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@gmail.com>

diff --git a/libavcodec/mpegaudiodec_float.c b/libavcodec/mpegaudiodec_float.c
index ddfa5e0daaa28db98afb5b4c5d3e57ea8225bb45..0defdf3af16ffbe01262fd4fb53c50e93617be11 100644 (file)
--- 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 98759b8e01e6a5e048a71adc4d2eed24dc41f7e0..d2b72497e10a25e47f0feccd6901736e68f06e6f 100644 (file)
--- 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;