48 #define LICENSE_PREFIX "libswresample license: "
53 if(!
s ||
s->in_convert)
55 s->channel_map = channel_map;
66 s->log_level_offset= log_offset;
105 if (
a->ch_count == 1)
111 memset(
a, 0,
sizeof(*
a));
115 s->in_buffer_index= 0;
116 s->in_buffer_count= 0;
117 s->resample_in_constraint= 0;
118 memset(
s->in.ch, 0,
sizeof(
s->in.ch));
119 memset(
s->out.ch, 0,
sizeof(
s->out.ch));
133 s->delayed_samples_fixup = 0;
142 s->resampler->free(&
s->resample);
154 char l1[1024], l2[1024];
171 if(
s->out_sample_rate <= 0){
177 s->used_ch_count =
s->user_used_ch_count;
180 s->out_ch_layout =
s->user_out_ch_layout;
182 s->int_sample_fmt=
s->user_int_sample_fmt;
184 s->dither.method =
s->user_dither_method;
192 av_log(
s,
AV_LOG_WARNING,
"Output channel layout 0x%"PRIx64
" is invalid or unsupported.\n",
s->out_ch_layout);
193 s->out_ch_layout = 0;
206 if(!
s->used_ch_count)
207 s->used_ch_count=
s->in.ch_count;
210 av_log(
s,
AV_LOG_WARNING,
"Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
216 if(!
s->out_ch_layout)
219 s->rematrix=
s->out_ch_layout !=
s->in_ch_layout ||
s->rematrix_volume!=1.0 ||
228 &&
s->out_sample_rate==
s->in_sample_rate
234 &&
s->out_sample_rate ==
s->in_sample_rate
259 if (!
s->async &&
s->min_compensation >= FLT_MAX/2)
262 s->outpts =
s->firstpts_in_samples *
s->out_sample_rate;
267 if (
s->min_compensation >= FLT_MAX/2)
268 s->min_compensation = 0.001;
269 if (
s->async > 1.0001) {
270 s->max_soft_compensation =
s->async / (double)
s->in_sample_rate;
275 s->resample =
s->resampler->init(
s->resample,
s->out_sample_rate,
s->in_sample_rate,
s->filter_size,
s->phase_shift,
s->linear_interp,
s->cutoff,
s->int_sample_fmt,
s->filter_type,
s->kaiser_beta,
s->precision,
s->cheby,
s->exact_rational);
281 s->resampler->free(&
s->resample);
295 if(!
s->used_ch_count)
296 s->used_ch_count=
s->in.ch_count;
300 if(!
s->
in.ch_count){
310 av_log(
s,
AV_LOG_ERROR,
"Output channel layout %s mismatches specified channel count %d\n", l2,
s->out.ch_count);
315 av_log(
s,
AV_LOG_ERROR,
"Input channel layout %s mismatches specified channel count %d\n", l1,
s->used_ch_count);
320 if ((!
s->out_ch_layout || !
s->in_ch_layout) &&
s->used_ch_count !=
s->out.ch_count && !
s->rematrix_custom) {
322 "but there is not enough information to do it\n", l1, l2);
333 s->drop_temp=
s->out;
338 if(!
s->resample && !
s->rematrix && !
s->channel_map && !
s->dither.method){
347 s->int_sample_fmt,
s->out.ch_count,
NULL, 0);
349 if (!
s->in_convert || !
s->out_convert) {
360 s->midbuf.ch_count=
s->used_ch_count;
362 s->in_buffer.ch_count=
s->used_ch_count;
364 if(!
s->resample_first){
365 s->midbuf.ch_count=
s->out.ch_count;
367 s->in_buffer.ch_count =
s->out.ch_count;
379 s->dither.noise =
s->preout;
380 s->dither.temp =
s->preout;
382 s->dither.noise.bps = 4;
384 s->dither.noise_scale = 1;
387 if(
s->rematrix ||
s->dither.method) {
404 if(count < 0 || count > INT_MAX/2/
a->bps/
a->ch_count)
407 if(
a->count >= count)
421 for(
i=0;
i<
a->ch_count;
i++){
422 a->ch[
i]=
a->data +
i*(
a->planar ? countb :
a->bps);
423 if(
a->count &&
a->planar) memcpy(
a->ch[
i], old.
ch[
i],
a->count*
a->bps);
425 if(
a->count && !
a->planar) memcpy(
a->ch[0], old.
ch[0],
a->count*
a->ch_count*
a->bps);
439 for(ch=0; ch<
out->ch_count; ch++)
440 memcpy(
out->ch[ch],
in->ch[ch], count*
out->bps);
442 memcpy(
out->ch[0],
in->ch[0], count*
out->ch_count*
out->bps);
448 memset(
out->ch, 0,
sizeof(
out->ch));
449 }
else if(
out->planar){
450 for(
i=0;
i<
out->ch_count;
i++)
451 out->ch[
i]= in_arg[
i];
453 for(
i=0;
i<
out->ch_count;
i++)
461 for(
i=0;
i<
out->ch_count;
i++)
462 in_arg[
i]=
out->ch[
i];
464 in_arg[0]=
out->ch[0];
475 for(ch=0; ch<
out->ch_count; ch++)
476 out->ch[ch]=
in->ch[ch] + count*
out->bps;
478 for(ch=
out->ch_count-1; ch>=0; ch--)
479 out->ch[ch]=
in->ch[0] + (ch + count*
out->ch_count) *
out->bps;
488 const AudioData * in_param,
int in_count){
501 border =
s->resampler->invert_initial_buffer(
s->resample, &
s->in_buffer,
502 &
in, in_count, &
s->in_buffer_index, &
s->in_buffer_count);
503 if (border == INT_MAX) {
505 }
else if (border < 0) {
510 s->resample_in_constraint = 0;
514 int ret,
size, consumed;
515 if(!
s->resample_in_constraint &&
s->in_buffer_count){
517 ret=
s->resampler->multiple_resample(
s->resample, &
out, out_count, &
tmp,
s->in_buffer_count, &consumed);
521 s->in_buffer_count -= consumed;
522 s->in_buffer_index += consumed;
526 if(
s->in_buffer_count <= border){
528 in_count +=
s->in_buffer_count;
529 s->in_buffer_count=0;
530 s->in_buffer_index=0;
535 if((
s->flushed || in_count > padless) && !
s->in_buffer_count){
536 s->in_buffer_index=0;
537 ret=
s->resampler->multiple_resample(
s->resample, &
out, out_count, &
in,
FFMAX(in_count-padless, 0), &consumed);
541 in_count -= consumed;
546 size=
s->in_buffer_index +
s->in_buffer_count + in_count;
547 if(
size >
s->in_buffer.count
550 copy(&
s->in_buffer, &
tmp,
s->in_buffer_count);
551 s->in_buffer_index=0;
558 if(
s->in_buffer_count &&
s->in_buffer_count+2 < count && out_count) count=
s->in_buffer_count+2;
560 buf_set(&
tmp, &
s->in_buffer,
s->in_buffer_index +
s->in_buffer_count);
562 s->in_buffer_count += count;
566 s->resample_in_constraint= 0;
567 if(
s->in_buffer_count != count || in_count)
577 s->resample_in_constraint= !!out_count;
599 if(
s->resample_first){
613 midbuf_tmp=
s->midbuf;
615 preout_tmp=
s->preout;
621 if(
s->resample_first ? !
s->resample : !
s->rematrix)
624 if(
s->resample_first ? !
s->rematrix : !
s->resample)
627 if(
s->int_sample_fmt ==
s->out_sample_fmt &&
s->out.planar
630 out_count=
FFMIN(out_count, in_count);
644 if(
s->resample_first){
658 if(
s->dither.method){
660 int dither_count=
FFMAX(out_count, 1<<16);
663 conv_src = &
s->dither.temp;
671 for(ch=0; ch<
s->dither.noise.ch_count; ch++)
672 if((ret=
swri_get_dither(
s,
s->dither.noise.ch[ch],
s->dither.noise.count, (12345678913579ULL*ch + 3141592) % 2718281828U,
s->dither.noise.fmt))<0)
676 if(
s->dither.noise_pos + out_count >
s->dither.noise.count)
677 s->dither.noise_pos = 0;
680 if (
s->mix_2_1_simd) {
681 int len1= out_count&~15;
686 s->mix_2_1_simd(conv_src->
ch[ch],
preout->
ch[ch],
s->dither.noise.ch[ch] +
s->dither.noise.bps *
s->dither.noise_pos,
s->native_simd_one, 0, 0, len1);
687 if(out_count != len1)
689 s->mix_2_1_f(conv_src->
ch[ch] + off,
preout->
ch[ch] + off,
s->dither.noise.ch[ch] +
s->dither.noise.bps *
s->dither.noise_pos + off,
s->native_one, 0, 0, out_count - len1);
692 s->mix_2_1_f(conv_src->
ch[ch],
preout->
ch[ch],
s->dither.noise.ch[ch] +
s->dither.noise.bps *
s->dither.noise_pos,
s->native_one, 0, 0, out_count);
695 switch(
s->int_sample_fmt) {
702 s->dither.noise_pos += out_count;
711 return !!
s->in_buffer.ch_count;
724 #if defined(ASSERT_LEVEL) && ASSERT_LEVEL >1
728 while(
s->drop_output > 0){
731 #define MAX_DROP_STEP 16384
736 s->drop_output *= -1;
738 s->drop_output *= -1;
741 s->drop_output -= ret;
742 if (!
s->drop_output && !out_arg)
754 s->resampler->flush(
s);
755 s->resample_in_constraint = 0;
757 }
else if(!
s->in_buffer_count){
767 if(ret>0 && !
s->drop_output)
768 s->outpts += ret * (int64_t)
s->in_sample_rate;
770 av_assert2(max_output < 0 || ret < 0 || ret <= max_output);
784 s->in_buffer_count -= ret;
785 s->in_buffer_index += ret;
788 if(!
s->in_buffer_count)
789 s->in_buffer_index = 0;
793 size=
s->in_buffer_index +
s->in_buffer_count + in_count - out_count;
795 if(in_count > out_count) {
796 if(
size >
s->in_buffer.count
799 copy(&
s->in_buffer, &
tmp,
s->in_buffer_count);
800 s->in_buffer_index=0;
816 buf_set(&
tmp, &
s->in_buffer,
s->in_buffer_index +
s->in_buffer_count);
818 s->in_buffer_count += in_count;
821 if(ret2>0 && !
s->drop_output)
822 s->outpts += ret2 * (int64_t)
s->in_sample_rate;
823 av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output);
830 s->drop_output += count;
832 if(
s->drop_output <= 0)
846 #define MAX_SILENCE_STEP 16384
856 if(
s->silence.planar)
for(
i=0;
i<
s->silence.ch_count;
i++) {
857 memset(
s->silence.ch[
i],
s->silence.bps==1 ? 0x80 : 0, count*
s->silence.bps);
859 memset(
s->silence.ch[0],
s->silence.bps==1 ? 0x80 : 0, count*
s->silence.bps*
s->silence.ch_count);
868 if (
s->resampler &&
s->resample){
869 return s->resampler->get_delay(
s,
base);
871 return (
s->in_buffer_count*
base + (
s->in_sample_rate>>1))/
s->in_sample_rate;
882 if (
s->resampler &&
s->resample) {
883 if (!
s->resampler->get_out_samples)
885 out_samples =
s->resampler->get_out_samples(
s, in_samples);
887 out_samples =
s->in_buffer_count + in_samples;
891 if (out_samples > INT_MAX)
900 if (!
s || compensation_distance < 0)
902 if (!compensation_distance && sample_delta)
910 if (!
s->resampler->set_compensation){
913 return s->resampler->set_compensation(
s->resample, sample_delta, compensation_distance);
922 s->outpts =
s->firstpts =
pts;
924 if(
s->min_compensation >= FLT_MAX) {
927 int64_t
delta =
pts -
swr_get_delay(
s,
s->in_sample_rate * (int64_t)
s->out_sample_rate) -
s->outpts +
s->drop_output*(int64_t)
s->in_sample_rate;
928 double fdelta =
delta /(
double)(
s->in_sample_rate * (int64_t)
s->out_sample_rate);
930 if(
fabs(fdelta) >
s->min_compensation) {
931 if(
s->outpts ==
s->firstpts ||
fabs(fdelta) >
s->min_hard_compensation){
938 }
else if(
s->soft_compensation_duration &&
s->max_soft_compensation) {
939 int duration =
s->out_sample_rate *
s->soft_compensation_duration;
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)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Audio format conversion routines.
AudioConvert * swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags)
Create an audio sample format converter context.
void swri_audio_convert_free(AudioConvert **ctx)
Free audio sample format converter context.
int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len)
Convert between audio sample formats.
simple assert() macros that are a bit more flexible than ISO C assert().
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
#define av_assert0(cond)
assert() equivalent, that is always enabled.
#define ss(width, name, subs,...)
audio channel layout utility functions
#define FFMPEG_CONFIGURATION
static __device__ float fabs(float a)
static void comp(unsigned char *dst, ptrdiff_t dst_stride, unsigned char *src, ptrdiff_t src_stride, int add)
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
int64_t av_get_default_channel_layout(int nb_channels)
Return default channel layout for a given number of channels.
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
#define AV_LOG_WARNING
Something somehow does not look correct.
#define AV_LOG_VERBOSE
Detailed information.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void * av_mallocz_array(size_t nmemb, size_t size)
Allocate a memory block for an array with av_mallocz().
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
Get the planar alternative form of the given sample format.
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_FLTP
float, planar
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
@ AV_SAMPLE_FMT_NB
Number of sample formats. DO NOT USE if linking dynamically.
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
@ AV_SAMPLE_FMT_DBLP
double, planar
@ AV_SAMPLE_FMT_S64P
signed 64 bits, planar
#define AV_NOPTS_VALUE
Undefined timestamp value.
int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map)
Set a customized input channel mapping.
#define SWR_FLAG_RESAMPLE
Force resampling even if equal sample rate.
const char * swresample_license(void)
Return the swr license.
int swr_drop_output(struct SwrContext *s, int count)
Drops the specified number of output samples.
int swr_inject_silence(struct SwrContext *s, int count)
Injects the specified number of silence samples.
int64_t swr_get_delay(struct SwrContext *s, int64_t base)
Gets the delay the next input sample will experience relative to the next output sample.
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
int swr_get_out_samples(struct SwrContext *s, int in_samples)
Find an upper bound on the number of samples that the next swr_convert call will output,...
av_cold void swr_close(SwrContext *s)
Closes the context so that swr_is_initialized() returns 0.
int64_t swr_next_pts(struct SwrContext *s, int64_t pts)
Convert the next timestamp from input to output timestamps are in 1/(in_sample_rate * out_sample_rate...
int swr_is_initialized(struct SwrContext *s)
Check whether an swr context has been initialized or not.
int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance)
Activate resampling compensation ("soft" compensation).
const char * swresample_configuration(void)
Return the swr build-time configuration.
struct SwrContext * swr_alloc_set_opts(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, int log_offset, void *log_ctx)
Allocate SwrContext if needed and set/reset common parameters.
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
unsigned swresample_version(void)
Return the LIBSWRESAMPLE_VERSION_INT constant.
av_cold struct SwrContext * swr_alloc(void)
Allocate SwrContext.
@ SWR_DITHER_NS
not part of API/ABI
@ SWR_ENGINE_SWR
SW Resampler.
@ SWR_ENGINE_SOXR
SoX Resampler.
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
common internal API header
#define attribute_align_arg
av_cold int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt)
int swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt)
struct Resampler const swri_resampler
#define LIBSWRESAMPLE_VERSION_INT
#define LIBSWRESAMPLE_VERSION_MICRO
int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy)
av_cold int swri_rematrix_init(SwrContext *s)
av_cold void swri_rematrix_free(SwrContext *s)
struct Resampler const swri_soxr_resampler
Audio buffer used for intermediate storage between conversion phases.
int ch_count
number of channels
int planar
1 if planar audio, 0 otherwise
uint8_t * data[AVRESAMPLE_MAX_CHANNELS]
data plane pointers
uint8_t * ch[SWR_CH_MAX]
samples buffer per channel
enum AVSampleFormat fmt
sample format
The libswresample context.
int user_out_ch_count
User set output channel count.
float max_soft_compensation
swr maximum soft compensation in seconds over soft_compensation_duration
int out_sample_rate
output sample rate
int in_buffer_index
cached buffer position
AudioData postin
post-input audio data: used for rematrix/resample
int in_sample_rate
input sample rate
AudioData midbuf
intermediate audio data (postin/preout)
void * log_ctx
parent logging context
AudioData preout
pre-output audio data: used for rematrix/resample
int64_t out_ch_layout
output channel layout
int64_t user_in_ch_layout
User set input channel layout.
enum AVSampleFormat in_sample_fmt
input sample format
int64_t in_ch_layout
input channel layout
struct AudioConvert * in_convert
input conversion context
enum AVSampleFormat out_sample_fmt
output sample format
int user_in_ch_count
User set input channel count.
const char swr_ffversion[]
static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count, AudioData *in, int in_count)
static void copy(AudioData *out, AudioData *in, int count)
int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, const uint8_t *in_arg[SWR_CH_MAX], int in_count)
static void buf_set(AudioData *out, AudioData *in, int count)
out may be equal in.
static void reversefill_audiodata(AudioData *out, uint8_t *in_arg[SWR_CH_MAX])
static void fill_audiodata(AudioData *out, uint8_t *in_arg[SWR_CH_MAX])
static int resample(SwrContext *s, AudioData *out_param, int out_count, const AudioData *in_param, int in_count)
static void clear_context(SwrContext *s)
static void free_temp(AudioData *a)
static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt)
int swri_realloc_audio(AudioData *a, int count)
void swri_noise_shaping_int16(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
void swri_noise_shaping_int32(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
void swri_noise_shaping_float(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)
void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count)