[FFmpeg-cvslog] qdm2: refactor joined stereo support

[Luca Barbato]

ffmpeg | branch: master | Luca Barbato <lu_zero at gentoo.org> | Tue Jul  9 14:44:02 2013 +0200| [adadc3f2443d25b375e21e801516ccfd78e0b080] | committer: Luca Barbato

qdm2: refactor joined stereo support

qdm2 does support only two channels. Loop over the run once.

Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable at libav.org

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=adadc3f2443d25b375e21e801516ccfd78e0b080
---

 libavcodec/qdm2.c |   22 ++++++++++++----------
 1 file changed, 12 insertions(+), 10 deletions(-)

diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c
index 6343ce3..dcfe621 100644
--- a/libavcodec/qdm2.c
+++ b/libavcodec/qdm2.c
@@ -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)