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)) {
 
  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)
 
const char const char void * val
 
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. 
 
#define AV_CH_LAYOUT_STEREO
 
struct AVFilterChannelLayouts * in_channel_layouts
 
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 
 
int(* filter_frame)(AVFilterLink *link, AVFrame *frame)
Filtering callback. 
 
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(). 
 
struct AVFilterChannelLayouts * out_channel_layouts
 
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)
 
static const AVFilterPad inputs[]
 
#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
 
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
 
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,...)
 
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. 
 
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. 
 
uint8_t ** extended_data
pointers to the data planes/channels. 
 
static const char * uri_table_unmap(LV2_URID_Map_Handle handle, LV2_URID urid)
 
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.