WMA: extend exponent range to 95

[frescor/ffmpeg.git] / libavcodec / alac.c

diff --git a/libavcodec/alac.c b/libavcodec/alac.c
index 288558dfdbb2b9e3c010f6b0c8113e7d2843a644..5c48a4b28f4a105a8a34dcc602e8bddf788ef90a 100644 (file)
--- a/libavcodec/alac.c
+++ b/libavcodec/alac.c
@@ -20,7 +20,7 @@
  */
 
 /**
- * @file alac.c
+ * @file libavcodec/alac.c
  * ALAC (Apple Lossless Audio Codec) decoder
  * @author 2005 David Hammerton
  *
@@ -53,9 +53,10 @@
 
 
 #include "avcodec.h"
-#include "bitstream.h"
+#include "get_bits.h"
 #include "bytestream.h"
 #include "unary.h"
+#include "mathops.h"
 
 #define ALAC_EXTRADATA_SIZE 36
 #define MAX_CHANNELS 2
@@ -68,7 +69,6 @@ typedef struct {
      * set this to 1 */
     int context_initialized;
 
-    int samplesize;
     int numchannels;
     int bytespersample;
 
@@ -116,6 +116,10 @@ static int alac_set_info(ALACContext *alac)
     alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
     ptr++;                          /* ??? */
     alac->setinfo_sample_size           = *ptr++;
+    if (alac->setinfo_sample_size > 32) {
+        av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n");
+        return -1;
+    }
     alac->setinfo_rice_historymult      = *ptr++;
     alac->setinfo_rice_initialhistory   = *ptr++;
     alac->setinfo_rice_kmodifier        = *ptr++;
@@ -200,7 +204,8 @@ static void bastardized_rice_decompress(ALACContext *alac,
 
         /* special case: there may be compressed blocks of 0 */
         if ((history < 128) && (output_count+1 < output_size)) {
-            int block_size, k;
+            int k;
+            unsigned int block_size;
 
             sign_modifier = 1;
 
@@ -209,6 +214,10 @@ static void bastardized_rice_decompress(ALACContext *alac,
             block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16);
 
             if (block_size > 0) {
+                if(block_size >= output_size - output_count){
+                    av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count);
+                    block_size= output_size - output_count - 1;
+                }
                 memset(&output_buffer[output_count+1], 0, block_size * 4);
                 output_count += block_size;
             }
@@ -221,11 +230,6 @@ static void bastardized_rice_decompress(ALACContext *alac,
     }
 }
 
-static inline int32_t extend_sign32(int32_t val, int bits)
-{
-    return (val << (32 - bits)) >> (32 - bits);
-}
-
 static inline int sign_only(int v)
 {
     return v ? FFSIGN(v) : 0;
@@ -265,7 +269,7 @@ static void predictor_decompress_fir_adapt(int32_t *error_buffer,
             prev_value = buffer_out[i];
             error_value = error_buffer[i+1];
             buffer_out[i+1] =
-                extend_sign32((prev_value + error_value), readsamplesize);
+                sign_extend((prev_value + error_value), readsamplesize);
         }
         return;
     }
@@ -276,7 +280,7 @@ static void predictor_decompress_fir_adapt(int32_t *error_buffer,
             int32_t val;
 
             val = buffer_out[i] + error_buffer[i+1];
-            val = extend_sign32(val, readsamplesize);
+            val = sign_extend(val, readsamplesize);
             buffer_out[i+1] = val;
         }
 
@@ -311,7 +315,7 @@ static void predictor_decompress_fir_adapt(int32_t *error_buffer,
             outval = (1 << (predictor_quantitization-1)) + sum;
             outval = outval >> predictor_quantitization;
             outval = outval + buffer_out[0] + error_val;
-            outval = extend_sign32(outval, readsamplesize);
+            outval = sign_extend(outval, readsamplesize);
 
             buffer_out[predictor_coef_num+1] = outval;
 
@@ -396,14 +400,16 @@ static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS],
 
 static int alac_decode_frame(AVCodecContext *avctx,
                              void *outbuffer, int *outputsize,
-                             const uint8_t *inbuffer, int input_buffer_size)
+                             AVPacket *avpkt)
 {
+    const uint8_t *inbuffer = avpkt->data;
+    int input_buffer_size = avpkt->size;
     ALACContext *alac = avctx->priv_data;
 
     int channels;
-    int32_t outputsamples;
+    unsigned int outputsamples;
     int hassize;
-    int readsamplesize;
+    unsigned int readsamplesize;
     int wasted_bytes;
     int isnotcompressed;
     uint8_t interlacing_shift;
@@ -453,12 +459,25 @@ static int alac_decode_frame(AVCodecContext *avctx,
 
     if (hassize) {
         /* now read the number of samples as a 32bit integer */
-        outputsamples = get_bits(&alac->gb, 32);
+        outputsamples = get_bits_long(&alac->gb, 32);
+        if(outputsamples > alac->setinfo_max_samples_per_frame){
+            av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame);
+            return -1;
+        }
     } else
         outputsamples = alac->setinfo_max_samples_per_frame;
 
+    if(outputsamples > *outputsize / alac->bytespersample){
+        av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
+        return -1;
+    }
+
     *outputsize = outputsamples * alac->bytespersample;
     readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1;
+    if (readsamplesize > MIN_CACHE_BITS) {
+        av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
+        return -1;
+    }
 
     if (!isnotcompressed) {
         /* so it is compressed */
@@ -518,37 +537,21 @@ static int alac_decode_frame(AVCodecContext *avctx,
         }
     } else {
         /* not compressed, easy case */
-        if (alac->setinfo_sample_size <= 16) {
-            int i, chan;
-            for (chan = 0; chan < channels; chan++)
-                for (i = 0; i < outputsamples; i++) {
-                    int32_t audiobits;
+        int i, chan;
+        for (i = 0; i < outputsamples; i++)
+            for (chan = 0; chan < channels; chan++) {
+                int32_t audiobits;
 
-                    audiobits = get_bits(&alac->gb, alac->setinfo_sample_size);
-                    audiobits = extend_sign32(audiobits, readsamplesize);
+                audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size);
 
-                    alac->outputsamples_buffer[chan][i] = audiobits;
-                }
-        } else {
-            int i, chan;
-            for (chan = 0; chan < channels; chan++)
-                for (i = 0; i < outputsamples; i++) {
-                    int32_t audiobits;
-
-                    audiobits = get_bits(&alac->gb, 16);
-                    /* special case of sign extension..
-                     * as we'll be ORing the low 16bits into this */
-                    audiobits = audiobits << 16;
-                    audiobits = audiobits >> (32 - alac->setinfo_sample_size);
-                    audiobits |= get_bits(&alac->gb, alac->setinfo_sample_size - 16);
-
-                    alac->outputsamples_buffer[chan][i] = audiobits;
-                }
-        }
+                alac->outputsamples_buffer[chan][i] = audiobits;
+            }
         /* wasted_bytes = 0; */
         interlacing_shift = 0;
         interlacing_leftweight = 0;
     }
+    if (get_bits(&alac->gb, 3) != 7)
+        av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
 
     switch(alac->setinfo_sample_size) {
     case 16:
@@ -578,6 +581,9 @@ static int alac_decode_frame(AVCodecContext *avctx,
         break;
     }
 
+    if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
+        av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb));
+
     return input_buffer_size;
 }
 
@@ -587,9 +593,9 @@ static av_cold int alac_decode_init(AVCodecContext * avctx)
     alac->avctx = avctx;
     alac->context_initialized = 0;
 
-    alac->samplesize = alac->avctx->bits_per_sample;
     alac->numchannels = alac->avctx->channels;
-    alac->bytespersample = (alac->samplesize / 8) * alac->numchannels;
+    alac->bytespersample = 2 * alac->numchannels;
+    avctx->sample_fmt = SAMPLE_FMT_S16;
 
     return 0;
 }
@@ -616,4 +622,5 @@ AVCodec alac_decoder = {
     NULL,
     alac_decode_close,
     alac_decode_frame,
+    .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
 };