fft: remove inline wrappers for function pointers

This removes the rather pointless wrappers (one not even inline)
for calling the fft_calc and related function pointers.

Signed-off-by: Mans Rullgard <mans@mansr.com>

libavcodec/aacdec.c

@@ -1750,7 +1750,7 @@ static void windowing_and_mdct_ltp(AACContext *ac, float *out,
         ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
         memset(in + 1024 + 576, 0, 448 * sizeof(float));
     }
-    ff_mdct_calc(&ac->mdct_ltp, out, in);
+    ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
 }
 
 /**
@@ -1839,9 +1839,9 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
     // imdct
     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
         for (i = 0; i < 1024; i += 128)
-            ff_imdct_half(&ac->mdct_small, buf + i, in + i);
+            ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
     } else
-        ff_imdct_half(&ac->mdct, buf, in);
+        ac->mdct.imdct_half(&ac->mdct, buf, in);
 
     /* window overlapping
      * NOTE: To simplify the overlapping code, all 'meaningless' short to long

libavcodec/aacenc.c

@@ -250,7 +250,7 @@ static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
             for (i = 0; i < 1024; i++)
                 sce->saved[i] = audio[i * chans];
         }
-        ff_mdct_calc(&s->mdct1024, sce->coeffs, output);
+        s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
     } else {
         for (k = 0; k < 1024; k += 128) {
             for (i = 448 + k; i < 448 + k + 256; i++)
@@ -259,7 +259,7 @@ static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
                                          : audio[(i-1024)*chans];
             s->dsp.vector_fmul        (output,     output, k ?  swindow : pwindow, 128);
             s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);
-            ff_mdct_calc(&s->mdct128, sce->coeffs + k, output);
+            s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + k, output);
         }
         for (i = 0; i < 1024; i++)
             sce->saved[i] = audio[i * chans];

libavcodec/aacsbr.c

@@ -1155,7 +1155,7 @@ static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct, const float *in,
         }
         z[64+63] = z[32];
 
-        ff_imdct_half(mdct, z, z+64);
+        mdct->imdct_half(mdct, z, z+64);
         for (k = 0; k < 32; k++) {
             W[1][i][k][0] = -z[63-k];
             W[1][i][k][1] = z[k];
@@ -1190,7 +1190,7 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
                 X[0][i][   n] = -X[0][i][n];
                 X[0][i][32+n] =  X[1][i][31-n];
             }
-            ff_imdct_half(mdct, mdct_buf[0], X[0][i]);
+            mdct->imdct_half(mdct, mdct_buf[0], X[0][i]);
             for (n = 0; n < 32; n++) {
                 v[     n] =  mdct_buf[0][63 - 2*n];
                 v[63 - n] = -mdct_buf[0][62 - 2*n];
@@ -1199,8 +1199,8 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
             for (n = 1; n < 64; n+=2) {
                 X[1][i][n] = -X[1][i][n];
             }
-            ff_imdct_half(mdct, mdct_buf[0], X[0][i]);
+            mdct->imdct_half(mdct, mdct_buf[0], X[0][i]);
-            ff_imdct_half(mdct, mdct_buf[1], X[1][i]);
+            mdct->imdct_half(mdct, mdct_buf[1], X[1][i]);
             for (n = 0; n < 64; n++) {
                 v[      n] = -mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];
                 v[127 - n] =  mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];

libavcodec/ac3dec.c

@@ -628,13 +628,13 @@ static inline void do_imdct(AC3DecodeContext *s, int channels)
             float *x = s->tmp_output+128;
             for(i=0; i<128; i++)
                 x[i] = s->transform_coeffs[ch][2*i];
-            ff_imdct_half(&s->imdct_256, s->tmp_output, x);
+            s->imdct_256.imdct_half(&s->imdct_256, s->tmp_output, x);
             s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 128);
             for(i=0; i<128; i++)
                 x[i] = s->transform_coeffs[ch][2*i+1];
-            ff_imdct_half(&s->imdct_256, s->delay[ch-1], x);
+            s->imdct_256.imdct_half(&s->imdct_256, s->delay[ch-1], x);
         } else {
-            ff_imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
+            s->imdct_512.imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
             s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 128);
             memcpy(s->delay[ch-1], s->tmp_output+128, 128*sizeof(float));
         }

libavcodec/ac3enc_float.c

@@ -74,7 +74,7 @@ static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
  */
 static void mdct512(AC3MDCTContext *mdct, float *out, float *in)
 {
-    ff_mdct_calc(&mdct->fft, out, in);
+    mdct->fft.mdct_calc(&mdct->fft, out, in);
 }
 
 

libavcodec/atrac1.c

@@ -99,7 +99,7 @@ static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
         for (i = 0; i < transf_size / 2; i++)
             FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
     }
-    ff_imdct_half(mdct_context, out, spec);
+    mdct_context->imdct_half(mdct_context, out, spec);
 }
 
 

libavcodec/atrac3.c

@@ -146,7 +146,7 @@ static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, int odd_band)
         /**
         * Reverse the odd bands before IMDCT, this is an effect of the QMF transform
         * or it gives better compression to do it this way.
-        * FIXME: It should be possible to handle this in ff_imdct_calc
+        * FIXME: It should be possible to handle this in imdct_calc
         * for that to happen a modification of the prerotation step of
         * all SIMD code and C code is needed.
         * Or fix the functions before so they generate a pre reversed spectrum.
@@ -156,7 +156,7 @@ static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, int odd_band)
             FFSWAP(float, pInput[i], pInput[255-i]);
     }
 
-    ff_imdct_calc(&q->mdct_ctx,pOutput,pInput);
+    q->mdct_ctx.imdct_calc(&q->mdct_ctx,pOutput,pInput);
 
     /* Perform windowing on the output. */
     dsp.vector_fmul(pOutput, pOutput, mdct_window, 512);

libavcodec/avfft.c

@@ -101,7 +101,7 @@ RDFTContext *av_rdft_init(int nbits, enum RDFTransformType trans)
 
 void av_rdft_calc(RDFTContext *s, FFTSample *data)
 {
-    ff_rdft_calc(s, data);
+    s->rdft_calc(s, data);
 }
 
 void av_rdft_end(RDFTContext *s)
@@ -128,7 +128,7 @@ DCTContext *av_dct_init(int nbits, enum DCTTransformType inverse)
 
 void av_dct_calc(DCTContext *s, FFTSample *data)
 {
-    ff_dct_calc(s, data);
+    s->dct_calc(s, data);
 }
 
 void av_dct_end(DCTContext *s)

libavcodec/binkaudio.c

@@ -223,11 +223,11 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
 
         if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
             coeffs[0] /= 0.5;
-            ff_dct_calc (&s->trans.dct,  coeffs);
+            s->trans.dct.dct_calc(&s->trans.dct,  coeffs);
             s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len);
         }
         else if (CONFIG_BINKAUDIO_RDFT_DECODER)
-            ff_rdft_calc(&s->trans.rdft, coeffs);
+            s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
     }
 
     s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,

libavcodec/cook.c

@@ -753,7 +753,7 @@ static void imlt_gain(COOKContext *q, float *inbuffer,
     int i;
 
     /* Inverse modified discrete cosine transform */
-    ff_imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
+    q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
 
     q->imlt_window (q, buffer1, gains_ptr, previous_buffer);
 

libavcodec/dct.c

@@ -59,7 +59,7 @@ static void ff_dst_calc_I_c(DCTContext *ctx, FFTSample *data)
     }
 
     data[n/2] *= 2;
-    ff_rdft_calc(&ctx->rdft, data);
+    ctx->rdft.rdft_calc(&ctx->rdft, data);
 
     data[0] *= 0.5f;
 
@@ -93,7 +93,7 @@ static void ff_dct_calc_I_c(DCTContext *ctx, FFTSample *data)
         data[n - i] = tmp1 + s;
     }
 
-    ff_rdft_calc(&ctx->rdft, data);
+    ctx->rdft.rdft_calc(&ctx->rdft, data);
     data[n] = data[1];
     data[1] = next;
 
@@ -121,7 +121,7 @@ static void ff_dct_calc_III_c(DCTContext *ctx, FFTSample *data)
 
     data[1] = 2 * next;
 
-    ff_rdft_calc(&ctx->rdft, data);
+    ctx->rdft.rdft_calc(&ctx->rdft, data);
 
     for (i = 0; i < n / 2; i++) {
         float tmp1 = data[i        ] * inv_n;
@@ -152,7 +152,7 @@ static void ff_dct_calc_II_c(DCTContext *ctx, FFTSample *data)
         data[n-i-1] = tmp1 - s;
     }
 
-    ff_rdft_calc(&ctx->rdft, data);
+    ctx->rdft.rdft_calc(&ctx->rdft, data);
 
     next = data[1] * 0.5;
     data[1] *= -1;
@@ -176,11 +176,6 @@ static void dct32_func(DCTContext *ctx, FFTSample *data)
     ctx->dct32(data, data);
 }
 
-void ff_dct_calc(DCTContext *s, FFTSample *data)
-{
-    s->dct_calc(s, data);
-}
-
 av_cold int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType inverse)
 {
     int n = 1 << nbits;

libavcodec/fft-test.c

@@ -327,20 +327,20 @@ int main(int argc, char **argv)
     case TRANSFORM_MDCT:
         if (do_inverse) {
             imdct_ref((float *)tab_ref, (float *)tab1, fft_nbits);
-            ff_imdct_calc(m, tab2, (float *)tab1);
+            m->imdct_calc(m, tab2, (float *)tab1);
             err = check_diff((float *)tab_ref, tab2, fft_size, scale);
         } else {
             mdct_ref((float *)tab_ref, (float *)tab1, fft_nbits);
 
-            ff_mdct_calc(m, tab2, (float *)tab1);
+            m->mdct_calc(m, tab2, (float *)tab1);
 
             err = check_diff((float *)tab_ref, tab2, fft_size / 2, scale);
         }
         break;
     case TRANSFORM_FFT:
         memcpy(tab, tab1, fft_size * sizeof(FFTComplex));
-        ff_fft_permute(s, tab);
+        s->fft_permute(s, tab);
-        ff_fft_calc(s, tab);
+        s->fft_calc(s, tab);
 
         fft_ref(tab_ref, tab1, fft_nbits);
         err = check_diff((float *)tab_ref, (float *)tab, fft_size * 2, 1.0);
@@ -357,7 +357,7 @@ int main(int argc, char **argv)
             memcpy(tab2, tab1, fft_size * sizeof(FFTSample));
             tab2[1] = tab1[fft_size_2].re;
 
-            ff_rdft_calc(r, tab2);
+            r->rdft_calc(r, tab2);
             fft_ref(tab_ref, tab1, fft_nbits);
             for (i = 0; i < fft_size; i++) {
                 tab[i].re = tab2[i];
@@ -369,7 +369,7 @@ int main(int argc, char **argv)
                 tab2[i]    = tab1[i].re;
                 tab1[i].im = 0;
             }
-            ff_rdft_calc(r, tab2);
+            r->rdft_calc(r, tab2);
             fft_ref(tab_ref, tab1, fft_nbits);
             tab_ref[0].im = tab_ref[fft_size_2].re;
             err = check_diff((float *)tab_ref, (float *)tab2, fft_size, 1.0);
@@ -377,7 +377,7 @@ int main(int argc, char **argv)
         break;
     case TRANSFORM_DCT:
         memcpy(tab, tab1, fft_size * sizeof(FFTComplex));
-        ff_dct_calc(d, tab);
+        d->dct_calc(d, tab);
         if (do_inverse) {
             idct_ref(tab_ref, tab1, fft_nbits);
         } else {
@@ -402,22 +402,22 @@ int main(int argc, char **argv)
                 switch (transform) {
                 case TRANSFORM_MDCT:
                     if (do_inverse) {
-                        ff_imdct_calc(m, (float *)tab, (float *)tab1);
+                        m->imdct_calc(m, (float *)tab, (float *)tab1);
                     } else {
-                        ff_mdct_calc(m, (float *)tab, (float *)tab1);
+                        m->mdct_calc(m, (float *)tab, (float *)tab1);
                     }
                     break;
                 case TRANSFORM_FFT:
                     memcpy(tab, tab1, fft_size * sizeof(FFTComplex));
-                    ff_fft_calc(s, tab);
+                    s->fft_calc(s, tab);
                     break;
                 case TRANSFORM_RDFT:
                     memcpy(tab2, tab1, fft_size * sizeof(FFTSample));
-                    ff_rdft_calc(r, tab2);
+                    r->rdft_calc(r, tab2);
                     break;
                 case TRANSFORM_DCT:
                     memcpy(tab2, tab1, fft_size * sizeof(FFTSample));
-                    ff_dct_calc(d, tab2);
+                    d->dct_calc(d, tab2);
                     break;
                 }
             }

libavcodec/fft.h

@@ -39,7 +39,14 @@ struct FFTContext {
     /* pre/post rotation tables */
     FFTSample *tcos;
     FFTSample *tsin;
+    /**
+     * Do the permutation needed BEFORE calling fft_calc().
+     */
     void (*fft_permute)(struct FFTContext *s, FFTComplex *z);
+    /**
+     * Do a complex FFT with the parameters defined in ff_fft_init(). The
+     * input data must be permuted before. No 1.0/sqrt(n) normalization is done.
+     */
     void (*fft_calc)(struct FFTContext *s, FFTComplex *z);
     void (*imdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input);
     void (*imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input);
@@ -115,40 +122,8 @@ void ff_fft_init_mmx(FFTContext *s);
 void ff_fft_init_arm(FFTContext *s);
 void ff_dct_init_mmx(DCTContext *s);
 
-/**
- * Do the permutation needed BEFORE calling ff_fft_calc().
- */
-static inline void ff_fft_permute(FFTContext *s, FFTComplex *z)
-{
-    s->fft_permute(s, z);
-}
-/**
- * Do a complex FFT with the parameters defined in ff_fft_init(). The
- * input data must be permuted before. No 1.0/sqrt(n) normalization is done.
- */
-static inline void ff_fft_calc(FFTContext *s, FFTComplex *z)
-{
-    s->fft_calc(s, z);
-}
 void ff_fft_end(FFTContext *s);
 
-/* MDCT computation */
-
-static inline void ff_imdct_calc(FFTContext *s, FFTSample *output, const FFTSample *input)
-{
-    s->imdct_calc(s, output, input);
-}
-static inline void ff_imdct_half(FFTContext *s, FFTSample *output, const FFTSample *input)
-{
-    s->imdct_half(s, output, input);
-}
-
-static inline void ff_mdct_calc(FFTContext *s, FFTSample *output,
-                                const FFTSample *input)
-{
-    s->mdct_calc(s, output, input);
-}
-
 /**
  * Maximum window size for ff_kbd_window_init.
  */
@@ -213,11 +188,6 @@ void ff_rdft_end(RDFTContext *s);
 
 void ff_rdft_init_arm(RDFTContext *s);
 
-static av_always_inline void ff_rdft_calc(RDFTContext *s, FFTSample *data)
-{
-    s->rdft_calc(s, data);
-}
-
 /* Discrete Cosine Transform */
 
 struct DCTContext {
@@ -239,7 +209,6 @@ struct DCTContext {
  * @note the first element of the input of DST-I is ignored
  */
 int  ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType type);
-void ff_dct_calc(DCTContext *s, FFTSample *data);
 void ff_dct_end (DCTContext *s);
 
 #endif /* AVCODEC_FFT_H */

libavcodec/imc.c

@@ -564,8 +564,8 @@ static void imc_imdct256(IMCContext *q) {
     }
 
     /* FFT */
-    ff_fft_permute(&q->fft, q->samples);
+    q->fft.fft_permute(&q->fft, q->samples);
-    ff_fft_calc (&q->fft, q->samples);
+    q->fft.fft_calc   (&q->fft, q->samples);
 
     /* postrotation, window and reorder */
     for(i = 0; i < COEFFS/2; i++){

libavcodec/mdct.c

@@ -146,7 +146,7 @@ void ff_imdct_half_c(FFTContext *s, FFTSample *output, const FFTSample *input)
         in1 += 2;
         in2 -= 2;
     }
-    ff_fft_calc(s, z);
+    s->fft_calc(s, z);
 
     /* post rotation + reordering */
     for(k = 0; k < n8; k++) {
@@ -213,7 +213,7 @@ void ff_mdct_calc_c(FFTContext *s, FFTSample *out, const FFTSample *input)
         CMUL(x[j].re, x[j].im, re, im, -tcos[n8 + i], tsin[n8 + i]);
     }
 
-    ff_fft_calc(s, x);
+    s->fft_calc(s, x);
 
     /* post rotation */
     for(i=0;i<n8;i++) {

libavcodec/nellymoserdec.c

@@ -121,7 +121,7 @@ static void nelly_decode_block(NellyMoserDecodeContext *s,
         memset(&aptr[NELLY_FILL_LEN], 0,
                (NELLY_BUF_LEN - NELLY_FILL_LEN) * sizeof(float));
 
-        ff_imdct_calc(&s->imdct_ctx, s->imdct_out, aptr);
+        s->imdct_ctx.imdct_calc(&s->imdct_ctx, s->imdct_out, aptr);
         /* XXX: overlapping and windowing should be part of a more
            generic imdct function */
         overlap_and_window(s, s->state, aptr, s->imdct_out);

libavcodec/nellymoserenc.c

@@ -116,13 +116,13 @@ static void apply_mdct(NellyMoserEncodeContext *s)
     s->dsp.vector_fmul(s->in_buff, s->buf[s->bufsel], ff_sine_128, NELLY_BUF_LEN);
     s->dsp.vector_fmul_reverse(s->in_buff + NELLY_BUF_LEN, s->buf[s->bufsel] + NELLY_BUF_LEN, ff_sine_128,
                                NELLY_BUF_LEN);
-    ff_mdct_calc(&s->mdct_ctx, s->mdct_out, s->in_buff);
+    s->mdct_ctx.mdct_calc(&s->mdct_ctx, s->mdct_out, s->in_buff);
 
     s->dsp.vector_fmul(s->buf[s->bufsel] + NELLY_BUF_LEN, s->buf[s->bufsel] + NELLY_BUF_LEN,
                        ff_sine_128, NELLY_BUF_LEN);
     s->dsp.vector_fmul_reverse(s->buf[s->bufsel] + 2 * NELLY_BUF_LEN, s->buf[1 - s->bufsel], ff_sine_128,
                                NELLY_BUF_LEN);
-    ff_mdct_calc(&s->mdct_ctx, s->mdct_out + NELLY_BUF_LEN, s->buf[s->bufsel] + NELLY_BUF_LEN);
+    s->mdct_ctx.mdct_calc(&s->mdct_ctx, s->mdct_out + NELLY_BUF_LEN, s->buf[s->bufsel] + NELLY_BUF_LEN);
 }
 
 static av_cold int encode_init(AVCodecContext *avctx)

libavcodec/qdm2.c

@@ -1588,7 +1588,7 @@ static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
     int i;
     q->fft.complex[channel][0].re *= 2.0f;
     q->fft.complex[channel][0].im = 0.0f;
-    ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
+    q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
     /* add samples to output buffer */
     for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
         q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;

libavcodec/rdft.c

@@ -65,8 +65,8 @@ static void ff_rdft_calc_c(RDFTContext* s, FFTSample* data)
     const FFTSample *tsin = s->tsin;
 
     if (!s->inverse) {
-        ff_fft_permute(&s->fft, (FFTComplex*)data);
+        s->fft.fft_permute(&s->fft, (FFTComplex*)data);
-        ff_fft_calc(&s->fft, (FFTComplex*)data);
+        s->fft.fft_calc(&s->fft, (FFTComplex*)data);
     }
     /* i=0 is a special case because of packing, the DC term is real, so we
        are going to throw the N/2 term (also real) in with it. */
@@ -91,8 +91,8 @@ static void ff_rdft_calc_c(RDFTContext* s, FFTSample* data)
     if (s->inverse) {
         data[0] *= k1;
         data[1] *= k1;
-        ff_fft_permute(&s->fft, (FFTComplex*)data);
+        s->fft.fft_permute(&s->fft, (FFTComplex*)data);
-        ff_fft_calc(&s->fft, (FFTComplex*)data);
+        s->fft.fft_calc(&s->fft, (FFTComplex*)data);
     }
 }
 

libavcodec/synth_filter.c

@@ -29,7 +29,7 @@ static void synth_filter_float(FFTContext *imdct,
     float *synth_buf= synth_buf_ptr + *synth_buf_offset;
     int i, j;
 
-    ff_imdct_half(imdct, synth_buf, in);
+    imdct->imdct_half(imdct, synth_buf, in);
 
     for (i = 0; i < 16; i++){
         float a= synth_buf2[i     ];

libavcodec/twinvq.c

@@ -608,6 +608,7 @@ static void dec_lpc_spectrum_inv(TwinContext *tctx, float *lsp,
 static void imdct_and_window(TwinContext *tctx, enum FrameType ftype, int wtype,
                             float *in, float *prev, int ch)
 {
+    FFTContext *mdct = &tctx->mdct_ctx[ftype];
     const ModeTab *mtab = tctx->mtab;
     int bsize = mtab->size / mtab->fmode[ftype].sub;
     int size  = mtab->size;
@@ -640,7 +641,7 @@ static void imdct_and_window(TwinContext *tctx, enum FrameType ftype, int wtype,
 
         wsize = types_sizes[wtype_to_wsize[sub_wtype]];
 
-        ff_imdct_half(&tctx->mdct_ctx[ftype], buf1 + bsize*j, in + bsize*j);
+        mdct->imdct_half(mdct, buf1 + bsize*j, in + bsize*j);
 
         tctx->dsp.vector_fmul_window(out2,
                                      prev_buf + (bsize-wsize)/2,

libavcodec/vorbis_dec.c

@@ -1448,7 +1448,7 @@ void vorbis_inverse_coupling(float *mag, float *ang, int blocksize)
 static int vorbis_parse_audio_packet(vorbis_context *vc)
 {
     GetBitContext *gb = &vc->gb;
-
+    FFTContext *mdct;
     uint_fast8_t previous_window = vc->previous_window;
     uint_fast8_t mode_number;
     uint_fast8_t blockflag;
@@ -1552,11 +1552,13 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
 
 // Dotproduct, MDCT
 
+    mdct = &vc->mdct[blockflag];
+
     for (j = vc->audio_channels-1;j >= 0; j--) {
         ch_floor_ptr = vc->channel_floors   + j           * blocksize / 2;
         ch_res_ptr   = vc->channel_residues + res_chan[j] * blocksize / 2;
         vc->dsp.vector_fmul(ch_floor_ptr, ch_floor_ptr, ch_res_ptr, blocksize / 2);
-        ff_imdct_half(&vc->mdct[blockflag], ch_res_ptr, ch_floor_ptr);
+        mdct->imdct_half(mdct, ch_res_ptr, ch_floor_ptr);
     }
 
 // Overlap/add, save data for next overlapping  FPMATH

libavcodec/vorbis_enc.c

@@ -935,7 +935,7 @@ static int apply_window_and_mdct(vorbis_enc_context *venc, const signed short *a
     }
 
     for (channel = 0; channel < venc->channels; channel++)
-        ff_mdct_calc(&venc->mdct[0], venc->coeffs + channel * window_len,
+        venc->mdct[0].mdct_calc(&venc->mdct[0], venc->coeffs + channel * window_len,
                      venc->samples + channel * window_len * 2);
 
     if (samples) {

libavcodec/wmadec.c

@@ -447,6 +447,7 @@ static int wma_decode_block(WMACodecContext *s)
     int coef_nb_bits, total_gain;
     int nb_coefs[MAX_CHANNELS];
     float mdct_norm;
+    FFTContext *mdct;
 
 #ifdef TRACE
     tprintf(s->avctx, "***decode_block: %d:%d\n", s->frame_count - 1, s->block_num);
@@ -742,12 +743,14 @@ static int wma_decode_block(WMACodecContext *s)
     }
 
 next:
+    mdct = &s->mdct_ctx[bsize];
+
     for(ch = 0; ch < s->nb_channels; ch++) {
         int n4, index;
 
         n4 = s->block_len / 2;
         if(s->channel_coded[ch]){
-            ff_imdct_calc(&s->mdct_ctx[bsize], s->output, s->coefs[ch]);
+            mdct->imdct_calc(mdct, s->output, s->coefs[ch]);
         }else if(!(s->ms_stereo && ch==1))
             memset(s->output, 0, sizeof(s->output));
 

libavcodec/wmaenc.c

@@ -77,6 +77,7 @@ static int encode_init(AVCodecContext * avctx){
 static void apply_window_and_mdct(AVCodecContext * avctx, const signed short * audio, int len) {
     WMACodecContext *s = avctx->priv_data;
     int window_index= s->frame_len_bits - s->block_len_bits;
+    FFTContext *mdct = &s->mdct_ctx[window_index];
     int i, j, channel;
     const float * win = s->windows[window_index];
     int window_len = 1 << s->block_len_bits;
@@ -89,7 +90,7 @@ static void apply_window_and_mdct(AVCodecContext * avctx, const signed short * a
             s->output[i+window_len]  = audio[j] / n * win[window_len - i - 1];
             s->frame_out[channel][i] = audio[j] / n * win[i];
         }
-        ff_mdct_calc(&s->mdct_ctx[window_index], s->coefs[channel], s->output);
+        mdct->mdct_calc(mdct, s->coefs[channel], s->output);
     }
 }
 

libavcodec/wmaprodec.c

@@ -1222,6 +1222,7 @@ static int decode_subframe(WMAProDecodeCtx *s)
             get_bits_count(&s->gb) - s->subframe_offset);
 
     if (transmit_coeffs) {
+        FFTContext *mdct = &s->mdct_ctx[av_log2(subframe_len) - WMAPRO_BLOCK_MIN_BITS];
         /** reconstruct the per channel data */
         inverse_channel_transform(s);
         for (i = 0; i < s->channels_for_cur_subframe; i++) {
@@ -1246,9 +1247,8 @@ static int decode_subframe(WMAProDecodeCtx *s)
                                           quant, end - start);
             }
 
-            /** apply imdct (ff_imdct_half == DCTIV with reverse) */
+            /** apply imdct (imdct_half == DCTIV with reverse) */
-            ff_imdct_half(&s->mdct_ctx[av_log2(subframe_len) - WMAPRO_BLOCK_MIN_BITS],
+            mdct->imdct_half(mdct, s->channel[c].coeffs, s->tmp);
-                          s->channel[c].coeffs, s->tmp);
         }
     }
 

libavcodec/wmavoice.c

@@ -558,7 +558,7 @@ 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 +601,8 @@ 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);
+    s->dct.dct_calc(&s->dct, lpcs);
-    ff_dct_calc(&s->dst, 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 +623,7 @@ 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 +693,8 @@ 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);
+        s->rdft.rdft_calc(&s->rdft, synth_pf);
-        ff_rdft_calc(&s->rdft, coeffs);
+        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 +702,7 @@ 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 */