ac3enc: move ff_ac3_encode_frame() to ac3enc_template.c

[ffmpeg] / libavcodec / apedec.c

diff --git a/libavcodec/apedec.c b/libavcodec/apedec.c
index b8d1e9e254230f3823abdda8c4eff7b028515d01..d5a7efd94d016486a2350d4512949669d24e31f5 100644 (file)
--- a/libavcodec/apedec.c
+++ b/libavcodec/apedec.c
@@ -3,20 +3,20 @@
  * Copyright (c) 2007 Benjamin Zores <ben@geexbox.org>
  *  based upon libdemac from Dave Chapman.
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -25,9 +25,10 @@
 #include "dsputil.h"
 #include "get_bits.h"
 #include "bytestream.h"
+#include "libavutil/audioconvert.h"
 
 /**
- * @file libavcodec/apedec.c
+ * @file
  * Monkey's Audio lossless audio decoder
  */
 
@@ -198,8 +199,8 @@ static av_cold int ape_decode_init(AVCodecContext * avctx)
     }
 
     dsputil_init(&s->dsp, avctx);
-    avctx->sample_fmt = SAMPLE_FMT_S16;
-    avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;
+    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+    avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
     return 0;
 }
 
@@ -211,6 +212,7 @@ static av_cold int ape_decode_close(AVCodecContext * avctx)
     for (i = 0; i < APE_FILTER_LEVELS; i++)
         av_freep(&s->filterbuf[i]);
 
+    av_freep(&s->data);
     return 0;
 }
 
@@ -648,22 +650,16 @@ static void init_filter(APEContext * ctx, APEFilter *f, int16_t * buf, int order
     do_init_filter(&f[1], buf + order * 3 + HISTORY_SIZE, order);
 }
 
-static inline void do_apply_filter(APEContext * ctx, int version, APEFilter *f, int32_t *data, int count, int order, int fracbits)
+static void do_apply_filter(APEContext * ctx, int version, APEFilter *f, int32_t *data, int count, int order, int fracbits)
 {
     int res;
     int absres;
 
     while (count--) {
         /* round fixedpoint scalar product */
-        res = (ctx->dsp.scalarproduct_int16(f->delay - order, f->coeffs, order, 0) + (1 << (fracbits - 1))) >> fracbits;
-
-        if (*data < 0)
-            ctx->dsp.add_int16(f->coeffs, f->adaptcoeffs - order, order);
-        else if (*data > 0)
-            ctx->dsp.sub_int16(f->coeffs, f->adaptcoeffs - order, order);
-
+        res = ctx->dsp.scalarproduct_and_madd_int16(f->coeffs, f->delay - order, f->adaptcoeffs - order, order, APESIGN(*data));
+        res = (res + (1 << (fracbits - 1))) >> fracbits;
         res += *data;
-
         *data++ = res;
 
         /* Update the output history */
@@ -882,9 +878,15 @@ static int ape_decode_frame(AVCodecContext * avctx,
     return bytes_used;
 }
 
-AVCodec ape_decoder = {
+static void ape_flush(AVCodecContext *avctx)
+{
+    APEContext *s = avctx->priv_data;
+    s->samples= 0;
+}
+
+AVCodec ff_ape_decoder = {
     "ape",
-    CODEC_TYPE_AUDIO,
+    AVMEDIA_TYPE_AUDIO,
     CODEC_ID_APE,
     sizeof(APEContext),
     ape_decode_init,
@@ -892,5 +894,6 @@ AVCodec ape_decoder = {
     ape_decode_close,
     ape_decode_frame,
     .capabilities = CODEC_CAP_SUBFRAMES,
+    .flush = ape_flush,
     .long_name = NULL_IF_CONFIG_SMALL("Monkey's Audio"),
 };