FFmpeg  4.0
flacdsp.h
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17  */
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 {
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);
38 
42 
43 #endif /* AVCODEC_FLACDSP_H */
static int shift(int a, int b)
Definition: sonic.c:82
const char * fmt
Definition: avisynth_c.h:769
channels
Definition: aptx.c:30
void(* lpc32)(int32_t *samples, const int coeffs[32], int order, int qlevel, int len)
Definition: flacdsp.h:31
uint8_t
void(* lpc16)(int32_t *samples, const int coeffs[32], int order, int qlevel, int len)
Definition: flacdsp.h:29
common internal API header
int32_t
void ff_flacdsp_init_x86(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps)
Definition: flacdsp_init.c:53
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
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
typedef void(RENAME(mix_any_func_type))
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=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;} 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_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);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_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps)
Definition: flacdsp.c:88
static double c[64]
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
unsigned bps
Definition: movenc.c:1456
static const int16_t coeffs[]
int len
void ff_flacdsp_init_arm(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps)
FILE * out
Definition: movenc.c:54
void(* decorrelate[4])(uint8_t **out, int32_t **in, int channels, int len, int shift)
Definition: flacdsp.h:27