Update internal FFmpeg copy.

1 files changed, 100 insertions, 0 deletions

diff --git a/contrib/ffmpeg/libavcodec/aac_parser.c b/contrib/ffmpeg/libavcodec/aac_parser.c
new file mode 100644
index 000000000..ac806931e
--- /dev/null
+++ b/contrib/ffmpeg/libavcodec/aac_parser.c
@@ -0,0 +1,100 @@
+/*
+ * Audio and Video frame extraction
+ * 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 "parser.h"
+#include "aac_ac3_parser.h"
+#include "bitstream.h"
+
+
+#define AAC_HEADER_SIZE 7
+
+
+static const int aac_sample_rates[16] = {
+    96000, 88200, 64000, 48000, 44100, 32000,
+    24000, 22050, 16000, 12000, 11025, 8000, 7350
+};
+
+static const int aac_channels[8] = {
+    0, 1, 2, 3, 4, 5, 6, 8
+};
+
+
+static int aac_sync(const uint8_t *buf, int *channels, int *sample_rate,
+                    int *bit_rate, int *samples)
+{
+    GetBitContext bits;
+    int size, rdb, ch, sr;
+
+    init_get_bits(&bits, buf, AAC_HEADER_SIZE * 8);
+
+    if(get_bits(&bits, 12) != 0xfff)
+        return 0;
+
+    skip_bits1(&bits);          /* id */
+    skip_bits(&bits, 2);        /* layer */
+    skip_bits1(&bits);          /* protection_absent */
+    skip_bits(&bits, 2);        /* profile_objecttype */
+    sr = get_bits(&bits, 4);    /* sample_frequency_index */
+    if(!aac_sample_rates[sr])
+        return 0;
+    skip_bits1(&bits);          /* private_bit */
+    ch = get_bits(&bits, 3);    /* channel_configuration */
+    if(!aac_channels[ch])
+        return 0;
+    skip_bits1(&bits);          /* original/copy */
+    skip_bits1(&bits);          /* home */
+
+    /* adts_variable_header */
+    skip_bits1(&bits);          /* copyright_identification_bit */
+    skip_bits1(&bits);          /* copyright_identification_start */
+    size = get_bits(&bits, 13); /* aac_frame_length */
+    if(size < AAC_HEADER_SIZE)
+        return 0;
+
+    skip_bits(&bits, 11);       /* adts_buffer_fullness */
+    rdb = get_bits(&bits, 2);   /* number_of_raw_data_blocks_in_frame */
+
+    *channels = aac_channels[ch];
+    *sample_rate = aac_sample_rates[sr];
+    *samples = (rdb + 1) * 1024;
+    *bit_rate = size * 8 * *sample_rate / *samples;
+
+    return size;
+}
+
+static int aac_parse_init(AVCodecParserContext *s1)
+{
+    AACAC3ParseContext *s = s1->priv_data;
+    s->inbuf_ptr = s->inbuf;
+    s->header_size = AAC_HEADER_SIZE;
+    s->sync = aac_sync;
+    return 0;
+}
+
+
+AVCodecParser aac_parser = {
+    { CODEC_ID_AAC },
+    sizeof(AACAC3ParseContext),
+    aac_parse_init,
+    ff_aac_ac3_parse,
+    NULL,
+};