avformat/avio: Add Metacube support

[ffmpeg] / libavcodec / qdm2.c

diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c
index 88b6b19d1123716a9dac520b6ef6079a8b20bac2..5e4e741e59a42ce34a60cf0d0c09adb90ff624f2 100644 (file)
--- a/libavcodec/qdm2.c
+++ b/libavcodec/qdm2.c
@@ -36,6 +36,8 @@
 #include <stdio.h>
 
 #include "libavutil/channel_layout.h"
+#include "libavutil/mem_internal.h"
+#include "libavutil/thread.h"
 
 #define BITSTREAM_READER_LE
 #include "avcodec.h"
@@ -205,7 +207,7 @@ static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
     value = get_vlc2(gb, vlc->table, vlc->bits, depth);
 
     /* stage-2, 3 bits exponent escape sequence */
-    if (value-- == 0)
+    if (value < 0)
         value = get_bits(gb, get_bits(gb, 3) + 1);
 
     /* stage-3, optional */
@@ -408,7 +410,12 @@ 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 +1291,10 @@ 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 +1336,9 @@ 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)
@@ -1582,22 +1596,14 @@ static void qdm2_synthesis_filter(QDM2Context *q, int index)
 
 /**
  * Init static data (does not depend on specific file)
- *
- * @param q    context
  */
 static av_cold void qdm2_init_static_data(void) {
-    static int done;
-
-    if(done)
-        return;
-
     qdm2_init_vlc();
-    ff_mpa_synth_init_float(ff_mpa_synth_window_float);
     softclip_table_init();
     rnd_table_init();
     init_noise_samples();
 
-    done = 1;
+    ff_mpa_synth_init_float();
 }
 
 /**
@@ -1605,12 +1611,11 @@ static av_cold void qdm2_init_static_data(void) {
  */
 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
 {
+    static AVOnce init_static_once = AV_ONCE_INIT;
     QDM2Context *s = avctx->priv_data;
     int tmp_val, tmp, size;
     GetByteContext gb;
 
-    qdm2_init_static_data();
-
     /* extradata parsing
 
     Structure:
@@ -1695,13 +1700,19 @@ 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 +1723,11 @@ 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 +1751,6 @@ 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;
@@ -1750,6 +1761,8 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
 
     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
 
+    ff_thread_once(&init_static_once, qdm2_init_static_data);
+
     return 0;
 }
 
@@ -1856,7 +1869,7 @@ static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
     return s->checksum_size;
 }
 
-AVCodec ff_qdm2_decoder = {
+const AVCodec ff_qdm2_decoder = {
     .name             = "qdm2",
     .long_name        = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
     .type             = AVMEDIA_TYPE_AUDIO,
@@ -1865,5 +1878,6 @@ AVCodec ff_qdm2_decoder = {
     .init             = qdm2_decode_init,
     .close            = qdm2_decode_close,
     .decode           = qdm2_decode_frame,
-    .capabilities     = AV_CODEC_CAP_DR1,
+    .capabilities     = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
+    .caps_internal    = FF_CODEC_CAP_INIT_THREADSAFE,
 };