/external/chromium_org/chromeos/ime/ |
H A D | fake_ime_keyboard.cc | 70 bool FakeImeKeyboard::SetAutoRepeatRate(const AutoRepeatRate& rate) { argument 71 last_auto_repeat_rate_ = rate;
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/external/chromium_org/third_party/opus/src/celt/ |
H A D | 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);
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/external/chromium_org/third_party/webrtc/common_audio/vad/ |
H A D | webrtc_vad.c | 106 int WebRtcVad_ValidRateAndFrameLength(int rate, int frame_length) { argument 115 if (kValidRates[i] == rate) {
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/external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
H A D | test_utils.h | 32 static inline int SamplesFromRate(int rate) { argument 33 return AudioProcessing::kChunkSizeMs * rate / 1000;
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_payload_registry_unittest.cc | 40 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
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/external/iproute2/tc/ |
H A D | 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);
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H A D | q_gred.c | 125 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)
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H A D | 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) {
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/external/libopus/celt/ |
H A D | 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);
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/external/mp4parser/isoparser/src/main/java/com/coremedia/iso/boxes/ |
H A D | ProgressiveDownloadInformationBox.java | 55 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 [all...] |
/external/opencv/cv/src/ |
H A D | _cv.h | 103 double *rate; member in struct:CvPyramid
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/external/ppp/pppd/plugins/pppoatm/ |
H A D | text2qos.c | 30 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;
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/external/qemu/audio/ |
H A D | 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 (%u bytes)\n", (int)sizeof (*rate)); 308 rate->opos = 0; 311 rate->opos_inc = ((uint64_t) inrate << 32) / outrate; 313 rate [all...] |
/external/qemu/distrib/sdl-1.2.15/src/audio/nds/ |
H A D | soundcommon.h | 54 u32 rate; member in struct:__anon28511 66 extern void SoundSystemInit(u32 rate,u32 buffersize,u8 channel,u8 format);
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/external/speex/libspeex/ |
H A D | vorbis_psy.h | 80 int rate; member in struct:__anon30113 91 VorbisPsy *vorbis_psy_init(int rate, int size);
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/external/webrtc/src/common_audio/vad/ |
H A D | vad_unittest.cc | 39 // 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
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/external/chromium_org/components/policy/core/common/cloud/ |
H A D | cloud_policy_refresh_scheduler_unittest.cc | 126 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 [all...] |
/external/chromium_org/remoting/client/ |
H A D | audio_player_unittest.cc | 93 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
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/external/chromium_org/remoting/codec/ |
H A D | audio_encoder_opus.cc | 21 // 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.
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H A D | 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_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 [all...] |
/external/chromium_org/third_party/WebKit/Source/platform/speech/ |
H A D | PlatformSpeechSynthesisUtterance.h | 61 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
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/external/chromium_org/third_party/leveldatabase/src/util/ |
H A D | bloom_test.cc | 118 // 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
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/external/chromium_org/third_party/libsrtp/srtp/test/ |
H A D | rdbx_driver.c | 71 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);
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/g722/ |
H A D | g722_decode.c | 159 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
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | Channel.cc | 399 double rate; local 402 rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime); 404 return rate;
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