71 #define OFFSET(x) offsetof(CompandContext, x)
72 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
75 {
"attacks",
"set time over which increase of volume is determined",
OFFSET(attacks),
AV_OPT_TYPE_STRING, { .str =
"0.3" }, 0, 0,
A },
76 {
"decays",
"set time over which decrease of volume is determined",
OFFSET(decays),
AV_OPT_TYPE_STRING, { .str =
"0.8" }, 0, 0,
A },
77 {
"points",
"set points of transfer function",
OFFSET(points),
AV_OPT_TYPE_STRING, { .str =
"-70/-70|-60/-20" }, 0, 0,
A },
81 {
"delay",
"set delay for samples before sending them to volume adjuster",
OFFSET(delay),
AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, 0, 20,
A },
138 for (p = item_str; *p; p++) {
139 if (*p ==
' ' || *p ==
'|')
157 double in_log, out_log;
160 if (in_lin < s->in_min_lin)
163 in_log = log(in_lin);
166 if (in_log <= s->segments[i].x)
170 out_log = cs->
y + in_log * (cs->
a * in_log + cs->
b);
179 const int channels = inlink->
channels;
201 for (chan = 0; chan < channels; chan++) {
206 for (i = 0; i < nb_samples; i++) {
213 if (frame != out_frame)
219 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
225 const int channels = inlink->
channels;
237 for (chan = 0; chan < channels; chan++) {
246 for (i = 0, oindex = 0; i < nb_samples; i++) {
247 const double in = src[i];
297 const int channels = outlink->
channels;
310 for (chan = 0; chan < channels; chan++) {
311 AVFrame *delay_frame =
s->delay_frame;
316 dindex =
s->delay_index;
319 dindex =
MOD(dindex + 1,
s->delay_samples);
323 s->delay_index = dindex;
334 char *p, *saveptr =
NULL;
335 const int channels = outlink->
channels;
336 int nb_attacks, nb_decays, nb_points;
337 int new_nb_items, num;
351 if (nb_attacks > channels || nb_decays > channels) {
353 "Number of attacks/decays bigger than number of channels.\n");
369 for (i = 0, new_nb_items = 0; i < nb_attacks; i++) {
370 char *tstr =
av_strtok(p,
" |", &saveptr);
378 nb_attacks = new_nb_items;
381 for (i = 0, new_nb_items = 0; i < nb_decays; i++) {
382 char *tstr =
av_strtok(p,
" |", &saveptr);
384 new_nb_items += sscanf(tstr,
"%lf", &s->
channels[i].
decay) == 1;
390 nb_decays = new_nb_items;
392 if (nb_attacks != nb_decays) {
394 "Number of attacks %d differs from number of decays %d.\n",
395 nb_attacks, nb_decays);
400 for (i = nb_decays; i < channels; i++) {
405 #define S(x) s->segments[2 * ((x) + 1)]
407 for (i = 0, new_nb_items = 0; i < nb_points; i++) {
408 char *tstr =
av_strtok(p,
" |", &saveptr);
410 if (sscanf(tstr,
"%lf/%lf", &
S(i).x, &
S(i).y) != 2) {
412 "Invalid and/or missing input/output value.\n");
416 if (i &&
S(i - 1).x >
S(i).x) {
418 "Transfer function input values must be increasing.\n");
429 if (num == 0 ||
S(num - 1).x)
433 #define S(x) s->segments[2 * (x)]
440 for (i = 2; i < num; i++) {
441 double g1 = (
S(i - 1).y -
S(i - 2).y) * (
S(i - 0).x -
S(i - 1).x);
442 double g2 = (
S(i - 0).y -
S(i - 1).y) * (
S(i - 1).x -
S(i - 2).x);
448 for (j = --i; j < num; j++)
458 #define L(x) s->segments[i - (x)]
460 double x, y, cx, cy, in1, in2, out1, out2, theta,
len,
r;
463 L(4).b = (
L(2).y -
L(4).y) / (
L(2).x -
L(4).x);
466 L(2).b = (
L(0).y -
L(2).y) / (
L(0).x -
L(2).x);
468 theta = atan2(
L(2).y -
L(4).y,
L(2).x -
L(4).x);
469 len =
hypot(
L(2).x -
L(4).x,
L(2).y -
L(4).y);
470 r =
FFMIN(radius, len);
471 L(3).x =
L(2).x - r * cos(theta);
472 L(3).y =
L(2).y - r * sin(theta);
474 theta = atan2(
L(0).y -
L(2).y,
L(0).x -
L(2).x);
475 len =
hypot(
L(0).x -
L(2).x,
L(0).y -
L(2).y);
476 r =
FFMIN(radius, len / 2);
477 x =
L(2).x + r * cos(theta);
478 y =
L(2).y + r * sin(theta);
480 cx = (
L(3).x +
L(2).x + x) / 3;
481 cy = (
L(3).y +
L(2).y + y) / 3;
488 in2 =
L(2).x -
L(3).x;
489 out2 =
L(2).y -
L(3).y;
490 L(3).a = (out2 / in2 - out1 / in1) / (in2 - in1);
491 L(3).b = out1 / in1 -
L(3).a * in1;
499 for (i = 0; i < channels; i++) {
502 if (cp->
attack > 1.0 / sample_rate)
506 if (cp->
decay > 1.0 / sample_rate)
582 "Compress or expand audio dynamic range."),
585 .priv_class = &compand_class,
static const AVFilterPad compand_inputs[]
This structure describes decoded (raw) audio or video data.
static const AVOption compand_options[]
static double get_volume(CompandContext *s, double in_lin)
Main libavfilter public API header.
AVFILTER_DEFINE_CLASS(compand)
static const AVFilterPad compand_outputs[]
static void update_volume(ChanParam *cp, double in)
int is_disabled
the enabled state from the last expression evaluation
static av_cold int init(AVFilterContext *ctx)
const char * name
Pad name.
AVFilterLink ** inputs
array of pointers to input links
#define av_assert0(cond)
assert() equivalent, that is always enabled.
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
static int compand_delay(AVFilterContext *ctx, AVFrame *frame)
static int query_formats(AVFilterContext *ctx)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
#define AVERROR_EOF
End of file.
A filter pad used for either input or output.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
A link between two filters.
static int compand_nodelay(AVFilterContext *ctx, AVFrame *frame)
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int sample_rate
samples per second
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
void * priv
private data for use by the filter
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link...
simple assert() macros that are a bit more flexible than ISO C assert().
static void * av_mallocz_array(size_t nmemb, size_t size)
uint64_t channel_layout
Channel layout of the audio data.
static int request_frame(AVFilterLink *outlink)
static av_const double hypot(double x, double y)
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
CompandSegment * segments
AVFilterContext * src
source filter
static const AVFilterPad outputs[]
int format
agreed upon media format
A list of supported channel layouts.
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
static const AVFilterPad inputs[]
AVSampleFormat
Audio sample formats.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
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
Describe the class of an AVClass context structure.
Rational number (pair of numerator and denominator).
const char * name
Filter name.
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok()...
static av_cold void uninit(AVFilterContext *ctx)
internal math functions header
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
uint64_t channel_layout
channel layout of current buffer (see libavutil/channel_layout.h)
int channels
Number of channels.
static int config_output(AVFilterLink *outlink)
static int compand_drain(AVFilterLink *outlink)
AVFilterContext * dst
dest filter
int(* compand)(AVFilterContext *ctx, AVFrame *frame)
static void count_items(char *item_str, int *nb_items)
static enum AVSampleFormat sample_fmts[]
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
uint8_t ** extended_data
pointers to the data planes/channels.
int nb_samples
number of audio samples (per channel) described by this frame
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
#define AV_NOPTS_VALUE
Undefined timestamp value.