OSDN Git Service

aacenc: remove the data arrays
authorYoung Han Lee <cpumaker@gmail.com>
Fri, 18 Feb 2011 00:33:11 +0000 (09:33 +0900)
committerRonald S. Bultje <rsbultje@gmail.com>
Mon, 7 Mar 2011 17:25:36 +0000 (12:25 -0500)
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
libavcodec/aacenc.c
libavcodec/aacenc.h

index 50dab1c..17ae6f9 100644 (file)
@@ -225,40 +225,41 @@ static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
     const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
     const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
     const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
+    float *output = sce->ret;
 
     if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
-        memcpy(s->output, sce->saved, sizeof(float)*1024);
+        memcpy(output, sce->saved, sizeof(float)*1024);
         if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
-            memset(s->output, 0, sizeof(s->output[0]) * 448);
+            memset(output, 0, sizeof(output[0]) * 448);
             for (i = 448; i < 576; i++)
-                s->output[i] = sce->saved[i] * pwindow[i - 448];
+                output[i] = sce->saved[i] * pwindow[i - 448];
             for (i = 576; i < 704; i++)
-                s->output[i] = sce->saved[i];
+                output[i] = sce->saved[i];
         }
         if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
             for (i = 0; i < 1024; i++) {
-                s->output[i+1024]         = audio[i * chans] * lwindow[1024 - i - 1];
+                output[i+1024]         = audio[i * chans] * lwindow[1024 - i - 1];
                 sce->saved[i] = audio[i * chans] * lwindow[i];
             }
         } else {
             for (i = 0; i < 448; i++)
-                s->output[i+1024]         = audio[i * chans];
+                output[i+1024]         = audio[i * chans];
             for (; i < 576; i++)
-                s->output[i+1024]         = audio[i * chans] * swindow[576 - i - 1];
-            memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
+                output[i+1024]         = audio[i * chans] * swindow[576 - i - 1];
+            memset(output+1024+576, 0, sizeof(output[0]) * 448);
             for (i = 0; i < 1024; i++)
                 sce->saved[i] = audio[i * chans];
         }
-        ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
+        ff_mdct_calc(&s->mdct1024, sce->coeffs, output);
     } else {
         for (k = 0; k < 1024; k += 128) {
             for (i = 448 + k; i < 448 + k + 256; i++)
-                s->output[i - 448 - k] = (i < 1024)
+                output[i - 448 - k] = (i < 1024)
                                          ? sce->saved[i]
                                          : audio[(i-1024)*chans];
-            s->dsp.vector_fmul        (s->output,     s->output, k ?  swindow : pwindow, 128);
-            s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
-            ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
+            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);
         }
         for (i = 0; i < 1024; i++)
             sce->saved[i] = audio[i * chans];
@@ -597,6 +598,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
             }
             for (j = 0; j < chans; j++) {
                 s->cur_channel = start_ch + j;
+                s->scoefs = cpe->ch[j].ret;
                 encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
             }
             start_ch += chans;
index 86c68d3..3559234 100644 (file)
@@ -52,8 +52,7 @@ typedef struct AACEncContext {
     FFTContext mdct1024;                         ///< long (1024 samples) frame transform context
     FFTContext mdct128;                          ///< short (128 samples) frame transform context
     DSPContext  dsp;
-    DECLARE_ALIGNED(16, FFTSample, output)[2048]; ///< temporary buffer for MDCT input coefficients
-    int16_t* samples;                            ///< saved preprocessed input
+    int16_t *samples;                            ///< saved preprocessed input
 
     int samplerate_index;                        ///< MPEG-4 samplerate index
 
@@ -64,8 +63,8 @@ typedef struct AACEncContext {
     int cur_channel;
     int last_frame;
     float lambda;
+    float *scoefs;                               ///< scaled coefficients
     DECLARE_ALIGNED(16, int,   qcoefs)[96];      ///< quantized coefficients
-    DECLARE_ALIGNED(16, float, scoefs)[1024];    ///< scaled coefficients
 } AACEncContext;
 
 #endif /* AVCODEC_AACENC_H */