On Thu, Jun 11, 2009 at 11:24:46AM -0400, Alex Converse wrote: > Hi all, > > The quantizer code in the aac encoder is kind of sloppy. This cleans it up. > > Regards, > > Alex Converse
> commit 35d52275e36c88a429da06e4f6c6ffdab22e3863 > Author: Alex Converse <[email protected]> > Date: Wed Apr 22 00:24:19 2009 -0400 > > Clean up quant sloppiness. > More cleanup > More quant cleanup > > diff --git a/libavcodec/aaccoder.c b/libavcodec/aaccoder.c > index d3a2755..b1ed259 100644 > --- a/libavcodec/aaccoder.c > +++ b/libavcodec/aaccoder.c > @@ -59,20 +59,26 @@ static const uint8_t* run_value_bits[2] = { > * @return absolute value of the quantized coefficient > * @see 3GPP TS26.403 5.6.2 "Scalefactor determination" > */ > +static av_always_inline int quant_clip(float coef, const float Q) > +{ > + int ret = pow(coef * Q, 0.75) + 0.4054; > + return FFMIN(ret, 8191); > +} > + > static av_always_inline int quant(float coef, const float Q) > { > - return av_clip((int)(pow(fabsf(coef) * Q, 0.75) + 0.4054), 0, 8191); > + return pow(coef * Q, 0.75) + 0.4054; > } > > #if 1 > > static av_always_inline int quant2(float coef, const float Q) > { > - return av_clip((int)(pow(fabsf(coef) * Q, 0.75)), 0, 8191); > + return pow(coef * Q, 0.75); > } { > -static const float aac_cb_range[12] = { 0, 3, 3, 3, 3, 9, 9, 8, 8, 13, 13, > 17}; > -static const float aac_cb_maxval[12] = {0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, > 16}; > +static const uint8_t aac_cb_range [12] = {0, 3, 3, 3, 3, 9, 9, 8, 8, 13, 13, > 17}; > +static const uint8_t aac_cb_maxval[12] = {0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, > 16}; } Commit this change separately > /** > * Calculate rate distortion cost for quantizing with given codebook > @@ -107,8 +113,8 @@ static float quantize_band_cost(const float *in, int > size, int scale_idx, int cb > int quants[4][2]; > mincost = 0.0f; > for(j = 0; j < dim; j++){ > - quants[j][0] = quant2(in[i+j], Q); > - quants[j][1] = quant (in[i+j], Q); > + quants[j][0] = quant2(fabsf(in[i+j]), Q); > + quants[j][1] = quant (fabsf(in[i+j]), Q); > for(k = 0; k < 2; k++){ > quants[j][k] = FFMIN(quants[j][k], maxval); > if(!IS_CODEBOOK_UNSIGNED(cb) && in[i+j] < 0.0f) > @@ -151,7 +157,7 @@ static float quantize_band_cost(const float *in, int > size, int scale_idx, int cb > di = t - 165140.0f; > curbits += 21; > }else{ > - int c = quant(t, Q); > + int c = quant_clip(t, Q); > di = t - c*cbrt(c)*IQ; > curbits += av_log2(c)*2 - 4 + 1; > } > @@ -210,8 +216,8 @@ static void quantize_and_encode_band(PutBitContext *pb, > const float *in, int siz > int quants[4][2]; > mincost = 0.0f; > for(j = 0; j < dim; j++){ > - quants[j][0] = av_clip(quant2(in[i+j], Q), -maxval, maxval); > - quants[j][1] = av_clip(quant (in[i+j], Q), -maxval, maxval); > + quants[j][0] = quant2(fabsf(in[i+j]), Q); > + quants[j][1] = quant (fabsf(in[i+j]), Q); > > for(k = 0; k < 2; k++){ > quants[j][k] = FFMIN(quants[j][k], maxval); > if(!IS_CODEBOOK_UNSIGNED(cb) && in[i+j] < 0.0f) > @@ -254,7 +260,7 @@ static void quantize_and_encode_band(PutBitContext *pb, > const float *in, int siz > di = t - 165140.0f; > curbits += 21; > }else{ > - int c = quant(t, Q); > + int c = quant_clip(t, Q); > di = t - c*cbrt(c)*IQ; > curbits += av_log2(c)*2 - 4 + 1; > } > @@ -286,7 +292,7 @@ static void quantize_and_encode_band(PutBitContext *pb, > const float *in, int siz > if(cb == ESC_BT){ > for(j = 0; j < 2; j++){ > if(ff_aac_codebook_vectors[cb-1][minidx*2+j] == 64.0f){ > - int coef = quant(in[i+j], Q); > + int coef = quant_clip(fabsf(in[i+j]), Q); > int len = av_log2(coef); > > put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2); > @@ -308,6 +314,7 @@ static float quantize_band_cost(const float *in, int > size, int scale_idx, int cb > const float lambda, const float uplim, int > *bits) > { > const float Q = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + > SCALE_DIV_512]; > + const float IQ = 1.0/Q; > int i, j, k; > float cost = 0; > const int dim = cb < FIRST_PAIR_BT ? 4 : 2; > @@ -340,7 +347,7 @@ static float quantize_band_cost(const float *in, int > size, int scale_idx, int cb > di = t - 165140.0f; > curbits += 21; > }else{ > - int c = quant(t, 1.0/Q); > + int c = quant_clip(t, IQ); > di = t - c*cbrt(c)*Q; > curbits += av_log2(c)*2 - 4 + 1; > } > @@ -413,7 +420,7 @@ static void quantize_and_encode_band(PutBitContext *pb, > const float *in, int siz > di = t - 165140.0f; > curbits += 21; > }else{ > - int c = quant(t, IQ); > + int c = quant_clip(t, IQ); > di = t - c*cbrt(c)*Q; > curbits += av_log2(c)*2 - 4 + 1; > } > @@ -445,7 +452,7 @@ static void quantize_and_encode_band(PutBitContext *pb, > const float *in, int siz > if(cb == ESC_BT){ > for(j = 0; j < 2; j++){ > if(ff_aac_codebook_vectors[cb-1][minidx*2+j] == 64.0f){ > - int coef = quant(in[i+j], IQ); > + int coef = quant_clip(fabsf(in[i+j]), IQ); > int len = av_log2(coef); > > put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2); No objections against introducing IQ where appropriate. Also commit separately anytime. As for having unlimited and limited quantizing - that's code duplication. I think it's better to clip it after the call. _______________________________________________ FFmpeg-soc mailing list [email protected] https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-soc
