/external/chromium_org/chrome/browser/resources/app_list/ |
plugin_manager.js | 106 * @param {number} samplingRate the sampling rate the plugin accepts. 120 * @param {number} samplingRate the sampling rate the plugin accepts.
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/external/chromium_org/content/renderer/media/ |
webrtc_audio_renderer.cc | 30 // current sample rate (set by the user) on Windows and Mac OS X. The listed 32 // any rate outside these ranges. 37 // TODO(leozwang): We want to use native sampling rate on Android to achieve 211 DVLOG(1) << "Audio output hardware sample rate: " << sample_rate; 214 // and 48000 is the preferred sample rate. Therefore, if 192000 is detected, 215 // we change the rate to 48000 instead. The consequence is that the native 231 // Verify that the reported output hardware sample rate is supported 237 DLOG(ERROR) << sample_rate << " is not a supported output rate.";
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/external/chromium_org/media/audio/alsa/ |
alsa_wrapper.cc | 76 unsigned int rate, int soft_resample, 82 rate,
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/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);
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/external/chromium_org/media/base/ |
audio_renderer.h | 55 // Start audio decoding and rendering at the current playback rate, executing 77 // Updates the current playback rate.
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video_renderer.h | 66 // Start audio decoding and rendering at the current playback rate, executing 91 // Updates the current playback rate.
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/external/chromium_org/net/quic/congestion_control/ |
channel_estimator.cc | 7 // To get information about bandwidth, our send rate for a pair of packets must 8 // be much faster (ideally back to back) than the receive rate. In that
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/external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
AudioBufferSourceNode.cpp | 48 // Arbitrary upper limit on playback rate. 241 // Sanity check that our playback rate isn't larger than the loop size. 460 // Incorporate buffer's sample-rate versus AudioContext's sample-rate. 461 // Normally it's not an issue because buffers are loaded at the AudioContext's sample-rate, but we can handle it in any case. 470 // Sanity check the total rate. It's very important that the resampler not get any bad rate values. 473 totalRate = 1; // zero rate is considered illegal
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MediaStreamAudioSourceNode.cpp | 67 // The sample-rate must be equal to the context's sample-rate.
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/external/chromium_org/third_party/WebKit/Source/platform/audio/ |
AudioDestination.h | 57 // Sample-rate conversion may happen in AudioDestination to the hardware sample-rate
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/external/chromium_org/third_party/opus/src/silk/ |
decoder_set_fs.c | 34 /* Set decoder sampling rate */ 88 /* unsupported sampling rate */
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/external/jmonkeyengine/engine/src/core/com/jme3/audio/ |
AudioData.java | 89 * @return The sample rate, or how many samples per second. 99 * @param sampleRate Sample rate, 44100, 22050, etc.
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/external/kernel-headers/original/asm-x86/ |
termbits.h | 143 #define BOTHER 0010000 /* non standard rate */ 159 #define CIBAUD 002003600000 /* input baud rate */
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/external/kernel-headers/original/linux/ |
atmdev.h | 18 /* OC3 link rate: 155520000 bps 21 max cell rate: 353207.547 cells/sec */ 23 /* 25 Mbps ATM cell rate (59111) */ 25 /* OC12 link rate: 622080000 bps 28 max cell rate: 1412830.188 cells/sec */ 55 /* get link rate */ 355 int link_rate; /* link rate (default: OC3) */ 370 #define ATM_OF_INRATE 2 /* Attempt in-rate delivery */
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/external/libvorbis/doc/vorbisfile/ |
vorbisfile_example_c.html | 61 fprintf(stderr,"\nBitstream is %d channel, %ldHz\n",vi->channels,vi->rate); 74 /* we don't bother dealing with sample rate changes, etc, but
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/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
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/external/qemu/distrib/sdl-1.2.15/docs/html/ |
sdlenablekeyrepeat.html | 83 >SDL_EnableKeyRepeat -- Set keyboard repeat rate.</DIV 123 >Enables or disables the keyboard repeat rate. <TT
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/external/robolectric/src/main/java/com/xtremelabs/robolectric/shadows/ |
ShadowMediaRecorder.java | 134 public void setVideoFrameRate(int rate) { 135 videoFrameRate = rate;
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/external/sonivox/arm-hybrid-22k/lib_src/ |
eas_wtsynth.h | 36 /* adjust the filter cutoff frequency to the sample rate */ 54 #error "_SAMPLE_RATE_XXXXX must be defined to valid rate"
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/external/sonivox/arm-wt-22k/lib_src/ |
eas_wtsynth.h | 36 /* adjust the filter cutoff frequency to the sample rate */ 54 #error "_SAMPLE_RATE_XXXXX must be defined to valid rate"
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/external/speex/libspeex/ |
vorbis_psy.h | 80 int rate; member in struct:__anon26381 91 VorbisPsy *vorbis_psy_init(int rate, int size);
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/external/srtp/test/ |
roc_driver.c | 113 printf("error: failure rate too high (%d bad estimates in %d trials)\n", 158 printf("error: failure rate too high (%d bad estimates in %d trials)\n",
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/external/tinycompress/ |
utils.c | 64 unsigned int compress_get_alsa_rate(unsigned int rate) 66 switch (rate) {
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/frameworks/av/media/libeffects/lvm/lib/Common/src/ |
LVM_FO_HPF.c | 61 /* w Sample rate in radians, where: */ 64 /* Fs is the sample rate in Hz */
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LVM_FO_LPF.c | 61 /* w Sample rate in radians, where: */ 64 /* Fs is the sample rate in Hz */
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