Rename ff_parse_eval() to ff_eval_expr().

[ffmpeg] / libavcodec / flacenc.c

diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c
index 164e0c03240b164400e38ba1a0c87819b513ef6f..89d40d5ac24fe20eeda779aa77ac7016764bab8b 100644 (file)
--- a/libavcodec/flacenc.c
+++ b/libavcodec/flacenc.c
@@ -352,7 +352,7 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
     /* initialize MD5 context */
     s->md5ctx = av_malloc(av_md5_size);
     if(!s->md5ctx)
-        return AVERROR_NOMEM;
+        return AVERROR(ENOMEM);
     av_md5_init(s->md5ctx);
 
     streaminfo = av_malloc(FLAC_STREAMINFO_SIZE);
@@ -552,69 +552,6 @@ static uint32_t calc_rice_params_lpc(RiceContext *rc, int pmin, int pmax,
     return bits;
 }
 
-/**
- * Apply Welch window function to audio block
- */
-static void apply_welch_window(const int32_t *data, int len, double *w_data)
-{
-    int i, n2;
-    double w;
-    double c;
-
-    assert(!(len&1)); //the optimization in r11881 does not support odd len
-                      //if someone wants odd len extend the change in r11881
-
-    n2 = (len >> 1);
-    c = 2.0 / (len - 1.0);
-
-    w_data+=n2;
-      data+=n2;
-    for(i=0; i<n2; i++) {
-        w = c - n2 + i;
-        w = 1.0 - (w * w);
-        w_data[-i-1] = data[-i-1] * w;
-        w_data[+i  ] = data[+i  ] * w;
-    }
-}
-
-/**
- * Calculates autocorrelation data from audio samples
- * A Welch window function is applied before calculation.
- */
-void ff_flac_compute_autocorr(const int32_t *data, int len, int lag,
-                              double *autoc)
-{
-    int i, j;
-    double tmp[len + lag + 1];
-    double *data1= tmp + lag;
-
-    apply_welch_window(data, len, data1);
-
-    for(j=0; j<lag; j++)
-        data1[j-lag]= 0.0;
-    data1[len] = 0.0;
-
-    for(j=0; j<lag; j+=2){
-        double sum0 = 1.0, sum1 = 1.0;
-        for(i=j; i<len; i++){
-            sum0 += data1[i] * data1[i-j];
-            sum1 += data1[i] * data1[i-j-1];
-        }
-        autoc[j  ] = sum0;
-        autoc[j+1] = sum1;
-    }
-
-    if(j==lag){
-        double sum = 1.0;
-        for(i=j-1; i<len; i+=2){
-            sum += data1[i  ] * data1[i-j  ]
-                 + data1[i+1] * data1[i-j+1];
-        }
-        autoc[j] = sum;
-    }
-}
-
-
 static void encode_residual_verbatim(int32_t *res, int32_t *smp, int n)
 {
     assert(n > 0);
@@ -1316,7 +1253,7 @@ static av_cold int flac_encode_close(AVCodecContext *avctx)
 
 AVCodec flac_encoder = {
     "flac",
-    CODEC_TYPE_AUDIO,
+    AVMEDIA_TYPE_AUDIO,
     CODEC_ID_FLAC,
     sizeof(FlacEncodeContext),
     flac_encode_init,
@@ -1324,6 +1261,6 @@ AVCodec flac_encoder = {
     flac_encode_close,
     NULL,
     .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
 };