+/*
+ * copyright (c) 2001 Fabrice Bellard
+ *
+ * This file is part of Libav.
+ *
+ * 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.
+ *
+ * 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * mpeg audio declarations for both encoder and decoder.
+ */
+
+#ifndef AVCODEC_MPEGAUDIO_H
+#define AVCODEC_MPEGAUDIO_H
+
+#ifndef CONFIG_FLOAT
+# define CONFIG_FLOAT 0
+#endif
+
+#include "avcodec.h"
+#include "get_bits.h"
+#include "dsputil.h"
+#include "dct.h"
+
+#define CONFIG_AUDIO_NONSHORT 0
+
+/* max frame size, in samples */
+#define MPA_FRAME_SIZE 1152
/* max compressed frame size */
-#define MPA_MAX_CODED_FRAME_SIZE 1200
+#define MPA_MAX_CODED_FRAME_SIZE 1792
-#define MPA_FRAME_SIZE 1152
#define MPA_MAX_CHANNELS 2
-#define SAMPLES_BUF_SIZE 4096
#define SBLIMIT 32 /* number of subbands */
-#define DCT_BITS 14 /* number of bits for the DCT */
-#define MUL(a,b) (((a) * (b)) >> DCT_BITS)
-#define FIX(a) ((int)((a) * (1 << DCT_BITS)))
-
-typedef struct MpegAudioContext {
- PutBitContext pb;
- int nb_channels;
- int freq, bit_rate;
- int lsf; /* 1 if mpeg2 low bitrate selected */
- int bitrate_index; /* bit rate */
- int freq_index;
- int frame_size; /* frame size, in bits, without padding */
- long long nb_samples; /* total number of samples encoded */
- /* padding computation */
- int frame_frac, frame_frac_incr, do_padding;
- short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
- int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
- int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
- unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
- /* code to group 3 scale factors */
- unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
- int sblimit; /* number of used subbands */
- const unsigned char *alloc_table;
-} MpegAudioContext;
+#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
+
+#if CONFIG_MPEGAUDIO_HP
+#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 CONFIG_FLOAT
+typedef float OUT_INT;
+#define OUT_FMT AV_SAMPLE_FMT_FLT
+#elif CONFIG_MPEGAUDIO_HP && 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)
+#define OUT_FMT AV_SAMPLE_FMT_S32
+#else
+typedef int16_t OUT_INT;
+#define OUT_MAX INT16_MAX
+#define OUT_MIN INT16_MIN
+#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15)
+#define OUT_FMT AV_SAMPLE_FMT_S16
+#endif
+
+#if CONFIG_FLOAT
+# define INTFLOAT float
+typedef float MPA_INT;
+#elif FRAC_BITS <= 15
+# define INTFLOAT int
+typedef int16_t MPA_INT;
+#else
+# define INTFLOAT int
+typedef int32_t MPA_INT;
+#endif
+
+#define BACKSTEP_SIZE 512
+#define EXTRABYTES 24
+
+/* layer 3 "granule" */
+typedef struct GranuleDef {
+ uint8_t scfsi;
+ int part2_3_length;
+ int big_values;
+ int global_gain;
+ int scalefac_compress;
+ uint8_t block_type;
+ uint8_t switch_point;
+ int table_select[3];
+ int subblock_gain[3];
+ uint8_t scalefac_scale;
+ uint8_t count1table_select;
+ int region_size[3]; /* number of huffman codes in each region */
+ int preflag;
+ int short_start, long_end; /* long/short band indexes */
+ uint8_t scale_factors[40];
+ INTFLOAT sb_hybrid[SBLIMIT * 18]; /* 576 samples */
+} GranuleDef;
+
+#define MPA_DECODE_HEADER \
+ int frame_size; \
+ int error_protection; \
+ int layer; \
+ int sample_rate; \
+ int sample_rate_index; /* between 0 and 8 */ \
+ int bit_rate; \
+ int nb_channels; \
+ int mode; \
+ int mode_ext; \
+ int lsf;
+
+typedef struct MPADecodeHeader {
+ MPA_DECODE_HEADER
+} MPADecodeHeader;
+
+typedef struct MPADecodeContext {
+ MPA_DECODE_HEADER
+ uint8_t last_buf[2*BACKSTEP_SIZE + EXTRABYTES];
+ int last_buf_size;
+ /* next header (used in free format parsing) */
+ uint32_t free_format_next_header;
+ GetBitContext gb;
+ GetBitContext in_gb;
+ DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
+ int synth_buf_offset[MPA_MAX_CHANNELS];
+ DECLARE_ALIGNED(16, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
+ INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
+ GranuleDef granules[2][2]; /* Used in Layer 3 */
+#ifdef DEBUG
+ int frame_count;
+#endif
+ int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
+ int dither_state;
+ int error_recognition;
+ AVCodecContext* avctx;
+#if CONFIG_FLOAT
+ DCTContext dct;
+#endif
+ void (*apply_window_mp3)(MPA_INT *synth_buf, MPA_INT *window,
+ int *dither_state, OUT_INT *samples, int incr);
+} 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, int *channels, int *frame_size, int *bitrate);
+extern MPA_INT ff_mpa_synth_window[];
+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,
+ INTFLOAT sb_samples[SBLIMIT]);
+
+void ff_mpa_synth_init_float(MPA_INT *window);
+void ff_mpa_synth_filter_float(MPADecodeContext *s,
+ MPA_INT *synth_buf_ptr, int *synth_buf_offset,
+ MPA_INT *window, int *dither_state,
+ OUT_INT *samples, int incr,
+ INTFLOAT sb_samples[SBLIMIT]);
+
+void ff_mpegaudiodec_init_mmx(MPADecodeContext *s);
+void ff_mpegaudiodec_init_altivec(MPADecodeContext *s);
+
+/* 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 /* AVCODEC_MPEGAUDIO_H */
projects / ffmpeg / blobdiff