FFmpeg
 All Data Structures Namespaces Files Functions Variables Typedefs Enumerations Enumerator Macros Groups Pages
swresample_internal.h
Go to the documentation of this file.
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 SWRESAMPLE_SWRESAMPLE_INTERNAL_H
22 #define SWRESAMPLE_SWRESAMPLE_INTERNAL_H
23 
24 #include "swresample.h"
26 #include "config.h"
27 
28 #define SWR_CH_MAX 64
29 
30 #define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */
31 
32 #define NS_TAPS 20
33 
34 #if ARCH_X86_64
35 typedef int64_t integer;
36 #else
37 typedef int integer;
38 #endif
39 
40 typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, integer index, integer len);
41 typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, integer index1, integer index2, integer len);
42 
43 typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len);
44 
45 typedef struct AudioData{
46  uint8_t *ch[SWR_CH_MAX]; ///< samples buffer per channel
47  uint8_t *data; ///< samples buffer
48  int ch_count; ///< number of channels
49  int bps; ///< bytes per sample
50  int count; ///< number of samples
51  int planar; ///< 1 if planar audio, 0 otherwise
52  enum AVSampleFormat fmt; ///< sample format
53 } AudioData;
54 
55 struct DitherContext {
56  int method;
57  int noise_pos;
58  float scale;
59  float noise_scale; ///< Noise scale
60  int ns_taps; ///< Noise shaping dither taps
61  float ns_scale; ///< Noise shaping dither scale
62  float ns_scale_1; ///< Noise shaping dither scale^-1
63  int ns_pos; ///< Noise shaping dither position
64  float ns_coeffs[NS_TAPS]; ///< Noise shaping filter coefficients
66  AudioData noise; ///< noise used for dithering
67  AudioData temp; ///< temporary storage when writing into the input buffer isn't possible
68  int output_sample_bits; ///< the number of used output bits, needed to scale dither correctly
69 };
70 
71 typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
72  double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta, double precision, int cheby);
73 typedef void (* resample_free_func)(struct ResampleContext **c);
74 typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
75 typedef int (* resample_flush_func)(struct SwrContext *c);
76 typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
77 typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
78 typedef int (* invert_initial_buffer_func)(struct ResampleContext *c, AudioData *dst, const AudioData *src, int src_size, int *dst_idx, int *dst_count);
79 typedef int64_t (* get_out_samples_func)(struct SwrContext *s, int in_samples);
80 
81 struct Resampler {
82  resample_init_func init;
90 };
91 
92 extern struct Resampler const swri_resampler;
93 extern struct Resampler const swri_soxr_resampler;
94 
95 struct SwrContext {
96  const AVClass *av_class; ///< AVClass used for AVOption and av_log()
97  int log_level_offset; ///< logging level offset
98  void *log_ctx; ///< parent logging context
99  enum AVSampleFormat in_sample_fmt; ///< input sample format
100  enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
101  enum AVSampleFormat out_sample_fmt; ///< output sample format
102  int64_t in_ch_layout; ///< input channel layout
103  int64_t out_ch_layout; ///< output channel layout
104  int in_sample_rate; ///< input sample rate
105  int out_sample_rate; ///< output sample rate
106  int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
107  float slev; ///< surround mixing level
108  float clev; ///< center mixing level
109  float lfe_mix_level; ///< LFE mixing level
110  float rematrix_volume; ///< rematrixing volume coefficient
111  float rematrix_maxval; ///< maximum value for rematrixing output
112  int matrix_encoding; /**< matrixed stereo encoding */
113  const int *channel_map; ///< channel index (or -1 if muted channel) map
114  int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
115  int engine;
116 
117  int user_in_ch_count; ///< User set input channel count
118  int user_out_ch_count; ///< User set output channel count
119  int user_used_ch_count; ///< User set used channel count
120  int64_t user_in_ch_layout; ///< User set input channel layout
121  int64_t user_out_ch_layout; ///< User set output channel layout
122  enum AVSampleFormat user_int_sample_fmt; ///< User set internal sample format
123 
125 
126  int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
127  int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
128  int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
129  double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
130  int filter_type; /**< swr resampling filter type */
131  double kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
132  double precision; /**< soxr resampling precision (in bits) */
133  int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
134 
135  float min_compensation; ///< swr minimum below which no compensation will happen
136  float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen
137  float soft_compensation_duration; ///< swr duration over which soft compensation is applied
138  float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration
139  float async; ///< swr simple 1 parameter async, similar to ffmpegs -async
140  int64_t firstpts_in_samples; ///< swr first pts in samples
141 
142  int resample_first; ///< 1 if resampling must come first, 0 if rematrixing
143  int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
144  int rematrix_custom; ///< flag to indicate that a custom matrix has been defined
145 
146  AudioData in; ///< input audio data
147  AudioData postin; ///< post-input audio data: used for rematrix/resample
148  AudioData midbuf; ///< intermediate audio data (postin/preout)
149  AudioData preout; ///< pre-output audio data: used for rematrix/resample
150  AudioData out; ///< converted output audio data
151  AudioData in_buffer; ///< cached audio data (convert and resample purpose)
152  AudioData silence; ///< temporary with silence
153  AudioData drop_temp; ///< temporary used to discard output
154  int in_buffer_index; ///< cached buffer position
155  int in_buffer_count; ///< cached buffer length
156  int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise
157  int flushed; ///< 1 if data is to be flushed and no further input is expected
158  int64_t outpts; ///< output PTS
159  int64_t firstpts; ///< first PTS
160  int drop_output; ///< number of output samples to drop
161  double delayed_samples_fixup; ///< soxr 0.1.1: needed to fixup delayed_samples after flush has been called.
162 
163  struct AudioConvert *in_convert; ///< input conversion context
164  struct AudioConvert *out_convert; ///< output conversion context
165  struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output)
166  struct ResampleContext *resample; ///< resampling context
167  struct Resampler const *resampler; ///< resampler virtual function table
168 
169  float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
174  int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients
175  uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients
178 
181 
183 
184  /* TODO: callbacks for ASM optimizations */
185 };
186 
189 
190 void swri_noise_shaping_int16 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
191 void swri_noise_shaping_int32 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
192 void swri_noise_shaping_float (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
193 void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
194 
198 int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
199 int swri_rematrix_init_x86(struct SwrContext *s);
200 
202 int swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt);
204 int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
205 
207  enum AVSampleFormat out_fmt,
208  enum AVSampleFormat in_fmt,
209  int channels);
211  enum AVSampleFormat out_fmt,
212  enum AVSampleFormat in_fmt,
213  int channels);
215  enum AVSampleFormat out_fmt,
216  enum AVSampleFormat in_fmt,
217  int channels);
218 
219 #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:631
AudioData temp
temporary storage when writing into the input buffer isn't possible
int out_sample_rate
output sample rate
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: audio_data.h:37
multiple_resample_func multiple_resample
static int linear(InterplayACMContext *s, unsigned ind, unsigned col)
Definition: interplayacm.c:115
struct Resampler const swri_resampler
Definition: resample.c:556
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)
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:170
double delayed_samples_fixup
soxr 0.1.1: needed to fixup delayed_samples after flush has been called.
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)
int log_level_offset
logging level offset
struct Resampler const * resampler
resampler virtual function table
float ns_errors[SWR_CH_MAX][2 *NS_TAPS]
uint8_t
float max_soft_compensation
swr maximum soft compensation in seconds over soft_compensation_duration
int user_out_ch_count
User set output channel count.
enum AVSampleFormat fmt
sample format
#define NS_TAPS
void * log_ctx
parent logging context
void swri_rematrix_free(SwrContext *s)
Definition: rematrix.c:456
void swri_audio_convert_init_arm(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels)
AudioData out
converted output audio data
int phase_shift
log2 of the number of entries in the resampling polyphase filterbank
int compensation_distance
Definition: resample.h:38
AudioData in
input audio data
uint8_t * native_simd_one
invert_initial_buffer_func invert_initial_buffer
float min_hard_compensation
swr minimum below which no silence inject / sample drop will happen
enum AVResampleFilterType filter_type
Definition: resample.h:42
enum AVSampleFormat out_sample_fmt
output sample format
void(* resample_free_func)(struct ResampleContext **c)
int in_buffer_count
cached buffer length
libswresample public header
AudioData postin
post-input audio data: used for rematrix/resample
int matrix_encoding
matrixed stereo encoding
float slev
surround mixing level
int output_sample_bits
the number of used output bits, needed to scale dither correctly
int64_t user_in_ch_layout
User set input channel layout.
av_warn_unused_result int swri_realloc_audio(AudioData *a, int count)
Definition: swresample.c:381
av_warn_unused_result int swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt)
Definition: dither.c:26
The libswresample context.
int swri_rematrix_init_x86(struct SwrContext *s)
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
GLsizei count
Definition: opengl_enc.c:109
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
audio channel layout utility functions
int flags
miscellaneous flags such as SWR_FLAG_RESAMPLE
int filter_type
swr resampling filter type
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)
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
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
int in_sample_rate
input sample rate
int bps
bytes per sample
av_warn_unused_result int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt)
Definition: dither.c:79
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
#define src
Definition: vp9dsp.c:530
int64_t(* get_out_samples_func)(struct SwrContext *s, int in_samples)
float ns_scale_1
Noise shaping dither scale^-1.
float noise_scale
Noise scale.
FILE * out
Definition: movenc-test.c:54
void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
int user_in_ch_count
User set input channel count.
enum AVSampleFormat user_int_sample_fmt
User set internal sample format.
int64_t outpts
output PTS
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int user_used_ch_count
User set used channel count.
int filter_size
length of each FIR filter in the resampling filterbank relative to the cutoff frequency ...
#define SWR_CH_MAX
uint8_t * data[AVRESAMPLE_MAX_CHANNELS]
data plane pointers
Definition: audio_data.h:39
double kaiser_beta
swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) ...
static const char * format
Definition: movenc-test.c:47
static unsigned int seed
Definition: videogen.c:78
float min_compensation
swr minimum below which no compensation will happen
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
int ns_pos
Noise shaping dither position.
Describe the class of an AVClass context structure.
Definition: log.h:67
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)
get_out_samples_func get_out_samples
av_warn_unused_result int swri_rematrix_init(SwrContext *s)
Definition: rematrix.c:360
enum AVSampleFormat in_sample_fmt
input sample format
struct Resampler const swri_soxr_resampler
int(* invert_initial_buffer_func)(struct ResampleContext *c, AudioData *dst, const AudioData *src, int src_size, int *dst_idx, int *dst_count)
int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy)
Definition: rematrix.c:463
uint8_t * native_one
#define av_warn_unused_result
Definition: attributes.h:56
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]
int cheby
soxr: if 1 then passband rolloff will be none (Chebyshev) & 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)
int len
float rematrix_maxval
maximum value for rematrixing output
struct AudioConvert * out_convert
output conversion context
float rematrix_volume
rematrixing volume coefficient
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
int planar
1 if planar audio, 0 otherwise
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 used_ch_count
number of used input channels (mapped channel count if channel_map, otherwise in.ch_count) ...
resample_init_func init
void swri_audio_convert_init_aarch64(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels)
int64_t user_out_ch_layout
User set output channel layout.
AudioData silence
temporary with silence
int resample_first
1 if resampling must come first, 0 if rematrixing
int ns_taps
Noise shaping dither taps.