gstreamer/ext/lv2/gstlv2source.c
Stefan Sauer a1bf2e17cc lv2: support CVPorts
CVPorts are ports that take a buffer. For now we just fill the buffers with
the control value.
2016-05-18 21:33:43 -07:00

666 lines
19 KiB
C

/* GStreamer
* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
* 2001 Steve Baker <stevebaker_org@yahoo.co.uk>
* 2003 Andy Wingo <wingo at pobox.com>
* 2016 Stefan Sauer <ensonic@users.sf.net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstlv2.h"
#include "gstlv2utils.h"
#include <string.h>
#include <math.h>
#include <glib.h>
#include <lilv/lilv.h>
#include <gst/audio/audio.h>
#include <gst/audio/audio-channels.h>
#include <gst/base/gstbasesrc.h>
GST_DEBUG_CATEGORY_EXTERN (lv2_debug);
#define GST_CAT_DEFAULT lv2_debug
typedef struct _GstLV2Source GstLV2Source;
typedef struct _GstLV2SourceClass GstLV2SourceClass;
struct _GstLV2Source
{
GstBaseSrc parent;
GstLV2 lv2;
/* audio parameters */
GstAudioInfo info;
gint samples_per_buffer;
/*< private > */
gboolean tags_pushed; /* send tags just once ? */
GstClockTimeDiff timestamp_offset; /* base offset */
GstClockTime next_time; /* next timestamp */
gint64 next_sample; /* next sample to send */
gint64 next_byte; /* next byte to send */
gint64 sample_stop;
gboolean check_seek_stop;
gboolean eos_reached;
gint generate_samples_per_buffer; /* used to generate a partial buffer */
gboolean can_activate_pull;
gboolean reverse; /* play backwards */
};
struct _GstLV2SourceClass
{
GstBaseSrcClass parent_class;
GstLV2Class lv2;
};
enum
{
GST_LV2_SOURCE_PROP_0,
GST_LV2_SOURCE_PROP_SAMPLES_PER_BUFFER,
GST_LV2_SOURCE_PROP_IS_LIVE,
GST_LV2_SOURCE_PROP_TIMESTAMP_OFFSET,
GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PUSH,
GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PULL,
GST_LV2_SOURCE_PROP_LAST
};
static GstBaseSrc *parent_class = NULL;
/* GstBasesrc vmethods implementation */
static gboolean
gst_lv2_source_set_caps (GstBaseSrc * base, GstCaps * caps)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps)) {
GST_ERROR_OBJECT (base, "received invalid caps");
return FALSE;
}
GST_DEBUG_OBJECT (lv2, "negotiated to caps %" GST_PTR_FORMAT, caps);
lv2->info = info;
gst_base_src_set_blocksize (base,
GST_AUDIO_INFO_BPF (&info) * lv2->samples_per_buffer);
if (!gst_lv2_setup (&lv2->lv2, GST_AUDIO_INFO_RATE (&info)))
goto no_instance;
return TRUE;
no_instance:
{
GST_ERROR_OBJECT (lv2, "could not create instance");
return FALSE;
}
}
static GstCaps *
gst_lv2_source_fixate (GstBaseSrc * base, GstCaps * caps)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
GstStructure *structure;
caps = gst_caps_make_writable (caps);
structure = gst_caps_get_structure (caps, 0);
GST_DEBUG_OBJECT (lv2, "fixating samplerate to %d", GST_AUDIO_DEF_RATE);
gst_structure_fixate_field_nearest_int (structure, "rate",
GST_AUDIO_DEF_RATE);
gst_structure_fixate_field_string (structure, "format", GST_AUDIO_NE (F32));
gst_structure_fixate_field_nearest_int (structure, "channels",
lv2->lv2.klass->out_group.ports->len);
caps = GST_BASE_SRC_CLASS (parent_class)->fixate (base, caps);
return caps;
}
static void
gst_lv2_source_get_times (GstBaseSrc * base, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
/* for live sources, sync on the timestamp of the buffer */
if (gst_base_src_is_live (base)) {
GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* get duration to calculate end time */
GstClockTime duration = GST_BUFFER_DURATION (buffer);
if (GST_CLOCK_TIME_IS_VALID (duration)) {
*end = timestamp + duration;
}
*start = timestamp;
}
} else {
*start = -1;
*end = -1;
}
}
/* seek to time, will be called when we operate in push mode. In pull mode we
* get the requested byte offset. */
static gboolean
gst_lv2_source_do_seek (GstBaseSrc * base, GstSegment * segment)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
GstClockTime time;
gint samplerate, bpf;
gint64 next_sample;
GST_DEBUG_OBJECT (lv2, "seeking %" GST_SEGMENT_FORMAT, segment);
time = segment->position;
lv2->reverse = (segment->rate < 0.0);
samplerate = GST_AUDIO_INFO_RATE (&lv2->info);
bpf = GST_AUDIO_INFO_BPF (&lv2->info);
/* now move to the time indicated, don't seek to the sample *after* the time */
next_sample = gst_util_uint64_scale_int (time, samplerate, GST_SECOND);
lv2->next_byte = next_sample * bpf;
if (samplerate == 0)
lv2->next_time = 0;
else
lv2->next_time =
gst_util_uint64_scale_round (next_sample, GST_SECOND, samplerate);
GST_DEBUG_OBJECT (lv2, "seeking next_sample=%" G_GINT64_FORMAT
" next_time=%" GST_TIME_FORMAT, next_sample,
GST_TIME_ARGS (lv2->next_time));
g_assert (lv2->next_time <= time);
lv2->next_sample = next_sample;
if (!lv2->reverse) {
if (GST_CLOCK_TIME_IS_VALID (segment->start)) {
segment->time = segment->start;
}
} else {
if (GST_CLOCK_TIME_IS_VALID (segment->stop)) {
segment->time = segment->stop;
}
}
if (GST_CLOCK_TIME_IS_VALID (segment->stop)) {
time = segment->stop;
lv2->sample_stop =
gst_util_uint64_scale_round (time, samplerate, GST_SECOND);
lv2->check_seek_stop = TRUE;
} else {
lv2->check_seek_stop = FALSE;
}
lv2->eos_reached = FALSE;
return TRUE;
}
static gboolean
gst_lv2_source_is_seekable (GstBaseSrc * base)
{
/* we're seekable... */
return TRUE;
}
static gboolean
gst_lv2_source_query (GstBaseSrc * base, GstQuery * query)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
if (!gst_audio_info_convert (&lv2->info, src_fmt, src_val, dest_fmt,
&dest_val)) {
GST_DEBUG_OBJECT (lv2, "query failed");
return FALSE;
}
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
res = TRUE;
break;
}
case GST_QUERY_SCHEDULING:
{
/* if we can operate in pull mode */
gst_query_set_scheduling (query, GST_SCHEDULING_FLAG_SEEKABLE, 1, -1, 0);
gst_query_add_scheduling_mode (query, GST_PAD_MODE_PUSH);
if (lv2->can_activate_pull)
gst_query_add_scheduling_mode (query, GST_PAD_MODE_PULL);
res = TRUE;
break;
}
default:
res = GST_BASE_SRC_CLASS (parent_class)->query (base, query);
break;
}
return res;
}
static inline void
gst_lv2_source_interleave_data (guint n_channels, gfloat * outdata,
guint samples, gfloat * indata)
{
guint i, j;
for (i = 0; i < n_channels; i++)
for (j = 0; j < samples; j++) {
outdata[j * n_channels + i] = indata[i * samples + j];
}
}
static GstFlowReturn
gst_lv2_source_fill (GstBaseSrc * base, guint64 offset,
guint length, GstBuffer * buffer)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
GstLV2SourceClass *klass = (GstLV2SourceClass *) GST_BASE_SRC_GET_CLASS (lv2);
GstLV2Class *lv2_class = &klass->lv2;
GstLV2Group *lv2_group;
GstLV2Port *lv2_port;
GstClockTime next_time;
gint64 next_sample, next_byte;
guint bytes, samples;
GstElementClass *eclass;
GstMapInfo map;
gint samplerate, bpf;
guint j, k, l;
gfloat *out = NULL, *cv = NULL, *mem;
gfloat val;
/* example for tagging generated data */
if (!lv2->tags_pushed) {
GstTagList *taglist;
taglist = gst_tag_list_new (GST_TAG_DESCRIPTION, "lv2 wave", NULL);
eclass = GST_ELEMENT_CLASS (parent_class);
if (eclass->send_event)
eclass->send_event (GST_ELEMENT (base), gst_event_new_tag (taglist));
else
gst_tag_list_unref (taglist);
lv2->tags_pushed = TRUE;
}
if (lv2->eos_reached) {
GST_INFO_OBJECT (lv2, "eos");
return GST_FLOW_EOS;
}
samplerate = GST_AUDIO_INFO_RATE (&lv2->info);
bpf = GST_AUDIO_INFO_BPF (&lv2->info);
/* if no length was given, use our default length in samples otherwise convert
* the length in bytes to samples. */
if (length == -1)
samples = lv2->samples_per_buffer;
else
samples = length / bpf;
/* if no offset was given, use our next logical byte */
if (offset == -1)
offset = lv2->next_byte;
/* now see if we are at the byteoffset we think we are */
if (offset != lv2->next_byte) {
GST_DEBUG_OBJECT (lv2, "seek to new offset %" G_GUINT64_FORMAT, offset);
/* we have a discont in the expected sample offset, do a 'seek' */
lv2->next_sample = offset / bpf;
lv2->next_time =
gst_util_uint64_scale_int (lv2->next_sample, GST_SECOND, samplerate);
lv2->next_byte = offset;
}
/* check for eos */
if (lv2->check_seek_stop &&
(lv2->sample_stop > lv2->next_sample) &&
(lv2->sample_stop < lv2->next_sample + samples)
) {
/* calculate only partial buffer */
lv2->generate_samples_per_buffer = lv2->sample_stop - lv2->next_sample;
next_sample = lv2->sample_stop;
lv2->eos_reached = TRUE;
GST_INFO_OBJECT (lv2, "eos reached");
} else {
/* calculate full buffer */
lv2->generate_samples_per_buffer = samples;
next_sample = lv2->next_sample + (lv2->reverse ? (-samples) : samples);
}
bytes = lv2->generate_samples_per_buffer * bpf;
next_byte = lv2->next_byte + (lv2->reverse ? (-bytes) : bytes);
next_time = gst_util_uint64_scale_int (next_sample, GST_SECOND, samplerate);
GST_LOG_OBJECT (lv2, "samplerate %d", samplerate);
GST_LOG_OBJECT (lv2,
"next_sample %" G_GINT64_FORMAT ", ts %" GST_TIME_FORMAT, next_sample,
GST_TIME_ARGS (next_time));
gst_buffer_set_size (buffer, bytes);
GST_BUFFER_OFFSET (buffer) = lv2->next_sample;
GST_BUFFER_OFFSET_END (buffer) = next_sample;
if (!lv2->reverse) {
GST_BUFFER_TIMESTAMP (buffer) = lv2->timestamp_offset + lv2->next_time;
GST_BUFFER_DURATION (buffer) = next_time - lv2->next_time;
} else {
GST_BUFFER_TIMESTAMP (buffer) = lv2->timestamp_offset + next_time;
GST_BUFFER_DURATION (buffer) = lv2->next_time - next_time;
}
gst_object_sync_values (GST_OBJECT (lv2), GST_BUFFER_TIMESTAMP (buffer));
lv2->next_time = next_time;
lv2->next_sample = next_sample;
lv2->next_byte = next_byte;
GST_LOG_OBJECT (lv2, "generating %u samples at ts %" GST_TIME_FORMAT,
samples, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
/* multi channel outputs */
lv2_group = &lv2_class->out_group;
if (lv2_group->ports->len > 1) {
out = g_new0 (gfloat, samples * lv2_group->ports->len);
for (j = 0; j < lv2_group->ports->len; ++j) {
lv2_port = &g_array_index (lv2_group->ports, GstLV2Port, j);
lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index,
out + (j * samples));
GST_LOG_OBJECT (lv2, "connected port %d/%d", j, lv2_group->ports->len);
}
} else {
lv2_port = &g_array_index (lv2_group->ports, GstLV2Port, 0);
lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index,
(gfloat *) map.data);
GST_LOG_OBJECT (lv2, "connected port 0");
}
/* cv ports */
cv = g_new (gfloat, samples * lv2_class->num_cv_in);
for (j = k = 0; j < lv2_class->control_in_ports->len; j++) {
lv2_port = &g_array_index (lv2_class->control_in_ports, GstLV2Port, j);
if (lv2_port->type != GST_LV2_PORT_CV)
continue;
mem = cv + (k * samples);
val = lv2->lv2.ports.control.in[j];
/* FIXME: use gst_control_binding_get_value_array */
for (l = 0; l < samples; l++)
mem[l] = val;
lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index, mem);
k++;
}
lilv_instance_run (lv2->lv2.instance, samples);
if (lv2_group->ports->len > 1) {
gst_lv2_source_interleave_data (lv2_group->ports->len,
(gfloat *) map.data, samples, out);
g_free (out);
}
g_free (cv);
gst_buffer_unmap (buffer, &map);
return GST_FLOW_OK;
}
static gboolean
gst_lv2_source_start (GstBaseSrc * base)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
lv2->next_sample = 0;
lv2->next_byte = 0;
lv2->next_time = 0;
lv2->check_seek_stop = FALSE;
lv2->eos_reached = FALSE;
lv2->tags_pushed = FALSE;
GST_INFO_OBJECT (base, "starting");
return TRUE;
}
static gboolean
gst_lv2_source_stop (GstBaseSrc * base)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
GST_INFO_OBJECT (base, "stopping");
return gst_lv2_cleanup (&lv2->lv2, (GstObject *) lv2);
}
/* GObject vmethods implementation */
static void
gst_lv2_source_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstLV2Source *self = (GstLV2Source *) object;
switch (prop_id) {
case GST_LV2_SOURCE_PROP_SAMPLES_PER_BUFFER:
self->samples_per_buffer = g_value_get_int (value);
gst_base_src_set_blocksize (GST_BASE_SRC (self),
GST_AUDIO_INFO_BPF (&self->info) * self->samples_per_buffer);
break;
case GST_LV2_SOURCE_PROP_IS_LIVE:
gst_base_src_set_live (GST_BASE_SRC (self), g_value_get_boolean (value));
break;
case GST_LV2_SOURCE_PROP_TIMESTAMP_OFFSET:
self->timestamp_offset = g_value_get_int64 (value);
break;
case GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PUSH:
GST_BASE_SRC (self)->can_activate_push = g_value_get_boolean (value);
break;
case GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PULL:
self->can_activate_pull = g_value_get_boolean (value);
break;
default:
gst_lv2_object_set_property (&self->lv2, object, prop_id, value, pspec);
break;
}
}
static void
gst_lv2_source_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstLV2Source *self = (GstLV2Source *) object;
switch (prop_id) {
case GST_LV2_SOURCE_PROP_SAMPLES_PER_BUFFER:
g_value_set_int (value, self->samples_per_buffer);
break;
case GST_LV2_SOURCE_PROP_IS_LIVE:
g_value_set_boolean (value, gst_base_src_is_live (GST_BASE_SRC (self)));
break;
case GST_LV2_SOURCE_PROP_TIMESTAMP_OFFSET:
g_value_set_int64 (value, self->timestamp_offset);
break;
case GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PUSH:
g_value_set_boolean (value, GST_BASE_SRC (self)->can_activate_push);
break;
case GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PULL:
g_value_set_boolean (value, self->can_activate_pull);
break;
default:
gst_lv2_object_get_property (&self->lv2, object, prop_id, value, pspec);
break;
}
}
static void
gst_lv2_source_finalize (GObject * object)
{
GstLV2Source *self = (GstLV2Source *) object;
gst_lv2_finalize (&self->lv2);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_lv2_source_base_init (gpointer g_class)
{
GstLV2SourceClass *klass = (GstLV2SourceClass *) g_class;
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstPadTemplate *pad_template;
GstCaps *srccaps;
gst_lv2_class_init (&klass->lv2, G_TYPE_FROM_CLASS (klass));
gst_lv2_element_class_set_metadata (&klass->lv2, element_class,
"Source/Audio/LV2");
srccaps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (F32),
"channels", G_TYPE_INT, klass->lv2.out_group.ports->len,
"rate", GST_TYPE_INT_RANGE, 1, G_MAXINT,
"layout", G_TYPE_STRING, "interleaved", NULL);
pad_template =
gst_pad_template_new (GST_BASE_TRANSFORM_SRC_NAME, GST_PAD_SRC,
GST_PAD_ALWAYS, srccaps);
gst_element_class_add_pad_template (element_class, pad_template);
gst_caps_unref (srccaps);
}
static void
gst_lv2_source_base_finalize (GstLV2SourceClass * lv2_class)
{
gst_lv2_class_finalize (&lv2_class->lv2);
}
static void
gst_lv2_source_class_init (GstLV2SourceClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstBaseSrcClass *src_class = (GstBaseSrcClass *) klass;
GST_DEBUG ("class_init %p", klass);
gobject_class->set_property = gst_lv2_source_set_property;
gobject_class->get_property = gst_lv2_source_get_property;
gobject_class->finalize = gst_lv2_source_finalize;
src_class->set_caps = gst_lv2_source_set_caps;
src_class->fixate = gst_lv2_source_fixate;
src_class->is_seekable = gst_lv2_source_is_seekable;
src_class->do_seek = gst_lv2_source_do_seek;
src_class->query = gst_lv2_source_query;
src_class->get_times = gst_lv2_source_get_times;
src_class->start = gst_lv2_source_start;
src_class->stop = gst_lv2_source_stop;
src_class->fill = gst_lv2_source_fill;
g_object_class_install_property (gobject_class,
GST_LV2_SOURCE_PROP_SAMPLES_PER_BUFFER,
g_param_spec_int ("samplesperbuffer", "Samples per buffer",
"Number of samples in each outgoing buffer", 1, G_MAXINT, 1024,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, GST_LV2_SOURCE_PROP_IS_LIVE,
g_param_spec_boolean ("is-live", "Is Live",
"Whether to act as a live source", FALSE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
GST_LV2_SOURCE_PROP_TIMESTAMP_OFFSET,
g_param_spec_int64 ("timestamp-offset", "Timestamp offset",
"An offset added to timestamps set on buffers (in ns)", G_MININT64,
G_MAXINT64, G_GINT64_CONSTANT (0),
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PUSH,
g_param_spec_boolean ("can-activate-push", "Can activate push",
"Can activate in push mode", TRUE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PULL,
g_param_spec_boolean ("can-activate-pull", "Can activate pull",
"Can activate in pull mode", FALSE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_lv2_class_install_properties (&klass->lv2, gobject_class,
GST_LV2_SOURCE_PROP_LAST);
}
static void
gst_lv2_source_init (GstLV2Source * self, GstLV2SourceClass * klass)
{
gst_lv2_init (&self->lv2, &klass->lv2);
gst_base_src_set_format (GST_BASE_SRC (self), GST_FORMAT_TIME);
gst_base_src_set_blocksize (GST_BASE_SRC (self), -1);
self->samples_per_buffer = 1024;
self->generate_samples_per_buffer = self->samples_per_buffer;
}
void
gst_lv2_source_register_element (GstPlugin * plugin, GstStructure * lv2_meta)
{
GTypeInfo info = {
sizeof (GstLV2SourceClass),
(GBaseInitFunc) gst_lv2_source_base_init,
(GBaseFinalizeFunc) gst_lv2_source_base_finalize,
(GClassInitFunc) gst_lv2_source_class_init,
NULL,
NULL,
sizeof (GstLV2Source),
0,
(GInstanceInitFunc) gst_lv2_source_init,
};
gst_lv2_register_element (plugin, GST_TYPE_BASE_SRC, &info, lv2_meta);
if (!parent_class)
parent_class = g_type_class_ref (GST_TYPE_BASE_SRC);
}