Get rid of some pointless '== NULL' / '!= 0' conditions in if statements.

[ffmpeg] / libavcodec / flacenc.c

diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c
index f51a4ee58ba1fc013b51b10923a8c83aef7bcb73..c1f862a68458397c795eb4bd35739052100d81c4 100644 (file)
--- a/libavcodec/flacenc.c
+++ b/libavcodec/flacenc.c
@@ -20,10 +20,9 @@
  */
 
 #include "libavutil/crc.h"
-#include "libavutil/lls.h"
 #include "libavutil/md5.h"
 #include "avcodec.h"
-#include "bitstream.h"
+#include "get_bits.h"
 #include "dsputil.h"
 #include "golomb.h"
 #include "lpc.h"
@@ -84,6 +83,7 @@ typedef struct FlacEncodeContext {
     int channels;
     int samplerate;
     int sr_code[2];
+    int max_blocksize;
     int min_framesize;
     int max_framesize;
     int max_encoded_framesize;
@@ -108,8 +108,8 @@ static void write_streaminfo(FlacEncodeContext *s, uint8_t *header)
     init_put_bits(&pb, header, FLAC_STREAMINFO_SIZE);
 
     /* streaminfo metadata block */
-    put_bits(&pb, 16, s->avctx->frame_size);
-    put_bits(&pb, 16, s->avctx->frame_size);
+    put_bits(&pb, 16, s->max_blocksize);
+    put_bits(&pb, 16, s->max_blocksize);
     put_bits(&pb, 24, s->min_framesize);
     put_bits(&pb, 24, s->max_framesize);
     put_bits(&pb, 20, s->samplerate);
@@ -327,6 +327,7 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
     } else {
         s->avctx->frame_size = select_blocksize(s->samplerate, s->options.block_time_ms);
     }
+    s->max_blocksize = s->avctx->frame_size;
     av_log(avctx, AV_LOG_DEBUG, " block size: %d\n", s->avctx->frame_size);
 
     /* set LPC precision */
@@ -345,11 +346,8 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
            s->options.lpc_coeff_precision);
 
     /* set maximum encoded frame size in verbatim mode */
-    if(s->channels == 2) {
-        s->max_framesize = 14 + ((s->avctx->frame_size * 33 + 7) >> 3);
-    } else {
-        s->max_framesize = 14 + (s->avctx->frame_size * s->channels * 2);
-    }
+    s->max_framesize = ff_flac_get_max_frame_size(s->avctx->frame_size,
+                                                  s->channels, 16);
 
     /* initialize MD5 context */
     s->md5ctx = av_malloc(av_md5_size);
@@ -554,69 +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=0; 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=0; 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);
@@ -1226,7 +1161,7 @@ static void output_frame_footer(FlacEncodeContext *s)
 
 static void update_md5_sum(FlacEncodeContext *s, int16_t *samples)
 {
-#ifdef WORDS_BIGENDIAN
+#if HAVE_BIGENDIAN
     int i;
     for(i = 0; i < s->frame.blocksize*s->channels; i++) {
         int16_t smp = le2me_16(samples[i]);
@@ -1326,6 +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)"),
 };