[FFmpeg-cvslog] avfilter: add lv2 wrapper filter
Paul B Mahol
git at videolan.org
Sun Nov 26 15:05:42 EET 2017
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Thu Nov 23 16:08:42 2017 +0100| [ffc01280be6316ae625972a0976fef077a3a0b51] | committer: Paul B Mahol
avfilter: add lv2 wrapper filter
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=ffc01280be6316ae625972a0976fef077a3a0b51
---
Changelog | 1 +
configure | 4 +
doc/filters.texi | 59 +++++
libavfilter/Makefile | 1 +
libavfilter/af_lv2.c | 602 +++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
7 files changed, 669 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 0b7e02392d..e3092e211f 100644
--- a/Changelog
+++ b/Changelog
@@ -20,6 +20,7 @@ version <next>:
- OpenCL overlay filter
- video mix filter
- video normalize filter
+- audio lv2 wrapper filter
version 3.4:
diff --git a/configure b/configure
index 7769427ffb..0cc97ebbbb 100755
--- a/configure
+++ b/configure
@@ -283,6 +283,7 @@ External library support:
--enable-libzimg enable z.lib, needed for zscale filter [no]
--enable-libzmq enable message passing via libzmq [no]
--enable-libzvbi enable teletext support via libzvbi [no]
+ --enable-lv2 enable LV2 audio filtering [no]
--disable-lzma disable lzma [autodetect]
--enable-decklink enable Blackmagic DeckLink I/O support [no]
--enable-libndi_newtek enable Newteck NDI I/O support [no]
@@ -1631,6 +1632,7 @@ EXTERNAL_LIBRARY_LIST="
libzimg
libzmq
libzvbi
+ lv2
mediacodec
openal
opengl
@@ -3229,6 +3231,7 @@ hqdn3d_filter_deps="gpl"
interlace_filter_deps="gpl"
kerndeint_filter_deps="gpl"
ladspa_filter_deps="ladspa libdl"
+lv2_filter_deps="lv2"
mcdeint_filter_deps="avcodec gpl"
movie_filter_deps="avcodec avformat"
mpdecimate_filter_deps="gpl"
@@ -5825,6 +5828,7 @@ enabled gmp && require gmp gmp.h mpz_export -lgmp
enabled gnutls && require_pkg_config gnutls gnutls gnutls/gnutls.h gnutls_global_init
enabled jni && { [ $target_os = "android" ] && check_header jni.h && enabled pthreads || die "ERROR: jni not found"; }
enabled ladspa && require_header ladspa.h
+enabled lv2 && require_pkg_config lv2 lilv-0 "lilv-0/lilv/lilv.h" lilv_world_new
enabled libiec61883 && require libiec61883 libiec61883/iec61883.h iec61883_cmp_connect -lraw1394 -lavc1394 -lrom1394 -liec61883
enabled libass && require_pkg_config libass libass ass/ass.h ass_library_init
enabled libbluray && require_pkg_config libbluray libbluray libbluray/bluray.h bd_open
diff --git a/doc/filters.texi b/doc/filters.texi
index fda789630b..476f014ac8 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3281,6 +3281,65 @@ lowpass=c=LFE
@end example
@end itemize
+ at section lv2
+
+Load a LV2 (LADSPA Version 2) plugin.
+
+To enable compilation of this filter you need to configure FFmpeg with
+ at code{--enable-lv2}.
+
+ at table @option
+ at item plugin, p
+Specifies the plugin URI. You may need to escape ':'.
+
+ at item controls, c
+Set the '|' separated list of controls which are zero or more floating point
+values that determine the behavior of the loaded plugin (for example delay,
+threshold or gain).
+If @option{controls} is set to @code{help}, all available controls and
+their valid ranges are printed.
+
+ at item sample_rate, s
+Specify the sample rate, default to 44100. Only used if plugin have
+zero inputs.
+
+ at item nb_samples, n
+Set the number of samples per channel per each output frame, default
+is 1024. Only used if plugin have zero inputs.
+
+ at item duration, d
+Set the minimum duration of the sourced audio. See
+ at ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
+for the accepted syntax.
+Note that the resulting duration may be greater than the specified duration,
+as the generated audio is always cut at the end of a complete frame.
+If not specified, or the expressed duration is negative, the audio is
+supposed to be generated forever.
+Only used if plugin have zero inputs.
+ at end table
+
+ at subsection Examples
+
+ at itemize
+ at item
+Apply bass enhancer plugin from Calf:
+ at example
+lv2=p=http\\\\://calf.sourceforge.net/plugins/BassEnhancer:c=amount=2
+ at end example
+
+ at item
+Apply bass vinyl plugin from Calf:
+ at example
+lv2=p=http\\\\://calf.sourceforge.net/plugins/Vinyl:c=drone=0.2|aging=0.5
+ at end example
+
+ at item
+Apply bit crusher plugin from ArtyFX:
+ at example
+lv2=p=http\\\\://www.openavproductions.com/artyfx#bitta:c=crush=0.3
+ at end example
+ at end itemize
+
@section mcompand
Multiband Compress or expand the audio's dynamic range.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index dd195d2538..0b77d7a01f 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -101,6 +101,7 @@ OBJS-$(CONFIG_JOIN_FILTER) += af_join.o
OBJS-$(CONFIG_LADSPA_FILTER) += af_ladspa.o
OBJS-$(CONFIG_LOUDNORM_FILTER) += af_loudnorm.o ebur128.o
OBJS-$(CONFIG_LOWPASS_FILTER) += af_biquads.o
+OBJS-$(CONFIG_LV2_FILTER) += af_lv2.o
OBJS-$(CONFIG_MCOMPAND_FILTER) += af_mcompand.o
OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
OBJS-$(CONFIG_REPLAYGAIN_FILTER) += af_replaygain.o
diff --git a/libavfilter/af_lv2.c b/libavfilter/af_lv2.c
new file mode 100644
index 0000000000..8a0a6fd888
--- /dev/null
+++ b/libavfilter/af_lv2.c
@@ -0,0 +1,602 @@
+/*
+ * Copyright (c) 2017 Paul B Mahol
+ * Copyright (c) 2007-2016 David Robillard <http://drobilla.net>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * LV2 wrapper
+ */
+
+#include <lilv/lilv.h>
+#include <lv2/lv2plug.in/ns/ext/atom/atom.h>
+#include <lv2/lv2plug.in/ns/ext/buf-size/buf-size.h>
+
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct URITable {
+ char **uris;
+ size_t n_uris;
+} URITable;
+
+typedef struct LV2Context {
+ const AVClass *class;
+ char *plugin_uri;
+ char *options;
+
+ unsigned nb_inputs;
+ unsigned nb_inputcontrols;
+ unsigned nb_outputs;
+
+ int sample_rate;
+ int nb_samples;
+ int64_t pts;
+ int64_t duration;
+
+ LilvWorld *world;
+ const LilvPlugin *plugin;
+ uint32_t nb_ports;
+ float *values;
+ URITable uri_table;
+ LV2_URID_Map map;
+ LV2_Feature map_feature;
+ LV2_URID_Unmap unmap;
+ LV2_Feature unmap_feature;
+ LV2_Atom_Sequence seq_in[2];
+ LV2_Atom_Sequence *seq_out;
+ const LV2_Feature *features[5];
+
+ float *mins;
+ float *maxes;
+ float *controls;
+
+ LilvInstance *instance;
+
+ LilvNode *atom_AtomPort;
+ LilvNode *atom_Sequence;
+ LilvNode *lv2_AudioPort;
+ LilvNode *lv2_CVPort;
+ LilvNode *lv2_ControlPort;
+ LilvNode *lv2_Optional;
+ LilvNode *lv2_InputPort;
+ LilvNode *lv2_OutputPort;
+ LilvNode *urid_map;
+ LilvNode *powerOf2BlockLength;
+ LilvNode *fixedBlockLength;
+ LilvNode *boundedBlockLength;
+} LV2Context;
+
+#define OFFSET(x) offsetof(LV2Context, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption lv2_options[] = {
+ { "plugin", "set plugin uri", OFFSET(plugin_uri), AV_OPT_TYPE_STRING, .flags = FLAGS },
+ { "p", "set plugin uri", OFFSET(plugin_uri), AV_OPT_TYPE_STRING, .flags = FLAGS },
+ { "controls", "set plugin options", OFFSET(options), AV_OPT_TYPE_STRING, .flags = FLAGS },
+ { "c", "set plugin options", OFFSET(options), AV_OPT_TYPE_STRING, .flags = FLAGS },
+ { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT32_MAX, FLAGS },
+ { "s", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT32_MAX, FLAGS },
+ { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
+ { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
+ { "duration", "set audio duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64=-1}, -1, INT64_MAX, FLAGS },
+ { "d", "set audio duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64=-1}, -1, INT64_MAX, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(lv2);
+
+static void uri_table_init(URITable *table)
+{
+ table->uris = NULL;
+ table->n_uris = 0;
+}
+
+static void uri_table_destroy(URITable *table)
+{
+ int i;
+
+ for (i = 0; i < table->n_uris; i++) {
+ av_freep(&table->uris[i]);
+ }
+
+ av_freep(&table->uris);
+}
+
+static LV2_URID uri_table_map(LV2_URID_Map_Handle handle, const char *uri)
+{
+ URITable *table = (URITable*)handle;
+ const size_t len = strlen(uri);
+ size_t i;
+ char **tmp;
+
+ for (i = 0; i < table->n_uris; i++) {
+ if (!strcmp(table->uris[i], uri)) {
+ return i + 1;
+ }
+ }
+
+ tmp = av_calloc(table->n_uris + 1, sizeof(char*));
+ if (!tmp)
+ return table->n_uris;
+ memcpy(tmp, table->uris, table->n_uris * sizeof(char**));
+
+ av_free(table->uris);
+ table->uris = tmp;
+ table->uris[table->n_uris] = av_malloc(len + 1);
+ if (!table->uris[table->n_uris])
+ return table->n_uris;
+
+ memcpy(table->uris[table->n_uris], uri, len + 1);
+ table->n_uris++;
+
+ return table->n_uris;
+}
+
+static const char *uri_table_unmap(LV2_URID_Map_Handle handle, LV2_URID urid)
+{
+ URITable *table = (URITable*)handle;
+
+ if (urid > 0 && urid <= table->n_uris) {
+ return table->uris[urid - 1];
+ }
+
+ return NULL;
+}
+
+static void connect_ports(LV2Context *s, AVFrame *in, AVFrame *out)
+{
+ int ich = 0, och = 0, i;
+
+ for (i = 0; i < s->nb_ports; i++) {
+ const LilvPort *port = lilv_plugin_get_port_by_index(s->plugin, i);
+
+ if (lilv_port_is_a(s->plugin, port, s->lv2_AudioPort) ||
+ lilv_port_is_a(s->plugin, port, s->lv2_CVPort)) {
+ if (lilv_port_is_a(s->plugin, port, s->lv2_InputPort)) {
+ lilv_instance_connect_port(s->instance, i, in->extended_data[ich++]);
+ } else if (lilv_port_is_a(s->plugin, port, s->lv2_OutputPort)) {
+ lilv_instance_connect_port(s->instance, i, out->extended_data[och++]);
+ } else {
+ av_log(s, AV_LOG_WARNING, "port %d neither input nor output, skipping\n", i);
+ }
+ } else if (lilv_port_is_a(s->plugin, port, s->atom_AtomPort)) {
+ if (lilv_port_is_a(s->plugin, port, s->lv2_InputPort)) {
+ lilv_instance_connect_port(s->instance, i, &s->seq_in);
+ } else {
+ lilv_instance_connect_port(s->instance, i, s->seq_out);
+ }
+ } else if (lilv_port_is_a(s->plugin, port, s->lv2_ControlPort)) {
+ lilv_instance_connect_port(s->instance, i, &s->controls[i]);
+ }
+ }
+
+ s->seq_in[0].atom.size = sizeof(LV2_Atom_Sequence_Body);
+ s->seq_in[0].atom.type = uri_table_map(&s->uri_table, LV2_ATOM__Sequence);
+ s->seq_out->atom.size = 9624;
+ s->seq_out->atom.type = uri_table_map(&s->uri_table, LV2_ATOM__Chunk);
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ LV2Context *s = ctx->priv;
+ AVFrame *out;
+
+ if (!s->nb_outputs ||
+ (av_frame_is_writable(in) && s->nb_inputs == s->nb_outputs)) {
+ out = in;
+ } else {
+ out = ff_get_audio_buffer(ctx->outputs[0], in->nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+ }
+
+ connect_ports(s, in, out);
+
+ lilv_instance_run(s->instance, in->nb_samples);
+
+ if (out != in)
+ av_frame_free(&in);
+
+ return ff_filter_frame(ctx->outputs[0], out);
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ LV2Context *s = ctx->priv;
+ AVFrame *out;
+ int64_t t;
+
+ if (ctx->nb_inputs)
+ return ff_request_frame(ctx->inputs[0]);
+
+ t = av_rescale(s->pts, AV_TIME_BASE, s->sample_rate);
+ if (s->duration >= 0 && t >= s->duration)
+ return AVERROR_EOF;
+
+ out = ff_get_audio_buffer(outlink, s->nb_samples);
+ if (!out)
+ return AVERROR(ENOMEM);
+
+ connect_ports(s, out, out);
+
+ lilv_instance_run(s->instance, out->nb_samples);
+
+ out->sample_rate = s->sample_rate;
+ out->pts = s->pts;
+ s->pts += s->nb_samples;
+
+ return ff_filter_frame(outlink, out);
+}
+
+static const LV2_Feature buf_size_features[3] = {
+ { LV2_BUF_SIZE__powerOf2BlockLength, NULL },
+ { LV2_BUF_SIZE__fixedBlockLength, NULL },
+ { LV2_BUF_SIZE__boundedBlockLength, NULL },
+};
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ LV2Context *s = ctx->priv;
+ char *p, *arg, *saveptr = NULL;
+ int i, sample_rate;
+
+ uri_table_init(&s->uri_table);
+ s->map.handle = &s->uri_table;
+ s->map.map = uri_table_map;
+ s->map_feature.URI = LV2_URID_MAP_URI;
+ s->map_feature.data = &s->map;
+ s->unmap.handle = &s->uri_table;
+ s->unmap.unmap = uri_table_unmap;
+ s->unmap_feature.URI = LV2_URID_UNMAP_URI;
+ s->unmap_feature.data = &s->unmap;
+ s->features[0] = &s->map_feature;
+ s->features[1] = &s->unmap_feature;
+ s->features[2] = &buf_size_features[0];
+ s->features[3] = &buf_size_features[1];
+ s->features[4] = &buf_size_features[2];
+
+ if (ctx->nb_inputs) {
+ AVFilterLink *inlink = ctx->inputs[0];
+
+ outlink->format = inlink->format;
+ outlink->sample_rate = sample_rate = inlink->sample_rate;
+ if (s->nb_inputs == s->nb_outputs) {
+ outlink->channel_layout = inlink->channel_layout;
+ outlink->channels = inlink->channels;
+ }
+
+ } else {
+ outlink->sample_rate = sample_rate = s->sample_rate;
+ outlink->time_base = (AVRational){1, s->sample_rate};
+ }
+
+ s->instance = lilv_plugin_instantiate(s->plugin, sample_rate, s->features);
+ if (!s->instance) {
+ av_log(s, AV_LOG_ERROR, "Failed to instantiate <%s>\n", lilv_node_as_uri(lilv_plugin_get_uri(s->plugin)));
+ return AVERROR(EINVAL);
+ }
+
+ s->mins = av_calloc(s->nb_ports, sizeof(float));
+ s->maxes = av_calloc(s->nb_ports, sizeof(float));
+ s->controls = av_calloc(s->nb_ports, sizeof(float));
+
+ if (!s->mins || !s->maxes || !s->controls)
+ return AVERROR(ENOMEM);
+
+ lilv_plugin_get_port_ranges_float(s->plugin, s->mins, s->maxes, s->controls);
+ s->seq_out = av_malloc(sizeof(LV2_Atom_Sequence) + 9624);
+ if (!s->seq_out)
+ return AVERROR(ENOMEM);
+
+ if (s->options && !strcmp(s->options, "help")) {
+ if (!s->nb_inputcontrols) {
+ av_log(ctx, AV_LOG_INFO,
+ "The '%s' plugin does not have any input controls.\n",
+ s->plugin_uri);
+ } else {
+ av_log(ctx, AV_LOG_INFO,
+ "The '%s' plugin has the following input controls:\n",
+ s->plugin_uri);
+ for (i = 0; i < s->nb_ports; i++) {
+ const LilvPort *port = lilv_plugin_get_port_by_index(s->plugin, i);
+ const LilvNode *symbol = lilv_port_get_symbol(s->plugin, port);
+ LilvNode *name = lilv_port_get_name(s->plugin, port);
+
+ if (lilv_port_is_a(s->plugin, port, s->lv2_InputPort) &&
+ lilv_port_is_a(s->plugin, port, s->lv2_ControlPort)) {
+ av_log(ctx, AV_LOG_INFO, "%s\t\t<float> (from %f to %f) (default %f)\t\t%s\n",
+ lilv_node_as_string(symbol), s->mins[i], s->maxes[i], s->controls[i],
+ lilv_node_as_string(name));
+ }
+
+ lilv_node_free(name);
+ }
+ }
+ return AVERROR_EXIT;
+ }
+
+ p = s->options;
+ while (s->options) {
+ const LilvPort *port;
+ LilvNode *sym;
+ float val;
+ char *str, *vstr;
+ int index;
+
+ if (!(arg = av_strtok(p, " |", &saveptr)))
+ break;
+ p = NULL;
+
+ vstr = strstr(arg, "=");
+ if (vstr == NULL) {
+ av_log(ctx, AV_LOG_ERROR, "Invalid syntax.\n");
+ return AVERROR(EINVAL);
+ }
+
+ vstr[0] = 0;
+ str = arg;
+ val = atof(vstr+1);
+ sym = lilv_new_string(s->world, str);
+ port = lilv_plugin_get_port_by_symbol(s->plugin, sym);
+ lilv_node_free(sym);
+ if (!port) {
+ av_log(s, AV_LOG_WARNING, "Unknown option: <%s>\n", str);
+ } else {
+ index = lilv_port_get_index(s->plugin, port);
+ s->controls[index] = val;
+ }
+ }
+
+ if (s->nb_inputs &&
+ (lilv_plugin_has_feature(s->plugin, s->powerOf2BlockLength) ||
+ lilv_plugin_has_feature(s->plugin, s->fixedBlockLength) ||
+ lilv_plugin_has_feature(s->plugin, s->boundedBlockLength))) {
+ AVFilterLink *inlink = ctx->inputs[0];
+
+ inlink->partial_buf_size = inlink->min_samples = inlink->max_samples = 4096;
+ }
+
+ return 0;
+}
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ LV2Context *s = ctx->priv;
+ const LilvPlugins *plugins;
+ const LilvPlugin *plugin;
+ AVFilterPad pad = { NULL };
+ LilvNode *uri;
+ int i;
+
+ s->world = lilv_world_new();
+ if (!s->world)
+ return AVERROR(ENOMEM);
+
+ uri = lilv_new_uri(s->world, s->plugin_uri);
+ if (!uri) {
+ av_log(s, AV_LOG_ERROR, "Invalid plugin URI <%s>\n", s->plugin_uri);
+ return AVERROR(EINVAL);
+ }
+
+ lilv_world_load_all(s->world);
+ plugins = lilv_world_get_all_plugins(s->world);
+ plugin = lilv_plugins_get_by_uri(plugins, uri);
+ lilv_node_free(uri);
+
+ if (!plugin) {
+ av_log(s, AV_LOG_ERROR, "Plugin <%s> not found\n", s->plugin_uri);
+ return AVERROR(EINVAL);
+ }
+
+ s->plugin = plugin;
+ s->nb_ports = lilv_plugin_get_num_ports(s->plugin);
+
+ s->lv2_InputPort = lilv_new_uri(s->world, LV2_CORE__InputPort);
+ s->lv2_OutputPort = lilv_new_uri(s->world, LV2_CORE__OutputPort);
+ s->lv2_AudioPort = lilv_new_uri(s->world, LV2_CORE__AudioPort);
+ s->lv2_ControlPort = lilv_new_uri(s->world, LV2_CORE__ControlPort);
+ s->lv2_Optional = lilv_new_uri(s->world, LV2_CORE__connectionOptional);
+ s->atom_AtomPort = lilv_new_uri(s->world, LV2_ATOM__AtomPort);
+ s->atom_Sequence = lilv_new_uri(s->world, LV2_ATOM__Sequence);
+ s->urid_map = lilv_new_uri(s->world, LV2_URID__map);
+ s->powerOf2BlockLength = lilv_new_uri(s->world, LV2_BUF_SIZE__powerOf2BlockLength);
+ s->fixedBlockLength = lilv_new_uri(s->world, LV2_BUF_SIZE__fixedBlockLength);
+ s->boundedBlockLength = lilv_new_uri(s->world, LV2_BUF_SIZE__boundedBlockLength);
+
+ for (i = 0; i < s->nb_ports; i++) {
+ const LilvPort *lport = lilv_plugin_get_port_by_index(s->plugin, i);
+ int is_input = 0;
+ int is_optional = 0;
+
+ is_optional = lilv_port_has_property(s->plugin, lport, s->lv2_Optional);
+
+ if (lilv_port_is_a(s->plugin, lport, s->lv2_InputPort)) {
+ is_input = 1;
+ } else if (!lilv_port_is_a(s->plugin, lport, s->lv2_OutputPort) && !is_optional) {
+ return AVERROR(EINVAL);
+ }
+
+ if (lilv_port_is_a(s->plugin, lport, s->lv2_ControlPort)) {
+ if (is_input) {
+ s->nb_inputcontrols++;
+ }
+ } else if (lilv_port_is_a(s->plugin, lport, s->lv2_AudioPort)) {
+ if (is_input) {
+ s->nb_inputs++;
+ } else {
+ s->nb_outputs++;
+ }
+ }
+ }
+
+ pad.type = AVMEDIA_TYPE_AUDIO;
+
+ if (s->nb_inputs) {
+ pad.name = av_asprintf("in0:%s:%u", s->plugin_uri, s->nb_inputs);
+ if (!pad.name)
+ return AVERROR(ENOMEM);
+
+ pad.filter_frame = filter_frame;
+ if (ff_insert_inpad(ctx, ctx->nb_inputs, &pad) < 0) {
+ av_freep(&pad.name);
+ return AVERROR(ENOMEM);
+ }
+ }
+
+ return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ LV2Context *s = ctx->priv;
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ AVFilterLink *outlink = ctx->outputs[0];
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE };
+ int ret;
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ return ret;
+
+ if (s->nb_inputs) {
+ formats = ff_all_samplerates();
+ if (!formats)
+ return AVERROR(ENOMEM);
+
+ ret = ff_set_common_samplerates(ctx, formats);
+ if (ret < 0)
+ return ret;
+ } else {
+ int sample_rates[] = { s->sample_rate, -1 };
+
+ ret = ff_set_common_samplerates(ctx, ff_make_format_list(sample_rates));
+ if (ret < 0)
+ return ret;
+ }
+
+ if (s->nb_inputs == 2 && s->nb_outputs == 2) {
+ layouts = NULL;
+ ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO);
+ if (ret < 0)
+ return ret;
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+ } else {
+ if (s->nb_inputs >= 1) {
+ AVFilterLink *inlink = ctx->inputs[0];
+ uint64_t inlayout = FF_COUNT2LAYOUT(s->nb_inputs);
+
+ layouts = NULL;
+ ret = ff_add_channel_layout(&layouts, inlayout);
+ if (ret < 0)
+ return ret;
+ ret = ff_channel_layouts_ref(layouts, &inlink->out_channel_layouts);
+ if (ret < 0)
+ return ret;
+
+ if (!s->nb_outputs) {
+ ret = ff_channel_layouts_ref(layouts, &outlink->in_channel_layouts);
+ if (ret < 0)
+ return ret;
+ }
+ }
+
+ if (s->nb_outputs >= 1) {
+ uint64_t outlayout = FF_COUNT2LAYOUT(s->nb_outputs);
+
+ layouts = NULL;
+ ret = ff_add_channel_layout(&layouts, outlayout);
+ if (ret < 0)
+ return ret;
+ ret = ff_channel_layouts_ref(layouts, &outlink->in_channel_layouts);
+ if (ret < 0)
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ LV2Context *s = ctx->priv;
+
+ lilv_node_free(s->powerOf2BlockLength);
+ lilv_node_free(s->fixedBlockLength);
+ lilv_node_free(s->boundedBlockLength);
+ lilv_node_free(s->urid_map);
+ lilv_node_free(s->atom_Sequence);
+ lilv_node_free(s->atom_AtomPort);
+ lilv_node_free(s->lv2_Optional);
+ lilv_node_free(s->lv2_ControlPort);
+ lilv_node_free(s->lv2_AudioPort);
+ lilv_node_free(s->lv2_OutputPort);
+ lilv_node_free(s->lv2_InputPort);
+ uri_table_destroy(&s->uri_table);
+ lilv_instance_free(s->instance);
+ lilv_world_free(s->world);
+ av_freep(&s->mins);
+ av_freep(&s->maxes);
+ av_freep(&s->controls);
+ av_freep(&s->seq_out);
+
+ if (ctx->nb_inputs)
+ av_freep(&ctx->input_pads[0].name);
+}
+
+static const AVFilterPad lv2_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ .request_frame = request_frame,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_lv2 = {
+ .name = "lv2",
+ .description = NULL_IF_CONFIG_SMALL("Apply LV2 effect."),
+ .priv_size = sizeof(LV2Context),
+ .priv_class = &lv2_class,
+ .init = init,
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .inputs = 0,
+ .outputs = lv2_outputs,
+ .flags = AVFILTER_FLAG_DYNAMIC_INPUTS,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index e09d841387..4c834f7381 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -112,6 +112,7 @@ static void register_all(void)
REGISTER_FILTER(LADSPA, ladspa, af);
REGISTER_FILTER(LOUDNORM, loudnorm, af);
REGISTER_FILTER(LOWPASS, lowpass, af);
+ REGISTER_FILTER(LV2, lv2, af);
REGISTER_FILTER(MCOMPAND, mcompand, af);
REGISTER_FILTER(PAN, pan, af);
REGISTER_FILTER(REPLAYGAIN, replaygain, af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index fdcf76befe..1d356a9a5a 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7
-#define LIBAVFILTER_VERSION_MINOR 4
+#define LIBAVFILTER_VERSION_MINOR 5
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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