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| author | Michael Niedermayer <michaelni@gmx.at> | 2011-03-21 03:09:46 +0100 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2011-03-21 03:33:28 +0100 |
| commit | d4a50a2100a2b3539d238e64af98e16f47b47a89 (patch) | |
| tree | 95bc08782f34783dfac34a73142a77adbce4db32 /libavcodec/wmavoice.c | |
| parent | 849f80fecbada7c512737e13edec9516541996e3 (diff) | |
| parent | cf752d027fafe691951fa00d9c25e2fda032b536 (diff) | |
| download | ffmpeg-d4a50a2100a2b3539d238e64af98e16f47b47a89.tar.gz | |
Merge remote-tracking branch 'newdev/master'
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/wmavoice.c')
| -rw-r--r-- | libavcodec/wmavoice.c | 19 |
1 files changed, 10 insertions, 9 deletions
diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c index 13011ec234..dfdfabbbb3 100644 --- a/libavcodec/wmavoice.c +++ b/libavcodec/wmavoice.c @@ -36,8 +36,9 @@ #include "acelp_filters.h" #include "lsp.h" #include "libavutil/lzo.h" -#include "avfft.h" -#include "fft.h" +#include "dct.h" +#include "rdft.h" +#include "sinewin.h" #define MAX_BLOCKS 8 ///< maximum number of blocks per frame #define MAX_LSPS 16 ///< maximum filter order @@ -558,7 +559,7 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs, int n, idx; /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */ - ff_rdft_calc(&s->rdft, lpcs); + s->rdft.rdft_calc(&s->rdft, lpcs); #define log_range(var, assign) do { \ float tmp = log10f(assign); var = tmp; \ max = FFMAX(max, tmp); min = FFMIN(min, tmp); \ @@ -601,8 +602,8 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs, * is a sinus input) by doing a phase shift (in theory, H(sin())=cos()). * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the * "moment" of the LPCs in this filter. */ - ff_dct_calc(&s->dct, lpcs); - ff_dct_calc(&s->dst, lpcs); + s->dct.dct_calc(&s->dct, lpcs); + s->dst.dct_calc(&s->dst, lpcs); /* Split out the coefficient indexes into phase/magnitude pairs */ idx = 255 + av_clip(lpcs[64], -255, 255); @@ -623,7 +624,7 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs, coeffs[1] = last_coeff; /* move into real domain */ - ff_rdft_calc(&s->irdft, coeffs); + s->irdft.rdft_calc(&s->irdft, coeffs); /* tilt correction and normalize scale */ memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder)); @@ -693,8 +694,8 @@ static void wiener_denoise(WMAVoiceContext *s, int fcb_type, /* apply coefficients (in frequency spectrum domain), i.e. complex * number multiplication */ memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size)); - ff_rdft_calc(&s->rdft, synth_pf); - ff_rdft_calc(&s->rdft, coeffs); + s->rdft.rdft_calc(&s->rdft, synth_pf); + s->rdft.rdft_calc(&s->rdft, coeffs); synth_pf[0] *= coeffs[0]; synth_pf[1] *= coeffs[1]; for (n = 1; n < 64; n++) { @@ -702,7 +703,7 @@ static void wiener_denoise(WMAVoiceContext *s, int fcb_type, synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1]; synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1]; } - ff_rdft_calc(&s->irdft, synth_pf); + s->irdft.rdft_calc(&s->irdft, synth_pf); } /* merge filter output with the history of previous runs */ |