/external/chromium_org/media/audio/cras/ |
H A D | 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; local 249 return GetVolumeRatioFromDecibels(dB); 255 // Convert from the passed volume ratio, to dB * 100. 256 double dB = GetDecibelsFromVolumeRatio(volume); local 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; local 272 return GetVolumeRatioFromDecibels(dB); 275 double CrasInputStream::GetVolumeRatioFromDecibels(double dB) cons [all...] |
H A D | 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.
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/external/libvorbis/lib/ |
H A D | 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,
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H A D | analysis.c | 70 void _analysis_output_always(char *base,int i,float *v,int n,int bark,int dB,ogg_int64_t off){ argument 90 if(dB){ 104 void _analysis_output(char *base,int i,float *v,int n,int bark,int dB, argument 106 if(analysis_noisy)_analysis_output_always(base,i,v,n,bark,dB,off);
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H A D | 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){ argument 208 if(dB && v[j]==0) 216 if(dB){
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/external/eigen/Eigen/src/LU/arch/ |
H A D | Inverse_SSE.h | 77 __m128 dA, dB, dC, dD; // determinant of the sub-matrices local 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, 215 __m128d dA, dB, dC, dD; // determinant of the sub-matrices local [all...] |
/external/libvorbis/doc/ |
H A D | 10-tables.tex | 6 \subsection{floor1\_inverse\_dB\_table} \label{vorbis:spec:floor1:inverse:dB:table}
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H A D | 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 t [all...] |
H A D | 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 rang [all...] |
/external/eigen/test/ |
H A D | sparse_solver.h | 187 DenseMatrix dB(size,rhsCols); 188 initSparse<Scalar>(density, dB, B, ForceNonZeroDiag); 193 check_sparse_solving(solver, A, dB, dA, dB); 194 check_sparse_solving(solver, halfA, dB, dA, dB); 195 check_sparse_solving(solver, A, B, dA, dB); 196 check_sparse_solving(solver, halfA, B, dA, dB); 271 DenseMatrix dB = DenseMatrix::Random(size,rhsCols); local 275 check_sparse_solving(solver, A, dB, d [all...] |
/external/qemu/distrib/sdl-1.2.15/src/video/ |
H A D | SDL_blit_0.c | 371 unsigned dR, dG, dB; local 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; local 423 pixel, dR, dG, dB); 424 ALPHA_BLEND(sR, sG, sB, A, dR, dG, dB); 425 ASSEMBLE_RGB(dst, dstbpp, dstfmt, dR, dG, dB);
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H A D | SDL_blit_1.c | 424 int dR, dG, dB; local 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; local 473 pixel, dR, dG, dB); 474 ALPHA_BLEND(sR, sG, sB, A, dR, dG, dB); 475 ASSEMBLE_RGB(dst, dstbpp, dstfmt, dR, dG, dB);
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H A D | 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 [all...] |
H A D | SDL_blit.h | 385 #define ALPHA_BLEND(sR, sG, sB, A, dR, dG, dB) \ 389 dB = (((sB-dB)*(A)+255)>>8)+dB; \
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/external/sonivox/arm-hybrid-22k/lib_src/ |
H A D | 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
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H A D | 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
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H A D | ARM-E_voice_gain_gnu.s | 152 MOV tmp0, tmp0, ASR #1 @ add 6dB headroom
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/external/sonivox/arm-wt-22k/lib_src/ |
H A D | 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
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H A D | 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
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H A D | ARM-E_voice_gain_gnu.s | 152 MOV tmp0, tmp0, ASR #1 @ add 6dB headroom
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/external/chromium_org/third_party/mesa/src/src/mesa/swrast/ |
H A D | s_blend.c | 495 GLfloat dR, dG, dB, dA; /* Dest factor */ local 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 [all...] |
/external/mesa3d/src/mesa/swrast/ |
H A D | s_blend.c | 495 GLfloat dR, dG, dB, dA; /* Dest factor */ local 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 [all...] |
/external/webrtc/src/modules/audio_processing/agc/ |
H A D | 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; local [all...] |
/external/jmonkeyengine/engine/src/core/com/jme3/audio/ |
H A D | Environment.java | 74 float dB = eaxDb / 2000f; 75 return FastMath.pow(10f, dB);
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/external/chromium_org/third_party/WebKit/Source/core/platform/graphics/ |
H A D | Color.cpp | 178 int dB = c1.blue() - c2.blue(); 179 return dR * dR + dG * dG + dB * dB;
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