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  /external/libvpx/vp8/encoder/x86/
quantize_mmx.asm 56 psubw mm3, mm0 ;gain the sign back
96 psubw mm7, mm4;gain the sign back
137 psubw mm7, mm4;gain the sign back
178 psubw mm7, mm4;gain the sign back
  /external/sonivox/arm-fm-22k/lib_src/
eas_pan.c 7 * Calculates left and right gain multipliers based on a pan value from -63 to +63
41 * Assign the left and right gain values corresponding to the given pan value.
55 * pGainLeft linear gain multiplier for left channel (15-bit fraction)
56 * pGainRight linear gain multiplier for left channel (15-bit fraction)
eas_pan.h 7 * Calculates left and right gain multipliers based on a pan value from -63 to +63
43 * Assign the left and right gain values corresponding to the given pan value.
57 * pGainLeft linear gain multiplier for left channel (15-bit fraction)
58 * pGainRight linear gain multiplier for left channel (15-bit fraction)
  /external/sonivox/arm-hybrid-22k/lib_src/
eas_pan.c 7 * Calculates left and right gain multipliers based on a pan value from -63 to +63
41 * Assign the left and right gain values corresponding to the given pan value.
55 * pGainLeft linear gain multiplier for left channel (15-bit fraction)
56 * pGainRight linear gain multiplier for left channel (15-bit fraction)
eas_pan.h 7 * Calculates left and right gain multipliers based on a pan value from -63 to +63
43 * Assign the left and right gain values corresponding to the given pan value.
57 * pGainLeft linear gain multiplier for left channel (15-bit fraction)
58 * pGainRight linear gain multiplier for left channel (15-bit fraction)
eas_wt_IPC_frame.h 74 EAS_I16 gainLeft; /* left channel gain */
75 EAS_I16 gainRight; /* right channel gain */
78 EAS_I16 gain; /* current voice gain */ member in struct:s_wt_config_tag
eas_wtengine.c 65 * Output gain for individual voice
78 EAS_I32 gain; local
99 gain = pWTIntFrame->prevGain << 16;
108 /* incremental gain step to prevent zipper noise */
110 gain += gainIncrement;
112 tmp2 = gain >> 16;
114 /* scale sample by gain */
464 * optimizations. It calls the interpolator, filter, and gain routines
535 /* apply gain, and left and right gain */
562 EAS_I32 gain; local
    [all...]
  /external/sonivox/arm-wt-22k/lib_src/
eas_pan.c 7 * Calculates left and right gain multipliers based on a pan value from -63 to +63
41 * Assign the left and right gain values corresponding to the given pan value.
55 * pGainLeft linear gain multiplier for left channel (15-bit fraction)
56 * pGainRight linear gain multiplier for left channel (15-bit fraction)
eas_pan.h 7 * Calculates left and right gain multipliers based on a pan value from -63 to +63
43 * Assign the left and right gain values corresponding to the given pan value.
57 * pGainLeft linear gain multiplier for left channel (15-bit fraction)
58 * pGainRight linear gain multiplier for left channel (15-bit fraction)
eas_wt_IPC_frame.h 74 EAS_I16 gainLeft; /* left channel gain */
75 EAS_I16 gainRight; /* right channel gain */
78 EAS_I16 gain; /* current voice gain */ member in struct:s_wt_config_tag
eas_wtengine.c 65 * Output gain for individual voice
78 EAS_I32 gain; local
99 gain = pWTIntFrame->prevGain << 16;
108 /* incremental gain step to prevent zipper noise */
110 gain += gainIncrement;
112 tmp2 = gain >> 16;
114 /* scale sample by gain */
464 * optimizations. It calls the interpolator, filter, and gain routines
535 /* apply gain, and left and right gain */
562 EAS_I32 gain; local
    [all...]
  /external/webkit/Source/WebCore/webaudio/
AudioGainNode.idl 31 readonly attribute AudioGain gain;
  /frameworks/base/media/libeffects/lvm/lib/SpectrumAnalyzer/src/
LVPSA_QPD.h 52 /* Apply downsampling, post gain, quasi peak filtering and write the levels values */
LVPSA_Tables.c 128 * Gain table
130 const LVM_INT16 LVPSA_GainTable[] = { 364, /* -15dB gain */
145 2048, /* 0dB gain */
160 11576}; /* +15dB gain */
  /frameworks/base/media/libeffects/lvm/lib/StereoWidening/src/
LVCS_StereoEnhancer.h 56 LVM_UINT16 MidGain; /* Middle gain in mobile speaker mode */
LVCS_Headphone_Coeffs.h 244 /* Reverb Gain Settings */
245 #define LVCS_HEADPHONE_DELAYGAIN 0.800000 /* Algorithm delay path gain */
246 #define LVCS_HEADPHONE_OUTPUTGAIN 1.000000 /* Algorithm output gain */
247 #define LVCS_HEADPHONE_PROCGAIN 18403 /* Processed path gain */
248 #define LVCS_HEADPHONE_UNPROCGAIN 18403 /* Unprocessed path gain */
249 #define LVCS_HEADPHONE_GAINCORRECT 1.009343 /* Delay mixer gain correction */
387 /* The Output Gain Correction */
393 #define LVCS_HEADPHONE_GAIN 6840 /* Unprocessed path gain */
396 #define LVCS_EX_HEADPHONE_GAIN 5108 /* EX Unprocessed path gain */
  /frameworks/base/media/libstagefright/codecs/amrwbenc/inc/
acelp.h 373 * gain prototypes *
376 Word16 G_pitch( /* (o) Q14 : Gain of pitch lag saturated to 1.2 */
379 Word16 g_coeff[], /* : Correlations need for gain quantization. */
395 Word16 * gain_pit, /* (i/o)Q14:Pitch gain. */
396 Word32 * gain_cod, /* (o) Q16 :Code gain. */
409 Word16 * gain_pit, /* (o) Q14 :Pitch gain. */
410 Word32 * gain_cod, /* (o) Q16 :Code gain. */
479 Word16 gain, /* (i) Q14 : adaptive codebook gain */
485 Word16 gain_pit, /* (i) Q14 : gain of pitch *
    [all...]
  /prebuilt/linux-x86/toolchain/i686-linux-glibc2.7-4.4.3/sysroot/usr/include/linux/
ixjuser.h 164 * The gain values range from 0 to 15 indicating +6dB to -24dB in 2dB
318 * table indexes are used in a cadence the frequency and gain variables will
469 * The input volume controls use gain with 32 levels from +12dB to -50dB
576 * The DAA Analog GAIN sets 2 parameters at one time, the receive gain (AGRR),
577 * and the transmit gain (AGX). OR together the components and pass them
583 #define AGRR00DB 0x00 /* Analog gain in receive direction 0dB */
584 #define AGRR3_5DB 0x10 /* Analog gain in receive direction 3.5dB */
585 #define AGRR06DB 0x30 /* Analog gain in receive direction 6dB */
587 #define AGX00DB 0x00 /* Analog gain in transmit direction 0dB *
    [all...]
  /system/media/audio_effects/include/audio_effects/
effect_agc.h 36 AGC_PARAM_COMP_GAIN, // gain in the compression range in millibel
  /frameworks/base/media/libstagefright/codecs/amrwbenc/src/
q_gain2.c 24 * An initial predicted gain, g_0, is first determined and the correction *
25 * factor alpha = gain / g_0 is quantized. *
26 * The pitch gain and the correction factor are vector quantized and the *
72 Word16 * gain_pit, /* (i/o)Q14: Pitch gain. */
73 Word32 * gain_cod, /* (o) Q16 : Code gain. */
100 size = size - 16; /* limit gain pitch to 1.0 */
112 j = j - 27; /* limit gain pitch to 1.0 */
313 *gain_pit = *p++; /* selected pitch gain in Q14 */
314 g_code = *p++; /* selected code gain in Q11 */
319 *gain_cod = L_tmp; /* gain of code in Q16 *
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dtx.c 167 Word16 log_en, gain, level, exp, exp0, tmp; local
238 /* the result corresponds to log2(gain) in Q10 */
247 /* Subtract 2 from log_en in Q9, i.e divide the gain by 2 (energy by 4) */
263 /* gain = level / sqrt(ener) * sqrt(L_FRAME) */
270 gain = extract_h(ener32);
272 gain = mult(level, gain); /* gain in Q15 */
281 tmp = mult(exc2[i], gain); /* Q0 * Q15 */
  /external/speex/libspeex/
nb_celp.h 93 spx_word32_t *pi_gain; /**< Gain of LPC filter at theta=pi (fe/2) */
135 spx_word16_t last_ol_gain; /**< Open-loop gain for previous frame */
144 spx_word32_t *pi_gain; /**< Gain of LPC filter at theta=pi (fe/2) */
153 spx_word16_t last_pitch_gain; /**< Pitch gain of last correctly decoded frame */
154 spx_word16_t pitch_gain_buf[3]; /**< Pitch gain of last decoded frames */
  /external/bluetooth/bluez/audio/
headset.c 949 static int headset_set_gain(struct audio_device *device, uint16_t gain, char type)
955 if (gain > 15) {
956 error("Invalid gain value: %u", gain);
962 if (slc->sp_gain == gain) {
963 DBG("Ignoring no-change in speaker gain");
968 slc->sp_gain = gain;
971 if (slc->mic_gain == gain) {
972 DBG("Ignoring no-change in microphone gain");
977 slc->mic_gain = gain;
999 dbus_uint16_t gain; local
1862 dbus_uint16_t gain; local
1887 dbus_uint16_t gain; local
1940 uint16_t gain; local
1953 uint16_t gain; local
2022 uint16_t gain; local
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  /frameworks/base/media/libstagefright/codecs/amrnb/enc/src/
g_adapt.h 107 Word16 prev_gc; /* previous code gain, Q1 */
109 Word16 ltpg_mem[LTPG_MEM_SIZE]; /* LTP coding gain history, Q13 */
118 /* initialize one instance of the gain adaptor
125 /* reset of gain adaptor state (i.e. set state memory to zero)
130 /* de-initialize gain adaptor state (i.e. free state struct)
137 * Purpose: calculate pitch/codebook gain adaptation factor alpha
144 Word16 ltpg, /* i : ltp coding gain (log2()), Q */
145 Word16 gain_cod, /* i : code gain, Q13 */
146 Word16 *alpha, /* o : gain adaptation factor, Q15 */
qgain795.h 108 GainAdaptState *adapt_st, /* i/o: gain adapter state structure */
117 Word16 exp_gcode0, /* i : predicted CB gain (exponent), Q0 */
118 Word16 frac_gcode0, /* i : predicted CB gain (fraction), Q15 */
120 Word16 cod_gain_frac, /* i : opt. codebook gain (fraction),Q15 */
121 Word16 cod_gain_exp, /* i : opt. codebook gain (exponent), Q0 */
122 Word16 gp_limit, /* i : pitch gain limit */
123 Word16 *gain_pit, /* i/o: Pitch gain (unquant/quant), Q14 */
124 Word16 *gain_cod, /* o : Code gain, Q1 */
130 /* (first gain pitch, then code pitch)*/

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