Re-tagging 0.13.0 release to fix a couple of bugs with the tarball.

git-svn-id: https://svn.musicpd.org/mpd/tags/release-0.13.0@6325 09075e82-0dd4-0310-85a5-a0d7c8717e4f

1 files changed, 152 insertions, 0 deletions

diff --git a/trunk/src/mp4ff/mp4sample.c b/trunk/src/mp4ff/mp4sample.c
new file mode 100644
index 000000000..5688a3a8f
--- /dev/null
+++ b/trunk/src/mp4ff/mp4sample.c
@@ -0,0 +1,152 @@
+/*
+** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
+** Copyright (C) 2003-2004 M. Bakker, Ahead Software AG, http://www.nero.com
+**
+** This program is free software; you can redistribute it and/or modify
+** it under the terms of the GNU General Public License as published by
+** the Free Software Foundation; either version 2 of the License, or
+** (at your option) any later version.
+**
+** This program 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 General Public License for more details.
+**
+** You should have received a copy of the GNU General Public License
+** along with this program; if not, write to the Free Software
+** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+**
+** Any non-GPL usage of this software or parts of this software is strictly
+** forbidden.
+**
+** Commercial non-GPL licensing of this software is possible.
+** For more info contact Ahead Software through Mpeg4AAClicense@nero.com.
+**
+** $Id: mp4sample.c,v 1.15 2004/01/11 15:52:19 menno Exp $
+**/
+
+#include <stdlib.h>
+#include "mp4ffint.h"
+
+
+static int32_t mp4ff_chunk_of_sample(const mp4ff_t *f, const int32_t track, const int32_t sample,
+                                     int32_t *chunk_sample, int32_t *chunk)
+{
+    int32_t total_entries = 0;
+    int32_t chunk2entry;
+    int32_t chunk1, chunk2, chunk1samples, range_samples, total = 0;
+
+    if (f->track[track] == NULL)
+    {
+        return -1;
+    }
+
+    total_entries = f->track[track]->stsc_entry_count;
+
+    chunk1 = 1;
+    chunk1samples = 0;
+    chunk2entry = 0;
+
+    do
+    {
+        chunk2 = f->track[track]->stsc_first_chunk[chunk2entry];
+        *chunk = chunk2 - chunk1;
+        range_samples = *chunk * chunk1samples;
+
+        if (sample < total + range_samples) break;
+
+        chunk1samples = f->track[track]->stsc_samples_per_chunk[chunk2entry];
+        chunk1 = chunk2;
+
+        if(chunk2entry < total_entries)
+        {
+            chunk2entry++;
+            total += range_samples;
+        }
+    } while (chunk2entry < total_entries);
+
+    if (chunk1samples)
+        *chunk = (sample - total) / chunk1samples + chunk1;
+    else
+        *chunk = 1;
+
+    *chunk_sample = total + (*chunk - chunk1) * chunk1samples;
+
+    return 0;
+}
+
+static int32_t mp4ff_chunk_to_offset(const mp4ff_t *f, const int32_t track, const int32_t chunk)
+{
+    const mp4ff_track_t * p_track = f->track[track];
+
+    if (p_track->stco_entry_count && (chunk > p_track->stco_entry_count))
+    {
+        return p_track->stco_chunk_offset[p_track->stco_entry_count - 1];
+    } else if (p_track->stco_entry_count) {
+        return p_track->stco_chunk_offset[chunk - 1];
+    } else {
+        return 8;
+    }
+
+    return 0;
+}
+
+static int32_t mp4ff_sample_range_size(const mp4ff_t *f, const int32_t track,
+                                       const int32_t chunk_sample, const int32_t sample)
+{
+    int32_t i, total;
+    const mp4ff_track_t * p_track = f->track[track];
+
+    if (p_track->stsz_sample_size)
+    {
+        return (sample - chunk_sample) * p_track->stsz_sample_size;
+    }
+    else
+    {
+        if (sample>=p_track->stsz_sample_count) return 0;//error
+
+        for(i = chunk_sample, total = 0; i < sample; i++)
+        {
+            total += p_track->stsz_table[i];
+        }
+    }
+
+    return total;
+}
+
+static int32_t mp4ff_sample_to_offset(const mp4ff_t *f, const int32_t track, const int32_t sample)
+{
+    int32_t chunk, chunk_sample, chunk_offset1, chunk_offset2;
+
+    mp4ff_chunk_of_sample(f, track, sample, &chunk_sample, &chunk);
+
+    chunk_offset1 = mp4ff_chunk_to_offset(f, track, chunk);
+    chunk_offset2 = chunk_offset1 + mp4ff_sample_range_size(f, track, chunk_sample, sample);
+
+    return chunk_offset2;
+}
+
+int32_t mp4ff_audio_frame_size(const mp4ff_t *f, const int32_t track, const int32_t sample)
+{
+    int32_t bytes;
+    const mp4ff_track_t * p_track = f->track[track];
+
+    if (p_track->stsz_sample_size)
+    {
+        bytes = p_track->stsz_sample_size;
+    } else {
+        bytes = p_track->stsz_table[sample];
+    }
+
+    return bytes;
+}
+
+int32_t mp4ff_set_sample_position(mp4ff_t *f, const int32_t track, const int32_t sample)
+{
+    int32_t offset;
+
+    offset = mp4ff_sample_to_offset(f, track, sample);
+    mp4ff_set_position(f, offset);
+
+    return 0;
+}