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flacdsp.h
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18 
19 #ifndef AVCODEC_FLACDSP_H
20 #define AVCODEC_FLACDSP_H
21 
22 #include <stdint.h>
23 #include "libavutil/internal.h"
24 #include "libavutil/samplefmt.h"
25 
26 typedef struct FLACDSPContext {
27  void (*decorrelate[4])(uint8_t **out, int32_t **in, int channels,
28  int len, int shift);
29  void (*lpc16)(int32_t *samples, const int coeffs[32], int order,
30  int qlevel, int len);
31  void (*lpc32)(int32_t *samples, const int coeffs[32], int order,
32  int qlevel, int len);
33  void (*lpc16_encode)(int32_t *res, const int32_t *smp, int len, int order,
34  const int32_t coefs[32], int shift);
35  void (*lpc32_encode)(int32_t *res, const int32_t *smp, int len, int order,
36  const int32_t coefs[32], int shift);
37 } FLACDSPContext;
38 
39 void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps);
40 void ff_flacdsp_init_arm(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps);
41 void ff_flacdsp_init_x86(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps);
42 
43 #endif /* AVCODEC_FLACDSP_H */
shift
static int shift(int a, int b)
Definition: sonic.c:82
fmt
const char * fmt
Definition: avisynth_c.h:769
FLACDSPContext::lpc32_encode
void(* lpc32_encode)(int32_t *res, const int32_t *smp, int len, int order, const int32_t coefs[32], int shift)
Definition: flacdsp.h:35
FLACDSPContext::lpc32
void(* lpc32)(int32_t *samples, const int coeffs[32], int order, int qlevel, int len)
Definition: flacdsp.h:31
uint8_t
uint8_t
Definition: audio_convert.c:194
FLACDSPContext::lpc16_encode
void(* lpc16_encode)(int32_t *res, const int32_t *smp, int len, int order, const int32_t coefs[32], int shift)
Definition: flacdsp.h:33
internal.h
common internal API header
void
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
int32_t
int32_t
Definition: audio_convert.c:194
ff_flacdsp_init_x86
void ff_flacdsp_init_x86(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps)
Definition: flacdsp_init.c:53
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
in
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
Definition: audio_convert.c:194
ff_flacdsp_init
void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps)
Definition: flacdsp.c:88
FLACDSPContext::lpc16
void(* lpc16)(int32_t *samples, const int coeffs[32], int order, int qlevel, int len)
Definition: flacdsp.h:29
c
static double c[64]
Definition: vsrc_mptestsrc.c:87
bps
unsigned bps
Definition: movenc.c:1410
coeffs
static const int16_t coeffs[]
Definition: simple_idct_mmi.c:40
len
int len
Definition: vorbis_enc_data.h:452
ff_flacdsp_init_arm
void ff_flacdsp_init_arm(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps)
Definition: flacdsp_init_arm.c:27
out
FILE * out
Definition: movenc.c:54
FLACDSPContext::decorrelate
void(* decorrelate[4])(uint8_t **out, int32_t **in, int channels, int len, int shift)
Definition: flacdsp.h:27
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