Fix overflow in pred16x16_plane x86 simd code. Fixes issue 2547.

[ffmpeg] / libavcodec / sonic.c

diff --git a/libavcodec/sonic.c b/libavcodec/sonic.c
index 79641ae5139c26d1ec0c842349b20201cdc915c4..aff155d57fab3780a0d5144ea022a1215a31b71a 100644 (file)
--- a/libavcodec/sonic.c
+++ b/libavcodec/sonic.c
@@ -19,11 +19,11 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 #include "avcodec.h"
-#include "bitstream.h"
+#include "get_bits.h"
 #include "golomb.h"
 
 /**
- * @file sonic.c
+ * @file
  * Simple free lossless/lossy audio codec
  * Based on Paul Francis Harrison's Bonk (http://www.logarithmic.net/pfh/bonk)
  * Written and designed by Alex Beregszaszi
@@ -408,7 +408,7 @@ static int predictor_calc_error(int *k, int *state, int order, int error)
     return x;
 }
 
-#ifdef CONFIG_ENCODERS
+#if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER
 // Heavily modified Levinson-Durbin algorithm which
 // copes better with quantization, and calculates the
 // actual whitened result as it goes.
@@ -479,12 +479,6 @@ static void modified_levinson_durbin(int *window, int window_entries,
 
     av_free(state);
 }
-#endif /* CONFIG_ENCODERS */
-
-static int samplerate_table[] =
-    { 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 };
-
-#ifdef CONFIG_ENCODERS
 
 static inline int code_samplerate(int samplerate)
 {
@@ -748,9 +742,12 @@ static int sonic_encode_frame(AVCodecContext *avctx,
     flush_put_bits(&pb);
     return (put_bits_count(&pb)+7)/8;
 }
-#endif //CONFIG_ENCODERS
+#endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */
+
+#if CONFIG_SONIC_DECODER
+static const int samplerate_table[] =
+    { 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 };
 
-#ifdef CONFIG_DECODERS
 static av_cold int sonic_decode_init(AVCodecContext *avctx)
 {
     SonicContext *s = avctx->priv_data;
@@ -828,6 +825,7 @@ static av_cold int sonic_decode_init(AVCodecContext *avctx)
     }
     s->int_samples = av_mallocz(4* s->frame_size);
 
+    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
     return 0;
 }
 
@@ -851,8 +849,10 @@ static av_cold int sonic_decode_close(AVCodecContext *avctx)
 
 static int sonic_decode_frame(AVCodecContext *avctx,
                             void *data, int *data_size,
-                            const uint8_t *buf, int buf_size)
+                            AVPacket *avpkt)
 {
+    const uint8_t *buf = avpkt->data;
+    int buf_size = avpkt->size;
     SonicContext *s = avctx->priv_data;
     GetBitContext gb;
     int i, quant, ch, j;
@@ -934,41 +934,44 @@ static int sonic_decode_frame(AVCodecContext *avctx,
 
     return (get_bits_count(&gb)+7)/8;
 }
-#endif
 
-#ifdef CONFIG_ENCODERS
+AVCodec sonic_decoder = {
+    "sonic",
+    AVMEDIA_TYPE_AUDIO,
+    CODEC_ID_SONIC,
+    sizeof(SonicContext),
+    sonic_decode_init,
+    NULL,
+    sonic_decode_close,
+    sonic_decode_frame,
+    .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
+};
+#endif /* CONFIG_SONIC_DECODER */
+
+#if CONFIG_SONIC_ENCODER
 AVCodec sonic_encoder = {
     "sonic",
-    CODEC_TYPE_AUDIO,
+    AVMEDIA_TYPE_AUDIO,
     CODEC_ID_SONIC,
     sizeof(SonicContext),
     sonic_encode_init,
     sonic_encode_frame,
     sonic_encode_close,
     NULL,
+    .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
 };
+#endif
 
+#if CONFIG_SONIC_LS_ENCODER
 AVCodec sonic_ls_encoder = {
     "sonicls",
-    CODEC_TYPE_AUDIO,
+    AVMEDIA_TYPE_AUDIO,
     CODEC_ID_SONIC_LS,
     sizeof(SonicContext),
     sonic_encode_init,
     sonic_encode_frame,
     sonic_encode_close,
     NULL,
-};
-#endif
-
-#ifdef CONFIG_DECODERS
-AVCodec sonic_decoder = {
-    "sonic",
-    CODEC_TYPE_AUDIO,
-    CODEC_ID_SONIC,
-    sizeof(SonicContext),
-    sonic_decode_init,
-    NULL,
-    sonic_decode_close,
-    sonic_decode_frame,
+    .long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"),
 };
 #endif