47 int64_t start, int64_t range,
int curve);
51 int curve0,
int curve1);
54 enum CurveType {
NONE = -1,
TRI,
QSIN,
ESIN,
HSIN,
LOG,
IPAR,
QUA,
CUB,
SQU,
CBR,
PAR,
EXP,
IQSIN,
IHSIN,
DESE,
DESI,
LOSI,
SINC,
ISINC,
NB_CURVES };
56 #define OFFSET(x) offsetof(AudioFadeContext, x)
57 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
58 #define TFLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
95 #define CUBE(a) ((a)*(a)*(a))
102 gain = sin(gain *
M_PI / 2.0);
106 gain = 0.6366197723675814 * asin(gain);
109 gain = 1.0 - cos(
M_PI / 4.0 * (
CUBE(2.0*gain - 1) + 1));
112 gain = (1.0 - cos(gain *
M_PI)) / 2.0;
116 gain = 0.3183098861837907 * acos(1 - 2 * gain);
120 gain =
exp(-11.512925464970227 * (1 - gain));
123 gain =
av_clipd(1 + 0.2 * log10(gain), 0, 1.0);
126 gain = 1 - sqrt(1 - gain);
129 gain = (1 - (1 - gain) * (1 - gain));
144 gain = gain <= 0.5 ?
cbrt(2 * gain) / 2: 1 -
cbrt(2 * (1 - gain)) / 2;
147 gain = gain <= 0.5 ?
CUBE(2 * gain) / 2: 1 -
CUBE(2 * (1 - gain)) / 2;
150 const double a = 1. / (1. - 0.787) - 1;
151 double A = 1. / (1.0 +
exp(0 -((gain-0.5) *
a * 2.0)));
152 double B = 1. / (1.0 +
exp(
a));
153 double C = 1. / (1.0 +
exp(0-
a));
154 gain = (
A -
B) / (
C -
B);
158 gain = gain >= 1.0 ? 1.0 : sin(
M_PI * (1.0 - gain)) / (
M_PI * (1.0 - gain));
161 gain = gain <= 0.0 ? 0.0 : 1.0 - sin(
M_PI * gain) / (
M_PI * gain);
171 #define FADE_PLANAR(name, type) \
172 static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
173 int nb_samples, int channels, int dir, \
174 int64_t start, int64_t range, int curve) \
178 for (i = 0; i < nb_samples; i++) { \
179 double gain = fade_gain(curve, start + i * dir, range); \
180 for (c = 0; c < channels; c++) { \
181 type *d = (type *)dst[c]; \
182 const type *s = (type *)src[c]; \
184 d[i] = s[i] * gain; \
189 #define FADE(name, type) \
190 static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \
191 int nb_samples, int channels, int dir, \
192 int64_t start, int64_t range, int curve) \
194 type *d = (type *)dst[0]; \
195 const type *s = (type *)src[0]; \
198 for (i = 0; i < nb_samples; i++) { \
199 double gain = fade_gain(curve, start + i * dir, range); \
200 for (c = 0; c < channels; c++, k++) \
201 d[k] = s[k] * gain; \
220 switch (outlink->format) {
241 #if CONFIG_AFADE_FILTER
243 static const AVOption afade_options[] = {
248 {
"start_sample",
"set number of first sample to start fading",
OFFSET(start_sample),
AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX,
TFLAGS },
287 if (INT64_MAX -
s->nb_samples <
s->start_sample)
301 if ((!
s->type && (
s->start_sample +
s->nb_samples < cur_sample)) ||
302 (
s->type && (cur_sample + nb_samples < s->start_sample)))
314 if ((!
s->type && (cur_sample + nb_samples < s->start_sample)) ||
315 (
s->type && (
s->start_sample +
s->nb_samples < cur_sample))) {
322 start = cur_sample -
s->start_sample;
324 start =
s->start_sample +
s->nb_samples - cur_sample;
328 s->type ? -1 : 1, start,
329 s->nb_samples,
s->curve);
339 char *res,
int res_len,
int flags)
350 static const AVFilterPad avfilter_af_afade_inputs[] = {
359 static const AVFilterPad avfilter_af_afade_outputs[] = {
374 .
inputs = avfilter_af_afade_inputs,
375 .
outputs = avfilter_af_afade_outputs,
376 .priv_class = &afade_class,
383 #if CONFIG_ACROSSFADE_FILTER
385 static const AVOption acrossfade_options[] = {
386 {
"nb_samples",
"set number of samples for cross fade duration",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10,
FLAGS },
387 {
"ns",
"set number of samples for cross fade duration",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10,
FLAGS },
421 #define CROSSFADE_PLANAR(name, type) \
422 static void crossfade_samples_## name ##p(uint8_t **dst, uint8_t * const *cf0, \
423 uint8_t * const *cf1, \
424 int nb_samples, int channels, \
425 int curve0, int curve1) \
429 for (i = 0; i < nb_samples; i++) { \
430 double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
431 double gain1 = fade_gain(curve1, i, nb_samples); \
432 for (c = 0; c < channels; c++) { \
433 type *d = (type *)dst[c]; \
434 const type *s0 = (type *)cf0[c]; \
435 const type *s1 = (type *)cf1[c]; \
437 d[i] = s0[i] * gain0 + s1[i] * gain1; \
442 #define CROSSFADE(name, type) \
443 static void crossfade_samples_## name (uint8_t **dst, uint8_t * const *cf0, \
444 uint8_t * const *cf1, \
445 int nb_samples, int channels, \
446 int curve0, int curve1) \
448 type *d = (type *)dst[0]; \
449 const type *s0 = (type *)cf0[0]; \
450 const type *s1 = (type *)cf1[0]; \
453 for (i = 0; i < nb_samples; i++) { \
454 double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
455 double gain1 = fade_gain(curve1, i, nb_samples); \
456 for (c = 0; c < channels; c++, k++) \
457 d[k] = s0[k] * gain0 + s1[k] * gain1; \
461 CROSSFADE_PLANAR(dbl,
double)
462 CROSSFADE_PLANAR(flt,
float)
463 CROSSFADE_PLANAR(s16, int16_t)
466 CROSSFADE(dbl,
double)
467 CROSSFADE(flt,
float)
468 CROSSFADE(s16, int16_t)
476 int ret = 0, nb_samples, status;
481 if (
s->crossfade_is_over) {
488 }
else if (ret < 0) {
503 if (nb_samples > 0) {
532 s->crossfade_samples(
out->extended_data, cf[0]->extended_data,
533 cf[1]->extended_data,
534 s->nb_samples,
out->channels,
535 s->curve,
s->curve2);
539 s->crossfade_is_over = 1;
554 s->fade_samples(
out->extended_data, cf[0]->extended_data,
s->nb_samples,
555 outlink->
channels, -1,
s->nb_samples - 1,
s->nb_samples,
s->curve);
574 s->fade_samples(
out->extended_data, cf[1]->extended_data,
s->nb_samples,
575 outlink->
channels, 1, 0,
s->nb_samples,
s->curve2);
579 s->crossfade_is_over = 1;
601 static int acrossfade_config_output(
AVFilterLink *outlink)
606 if (
ctx->inputs[0]->sample_rate !=
ctx->inputs[1]->sample_rate) {
608 "Inputs must have the same sample rate "
609 "%d for in0 vs %d for in1\n",
610 ctx->inputs[0]->sample_rate,
ctx->inputs[1]->sample_rate);
619 switch (outlink->
format) {
635 static const AVFilterPad avfilter_af_acrossfade_inputs[] = {
637 .
name =
"crossfade0",
641 .name =
"crossfade1",
647 static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
651 .config_props = acrossfade_config_output,
657 .
name =
"acrossfade",
662 .priv_class = &acrossfade_class,
663 .
inputs = avfilter_af_acrossfade_inputs,
664 .
outputs = avfilter_af_acrossfade_outputs,
static enum AVSampleFormat sample_fmts[]
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
static int activate(AVFilterContext *ctx)
#define FADE_PLANAR(name, type)
static double fade_gain(int curve, int64_t index, int64_t range)
static int query_formats(AVFilterContext *ctx)
static int config_output(AVFilterLink *outlink)
AVFilter ff_af_acrossfade
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
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
int ff_outlink_get_status(AVFilterLink *link)
Get the status on an output link.
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options.
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link's FIFO and update the link's stats.
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
int ff_inlink_queued_samples(AVFilterLink *link)
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Main libavfilter public API header.
static av_cold int init(AVCodecContext *avctx)
#define flags(name, subs,...)
static int filter_frame(DBEDecodeContext *s, AVFrame *frame)
static int64_t start_time
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
#define AVERROR_EOF
End of file.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_FLTP
float, planar
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
@ AV_SAMPLE_FMT_S32
signed 32 bits
@ AV_SAMPLE_FMT_DBLP
double, planar
@ AV_SAMPLE_FMT_DBL
double
@ AV_SAMPLE_FMT_S16
signed 16 bits
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
#define AV_NOPTS_VALUE
Undefined timestamp value.
#define AV_TIME_BASE
Internal time base represented as integer.
#define AVFILTER_DEFINE_CLASS(fname)
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
enum MovChannelLayoutTag * layouts
typedef void(RENAME(mix_any_func_type))
Describe the class of an AVClass context structure.
A list of supported channel layouts.
void * priv
private data for use by the filter
AVFilterLink ** outputs
array of pointers to output links
A link between two filters.
int channels
Number of channels.
AVFilterContext * src
source filter
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link.
int sample_rate
samples per second
uint64_t channel_layout
channel layout of current buffer (see libavutil/channel_layout.h)
AVFilterContext * dst
dest filter
int format
agreed upon media format
A filter pad used for either input or output.
const char * name
Pad name.
const char * name
Filter name.
This structure describes decoded (raw) audio or video data.
int nb_samples
number of audio samples (per channel) described by this frame
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
int channels
number of audio channels, only used for audio.
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames,...
uint8_t ** extended_data
pointers to the data planes/channels.
Rational number (pair of numerator and denominator).
void(* fade_samples)(uint8_t **dst, uint8_t *const *src, int nb_samples, int channels, int direction, int64_t start, int64_t range, int curve)
void(* crossfade_samples)(uint8_t **dst, uint8_t *const *cf0, uint8_t *const *cf1, int nb_samples, int channels, int curve0, int curve1)