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11c74ccec6
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>
364 lines
12 KiB
C
364 lines
12 KiB
C
/*
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* GStreamer
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* Copyright (C) 2017 Collabora Inc.
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* Copyright (C) 2021 Igalia S.L.
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* Author: Philippe Normand <philn@igalia.com>
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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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/**
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* SECTION:element-fakeaudiosink
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* @title: fakeaudiosink
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*
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* This element is the same as fakesink but will pretend to act as an audio sink
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* supporting the `GstStreamVolume` interface. This is useful for throughput
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* testing while creating a new pipeline or for CI purposes on machines not
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* running a real audio daemon.
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*
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* ## Example launch lines
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* |[
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* gst-launch-1.0 audiotestsrc ! fakeaudiosink
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* ]|
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*
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* Since: 1.20
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*/
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#include "gstdebugutilsbadelements.h"
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#include "gstfakeaudiosink.h"
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#include "gstfakesinkutils.h"
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#include <gst/audio/audio.h>
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typedef enum
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{
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FAKE_SINK_STATE_ERROR_NONE = 0,
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FAKE_SINK_STATE_ERROR_NULL_READY,
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FAKE_SINK_STATE_ERROR_READY_PAUSED,
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FAKE_SINK_STATE_ERROR_PAUSED_PLAYING,
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FAKE_SINK_STATE_ERROR_PLAYING_PAUSED,
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FAKE_SINK_STATE_ERROR_PAUSED_READY,
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FAKE_SINK_STATE_ERROR_READY_NULL
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} GstFakeSinkStateError;
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#define DEFAULT_DROP_OUT_OF_SEGMENT TRUE
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#define DEFAULT_STATE_ERROR FAKE_SINK_STATE_ERROR_NONE
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#define DEFAULT_SILENT TRUE
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#define DEFAULT_DUMP FALSE
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#define DEFAULT_SIGNAL_HANDOFFS FALSE
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#define DEFAULT_LAST_MESSAGE NULL
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#define DEFAULT_CAN_ACTIVATE_PUSH TRUE
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#define DEFAULT_CAN_ACTIVATE_PULL FALSE
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#define DEFAULT_NUM_BUFFERS -1
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/**
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* GstFakeAudioSinkStateError:
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*
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* Proxy for GstFakeSinkError.
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*
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* Since: 1.22
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*/
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#define GST_TYPE_FAKE_AUDIO_SINK_STATE_ERROR (gst_fake_audio_sink_state_error_get_type())
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static GType
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gst_fake_audio_sink_state_error_get_type (void)
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{
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static GType fakeaudiosink_state_error_type = 0;
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static const GEnumValue fakeaudiosink_state_error[] = {
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{FAKE_SINK_STATE_ERROR_NONE, "No state change errors", "none"},
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{FAKE_SINK_STATE_ERROR_NULL_READY,
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"Fail state change from NULL to READY", "null-to-ready"},
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{FAKE_SINK_STATE_ERROR_READY_PAUSED,
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"Fail state change from READY to PAUSED", "ready-to-paused"},
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{FAKE_SINK_STATE_ERROR_PAUSED_PLAYING,
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"Fail state change from PAUSED to PLAYING", "paused-to-playing"},
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{FAKE_SINK_STATE_ERROR_PLAYING_PAUSED,
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"Fail state change from PLAYING to PAUSED", "playing-to-paused"},
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{FAKE_SINK_STATE_ERROR_PAUSED_READY,
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"Fail state change from PAUSED to READY", "paused-to-ready"},
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{FAKE_SINK_STATE_ERROR_READY_NULL,
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"Fail state change from READY to NULL", "ready-to-null"},
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{0, NULL, NULL},
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};
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if (!fakeaudiosink_state_error_type) {
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fakeaudiosink_state_error_type =
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g_enum_register_static ("GstFakeAudioSinkStateError",
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fakeaudiosink_state_error);
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}
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return fakeaudiosink_state_error_type;
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}
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enum
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{
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PROP_0,
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PROP_VOLUME,
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PROP_MUTE,
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PROP_STATE_ERROR,
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PROP_SILENT,
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PROP_DUMP,
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PROP_SIGNAL_HANDOFFS,
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PROP_DROP_OUT_OF_SEGMENT,
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PROP_LAST_MESSAGE,
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PROP_CAN_ACTIVATE_PUSH,
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PROP_CAN_ACTIVATE_PULL,
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PROP_NUM_BUFFERS,
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PROP_LAST
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};
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enum
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{
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SIGNAL_HANDOFF,
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SIGNAL_PREROLL_HANDOFF,
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LAST_SIGNAL
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};
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static guint gst_fake_audio_sink_signals[LAST_SIGNAL] = { 0 };
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static GParamSpec *pspec_last_message = NULL;
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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)));
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G_DEFINE_TYPE_WITH_CODE (GstFakeAudioSink, gst_fake_audio_sink, GST_TYPE_BIN,
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G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL);
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);
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GST_ELEMENT_REGISTER_DEFINE (fakeaudiosink, "fakeaudiosink",
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GST_RANK_NONE, gst_fake_audio_sink_get_type ());
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static void
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gst_fake_audio_sink_proxy_handoff (GstElement * element, GstBuffer * buffer,
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GstPad * pad, GstFakeAudioSink * self)
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{
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g_signal_emit (self, gst_fake_audio_sink_signals[SIGNAL_HANDOFF], 0,
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buffer, self->sinkpad);
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}
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static void
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gst_fake_audio_sink_proxy_preroll_handoff (GstElement * element,
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GstBuffer * buffer, GstPad * pad, GstFakeAudioSink * self)
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{
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g_signal_emit (self, gst_fake_audio_sink_signals[SIGNAL_PREROLL_HANDOFF], 0,
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buffer, self->sinkpad);
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}
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static void
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gst_fake_audio_sink_proxy_last_message (GstElement * element)
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{
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g_object_notify_by_pspec ((GObject *) element, pspec_last_message);
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}
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static void
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gst_fake_audio_sink_init (GstFakeAudioSink * self)
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{
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GstElement *child;
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GstPadTemplate *template = gst_static_pad_template_get (&sink_factory);
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self->volume = 1.0;
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self->mute = FALSE;
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child = gst_element_factory_make ("fakesink", "sink");
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if (child) {
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GstPad *sink_pad = gst_element_get_static_pad (child, "sink");
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GstPad *ghost_pad;
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/* mimic GstAudioSink base class */
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g_object_set (child, "qos", TRUE, "sync", TRUE, NULL);
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gst_bin_add (GST_BIN_CAST (self), child);
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self->sinkpad = ghost_pad =
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gst_ghost_pad_new_from_template ("sink", sink_pad, template);
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gst_object_unref (template);
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gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad);
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gst_object_unref (sink_pad);
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self->child = child;
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g_signal_connect (child, "notify::last-message",
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G_CALLBACK (gst_fake_audio_sink_proxy_last_message), self);
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g_signal_connect (child, "handoff",
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G_CALLBACK (gst_fake_audio_sink_proxy_handoff), self);
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g_signal_connect (child, "preroll-handoff",
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G_CALLBACK (gst_fake_audio_sink_proxy_preroll_handoff), self);
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} else {
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g_warning ("Check your GStreamer installation, "
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"core element 'fakesink' is missing.");
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}
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}
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static void
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gst_fake_audio_sink_get_property (GObject * object, guint property_id,
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GValue * value, GParamSpec * pspec)
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{
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GstFakeAudioSink *self = GST_FAKE_AUDIO_SINK (object);
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switch (property_id) {
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case PROP_VOLUME:
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g_value_set_double (value, self->volume);
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break;
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case PROP_MUTE:
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g_value_set_boolean (value, self->mute);
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break;
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default:
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g_object_get_property (G_OBJECT (self->child), pspec->name, value);
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break;
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}
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}
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static void
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gst_fake_audio_sink_set_property (GObject * object, guint property_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstFakeAudioSink *self = GST_FAKE_AUDIO_SINK (object);
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switch (property_id) {
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case PROP_VOLUME:
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self->volume = g_value_get_double (value);
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break;
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case PROP_MUTE:
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self->mute = g_value_get_boolean (value);
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break;
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default:
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g_object_set_property (G_OBJECT (self->child), pspec->name, value);
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break;
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}
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}
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static void
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gst_fake_audio_sink_class_init (GstFakeAudioSinkClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GObjectClass *base_sink_class;
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object_class->get_property = gst_fake_audio_sink_get_property;
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object_class->set_property = gst_fake_audio_sink_set_property;
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/**
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* GstFakeAudioSink:volume
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*
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* Control the audio volume
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*
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* Since: 1.20
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*/
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g_object_class_install_property (object_class, PROP_VOLUME,
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g_param_spec_double ("volume", "Volume", "The audio volume, 1.0=100%",
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0, 10, 1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstFakeAudioSink:mute
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*
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* Control the mute state
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*
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* Since: 1.20
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*/
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g_object_class_install_property (object_class, PROP_MUTE,
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g_param_spec_boolean ("mute", "Mute",
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"Mute the audio channel without changing the volume", FALSE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstFakeAudioSink::handoff:
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* @fakeaudiosink: the fakeaudiosink instance
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* @buffer: the buffer that just has been received
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* @pad: the pad that received it
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*
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* This signal gets emitted before unreffing the buffer.
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*
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* Since: 1.22
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*/
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gst_fake_audio_sink_signals[SIGNAL_HANDOFF] =
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g_signal_new ("handoff", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
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G_STRUCT_OFFSET (GstFakeAudioSinkClass, handoff), NULL, NULL,
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NULL, G_TYPE_NONE, 2, GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE,
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GST_TYPE_PAD);
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/**
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* GstFakeAudioSink::preroll-handoff:
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* @fakeaudiosink: the fakeaudiosink instance
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* @buffer: the buffer that just has been received
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* @pad: the pad that received it
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*
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* This signal gets emitted before unreffing the buffer.
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*
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* Since: 1.22
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*/
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gst_fake_audio_sink_signals[SIGNAL_PREROLL_HANDOFF] =
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g_signal_new ("preroll-handoff", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstFakeAudioSinkClass,
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preroll_handoff), NULL, NULL, NULL, G_TYPE_NONE, 2,
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GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE, GST_TYPE_PAD);
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g_object_class_install_property (object_class, PROP_STATE_ERROR,
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g_param_spec_enum ("state-error", "State Error",
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"Generate a state change error", GST_TYPE_FAKE_AUDIO_SINK_STATE_ERROR,
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DEFAULT_STATE_ERROR, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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pspec_last_message = g_param_spec_string ("last-message", "Last Message",
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"The message describing current status", DEFAULT_LAST_MESSAGE,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS);
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g_object_class_install_property (object_class, PROP_LAST_MESSAGE,
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pspec_last_message);
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g_object_class_install_property (object_class, PROP_SIGNAL_HANDOFFS,
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g_param_spec_boolean ("signal-handoffs", "Signal handoffs",
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"Send a signal before unreffing the buffer", DEFAULT_SIGNAL_HANDOFFS,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (object_class, PROP_DROP_OUT_OF_SEGMENT,
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g_param_spec_boolean ("drop-out-of-segment",
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"Drop out-of-segment buffers",
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"Drop and don't render / hand off out-of-segment buffers",
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DEFAULT_DROP_OUT_OF_SEGMENT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (object_class, PROP_SILENT,
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g_param_spec_boolean ("silent", "Silent",
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"Don't produce last_message events", DEFAULT_SILENT,
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G_PARAM_READWRITE | GST_PARAM_MUTABLE_PLAYING |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (object_class, PROP_DUMP,
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g_param_spec_boolean ("dump", "Dump", "Dump buffer contents to stdout",
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DEFAULT_DUMP,
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G_PARAM_READWRITE | GST_PARAM_MUTABLE_PLAYING |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (object_class, PROP_CAN_ACTIVATE_PUSH,
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g_param_spec_boolean ("can-activate-push", "Can activate push",
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"Can activate in push mode", DEFAULT_CAN_ACTIVATE_PUSH,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (object_class, PROP_CAN_ACTIVATE_PULL,
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g_param_spec_boolean ("can-activate-pull", "Can activate pull",
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"Can activate in pull mode", DEFAULT_CAN_ACTIVATE_PULL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (object_class, PROP_NUM_BUFFERS,
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g_param_spec_int ("num-buffers", "num-buffers",
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"Number of buffers to accept going EOS", -1, G_MAXINT,
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DEFAULT_NUM_BUFFERS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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base_sink_class = g_type_class_ref (GST_TYPE_BASE_SINK);
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gst_util_proxy_class_properties (object_class, base_sink_class, PROP_LAST);
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g_type_class_unref (base_sink_class);
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gst_element_class_add_static_pad_template (element_class, &sink_factory);
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gst_element_class_set_static_metadata (element_class, "Fake Audio Sink",
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"Audio/Sink", "Fake audio renderer",
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"Philippe Normand <philn@igalia.com>");
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gst_type_mark_as_plugin_api (GST_TYPE_FAKE_AUDIO_SINK_STATE_ERROR, 0);
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}
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