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27 #include <lilv/lilv.h>
28 #include <lv2/lv2plug.in/ns/ext/atom/atom.h>
29 #include <lv2/lv2plug.in/ns/ext/buf-size/buf-size.h>
91 #define OFFSET(x) offsetof(LV2Context, x)
92 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
101 {
"nb_samples",
"set the number of samples per requested frame",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX,
FLAGS },
102 {
"n",
"set the number of samples per requested frame",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX,
FLAGS },
120 for (
i = 0;
i <
table->n_uris;
i++) {
130 const size_t len = strlen(uri);
134 for (
i = 0;
i <
table->n_uris;
i++) {
135 if (!strcmp(
table->uris[
i], uri)) {
142 return table->n_uris;
149 return table->n_uris;
154 return table->n_uris;
161 if (urid > 0 && urid <= table->n_uris) {
162 return table->uris[urid - 1];
170 int ich = 0, och = 0,
i;
172 for (
i = 0;
i <
s->nb_ports;
i++) {
173 const LilvPort *port = lilv_plugin_get_port_by_index(
s->plugin,
i);
175 if (lilv_port_is_a(
s->plugin, port,
s->lv2_AudioPort) ||
176 lilv_port_is_a(
s->plugin, port,
s->lv2_CVPort)) {
177 if (lilv_port_is_a(
s->plugin, port,
s->lv2_InputPort)) {
178 lilv_instance_connect_port(
s->instance,
i,
in->extended_data[ich++]);
179 }
else if (lilv_port_is_a(
s->plugin, port,
s->lv2_OutputPort)) {
180 lilv_instance_connect_port(
s->instance,
i,
out->extended_data[och++]);
184 }
else if (lilv_port_is_a(
s->plugin, port,
s->atom_AtomPort)) {
185 if (lilv_port_is_a(
s->plugin, port,
s->lv2_InputPort)) {
186 lilv_instance_connect_port(
s->instance,
i, &
s->seq_in);
188 lilv_instance_connect_port(
s->instance,
i,
s->seq_out);
190 }
else if (lilv_port_is_a(
s->plugin, port,
s->lv2_ControlPort)) {
191 lilv_instance_connect_port(
s->instance,
i, &
s->controls[
i]);
195 s->seq_in[0].atom.size =
sizeof(LV2_Atom_Sequence_Body);
196 s->seq_in[0].atom.type =
uri_table_map(&
s->uri_table, LV2_ATOM__Sequence);
197 s->seq_out->atom.size = 9624;
207 if (!
s->nb_outputs ||
221 lilv_instance_run(
s->instance,
in->nb_samples);
240 if (
s->duration >= 0 && t >=
s->duration)
249 lilv_instance_run(
s->instance,
out->nb_samples);
251 out->sample_rate =
s->sample_rate;
253 s->pts +=
s->nb_samples;
259 { LV2_BUF_SIZE__powerOf2BlockLength,
NULL },
260 { LV2_BUF_SIZE__fixedBlockLength,
NULL },
261 { LV2_BUF_SIZE__boundedBlockLength,
NULL },
268 char *p, *
arg, *saveptr =
NULL;
272 s->map.handle = &
s->uri_table;
274 s->map_feature.URI = LV2_URID_MAP_URI;
275 s->map_feature.data = &
s->map;
276 s->unmap.handle = &
s->uri_table;
278 s->unmap_feature.URI = LV2_URID_UNMAP_URI;
279 s->unmap_feature.data = &
s->unmap;
280 s->features[0] = &
s->map_feature;
281 s->features[1] = &
s->unmap_feature;
286 if (
ctx->nb_inputs) {
291 if (
s->nb_inputs ==
s->nb_outputs) {
301 s->instance = lilv_plugin_instantiate(
s->plugin,
sample_rate,
s->features);
303 av_log(
s,
AV_LOG_ERROR,
"Failed to instantiate <%s>\n", lilv_node_as_uri(lilv_plugin_get_uri(
s->plugin)));
309 s->controls =
av_calloc(
s->nb_ports,
sizeof(
float));
311 if (!
s->mins || !
s->maxes || !
s->controls)
314 lilv_plugin_get_port_ranges_float(
s->plugin,
s->mins,
s->maxes,
s->controls);
315 s->seq_out =
av_malloc(
sizeof(LV2_Atom_Sequence) + 9624);
319 if (
s->options && !strcmp(
s->options,
"help")) {
320 if (!
s->nb_inputcontrols) {
322 "The '%s' plugin does not have any input controls.\n",
326 "The '%s' plugin has the following input controls:\n",
328 for (
i = 0;
i <
s->nb_ports;
i++) {
329 const LilvPort *port = lilv_plugin_get_port_by_index(
s->plugin,
i);
330 const LilvNode *symbol = lilv_port_get_symbol(
s->plugin, port);
331 LilvNode *
name = lilv_port_get_name(
s->plugin, port);
333 if (lilv_port_is_a(
s->plugin, port,
s->lv2_InputPort) &&
334 lilv_port_is_a(
s->plugin, port,
s->lv2_ControlPort)) {
336 lilv_node_as_string(symbol),
s->mins[
i],
s->maxes[
i],
s->controls[
i],
337 lilv_node_as_string(
name));
340 lilv_node_free(
name);
348 const LilvPort *port;
358 vstr = strstr(
arg,
"=");
367 sym = lilv_new_string(
s->world, str);
368 port = lilv_plugin_get_port_by_symbol(
s->plugin, sym);
373 index = lilv_port_get_index(
s->plugin, port);
379 (lilv_plugin_has_feature(
s->plugin,
s->powerOf2BlockLength) ||
380 lilv_plugin_has_feature(
s->plugin,
s->fixedBlockLength) ||
381 lilv_plugin_has_feature(
s->plugin,
s->boundedBlockLength))) {
393 const LilvPlugins *plugins;
394 const LilvPlugin *plugin;
399 s->world = lilv_world_new();
403 uri = lilv_new_uri(
s->world,
s->plugin_uri);
409 lilv_world_load_all(
s->world);
410 plugins = lilv_world_get_all_plugins(
s->world);
411 plugin = lilv_plugins_get_by_uri(plugins, uri);
420 s->nb_ports = lilv_plugin_get_num_ports(
s->plugin);
422 s->lv2_InputPort = lilv_new_uri(
s->world, LV2_CORE__InputPort);
423 s->lv2_OutputPort = lilv_new_uri(
s->world, LV2_CORE__OutputPort);
424 s->lv2_AudioPort = lilv_new_uri(
s->world, LV2_CORE__AudioPort);
425 s->lv2_ControlPort = lilv_new_uri(
s->world, LV2_CORE__ControlPort);
426 s->lv2_Optional = lilv_new_uri(
s->world, LV2_CORE__connectionOptional);
427 s->atom_AtomPort = lilv_new_uri(
s->world, LV2_ATOM__AtomPort);
428 s->atom_Sequence = lilv_new_uri(
s->world, LV2_ATOM__Sequence);
429 s->urid_map = lilv_new_uri(
s->world, LV2_URID__map);
430 s->powerOf2BlockLength = lilv_new_uri(
s->world, LV2_BUF_SIZE__powerOf2BlockLength);
431 s->fixedBlockLength = lilv_new_uri(
s->world, LV2_BUF_SIZE__fixedBlockLength);
432 s->boundedBlockLength = lilv_new_uri(
s->world, LV2_BUF_SIZE__boundedBlockLength);
434 for (
i = 0;
i <
s->nb_ports;
i++) {
435 const LilvPort *lport = lilv_plugin_get_port_by_index(
s->plugin,
i);
439 is_optional = lilv_port_has_property(
s->plugin, lport,
s->lv2_Optional);
441 if (lilv_port_is_a(
s->plugin, lport,
s->lv2_InputPort)) {
443 }
else if (!lilv_port_is_a(
s->plugin, lport,
s->lv2_OutputPort) && !is_optional) {
447 if (lilv_port_is_a(
s->plugin, lport,
s->lv2_ControlPort)) {
449 s->nb_inputcontrols++;
451 }
else if (lilv_port_is_a(
s->plugin, lport,
s->lv2_AudioPort)) {
510 if (
s->nb_inputs == 2 &&
s->nb_outputs == 2) {
519 if (
s->nb_inputs >= 1) {
531 if (!
s->nb_outputs) {
538 if (
s->nb_outputs >= 1) {
558 lilv_node_free(
s->powerOf2BlockLength);
559 lilv_node_free(
s->fixedBlockLength);
560 lilv_node_free(
s->boundedBlockLength);
561 lilv_node_free(
s->urid_map);
562 lilv_node_free(
s->atom_Sequence);
563 lilv_node_free(
s->atom_AtomPort);
564 lilv_node_free(
s->lv2_Optional);
565 lilv_node_free(
s->lv2_ControlPort);
566 lilv_node_free(
s->lv2_AudioPort);
567 lilv_node_free(
s->lv2_OutputPort);
568 lilv_node_free(
s->lv2_InputPort);
570 lilv_instance_free(
s->instance);
571 lilv_world_free(
s->world);
595 .priv_class = &lv2_class,
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
@ AV_SAMPLE_FMT_FLTP
float, planar
A list of supported channel layouts.
#define AV_LOG_WARNING
Something somehow does not look correct.
LilvNode * lv2_OutputPort
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
struct AVFilterChannelLayouts * in_channel_layouts
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static enum AVSampleFormat sample_fmts[]
enum MovChannelLayoutTag * layouts
#define AVERROR_EOF
End of file.
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_asprintf(const char *fmt,...)
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
static const AVFilterPad lv2_outputs[]
This structure describes decoded (raw) audio or video data.
static const uint16_t table[]
static av_cold int init(AVFilterContext *ctx)
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
static void connect_ports(LV2Context *s, AVFrame *in, AVFrame *out)
const char * name
Filter name.
static const AVOption lv2_options[]
A link between two filters.
int channels
Number of channels.
static int ff_insert_inpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new input pad for the filter.
LilvNode * lv2_ControlPort
LilvNode * fixedBlockLength
#define AVFILTER_FLAG_DYNAMIC_INPUTS
The number of the filter inputs is not determined just by AVFilter.inputs.
#define AV_CH_LAYOUT_STEREO
A filter pad used for either input or output.
static int query_formats(AVFilterContext *ctx)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
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().
const LV2_Feature * features[5]
static const AVFilterPad outputs[]
static av_cold void uninit(AVFilterContext *ctx)
Describe the class of an AVClass context structure.
LilvNode * boundedBlockLength
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Rational number (pair of numerator and denominator).
uint64_t channel_layout
channel layout of current buffer (see libavutil/channel_layout.h)
int(* filter_frame)(AVFilterLink *link, AVFrame *frame)
Filtering callback.
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
static void uri_table_init(URITable *table)
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static void uri_table_destroy(URITable *table)
LV2_Atom_Sequence * seq_out
unsigned nb_inputcontrols
const OptionDef options[]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
int format
agreed upon media format
static LV2_URID uri_table_map(LV2_URID_Map_Handle handle, const char *uri)
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
AVFilterContext * src
source filter
const char const char void * val
static int request_frame(AVFilterLink *outlink)
#define AV_LOG_INFO
Standard information.
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 sample_rate
samples per second
#define i(width, name, range_min, range_max)
#define AV_TIME_BASE
Internal time base represented as integer.
AVSampleFormat
Audio sample formats.
const char * name
Pad name.
AVFILTER_DEFINE_CLASS(lv2)
LilvNode * powerOf2BlockLength
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
enum AVMediaType type
AVFilterPad type.
static int config_output(AVFilterLink *outlink)
static const LV2_Feature buf_size_features[3]
static const char * uri_table_unmap(LV2_URID_Map_Handle handle, LV2_URID urid)
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link.
#define flags(name, subs,...)
LV2_Atom_Sequence seq_in[2]
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
const LilvPlugin * plugin
LV2_Feature unmap_feature