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> 67 LV2_Atom_Sequence seq_in[2];
69 const LV2_Feature *features[5];
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)) {
143 memcpy(tmp, table->
uris, table->
n_uris *
sizeof(
char**));
151 memcpy(table->
uris[table->
n_uris], uri, len + 1);
161 if (urid > 0 && urid <= table->
n_uris) {
162 return table->
uris[urid - 1];
170 int ich = 0, och = 0,
i;
173 const LilvPort *port = lilv_plugin_get_port_by_index(s->
plugin,
i);
195 s->
seq_in[0].atom.size =
sizeof(LV2_Atom_Sequence_Body);
259 { LV2_BUF_SIZE__powerOf2BlockLength,
NULL },
260 { LV2_BUF_SIZE__fixedBlockLength, NULL },
261 { LV2_BUF_SIZE__boundedBlockLength, NULL },
268 char *p, *
arg, *saveptr =
NULL;
322 "The '%s' plugin does not have any input controls.\n",
326 "The '%s' plugin has the following input controls:\n",
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);
337 lilv_node_as_string(name));
340 lilv_node_free(name);
348 const LilvPort *port;
354 if (!(arg =
av_strtok(p,
" |", &saveptr)))
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);
393 const LilvPlugins *plugins;
394 const LilvPlugin *plugin;
399 s->
world = lilv_world_new();
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);
435 const LilvPort *lport = lilv_plugin_get_port_by_index(s->
plugin, i);
571 lilv_world_free(s->
world);
595 .priv_class = &lv2_class,
static void connect_ports(LV2Context *s, AVFrame *in, AVFrame *out)
This structure describes decoded (raw) audio or video data.
#define AV_LOG_WARNING
Something somehow does not look correct.
Main libavfilter public API header.
int max_samples
Maximum number of samples to filter at once.
LilvNode * lv2_ControlPort
#define AVFILTER_FLAG_DYNAMIC_INPUTS
The number of the filter inputs is not determined just by AVFilter.inputs.
static int query_formats(AVFilterContext *ctx)
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
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static const AVOption lv2_options[]
static av_cold void uninit(AVFilterContext *ctx)
static const AVFilterPad lv2_outputs[]
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
const LV2_Feature * features[5]
#define AVERROR_EOF
End of file.
static const uint16_t table[]
A filter pad used for either input or output.
A link between two filters.
static void uri_table_init(URITable *table)
AVFilterPad * input_pads
array of input pads
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int min_samples
Minimum number of samples to filter at once.
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. ...
LilvNode * fixedBlockLength
void * priv
private data for use by the filter
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 int request_frame(AVFilterLink *outlink)
char * av_asprintf(const char *fmt,...)
audio channel layout utility functions
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.
LilvNode * boundedBlockLength
AVFilterContext * src
source filter
int partial_buf_size
Size of the partial buffer to allocate.
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
static const AVFilterPad outputs[]
int format
agreed upon media format
A list of supported channel layouts.
unsigned nb_inputcontrols
#define AV_LOG_INFO
Standard information.
AVSampleFormat
Audio sample formats.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
LV2_Atom_Sequence * seq_out
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
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
static void uri_table_destroy(URITable *table)
Describe the class of an AVClass context structure.
int sample_rate
Sample rate of the audio data.
Rational number (pair of numerator and denominator).
const char * name
Filter name.
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
static LV2_URID uri_table_map(LV2_URID_Map_Handle handle, const char *uri)
#define flags(name, subs,...)
int(* filter_frame)(AVFilterLink *link, AVFrame *frame)
Filtering callback.
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()...
LV2_Atom_Sequence seq_in[2]
const OptionDef options[]
static int config_output(AVFilterLink *outlink)
uint64_t channel_layout
channel layout of current buffer (see libavutil/channel_layout.h)
static const LV2_Feature buf_size_features[3]
int channels
Number of channels.
AVFilterFormatsConfig incfg
Lists of supported formats / etc.
LilvNode * powerOf2BlockLength
const LilvPlugin * plugin
AVFilterContext * dst
dest filter
AVFILTER_DEFINE_CLASS(lv2)
LilvNode * lv2_OutputPort
static enum AVSampleFormat sample_fmts[]
LV2_Feature unmap_feature
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.
static const char * uri_table_unmap(LV2_URID_Map_Handle handle, LV2_URID urid)
static double val(void *priv, double ch)
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.
static av_cold int init(AVFilterContext *ctx)
static int ff_insert_inpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new input pad for the filter.