| /external/chromium_org/media/audio/cras/ |
| cras_input.cc | 246 // Capture gain is returned as dB * 100 (150 => 1.5dBFS). Convert the dB
248 double dB = cras_client_get_system_max_capture_gain(client_) / 100.0;
249 return GetVolumeRatioFromDecibels(dB);
255 // Convert from the passed volume ratio, to dB * 100.
256 double dB = GetDecibelsFromVolumeRatio(volume);
257 cras_client_set_system_capture_gain(client_, static_cast<long>(dB * 100.0));
271 long dB = cras_client_get_system_capture_gain(client_) / 100.0;
272 return GetVolumeRatioFromDecibels(dB);
275 double CrasInputStream::GetVolumeRatioFromDecibels(double dB) const
[all...] |
| cras_input.h | 69 // Convert from dB * 100 to a volume ratio.
70 double GetVolumeRatioFromDecibels(double dB) const;
72 // Convert from a volume ratio to dB.
|
| /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){
|
| /external/eigen/Eigen/src/LU/arch/ |
| Inverse_SSE.h | 77 __m128 dA, dB, dC, dD; // determinant of the sub-matrices
91 // dB = |B|
92 dB = _mm_mul_ps(_mm_shuffle_ps(B, B, 0x5F),B);
93 dB = _mm_sub_ss(dB, _mm_movehl_ps(dB,dB));
116 d2 = _mm_mul_ss(dB,dC);
143 iB = _mm_sub_ps(_mm_mul_ps(C,_mm_shuffle_ps(dB,dB,0)), iB)
[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/eigen/test/ |
| sparse_solver.h | 14 void check_sparse_solving(Solver& solver, const typename Solver::MatrixType& A, const Rhs& b, const DenseMat& dA, const DenseRhs& db)
19 DenseRhs refX = dA.lu().solve(db);
63 DenseRhs x(db.rows(), db.cols());
64 DenseRhs b(db), oldb(db);
187 DenseMatrix dB(size,rhsCols);
188 initSparse<Scalar>(density, dB, B, ForceNonZeroDiag);
193 check_sparse_solving(solver, A, dB, dA, dB);
[all...] |
| /external/qemu/distrib/sdl-1.2.15/src/video/ |
| SDL_blit_0.c | 371 unsigned dR, dG, dB;
376 pixel, dR, dG, dB);
377 ALPHA_BLEND(sR, sG, sB, A, dR, dG, dB);
378 ASSEMBLE_RGB(dst, dstbpp, dstfmt, dR, dG, dB);
417 int dR, dG, dB;
423 pixel, dR, dG, dB);
424 ALPHA_BLEND(sR, sG, sB, A, dR, dG, dB);
425 ASSEMBLE_RGB(dst, dstbpp, dstfmt, dR, dG, dB);
|
| SDL_blit_1.c | 424 int dR, dG, dB;
432 pixel, dR, dG, dB);
433 ALPHA_BLEND(sR, sG, sB, A, dR, dG, dB);
434 ASSEMBLE_RGB(dst, dstbpp, dstfmt, dR, dG, dB);
464 int dR, dG, dB;
473 pixel, dR, dG, dB);
474 ALPHA_BLEND(sR, sG, sB, A, dR, dG, dB);
475 ASSEMBLE_RGB(dst, dstbpp, dstfmt, dR, dG, dB);
|
| SDL_blit_A.c | 90 unsigned dB;
94 dB = dstfmt->palette->colors[*dst].b;
95 ALPHA_BLEND(sR, sG, sB, A, dR, dG, dB);
98 dB &= 0xff;
103 ((dB>>6)<<(0));
107 ((dB>>6)<<(0))];
143 unsigned dB;
147 dB = dstfmt->palette->colors[*dst].b;
148 ALPHA_BLEND(sR, sG, sB, sA, dR, dG, dB);
151 dB &= 0xff
[all...] |
| SDL_blit.h | 385 #define ALPHA_BLEND(sR, sG, sB, A, dR, dG, dB) \
389 dB = (((sB-dB)*(A)+255)>>8)+dB; \
|
| /external/sonivox/arm-hybrid-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
|
| /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
|
| /external/chromium_org/third_party/mesa/src/src/mesa/swrast/ |
| s_blend.c | 495 GLfloat dR, dG, dB, dA; /* Dest factor */
629 dR = dG = dB = 0.0F;
632 dR = dG = dB = 1.0F;
637 dB = Bs;
642 dB = 1.0F - Bs;
645 dR = dG = dB = As;
648 dR = dG = dB = 1.0F - As;
651 dR = dG = dB = Ad;
654 dR = dG = dB = 1.0F - Ad;
659 dB = ctx->Color.BlendColor[2]
[all...] |
| /external/mesa3d/src/mesa/swrast/ |
| s_blend.c | 495 GLfloat dR, dG, dB, dA; /* Dest factor */
629 dR = dG = dB = 0.0F;
632 dR = dG = dB = 1.0F;
637 dB = Bs;
642 dB = 1.0F - Bs;
645 dR = dG = dB = As;
648 dR = dG = dB = 1.0F - As;
651 dR = dG = dB = Ad;
654 dR = dG = dB = 1.0F - Ad;
659 dB = ctx->Color.BlendColor[2]
[all...] |
| /external/webrtc/src/modules/audio_processing/agc/ |
| 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)');
134 // Calculate a denominator used in the exponential part to convert from dB to linear scale:
270 // start out with 0 dB gain
542 // multiply by 253/256 ==> -0.1 dB
648 state->stdLongTerm = 0; // standard deviation of input level in dB
655 state->stdShortTerm = 0; // short-term standard deviation of input level in dB
674 WebRtc_Word16 zeros, dB;
[all...] |
| /external/jmonkeyengine/engine/src/core/com/jme3/audio/ |
| Environment.java | 74 float dB = eaxDb / 2000f;
75 return FastMath.pow(10f, dB);
|
| /external/chromium_org/third_party/WebKit/Source/core/platform/graphics/ |
| Color.cpp | 178 int dB = c1.blue() - c2.blue();
179 return dR * dR + dG * dG + dB * dB;
|
| /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>
|
| /system/core/libpixelflinger/ |
| scanline.cpp | [all...] |
