FFmpeg  2.1.1
swresample_internal.h
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1 /*
2  * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
3  *
4  * This file is part of libswresample
5  *
6  * libswresample is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * libswresample is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with libswresample; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #ifndef SWR_INTERNAL_H
22 #define SWR_INTERNAL_H
23 
24 #include "swresample.h"
25 #include "libavutil/channel_layout.h"
26 #include "config.h"
27 
28 #define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */
29 
30 #define NS_TAPS 20
31 
32 #if ARCH_X86_64
33 typedef int64_t integer;
34 #else
35 typedef int integer;
36 #endif
37 
38 typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, integer index, integer len);
39 typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, integer index1, integer index2, integer len);
40 
41 typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len);
42 
43 typedef struct AudioData{
44  uint8_t *ch[SWR_CH_MAX]; ///< samples buffer per channel
45  uint8_t *data; ///< samples buffer
46  int ch_count; ///< number of channels
47  int bps; ///< bytes per sample
48  int count; ///< number of samples
49  int planar; ///< 1 if planar audio, 0 otherwise
50  enum AVSampleFormat fmt; ///< sample format
51 } AudioData;
52 
53 struct DitherContext {
55  int noise_pos;
56  float scale;
57  float noise_scale; ///< Noise scale
58  int ns_taps; ///< Noise shaping dither taps
59  float ns_scale; ///< Noise shaping dither scale
60  float ns_scale_1; ///< Noise shaping dither scale^-1
61  int ns_pos; ///< Noise shaping dither position
62  float ns_coeffs[NS_TAPS]; ///< Noise shaping filter coefficients
64  AudioData noise; ///< noise used for dithering
65  AudioData temp; ///< temporary storage when writing into the input buffer isnt possible
66  int output_sample_bits; ///< the number of used output bits, needed to scale dither correctly
67 };
68 
69 struct SwrContext {
70  const AVClass *av_class; ///< AVClass used for AVOption and av_log()
71  int log_level_offset; ///< logging level offset
72  void *log_ctx; ///< parent logging context
73  enum AVSampleFormat in_sample_fmt; ///< input sample format
74  enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
75  enum AVSampleFormat out_sample_fmt; ///< output sample format
76  int64_t in_ch_layout; ///< input channel layout
77  int64_t out_ch_layout; ///< output channel layout
78  int in_sample_rate; ///< input sample rate
79  int out_sample_rate; ///< output sample rate
80  int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
81  float slev; ///< surround mixing level
82  float clev; ///< center mixing level
83  float lfe_mix_level; ///< LFE mixing level
84  float rematrix_volume; ///< rematrixing volume coefficient
85  float rematrix_maxval; ///< maximum value for rematrixing output
86  enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */
87  const int *channel_map; ///< channel index (or -1 if muted channel) map
88  int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
90 
92 
93  int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
94  int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
95  int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
96  double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
97  enum SwrFilterType filter_type; /**< swr resampling filter type */
98  int kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
99  double precision; /**< soxr resampling precision (in bits) */
100  int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
101 
102  float min_compensation; ///< swr minimum below which no compensation will happen
103  float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen
104  float soft_compensation_duration; ///< swr duration over which soft compensation is applied
105  float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration
106  float async; ///< swr simple 1 parameter async, similar to ffmpegs -async
107  int64_t firstpts_in_samples; ///< swr first pts in samples
108 
109  int resample_first; ///< 1 if resampling must come first, 0 if rematrixing
110  int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
111  int rematrix_custom; ///< flag to indicate that a custom matrix has been defined
112 
113  AudioData in; ///< input audio data
114  AudioData postin; ///< post-input audio data: used for rematrix/resample
115  AudioData midbuf; ///< intermediate audio data (postin/preout)
116  AudioData preout; ///< pre-output audio data: used for rematrix/resample
117  AudioData out; ///< converted output audio data
118  AudioData in_buffer; ///< cached audio data (convert and resample purpose)
119  AudioData silence; ///< temporary with silence
120  AudioData drop_temp; ///< temporary used to discard output
121  int in_buffer_index; ///< cached buffer position
122  int in_buffer_count; ///< cached buffer length
123  int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise
124  int flushed; ///< 1 if data is to be flushed and no further input is expected
125  int64_t outpts; ///< output PTS
126  int64_t firstpts; ///< first PTS
127  int drop_output; ///< number of output samples to drop
128 
129  struct AudioConvert *in_convert; ///< input conversion context
130  struct AudioConvert *out_convert; ///< output conversion context
131  struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output)
132  struct ResampleContext *resample; ///< resampling context
133  struct Resampler const *resampler; ///< resampler virtual function table
134 
135  float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
140  int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients
141  uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients
144 
147 
149 
150  /* TODO: callbacks for ASM optimizations */
151 };
152 
153 typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
154  double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby);
155 typedef void (* resample_free_func)(struct ResampleContext **c);
156 typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
157 typedef int (* resample_flush_func)(struct SwrContext *c);
158 typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
159 typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
160 
161 struct Resampler {
162  resample_init_func init;
168 };
169 
170 extern struct Resampler const swri_resampler;
171 
173 int swri_resample_int16(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
174 int swri_resample_int32(struct ResampleContext *c, int32_t *dst, const int32_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
175 int swri_resample_float(struct ResampleContext *c, float *dst, const float *src, int *consumed, int src_size, int dst_size, int update_ctx);
176 int swri_resample_double(struct ResampleContext *c,double *dst, const double *src, int *consumed, int src_size, int dst_size, int update_ctx);
177 
178 void swri_noise_shaping_int16 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
179 void swri_noise_shaping_int32 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
180 void swri_noise_shaping_float (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
181 void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
182 
185 int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
186 void swri_rematrix_init_x86(struct SwrContext *s);
187 
188 void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt);
189 int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
190 
192  enum AVSampleFormat out_fmt,
193  enum AVSampleFormat in_fmt,
194  int channels);
196  enum AVSampleFormat out_fmt,
197  enum AVSampleFormat in_fmt,
198  int channels);
199 #endif
struct AudioConvert * in_convert
input conversion context
const AVClass * av_class
AVClass used for AVOption and av_log()
struct AudioConvert * full_convert
full conversion context (single conversion for input and output)
int(* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance)
const char * s
Definition: avisynth_c.h:668
AudioData temp
temporary storage when writing into the input buffer isnt possible
int out_sample_rate
output sample rate
enum SwrFilterType filter_type
swr resampling filter type
enum AVSampleFormat int_sample_fmt
internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
Audio buffer used for intermediate storage between conversion phases.
Definition: oss_audio.c:47
enum AVResampleDitherMethod method
Definition: dither.c:56
multiple_resample_func multiple_resample
libswresample public header
int count
number of samples
int ch_count
number of channels
void( mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, integer index1, integer index2, integer len)
int swri_resample_float(struct ResampleContext *c, float *dst, const float *src, int *consumed, int src_size, int dst_size, int update_ctx)
float soft_compensation_duration
swr duration over which soft compensation is applied
int rematrix_custom
flag to indicate that a custom matrix has been defined
SwrFilterType
Resampling Filter Types.
Definition: swresample.h:134
int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy)
Definition: rematrix.c:421
int in_buffer_index
cached buffer position
void( mix_1_1_func_type)(void *out, const void *in, void *coeffp, integer index, integer len)
AudioData in_buffer
cached audio data (convert and resample purpose)
int resample_in_constraint
1 if the input end was reach before the output end, 0 otherwise
struct ResampleContext * resample
resampling context
float ns_scale
Noise shaping dither scale.
float ns_coeffs[NS_TAPS]
Noise shaping filter coefficients.
float async
swr simple 1 parameter async, similar to ffmpegs -async
const int * channel_map
channel index (or -1 if muted channel) map
int(* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed)
av_cold void swri_audio_convert_init_arm(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels)
int log_level_offset
logging level offset
struct Resampler const * resampler
resampler virtual function table
float ns_errors[SWR_CH_MAX][2 *NS_TAPS]
enum AVSampleFormat format
Definition: resample.c:50
av_cold int swri_rematrix_init(SwrContext *s)
Definition: rematrix.c:343
uint8_t
float max_soft_compensation
swr maximum soft compensation in seconds over soft_compensation_duration
enum AVSampleFormat fmt
sample format
#define NS_TAPS
SwrDitherType
Definition: swresample.h:109
void * log_ctx
parent logging context
AudioData out
converted output audio data
int swri_realloc_audio(AudioData *a, int count)
Definition: swresample.c:445
int phase_shift
log2 of the number of entries in the resampling polyphase filterbank
int compensation_distance
Definition: resample.c:39
AudioData in
input audio data
uint8_t * native_simd_one
int swri_resample_double(struct ResampleContext *c, double *dst, const double *src, int *consumed, int src_size, int dst_size, int update_ctx)
float min_hard_compensation
swr minimum below which no silence inject / sample drop will happen
enum AVResampleFilterType filter_type
Definition: resample.c:43
struct Resampler const swri_resampler
Definition: resample.c:365
enum AVSampleFormat out_sample_fmt
output sample format
void(* resample_free_func)(struct ResampleContext **c)
int in_buffer_count
cached buffer length
SwrEngine
Resampling Engines.
Definition: swresample.h:127
AudioData postin
post-input audio data: used for rematrix/resample
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){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));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=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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);ff_audio_convert_init_arm(ac);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_dlog(ac->avr,"%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
float slev
surround mixing level
int output_sample_bits
the number of used output bits, needed to scale dither correctly
double cutoff
resampling cutoff frequency (swr: 6dB point; soxr: 0dB point).
float clev
center mixing level
void swri_audio_convert_init_x86(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels)
mix_2_1_func_type * mix_2_1_simd
resample_flush_func flush
int64_t firstpts
first PTS
AudioData preout
pre-output audio data: used for rematrix/resample
int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]
17.15 fixed point rematrixing coefficients
AudioData midbuf
intermediate audio data (postin/preout)
resample_free_func free
void swri_rematrix_init_x86(struct SwrContext *s)
int flags
miscellaneous flags such as SWR_FLAG_RESAMPLE
void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt)
Definition: dither.c:26
int drop_output
number of output samples to drop
int linear_interp
if 1 then the resampling FIR filter will be linearly interpolated
mix_1_1_func_type * mix_1_1_f
void swri_noise_shaping_int32(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
double precision
soxr resampling precision (in bits)
mix_1_1_func_type * mix_1_1_simd
AudioData noise
noise used for dithering
int32_t
int64_t out_ch_layout
output channel layout
enum AVMatrixEncoding matrix_encoding
matrixed stereo encoding
int in_sample_rate
input sample rate
int bps
bytes per sample
int(* resample_flush_func)(struct SwrContext *c)
int rematrix
flag to indicate if rematrixing is needed (basically if input and output layouts mismatch) ...
mix_any_func_type * mix_any_f
uint8_t * native_matrix
int64_t(* get_delay_func)(struct SwrContext *s, int64_t base)
set_compensation_func set_compensation
float ns_scale_1
Noise shaping dither scale^-1.
float noise_scale
Noise scale.
void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:49
int64_t outpts
output PTS
AVS_Value src
Definition: avisynth_c.h:523
typedef void(RENAME(mix_any_func_type))
int filter_size
length of each FIR filter in the resampling filterbank relative to the cutoff frequency ...
int swri_resample_int32(struct ResampleContext *c, int32_t *dst, const int32_t *src, int *consumed, int src_size, int dst_size, int update_ctx)
uint8_t * data[AVRESAMPLE_MAX_CHANNELS]
data plane pointers
Definition: audio_data.h:37
static unsigned int seed
Definition: videogen.c:78
float min_compensation
swr minimum below which no compensation will happen
int ns_pos
Noise shaping dither position.
Describe the class of an AVClass context structure.
Definition: log.h:50
enum SwrEngine engine
struct DitherContext dither
int index
Definition: gxfenc.c:89
void swri_noise_shaping_float(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
enum AVSampleFormat in_sample_fmt
input sample format
uint8_t * native_one
int flushed
1 if data is to be flushed and no further input is expected
int64_t in_ch_layout
input channel layout
uint8_t * native_simd_matrix
static double c[64]
#define SWR_CH_MAX
Maximum number of channels.
Definition: swresample.h:102
int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt)
Definition: dither.c:75
int cheby
soxr: if 1 then passband rolloff will be none (Chebyshev) &amp; irrational ratio approximation precision ...
get_delay_func get_delay
float lfe_mix_level
LFE mixing level.
void( mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len)
void swri_noise_shaping_int16(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
av_cold void swri_rematrix_free(SwrContext *s)
Definition: rematrix.c:414
int len
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){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));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=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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);ff_audio_convert_init_arm(ac);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_dlog(ac->avr,"%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-> out
float rematrix_maxval
maximum value for rematrixing output
struct AudioConvert * out_convert
output conversion context
float rematrix_volume
rematrixing volume coefficient
int kaiser_beta
swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) ...
float matrix[SWR_CH_MAX][SWR_CH_MAX]
floating point rematrixing coefficients
mix_2_1_func_type * mix_2_1_f
int64_t firstpts_in_samples
swr first pts in samples
int integer
void INT64 INT64 count
Definition: avisynth_c.h:594
int planar
1 if planar audio, 0 otherwise
AVMatrixEncoding
AudioData drop_temp
temporary used to discard output
uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]
Lists of input channels per output channel that have non zero rematrixing coefficients.
uint8_t * ch[SWR_CH_MAX]
samples buffer per channel
int swri_resample_int16(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx)
int used_ch_count
number of used input channels (mapped channel count if channel_map, otherwise in.ch_count) ...
resample_init_func init
AudioData silence
temporary with silence
int resample_first
1 if resampling must come first, 0 if rematrixing
int ns_taps
Noise shaping dither taps.