| /frameworks/data-binding/ |
| buildForGradlePlugin.sh | 1 ./gradlew :dB:compiler:uploadArchives :dB:compilerCommon:uploadArchives :dB:baseLibrary:uploadArchives -PaddRemoteRepos=true -PuseReleaseVersion=true -PforGradlePlugin=true
|
| /external/libvorbis/lib/ |
| misc.h | 27 extern void _analysis_output(char *base,int i,float *v,int n,int bark,int dB,
29 extern void _analysis_output_always(char *base,int i,float *v,int n,int bark,int dB,
|
| analysis.c | 70 void _analysis_output_always(char *base,int i,float *v,int n,int bark,int dB,ogg_int64_t off){
90 if(dB){
104 void _analysis_output(char *base,int i,float *v,int n,int bark,int dB,
106 if(analysis_noisy)_analysis_output_always(base,i,v,n,bark,dB,off);
|
| psytune.c | 78 /* y: 0 10 20 30 40 50 60 70 80 90 100 dB */
199 void analysis(char *base,int i,float *v,int n,int bark,int dB){
208 if(dB && v[j]==0)
216 if(dB){
|
| /cts/apps/CtsVerifier/src/com/android/cts/verifier/audio/ |
| WavAnalyzer.java | 15 private double[] dB; // Average response
47 dB = measurePipStrength();
56 result = responsePassesHifiTest(dB);
97 boolean responsePassesHifiTest(double[] dB) {
98 for (int i = 0; i < dB.length; i++) {
100 if (Double.isNaN(dB[i])) {
107 if (Util.mean(dB) - Util.mean(noiseDB) < Common.SIGNAL_MIN_STRENGTH_DB_ABOVE_NOISE) {
114 threshold = dB[indexOf2000Hz] + Common.PASSING_THRESHOLD_DB;
118 System.arraycopy(dB, indexOf18500Hz, responseInRange, 0, responseInRange.length);
211 double[] dB = new double[Common.PIP_NUM]
[all...] |
| HifiUltrasoundSpeakerTestActivity.java | 302 double[] dB = wavAnalyzerTask.getDB();
305 dBWrap[i] = new Double(dB[i]);
|
| HifiUltrasoundTestActivity.java | 261 double[] dB = wavAnalyzerTask.getDB();
264 dBWrap[i] = new Double(dB[i]);
|
| /compatibility/cdd/5_multimedia/ |
| 5_11_unprocessed-audio.md | 15 characteristics in the mid-frequency range: specifically ±10dB from
20 range: specifically from ±20 dB from 5 Hz to 100 Hz compared to the
25 range: specifically from ±30 dB from 7000 Hz to 22 KHz compared to the
30 tone source played at 94 dB Sound Pressure Level (SPL) yields a response with
31 RMS of 520 for 16 bit-samples (or -36 dB Full Scale for floating point/double
35 * [C-1-6] MUST have a signal-to-noise ratio (SNR) at 60 dB or higher for
37 (whereas the SNR is measured as the difference between 94 dB SPL and equivalent
41 1% for 1 kHZ at 90 dB SPL input level at each and every microphone used to
|
| 5_4_audio-recording.md | 52 amplitude versus frequency characteristics: specifically, ±3 dB, from 100 Hz
55 such that a 90 dB sound power level (SPL) source at 1000 Hz yields RMS of
58 levels linearly track input SPL changes over at least a 30 dB range from -18
59 dB to +12 dB re 90 dB SPL at the microphone.
61 distortion (THD) less than 1% for 1 kHz at 90 dB SPL input level at the
|
| /external/eigen/Eigen/src/LU/arch/ |
| Inverse_SSE.h | 79 __m128 dA, dB, dC, dD; // determinant of the sub-matrices
93 // dB = |B|
94 dB = _mm_mul_ps(_mm_shuffle_ps(B, B, 0x5F),B);
95 dB = _mm_sub_ss(dB, _mm_movehl_ps(dB,dB));
118 d2 = _mm_mul_ss(dB,dC);
145 iB = _mm_sub_ps(_mm_mul_ps(C,_mm_shuffle_ps(dB,dB,0)), iB)
[all...] |
| /external/eigen/unsupported/test/ |
| kronecker_product.cpp | 196 MatrixXf dA(ra,ca), dB(rb,cb), dC;
198 initSparse(density, dB, sB);
201 dC = kroneckerProduct(dA,dB);
205 dC = kroneckerProduct(dA.transpose(),dB);
209 dC = kroneckerProduct(dA.transpose(),dB.transpose());
213 dC = kroneckerProduct(dA,dB.transpose());
217 dC = kroneckerProduct(dA,dB);
221 dC = kroneckerProduct(dA,dB);
224 sC2 = kroneckerProduct(sA,dB);
225 dC = kroneckerProduct(dA,dB);
[all...] |
| /external/libvorbis/doc/ |
| 10-tables.tex | 6 \subsection{floor1\_inverse\_dB\_table} \label{vorbis:spec:floor1:inverse:dB:table}
|
| 01-introduction.tex | 209 representation on a dB amplitude scale and Bark frequency scale.
211 representation on a dB amplitude scale and linear frequency scale.
462 However, floor vector values can span \~{}140dB (\~{}24 bits unsigned), and
463 the audio spectrum vector should represent a minimum of 120dB (\~{}21
466 $-140$dB, it must be able to span 0 to $+140$dB. For the residue vector
467 to reach full scale if the floor is nailed at 0dB, it must be able to
468 represent $-140$dB to $+0$dB. Thus, in order to handle full range
469 dynamics, a residue vector may span $-140$dB to $+140$dB entirely withi
[all...] |
| 04-codec.tex | 550 However, floor vector values can span \~140dB (\~24 bits unsigned), and
551 the audio spectrum vector should represent a minimum of 120dB (\~21
554 $-140$dB, it must be able to span 0 to $+140$dB. For the residue vector
555 to reach full scale if the floor is nailed at 0dB, it must be able to
556 represent $-140$dB to $+0$dB. Thus, in order to handle full range
557 dynamics, a residue vector may span $-140$dB to $+140$dB entirely within
558 spec. A 280dB range is approximately 48 bits with sign; thus th
[all...] |
| /external/pdfium/fpdfsdk/pdfwindow/ |
| cpwl_color.cpp | 40 CPWL_Color ConvertRGB2GRAY(FX_FLOAT dR, FX_FLOAT dG, FX_FLOAT dB) {
41 if (!InRange(dR) || !InRange(dG) || !InRange(dB))
43 return CPWL_Color(COLORTYPE_GRAY, 0.3f * dR + 0.59f * dG + 0.11f * dB);
54 CPWL_Color ConvertRGB2CMYK(FX_FLOAT dR, FX_FLOAT dG, FX_FLOAT dB) {
55 if (!InRange(dR) || !InRange(dG) || !InRange(dB))
60 FX_FLOAT y = 1.0f - dB;
|
| /external/eigen/test/ |
| sparse_solver.h | 42 void check_sparse_solving(Solver& solver, const typename Solver::MatrixType& A, const Rhs& b, const DenseMat& dA, const DenseRhs& db)
48 DenseRhs refX = dA.householderQr().solve(db);
92 Map<const DenseRhs> bm(db.data(), db.rows(), db.cols());
114 DenseRhs oldb(db);
116 x.block(0,0,x.rows(),x.cols()) = solver.solve(db.block(0,0,db.rows(),db.cols()));
117 VERIFY(oldb.isApprox(db) && "sparse solver testing: the rhs should not be modified!")
[all...] |
| /external/skia/src/pathops/ |
| SkPathOpsPoint.h | 194 SkDPoint dA, dB;
196 dB.set(b);
197 double dist = dA.distance(dB); // OPTIMIZATION: can we compare against distSq instead ?
246 SkDPoint dA, dB;
248 dB.set(b);
249 double dist = dA.distance(dB); // OPTIMIZATION: can we compare against distSq instead ?
|
| /external/sonivox/arm-wt-22k/lib_src/ |
| ARM-E_interpolate_loop_gnu.s | 97 @ This section performs a gain adjustment of -12dB for 16-bit samples
98 @ or +36dB for 8-bit samples. For a high quality synthesizer, the output
105 MOV tmp0, tmp0, LSL #6 @ boost 8-bit signal by 36dB
107 MOV tmp0, tmp0, ASR #2 @ reduce 16-bit signal by 12dB
|
| ARM-E_interpolate_noloop_gnu.s | 89 @ This section performs a gain adjustment of -12dB for 16-bit samples
90 @ or +36dB for 8-bit samples. For a high quality synthesizer, the output
97 MOV tmp0, tmp0, LSL #6 @ boost 8-bit signal by 36dB
99 MOV tmp0, tmp0, ASR #2 @ reduce 16-bit signal by 12dB
|
| ARM-E_voice_gain_gnu.s | 152 MOV tmp0, tmp0, ASR #1 @ add 6dB headroom
|
| /external/mesa3d/src/mesa/swrast/ |
| s_blend.c | 492 GLfloat dR, dG, dB, dA; /* Dest factor */
626 dR = dG = dB = 0.0F;
629 dR = dG = dB = 1.0F;
634 dB = Bs;
639 dB = 1.0F - Bs;
642 dR = dG = dB = As;
645 dR = dG = dB = 1.0F - As;
648 dR = dG = dB = Ad;
651 dR = dG = dB = 1.0F - Ad;
656 dB = ctx->Color.BlendColor[2]
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/agc/legacy/ |
| digital_agc.c | 34 // subplot(121); plot(in, out); axis([-30, 0, -5, 20]); grid on; xlabel('Input (dB)'); ylabel('Gain (dB)');
35 // subplot(122); plot(in, in+out); axis([-30, 0, -30, 5]); grid on; xlabel('Input (dB)'); ylabel('Output (dB)');
133 // Calculate a denominator used in the exponential part to convert from dB to linear scale:
266 // start out with 0 dB gain
536 // multiply by 253/256 ==> -0.1 dB
618 state->stdLongTerm = 0; // standard deviation of input level in dB
625 state->stdShortTerm = 0; // short-term standard deviation of input level in dB
644 int16_t zeros, dB;
[all...] |
| /hardware/qcom/audio/legacy/libalsa-intf/ |
| alsa_mixer.c | 306 static void print_dB(long dB)
308 ALOGV("%li.%02lidB", dB / 100, (dB < 0 ? -dB : dB) % 100);
545 ALOGV("tlv db linear: b4 %d\n", percent);
552 ALOGV("tlv db linear: %d %d %d\n", percent, min, max);
609 * <volume><dB>
|
| /external/swiftshader/src/Renderer/ |
| ETC_Decoder.cpp | 129 int b = (B + dB);
184 signed char dB : 3;
356 int b2 = extend_5to8bits(B + dB);
|
| /system/core/libpixelflinger/ |
| scanline.cpp | [all...] |
