X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Falacenc.c;h=fe03bb7dad11c69400ee5a99c14c5e27f0658a08;hb=492cc1bef3d1b47b576cae8686b196368290ffe6;hp=0fad99febdb0cc7e785955d31a125243f6a68c34;hpb=edac49daf5f703aa4e742ecdd747658e82d91b33;p=ffmpeg

diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c
index 0fad99febdb..fe03bb7dad1 100644
--- a/libavcodec/alacenc.c
+++ b/libavcodec/alacenc.c
@@ -2,20 +2,20 @@
  * ALAC audio encoder
  * Copyright (c) 2008  Jaikrishnan Menon <realityman@gmx.net>
  *
- * 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
  */
 
@@ -51,11 +51,11 @@ typedef struct RiceContext {
     int rice_modifier;
 } RiceContext;
 
-typedef struct LPCContext {
+typedef struct AlacLPCContext {
     int lpc_order;
     int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
     int lpc_quant;
-} LPCContext;
+} AlacLPCContext;
 
 typedef struct AlacEncodeContext {
     int compression_level;
@@ -69,8 +69,8 @@ typedef struct AlacEncodeContext {
     int interlacing_leftweight;
     PutBitContext pbctx;
     RiceContext rc;
-    LPCContext lpc[MAX_CHANNELS];
-    DSPContext dspctx;
+    AlacLPCContext lpc[MAX_CHANNELS];
+    LPCContext lpc_ctx;
     AVCodecContext *avctx;
 } AlacEncodeContext;
 
@@ -141,12 +141,12 @@ static void calc_predictor_params(AlacEncodeContext *s, int ch)
         s->lpc[ch].lpc_coeff[4] =   80;
         s->lpc[ch].lpc_coeff[5] =  -25;
     } else {
-        opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch],
+        opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch],
                                       s->avctx->frame_size,
                                       s->min_prediction_order,
                                       s->max_prediction_order,
                                       ALAC_MAX_LPC_PRECISION, coefs, shift,
-                                      AV_LPC_TYPE_LEVINSON, 0,
+                                      FF_LPC_TYPE_LEVINSON, 0,
                                       ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
 
         s->lpc[ch].lpc_order = opt_order;
@@ -237,7 +237,7 @@ static void alac_stereo_decorrelation(AlacEncodeContext *s)
 static void alac_linear_predictor(AlacEncodeContext *s, int ch)
 {
     int i;
-    LPCContext lpc = s->lpc[ch];
+    AlacLPCContext lpc = s->lpc[ch];
 
     if(lpc.lpc_order == 31) {
         s->predictor_buf[0] = s->sample_buf[ch][0];
@@ -378,12 +378,13 @@ static void write_compressed_frame(AlacEncodeContext *s)
 static av_cold int alac_encode_init(AVCodecContext *avctx)
 {
     AlacEncodeContext *s    = avctx->priv_data;
+    int ret;
     uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
 
     avctx->frame_size      = DEFAULT_FRAME_SIZE;
     avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE;
 
-    if(avctx->sample_fmt != SAMPLE_FMT_S16) {
+    if(avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
         av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
         return -1;
     }
@@ -455,9 +456,10 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
     avctx->coded_frame->key_frame = 1;
 
     s->avctx = avctx;
-    dsputil_init(&s->dspctx, avctx);
+    ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size, s->max_prediction_order,
+                      FF_LPC_TYPE_LEVINSON);
 
-    return 0;
+    return ret;
 }
 
 static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
@@ -513,21 +515,23 @@ verbatim:
 
 static av_cold int alac_encode_close(AVCodecContext *avctx)
 {
+    AlacEncodeContext *s = avctx->priv_data;
+    ff_lpc_end(&s->lpc_ctx);
     av_freep(&avctx->extradata);
     avctx->extradata_size = 0;
     av_freep(&avctx->coded_frame);
     return 0;
 }
 
-AVCodec alac_encoder = {
-    "alac",
-    AVMEDIA_TYPE_AUDIO,
-    CODEC_ID_ALAC,
-    sizeof(AlacEncodeContext),
-    alac_encode_init,
-    alac_encode_frame,
-    alac_encode_close,
+AVCodec ff_alac_encoder = {
+    .name           = "alac",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_ALAC,
+    .priv_data_size = sizeof(AlacEncodeContext),
+    .init           = alac_encode_init,
+    .encode         = alac_encode_frame,
+    .close          = alac_encode_close,
     .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
-    .sample_fmts = (const enum SampleFormat[]){ SAMPLE_FMT_S16, SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
 };