Index: webrtc/modules/audio_processing/agc/legacy/digital_agc.c |
diff --git a/webrtc/modules/audio_processing/agc/legacy/digital_agc.c b/webrtc/modules/audio_processing/agc/legacy/digital_agc.c |
index 0881af11dbf4566820100f99c322081fe9d97da5..2ca967a4aae1a0d0a00758b4861a4838b1d11d46 100644 |
--- a/webrtc/modules/audio_processing/agc/legacy/digital_agc.c |
+++ b/webrtc/modules/audio_processing/agc/legacy/digital_agc.c |
@@ -27,269 +27,254 @@ |
// zeros = 0:31; lvl = 2.^(1-zeros); |
// A = -10*log10(lvl) * (CompRatio - 1) / CompRatio; |
// B = MaxGain - MinGain; |
-// gains = round(2^16*10.^(0.05 * (MinGain + B * ( log(exp(-Knee*A)+exp(-Knee*B)) - log(1+exp(-Knee*B)) ) / log(1/(1+exp(Knee*B)))))); |
+// gains = round(2^16*10.^(0.05 * (MinGain + B * ( |
+// log(exp(-Knee*A)+exp(-Knee*B)) - log(1+exp(-Knee*B)) ) / |
+// log(1/(1+exp(Knee*B)))))); |
// fprintf(1, '\t%i, %i, %i, %i,\n', gains); |
-// % Matlab code for plotting the gain and input/output level characteristic (copy/paste the following 3 lines): |
+// % Matlab code for plotting the gain and input/output level characteristic |
+// (copy/paste the following 3 lines): |
// in = 10*log10(lvl); out = 20*log10(gains/65536); |
-// subplot(121); plot(in, out); axis([-30, 0, -5, 20]); grid on; xlabel('Input (dB)'); ylabel('Gain (dB)'); |
-// subplot(122); plot(in, in+out); axis([-30, 0, -30, 5]); grid on; xlabel('Input (dB)'); ylabel('Output (dB)'); |
+// subplot(121); plot(in, out); axis([-30, 0, -5, 20]); grid on; xlabel('Input |
+// (dB)'); ylabel('Gain (dB)'); |
+// subplot(122); plot(in, in+out); axis([-30, 0, -30, 5]); grid on; |
+// xlabel('Input (dB)'); ylabel('Output (dB)'); |
// zoom on; |
// Generator table for y=log2(1+e^x) in Q8. |
enum { kGenFuncTableSize = 128 }; |
static const uint16_t kGenFuncTable[kGenFuncTableSize] = { |
- 256, 485, 786, 1126, 1484, 1849, 2217, 2586, |
- 2955, 3324, 3693, 4063, 4432, 4801, 5171, 5540, |
- 5909, 6279, 6648, 7017, 7387, 7756, 8125, 8495, |
- 8864, 9233, 9603, 9972, 10341, 10711, 11080, 11449, |
- 11819, 12188, 12557, 12927, 13296, 13665, 14035, 14404, |
- 14773, 15143, 15512, 15881, 16251, 16620, 16989, 17359, |
- 17728, 18097, 18466, 18836, 19205, 19574, 19944, 20313, |
- 20682, 21052, 21421, 21790, 22160, 22529, 22898, 23268, |
- 23637, 24006, 24376, 24745, 25114, 25484, 25853, 26222, |
- 26592, 26961, 27330, 27700, 28069, 28438, 28808, 29177, |
- 29546, 29916, 30285, 30654, 31024, 31393, 31762, 32132, |
- 32501, 32870, 33240, 33609, 33978, 34348, 34717, 35086, |
- 35456, 35825, 36194, 36564, 36933, 37302, 37672, 38041, |
- 38410, 38780, 39149, 39518, 39888, 40257, 40626, 40996, |
- 41365, 41734, 42104, 42473, 42842, 43212, 43581, 43950, |
- 44320, 44689, 45058, 45428, 45797, 46166, 46536, 46905 |
-}; |
- |
-static const int16_t kAvgDecayTime = 250; // frames; < 3000 |
- |
-int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16 |
- int16_t digCompGaindB, // Q0 |
- int16_t targetLevelDbfs,// Q0 |
+ 256, 485, 786, 1126, 1484, 1849, 2217, 2586, 2955, 3324, 3693, |
+ 4063, 4432, 4801, 5171, 5540, 5909, 6279, 6648, 7017, 7387, 7756, |
+ 8125, 8495, 8864, 9233, 9603, 9972, 10341, 10711, 11080, 11449, 11819, |
+ 12188, 12557, 12927, 13296, 13665, 14035, 14404, 14773, 15143, 15512, 15881, |
+ 16251, 16620, 16989, 17359, 17728, 18097, 18466, 18836, 19205, 19574, 19944, |
+ 20313, 20682, 21052, 21421, 21790, 22160, 22529, 22898, 23268, 23637, 24006, |
+ 24376, 24745, 25114, 25484, 25853, 26222, 26592, 26961, 27330, 27700, 28069, |
+ 28438, 28808, 29177, 29546, 29916, 30285, 30654, 31024, 31393, 31762, 32132, |
+ 32501, 32870, 33240, 33609, 33978, 34348, 34717, 35086, 35456, 35825, 36194, |
+ 36564, 36933, 37302, 37672, 38041, 38410, 38780, 39149, 39518, 39888, 40257, |
+ 40626, 40996, 41365, 41734, 42104, 42473, 42842, 43212, 43581, 43950, 44320, |
+ 44689, 45058, 45428, 45797, 46166, 46536, 46905}; |
+ |
+static const int16_t kAvgDecayTime = 250; // frames; < 3000 |
+ |
+int32_t WebRtcAgc_CalculateGainTable(int32_t* gainTable, // Q16 |
+ int16_t digCompGaindB, // Q0 |
+ int16_t targetLevelDbfs, // Q0 |
uint8_t limiterEnable, |
- int16_t analogTarget) // Q0 |
+ int16_t analogTarget) // Q0 |
{ |
- // This function generates the compressor gain table used in the fixed digital part. |
- uint32_t tmpU32no1, tmpU32no2, absInLevel, logApprox; |
- int32_t inLevel, limiterLvl; |
- int32_t tmp32, tmp32no1, tmp32no2, numFIX, den, y32; |
- const uint16_t kLog10 = 54426; // log2(10) in Q14 |
- const uint16_t kLog10_2 = 49321; // 10*log10(2) in Q14 |
- const uint16_t kLogE_1 = 23637; // log2(e) in Q14 |
- uint16_t constMaxGain; |
- uint16_t tmpU16, intPart, fracPart; |
- const int16_t kCompRatio = 3; |
- const int16_t kSoftLimiterLeft = 1; |
- int16_t limiterOffset = 0; // Limiter offset |
- int16_t limiterIdx, limiterLvlX; |
- int16_t constLinApprox, zeroGainLvl, maxGain, diffGain; |
- int16_t i, tmp16, tmp16no1; |
- int zeros, zerosScale; |
- |
- // Constants |
-// kLogE_1 = 23637; // log2(e) in Q14 |
-// kLog10 = 54426; // log2(10) in Q14 |
-// kLog10_2 = 49321; // 10*log10(2) in Q14 |
- |
- // Calculate maximum digital gain and zero gain level |
- tmp32no1 = (digCompGaindB - analogTarget) * (kCompRatio - 1); |
- tmp16no1 = analogTarget - targetLevelDbfs; |
- tmp16no1 += WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio); |
- maxGain = WEBRTC_SPL_MAX(tmp16no1, (analogTarget - targetLevelDbfs)); |
- tmp32no1 = maxGain * kCompRatio; |
- zeroGainLvl = digCompGaindB; |
- zeroGainLvl -= WebRtcSpl_DivW32W16ResW16(tmp32no1 + ((kCompRatio - 1) >> 1), |
- kCompRatio - 1); |
- if ((digCompGaindB <= analogTarget) && (limiterEnable)) |
- { |
- zeroGainLvl += (analogTarget - digCompGaindB + kSoftLimiterLeft); |
- limiterOffset = 0; |
+ // This function generates the compressor gain table used in the fixed digital |
+ // part. |
+ uint32_t tmpU32no1, tmpU32no2, absInLevel, logApprox; |
+ int32_t inLevel, limiterLvl; |
+ int32_t tmp32, tmp32no1, tmp32no2, numFIX, den, y32; |
+ const uint16_t kLog10 = 54426; // log2(10) in Q14 |
+ const uint16_t kLog10_2 = 49321; // 10*log10(2) in Q14 |
+ const uint16_t kLogE_1 = 23637; // log2(e) in Q14 |
+ uint16_t constMaxGain; |
+ uint16_t tmpU16, intPart, fracPart; |
+ const int16_t kCompRatio = 3; |
+ const int16_t kSoftLimiterLeft = 1; |
+ int16_t limiterOffset = 0; // Limiter offset |
+ int16_t limiterIdx, limiterLvlX; |
+ int16_t constLinApprox, zeroGainLvl, maxGain, diffGain; |
+ int16_t i, tmp16, tmp16no1; |
+ int zeros, zerosScale; |
+ |
+ // Constants |
+ // kLogE_1 = 23637; // log2(e) in Q14 |
+ // kLog10 = 54426; // log2(10) in Q14 |
+ // kLog10_2 = 49321; // 10*log10(2) in Q14 |
+ |
+ // Calculate maximum digital gain and zero gain level |
+ tmp32no1 = (digCompGaindB - analogTarget) * (kCompRatio - 1); |
+ tmp16no1 = analogTarget - targetLevelDbfs; |
+ tmp16no1 += |
+ WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio); |
+ maxGain = WEBRTC_SPL_MAX(tmp16no1, (analogTarget - targetLevelDbfs)); |
+ tmp32no1 = maxGain * kCompRatio; |
+ zeroGainLvl = digCompGaindB; |
+ zeroGainLvl -= WebRtcSpl_DivW32W16ResW16(tmp32no1 + ((kCompRatio - 1) >> 1), |
+ kCompRatio - 1); |
+ if ((digCompGaindB <= analogTarget) && (limiterEnable)) { |
+ zeroGainLvl += (analogTarget - digCompGaindB + kSoftLimiterLeft); |
+ limiterOffset = 0; |
+ } |
+ |
+ // Calculate the difference between maximum gain and gain at 0dB0v: |
+ // diffGain = maxGain + (compRatio-1)*zeroGainLvl/compRatio |
+ // = (compRatio-1)*digCompGaindB/compRatio |
+ tmp32no1 = digCompGaindB * (kCompRatio - 1); |
+ diffGain = |
+ WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio); |
+ if (diffGain < 0 || diffGain >= kGenFuncTableSize) { |
+ assert(0); |
+ return -1; |
+ } |
+ |
+ // Calculate the limiter level and index: |
+ // limiterLvlX = analogTarget - limiterOffset |
+ // limiterLvl = targetLevelDbfs + limiterOffset/compRatio |
+ limiterLvlX = analogTarget - limiterOffset; |
+ limiterIdx = 2 + WebRtcSpl_DivW32W16ResW16((int32_t)limiterLvlX * (1 << 13), |
+ kLog10_2 / 2); |
+ tmp16no1 = |
+ WebRtcSpl_DivW32W16ResW16(limiterOffset + (kCompRatio >> 1), kCompRatio); |
+ limiterLvl = targetLevelDbfs + tmp16no1; |
+ |
+ // Calculate (through table lookup): |
+ // constMaxGain = log2(1+2^(log2(e)*diffGain)); (in Q8) |
+ constMaxGain = kGenFuncTable[diffGain]; // in Q8 |
+ |
+ // Calculate a parameter used to approximate the fractional part of 2^x with a |
+ // piecewise linear function in Q14: |
+ // constLinApprox = round(3/2*(4*(3-2*sqrt(2))/(log(2)^2)-0.5)*2^14); |
+ constLinApprox = 22817; // in Q14 |
+ |
+ // Calculate a denominator used in the exponential part to convert from dB to |
+ // linear scale: |
+ // den = 20*constMaxGain (in Q8) |
+ den = WEBRTC_SPL_MUL_16_U16(20, constMaxGain); // in Q8 |
+ |
+ for (i = 0; i < 32; i++) { |
+ // Calculate scaled input level (compressor): |
+ // inLevel = |
+ // fix((-constLog10_2*(compRatio-1)*(1-i)+fix(compRatio/2))/compRatio) |
+ tmp16 = (int16_t)((kCompRatio - 1) * (i - 1)); // Q0 |
+ tmp32 = WEBRTC_SPL_MUL_16_U16(tmp16, kLog10_2) + 1; // Q14 |
+ inLevel = WebRtcSpl_DivW32W16(tmp32, kCompRatio); // Q14 |
+ |
+ // Calculate diffGain-inLevel, to map using the genFuncTable |
+ inLevel = (int32_t)diffGain * (1 << 14) - inLevel; // Q14 |
+ |
+ // Make calculations on abs(inLevel) and compensate for the sign afterwards. |
+ absInLevel = (uint32_t)WEBRTC_SPL_ABS_W32(inLevel); // Q14 |
+ |
+ // LUT with interpolation |
+ intPart = (uint16_t)(absInLevel >> 14); |
+ fracPart = |
+ (uint16_t)(absInLevel & 0x00003FFF); // extract the fractional part |
+ tmpU16 = kGenFuncTable[intPart + 1] - kGenFuncTable[intPart]; // Q8 |
+ tmpU32no1 = tmpU16 * fracPart; // Q22 |
+ tmpU32no1 += (uint32_t)kGenFuncTable[intPart] << 14; // Q22 |
+ logApprox = tmpU32no1 >> 8; // Q14 |
+ // Compensate for negative exponent using the relation: |
+ // log2(1 + 2^-x) = log2(1 + 2^x) - x |
+ if (inLevel < 0) { |
+ zeros = WebRtcSpl_NormU32(absInLevel); |
+ zerosScale = 0; |
+ if (zeros < 15) { |
+ // Not enough space for multiplication |
+ tmpU32no2 = absInLevel >> (15 - zeros); // Q(zeros-1) |
+ tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no2, kLogE_1); // Q(zeros+13) |
+ if (zeros < 9) { |
+ zerosScale = 9 - zeros; |
+ tmpU32no1 >>= zerosScale; // Q(zeros+13) |
+ } else { |
+ tmpU32no2 >>= zeros - 9; // Q22 |
+ } |
+ } else { |
+ tmpU32no2 = WEBRTC_SPL_UMUL_32_16(absInLevel, kLogE_1); // Q28 |
+ tmpU32no2 >>= 6; // Q22 |
+ } |
+ logApprox = 0; |
+ if (tmpU32no2 < tmpU32no1) { |
+ logApprox = (tmpU32no1 - tmpU32no2) >> (8 - zerosScale); // Q14 |
+ } |
} |
+ numFIX = (maxGain * constMaxGain) * (1 << 6); // Q14 |
+ numFIX -= (int32_t)logApprox * diffGain; // Q14 |
- // Calculate the difference between maximum gain and gain at 0dB0v: |
- // diffGain = maxGain + (compRatio-1)*zeroGainLvl/compRatio |
- // = (compRatio-1)*digCompGaindB/compRatio |
- tmp32no1 = digCompGaindB * (kCompRatio - 1); |
- diffGain = WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio); |
- if (diffGain < 0 || diffGain >= kGenFuncTableSize) |
+ // Calculate ratio |
+ // Shift |numFIX| as much as possible. |
+ // Ensure we avoid wrap-around in |den| as well. |
+ if (numFIX > (den >> 8)) // |den| is Q8. |
{ |
- assert(0); |
- return -1; |
+ zeros = WebRtcSpl_NormW32(numFIX); |
+ } else { |
+ zeros = WebRtcSpl_NormW32(den) + 8; |
} |
- |
- // Calculate the limiter level and index: |
- // limiterLvlX = analogTarget - limiterOffset |
- // limiterLvl = targetLevelDbfs + limiterOffset/compRatio |
- limiterLvlX = analogTarget - limiterOffset; |
- limiterIdx = |
- 2 + WebRtcSpl_DivW32W16ResW16((int32_t)limiterLvlX * (1 << 13), |
- kLog10_2 / 2); |
- tmp16no1 = WebRtcSpl_DivW32W16ResW16(limiterOffset + (kCompRatio >> 1), kCompRatio); |
- limiterLvl = targetLevelDbfs + tmp16no1; |
- |
- // Calculate (through table lookup): |
- // constMaxGain = log2(1+2^(log2(e)*diffGain)); (in Q8) |
- constMaxGain = kGenFuncTable[diffGain]; // in Q8 |
- |
- // Calculate a parameter used to approximate the fractional part of 2^x with a |
- // piecewise linear function in Q14: |
- // constLinApprox = round(3/2*(4*(3-2*sqrt(2))/(log(2)^2)-0.5)*2^14); |
- constLinApprox = 22817; // in Q14 |
- |
- // Calculate a denominator used in the exponential part to convert from dB to linear scale: |
- // den = 20*constMaxGain (in Q8) |
- den = WEBRTC_SPL_MUL_16_U16(20, constMaxGain); // in Q8 |
- |
- for (i = 0; i < 32; i++) |
- { |
- // Calculate scaled input level (compressor): |
- // inLevel = fix((-constLog10_2*(compRatio-1)*(1-i)+fix(compRatio/2))/compRatio) |
- tmp16 = (int16_t)((kCompRatio - 1) * (i - 1)); // Q0 |
- tmp32 = WEBRTC_SPL_MUL_16_U16(tmp16, kLog10_2) + 1; // Q14 |
- inLevel = WebRtcSpl_DivW32W16(tmp32, kCompRatio); // Q14 |
- |
- // Calculate diffGain-inLevel, to map using the genFuncTable |
- inLevel = (int32_t)diffGain * (1 << 14) - inLevel; // Q14 |
- |
- // Make calculations on abs(inLevel) and compensate for the sign afterwards. |
- absInLevel = (uint32_t)WEBRTC_SPL_ABS_W32(inLevel); // Q14 |
- |
- // LUT with interpolation |
- intPart = (uint16_t)(absInLevel >> 14); |
- fracPart = (uint16_t)(absInLevel & 0x00003FFF); // extract the fractional part |
- tmpU16 = kGenFuncTable[intPart + 1] - kGenFuncTable[intPart]; // Q8 |
- tmpU32no1 = tmpU16 * fracPart; // Q22 |
- tmpU32no1 += (uint32_t)kGenFuncTable[intPart] << 14; // Q22 |
- logApprox = tmpU32no1 >> 8; // Q14 |
- // Compensate for negative exponent using the relation: |
- // log2(1 + 2^-x) = log2(1 + 2^x) - x |
- if (inLevel < 0) |
- { |
- zeros = WebRtcSpl_NormU32(absInLevel); |
- zerosScale = 0; |
- if (zeros < 15) |
- { |
- // Not enough space for multiplication |
- tmpU32no2 = absInLevel >> (15 - zeros); // Q(zeros-1) |
- tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no2, kLogE_1); // Q(zeros+13) |
- if (zeros < 9) |
- { |
- zerosScale = 9 - zeros; |
- tmpU32no1 >>= zerosScale; // Q(zeros+13) |
- } else |
- { |
- tmpU32no2 >>= zeros - 9; // Q22 |
- } |
- } else |
- { |
- tmpU32no2 = WEBRTC_SPL_UMUL_32_16(absInLevel, kLogE_1); // Q28 |
- tmpU32no2 >>= 6; // Q22 |
- } |
- logApprox = 0; |
- if (tmpU32no2 < tmpU32no1) |
- { |
- logApprox = (tmpU32no1 - tmpU32no2) >> (8 - zerosScale); //Q14 |
- } |
- } |
- numFIX = (maxGain * constMaxGain) * (1 << 6); // Q14 |
- numFIX -= (int32_t)logApprox * diffGain; // Q14 |
- |
- // Calculate ratio |
- // Shift |numFIX| as much as possible. |
- // Ensure we avoid wrap-around in |den| as well. |
- if (numFIX > (den >> 8)) // |den| is Q8. |
- { |
- zeros = WebRtcSpl_NormW32(numFIX); |
- } else |
- { |
- zeros = WebRtcSpl_NormW32(den) + 8; |
- } |
- numFIX *= 1 << zeros; // Q(14+zeros) |
- |
- // Shift den so we end up in Qy1 |
- tmp32no1 = WEBRTC_SPL_SHIFT_W32(den, zeros - 8); // Q(zeros) |
- if (numFIX < 0) |
- { |
- numFIX -= tmp32no1 / 2; |
- } else |
- { |
- numFIX += tmp32no1 / 2; |
- } |
- y32 = numFIX / tmp32no1; // in Q14 |
- if (limiterEnable && (i < limiterIdx)) |
- { |
- tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2); // Q14 |
- tmp32 -= limiterLvl * (1 << 14); // Q14 |
- y32 = WebRtcSpl_DivW32W16(tmp32 + 10, 20); |
- } |
- if (y32 > 39000) |
- { |
- tmp32 = (y32 >> 1) * kLog10 + 4096; // in Q27 |
- tmp32 >>= 13; // In Q14. |
- } else |
- { |
- tmp32 = y32 * kLog10 + 8192; // in Q28 |
- tmp32 >>= 14; // In Q14. |
- } |
- tmp32 += 16 << 14; // in Q14 (Make sure final output is in Q16) |
- |
- // Calculate power |
- if (tmp32 > 0) |
- { |
- intPart = (int16_t)(tmp32 >> 14); |
- fracPart = (uint16_t)(tmp32 & 0x00003FFF); // in Q14 |
- if ((fracPart >> 13) != 0) |
- { |
- tmp16 = (2 << 14) - constLinApprox; |
- tmp32no2 = (1 << 14) - fracPart; |
- tmp32no2 *= tmp16; |
- tmp32no2 >>= 13; |
- tmp32no2 = (1 << 14) - tmp32no2; |
- } else |
- { |
- tmp16 = constLinApprox - (1 << 14); |
- tmp32no2 = (fracPart * tmp16) >> 13; |
- } |
- fracPart = (uint16_t)tmp32no2; |
- gainTable[i] = |
- (1 << intPart) + WEBRTC_SPL_SHIFT_W32(fracPart, intPart - 14); |
- } else |
- { |
- gainTable[i] = 0; |
- } |
+ numFIX *= 1 << zeros; // Q(14+zeros) |
+ |
+ // Shift den so we end up in Qy1 |
+ tmp32no1 = WEBRTC_SPL_SHIFT_W32(den, zeros - 8); // Q(zeros) |
+ if (numFIX < 0) { |
+ numFIX -= tmp32no1 / 2; |
+ } else { |
+ numFIX += tmp32no1 / 2; |
+ } |
+ y32 = numFIX / tmp32no1; // in Q14 |
+ if (limiterEnable && (i < limiterIdx)) { |
+ tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2); // Q14 |
+ tmp32 -= limiterLvl * (1 << 14); // Q14 |
+ y32 = WebRtcSpl_DivW32W16(tmp32 + 10, 20); |
} |
+ if (y32 > 39000) { |
+ tmp32 = (y32 >> 1) * kLog10 + 4096; // in Q27 |
+ tmp32 >>= 13; // In Q14. |
+ } else { |
+ tmp32 = y32 * kLog10 + 8192; // in Q28 |
+ tmp32 >>= 14; // In Q14. |
+ } |
+ tmp32 += 16 << 14; // in Q14 (Make sure final output is in Q16) |
+ |
+ // Calculate power |
+ if (tmp32 > 0) { |
+ intPart = (int16_t)(tmp32 >> 14); |
+ fracPart = (uint16_t)(tmp32 & 0x00003FFF); // in Q14 |
+ if ((fracPart >> 13) != 0) { |
+ tmp16 = (2 << 14) - constLinApprox; |
+ tmp32no2 = (1 << 14) - fracPart; |
+ tmp32no2 *= tmp16; |
+ tmp32no2 >>= 13; |
+ tmp32no2 = (1 << 14) - tmp32no2; |
+ } else { |
+ tmp16 = constLinApprox - (1 << 14); |
+ tmp32no2 = (fracPart * tmp16) >> 13; |
+ } |
+ fracPart = (uint16_t)tmp32no2; |
+ gainTable[i] = |
+ (1 << intPart) + WEBRTC_SPL_SHIFT_W32(fracPart, intPart - 14); |
+ } else { |
+ gainTable[i] = 0; |
+ } |
+ } |
- return 0; |
+ return 0; |
} |
int32_t WebRtcAgc_InitDigital(DigitalAgc* stt, int16_t agcMode) { |
- if (agcMode == kAgcModeFixedDigital) |
- { |
- // start at minimum to find correct gain faster |
- stt->capacitorSlow = 0; |
- } else |
- { |
- // start out with 0 dB gain |
- stt->capacitorSlow = 134217728; // (int32_t)(0.125f * 32768.0f * 32768.0f); |
- } |
- stt->capacitorFast = 0; |
- stt->gain = 65536; |
- stt->gatePrevious = 0; |
- stt->agcMode = agcMode; |
+ if (agcMode == kAgcModeFixedDigital) { |
+ // start at minimum to find correct gain faster |
+ stt->capacitorSlow = 0; |
+ } else { |
+ // start out with 0 dB gain |
+ stt->capacitorSlow = 134217728; // (int32_t)(0.125f * 32768.0f * 32768.0f); |
+ } |
+ stt->capacitorFast = 0; |
+ stt->gain = 65536; |
+ stt->gatePrevious = 0; |
+ stt->agcMode = agcMode; |
#ifdef WEBRTC_AGC_DEBUG_DUMP |
- stt->frameCounter = 0; |
+ stt->frameCounter = 0; |
#endif |
- // initialize VADs |
- WebRtcAgc_InitVad(&stt->vadNearend); |
- WebRtcAgc_InitVad(&stt->vadFarend); |
+ // initialize VADs |
+ WebRtcAgc_InitVad(&stt->vadNearend); |
+ WebRtcAgc_InitVad(&stt->vadFarend); |
- return 0; |
+ return 0; |
} |
int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* stt, |
const int16_t* in_far, |
size_t nrSamples) { |
- assert(stt != NULL); |
- // VAD for far end |
- WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples); |
+ assert(stt != NULL); |
+ // VAD for far end |
+ WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples); |
- return 0; |
+ return 0; |
} |
int32_t WebRtcAgc_ProcessDigital(DigitalAgc* stt, |
@@ -298,476 +283,408 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc* stt, |
int16_t* const* out, |
uint32_t FS, |
int16_t lowlevelSignal) { |
- // array for gains (one value per ms, incl start & end) |
- int32_t gains[11]; |
- |
- int32_t out_tmp, tmp32; |
- int32_t env[10]; |
- int32_t max_nrg; |
- int32_t cur_level; |
- int32_t gain32, delta; |
- int16_t logratio; |
- int16_t lower_thr, upper_thr; |
- int16_t zeros = 0, zeros_fast, frac = 0; |
- int16_t decay; |
- int16_t gate, gain_adj; |
- int16_t k; |
- size_t n, i, L; |
- int16_t L2; // samples/subframe |
- |
- // determine number of samples per ms |
- if (FS == 8000) |
- { |
- L = 8; |
- L2 = 3; |
- } else if (FS == 16000 || FS == 32000 || FS == 48000) |
- { |
- L = 16; |
- L2 = 4; |
- } else |
- { |
- return -1; |
- } |
- |
- for (i = 0; i < num_bands; ++i) |
- { |
- if (in_near[i] != out[i]) |
- { |
- // Only needed if they don't already point to the same place. |
- memcpy(out[i], in_near[i], 10 * L * sizeof(in_near[i][0])); |
- } |
- } |
- // VAD for near end |
- logratio = WebRtcAgc_ProcessVad(&stt->vadNearend, out[0], L * 10); |
- |
- // Account for far end VAD |
- if (stt->vadFarend.counter > 10) |
- { |
- tmp32 = 3 * logratio; |
- logratio = (int16_t)((tmp32 - stt->vadFarend.logRatio) >> 2); |
+ // array for gains (one value per ms, incl start & end) |
+ int32_t gains[11]; |
+ |
+ int32_t out_tmp, tmp32; |
+ int32_t env[10]; |
+ int32_t max_nrg; |
+ int32_t cur_level; |
+ int32_t gain32, delta; |
+ int16_t logratio; |
+ int16_t lower_thr, upper_thr; |
+ int16_t zeros = 0, zeros_fast, frac = 0; |
+ int16_t decay; |
+ int16_t gate, gain_adj; |
+ int16_t k; |
+ size_t n, i, L; |
+ int16_t L2; // samples/subframe |
+ |
+ // determine number of samples per ms |
+ if (FS == 8000) { |
+ L = 8; |
+ L2 = 3; |
+ } else if (FS == 16000 || FS == 32000 || FS == 48000) { |
+ L = 16; |
+ L2 = 4; |
+ } else { |
+ return -1; |
+ } |
+ |
+ for (i = 0; i < num_bands; ++i) { |
+ if (in_near[i] != out[i]) { |
+ // Only needed if they don't already point to the same place. |
+ memcpy(out[i], in_near[i], 10 * L * sizeof(in_near[i][0])); |
} |
- |
- // Determine decay factor depending on VAD |
- // upper_thr = 1.0f; |
- // lower_thr = 0.25f; |
- upper_thr = 1024; // Q10 |
- lower_thr = 0; // Q10 |
- if (logratio > upper_thr) |
- { |
- // decay = -2^17 / DecayTime; -> -65 |
- decay = -65; |
- } else if (logratio < lower_thr) |
- { |
- decay = 0; |
- } else |
- { |
- // decay = (int16_t)(((lower_thr - logratio) |
- // * (2^27/(DecayTime*(upper_thr-lower_thr)))) >> 10); |
- // SUBSTITUTED: 2^27/(DecayTime*(upper_thr-lower_thr)) -> 65 |
- tmp32 = (lower_thr - logratio) * 65; |
- decay = (int16_t)(tmp32 >> 10); |
+ } |
+ // VAD for near end |
+ logratio = WebRtcAgc_ProcessVad(&stt->vadNearend, out[0], L * 10); |
+ |
+ // Account for far end VAD |
+ if (stt->vadFarend.counter > 10) { |
+ tmp32 = 3 * logratio; |
+ logratio = (int16_t)((tmp32 - stt->vadFarend.logRatio) >> 2); |
+ } |
+ |
+ // Determine decay factor depending on VAD |
+ // upper_thr = 1.0f; |
+ // lower_thr = 0.25f; |
+ upper_thr = 1024; // Q10 |
+ lower_thr = 0; // Q10 |
+ if (logratio > upper_thr) { |
+ // decay = -2^17 / DecayTime; -> -65 |
+ decay = -65; |
+ } else if (logratio < lower_thr) { |
+ decay = 0; |
+ } else { |
+ // decay = (int16_t)(((lower_thr - logratio) |
+ // * (2^27/(DecayTime*(upper_thr-lower_thr)))) >> 10); |
+ // SUBSTITUTED: 2^27/(DecayTime*(upper_thr-lower_thr)) -> 65 |
+ tmp32 = (lower_thr - logratio) * 65; |
+ decay = (int16_t)(tmp32 >> 10); |
+ } |
+ |
+ // adjust decay factor for long silence (detected as low standard deviation) |
+ // This is only done in the adaptive modes |
+ if (stt->agcMode != kAgcModeFixedDigital) { |
+ if (stt->vadNearend.stdLongTerm < 4000) { |
+ decay = 0; |
+ } else if (stt->vadNearend.stdLongTerm < 8096) { |
+ // decay = (int16_t)(((stt->vadNearend.stdLongTerm - 4000) * decay) >> |
+ // 12); |
+ tmp32 = (stt->vadNearend.stdLongTerm - 4000) * decay; |
+ decay = (int16_t)(tmp32 >> 12); |
} |
- // adjust decay factor for long silence (detected as low standard deviation) |
- // This is only done in the adaptive modes |
- if (stt->agcMode != kAgcModeFixedDigital) |
- { |
- if (stt->vadNearend.stdLongTerm < 4000) |
- { |
- decay = 0; |
- } else if (stt->vadNearend.stdLongTerm < 8096) |
- { |
- // decay = (int16_t)(((stt->vadNearend.stdLongTerm - 4000) * decay) >> 12); |
- tmp32 = (stt->vadNearend.stdLongTerm - 4000) * decay; |
- decay = (int16_t)(tmp32 >> 12); |
- } |
- |
- if (lowlevelSignal != 0) |
- { |
- decay = 0; |
- } |
+ if (lowlevelSignal != 0) { |
+ decay = 0; |
} |
+ } |
#ifdef WEBRTC_AGC_DEBUG_DUMP |
- stt->frameCounter++; |
- fprintf(stt->logFile, |
- "%5.2f\t%d\t%d\t%d\t", |
- (float)(stt->frameCounter) / 100, |
- logratio, |
- decay, |
- stt->vadNearend.stdLongTerm); |
+ stt->frameCounter++; |
+ fprintf(stt->logFile, "%5.2f\t%d\t%d\t%d\t", (float)(stt->frameCounter) / 100, |
+ logratio, decay, stt->vadNearend.stdLongTerm); |
#endif |
- // Find max amplitude per sub frame |
- // iterate over sub frames |
- for (k = 0; k < 10; k++) |
- { |
- // iterate over samples |
- max_nrg = 0; |
- for (n = 0; n < L; n++) |
- { |
- int32_t nrg = out[0][k * L + n] * out[0][k * L + n]; |
- if (nrg > max_nrg) |
- { |
- max_nrg = nrg; |
- } |
- } |
- env[k] = max_nrg; |
+ // Find max amplitude per sub frame |
+ // iterate over sub frames |
+ for (k = 0; k < 10; k++) { |
+ // iterate over samples |
+ max_nrg = 0; |
+ for (n = 0; n < L; n++) { |
+ int32_t nrg = out[0][k * L + n] * out[0][k * L + n]; |
+ if (nrg > max_nrg) { |
+ max_nrg = nrg; |
+ } |
+ } |
+ env[k] = max_nrg; |
+ } |
+ |
+ // Calculate gain per sub frame |
+ gains[0] = stt->gain; |
+ for (k = 0; k < 10; k++) { |
+ // Fast envelope follower |
+ // decay time = -131000 / -1000 = 131 (ms) |
+ stt->capacitorFast = |
+ AGC_SCALEDIFF32(-1000, stt->capacitorFast, stt->capacitorFast); |
+ if (env[k] > stt->capacitorFast) { |
+ stt->capacitorFast = env[k]; |
+ } |
+ // Slow envelope follower |
+ if (env[k] > stt->capacitorSlow) { |
+ // increase capacitorSlow |
+ stt->capacitorSlow = AGC_SCALEDIFF32(500, (env[k] - stt->capacitorSlow), |
+ stt->capacitorSlow); |
+ } else { |
+ // decrease capacitorSlow |
+ stt->capacitorSlow = |
+ AGC_SCALEDIFF32(decay, stt->capacitorSlow, stt->capacitorSlow); |
} |
- // Calculate gain per sub frame |
- gains[0] = stt->gain; |
- for (k = 0; k < 10; k++) |
- { |
- // Fast envelope follower |
- // decay time = -131000 / -1000 = 131 (ms) |
- stt->capacitorFast = AGC_SCALEDIFF32(-1000, stt->capacitorFast, stt->capacitorFast); |
- if (env[k] > stt->capacitorFast) |
- { |
- stt->capacitorFast = env[k]; |
- } |
- // Slow envelope follower |
- if (env[k] > stt->capacitorSlow) |
- { |
- // increase capacitorSlow |
- stt->capacitorSlow |
- = AGC_SCALEDIFF32(500, (env[k] - stt->capacitorSlow), stt->capacitorSlow); |
- } else |
- { |
- // decrease capacitorSlow |
- stt->capacitorSlow |
- = AGC_SCALEDIFF32(decay, stt->capacitorSlow, stt->capacitorSlow); |
- } |
- |
- // use maximum of both capacitors as current level |
- if (stt->capacitorFast > stt->capacitorSlow) |
- { |
- cur_level = stt->capacitorFast; |
- } else |
- { |
- cur_level = stt->capacitorSlow; |
- } |
- // Translate signal level into gain, using a piecewise linear approximation |
- // find number of leading zeros |
- zeros = WebRtcSpl_NormU32((uint32_t)cur_level); |
- if (cur_level == 0) |
- { |
- zeros = 31; |
- } |
- tmp32 = (cur_level << zeros) & 0x7FFFFFFF; |
- frac = (int16_t)(tmp32 >> 19); // Q12. |
- tmp32 = (stt->gainTable[zeros-1] - stt->gainTable[zeros]) * frac; |
- gains[k + 1] = stt->gainTable[zeros] + (tmp32 >> 12); |
+ // use maximum of both capacitors as current level |
+ if (stt->capacitorFast > stt->capacitorSlow) { |
+ cur_level = stt->capacitorFast; |
+ } else { |
+ cur_level = stt->capacitorSlow; |
+ } |
+ // Translate signal level into gain, using a piecewise linear approximation |
+ // find number of leading zeros |
+ zeros = WebRtcSpl_NormU32((uint32_t)cur_level); |
+ if (cur_level == 0) { |
+ zeros = 31; |
+ } |
+ tmp32 = (cur_level << zeros) & 0x7FFFFFFF; |
+ frac = (int16_t)(tmp32 >> 19); // Q12. |
+ tmp32 = (stt->gainTable[zeros - 1] - stt->gainTable[zeros]) * frac; |
+ gains[k + 1] = stt->gainTable[zeros] + (tmp32 >> 12); |
#ifdef WEBRTC_AGC_DEBUG_DUMP |
- if (k == 0) { |
- fprintf(stt->logFile, |
- "%d\t%d\t%d\t%d\t%d\n", |
- env[0], |
- cur_level, |
- stt->capacitorFast, |
- stt->capacitorSlow, |
- zeros); |
- } |
+ if (k == 0) { |
+ fprintf(stt->logFile, "%d\t%d\t%d\t%d\t%d\n", env[0], cur_level, |
+ stt->capacitorFast, stt->capacitorSlow, zeros); |
+ } |
#endif |
+ } |
+ |
+ // Gate processing (lower gain during absence of speech) |
+ zeros = (zeros << 9) - (frac >> 3); |
+ // find number of leading zeros |
+ zeros_fast = WebRtcSpl_NormU32((uint32_t)stt->capacitorFast); |
+ if (stt->capacitorFast == 0) { |
+ zeros_fast = 31; |
+ } |
+ tmp32 = (stt->capacitorFast << zeros_fast) & 0x7FFFFFFF; |
+ zeros_fast <<= 9; |
+ zeros_fast -= (int16_t)(tmp32 >> 22); |
+ |
+ gate = 1000 + zeros_fast - zeros - stt->vadNearend.stdShortTerm; |
+ |
+ if (gate < 0) { |
+ stt->gatePrevious = 0; |
+ } else { |
+ tmp32 = stt->gatePrevious * 7; |
+ gate = (int16_t)((gate + tmp32) >> 3); |
+ stt->gatePrevious = gate; |
+ } |
+ // gate < 0 -> no gate |
+ // gate > 2500 -> max gate |
+ if (gate > 0) { |
+ if (gate < 2500) { |
+ gain_adj = (2500 - gate) >> 5; |
+ } else { |
+ gain_adj = 0; |
} |
- |
- // Gate processing (lower gain during absence of speech) |
- zeros = (zeros << 9) - (frac >> 3); |
- // find number of leading zeros |
- zeros_fast = WebRtcSpl_NormU32((uint32_t)stt->capacitorFast); |
- if (stt->capacitorFast == 0) |
- { |
- zeros_fast = 31; |
+ for (k = 0; k < 10; k++) { |
+ if ((gains[k + 1] - stt->gainTable[0]) > 8388608) { |
+ // To prevent wraparound |
+ tmp32 = (gains[k + 1] - stt->gainTable[0]) >> 8; |
+ tmp32 *= 178 + gain_adj; |
+ } else { |
+ tmp32 = (gains[k + 1] - stt->gainTable[0]) * (178 + gain_adj); |
+ tmp32 >>= 8; |
+ } |
+ gains[k + 1] = stt->gainTable[0] + tmp32; |
} |
- tmp32 = (stt->capacitorFast << zeros_fast) & 0x7FFFFFFF; |
- zeros_fast <<= 9; |
- zeros_fast -= (int16_t)(tmp32 >> 22); |
- |
- gate = 1000 + zeros_fast - zeros - stt->vadNearend.stdShortTerm; |
- |
- if (gate < 0) |
- { |
- stt->gatePrevious = 0; |
- } else |
- { |
- tmp32 = stt->gatePrevious * 7; |
- gate = (int16_t)((gate + tmp32) >> 3); |
- stt->gatePrevious = gate; |
+ } |
+ |
+ // Limit gain to avoid overload distortion |
+ for (k = 0; k < 10; k++) { |
+ // To prevent wrap around |
+ zeros = 10; |
+ if (gains[k + 1] > 47453132) { |
+ zeros = 16 - WebRtcSpl_NormW32(gains[k + 1]); |
} |
- // gate < 0 -> no gate |
- // gate > 2500 -> max gate |
- if (gate > 0) |
- { |
- if (gate < 2500) |
- { |
- gain_adj = (2500 - gate) >> 5; |
- } else |
- { |
- gain_adj = 0; |
- } |
- for (k = 0; k < 10; k++) |
- { |
- if ((gains[k + 1] - stt->gainTable[0]) > 8388608) |
- { |
- // To prevent wraparound |
- tmp32 = (gains[k + 1] - stt->gainTable[0]) >> 8; |
- tmp32 *= 178 + gain_adj; |
- } else |
- { |
- tmp32 = (gains[k+1] - stt->gainTable[0]) * (178 + gain_adj); |
- tmp32 >>= 8; |
- } |
- gains[k + 1] = stt->gainTable[0] + tmp32; |
- } |
+ gain32 = (gains[k + 1] >> zeros) + 1; |
+ gain32 *= gain32; |
+ // check for overflow |
+ while (AGC_MUL32((env[k] >> 12) + 1, gain32) > |
+ WEBRTC_SPL_SHIFT_W32((int32_t)32767, 2 * (1 - zeros + 10))) { |
+ // multiply by 253/256 ==> -0.1 dB |
+ if (gains[k + 1] > 8388607) { |
+ // Prevent wrap around |
+ gains[k + 1] = (gains[k + 1] / 256) * 253; |
+ } else { |
+ gains[k + 1] = (gains[k + 1] * 253) / 256; |
+ } |
+ gain32 = (gains[k + 1] >> zeros) + 1; |
+ gain32 *= gain32; |
} |
- |
- // Limit gain to avoid overload distortion |
- for (k = 0; k < 10; k++) |
- { |
- // To prevent wrap around |
- zeros = 10; |
- if (gains[k + 1] > 47453132) |
- { |
- zeros = 16 - WebRtcSpl_NormW32(gains[k + 1]); |
- } |
- gain32 = (gains[k + 1] >> zeros) + 1; |
- gain32 *= gain32; |
- // check for overflow |
- while (AGC_MUL32((env[k] >> 12) + 1, gain32) |
- > WEBRTC_SPL_SHIFT_W32((int32_t)32767, 2 * (1 - zeros + 10))) |
- { |
- // multiply by 253/256 ==> -0.1 dB |
- if (gains[k + 1] > 8388607) |
- { |
- // Prevent wrap around |
- gains[k + 1] = (gains[k+1] / 256) * 253; |
- } else |
- { |
- gains[k + 1] = (gains[k+1] * 253) / 256; |
- } |
- gain32 = (gains[k + 1] >> zeros) + 1; |
- gain32 *= gain32; |
- } |
+ } |
+ // gain reductions should be done 1 ms earlier than gain increases |
+ for (k = 1; k < 10; k++) { |
+ if (gains[k] > gains[k + 1]) { |
+ gains[k] = gains[k + 1]; |
} |
- // gain reductions should be done 1 ms earlier than gain increases |
- for (k = 1; k < 10; k++) |
- { |
- if (gains[k] > gains[k + 1]) |
- { |
- gains[k] = gains[k + 1]; |
- } |
+ } |
+ // save start gain for next frame |
+ stt->gain = gains[10]; |
+ |
+ // Apply gain |
+ // handle first sub frame separately |
+ delta = (gains[1] - gains[0]) * (1 << (4 - L2)); |
+ gain32 = gains[0] * (1 << 4); |
+ // iterate over samples |
+ for (n = 0; n < L; n++) { |
+ for (i = 0; i < num_bands; ++i) { |
+ tmp32 = out[i][n] * ((gain32 + 127) >> 7); |
+ out_tmp = tmp32 >> 16; |
+ if (out_tmp > 4095) { |
+ out[i][n] = (int16_t)32767; |
+ } else if (out_tmp < -4096) { |
+ out[i][n] = (int16_t)-32768; |
+ } else { |
+ tmp32 = out[i][n] * (gain32 >> 4); |
+ out[i][n] = (int16_t)(tmp32 >> 16); |
+ } |
} |
- // save start gain for next frame |
- stt->gain = gains[10]; |
- |
- // Apply gain |
- // handle first sub frame separately |
- delta = (gains[1] - gains[0]) * (1 << (4 - L2)); |
- gain32 = gains[0] * (1 << 4); |
+ // |
+ |
+ gain32 += delta; |
+ } |
+ // iterate over subframes |
+ for (k = 1; k < 10; k++) { |
+ delta = (gains[k + 1] - gains[k]) * (1 << (4 - L2)); |
+ gain32 = gains[k] * (1 << 4); |
// iterate over samples |
- for (n = 0; n < L; n++) |
- { |
- for (i = 0; i < num_bands; ++i) |
- { |
- tmp32 = out[i][n] * ((gain32 + 127) >> 7); |
- out_tmp = tmp32 >> 16; |
- if (out_tmp > 4095) |
- { |
- out[i][n] = (int16_t)32767; |
- } else if (out_tmp < -4096) |
- { |
- out[i][n] = (int16_t)-32768; |
- } else |
- { |
- tmp32 = out[i][n] * (gain32 >> 4); |
- out[i][n] = (int16_t)(tmp32 >> 16); |
- } |
- } |
- // |
- |
- gain32 += delta; |
- } |
- // iterate over subframes |
- for (k = 1; k < 10; k++) |
- { |
- delta = (gains[k+1] - gains[k]) * (1 << (4 - L2)); |
- gain32 = gains[k] * (1 << 4); |
- // iterate over samples |
- for (n = 0; n < L; n++) |
- { |
- for (i = 0; i < num_bands; ++i) |
- { |
- tmp32 = out[i][k * L + n] * (gain32 >> 4); |
- out[i][k * L + n] = (int16_t)(tmp32 >> 16); |
- } |
- gain32 += delta; |
- } |
+ for (n = 0; n < L; n++) { |
+ for (i = 0; i < num_bands; ++i) { |
+ tmp32 = out[i][k * L + n] * (gain32 >> 4); |
+ out[i][k * L + n] = (int16_t)(tmp32 >> 16); |
+ } |
+ gain32 += delta; |
} |
+ } |
- return 0; |
+ return 0; |
} |
void WebRtcAgc_InitVad(AgcVad* state) { |
- int16_t k; |
- |
- state->HPstate = 0; // state of high pass filter |
- state->logRatio = 0; // log( P(active) / P(inactive) ) |
- // average input level (Q10) |
- state->meanLongTerm = 15 << 10; |
- |
- // variance of input level (Q8) |
- state->varianceLongTerm = 500 << 8; |
- |
- state->stdLongTerm = 0; // standard deviation of input level in dB |
- // short-term average input level (Q10) |
- state->meanShortTerm = 15 << 10; |
- |
- // short-term variance of input level (Q8) |
- state->varianceShortTerm = 500 << 8; |
- |
- state->stdShortTerm = 0; // short-term standard deviation of input level in dB |
- state->counter = 3; // counts updates |
- for (k = 0; k < 8; k++) |
- { |
- // downsampling filter |
- state->downState[k] = 0; |
- } |
+ int16_t k; |
+ |
+ state->HPstate = 0; // state of high pass filter |
+ state->logRatio = 0; // log( P(active) / P(inactive) ) |
+ // average input level (Q10) |
+ state->meanLongTerm = 15 << 10; |
+ |
+ // variance of input level (Q8) |
+ state->varianceLongTerm = 500 << 8; |
+ |
+ state->stdLongTerm = 0; // standard deviation of input level in dB |
+ // short-term average input level (Q10) |
+ state->meanShortTerm = 15 << 10; |
+ |
+ // short-term variance of input level (Q8) |
+ state->varianceShortTerm = 500 << 8; |
+ |
+ state->stdShortTerm = |
+ 0; // short-term standard deviation of input level in dB |
+ state->counter = 3; // counts updates |
+ for (k = 0; k < 8; k++) { |
+ // downsampling filter |
+ state->downState[k] = 0; |
+ } |
} |
int16_t WebRtcAgc_ProcessVad(AgcVad* state, // (i) VAD state |
const int16_t* in, // (i) Speech signal |
- size_t nrSamples) // (i) number of samples |
+ size_t nrSamples) // (i) number of samples |
{ |
- int32_t out, nrg, tmp32, tmp32b; |
- uint16_t tmpU16; |
- int16_t k, subfr, tmp16; |
- int16_t buf1[8]; |
- int16_t buf2[4]; |
- int16_t HPstate; |
- int16_t zeros, dB; |
- |
- // process in 10 sub frames of 1 ms (to save on memory) |
- nrg = 0; |
- HPstate = state->HPstate; |
- for (subfr = 0; subfr < 10; subfr++) |
- { |
- // downsample to 4 kHz |
- if (nrSamples == 160) |
- { |
- for (k = 0; k < 8; k++) |
- { |
- tmp32 = (int32_t)in[2 * k] + (int32_t)in[2 * k + 1]; |
- tmp32 >>= 1; |
- buf1[k] = (int16_t)tmp32; |
- } |
- in += 16; |
- |
- WebRtcSpl_DownsampleBy2(buf1, 8, buf2, state->downState); |
- } else |
- { |
- WebRtcSpl_DownsampleBy2(in, 8, buf2, state->downState); |
- in += 8; |
- } |
- |
- // high pass filter and compute energy |
- for (k = 0; k < 4; k++) |
- { |
- out = buf2[k] + HPstate; |
- tmp32 = 600 * out; |
- HPstate = (int16_t)((tmp32 >> 10) - buf2[k]); |
- nrg += (out * out) >> 6; |
- } |
+ int32_t out, nrg, tmp32, tmp32b; |
+ uint16_t tmpU16; |
+ int16_t k, subfr, tmp16; |
+ int16_t buf1[8]; |
+ int16_t buf2[4]; |
+ int16_t HPstate; |
+ int16_t zeros, dB; |
+ |
+ // process in 10 sub frames of 1 ms (to save on memory) |
+ nrg = 0; |
+ HPstate = state->HPstate; |
+ for (subfr = 0; subfr < 10; subfr++) { |
+ // downsample to 4 kHz |
+ if (nrSamples == 160) { |
+ for (k = 0; k < 8; k++) { |
+ tmp32 = (int32_t)in[2 * k] + (int32_t)in[2 * k + 1]; |
+ tmp32 >>= 1; |
+ buf1[k] = (int16_t)tmp32; |
+ } |
+ in += 16; |
+ |
+ WebRtcSpl_DownsampleBy2(buf1, 8, buf2, state->downState); |
+ } else { |
+ WebRtcSpl_DownsampleBy2(in, 8, buf2, state->downState); |
+ in += 8; |
} |
- state->HPstate = HPstate; |
- // find number of leading zeros |
- if (!(0xFFFF0000 & nrg)) |
- { |
- zeros = 16; |
- } else |
- { |
- zeros = 0; |
- } |
- if (!(0xFF000000 & (nrg << zeros))) |
- { |
- zeros += 8; |
- } |
- if (!(0xF0000000 & (nrg << zeros))) |
- { |
- zeros += 4; |
- } |
- if (!(0xC0000000 & (nrg << zeros))) |
- { |
- zeros += 2; |
+ // high pass filter and compute energy |
+ for (k = 0; k < 4; k++) { |
+ out = buf2[k] + HPstate; |
+ tmp32 = 600 * out; |
+ HPstate = (int16_t)((tmp32 >> 10) - buf2[k]); |
+ nrg += (out * out) >> 6; |
} |
- if (!(0x80000000 & (nrg << zeros))) |
- { |
- zeros += 1; |
- } |
- |
- // energy level (range {-32..30}) (Q10) |
- dB = (15 - zeros) << 11; |
- |
- // Update statistics |
- |
- if (state->counter < kAvgDecayTime) |
- { |
- // decay time = AvgDecTime * 10 ms |
- state->counter++; |
- } |
- |
- // update short-term estimate of mean energy level (Q10) |
- tmp32 = state->meanShortTerm * 15 + dB; |
- state->meanShortTerm = (int16_t)(tmp32 >> 4); |
- |
- // update short-term estimate of variance in energy level (Q8) |
- tmp32 = (dB * dB) >> 12; |
- tmp32 += state->varianceShortTerm * 15; |
- state->varianceShortTerm = tmp32 / 16; |
- |
- // update short-term estimate of standard deviation in energy level (Q10) |
- tmp32 = state->meanShortTerm * state->meanShortTerm; |
- tmp32 = (state->varianceShortTerm << 12) - tmp32; |
- state->stdShortTerm = (int16_t)WebRtcSpl_Sqrt(tmp32); |
- |
- // update long-term estimate of mean energy level (Q10) |
- tmp32 = state->meanLongTerm * state->counter + dB; |
- state->meanLongTerm = WebRtcSpl_DivW32W16ResW16( |
- tmp32, WebRtcSpl_AddSatW16(state->counter, 1)); |
- |
- // update long-term estimate of variance in energy level (Q8) |
- tmp32 = (dB * dB) >> 12; |
- tmp32 += state->varianceLongTerm * state->counter; |
- state->varianceLongTerm = WebRtcSpl_DivW32W16( |
- tmp32, WebRtcSpl_AddSatW16(state->counter, 1)); |
- |
- // update long-term estimate of standard deviation in energy level (Q10) |
- tmp32 = state->meanLongTerm * state->meanLongTerm; |
- tmp32 = (state->varianceLongTerm << 12) - tmp32; |
- state->stdLongTerm = (int16_t)WebRtcSpl_Sqrt(tmp32); |
- |
- // update voice activity measure (Q10) |
- tmp16 = 3 << 12; |
- // TODO(bjornv): (dB - state->meanLongTerm) can overflow, e.g., in |
- // ApmTest.Process unit test. Previously the macro WEBRTC_SPL_MUL_16_16() |
- // was used, which did an intermediate cast to (int16_t), hence losing |
- // significant bits. This cause logRatio to max out positive, rather than |
- // negative. This is a bug, but has very little significance. |
- tmp32 = tmp16 * (int16_t)(dB - state->meanLongTerm); |
- tmp32 = WebRtcSpl_DivW32W16(tmp32, state->stdLongTerm); |
- tmpU16 = (13 << 12); |
- tmp32b = WEBRTC_SPL_MUL_16_U16(state->logRatio, tmpU16); |
- tmp32 += tmp32b >> 10; |
- |
- state->logRatio = (int16_t)(tmp32 >> 6); |
- |
- // limit |
- if (state->logRatio > 2048) |
- { |
- state->logRatio = 2048; |
- } |
- if (state->logRatio < -2048) |
- { |
- state->logRatio = -2048; |
- } |
- |
- return state->logRatio; // Q10 |
+ } |
+ state->HPstate = HPstate; |
+ |
+ // find number of leading zeros |
+ if (!(0xFFFF0000 & nrg)) { |
+ zeros = 16; |
+ } else { |
+ zeros = 0; |
+ } |
+ if (!(0xFF000000 & (nrg << zeros))) { |
+ zeros += 8; |
+ } |
+ if (!(0xF0000000 & (nrg << zeros))) { |
+ zeros += 4; |
+ } |
+ if (!(0xC0000000 & (nrg << zeros))) { |
+ zeros += 2; |
+ } |
+ if (!(0x80000000 & (nrg << zeros))) { |
+ zeros += 1; |
+ } |
+ |
+ // energy level (range {-32..30}) (Q10) |
+ dB = (15 - zeros) << 11; |
+ |
+ // Update statistics |
+ |
+ if (state->counter < kAvgDecayTime) { |
+ // decay time = AvgDecTime * 10 ms |
+ state->counter++; |
+ } |
+ |
+ // update short-term estimate of mean energy level (Q10) |
+ tmp32 = state->meanShortTerm * 15 + dB; |
+ state->meanShortTerm = (int16_t)(tmp32 >> 4); |
+ |
+ // update short-term estimate of variance in energy level (Q8) |
+ tmp32 = (dB * dB) >> 12; |
+ tmp32 += state->varianceShortTerm * 15; |
+ state->varianceShortTerm = tmp32 / 16; |
+ |
+ // update short-term estimate of standard deviation in energy level (Q10) |
+ tmp32 = state->meanShortTerm * state->meanShortTerm; |
+ tmp32 = (state->varianceShortTerm << 12) - tmp32; |
+ state->stdShortTerm = (int16_t)WebRtcSpl_Sqrt(tmp32); |
+ |
+ // update long-term estimate of mean energy level (Q10) |
+ tmp32 = state->meanLongTerm * state->counter + dB; |
+ state->meanLongTerm = |
+ WebRtcSpl_DivW32W16ResW16(tmp32, WebRtcSpl_AddSatW16(state->counter, 1)); |
+ |
+ // update long-term estimate of variance in energy level (Q8) |
+ tmp32 = (dB * dB) >> 12; |
+ tmp32 += state->varianceLongTerm * state->counter; |
+ state->varianceLongTerm = |
+ WebRtcSpl_DivW32W16(tmp32, WebRtcSpl_AddSatW16(state->counter, 1)); |
+ |
+ // update long-term estimate of standard deviation in energy level (Q10) |
+ tmp32 = state->meanLongTerm * state->meanLongTerm; |
+ tmp32 = (state->varianceLongTerm << 12) - tmp32; |
+ state->stdLongTerm = (int16_t)WebRtcSpl_Sqrt(tmp32); |
+ |
+ // update voice activity measure (Q10) |
+ tmp16 = 3 << 12; |
+ // TODO(bjornv): (dB - state->meanLongTerm) can overflow, e.g., in |
+ // ApmTest.Process unit test. Previously the macro WEBRTC_SPL_MUL_16_16() |
+ // was used, which did an intermediate cast to (int16_t), hence losing |
+ // significant bits. This cause logRatio to max out positive, rather than |
+ // negative. This is a bug, but has very little significance. |
+ tmp32 = tmp16 * (int16_t)(dB - state->meanLongTerm); |
+ tmp32 = WebRtcSpl_DivW32W16(tmp32, state->stdLongTerm); |
+ tmpU16 = (13 << 12); |
+ tmp32b = WEBRTC_SPL_MUL_16_U16(state->logRatio, tmpU16); |
+ tmp32 += tmp32b >> 10; |
+ |
+ state->logRatio = (int16_t)(tmp32 >> 6); |
+ |
+ // limit |
+ if (state->logRatio > 2048) { |
+ state->logRatio = 2048; |
+ } |
+ if (state->logRatio < -2048) { |
+ state->logRatio = -2048; |
+ } |
+ |
+ return state->logRatio; // Q10 |
} |