Remove the add bias hack for the C version of DSPContext.float_to_int16_*().

1 files changed, 9 insertions, 29 deletions

diff --git a/libavcodec/vorbis_dec.c b/libavcodec/vorbis_dec.c
index cdb485a9c9..69b784c668 100644
--- a/libavcodec/vorbis_dec.c
+++ b/libavcodec/vorbis_dec.c
@@ -153,8 +153,7 @@ typedef struct vorbis_context_s {
     float        *channel_residues;
     float        *channel_floors;
     float        *saved;
-    uint_fast32_t add_bias; // for float->int conversion
-    uint_fast32_t exp_bias;
+    float         scale_bias; // for float->int conversion
 } vorbis_context;
 
 /* Helper functions */
@@ -932,8 +931,8 @@ static int vorbis_parse_id_hdr(vorbis_context *vc)
     vc->saved            =  av_mallocz((vc->blocksize[1] / 4) * vc->audio_channels * sizeof(float));
     vc->previous_window  = 0;
 
-    ff_mdct_init(&vc->mdct[0], bl0, 1, vc->exp_bias ? -(1 << 15) : -1.0);
-    ff_mdct_init(&vc->mdct[1], bl1, 1, vc->exp_bias ? -(1 << 15) : -1.0);
+    ff_mdct_init(&vc->mdct[0], bl0, 1, -vc->scale_bias);
+    ff_mdct_init(&vc->mdct[1], bl1, 1, -vc->scale_bias);
 
     AV_DEBUG(" vorbis version %d \n audio_channels %d \n audio_samplerate %d \n bitrate_max %d \n bitrate_nom %d \n bitrate_min %d \n blk_0 %d blk_1 %d \n ",
             vc->version, vc->audio_channels, vc->audio_samplerate, vc->bitrate_maximum, vc->bitrate_nominal, vc->bitrate_minimum, vc->blocksize[0], vc->blocksize[1]);
@@ -963,13 +962,7 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
     vc->avccontext = avccontext;
     dsputil_init(&vc->dsp, avccontext);
 
-    if (vc->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
-        vc->add_bias = 385;
-        vc->exp_bias = 0;
-    } else {
-        vc->add_bias = 0;
-        vc->exp_bias = 15 << 23;
-    }
+    vc->scale_bias = 32768.0f;
 
     if (!headers_len) {
         av_log(avccontext, AV_LOG_ERROR, "Extradata missing.\n");
@@ -1453,18 +1446,6 @@ void vorbis_inverse_coupling(float *mag, float *ang, int blocksize)
     }
 }
 
-static void copy_normalize(float *dst, float *src, int len, int exp_bias,
-                           float add_bias)
-{
-    int i;
-    if (exp_bias) {
-        memcpy(dst, src, len * sizeof(float));
-    } else {
-        for (i = 0; i < len; i++)
-            dst[i] = src[i] + add_bias;
-    }
-}
-
 // Decode the audio packet using the functions above
 
 static int vorbis_parse_audio_packet(vorbis_context *vc)
@@ -1484,7 +1465,6 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
     uint_fast8_t res_chan[255];
     uint_fast8_t res_num = 0;
     int_fast16_t retlen  = 0;
-    float fadd_bias = vc->add_bias;
 
     if (get_bits1(gb)) {
         av_log(vc->avccontext, AV_LOG_ERROR, "Not a Vorbis I audio packet.\n");
@@ -1595,13 +1575,13 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
         const float *win  = vc->win[blockflag & previous_window];
 
         if (blockflag == previous_window) {
-            vc->dsp.vector_fmul_window(ret, saved, buf, win, fadd_bias, blocksize / 4);
+            vc->dsp.vector_fmul_window(ret, saved, buf, win, 0, blocksize / 4);
         } else if (blockflag > previous_window) {
-            vc->dsp.vector_fmul_window(ret, saved, buf, win, fadd_bias, bs0 / 4);
-            copy_normalize(ret+bs0/2, buf+bs0/4, (bs1-bs0)/4, vc->exp_bias, fadd_bias);
+            vc->dsp.vector_fmul_window(ret, saved, buf, win, 0, bs0 / 4);
+            memcpy(ret+bs0/2, buf+bs0/4, ((bs1-bs0)/4) * sizeof(float));
         } else {
-            copy_normalize(ret, saved, (bs1 - bs0) / 4, vc->exp_bias, fadd_bias);
-            vc->dsp.vector_fmul_window(ret + (bs1 - bs0) / 4, saved + (bs1 - bs0) / 4, buf, win, fadd_bias, bs0 / 4);
+            memcpy(ret, saved, ((bs1 - bs0) / 4) * sizeof(float));
+            vc->dsp.vector_fmul_window(ret + (bs1 - bs0) / 4, saved + (bs1 - bs0) / 4, buf, win, 0, bs0 / 4);
         }
         memcpy(saved, buf + blocksize / 4, blocksize / 4 * sizeof(float));
     }