/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | lpc_analysis.c | 147 /* Find average pitch gain */ 156 /* If pitch gain is low and energy constant - increase noise level*/ 199 /* If pitch gain is low and energy constant - increase noise level*/ 323 /* add hearing threshold and compute the gain */ 350 /* add hearing threshold and compute of the gain */ 483 * -gain : pointer to a buffer where LP gains are written. 491 double* gain, 532 /* add hearing threshold and compute the gain */ 533 gain[subFrameCntr] = S_N_R / (sqrt(res_nrg) / *varscale + H_T_H); 487 WebRtcIsac_GetLpcGain( double signal_noise_ratio, const double* filtCoeffVecs, int numVecs, double* gain, double corrMat[][UB_LPC_ORDER + 1], const double* varscale) argument
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/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
H A D | lpc_analysis.c | 147 /* Find average pitch gain */ 156 /* If pitch gain is low and energy constant - increase noise level*/ 199 /* If pitch gain is low and energy constant - increase noise level*/ 323 /* add hearing threshold and compute the gain */ 350 /* add hearing threshold and compute of the gain */ 483 * -gain : pointer to a buffer where LP gains are written. 491 double* gain, 532 /* add hearing threshold and compute the gain */ 533 gain[subFrameCntr] = S_N_R / (sqrt(res_nrg) / *varscale + H_T_H); 487 WebRtcIsac_GetLpcGain( double signal_noise_ratio, const double* filtCoeffVecs, int numVecs, double* gain, double corrMat[][UB_LPC_ORDER + 1], const double* varscale) argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | pitch_filter.c | 40 int16_t gain, 87 // Get old lag and gain value from memory.
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/external/chromium_org/third_party/webrtc/modules/audio_processing/include/ |
H A D | audio_processing.h | 270 // necessary, to provide if gain control is enabled. On the server-side this 536 // The automatic gain control (AGC) component brings the signal to an 537 // appropriate range. This is done by applying a digital gain directly and, in 538 // the analog mode, prescribing an analog gain to be applied at the audio HAL. 562 // It consists of an analog gain prescription for the audio device and a 576 // short time-window of the input signal. It applies a fixed gain through 577 // most of the input level range, and compresses (gradually reduces gain 580 // predictable, so that a known gain can be applied. 597 // Sets the maximum |gain| the digital compression stage may apply, in dB. A 600 virtual int set_compression_gain_db(int gain) [all...] |
H A D | mock_audio_processing.h | 94 int(int gain));
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/external/kernel-headers/original/uapi/linux/ |
H A D | omap3isp.h | 315 * @offset: Table Offset of the gain table. 328 * @size: Size of LSC gain table. Filled when loaded from userspace. 346 * @obgain: Optical black average gain. 405 * @lsc: Pointer to LSC gain table. 487 * @gain: Gain. 492 __u8 gain; member in struct:omap3isp_prev_csup 499 * @dgain: Digital gain (U10Q8). 500 * @coef3: White balance gain - COEF 3 (U8Q5). 501 * @coef2: White balance gain - COEF 2 (U8Q5). 502 * @coef1: White balance gain [all...] |
/external/kernel-headers/original/uapi/sound/ |
H A D | compress_params.h | 293 * @gain: Add replay gain tags 310 __u32 gain; member in struct:snd_enc_flac
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/external/sonivox/arm-fm-22k/lib_src/ |
H A D | eas_synth.h | 278 EAS_I16 gain; /* current gain */ member in struct:s_synth_voice_tag
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H A D | eas_public.c | 1820 * Set the master gain for the mix engine in 1dB increments 1824 * volume - the desired master gain (100 is max) 1837 EAS_I16 gain; local 1852 /* get gain offset */ 1859 gain = EAS_VolumeToGain(volume - STREAM_VOLUME_HEADROOM); 1862 return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_VOLUME, gain); 1872 gain = EAS_VolumeToGain(volume - STREAM_VOLUME_HEADROOM); 1873 pEASData->masterGain = gain;
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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
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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
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H A D | eas_synth.h | 278 EAS_I16 gain; /* current gain */ member in struct:s_synth_voice_tag
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H A D | eas_public.c | 1820 * Set the master gain for the mix engine in 1dB increments 1824 * volume - the desired master gain (100 is max) 1837 EAS_I16 gain; local 1852 /* get gain offset */ 1859 gain = EAS_VolumeToGain(volume - STREAM_VOLUME_HEADROOM); 1862 return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_VOLUME, gain); 1872 gain = EAS_VolumeToGain(volume - STREAM_VOLUME_HEADROOM); 1873 pEASData->masterGain = gain;
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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
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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
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H A D | eas_synth.h | 278 EAS_I16 gain; /* current gain */ member in struct:s_synth_voice_tag
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H A D | eas_public.c | 1835 * Set the master gain for the mix engine in 1dB increments 1839 * volume - the desired master gain (100 is max) 1852 EAS_I16 gain; local 1867 /* get gain offset */ 1874 gain = EAS_VolumeToGain(volume - STREAM_VOLUME_HEADROOM); 1877 return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_VOLUME, gain); 1887 gain = EAS_VolumeToGain(volume - STREAM_VOLUME_HEADROOM); 1888 pEASData->masterGain = gain;
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H A D | eas_mdls.c | 219 EAS_I32 gain; member in struct:__anon30964 354 0, /* Mod LFO to gain: 0 dB */ 355 0, /* Mod LFO CC1 to gain: 0 dB */ 356 0, /* Mod LFO channel pressure to gain: 0 dB */ 357 960, /* Velocity to gain: 96 dB */ 955 p->gain = 0; 1086 /* get gain */ 1087 if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &p->gain, EAS_FALSE)) != EAS_SUCCESS) 1089 if (p->gain > 0) 1091 { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Positive gain [ [all...] |
/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
H A D | pitch_filter.c | 51 WebRtc_Word16 gain, 98 // Get old lag and gain value from memory.
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/external/webrtc/src/modules/audio_processing/interface/ |
H A D | audio_processing.h | 164 // necessary, to provide if gain control is enabled. On the server-side this 395 // The automatic gain control (AGC) component brings the signal to an 396 // appropriate range. This is done by applying a digital gain directly and, in 397 // the analog mode, prescribing an analog gain to be applied at the audio HAL. 421 // It consists of an analog gain prescription for the audio device and a 435 // short time-window of the input signal. It applies a fixed gain through 436 // most of the input level range, and compresses (gradually reduces gain 439 // predictable, so that a known gain can be applied. 456 // Sets the maximum |gain| the digital compression stage may apply, in dB. A 459 virtual int set_compression_gain_db(int gain) [all...] |
/external/chromium_org/chromeos/audio/ |
H A D | cras_audio_handler.cc | 316 // TODO(jennyz): Should we set all input devices' gain to the same level? 535 // TODO(rkc,jennyz): Set input gain once we decide on how to store 536 // the gain values since the range and step are both device specific. 569 // devices. For input devices, we don't restore their gain value so far. 571 // we should persist input gain value in prefs. 644 void CrasAudioHandler::SetInputNodeGain(uint64 node_id, int gain) { argument 646 SetInputNodeGain(node_id, gain); 655 // NOTE: We do not sanitize input gain values since the range is completely
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/external/chromium_org/third_party/WebKit/Source/platform/graphics/filters/ |
H A D | FEConvolveMatrix.cpp | 525 SkScalar gain = SkFloatToScalar(1.0f / m_divisor); local 534 return adoptRef(SkMatrixConvolutionImageFilter::Create(kernelSize, kernel.get(), gain, bias, target, tileMode, convolveAlpha, input.get(), &cropRect));
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/external/pdfium/core/src/fxcodec/fx_libopenjpeg/libopenjpeg20/ |
H A D | dwt.c | 480 /* Get gain of 5-3 wavelet transform. */ 506 /* Get gain of 9-7 wavelet transform. */ 525 OPJ_UINT32 resno, level, orient, gain; local 530 gain = (tccp->qmfbid == 0) ? 0 : ((orient == 0) ? 0 : (((orient == 1) || (orient == 2)) ? 1 : 2)); 535 stepsize = (1 << (gain)) / norm; 537 opj_dwt_encode_stepsize((OPJ_INT32) floor(stepsize * 8192.0), (OPJ_INT32)(prec + gain), &tccp->stepsizes[bandno]);
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/external/chromium_org/third_party/opus/src/src/ |
H A D | opus_decoder.c | 555 opus_val32 gain; local 556 gain = celt_exp2(MULT16_16_P15(QCONST16(6.48814081e-4f, 25), st->decode_gain)); 560 x = MULT16_32_P16(pcm[i],gain);
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/external/libopus/src/ |
H A D | opus_decoder.c | 555 opus_val32 gain; local 556 gain = celt_exp2(MULT16_16_P15(QCONST16(6.48814081e-4f, 25), st->decode_gain)); 560 x = MULT16_32_P16(pcm[i],gain);
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