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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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648 lines
19 KiB
C
648 lines
19 KiB
C
/* GStreamer
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* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
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* 2001 Steve Baker <stevebaker_org@yahoo.co.uk>
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* 2003 Andy Wingo <wingo at pobox.com>
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* 2016 Stefan Sauer <ensonic@users.sf.net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstlv2.h"
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#include "gstlv2utils.h"
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#include <string.h>
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#include <math.h>
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#include <glib.h>
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#include <lilv/lilv.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/audio-channels.h>
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#include <gst/base/gstbasesrc.h>
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GST_DEBUG_CATEGORY_EXTERN (lv2_debug);
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#define GST_CAT_DEFAULT lv2_debug
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typedef struct _GstLV2Source GstLV2Source;
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typedef struct _GstLV2SourceClass GstLV2SourceClass;
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struct _GstLV2Source
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{
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GstBaseSrc parent;
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GstLV2 lv2;
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/* audio parameters */
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GstAudioInfo info;
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gint samples_per_buffer;
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/*< private > */
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gboolean tags_pushed; /* send tags just once ? */
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GstClockTimeDiff timestamp_offset; /* base offset */
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GstClockTime next_time; /* next timestamp */
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gint64 next_sample; /* next sample to send */
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gint64 next_byte; /* next byte to send */
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gint64 sample_stop;
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gboolean check_seek_stop;
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gboolean eos_reached;
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gint generate_samples_per_buffer; /* used to generate a partial buffer */
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gboolean can_activate_pull;
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gboolean reverse; /* play backwards */
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};
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struct _GstLV2SourceClass
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{
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GstBaseSrcClass parent_class;
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GstLV2Class lv2;
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};
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enum
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{
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GST_LV2_SOURCE_PROP_0,
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GST_LV2_SOURCE_PROP_SAMPLES_PER_BUFFER,
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GST_LV2_SOURCE_PROP_IS_LIVE,
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GST_LV2_SOURCE_PROP_TIMESTAMP_OFFSET,
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GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PUSH,
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GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PULL,
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GST_LV2_SOURCE_PROP_LAST
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};
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static GstBaseSrc *parent_class = NULL;
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/* GstBasesrc vmethods implementation */
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static gboolean
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gst_lv2_source_set_caps (GstBaseSrc * base, GstCaps * caps)
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{
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GstLV2Source *lv2 = (GstLV2Source *) base;
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GstAudioInfo info;
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if (!gst_audio_info_from_caps (&info, caps)) {
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GST_ERROR_OBJECT (base, "received invalid caps");
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return FALSE;
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}
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GST_DEBUG_OBJECT (lv2, "negotiated to caps %" GST_PTR_FORMAT, caps);
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lv2->info = info;
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gst_base_src_set_blocksize (base,
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GST_AUDIO_INFO_BPF (&info) * lv2->samples_per_buffer);
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if (!gst_lv2_setup (&lv2->lv2, GST_AUDIO_INFO_RATE (&info)))
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goto no_instance;
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return TRUE;
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no_instance:
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{
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GST_ERROR_OBJECT (lv2, "could not create instance");
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return FALSE;
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}
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}
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static GstCaps *
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gst_lv2_source_fixate (GstBaseSrc * base, GstCaps * caps)
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{
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GstLV2Source *lv2 = (GstLV2Source *) base;
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GstStructure *structure;
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caps = gst_caps_make_writable (caps);
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structure = gst_caps_get_structure (caps, 0);
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GST_DEBUG_OBJECT (lv2, "fixating samplerate to %d", GST_AUDIO_DEF_RATE);
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gst_structure_fixate_field_nearest_int (structure, "rate",
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GST_AUDIO_DEF_RATE);
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gst_structure_fixate_field_string (structure, "format", GST_AUDIO_NE (F32));
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gst_structure_fixate_field_nearest_int (structure, "channels",
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lv2->lv2.klass->out_group.ports->len);
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caps = GST_BASE_SRC_CLASS (parent_class)->fixate (base, caps);
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return caps;
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}
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static void
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gst_lv2_source_get_times (GstBaseSrc * base, GstBuffer * buffer,
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GstClockTime * start, GstClockTime * end)
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{
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/* for live sources, sync on the timestamp of the buffer */
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if (gst_base_src_is_live (base)) {
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GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
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if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
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/* get duration to calculate end time */
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GstClockTime duration = GST_BUFFER_DURATION (buffer);
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if (GST_CLOCK_TIME_IS_VALID (duration)) {
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*end = timestamp + duration;
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}
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*start = timestamp;
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}
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} else {
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*start = -1;
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*end = -1;
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}
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}
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/* seek to time, will be called when we operate in push mode. In pull mode we
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* get the requested byte offset. */
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static gboolean
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gst_lv2_source_do_seek (GstBaseSrc * base, GstSegment * segment)
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{
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GstLV2Source *lv2 = (GstLV2Source *) base;
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GstClockTime time;
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gint samplerate, bpf;
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gint64 next_sample;
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GST_DEBUG_OBJECT (lv2, "seeking %" GST_SEGMENT_FORMAT, segment);
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time = segment->position;
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lv2->reverse = (segment->rate < 0.0);
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samplerate = GST_AUDIO_INFO_RATE (&lv2->info);
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bpf = GST_AUDIO_INFO_BPF (&lv2->info);
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/* now move to the time indicated, don't seek to the sample *after* the time */
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next_sample = gst_util_uint64_scale_int (time, samplerate, GST_SECOND);
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lv2->next_byte = next_sample * bpf;
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if (samplerate == 0)
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lv2->next_time = 0;
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else
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lv2->next_time =
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gst_util_uint64_scale_round (next_sample, GST_SECOND, samplerate);
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GST_DEBUG_OBJECT (lv2, "seeking next_sample=%" G_GINT64_FORMAT
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" next_time=%" GST_TIME_FORMAT, next_sample,
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GST_TIME_ARGS (lv2->next_time));
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g_assert (lv2->next_time <= time);
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lv2->next_sample = next_sample;
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if (!lv2->reverse) {
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if (GST_CLOCK_TIME_IS_VALID (segment->start)) {
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segment->time = segment->start;
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}
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} else {
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if (GST_CLOCK_TIME_IS_VALID (segment->stop)) {
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segment->time = segment->stop;
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}
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}
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if (GST_CLOCK_TIME_IS_VALID (segment->stop)) {
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time = segment->stop;
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lv2->sample_stop =
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gst_util_uint64_scale_round (time, samplerate, GST_SECOND);
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lv2->check_seek_stop = TRUE;
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} else {
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lv2->check_seek_stop = FALSE;
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}
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lv2->eos_reached = FALSE;
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return TRUE;
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}
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static gboolean
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gst_lv2_source_is_seekable (GstBaseSrc * base)
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{
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/* we're seekable... */
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return TRUE;
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}
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static gboolean
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gst_lv2_source_query (GstBaseSrc * base, GstQuery * query)
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{
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GstLV2Source *lv2 = (GstLV2Source *) base;
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gboolean res = FALSE;
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_CONVERT:
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{
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GstFormat src_fmt, dest_fmt;
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gint64 src_val, dest_val;
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gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
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if (!gst_audio_info_convert (&lv2->info, src_fmt, src_val, dest_fmt,
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&dest_val)) {
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GST_DEBUG_OBJECT (lv2, "query failed");
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return FALSE;
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}
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gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
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res = TRUE;
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break;
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}
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case GST_QUERY_SCHEDULING:
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{
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/* if we can operate in pull mode */
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gst_query_set_scheduling (query, GST_SCHEDULING_FLAG_SEEKABLE, 1, -1, 0);
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gst_query_add_scheduling_mode (query, GST_PAD_MODE_PUSH);
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if (lv2->can_activate_pull)
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gst_query_add_scheduling_mode (query, GST_PAD_MODE_PULL);
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res = TRUE;
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break;
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}
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default:
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res = GST_BASE_SRC_CLASS (parent_class)->query (base, query);
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break;
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}
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return res;
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}
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static inline void
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gst_lv2_source_interleave_data (guint n_channels, gfloat * outdata,
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guint samples, gfloat * indata)
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{
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guint i, j;
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for (i = 0; i < n_channels; i++)
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for (j = 0; j < samples; j++) {
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outdata[j * n_channels + i] = indata[i * samples + j];
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}
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}
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static GstFlowReturn
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gst_lv2_source_fill (GstBaseSrc * base, guint64 offset,
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guint length, GstBuffer * buffer)
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{
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GstLV2Source *lv2 = (GstLV2Source *) base;
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GstLV2SourceClass *lv2_class =
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(GstLV2SourceClass *) GST_BASE_SRC_GET_CLASS (lv2);
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GstLV2Group *lv2_group;
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GstLV2Port *lv2_port;
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GstClockTime next_time;
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gint64 next_sample, next_byte;
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guint bytes, samples;
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GstElementClass *eclass;
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GstMapInfo map;
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gint samplerate, bpf;
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guint j;
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gfloat *out = NULL;
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/* example for tagging generated data */
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if (!lv2->tags_pushed) {
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GstTagList *taglist;
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taglist = gst_tag_list_new (GST_TAG_DESCRIPTION, "lv2 wave", NULL);
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eclass = GST_ELEMENT_CLASS (parent_class);
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if (eclass->send_event)
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eclass->send_event (GST_ELEMENT (base), gst_event_new_tag (taglist));
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else
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gst_tag_list_unref (taglist);
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lv2->tags_pushed = TRUE;
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}
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if (lv2->eos_reached) {
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GST_INFO_OBJECT (lv2, "eos");
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return GST_FLOW_EOS;
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}
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samplerate = GST_AUDIO_INFO_RATE (&lv2->info);
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bpf = GST_AUDIO_INFO_BPF (&lv2->info);
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/* if no length was given, use our default length in samples otherwise convert
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* the length in bytes to samples. */
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if (length == -1)
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samples = lv2->samples_per_buffer;
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else
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samples = length / bpf;
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/* if no offset was given, use our next logical byte */
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if (offset == -1)
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offset = lv2->next_byte;
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/* now see if we are at the byteoffset we think we are */
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if (offset != lv2->next_byte) {
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GST_DEBUG_OBJECT (lv2, "seek to new offset %" G_GUINT64_FORMAT, offset);
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/* we have a discont in the expected sample offset, do a 'seek' */
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lv2->next_sample = offset / bpf;
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lv2->next_time =
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gst_util_uint64_scale_int (lv2->next_sample, GST_SECOND, samplerate);
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lv2->next_byte = offset;
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}
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/* check for eos */
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if (lv2->check_seek_stop &&
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(lv2->sample_stop > lv2->next_sample) &&
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(lv2->sample_stop < lv2->next_sample + samples)
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) {
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/* calculate only partial buffer */
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lv2->generate_samples_per_buffer = lv2->sample_stop - lv2->next_sample;
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next_sample = lv2->sample_stop;
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lv2->eos_reached = TRUE;
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GST_INFO_OBJECT (lv2, "eos reached");
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} else {
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/* calculate full buffer */
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lv2->generate_samples_per_buffer = samples;
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next_sample = lv2->next_sample + (lv2->reverse ? (-samples) : samples);
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}
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bytes = lv2->generate_samples_per_buffer * bpf;
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next_byte = lv2->next_byte + (lv2->reverse ? (-bytes) : bytes);
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next_time = gst_util_uint64_scale_int (next_sample, GST_SECOND, samplerate);
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GST_LOG_OBJECT (lv2, "samplerate %d", samplerate);
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GST_LOG_OBJECT (lv2,
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"next_sample %" G_GINT64_FORMAT ", ts %" GST_TIME_FORMAT, next_sample,
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GST_TIME_ARGS (next_time));
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gst_buffer_set_size (buffer, bytes);
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GST_BUFFER_OFFSET (buffer) = lv2->next_sample;
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GST_BUFFER_OFFSET_END (buffer) = next_sample;
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if (!lv2->reverse) {
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GST_BUFFER_TIMESTAMP (buffer) = lv2->timestamp_offset + lv2->next_time;
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GST_BUFFER_DURATION (buffer) = next_time - lv2->next_time;
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} else {
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GST_BUFFER_TIMESTAMP (buffer) = lv2->timestamp_offset + next_time;
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GST_BUFFER_DURATION (buffer) = lv2->next_time - next_time;
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}
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gst_object_sync_values (GST_OBJECT (lv2), GST_BUFFER_TIMESTAMP (buffer));
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lv2->next_time = next_time;
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lv2->next_sample = next_sample;
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lv2->next_byte = next_byte;
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GST_INFO_OBJECT (lv2, "generating %u samples at ts %" GST_TIME_FORMAT,
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samples, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
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gst_buffer_map (buffer, &map, GST_MAP_WRITE);
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/* multi channel outputs */
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lv2_group = &lv2_class->lv2.out_group;
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if (lv2_group->ports->len > 1) {
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out = g_new0 (gfloat, samples * lv2_group->ports->len);
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for (j = 0; j < lv2_group->ports->len; ++j) {
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lv2_port = &g_array_index (lv2_group->ports, GstLV2Port, j);
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lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index,
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out + (j * samples));
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GST_INFO_OBJECT (lv2, "connected port %d/%d", j, lv2_group->ports->len);
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}
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} else {
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lv2_port = &g_array_index (lv2_group->ports, GstLV2Port, 0);
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lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index,
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(gfloat *) map.data);
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GST_INFO_OBJECT (lv2, "connected port 0");
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}
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lilv_instance_run (lv2->lv2.instance, samples);
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if (lv2_group->ports->len > 1) {
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gst_lv2_source_interleave_data (lv2_group->ports->len,
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(gfloat *) map.data, samples, out);
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g_free (out);
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}
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gst_buffer_unmap (buffer, &map);
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return GST_FLOW_OK;
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}
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static gboolean
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gst_lv2_source_start (GstBaseSrc * base)
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{
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GstLV2Source *lv2 = (GstLV2Source *) base;
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lv2->next_sample = 0;
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lv2->next_byte = 0;
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lv2->next_time = 0;
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lv2->check_seek_stop = FALSE;
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lv2->eos_reached = FALSE;
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lv2->tags_pushed = FALSE;
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GST_INFO_OBJECT (base, "starting");
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return TRUE;
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}
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static gboolean
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gst_lv2_source_stop (GstBaseSrc * base)
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{
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GstLV2Source *lv2 = (GstLV2Source *) base;
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GST_INFO_OBJECT (base, "stopping");
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return gst_lv2_cleanup (&lv2->lv2, (GstObject *) lv2);
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}
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/* GObject vmethods implementation */
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static void
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gst_lv2_source_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstLV2Source *self = (GstLV2Source *) object;
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switch (prop_id) {
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case GST_LV2_SOURCE_PROP_SAMPLES_PER_BUFFER:
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self->samples_per_buffer = g_value_get_int (value);
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gst_base_src_set_blocksize (GST_BASE_SRC (self),
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GST_AUDIO_INFO_BPF (&self->info) * self->samples_per_buffer);
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break;
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case GST_LV2_SOURCE_PROP_IS_LIVE:
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gst_base_src_set_live (GST_BASE_SRC (self), g_value_get_boolean (value));
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break;
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case GST_LV2_SOURCE_PROP_TIMESTAMP_OFFSET:
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self->timestamp_offset = g_value_get_int64 (value);
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break;
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case GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PUSH:
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GST_BASE_SRC (self)->can_activate_push = g_value_get_boolean (value);
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break;
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case GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PULL:
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self->can_activate_pull = g_value_get_boolean (value);
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break;
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default:
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gst_lv2_object_set_property (&self->lv2, object, prop_id, value, pspec);
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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;
|
|
|
|
// FIXME: basesrc methods
|
|
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);
|
|
}
|