avdevice/decklink_dec: fix build with older SDK

[ffmpeg] / libavcodec / qdm2.c

diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c
index 88b6b19d1123716a9dac520b6ef6079a8b20bac2..657b2da64dc8fa2721f37b25a70cf43e6f31bd88 100644 (file)
--- a/libavcodec/qdm2.c
+++ b/libavcodec/qdm2.c
@@ -408,7 +408,12 @@ static int fix_coding_method_array(int sb, int channels,
             }
             for (k = 0; k < run; k++) {
                 if (j + k < 128) {
-                    if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) {
+                    int sbjk = sb + (j + k) / 64;
+                    if (sbjk > 29) {
+                        SAMPLES_NEEDED
+                        continue;
+                    }
+                    if (coding_method[ch][sbjk][(j + k) % 64] > coding_method[ch][sb][j]) {
                         if (k > 0) {
                             SAMPLES_NEEDED
                             //not debugged, almost never used
@@ -1284,6 +1289,10 @@ static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
             }
             offset += (n - 2);
         } else {
+            if (local_int_10 <= 2) {
+                av_log(NULL, AV_LOG_ERROR, "qdm2_fft_decode_tones() stuck\n");
+                return;
+            }
             offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
             while (offset >= (local_int_10 - 1)) {
                 offset       += (1 - (local_int_10 - 1));
@@ -1325,6 +1334,9 @@ static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
         if (q->frequency_range > (local_int_14 + 1)) {
             int sub_packet = (local_int_20 + local_int_28);
 
+            if (q->fft_coefs_index + stereo >= FF_ARRAY_ELEMS(q->fft_coefs))
+                return;
+
             qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
                                       channel, exp, phase);
             if (stereo)
@@ -1695,13 +1707,19 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
     s->group_size = bytestream2_get_be32(&gb);
     s->fft_size = bytestream2_get_be32(&gb);
     s->checksum_size = bytestream2_get_be32(&gb);
-    if (s->checksum_size >= 1U << 28) {
-        av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
+    if (s->checksum_size >= 1U << 28 || s->checksum_size <= 1) {
+        av_log(avctx, AV_LOG_ERROR, "data block size invalid (%u)\n", s->checksum_size);
         return AVERROR_INVALIDDATA;
     }
 
     s->fft_order = av_log2(s->fft_size) + 1;
 
+    // Fail on unknown fft order
+    if ((s->fft_order < 7) || (s->fft_order > 9)) {
+        avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
+        return AVERROR_PATCHWELCOME;
+    }
+
     // something like max decodable tones
     s->group_order = av_log2(s->group_size) + 1;
     s->frame_size = s->group_size / 16; // 16 iterations per super block
@@ -1712,6 +1730,11 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
     s->sub_sampling = s->fft_order - 7;
     s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
 
+    if (s->frame_size * 4 >> s->sub_sampling > MPA_FRAME_SIZE) {
+        avpriv_request_sample(avctx, "large frames");
+        return AVERROR_PATCHWELCOME;
+    }
+
     switch ((s->sub_sampling * 2 + s->channels - 1)) {
         case 0: tmp = 40; break;
         case 1: tmp = 48; break;
@@ -1735,11 +1758,6 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
     else
         s->coeff_per_sb_select = 2;
 
-    // Fail on unknown fft order
-    if ((s->fft_order < 7) || (s->fft_order > 9)) {
-        avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
-        return AVERROR_PATCHWELCOME;
-    }
     if (s->fft_size != (1 << (s->fft_order - 1))) {
         av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
         return AVERROR_INVALIDDATA;