roqvideo: Remove a totally unused dspcontext

[ffmpeg] / libavcodec / mpegaudiodec.c

diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c
index d82432c775492108a8d9bb63599e05853aa79c22..6c1e8af845b7acffebc16441376d571b7d8eabea 100644 (file)
--- a/libavcodec/mpegaudiodec.c
+++ b/libavcodec/mpegaudiodec.c
@@ -29,6 +29,7 @@
 #include "get_bits.h"
 #include "mathops.h"
 #include "mpegaudiodsp.h"
+#include "dsputil.h"
 
 /*
  * TODO:
@@ -80,6 +81,7 @@ typedef struct MPADecodeContext {
     int err_recognition;
     AVCodecContext* avctx;
     MPADSPContext mpadsp;
+    DSPContext dsp;
     AVFrame frame;
 } MPADecodeContext;
 
@@ -172,9 +174,12 @@ static void ff_region_offset2size(GranuleDef *g)
 
 static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g)
 {
-    if (g->block_type == 2)
-        g->region_size[0] = (36 / 2);
-    else {
+    if (g->block_type == 2) {
+        if (s->sample_rate_index != 8)
+            g->region_size[0] = (36 / 2);
+        else
+            g->region_size[0] = (72 / 2);
+    } else {
         if (s->sample_rate_index <= 2)
             g->region_size[0] = (36 / 2);
         else if (s->sample_rate_index != 8)
@@ -199,14 +204,12 @@ static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
     if (g->block_type == 2) {
         if (g->switch_point) {
             /* if switched mode, we handle the 36 first samples as
-                long blocks.  For 8000Hz, we handle the 48 first
-                exponents as long blocks (XXX: check this!) */
+                long blocks.  For 8000Hz, we handle the 72 first
+                exponents as long blocks */
             if (s->sample_rate_index <= 2)
                 g->long_end = 8;
-            else if (s->sample_rate_index != 8)
-                g->long_end = 6;
             else
-                g->long_end = 4; /* 8000 Hz */
+                g->long_end = 6;
 
             g->short_start = 2 + (s->sample_rate_index != 8);
         } else {
@@ -302,11 +305,8 @@ static av_cold void decode_init_static(void)
     for (i = 1; i < 16; i++) {
         const HuffTable *h = &mpa_huff_tables[i];
         int xsize, x, y;
-        uint8_t  tmp_bits [512];
-        uint16_t tmp_codes[512];
-
-        memset(tmp_bits , 0, sizeof(tmp_bits ));
-        memset(tmp_codes, 0, sizeof(tmp_codes));
+        uint8_t  tmp_bits [512] = { 0 };
+        uint16_t tmp_codes[512] = { 0 };
 
         xsize = h->xsize;
 
@@ -432,6 +432,7 @@ static av_cold int decode_init(AVCodecContext * avctx)
     s->avctx = avctx;
 
     ff_mpadsp_init(&s->mpadsp);
+    ff_dsputil_init(&s->dsp, avctx);
 
     avctx->sample_fmt= OUT_FMT;
     s->err_recognition = avctx->err_recognition;
@@ -1018,7 +1019,7 @@ static void reorder_block(MPADecodeContext *s, GranuleDef *g)
         if (s->sample_rate_index != 8)
             ptr = g->sb_hybrid + 36;
         else
-            ptr = g->sb_hybrid + 48;
+            ptr = g->sb_hybrid + 72;
     } else {
         ptr = g->sb_hybrid;
     }
@@ -1153,6 +1154,9 @@ found2:
         /* ms stereo ONLY */
         /* NOTE: the 1/sqrt(2) normalization factor is included in the
            global gain */
+#if CONFIG_FLOAT
+       s-> dsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
+#else
         tab0 = g0->sb_hybrid;
         tab1 = g1->sb_hybrid;
         for (i = 0; i < 576; i++) {
@@ -1161,6 +1165,7 @@ found2:
             tab0[i] = tmp0 + tmp1;
             tab1[i] = tmp0 - tmp1;
         }
+#endif
     }
 }
 
@@ -1529,7 +1534,7 @@ static int mp_decode_layer3(MPADecodeContext *s)
             huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
         } /* ch */
 
-        if (s->nb_channels == 2)
+        if (s->mode == MPA_JSTEREO)
             compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
 
         for (ch = 0; ch < s->nb_channels; ch++) {
@@ -1653,7 +1658,6 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
     avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
     if (!avctx->bit_rate)
         avctx->bit_rate = s->bit_rate;
-    avctx->sub_id = s->layer;
 
     if (s->frame_size <= 0 || s->frame_size > buf_size) {
         av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
@@ -1726,11 +1730,14 @@ static int decode_frame_adu(AVCodecContext *avctx, void *data,
     avctx->channels    = s->nb_channels;
     if (!avctx->bit_rate)
         avctx->bit_rate = s->bit_rate;
-    avctx->sub_id = s->layer;
 
     s->frame_size = len;
 
     out_size = mp_decode_frame(s, NULL, buf, buf_size);
+    if (out_size < 0) {
+        av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
+        return AVERROR_INVALIDDATA;
+    }
 
     *got_frame_ptr   = 1;
     *(AVFrame *)data = s->frame;