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authorJustin Ruggles <justin.ruggles@gmail.com>2011-02-23 13:11:11 -0500
committerRonald S. Bultje <rsbultje@gmail.com>2011-02-23 21:51:44 -0500
commit1e86d685e0935077766c645e49b8533d41ca11cb (patch)
tree51649cfa9ad8d915601ca7aea4298a579c007fe9 /libavcodec/vmdav.c
parent1574eff3d23ad799d25454a449b01f94795495ea (diff)
downloadffmpeg-1e86d685e0935077766c645e49b8533d41ca11cb.tar.gz
vmdaudio: add out_bps to VmdAudioContext and use it to replace hard-coded sample size.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Diffstat (limited to 'libavcodec/vmdav.c')
-rw-r--r--libavcodec/vmdav.c8
1 files changed, 5 insertions, 3 deletions
diff --git a/libavcodec/vmdav.c b/libavcodec/vmdav.c
index 446e2c1dbc..bd71ae46c7 100644
--- a/libavcodec/vmdav.c
+++ b/libavcodec/vmdav.c
@@ -420,6 +420,7 @@ static av_cold int vmdvideo_decode_end(AVCodecContext *avctx)
typedef struct VmdAudioContext {
AVCodecContext *avctx;
+ int out_bps;
int channels;
int bits;
int block_align;
@@ -451,6 +452,7 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
s->bits = avctx->bits_per_coded_sample;
s->block_align = avctx->block_align;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ s->out_bps = av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3;
av_log(s->avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, block align = %d, sample rate = %d\n",
s->channels, s->bits, s->block_align, avctx->sample_rate);
@@ -480,7 +482,7 @@ static int vmdaudio_loadsound(VmdAudioContext *s, unsigned char *data,
const uint8_t *buf, int silent_chunks, int data_size)
{
int i;
- int silent_size = s->block_align * silent_chunks * 2;
+ int silent_size = s->block_align * silent_chunks * s->out_bps;
if (silent_chunks) {
memset(data, 0, silent_size);
@@ -496,7 +498,7 @@ static int vmdaudio_loadsound(VmdAudioContext *s, unsigned char *data,
}
}
- return silent_size + data_size * 2;
+ return silent_size + data_size * s->out_bps;
}
static int vmdaudio_decode_frame(AVCodecContext *avctx,
@@ -535,7 +537,7 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx,
}
/* ensure output buffer is large enough */
- if (*data_size < (s->block_align*silent_chunks + buf_size) * 2)
+ if (*data_size < (s->block_align*silent_chunks + buf_size) * s->out_bps)
return -1;
*data_size = vmdaudio_loadsound(s, output_samples, buf, silent_chunks, buf_size);