mpc.h
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1 /*
2  * Musepack decoder
3  * Copyright (c) 2006 Konstantin Shishkov
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * Musepack decoder
25  * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
26  * divided into 32 subbands.
27  */
28 
29 #ifndef AVCODEC_MPC_H
30 #define AVCODEC_MPC_H
31 
32 #include "libavutil/lfg.h"
33 #include "avcodec.h"
34 #include "get_bits.h"
35 #include "dsputil.h"
36 #include "mpegaudio.h"
37 #include "mpegaudiodsp.h"
38 
39 #define BANDS 32
40 #define SAMPLES_PER_BAND 36
41 #define MPC_FRAME_SIZE (BANDS * SAMPLES_PER_BAND)
42 
43 /** Subband structure - hold all variables for each subband */
44 typedef struct Band {
45  int msf; ///< mid-stereo flag
46  int res[2];
47  int scfi[2];
48  int scf_idx[2][3];
49  int Q[2];
50 }Band;
51 
52 typedef struct MPCContext {
56  int IS, MSS, gapless;
58  int maxbands, last_max_band;
60  int oldDSCF[2][BANDS];
61  Band bands[BANDS];
62  int Q[2][MPC_FRAME_SIZE];
63  int cur_frame, frames;
65  int buf_size;
68  /* for synthesis */
69  DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2];
70  int synth_buf_offset[MPA_MAX_CHANNELS];
71  DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
72 } MPCContext;
73 
74 void ff_mpc_init(void);
75 void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, int16_t **out, int channels);
76 
77 #endif /* AVCODEC_MPC_H */
Definition: lfg.h:25
int Q[2]
Definition: mpc.h:49
#define SBLIMIT
Definition: mpegaudio.h:43
MPADSPContext mpadsp
Definition: mpc.h:54
int msf
mid-stereo flag
Definition: mpc.h:45
int res[2]
Definition: mpc.h:46
#define DECLARE_ALIGNED(n, t, v)
Definition: mem.h:59
AVLFG rnd
Definition: mpc.h:66
struct Band Band
Subband structure - hold all variables for each subband.
int buf_size
Definition: mpc.h:65
int32_t MPA_INT
Definition: mpegaudio.h:71
int lastframelen
Definition: mpc.h:57
uint8_t
int scf_idx[2][3]
Definition: mpc.h:48
bitstream reader API header.
void ff_mpc_init(void)
int last_bits_used
Definition: mpc.h:59
int maxbands
Definition: mpc.h:58
external API header
#define MPA_MAX_CHANNELS
Definition: mpegaudio.h:41
uint8_t * bits
Definition: mpc.h:64
#define MPC_FRAME_SIZE
Definition: mpc.h:41
void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, int16_t **out, int channels)
#define BANDS
Definition: mpc.h:39
int32_t
int scfi[2]
Definition: mpc.h:47
DSPContext dsp
Definition: mpc.h:53
static double c[64]
mpeg audio declarations for both encoder and decoder.
int MSS
Definition: mpc.h:56
DSP utils.
int frames
Definition: mpc.h:63
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);return NULL;}return ac;}in_planar=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> out
struct MPCContext MPCContext
GetBitContext gb
Definition: mpc.h:55
Subband structure - hold all variables for each subband.
Definition: mpc.h:44
Definition: mpc.h:52
DSPContext.
Definition: dsputil.h:127
int frames_to_skip
Definition: mpc.h:67