qdm2: refactor joined stereo support
[ffmpeg.git] / libavcodec / qdm2.c
index 6343ce3..dcfe621 100644 (file)
@@ -807,7 +807,7 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
                                        int length, int sb_min, int sb_max)
 {
     int sb, j, k, n, ch, run, channels;
-    int joined_stereo, zero_encoding, chs;
+    int joined_stereo, zero_encoding;
     int type34_first;
     float type34_div = 0;
     float type34_predictor;
@@ -961,16 +961,18 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
                 }
 
                 if (joined_stereo) {
-                    float tmp[10][MPA_MAX_CHANNELS];
-
-                    for (k = 0; k < run; k++) {
-                        tmp[k][0] = samples[k];
-                        tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
+                    for (k = 0; k < run && j + k < 128; k++) {
+                        q->sb_samples[0][j + k][sb] =
+                            q->tone_level[0][sb][(j + k) / 2] * samples[k];
+                        if (q->nb_channels == 2) {
+                            if (sign_bits[(j + k) / 8])
+                                q->sb_samples[1][j + k][sb] =
+                                    q->tone_level[1][sb][(j + k) / 2] * -samples[k];
+                            else
+                                q->sb_samples[1][j + k][sb] =
+                                    q->tone_level[1][sb][(j + k) / 2] * samples[k];
+                        }
                     }
-                    for (chs = 0; chs < q->nb_channels; chs++)
-                        for (k = 0; k < run; k++)
-                            if ((j + k) < 128)
-                                q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
                 } else {
                     for (k = 0; k < run; k++)
                         if ((j + k) < 128)