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| author | Ronald S. Bultje <rsbultje@gmail.com> | 2013-01-20 15:41:52 -0800 |
|---|---|---|
| committer | Ronald S. Bultje <rsbultje@gmail.com> | 2013-01-22 11:55:42 -0800 |
| commit | d56668bd80075615b89aff652fe8a576bf853ceb (patch) | |
| tree | 9da3ed036b716dbaf33f5c9869578bedb6e393a2 /libavcodec/wmavoice.c | |
| parent | 5959bfaca396ecaf63a8123055f499688b79cae3 (diff) | |
| download | ffmpeg-d56668bd80075615b89aff652fe8a576bf853ceb.tar.gz | |
floatdsp: move scalarproduct_float from dsputil to avfloatdsp.
This makes the aac decoder and all voice codecs independent of dsputil.
Diffstat (limited to 'libavcodec/wmavoice.c')
| -rw-r--r-- | libavcodec/wmavoice.c | 16 |
1 files changed, 9 insertions, 7 deletions
diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c index 08d0600200..ba778cda31 100644 --- a/libavcodec/wmavoice.c +++ b/libavcodec/wmavoice.c @@ -30,8 +30,8 @@ #include <math.h> #include "libavutil/channel_layout.h" +#include "libavutil/float_dsp.h" #include "libavutil/mem.h" -#include "dsputil.h" #include "avcodec.h" #include "internal.h" #include "get_bits.h" @@ -523,7 +523,7 @@ static int kalman_smoothen(WMAVoiceContext *s, int pitch, /* find best fitting point in history */ do { - dot = ff_scalarproduct_float_c(in, ptr, size); + dot = avpriv_scalarproduct_float_c(in, ptr, size); if (dot > optimal_gain) { optimal_gain = dot; best_hist_ptr = ptr; @@ -532,7 +532,7 @@ static int kalman_smoothen(WMAVoiceContext *s, int pitch, if (optimal_gain <= 0) return -1; - dot = ff_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size); + dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size); if (dot <= 0) // would be 1.0 return -1; @@ -562,8 +562,8 @@ static float tilt_factor(const float *lpcs, int n_lpcs) { float rh0, rh1; - rh0 = 1.0 + ff_scalarproduct_float_c(lpcs, lpcs, n_lpcs); - rh1 = lpcs[0] + ff_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1); + rh0 = 1.0 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs); + rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1); return rh1 / rh0; } @@ -656,7 +656,8 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs, -1.8 * tilt_factor(coeffs, remainder - 1), coeffs, remainder); } - sq = (1.0 / 64.0) * sqrtf(1 / ff_scalarproduct_float_c(coeffs, coeffs, remainder)); + sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs, + remainder)); for (n = 0; n < remainder; n++) coeffs[n] *= sq; } @@ -1320,7 +1321,8 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, /* Calculate gain for adaptive & fixed codebook signal. * see ff_amr_set_fixed_gain(). */ idx = get_bits(gb, 7); - fcb_gain = expf(ff_scalarproduct_float_c(s->gain_pred_err, gain_coeff, 6) - + fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err, + gain_coeff, 6) - 5.2409161640 + wmavoice_gain_codebook_fcb[idx]); acb_gain = wmavoice_gain_codebook_acb[idx]; pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx], |