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aacdec: remove sf_scale and sf_offset.

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Instead, scalefactors are adjusted by the offset amount, removing the need
for sf_scale, and the MDCT scales are adjusted to compensate for the higher
scalefactors. Floating-point output will be handled by modifying the MDCT
scales.
pull/2/head
Alex Converse 14 years ago committed by Justin Ruggles
parent 6271794041
commit 767848d761
2 changed files with 8 additions and 20 deletions
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  1. 2
      libavcodec/aac.h
  2. 26
      libavcodec/aacdec.c

2
libavcodec/aac.h
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@ -293,8 +293,6 @@ typedef struct {
* @{
*/
float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
/** @} */
DECLARE_ALIGNED(32, float, temp)[128];

26
libavcodec/aacdec.c
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@ -578,12 +578,6 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ac->random_state = 0x1f2e3d4c;
// -1024 - Compensate wrong IMDCT method.
// 60 - Required to scale values to the correct range [-32768,32767]
// for float to int16 conversion. (1 << (60 / 4)) == 32768
ac->sf_scale = 1. / -1024.;
ac->sf_offset = 60;
ff_aac_tableinit();
INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
@ -591,9 +585,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
352);
ff_mdct_init(&ac->mdct, 11, 1, 1.0);
ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
ff_mdct_init(&ac->mdct_ltp, 11, 0, 2.0);
ff_mdct_init(&ac->mdct, 11, 1, 1.0/1024.0);
ff_mdct_init(&ac->mdct_small, 8, 1, 1.0/128.0);
ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0);
// window initialization
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
@ -651,7 +645,7 @@ static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
int sfb;
ltp->lag = get_bits(gb, 11);
ltp->coef = ltp_coef[get_bits(gb, 3)] * ac->sf_scale;
ltp->coef = ltp_coef[get_bits(gb, 3)];
for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
ltp->used[sfb] = get_bits1(gb);
}
@ -789,7 +783,6 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
enum BandType band_type[120],
int band_type_run_end[120])
{
const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
int g, i, idx = 0;
int offset[3] = { global_gain, global_gain - 90, 0 };
int clipped_offset;
@ -826,7 +819,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
"artifact, there may be a bug in the decoder. ",
offset[1], clipped_offset);
}
sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + sf_offset - 100 + POW_SF2_ZERO];
sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
}
} else {
for (; i < run_end; i++, idx++) {
@ -836,7 +829,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
"%s (%d) out of range.\n", sf_str[0], offset[0]);
return -1;
}
sf[idx] = -ff_aac_pow2sf_tab[offset[0] + sf_offset - 200 + POW_SF2_ZERO];
sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
}
}
}
@ -1247,7 +1240,6 @@ static av_always_inline float flt16_trunc(float pf)
}
static av_always_inline void predict(PredictorState *ps, float *coef,
float sf_scale, float inv_sf_scale,
int output_enable)
{
const float a = 0.953125; // 61.0 / 64
@ -1264,9 +1256,9 @@ static av_always_inline void predict(PredictorState *ps, float *coef,
pv = flt16_round(k1 * r0 + k2 * r1);
if (output_enable)
*coef += pv * sf_scale;
*coef += pv;
e0 = *coef * inv_sf_scale;
e0 = *coef;
e1 = e0 - k1 * r0;
ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
@ -1284,7 +1276,6 @@ static av_always_inline void predict(PredictorState *ps, float *coef,
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
{
int sfb, k;
float sf_scale = ac->sf_scale, inv_sf_scale = 1 / ac->sf_scale;
if (!sce->ics.predictor_initialized) {
reset_all_predictors(sce->predictor_state);
@ -1295,7 +1286,6 @@ static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
predict(&sce->predictor_state[k], &sce->coeffs[k],
sf_scale, inv_sf_scale,
sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
}
}

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