[FFmpeg-cvslog] avcodec/mpegaudiodec_float: Avoid indirection with float dsp function
Andreas Rheinhardt
git at videolan.org
Sat Sep 19 20:24:40 EEST 2020
ffmpeg | branch: master | Andreas Rheinhardt <andreas.rheinhardt at gmail.com> | Tue Sep 15 08:16:13 2020 +0200| [e9831b1e985a61f1f0089eccc877f2e5add2d58c] | committer: Andreas Rheinhardt
avcodec/mpegaudiodec_float: Avoid indirection with float dsp function
Do this by only keeping the only function pointer from the
AVFloatDSPContext that is needed lateron. This also allows to remove the
decoders' close function.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e9831b1e985a61f1f0089eccc877f2e5add2d58c
---
libavcodec/mpegaudiodec_float.c | 4 ----
libavcodec/mpegaudiodec_template.c | 30 +++++++++++-------------------
2 files changed, 11 insertions(+), 23 deletions(-)
diff --git a/libavcodec/mpegaudiodec_float.c b/libavcodec/mpegaudiodec_float.c
index ddfa5e0daa..0defdf3af1 100644
--- a/libavcodec/mpegaudiodec_float.c
+++ b/libavcodec/mpegaudiodec_float.c
@@ -46,7 +46,6 @@ AVCodec ff_mp1float_decoder = {
.id = AV_CODEC_ID_MP1,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
- .close = decode_close,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush,
@@ -64,7 +63,6 @@ AVCodec ff_mp2float_decoder = {
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
- .close = decode_close,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
@@ -80,7 +78,6 @@ AVCodec ff_mp3float_decoder = {
.id = AV_CODEC_ID_MP3,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
- .close = decode_close,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush,
@@ -97,7 +94,6 @@ AVCodec ff_mp3adufloat_decoder = {
.id = AV_CODEC_ID_MP3ADU,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
- .close = decode_close,
.decode = decode_frame_adu,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush,
diff --git a/libavcodec/mpegaudiodec_template.c b/libavcodec/mpegaudiodec_template.c
index 98759b8e01..d2b72497e1 100644
--- a/libavcodec/mpegaudiodec_template.c
+++ b/libavcodec/mpegaudiodec_template.c
@@ -87,7 +87,7 @@ typedef struct MPADecodeContext {
int err_recognition;
AVCodecContext* avctx;
MPADSPContext mpadsp;
- AVFloatDSPContext *fdsp;
+ void (*butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len);
AVFrame *frame;
uint32_t crc;
} MPADecodeContext;
@@ -410,16 +410,6 @@ static av_cold void decode_init_static(void)
}
}
-#if USE_FLOATS
-static av_cold int decode_close(AVCodecContext * avctx)
-{
- MPADecodeContext *s = avctx->priv_data;
- av_freep(&s->fdsp);
-
- return 0;
-}
-#endif
-
static av_cold int decode_init(AVCodecContext * avctx)
{
static int initialized_tables = 0;
@@ -433,9 +423,14 @@ static av_cold int decode_init(AVCodecContext * avctx)
s->avctx = avctx;
#if USE_FLOATS
- s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
- if (!s->fdsp)
- return AVERROR(ENOMEM);
+ {
+ AVFloatDSPContext *fdsp;
+ fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+ if (!fdsp)
+ return AVERROR(ENOMEM);
+ s->butterflies_float = fdsp->butterflies_float;
+ av_free(fdsp);
+ }
#endif
ff_mpadsp_init(&s->mpadsp);
@@ -1188,7 +1183,7 @@ found2:
/* NOTE: the 1/sqrt(2) normalization factor is included in the
global gain */
#if USE_FLOATS
- s->fdsp->butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
+ s->butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
#else
tab0 = g0->sb_hybrid;
tab1 = g1->sb_hybrid;
@@ -1871,9 +1866,6 @@ static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
MP3On4DecodeContext *s = avctx->priv_data;
int i;
- if (s->mp3decctx[0])
- av_freep(&s->mp3decctx[0]->fdsp);
-
for (i = 0; i < s->frames; i++)
av_freep(&s->mp3decctx[i]);
@@ -1938,7 +1930,7 @@ static av_cold int decode_init_mp3on4(AVCodecContext * avctx)
s->mp3decctx[i]->adu_mode = 1;
s->mp3decctx[i]->avctx = avctx;
s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
- s->mp3decctx[i]->fdsp = s->mp3decctx[0]->fdsp;
+ s->mp3decctx[i]->butterflies_float = s->mp3decctx[0]->butterflies_float;
}
return 0;
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