WMA: extend exponent range to 95

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

diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c
index 04e81c6faadf1125c518736a94245d872789d840..0ebc23d6cfe5679d5fbe8b2455b94b6ffa025432 100644 (file)
--- a/libavcodec/flacdec.c
+++ b/libavcodec/flacdec.c
@@ -38,7 +38,7 @@
 #include "libavutil/crc.h"
 #include "avcodec.h"
 #include "internal.h"
-#include "bitstream.h"
+#include "get_bits.h"
 #include "bytestream.h"
 #include "golomb.h"
 #include "flac.h"
@@ -125,6 +125,10 @@ static av_cold int flac_decode_init(AVCodecContext *avctx)
 
     /* initialize based on the demuxer-supplied streamdata header */
     ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
+    if (s->bps > 16)
+        avctx->sample_fmt = SAMPLE_FMT_S32;
+    else
+        avctx->sample_fmt = SAMPLE_FMT_S16;
     allocate_buffers(s);
     s->got_streaminfo = 1;
 
@@ -186,10 +190,6 @@ void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
     avctx->channels = s->channels;
     avctx->sample_rate = s->samplerate;
     avctx->bits_per_raw_sample = s->bps;
-    if (s->bps > 16)
-        avctx->sample_fmt = SAMPLE_FMT_S32;
-    else
-        avctx->sample_fmt = SAMPLE_FMT_S16;
 
     s->samples  = get_bits_long(&gb, 32) << 4;
     s->samples |= get_bits(&gb, 4);
@@ -358,7 +358,7 @@ static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order)
 {
     int i, j;
     int coeff_prec, qlevel;
-    int coeffs[pred_order];
+    int coeffs[32];
     int32_t *decoded = s->decoded[channel];
 
     /* warm up samples */
@@ -448,7 +448,7 @@ static inline int decode_subframe(FLACContext *s, int channel)
         s->curr_bps -= wasted;
     }
     if (s->curr_bps > 32) {
-        ff_log_missing_feature(s->avctx, "decorrelated bit depth > 32", 0);
+        av_log_missing_feature(s->avctx, "decorrelated bit depth > 32", 0);
         return -1;
     }
 
@@ -636,8 +636,10 @@ static int decode_frame(FLACContext *s)
 
 static int flac_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;
     FLACContext *s = avctx->priv_data;
     int i, j = 0, input_buf_size = 0, bytes_read = 0;
     int16_t *samples_16 = data;
@@ -804,7 +806,9 @@ AVCodec flac_decoder = {
     NULL,
     flac_decode_close,
     flac_decode_frame,
-    CODEC_CAP_DELAY,
+    CODEC_CAP_DELAY | CODEC_CAP_SUBFRAMES, /* FIXME: add a FLAC parser so that
+                                              we will not need to use either
+                                              of these capabilities */
     .flush= flac_flush,
     .long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
 };