| /frameworks/base/media/tests/audiotests/ |
| shared_mem_test.h | 16 void Generate(short *buffer, long bufferSz, long amplitude, unsigned long &phi, long dPhi);
18 short ComputeSine(long amplitude, long phi);
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| shared_mem_test.cpp | 80 long amplitude; local
86 amplitude = 1000;
88 Generate(smpBuf, BUF_SZ, amplitude, phi, dPhi); // fill buffer
134 void AudioTrackTest::Generate(short *buffer, long bufferSz, long amplitude, unsigned long &phi, long dPhi)
142 buffer[i0] = ComputeSine( amplitude, phi);
153 short AudioTrackTest::ComputeSine(long amplitude, long phi)
159 sample = (amplitude*sin1024[(phi>>22) & 0x3ff]) >> 15;
162 l1 = (amplitude*sin1024[((phi>>22) + 256) & 0x3ff]) >> 15; // 2^15*cosine
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| /external/jmonkeyengine/engine/src/terrain/com/jme3/terrain/noise/fractal/ |
| FractalSum.java | 39 * of the noise value with an ever decreasing (0 < roughness < 1) amplitude
54 private float amplitude;
field in class:FractalSum
63 this.amplitude = 1.0f;
73 for (float f = this.frequency, a = this.amplitude; f < this.maxFreq; f *= this.lacunarity, a *= this.roughness) {
118 return this.amplitude;
122 public Fractal setAmplitude(final float amplitude) {
123 this.amplitude = amplitude;
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| /frameworks/av/cmds/stagefright/ |
| SineSource.cpp | 80 int16_t amplitude = (int16_t)(32767.0 * sin(x)); local
82 *ptr++ = amplitude;
84 *ptr++ = amplitude;
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| /external/libvorbis/lib/ |
| tone.c | 7 fprintf(stderr,"tone <frequency_Hz>,[<amplitude>] [<frequency_Hz>,[<amplitude>]...]\n");
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| /external/libvorbis/doc/ |
| 06-floor0.tex | 51 decoding the curve amplitude and \varname{[floor0_order]} LSP
58 1) [amplitude] = read an unsigned integer of [floor0_amplitude_bits] bits
59 2) if ( [amplitude] is greater than zero ) \{
78 \item An \varname{[amplitude]} value of zero must result in a return code that indicates this channel is unused in this frame (the output of the channel will be all-zeroes in synthesis). Several later stages of decode don't occur for an unused channel.
82 \varname{[amplitude]} value had read zero at the beginning of decode.
106 Given an \varname{[amplitude]} integer and \varname{[coefficients]}
113 If the value \varname{[amplitude]} is zero, the return value is a
148 (dB) amplitude scale, mapping it to linear amplitude in the last step:
175 \exp \left( .11512925 \left(\frac{\mathtt{amplitude} \cdot \mathtt{floor0\_amplitute\_offset}}{(2^{\mathtt{floor0\_amplitude\_bits}}-1)\sqrt{p+q}
[all...] |
| /cts/suite/audio_quality/lib/src/task/ |
| TaskSound.cpp | 79 int amplitude = atoi(tokens->at(1).string()); local
83 buffer = AudioSignalFactory::generateSineWave(AudioHardware::E2BPS, amplitude,
90 int amplitude = atoi(tokens->at(1).string()); local
93 buffer = AudioSignalFactory::generateWhiteNoise(AudioHardware::E2BPS, amplitude,
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| /external/chromium_org/third_party/WebKit/Source/core/svg/ |
| SVGComponentTransferFunctionElement.cpp | 38 DEFINE_ANIMATED_NUMBER(SVGComponentTransferFunctionElement, SVGNames::amplitudeAttr, Amplitude, amplitude)
47 REGISTER_LOCAL_ANIMATED_PROPERTY(amplitude)
146 func.amplitude = amplitudeCurrentValue();
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| SVGComponentTransferFunctionElement.h | 91 DECLARE_ANIMATED_NUMBER(Amplitude, amplitude)
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| /external/chromium_org/third_party/opus/src/silk/ |
| shell_coder.c | 50 const opus_int p_child1, /* I pulse amplitude of first child subframe */
51 const opus_int p, /* I pulse amplitude of current subframe */
61 opus_int *p_child1, /* O pulse amplitude of first child subframe */
62 opus_int *p_child2, /* O pulse amplitude of second child subframe */
64 const opus_int p, /* I pulse amplitude of current subframe */
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| /cts/suite/audio_quality/test_description/ |
| dut_playback_spectrum.xml | 20 <!-- input: peak amplitude, duration in msec, sampling rate, high frequency, output: generated sound-->
30 <!-- input: host record, device record, samping rate, low frequency in Hz, high frequency in Hz, allowed error for pass in smaller side, allowed error in bigger side%, output: min value in lower side calculated normalized to 1.0, max value in higher side, calculated amplitude ratio in mannitude only between low f to high f -->
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| dut_recording_spectrum.xml | 20 <!-- input: peak amplitude, duration in msec, sampling rate, high frequency, output: generated sound-->
34 <!-- input: host record, device record, samping rate, low frequency in Hz, high frequency in Hz, allowed error for pass in smaller side, allowed error in bigger side%, output: min value in lower side calculated normalized to 1.0, max value in higher side, calculated amplitude ratio in mannitude only between low f to high f -->
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| /cts/suite/audio_quality/test_description/experimental/ |
| dut_playback_spectrum_chirp.xml | 20 <!-- input: peak amplitude, duration in msec, sampling rate, high frequency, output: generated sound-->
29 <!-- input: host record, device record, samping rate, low frequency in Hz, high frequency in Hz, allowed error for pass in smaller side, allowed error in bigger side%, output: min value in lower side calculated normalized to 1.0, max value in higher side, calculated amplitude ratio in mannitude only between low f to high f -->
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| ref_playback_spectrum.xml | 20 <!-- input: peak amplitude, duration in msec, sampling rate, high frequency, output: generated sound-->
29 <!-- input: host record, device record, samping rate, low frequency in Hz, high frequency in Hz, allowed error for pass in smaller side, allowed error in bigger side%, output: min value in lower side calculated normalized to 1.0, max value in higher side, calculated amplitude ratio in mannitude only between low f to high f -->
|
| ref_playback_spectrum_chirp.xml | 20 <!-- input: peak amplitude, duration in msec, sampling rate, high frequency, output: generated sound-->
29 <!-- input: host record, device record, samping rate, low frequency in Hz, high frequency in Hz, allowed error for pass in smaller side, allowed error in bigger side%, output: min value in lower side calculated normalized to 1.0, max value in higher side, calculated amplitude ratio in mannitude only between low f to high f -->
|
| /external/chromium_org/third_party/WebKit/Source/platform/graphics/filters/ |
| FEComponentTransfer.h | 46 , amplitude(0)
56 float amplitude; member in struct:WebCore::ComponentTransferFunction
|
| /external/libvorbis/vq/ |
| metrics.c | 26 mean and mean squared amplitude
138 fprintf(stderr,"\tmean sample amplitude: %g\n",
140 fprintf(stderr,"\tmean squared sample amplitude: %g\n\n",
212 float amplitude=0.f; local
217 amplitude=a[j*step]-(b->c->q_sequencep?last:0);
218 meanamplitude_acc+=fabs(amplitude);
219 meanamplitudesq_acc+=amplitude*amplitude;
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| /device/asus/flo/ |
| elan-touchscreen.idc | 33 touch.pressure.calibration = amplitude
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| touch_dev.idc | 29 touch.pressure.calibration = amplitude
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| /device/asus/grouper/ |
| elan-touchscreen.idc | 33 touch.pressure.calibration = amplitude
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| /device/lge/hammerhead/ |
| touch_dev.idc | 29 touch.pressure.calibration = amplitude
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| /device/lge/mako/ |
| touch_dev.idc | 29 touch.pressure.calibration = amplitude
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| /device/samsung/manta/ |
| Atmel_maXTouch_Touchscreen.idc | 29 touch.pressure.calibration = amplitude
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| /external/chromium_org/media/cast/test/ |
| audio_utility.h | 55 // low-amplitude noise), count the number of times the sine wave crosses
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| /prebuilts/gcc/linux-x86/host/i686-linux-glibc2.7-4.4.3/sysroot/usr/include/alsa/sound/ |
| asound_fm.h | 45 unsigned char am; /* amplitude modulation */
82 unsigned char am_depth; /* amplitude modulation depth (1=hi) */
|
