate.packet_overhead, *argv, 0))
504 const struct tc_netem_rate *rate = NULL; local
[all...] |
| /external/iproute2/testsuite/tests/ |
| cbq.t | 4 $TC class add dev $DEV parent 10:0 classid 10:12 cbq bandwidth 100mbit rate 100mbit allot 1514 prio 3 maxburst 1 avpkt 500 bounded
9 $TC class add dev $DEV parent 10:0 classid 10:12 cbq bandwidth 100mbit rate 100mbit allot 1514 prio 3 maxburst 1 avpkt 500 bounded
|
| /external/qemu/distrib/sdl-1.2.15/docs/man3/ |
| SDL_EnableKeyRepeat.3 | 3 SDL_EnableKeyRepeat \- Set keyboard repeat rate\&.
11 Enables or disables the keyboard repeat rate\&. \fBdelay\fR specifies how long the key must be pressed before it begins repeating, it then repeats at the speed specified by \fBinterval\fR\&. Both \fBdelay\fR and \fBinterval\fR are expressed in milliseconds\&.
|
| /external/chromium/webkit/glue/media/ |
| audio_decoder.cc | 41 // TODO(crogers) : do sample-rate conversion with FFmpeg.
43 // the WebAudioBus at the file's sample-rate.
53 << " sample rate: " << file_sample_rate
56 // Change to destination sample-rate.
|
