58 const LADSPA_Descriptor *
desc;
72 #define OFFSET(x) offsetof(LADSPAContext, x) 73 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM 113 const LADSPA_PortRangeHint *
h = s->
desc->PortRangeHints + map[ctl];
115 av_log(ctx, level,
"c%i: %s [", ctl, s->
desc->PortNames[map[ctl]]);
117 if (LADSPA_IS_HINT_TOGGLED(h->HintDescriptor)) {
118 av_log(ctx, level,
"toggled (1 or 0)");
120 if (LADSPA_IS_HINT_HAS_DEFAULT(h->HintDescriptor))
121 av_log(ctx, level,
" (default %i)", (
int)values[ctl]);
123 if (LADSPA_IS_HINT_INTEGER(h->HintDescriptor)) {
124 av_log(ctx, level,
"<int>");
126 if (LADSPA_IS_HINT_BOUNDED_BELOW(h->HintDescriptor))
127 av_log(ctx, level,
", min: %i", (
int)h->LowerBound);
129 if (LADSPA_IS_HINT_BOUNDED_ABOVE(h->HintDescriptor))
130 av_log(ctx, level,
", max: %i", (
int)h->UpperBound);
133 av_log(ctx, level,
" (value %d)", (
int)values[ctl]);
134 else if (LADSPA_IS_HINT_HAS_DEFAULT(h->HintDescriptor))
135 av_log(ctx, level,
" (default %d)", (
int)values[ctl]);
137 av_log(ctx, level,
"<float>");
139 if (LADSPA_IS_HINT_BOUNDED_BELOW(h->HintDescriptor))
140 av_log(ctx, level,
", min: %f", h->LowerBound);
142 if (LADSPA_IS_HINT_BOUNDED_ABOVE(h->HintDescriptor))
143 av_log(ctx, level,
", max: %f", h->UpperBound);
146 av_log(ctx, level,
" (value %f)", values[ctl]);
147 else if (LADSPA_IS_HINT_HAS_DEFAULT(h->HintDescriptor))
148 av_log(ctx, level,
" (default %f)", values[ctl]);
151 if (LADSPA_IS_HINT_SAMPLE_RATE(h->HintDescriptor))
152 av_log(ctx, level,
", multiple of sample rate");
154 if (LADSPA_IS_HINT_LOGARITHMIC(h->HintDescriptor))
155 av_log(ctx, level,
", logarithmic scale");
158 av_log(ctx, level,
"]\n");
166 int i,
h, p, new_out_samples;
173 !(s->
desc->Properties & LADSPA_PROPERTY_INPLACE_BROKEN))) {
215 new_out_samples -= trim;
219 if (new_out_samples <= 0) {
222 }
else if (new_out_samples < out->
nb_samples) {
224 for (
int ch = 0; ch < out->
channels; ch++)
226 sizeof(
float) * new_out_samples);
282 const LADSPA_PortRangeHint *
h = s->
desc->PortRangeHints + map[ctl];
283 const LADSPA_Data lower = h->LowerBound;
284 const LADSPA_Data upper = h->UpperBound;
286 if (LADSPA_IS_HINT_DEFAULT_MINIMUM(h->HintDescriptor)) {
288 }
else if (LADSPA_IS_HINT_DEFAULT_MAXIMUM(h->HintDescriptor)) {
290 }
else if (LADSPA_IS_HINT_DEFAULT_0(h->HintDescriptor)) {
292 }
else if (LADSPA_IS_HINT_DEFAULT_1(h->HintDescriptor)) {
294 }
else if (LADSPA_IS_HINT_DEFAULT_100(h->HintDescriptor)) {
296 }
else if (LADSPA_IS_HINT_DEFAULT_440(h->HintDescriptor)) {
298 }
else if (LADSPA_IS_HINT_DEFAULT_LOW(h->HintDescriptor)) {
299 if (LADSPA_IS_HINT_LOGARITHMIC(h->HintDescriptor))
300 values[ctl] =
exp(log(lower) * 0.75 + log(upper) * 0.25);
302 values[ctl] = lower * 0.75 + upper * 0.25;
303 }
else if (LADSPA_IS_HINT_DEFAULT_MIDDLE(h->HintDescriptor)) {
304 if (LADSPA_IS_HINT_LOGARITHMIC(h->HintDescriptor))
305 values[ctl] =
exp(log(lower) * 0.5 + log(upper) * 0.5);
307 values[ctl] = lower * 0.5 + upper * 0.5;
308 }
else if (LADSPA_IS_HINT_DEFAULT_HIGH(h->HintDescriptor)) {
309 if (LADSPA_IS_HINT_LOGARITHMIC(h->HintDescriptor))
310 values[ctl] =
exp(log(lower) * 0.25 + log(upper) * 0.75);
312 values[ctl] = lower * 0.25 + upper * 0.75;
341 if (s->
desc->activate)
387 LADSPA_PortDescriptor pd;
390 for (i = 0; i < desc->PortCount; i++) {
391 pd = desc->PortDescriptors[
i];
393 if (LADSPA_IS_PORT_AUDIO(pd)) {
394 if (LADSPA_IS_PORT_INPUT(pd)) {
396 }
else if (LADSPA_IS_PORT_OUTPUT(pd)) {
403 static void *
try_load(
const char *dir,
const char *soname)
409 ret = dlopen(path, RTLD_LOCAL|RTLD_NOW);
419 const char *label = s->
desc->Label;
420 LADSPA_PortRangeHint *
h = (LADSPA_PortRangeHint *)s->
desc->PortRangeHints +
429 if (LADSPA_IS_HINT_BOUNDED_BELOW(h->HintDescriptor) &&
432 "%s: input control c%ld is below lower boundary of %0.4f.\n",
433 label, port, h->LowerBound);
437 if (LADSPA_IS_HINT_BOUNDED_ABOVE(h->HintDescriptor) &&
438 value > h->UpperBound) {
440 "%s: input control c%ld is above upper boundary of %0.4f.\n",
441 label, port, h->UpperBound);
453 LADSPA_Descriptor_Function descriptor_fn;
454 const LADSPA_Descriptor *
desc;
455 LADSPA_PortDescriptor pd;
457 char *p, *
arg, *saveptr =
NULL;
458 unsigned long nb_ports;
471 char *paths =
av_strdup(getenv(
"LADSPA_PATH"));
472 const char *home_path = getenv(
"HOME");
473 const char *separator =
":";
505 descriptor_fn = dlsym(s->
dl_handle,
"ladspa_descriptor");
506 if (!descriptor_fn) {
518 for (i = 0; desc = descriptor_fn(i); i++) {
532 desc = descriptor_fn(i);
538 if (desc->Label && !strcmp(desc->Label, s->
plugin))
544 nb_ports = desc->PortCount;
557 for (i = 0; i < nb_ports; i++) {
558 pd = desc->PortDescriptors[
i];
560 if (LADSPA_IS_PORT_AUDIO(pd)) {
561 if (LADSPA_IS_PORT_INPUT(pd)) {
564 }
else if (LADSPA_IS_PORT_OUTPUT(pd)) {
568 }
else if (LADSPA_IS_PORT_CONTROL(pd)) {
569 if (LADSPA_IS_PORT_INPUT(pd)) {
572 if (LADSPA_IS_HINT_HAS_DEFAULT(desc->PortRangeHints[i].HintDescriptor))
578 }
else if (LADSPA_IS_PORT_OUTPUT(pd)) {
589 "The '%s' plugin does not have any input controls.\n",
593 "The '%s' plugin has the following input controls:\n",
607 if (!(arg =
av_strtok(p,
" |", &saveptr)))
611 if (
av_sscanf(arg,
"c%d=%f", &i, &val) != 2) {
750 if (s->
desc->deactivate)
752 if (s->
desc->cleanup)
773 char *res,
int res_len,
int flags)
798 .priv_class = &ladspa_class,
static void set_default_ctl_value(LADSPAContext *s, int ctl, unsigned long *map, LADSPA_Data *values)
static int request_frame(AVFilterLink *outlink)
This structure describes decoded (raw) audio or video data.
static int config_output(AVFilterLink *outlink)
Main libavfilter public API header.
#define AVFILTER_FLAG_DYNAMIC_INPUTS
The number of the filter inputs is not determined just by AVFilter.inputs.
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
enum AVMediaType type
AVFilterPad type.
AVFilterFormatsConfig outcfg
Lists of supported formats / etc.
#define AV_CH_LAYOUT_STEREO
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
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.
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
GLsizei GLboolean const GLfloat * value
static void count_ports(const LADSPA_Descriptor *desc, unsigned long *nb_inputs, unsigned long *nb_outputs)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
#define AVERROR_EOF
End of file.
#define AV_LOG_VERBOSE
Detailed information.
AVFILTER_DEFINE_CLASS(ladspa)
A filter pad used for either input or output.
static void * av_x_if_null(const void *p, const void *x)
Return x default pointer in case p is NULL.
A link between two filters.
AVFilterPad * input_pads
array of input pads
#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
const LADSPA_Descriptor * desc
#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().
int av_sscanf(const char *string, const char *format,...)
See libc sscanf manual for more information.
char * av_asprintf(const char *fmt,...)
int(* config_props)(AVFilterLink *link)
Link configuration callback.
int channels
number of audio channels, only used for audio.
audio channel layout utility functions
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
unsigned nb_inputs
number of input pads
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
#define AV_TIME_BASE
Internal time base represented as integer.
int av_strcasecmp(const char *a, const char *b)
Locale-independent case-insensitive compare.
static av_cold int init(AVFilterContext *ctx)
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return values
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
static int set_control(AVFilterContext *ctx, unsigned long port, LADSPA_Data value)
AVFilterContext * src
source filter
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
static av_cold void uninit(AVFilterContext *ctx)
static const AVFilterPad outputs[]
int format
agreed upon media format
A list of supported channel layouts.
static int find_latency(AVFilterContext *ctx, LADSPAContext *s)
static int config_input(AVFilterLink *inlink)
static void * try_load(const char *dir, const char *soname)
static const AVOption ladspa_options[]
#define AV_LOG_INFO
Standard information.
unsigned long nb_outputcontrols
char * av_strdup(const char *s)
Duplicate a string.
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,(*(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
Describe the class of an AVClass context structure.
int sample_rate
Sample rate of the audio data.
Rational number (pair of numerator and denominator).
static int query_formats(AVFilterContext *ctx)
const char * name
Filter name.
const VDPAUPixFmtMap * map
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a link
static int connect_ports(AVFilterContext *ctx, AVFilterLink *link)
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
#define flags(name, subs,...)
int(* filter_frame)(AVFilterLink *link, AVFrame *frame)
Filtering callback.
static void print_ctl_info(AVFilterContext *ctx, int level, LADSPAContext *s, int ctl, unsigned long *map, LADSPA_Data *values, int print)
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
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()...
uint64_t channel_layout
channel layout of current buffer (see libavutil/channel_layout.h)
int channels
Number of channels.
AVFilterFormatsConfig incfg
Lists of supported formats / etc.
AVFilterContext * dst
dest filter
static const AVFilterPad ladspa_outputs[]
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.
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
uint8_t ** extended_data
pointers to the data planes/channels.
#define AVERROR_EXTERNAL
Generic error in an external library.
static double val(void *priv, double ch)
unsigned long nb_inputcontrols
int nb_samples
number of audio samples (per channel) described by this frame
static void print(AVTreeNode *t, int depth)
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
static int ff_insert_inpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new input pad for the filter.