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-rw-r--r--libavcodec/ra144enc.c20
1 files changed, 10 insertions, 10 deletions
diff --git a/libavcodec/ra144enc.c b/libavcodec/ra144enc.c
index c05c2435d7..2eac343165 100644
--- a/libavcodec/ra144enc.c
+++ b/libavcodec/ra144enc.c
@@ -2,20 +2,20 @@
* Real Audio 1.0 (14.4K) encoder
* Copyright (c) 2010 Francesco Lavra <francescolavra@interfree.it>
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -197,8 +197,8 @@ static void create_adapt_vect(float *vect, const int16_t *cb, int lag)
static int adaptive_cb_search(const int16_t *adapt_cb, float *work,
const float *coefs, float *data)
{
- int i, best_vect;
- float score, gain, best_score, best_gain;
+ int i, av_uninit(best_vect);
+ float score, gain, best_score, av_uninit(best_gain);
float exc[BLOCKSIZE];
gain = best_score = 0;
@@ -336,7 +336,8 @@ static void ra144_encode_subblock(RA144Context *ractx,
float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE];
int16_t cba_vect[BLOCKSIZE];
int cba_idx, cb1_idx, cb2_idx, gain;
- int i, n, m[3];
+ int i, n;
+ unsigned m[3];
float g[3];
float error, best_error;
@@ -446,10 +447,8 @@ static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
if (ractx->last_frame)
return 0;
- if ((ret = ff_alloc_packet(avpkt, FRAMESIZE))) {
- av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ if ((ret = ff_alloc_packet2(avctx, avpkt, FRAMESIZE)) < 0)
return ret;
- }
/**
* Since the LPC coefficients are calculated on a frame centered over the
@@ -554,5 +553,6 @@ AVCodec ff_ra_144_encoder = {
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
+ .supported_samplerates = (const int[]){ 8000, 0 },
.long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
};