ansi: stop using deprecated avcodec_set_dimensions

[ffmpeg] / libavcodec / qdm2.c

diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c
index fd1e7bb99cfc8dc2457cdf7b5972f710cd316d41..04e7def067ce20e6c39f875ae2b30055327f8e1a 100644 (file)
--- a/libavcodec/qdm2.c
+++ b/libavcodec/qdm2.c
@@ -525,8 +525,8 @@ static void build_sb_samples_from_noise(QDM2Context *q, int sb)
  * @param channels         number of channels
  * @param coding_method    q->coding_method[0][0][0]
  */
-static void fix_coding_method_array(int sb, int channels,
-                                    sb_int8_array coding_method)
+static int fix_coding_method_array(int sb, int channels,
+                                   sb_int8_array coding_method)
 {
     int j, k;
     int ch;
@@ -534,6 +534,8 @@ static void fix_coding_method_array(int sb, int channels,
 
     for (ch = 0; ch < channels; ch++) {
         for (j = 0; j < 64; ) {
+            if (coding_method[ch][sb][j] < 8)
+                return -1;
             if ((coding_method[ch][sb][j] - 8) > 22) {
                 run      = 1;
                 case_val = 8;
@@ -579,6 +581,7 @@ static void fix_coding_method_array(int sb, int channels,
             j += run;
         }
     }
+    return 0;
 }
 
 /**
@@ -807,7 +810,7 @@ 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;
@@ -816,14 +819,12 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
     if (length == 0) {
         // If no data use noise
         for (sb=sb_min; sb < sb_max; sb++)
-            build_sb_samples_from_noise (q, sb);
+            build_sb_samples_from_noise(q, sb);
 
         return;
     }
 
     for (sb = sb_min; sb < sb_max; sb++) {
-        FIX_NOISE_IDX(q->noise_idx);
-
         channels = q->nb_channels;
 
         if (q->nb_channels <= 1 || sb < 12)
@@ -831,22 +832,27 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
         else if (sb >= 24)
             joined_stereo = 1;
         else
-            joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0;
+            joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
 
         if (joined_stereo) {
             if (get_bits_left(gb) >= 16)
                 for (j = 0; j < 16; j++)
-                    sign_bits[j] = get_bits1 (gb);
+                    sign_bits[j] = get_bits1(gb);
 
             for (j = 0; j < 64; j++)
                 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
                     q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
 
-            fix_coding_method_array(sb, q->nb_channels, q->coding_method);
+            if (fix_coding_method_array(sb, q->nb_channels,
+                                            q->coding_method)) {
+                build_sb_samples_from_noise(q, sb);
+                continue;
+            }
             channels = 1;
         }
 
         for (ch = 0; ch < channels; ch++) {
+            FIX_NOISE_IDX(q->noise_idx);
             zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
             type34_predictor = 0.0;
             type34_first = 1;
@@ -962,16 +968,18 @@ 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)
@@ -1031,7 +1039,7 @@ static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
  * @param q         context
  * @param gb        bitreader context
  */
-static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
+static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
 {
     int sb, j, k, n, ch;
 
@@ -1994,6 +2002,7 @@ static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
 
 AVCodec ff_qdm2_decoder = {
     .name             = "qdm2",
+    .long_name        = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
     .type             = AVMEDIA_TYPE_AUDIO,
     .id               = AV_CODEC_ID_QDM2,
     .priv_data_size   = sizeof(QDM2Context),
@@ -2002,5 +2011,4 @@ AVCodec ff_qdm2_decoder = {
     .close            = qdm2_decode_close,
     .decode           = qdm2_decode_frame,
     .capabilities     = CODEC_CAP_DR1,
-    .long_name        = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
 };