gstreamer/subprojects/gst-plugins-bad/gst/debugutils/gstfakeaudiosink.c
Mathieu Duponchelle 11c74ccec6 fake{video|audio}sink: don't proxy properties at instance init.
Instead proxy properties from the GstBaseSink class at class_init time,
and duplicate the rest of the fakesink properties manually.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1442

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3073>
2022-10-19 09:12:11 +00:00

364 lines
12 KiB
C

/*
* GStreamer
* Copyright (C) 2017 Collabora Inc.
* Copyright (C) 2021 Igalia S.L.
* Author: Philippe Normand <philn@igalia.com>
*
* 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.
*/
/**
* SECTION:element-fakeaudiosink
* @title: fakeaudiosink
*
* This element is the same as fakesink but will pretend to act as an audio sink
* supporting the `GstStreamVolume` interface. This is useful for throughput
* testing while creating a new pipeline or for CI purposes on machines not
* running a real audio daemon.
*
* ## Example launch lines
* |[
* gst-launch-1.0 audiotestsrc ! fakeaudiosink
* ]|
*
* Since: 1.20
*/
#include "gstdebugutilsbadelements.h"
#include "gstfakeaudiosink.h"
#include "gstfakesinkutils.h"
#include <gst/audio/audio.h>
typedef enum
{
FAKE_SINK_STATE_ERROR_NONE = 0,
FAKE_SINK_STATE_ERROR_NULL_READY,
FAKE_SINK_STATE_ERROR_READY_PAUSED,
FAKE_SINK_STATE_ERROR_PAUSED_PLAYING,
FAKE_SINK_STATE_ERROR_PLAYING_PAUSED,
FAKE_SINK_STATE_ERROR_PAUSED_READY,
FAKE_SINK_STATE_ERROR_READY_NULL
} GstFakeSinkStateError;
#define DEFAULT_DROP_OUT_OF_SEGMENT TRUE
#define DEFAULT_STATE_ERROR FAKE_SINK_STATE_ERROR_NONE
#define DEFAULT_SILENT TRUE
#define DEFAULT_DUMP FALSE
#define DEFAULT_SIGNAL_HANDOFFS FALSE
#define DEFAULT_LAST_MESSAGE NULL
#define DEFAULT_CAN_ACTIVATE_PUSH TRUE
#define DEFAULT_CAN_ACTIVATE_PULL FALSE
#define DEFAULT_NUM_BUFFERS -1
/**
* GstFakeAudioSinkStateError:
*
* Proxy for GstFakeSinkError.
*
* Since: 1.22
*/
#define GST_TYPE_FAKE_AUDIO_SINK_STATE_ERROR (gst_fake_audio_sink_state_error_get_type())
static GType
gst_fake_audio_sink_state_error_get_type (void)
{
static GType fakeaudiosink_state_error_type = 0;
static const GEnumValue fakeaudiosink_state_error[] = {
{FAKE_SINK_STATE_ERROR_NONE, "No state change errors", "none"},
{FAKE_SINK_STATE_ERROR_NULL_READY,
"Fail state change from NULL to READY", "null-to-ready"},
{FAKE_SINK_STATE_ERROR_READY_PAUSED,
"Fail state change from READY to PAUSED", "ready-to-paused"},
{FAKE_SINK_STATE_ERROR_PAUSED_PLAYING,
"Fail state change from PAUSED to PLAYING", "paused-to-playing"},
{FAKE_SINK_STATE_ERROR_PLAYING_PAUSED,
"Fail state change from PLAYING to PAUSED", "playing-to-paused"},
{FAKE_SINK_STATE_ERROR_PAUSED_READY,
"Fail state change from PAUSED to READY", "paused-to-ready"},
{FAKE_SINK_STATE_ERROR_READY_NULL,
"Fail state change from READY to NULL", "ready-to-null"},
{0, NULL, NULL},
};
if (!fakeaudiosink_state_error_type) {
fakeaudiosink_state_error_type =
g_enum_register_static ("GstFakeAudioSinkStateError",
fakeaudiosink_state_error);
}
return fakeaudiosink_state_error_type;
}
enum
{
PROP_0,
PROP_VOLUME,
PROP_MUTE,
PROP_STATE_ERROR,
PROP_SILENT,
PROP_DUMP,
PROP_SIGNAL_HANDOFFS,
PROP_DROP_OUT_OF_SEGMENT,
PROP_LAST_MESSAGE,
PROP_CAN_ACTIVATE_PUSH,
PROP_CAN_ACTIVATE_PULL,
PROP_NUM_BUFFERS,
PROP_LAST
};
enum
{
SIGNAL_HANDOFF,
SIGNAL_PREROLL_HANDOFF,
LAST_SIGNAL
};
static guint gst_fake_audio_sink_signals[LAST_SIGNAL] = { 0 };
static GParamSpec *pspec_last_message = NULL;
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)));
G_DEFINE_TYPE_WITH_CODE (GstFakeAudioSink, gst_fake_audio_sink, GST_TYPE_BIN,
G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL);
);
GST_ELEMENT_REGISTER_DEFINE (fakeaudiosink, "fakeaudiosink",
GST_RANK_NONE, gst_fake_audio_sink_get_type ());
static void
gst_fake_audio_sink_proxy_handoff (GstElement * element, GstBuffer * buffer,
GstPad * pad, GstFakeAudioSink * self)
{
g_signal_emit (self, gst_fake_audio_sink_signals[SIGNAL_HANDOFF], 0,
buffer, self->sinkpad);
}
static void
gst_fake_audio_sink_proxy_preroll_handoff (GstElement * element,
GstBuffer * buffer, GstPad * pad, GstFakeAudioSink * self)
{
g_signal_emit (self, gst_fake_audio_sink_signals[SIGNAL_PREROLL_HANDOFF], 0,
buffer, self->sinkpad);
}
static void
gst_fake_audio_sink_proxy_last_message (GstElement * element)
{
g_object_notify_by_pspec ((GObject *) element, pspec_last_message);
}
static void
gst_fake_audio_sink_init (GstFakeAudioSink * self)
{
GstElement *child;
GstPadTemplate *template = gst_static_pad_template_get (&sink_factory);
self->volume = 1.0;
self->mute = FALSE;
child = gst_element_factory_make ("fakesink", "sink");
if (child) {
GstPad *sink_pad = gst_element_get_static_pad (child, "sink");
GstPad *ghost_pad;
/* mimic GstAudioSink base class */
g_object_set (child, "qos", TRUE, "sync", TRUE, NULL);
gst_bin_add (GST_BIN_CAST (self), child);
self->sinkpad = ghost_pad =
gst_ghost_pad_new_from_template ("sink", sink_pad, template);
gst_object_unref (template);
gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad);
gst_object_unref (sink_pad);
self->child = child;
g_signal_connect (child, "notify::last-message",
G_CALLBACK (gst_fake_audio_sink_proxy_last_message), self);
g_signal_connect (child, "handoff",
G_CALLBACK (gst_fake_audio_sink_proxy_handoff), self);
g_signal_connect (child, "preroll-handoff",
G_CALLBACK (gst_fake_audio_sink_proxy_preroll_handoff), self);
} else {
g_warning ("Check your GStreamer installation, "
"core element 'fakesink' is missing.");
}
}
static void
gst_fake_audio_sink_get_property (GObject * object, guint property_id,
GValue * value, GParamSpec * pspec)
{
GstFakeAudioSink *self = GST_FAKE_AUDIO_SINK (object);
switch (property_id) {
case PROP_VOLUME:
g_value_set_double (value, self->volume);
break;
case PROP_MUTE:
g_value_set_boolean (value, self->mute);
break;
default:
g_object_get_property (G_OBJECT (self->child), pspec->name, value);
break;
}
}
static void
gst_fake_audio_sink_set_property (GObject * object, guint property_id,
const GValue * value, GParamSpec * pspec)
{
GstFakeAudioSink *self = GST_FAKE_AUDIO_SINK (object);
switch (property_id) {
case PROP_VOLUME:
self->volume = g_value_get_double (value);
break;
case PROP_MUTE:
self->mute = g_value_get_boolean (value);
break;
default:
g_object_set_property (G_OBJECT (self->child), pspec->name, value);
break;
}
}
static void
gst_fake_audio_sink_class_init (GstFakeAudioSinkClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GObjectClass *base_sink_class;
object_class->get_property = gst_fake_audio_sink_get_property;
object_class->set_property = gst_fake_audio_sink_set_property;
/**
* GstFakeAudioSink:volume
*
* Control the audio volume
*
* Since: 1.20
*/
g_object_class_install_property (object_class, PROP_VOLUME,
g_param_spec_double ("volume", "Volume", "The audio volume, 1.0=100%",
0, 10, 1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstFakeAudioSink:mute
*
* Control the mute state
*
* Since: 1.20
*/
g_object_class_install_property (object_class, PROP_MUTE,
g_param_spec_boolean ("mute", "Mute",
"Mute the audio channel without changing the volume", FALSE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstFakeAudioSink::handoff:
* @fakeaudiosink: the fakeaudiosink instance
* @buffer: the buffer that just has been received
* @pad: the pad that received it
*
* This signal gets emitted before unreffing the buffer.
*
* Since: 1.22
*/
gst_fake_audio_sink_signals[SIGNAL_HANDOFF] =
g_signal_new ("handoff", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstFakeAudioSinkClass, handoff), NULL, NULL,
NULL, G_TYPE_NONE, 2, GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE,
GST_TYPE_PAD);
/**
* GstFakeAudioSink::preroll-handoff:
* @fakeaudiosink: the fakeaudiosink instance
* @buffer: the buffer that just has been received
* @pad: the pad that received it
*
* This signal gets emitted before unreffing the buffer.
*
* Since: 1.22
*/
gst_fake_audio_sink_signals[SIGNAL_PREROLL_HANDOFF] =
g_signal_new ("preroll-handoff", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstFakeAudioSinkClass,
preroll_handoff), NULL, NULL, NULL, G_TYPE_NONE, 2,
GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE, GST_TYPE_PAD);
g_object_class_install_property (object_class, PROP_STATE_ERROR,
g_param_spec_enum ("state-error", "State Error",
"Generate a state change error", GST_TYPE_FAKE_AUDIO_SINK_STATE_ERROR,
DEFAULT_STATE_ERROR, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
pspec_last_message = g_param_spec_string ("last-message", "Last Message",
"The message describing current status", DEFAULT_LAST_MESSAGE,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS);
g_object_class_install_property (object_class, PROP_LAST_MESSAGE,
pspec_last_message);
g_object_class_install_property (object_class, PROP_SIGNAL_HANDOFFS,
g_param_spec_boolean ("signal-handoffs", "Signal handoffs",
"Send a signal before unreffing the buffer", DEFAULT_SIGNAL_HANDOFFS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (object_class, PROP_DROP_OUT_OF_SEGMENT,
g_param_spec_boolean ("drop-out-of-segment",
"Drop out-of-segment buffers",
"Drop and don't render / hand off out-of-segment buffers",
DEFAULT_DROP_OUT_OF_SEGMENT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (object_class, PROP_SILENT,
g_param_spec_boolean ("silent", "Silent",
"Don't produce last_message events", DEFAULT_SILENT,
G_PARAM_READWRITE | GST_PARAM_MUTABLE_PLAYING |
G_PARAM_STATIC_STRINGS));
g_object_class_install_property (object_class, PROP_DUMP,
g_param_spec_boolean ("dump", "Dump", "Dump buffer contents to stdout",
DEFAULT_DUMP,
G_PARAM_READWRITE | GST_PARAM_MUTABLE_PLAYING |
G_PARAM_STATIC_STRINGS));
g_object_class_install_property (object_class, PROP_CAN_ACTIVATE_PUSH,
g_param_spec_boolean ("can-activate-push", "Can activate push",
"Can activate in push mode", DEFAULT_CAN_ACTIVATE_PUSH,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (object_class, PROP_CAN_ACTIVATE_PULL,
g_param_spec_boolean ("can-activate-pull", "Can activate pull",
"Can activate in pull mode", DEFAULT_CAN_ACTIVATE_PULL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (object_class, PROP_NUM_BUFFERS,
g_param_spec_int ("num-buffers", "num-buffers",
"Number of buffers to accept going EOS", -1, G_MAXINT,
DEFAULT_NUM_BUFFERS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
base_sink_class = g_type_class_ref (GST_TYPE_BASE_SINK);
gst_util_proxy_class_properties (object_class, base_sink_class, PROP_LAST);
g_type_class_unref (base_sink_class);
gst_element_class_add_static_pad_template (element_class, &sink_factory);
gst_element_class_set_static_metadata (element_class, "Fake Audio Sink",
"Audio/Sink", "Fake audio renderer",
"Philippe Normand <philn@igalia.com>");
gst_type_mark_as_plugin_api (GST_TYPE_FAKE_AUDIO_SINK_STATE_ERROR, 0);
}