Searched refs:dB (Results 1 - 25 of 37) sorted by relevance

12

/external/chromium_org/media/audio/cras/
H A Dcras_input.cc246 // 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 Dcras_input.h69 // 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/
H A Dmisc.h27 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,
H A Danalysis.c70 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);
H A Dpsytune.c78 /* 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){
/external/eigen/Eigen/src/LU/arch/
H A DInverse_SSE.h77 __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 D10-tables.tex6 \subsection{floor1\_inverse\_dB\_table} \label{vorbis:spec:floor1:inverse:dB:table}
H A D01-introduction.tex209 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 D04-codec.tex550 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 Dsparse_solver.h187 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 DSDL_blit_0.c371 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);
H A DSDL_blit_1.c424 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);
H A DSDL_blit_A.c90 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 DSDL_blit.h385 #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/
H A DARM-E_interpolate_loop_gnu.s97 @ 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
H A DARM-E_interpolate_noloop_gnu.s89 @ 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
H A DARM-E_voice_gain_gnu.s152 MOV tmp0, tmp0, ASR #1 @ add 6dB headroom
/external/sonivox/arm-wt-22k/lib_src/
H A DARM-E_interpolate_loop_gnu.s97 @ 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
H A DARM-E_interpolate_noloop_gnu.s89 @ 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
H A DARM-E_voice_gain_gnu.s152 MOV tmp0, tmp0, ASR #1 @ add 6dB headroom
/external/chromium_org/third_party/mesa/src/src/mesa/swrast/
H A Ds_blend.c495 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 Ds_blend.c495 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 Ddigital_agc.c34 // 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 DEnvironment.java74 float dB = eaxDb / 2000f;
75 return FastMath.pow(10f, dB);
/external/chromium_org/third_party/WebKit/Source/core/platform/graphics/
H A DColor.cpp178 int dB = c1.blue() - c2.blue();
179 return dR * dR + dG * dG + dB * dB;

Completed in 492 milliseconds

12