diff options
| 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/amrnbdec.c | |
| parent | 5959bfaca396ecaf63a8123055f499688b79cae3 (diff) | |
floatdsp: move scalarproduct_float from dsputil to avfloatdsp.
This makes the aac decoder and all voice codecs independent of dsputil.
Diffstat (limited to 'libavcodec/amrnbdec.c')
| -rw-r--r-- | libavcodec/amrnbdec.c | 20 |
1 files changed, 10 insertions, 10 deletions
diff --git a/libavcodec/amrnbdec.c b/libavcodec/amrnbdec.c index 5c359a8f3d..7db12dd001 100644 --- a/libavcodec/amrnbdec.c +++ b/libavcodec/amrnbdec.c @@ -44,8 +44,8 @@ #include <math.h> #include "libavutil/channel_layout.h" +#include "libavutil/float_dsp.h" #include "avcodec.h" -#include "dsputil.h" #include "libavutil/common.h" #include "celp_filters.h" #include "acelp_filters.h" @@ -794,8 +794,8 @@ static int synthesis(AMRContext *p, float *lpc, // emphasize pitch vector contribution if (p->pitch_gain[4] > 0.5 && !overflow) { - float energy = ff_scalarproduct_float_c(excitation, excitation, - AMR_SUBFRAME_SIZE); + float energy = avpriv_scalarproduct_float_c(excitation, excitation, + AMR_SUBFRAME_SIZE); float pitch_factor = p->pitch_gain[4] * (p->cur_frame_mode == MODE_12k2 ? @@ -871,8 +871,8 @@ static float tilt_factor(float *lpc_n, float *lpc_d) ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE, LP_FILTER_ORDER); - rh0 = ff_scalarproduct_float_c(hf, hf, AMR_TILT_RESPONSE); - rh1 = ff_scalarproduct_float_c(hf, hf + 1, AMR_TILT_RESPONSE - 1); + rh0 = avpriv_scalarproduct_float_c(hf, hf, AMR_TILT_RESPONSE); + rh1 = avpriv_scalarproduct_float_c(hf, hf + 1, AMR_TILT_RESPONSE - 1); // The spec only specifies this check for 12.2 and 10.2 kbit/s // modes. But in the ref source the tilt is always non-negative. @@ -892,8 +892,8 @@ static void postfilter(AMRContext *p, float *lpc, float *buf_out) int i; float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input - float speech_gain = ff_scalarproduct_float_c(samples, samples, - AMR_SUBFRAME_SIZE); + float speech_gain = avpriv_scalarproduct_float_c(samples, samples, + AMR_SUBFRAME_SIZE); float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter const float *gamma_n, *gamma_d; // Formant filter factor table @@ -998,9 +998,9 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data, p->fixed_gain[4] = ff_amr_set_fixed_gain(fixed_gain_factor, - ff_scalarproduct_float_c(p->fixed_vector, - p->fixed_vector, - AMR_SUBFRAME_SIZE) / + avpriv_scalarproduct_float_c(p->fixed_vector, + p->fixed_vector, + AMR_SUBFRAME_SIZE) / AMR_SUBFRAME_SIZE, p->prediction_error, energy_mean[p->cur_frame_mode], energy_pred_fac); |