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authorNathan Caldwell <saintdev@gmail.com>2011-12-17 18:45:55 -0700
committerAlex Converse <alex.converse@gmail.com>2012-01-23 11:40:46 -0800
commit025ccf1f8bd669f45b628acf51e7febcb1fddd3b (patch)
tree01729ec4ed66b3c439fe963b3c3e8a512034af31 /libavcodec/aacenc.c
parent6381f913d19d78513bab06fad7c50548975475a3 (diff)
downloadffmpeg-025ccf1f8bd669f45b628acf51e7febcb1fddd3b.tar.gz
aacenc: Request normalized float samples instead of converting s16 samples to float.
Signed-off-by: Alex Converse <alex.converse@gmail.com>
Diffstat (limited to 'libavcodec/aacenc.c')
-rw-r--r--libavcodec/aacenc.c12
1 files changed, 6 insertions, 6 deletions
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index 33396b2e83..3bd04bffe1 100644
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -167,7 +167,7 @@ static void put_audio_specific_config(AVCodecContext *avctx)
}
static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
- SingleChannelElement *sce, short *audio)
+ SingleChannelElement *sce, float *audio)
{
int i, k;
const int chans = avctx->channels;
@@ -434,7 +434,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
uint8_t *frame, int buf_size, void *data)
{
AACEncContext *s = avctx->priv_data;
- int16_t *samples = s->samples, *samples2, *la;
+ float *samples = s->samples, *samples2, *la;
ChannelElement *cpe;
int i, ch, w, g, chans, tag, start_ch;
int chan_el_counter[4];
@@ -452,7 +452,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
for (i = 0; i < s->chan_map[0]; i++) {
tag = s->chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
- ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch,
+ ff_psy_preprocess(s->psypp, (float*)data + start_ch,
samples2 + start_ch, start_ch, chans);
start_ch += chans;
}
@@ -621,9 +621,9 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
ff_init_ff_sine_windows(10);
ff_init_ff_sine_windows(7);
- if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 1.0))
+ if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))
return ret;
- if (ret = ff_mdct_init(&s->mdct128, 8, 0, 1.0))
+ if (ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0))
return ret;
return 0;
@@ -722,7 +722,7 @@ AVCodec ff_aac_encoder = {
.encode = aac_encode_frame,
.close = aac_encode_end,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
- .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+ .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
.priv_class = &aacenc_class,
};