1 files changed, 104 insertions, 0 deletions
diff --git a/ffmpeg1/libavcodec/aac_ac3_parser.c b/ffmpeg1/libavcodec/aac_ac3_parser.c
new file mode 100644
index 0000000..6f1e188
--- /dev/null
+++ b/ffmpeg1/libavcodec/aac_ac3_parser.c
@@ -0,0 +1,104 @@
+/*
+ * Common AAC and AC-3 parser
+ * Copyright (c) 2003 Fabrice Bellard
+ * Copyright (c) 2003 Michael Niedermayer
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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,
+ * 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
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/common.h"
+#include "parser.h"
+#include "aac_ac3_parser.h"
+
+int ff_aac_ac3_parse(AVCodecParserContext *s1,
+                     AVCodecContext *avctx,
+                     const uint8_t **poutbuf, int *poutbuf_size,
+                     const uint8_t *buf, int buf_size)
+{
+    AACAC3ParseContext *s = s1->priv_data;
+    ParseContext *pc = &s->pc;
+    int len, i;
+    int new_frame_start;
+
+get_next:
+    i=END_NOT_FOUND;
+    if(s->remaining_size <= buf_size){
+        if(s->remaining_size && !s->need_next_header){
+            i= s->remaining_size;
+            s->remaining_size = 0;
+        }else{ //we need a header first
+            len=0;
+            for(i=s->remaining_size; i<buf_size; i++){
+                s->state = (s->state<<8) + buf[i];
+                if((len=s->sync(s->state, s, &s->need_next_header, &new_frame_start)))
+                    break;
+            }
+            if(len<=0){
+                i=END_NOT_FOUND;
+            }else{
+                s->state=0;
+                i-= s->header_size -1;
+                s->remaining_size = len;
+                if(!new_frame_start || pc->index+i<=0){
+                    s->remaining_size += i;
+                    goto get_next;
+                }
+            }
+        }
+    }
+
+    if(ff_combine_frame(pc, i, &buf, &buf_size)<0){
+        s->remaining_size -= FFMIN(s->remaining_size, buf_size);
+        *poutbuf = NULL;
+        *poutbuf_size = 0;
+        return buf_size;
+    }
+
+    *poutbuf = buf;
+    *poutbuf_size = buf_size;
+
+    /* update codec info */
+    if(s->codec_id)
+        avctx->codec_id = s->codec_id;
+
+    /* Due to backwards compatible HE-AAC the sample rate, channel count,
+       and total number of samples found in an AAC ADTS header are not
+       reliable. Bit rate is still accurate because the total frame duration in
+       seconds is still correct (as is the number of bits in the frame). */
+    if (avctx->codec_id != AV_CODEC_ID_AAC) {
+        avctx->sample_rate = s->sample_rate;
+
+        /* allow downmixing to stereo (or mono for AC-3) */
+        if(avctx->request_channels > 0 &&
+                avctx->request_channels < s->channels &&
+                (avctx->request_channels <= 2 ||
+                (avctx->request_channels == 1 &&
+                (avctx->codec_id == AV_CODEC_ID_AC3 ||
+                 avctx->codec_id == AV_CODEC_ID_EAC3)))) {
+            avctx->channels = avctx->request_channels;
+        } else {
+            avctx->channels = s->channels;
+            avctx->channel_layout = s->channel_layout;
+        }
+        s1->duration = s->samples;
+        avctx->audio_service_type = s->service_type;
+    }
+
+    avctx->bit_rate = s->bit_rate;
+
+    return i;
+}