libvorbis takes float input, so we can just deinterleave/reorder the input
as-is instead of also converting.
---
libavcodec/libvorbis.c | 6 +++---
1 files changed, 3 insertions(+), 3 deletions(-)
diff --git a/libavcodec/libvorbis.c b/libavcodec/libvorbis.c
index 1fc9e1d..955c249 100644
--- a/libavcodec/libvorbis.c
+++ b/libavcodec/libvorbis.c
@@ -235,7 +235,7 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx,
unsigned char *packets,
{
OggVorbisContext *s = avctx->priv_data;
ogg_packet op;
- signed short *audio = data;
+ float *audio = data;
int pkt_size;
/* send samples to libvorbis */
@@ -250,7 +250,7 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx,
unsigned char *packets,
int co = (channels > 8) ? c :
ff_vorbis_encoding_channel_layout_offsets[channels -
1][c];
for (i = 0; i < samples; i++)
- buffer[c][i] = audio[i * channels + co] / 32768.f;
+ buffer[c][i] = audio[i * channels + co];
}
vorbis_analysis_wrote(&s->vd, samples);
} else {
@@ -314,7 +314,7 @@ AVCodec ff_libvorbis_encoder = {
.encode = oggvorbis_encode_frame,
.close = oggvorbis_encode_close,
.capabilities = CODEC_CAP_DELAY,
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
.priv_class = &class,
--
1.7.1
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