Searched defs:rate (Results 26 - 50 of 262) sorted by relevance

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/external/chromium_org/chromeos/ime/
H A Dfake_ime_keyboard.cc70 bool FakeImeKeyboard::SetAutoRepeatRate(const AutoRepeatRate& rate) { argument
71 last_auto_repeat_rate_ = rate;
/external/chromium_org/third_party/opus/src/celt/
H A Dopus_custom_demo.c51 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/chromium_org/third_party/webrtc/common_audio/vad/
H A Dwebrtc_vad.c106 int WebRtcVad_ValidRateAndFrameLength(int rate, int frame_length) { argument
115 if (kValidRates[i] == rate) {
/external/chromium_org/third_party/webrtc/modules/audio_processing/test/
H A Dtest_utils.h32 static inline int SamplesFromRate(int rate) { argument
33 return AudioProcessing::kChunkSizeMs * rate / 1000;
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/
H A Drtp_payload_registry_unittest.cc40 uint8_t payload_type, uint32_t rate) {
44 { kTypicalFrequency, kTypicalChannels, rate }
55 rate)).WillOnce(Return(returned_payload_on_heap));
140 // Ok, update the rate for one of the codecs. If either the incoming rate or
141 // the stored rate is zero it's not really an error to register the same
39 ExpectReturnOfTypicalAudioPayload( uint8_t payload_type, uint32_t rate) argument
/external/iproute2/tc/
H A Dq_choke.c41 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);
H A Dq_gred.c125 unsigned rate = 0; local
201 if (get_rate(&rate, *argv)) {
217 if (rate == 0)
218 get_rate(&rate, "10Mbit");
244 if ((wlog = tc_red_eval_idle_damping(opt.Wlog, avpkt, rate, sbuf)) < 0)
H A Dq_red.c42 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) {
/external/libopus/celt/
H A Dopus_custom_demo.c51 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/mp4parser/isoparser/src/main/java/com/coremedia/iso/boxes/
H A DProgressiveDownloadInformationBox.java55 long rate; field in class:ProgressiveDownloadInformationBox.Entry
58 public Entry(long rate, long initialDelay) { argument
59 this.rate = rate;
64 return rate;
67 public void setRate(long rate) { argument
68 this.rate = rate;
82 "rate=" + rate
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/external/opencv/cv/src/
H A D_cv.h103 double *rate; member in struct:CvPyramid
/external/ppp/pppd/plugins/pppoatm/
H A Dtext2qos.c30 unsigned int rate,fract; local
37 rate = strtoul(*text,&end,10);
47 if (rate > UINT_MAX/1000) return RATE_ERROR;
48 rate *= 1000;
63 rate += fract;
69 rate = (rate+(up ? 8*ATM_CELL_PAYLOAD-1 : 0))/8/
74 if (rate > INT_MAX) return RATE_ERROR;
76 return rate;
/external/qemu/audio/
H A Dmixeng.c270 * 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 (%u bytes)\n", (int)sizeof (*rate));
308 rate->opos = 0;
311 rate->opos_inc = ((uint64_t) inrate << 32) / outrate;
313 rate
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/external/qemu/distrib/sdl-1.2.15/src/audio/nds/
H A Dsoundcommon.h54 u32 rate; member in struct:__anon28511
66 extern void SoundSystemInit(u32 rate,u32 buffersize,u8 channel,u8 format);
/external/speex/libspeex/
H A Dvorbis_psy.h80 int rate; member in struct:__anon30113
91 VorbisPsy *vorbis_psy_init(int rate, int size);
/external/webrtc/src/common_audio/vad/
H A Dvad_unittest.cc39 // Returns true if the rate and frame length combination is valid.
40 bool ValidRatesAndFrameLengths(int16_t rate, int16_t frame_length) { argument
41 if (rate == 8000) {
46 } else if (rate == 16000) {
52 if (rate == 32000) {
135 // Invalid sampling rate
142 // Loop through sampling rate and frame length combinations
/external/chromium_org/components/policy/core/common/cloud/
H A Dcloud_policy_refresh_scheduler_unittest.cc126 base::TimeDelta rate = base::TimeDelta::FromMilliseconds( local
130 CheckTimingWithAge(rate,
198 // Max out the request rate.
217 // The scheduler has scheduled refreshes at the initial refresh rate.
234 // The next refresh has been scheduled using a lower refresh rate.
263 // The next refresh has been scheduled at the normal rate.
280 // The next refresh has been scheduled using a lower refresh rate.
284 // refresh is rescheduled at the lower rate again; after executing all
288 // The next refresh has been scheduled using a lower refresh rate.
307 // The next refresh has been scheduled using a lower refresh rate
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/external/chromium_org/remoting/client/
H A Daudio_player_unittest.cc93 AudioPacket::SamplingRate rate, int samples) {
96 packet->set_sampling_rate(rate);
144 // New packet with different sampling rate causes previous samples to
92 CreatePacketWithSamplingRate( AudioPacket::SamplingRate rate, int samples) argument
/external/chromium_org/remoting/codec/
H A Daudio_encoder_opus.cc21 // Opus doesn't support 44100 sampling rate so we always resample to 48kHz.
29 // Number of samples per frame when using default sampling rate.
36 bool IsSupportedSampleRate(int rate) { argument
37 return rate == 44100 || rate == 48000;
85 // Drop leftover data because it's for different sampling rate.
H A Daudio_encoder_opus_unittest.cc27 // 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_h
54 GetSampleValue( AudioPacket::SamplingRate rate, double frequency_hz, double pos, int channel) argument
66 CreatePacket( int samples, AudioPacket::SamplingRate rate, double frequency_hz, int pos) argument
106 ValidateReceivedData(int samples, AudioPacket::SamplingRate rate, double frequency_hz, const std::vector<int16>& received_data) argument
127 TestEncodeDecode(int packet_size, double frequency_hz, AudioPacket::SamplingRate rate) argument
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/external/chromium_org/third_party/WebKit/Source/platform/speech/
H A DPlatformSpeechSynthesisUtterance.h61 float rate() const { return m_rate; } function in class:WebCore::FINAL
62 void setRate(float rate) { m_rate = std::max(std::min(10.0f, rate), 0.1f); } argument
/external/chromium_org/third_party/leveldatabase/src/util/
H A Dbloom_test.cc118 // Count number of filters that significantly exceed the false positive rate
138 // Check false positive rate
139 double rate = FalsePositiveRate(); local
142 rate*100.0, length, static_cast<int>(FilterSize()));
144 ASSERT_LE(rate, 0.02); // Must not be over 2%
145 if (rate > 0.0125) mediocre_filters++; // Allowed, but not too often
/external/chromium_org/third_party/libsrtp/srtp/test/
H A Drdbx_driver.c71 double rate; local
122 rate = rdbx_check_adds_per_second(1 << 18, 128);
123 printf("rdbx_check/replay_adds per second (ws=128): %e\n", rate);
124 rate = rdbx_check_adds_per_second(1 << 18, 1024);
125 printf("rdbx_check/replay_adds per second (ws=1024): %e\n", rate);
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/g722/
H A Dg722_decode.c159 int rate,
168 if (rate == 48000)
170 else if (rate == 56000)
158 WebRtc_g722_decode_init(g722_decode_state_t *s, int rate, int options) argument
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/
H A DChannel.cc399 double rate; local
402 rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
404 return rate;

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