configure: Document --enable-libfontconfig

[ffmpeg] / libavcodec / dcadsp.c

diff --git a/libavcodec/dcadsp.c b/libavcodec/dcadsp.c
index dd4994d2766ebe628e3555f7327b0db3e3089bcd..beec2007f63587b130956911995fafcb6692ff80 100644 (file)
--- a/libavcodec/dcadsp.c
+++ b/libavcodec/dcadsp.c
@@ -2,50 +2,133 @@
  * Copyright (c) 2004 Gildas Bazin
  * Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #include "config.h"
+
+#include "libavutil/attributes.h"
+#include "libavutil/intreadwrite.h"
+
 #include "dcadsp.h"
+#include "dcamath.h"
+
+static void decode_hf_c(int32_t dst[DCA_SUBBANDS][SAMPLES_PER_SUBBAND],
+                        const int32_t vq_num[DCA_SUBBANDS],
+                        const int8_t hf_vq[1024][32], intptr_t vq_offset,
+                        int32_t scale[DCA_SUBBANDS][2],
+                        intptr_t start, intptr_t end)
+{
+    int i, j;
+
+    for (j = start; j < end; j++) {
+        const int8_t *ptr = &hf_vq[vq_num[j]][vq_offset];
+        for (i = 0; i < 8; i++)
+            dst[j][i] = ptr[i] * scale[j][0] + 8 >> 4;
+    }
+}
 
-static void dca_lfe_fir_c(float *out, const float *in, const float *coefs,
-                          int decifactor, float scale)
+static inline void dca_lfe_fir(float *out, const float *in, const float *coefs,
+                               int decifactor)
 {
-    float *out2 = out + decifactor;
-    const float *cf0 = coefs;
-    const float *cf1 = coefs + 256;
+    float *out2    = out + 2 * decifactor - 1;
+    int num_coeffs = 256 / decifactor;
     int j, k;
 
     /* One decimated sample generates 2*decifactor interpolated ones */
     for (k = 0; k < decifactor; k++) {
         float v0 = 0.0;
         float v1 = 0.0;
-        for (j = 0; j < 256 / decifactor; j++) {
-            float s = in[-j];
-            v0 += s * *cf0++;
-            v1 += s * *--cf1;
+        for (j = 0; j < num_coeffs; j++, coefs++) {
+            v0 += in[-j]                 * *coefs;
+            v1 += in[j + 1 - num_coeffs] * *coefs;
         }
-        *out++  = v0 * scale;
-        *out2++ = v1 * scale;
+        *out++  = v0;
+        *out2-- = v1;
     }
 }
 
-void ff_dcadsp_init(DCADSPContext *s)
+static void dca_qmf_32_subbands(float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], int sb_act,
+                                SynthFilterContext *synth, FFTContext *imdct,
+                                float synth_buf_ptr[512],
+                                int *synth_buf_offset, float synth_buf2[32],
+                                const float window[512], float *samples_out,
+                                float raXin[32], float scale)
 {
-    s->lfe_fir = dca_lfe_fir_c;
-    if (ARCH_ARM) ff_dcadsp_init_arm(s);
+    int i;
+    int subindex;
+
+    for (i = sb_act; i < 32; i++)
+        raXin[i] = 0.0;
+
+    /* Reconstructed channel sample index */
+    for (subindex = 0; subindex < 8; subindex++) {
+        /* Load in one sample from each subband and clear inactive subbands */
+        for (i = 0; i < sb_act; i++) {
+            unsigned sign = (i - 1) & 2;
+            uint32_t v    = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
+            AV_WN32A(&raXin[i], v);
+        }
+
+        synth->synth_filter_float(imdct, synth_buf_ptr, synth_buf_offset,
+                                  synth_buf2, window, samples_out, raXin,
+                                  scale);
+        samples_out += 32;
+    }
+}
+
+static void dequantize_c(int32_t *samples, uint32_t step_size, uint32_t scale)
+{
+    int64_t step = (int64_t)step_size * scale;
+    int shift, i;
+    int32_t step_scale;
+
+    if (step > (1 << 23))
+        shift = av_log2(step >> 23) + 1;
+    else
+        shift = 0;
+    step_scale = (int32_t)(step >> shift);
+
+    for (i = 0; i < SAMPLES_PER_SUBBAND; i++)
+        samples[i] = dca_clip23(dca_norm((int64_t)samples[i] * step_scale, 22 - shift));
+}
+
+static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs)
+{
+    dca_lfe_fir(out, in, coefs, 32);
+}
+
+static void dca_lfe_fir1_c(float *out, const float *in, const float *coefs)
+{
+    dca_lfe_fir(out, in, coefs, 64);
+}
+
+av_cold void ff_dcadsp_init(DCADSPContext *s)
+{
+    s->lfe_fir[0]      = dca_lfe_fir0_c;
+    s->lfe_fir[1]      = dca_lfe_fir1_c;
+    s->qmf_32_subbands = dca_qmf_32_subbands;
+    s->decode_hf       = decode_hf_c;
+    s->dequantize      = dequantize_c;
+
+    if (ARCH_AARCH64)
+        ff_dcadsp_init_aarch64(s);
+    if (ARCH_ARM)
+        ff_dcadsp_init_arm(s);
+    if (ARCH_X86)
+        ff_dcadsp_init_x86(s);
 }