aacdec: remove sf_scale and sf_offset.

[ffmpeg] / libavcodec / vmdav.c

diff --git a/libavcodec/vmdav.c b/libavcodec/vmdav.c
index b3103ec220c8ca04200f114f5d4832d529c51726..710c2028cf15cf665706e842033b244aca86f90c 100644 (file)
--- a/libavcodec/vmdav.c
+++ b/libavcodec/vmdav.c
@@ -2,20 +2,20 @@
  * Sierra VMD Audio & Video Decoders
  * Copyright (C) 2004 the ffmpeg project
  *
- * 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
  */
 
@@ -420,9 +420,7 @@ static av_cold int vmdvideo_decode_end(AVCodecContext *avctx)
 
 typedef struct VmdAudioContext {
     AVCodecContext *avctx;
-    int channels;
-    int bits;
-    int block_align;
+    int out_bps;
     int predictors[2];
 } VmdAudioContext;
 
@@ -447,13 +445,16 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
     VmdAudioContext *s = avctx->priv_data;
 
     s->avctx = avctx;
-    s->channels = avctx->channels;
-    s->bits = avctx->bits_per_coded_sample;
-    s->block_align = avctx->block_align;
-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+    if (avctx->bits_per_coded_sample == 16)
+        avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+    else
+        avctx->sample_fmt = AV_SAMPLE_FMT_U8;
+    s->out_bps = av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3;
 
-    av_log(s->avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, block align = %d, sample rate = %d\n",
-            s->channels, s->bits, s->block_align, avctx->sample_rate);
+    av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
+           "block align = %d, sample rate = %d\n",
+           avctx->channels, avctx->bits_per_coded_sample, avctx->block_align,
+           avctx->sample_rate);
 
     return 0;
 }
@@ -479,24 +480,20 @@ static void vmdaudio_decode_audio(VmdAudioContext *s, unsigned char *data,
 static int vmdaudio_loadsound(VmdAudioContext *s, unsigned char *data,
     const uint8_t *buf, int silent_chunks, int data_size)
 {
-    int i;
-    int silent_size = s->block_align * silent_chunks * 2;
+    int silent_size = s->avctx->block_align * silent_chunks * s->out_bps;
 
     if (silent_chunks) {
-        memset(data, 0, silent_size);
+        memset(data, s->out_bps == 2 ? 0x00 : 0x80, silent_size);
         data += silent_size;
     }
-        if (s->bits == 16)
-            vmdaudio_decode_audio(s, data, buf, data_size, s->channels == 2);
-        else {
-            /* copy the data but convert it to signed */
-            for (i = 0; i < data_size; i++){
-                *data++ = buf[i] + 0x80;
-                *data++ = buf[i] + 0x80;
-            }
-        }
+    if (s->avctx->bits_per_coded_sample == 16)
+        vmdaudio_decode_audio(s, data, buf, data_size, s->avctx->channels == 2);
+    else {
+        /* just copy the data */
+        memcpy(data, buf, data_size);
+    }
 
-    return silent_size + data_size * 2;
+    return silent_size + data_size * s->out_bps;
 }
 
 static int vmdaudio_decode_frame(AVCodecContext *avctx,
@@ -506,7 +503,7 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx,
     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
     VmdAudioContext *s = avctx->priv_data;
-    int block_type;
+    int block_type, silent_chunks;
     unsigned char *output_samples = (unsigned char *)data;
 
     if (buf_size < 16) {
@@ -523,23 +520,23 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx,
     buf      += 16;
     buf_size -= 16;
 
-    if (block_type == BLOCK_TYPE_AUDIO) {
-        /* the chunk contains audio */
-        *data_size = vmdaudio_loadsound(s, output_samples, buf, 0, buf_size);
-    } else if (block_type == BLOCK_TYPE_INITIAL) {
-        /* initial chunk, may contain audio and silence */
+    silent_chunks = 0;
+    if (block_type == BLOCK_TYPE_INITIAL) {
         uint32_t flags = AV_RB32(buf);
-        int silent_chunks = av_popcount(flags);
+        silent_chunks  = av_popcount(flags);
         buf      += 4;
         buf_size -= 4;
-        if(*data_size < (s->block_align*silent_chunks + buf_size) * 2)
-            return -1;
-        *data_size = vmdaudio_loadsound(s, output_samples, buf, silent_chunks, buf_size);
     } else if (block_type == BLOCK_TYPE_SILENCE) {
-        /* silent chunk */
-        *data_size = vmdaudio_loadsound(s, output_samples, buf, 1, 0);
+        silent_chunks = 1;
+        buf_size = 0; // should already be zero but set it just to be sure
     }
 
+    /* ensure output buffer is large enough */
+    if (*data_size < (avctx->block_align*silent_chunks + buf_size) * s->out_bps)
+        return -1;
+
+    *data_size = vmdaudio_loadsound(s, output_samples, buf, silent_chunks, buf_size);
+
     return avpkt->size;
 }