1 files changed, 155 insertions, 0 deletions
diff --git a/contrib/ffmpeg/libavcodec/mpegaudio.h b/contrib/ffmpeg/libavcodec/mpegaudio.h
new file mode 100644
index 000000000..6d602a1dc
--- /dev/null
+++ b/contrib/ffmpeg/libavcodec/mpegaudio.h
@@ -0,0 +1,155 @@
+/*
+ * copyright (c) 2001 Fabrice Bellard
+ *
+ * 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
+ */
+
+/**
+ * @file mpegaudio.h
+ * mpeg audio declarations for both encoder and decoder.
+ */
+
+#ifndef FFMPEG_MPEGAUDIO_H
+#define FFMPEG_MPEGAUDIO_H
+
+#include "avcodec.h"
+#include "bitstream.h"
+#include "dsputil.h"
+
+/* max frame size, in samples */
+#define MPA_FRAME_SIZE 1152
+
+/* max compressed frame size */
+#define MPA_MAX_CODED_FRAME_SIZE 1792
+
+#define MPA_MAX_CHANNELS 2
+
+#define SBLIMIT 32 /* number of subbands */
+
+#define MPA_STEREO  0
+#define MPA_JSTEREO 1
+#define MPA_DUAL    2
+#define MPA_MONO    3
+
+/* header + layer + bitrate + freq + lsf/mpeg25 */
+#define SAME_HEADER_MASK \
+   (0xffe00000 | (3 << 17) | (0xf << 12) | (3 << 10) | (3 << 19))
+
+#define MP3_MASK 0xFFFE0CCF
+
+/* define USE_HIGHPRECISION to have a bit exact (but slower) mpeg
+   audio decoder */
+
+#ifdef USE_HIGHPRECISION
+#define FRAC_BITS   23   /* fractional bits for sb_samples and dct */
+#define WFRAC_BITS  16   /* fractional bits for window */
+#else
+#define FRAC_BITS   15   /* fractional bits for sb_samples and dct */
+#define WFRAC_BITS  14   /* fractional bits for window */
+#endif
+
+#define FRAC_ONE    (1 << FRAC_BITS)
+
+#define FIX(a)   ((int)((a) * FRAC_ONE))
+
+#if defined(USE_HIGHPRECISION) && defined(CONFIG_AUDIO_NONSHORT)
+typedef int32_t OUT_INT;
+#define OUT_MAX INT32_MAX
+#define OUT_MIN INT32_MIN
+#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 31)
+#else
+typedef int16_t OUT_INT;
+#define OUT_MAX INT16_MAX
+#define OUT_MIN INT16_MIN
+#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15)
+#endif
+
+#if FRAC_BITS <= 15
+typedef int16_t MPA_INT;
+#else
+typedef int32_t MPA_INT;
+#endif
+
+#define BACKSTEP_SIZE 512
+#define EXTRABYTES 24
+
+struct GranuleDef;
+
+typedef struct MPADecodeContext {
+    DECLARE_ALIGNED_8(uint8_t, last_buf[2*BACKSTEP_SIZE + EXTRABYTES]);
+    int last_buf_size;
+    int frame_size;
+    /* next header (used in free format parsing) */
+    uint32_t free_format_next_header;
+    int error_protection;
+    int layer;
+    int sample_rate;
+    int sample_rate_index; /* between 0 and 8 */
+    int bit_rate;
+    GetBitContext gb;
+    GetBitContext in_gb;
+    int nb_channels;
+    int mode;
+    int mode_ext;
+    int lsf;
+    DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512 * 2]);
+    int synth_buf_offset[MPA_MAX_CHANNELS];
+    DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT]);
+    int32_t mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
+#ifdef DEBUG
+    int frame_count;
+#endif
+    void (*compute_antialias)(struct MPADecodeContext *s, struct GranuleDef *g);
+    int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
+    int dither_state;
+    int error_resilience;
+    AVCodecContext* avctx;
+} MPADecodeContext;
+
+/* layer 3 huffman tables */
+typedef struct HuffTable {
+    int xsize;
+    const uint8_t *bits;
+    const uint16_t *codes;
+} HuffTable;
+
+int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf);
+int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate);
+void ff_mpa_synth_init(MPA_INT *window);
+void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
+                         MPA_INT *window, int *dither_state,
+                         OUT_INT *samples, int incr,
+                         int32_t sb_samples[SBLIMIT]);
+
+/* fast header check for resync */
+static inline int ff_mpa_check_header(uint32_t header){
+    /* header */
+    if ((header & 0xffe00000) != 0xffe00000)
+        return -1;
+    /* layer check */
+    if ((header & (3<<17)) == 0)
+        return -1;
+    /* bit rate */
+    if ((header & (0xf<<12)) == 0xf<<12)
+        return -1;
+    /* frequency */
+    if ((header & (3<<10)) == 3<<10)
+        return -1;
+    return 0;
+}
+
+#endif /* FFMPEG_MPEGAUDIO_H */