mopidy.audio.actor — Mopidy 3.4.1 documentation

# Source code for mopidy.audio.actor

    import logging
    import os
    import threading

    import pykka

    from mopidy import exceptions
    from mopidy.audio import tags as tags_lib
    from mopidy.audio import utils
    from mopidy.audio.constants import PlaybackState
    from mopidy.audio.listener import AudioListener
    from mopidy.internal import process
    from mopidy.internal.gi import GLib, GObject, Gst, GstPbutils

    logger = logging.getLogger(__name__)

    # This logger is only meant for debug logging of low level GStreamer info such
    # as callbacks, event, messages and direct interaction with GStreamer such as
    # set_state() on a pipeline.
    gst_logger = logging.getLogger("mopidy.audio.gst")

    _GST_PLAY_FLAGS_AUDIO = 0x02
    _GST_PLAY_FLAGS_DOWNLOAD = 0x80

    _GST_STATE_MAPPING = {
        Gst.State.PLAYING: PlaybackState.PLAYING,
        Gst.State.PAUSED: PlaybackState.PAUSED,
        Gst.State.NULL: PlaybackState.STOPPED,
    }

    # TODO: expose this as a property on audio?
    class _Appsrc:

        """Helper class for dealing with appsrc based playback."""

        def __init__(self):
            self._signals = utils.Signals()
            self.reset()

        def reset(self):
            """Reset the helper.

            Should be called whenever the source changes and we are not setting up
            a new appsrc.
            """
            self.prepare(None, None, None, None)

        def prepare(self, caps, need_data, enough_data, seek_data):
            """Store info we will need when the appsrc element gets installed."""
            self._signals.clear()
            self._source = None
            self._caps = caps
            self._need_data_callback = need_data
            self._seek_data_callback = seek_data
            self._enough_data_callback = enough_data

        def configure(self, source):
            """Configure the supplied source for use.

            Should be called whenever we get a new appsrc.
            """
            source.set_property("caps", self._caps)
            source.set_property("format", "time")
            source.set_property("stream-type", "seekable")
            source.set_property("max-bytes", 1 << 20)  # 1MB
            source.set_property("min-percent", 50)

            if self._need_data_callback:
                self._signals.connect(
                    source, "need-data", self._on_signal, self._need_data_callback
                )
            if self._seek_data_callback:
                self._signals.connect(
                    source, "seek-data", self._on_signal, self._seek_data_callback
                )
            if self._enough_data_callback:
                self._signals.connect(
                    source,
                    "enough-data",
                    self._on_signal,
                    None,
                    self._enough_data_callback,
                )

            self._source = source

        def push(self, buffer_):
            if self._source is None:
                return False

            if buffer_ is None:
                gst_logger.debug("Sending appsrc end-of-stream event.")
                result = self._source.emit("end-of-stream")
                return result == Gst.FlowReturn.OK
            else:
                result = self._source.emit("push-buffer", buffer_)
                return result == Gst.FlowReturn.OK

        def _on_signal(self, element, clocktime, func):
            # This shim is used to ensure we always return true, and also handles
            # that not all the callbacks have a time argument.
            if clocktime is None:
                func()
            else:
                func(utils.clocktime_to_millisecond(clocktime))
            return True

    # TODO: expose this as a property on audio when #790 gets further along.
    class _Outputs(Gst.Bin):
        def __init__(self):
            Gst.Bin.__init__(self)
            # TODO gst1: Set 'outputs' as the Bin name for easier debugging

            self._tee = Gst.ElementFactory.make("tee")
            self.add(self._tee)

            ghost_pad = Gst.GhostPad.new("sink", self._tee.get_static_pad("sink"))
            self.add_pad(ghost_pad)

        def add_output(self, description):
            # XXX This only works for pipelines not in use until #790 gets done.
            try:
                output = Gst.parse_bin_from_description(
                    description, ghost_unlinked_pads=True
                )
            except GLib.GError as ex:
                logger.error(
                    'Failed to create audio output "%s": %s', description, ex
                )
                raise exceptions.AudioException(ex)

            self._add(output)
            logger.info('Audio output set to "%s"', description)

        def _add(self, element):
            queue = Gst.ElementFactory.make("queue")
            self.add(element)
            self.add(queue)
            queue.link(element)
            self._tee.link(queue)

    class SoftwareMixer:
        def __init__(self, mixer):
            self._mixer = mixer
            self._element = None
            self._last_volume = None
            self._last_mute = None
            self._signals = utils.Signals()

        def setup(self, element, mixer_ref):
            self._element = element
            self._mixer.setup(mixer_ref)

        def teardown(self):
            self._signals.clear()
            self._mixer.teardown()

        def get_volume(self):
            return int(round(self._element.get_property("volume") * 100))

        def set_volume(self, volume):
            self._element.set_property("volume", volume / 100.0)
            self._mixer.trigger_volume_changed(self.get_volume())

        def get_mute(self):
            return self._element.get_property("mute")

        def set_mute(self, mute):
            self._element.set_property("mute", bool(mute))
            self._mixer.trigger_mute_changed(self.get_mute())

    class _Handler:
        def __init__(self, audio):
            self._audio = audio
            self._element = None
            self._pad = None
            self._message_handler_id = None
            self._event_handler_id = None

        def setup_message_handling(self, element):
            self._element = element
            bus = element.get_bus()
            bus.add_signal_watch()
            self._message_handler_id = bus.connect("message", self.on_message)

        def setup_event_handling(self, pad):
            self._pad = pad
            self._event_handler_id = pad.add_probe(
                Gst.PadProbeType.EVENT_BOTH, self.on_pad_event
            )

        def teardown_message_handling(self):
            bus = self._element.get_bus()
            bus.remove_signal_watch()
            bus.disconnect(self._message_handler_id)
            self._message_handler_id = None

        def teardown_event_handling(self):
            self._pad.remove_probe(self._event_handler_id)
            self._event_handler_id = None

        def on_message(self, bus, msg):
            if msg.type == Gst.MessageType.STATE_CHANGED:
                if msg.src != self._element:
                    return
                old_state, new_state, pending_state = msg.parse_state_changed()
                self.on_playbin_state_changed(old_state, new_state, pending_state)
            elif msg.type == Gst.MessageType.BUFFERING:
                self.on_buffering(msg.parse_buffering(), msg.get_structure())
            elif msg.type == Gst.MessageType.EOS:
                self.on_end_of_stream()
            elif msg.type == Gst.MessageType.ERROR:
                error, debug = msg.parse_error()
                self.on_error(error, debug)
            elif msg.type == Gst.MessageType.WARNING:
                error, debug = msg.parse_warning()
                self.on_warning(error, debug)
            elif msg.type == Gst.MessageType.ASYNC_DONE:
                self.on_async_done()
            elif msg.type == Gst.MessageType.TAG:
                taglist = msg.parse_tag()
                self.on_tag(taglist)
            elif msg.type == Gst.MessageType.ELEMENT:
                if GstPbutils.is_missing_plugin_message(msg):
                    self.on_missing_plugin(msg)
            elif msg.type == Gst.MessageType.STREAM_START:
                self.on_stream_start()

        def on_pad_event(self, pad, pad_probe_info):
            event = pad_probe_info.get_event()
            if event.type == Gst.EventType.SEGMENT:
                self.on_segment(event.parse_segment())
            return Gst.PadProbeReturn.OK

        def on_playbin_state_changed(self, old_state, new_state, pending_state):
            gst_logger.debug(
                "Got STATE_CHANGED bus message: old=%s new=%s pending=%s",
                old_state.value_name,
                new_state.value_name,
                pending_state.value_name,
            )

            if new_state == Gst.State.READY and pending_state == Gst.State.NULL:
                # XXX: We're not called on the last state change when going down to
                # NULL, so we rewrite the second to last call to get the expected
                # behavior.
                new_state = Gst.State.NULL
                pending_state = Gst.State.VOID_PENDING

            if pending_state != Gst.State.VOID_PENDING:
                return  # Ignore intermediate state changes

            if new_state == Gst.State.READY:
                return  # Ignore READY state as it's GStreamer specific

            new_state = _GST_STATE_MAPPING[new_state]
            old_state, self._audio.state = self._audio.state, new_state

            target_state = _GST_STATE_MAPPING.get(self._audio._target_state)
            if target_state is None:
                # XXX: Workaround for #1430, to be fixed properly by #1222.
                logger.warning("Race condition happened. See #1222 and #1430.")
                return
            if target_state == new_state:
                target_state = None

            logger.debug(
                "Audio event: state_changed(old_state=%s, new_state=%s, "
                "target_state=%s)",
                old_state,
                new_state,
                target_state,
            )
            AudioListener.send(
                "state_changed",
                old_state=old_state,
                new_state=new_state,
                target_state=target_state,
            )
            if new_state == PlaybackState.STOPPED:
                logger.debug("Audio event: stream_changed(uri=None)")
                AudioListener.send("stream_changed", uri=None)

            if "GST_DEBUG_DUMP_DOT_DIR" in os.environ:
                Gst.debug_bin_to_dot_file(
                    self._audio._playbin, Gst.DebugGraphDetails.ALL, "mopidy"
                )

        def on_buffering(self, percent, structure=None):
            if self._audio._target_state < Gst.State.PAUSED:
                gst_logger.debug("Skip buffering during track change.")
                return

            if structure is not None and structure.has_field("buffering-mode"):
                buffering_mode = structure.get_enum(
                    "buffering-mode", Gst.BufferingMode
                )
                if buffering_mode == Gst.BufferingMode.LIVE:
                    return  # Live sources stall in paused.

            level = logging.getLevelName("TRACE")
            if percent < 10 and not self._audio._buffering:
                self._audio._playbin.set_state(Gst.State.PAUSED)
                self._audio._buffering = True
                level = logging.DEBUG
            if percent == 100:
                self._audio._buffering = False
                if self._audio._target_state == Gst.State.PLAYING:
                    self._audio._playbin.set_state(Gst.State.PLAYING)
                level = logging.DEBUG

            gst_logger.log(
                level, "Got BUFFERING bus message: percent=%d%%", percent
            )

        def on_end_of_stream(self):
            gst_logger.debug("Got EOS (end of stream) bus message.")
            logger.debug("Audio event: reached_end_of_stream()")
            self._audio._tags = {}
            AudioListener.send("reached_end_of_stream")

        def on_error(self, error, debug):
            gst_logger.error(f"GStreamer error: {error.message}")
            gst_logger.debug(
                f"Got ERROR bus message: error={error!r} debug={debug!r}"
            )

            # TODO: is this needed?
            self._audio.stop_playback()

        def on_warning(self, error, debug):
            gst_logger.warning(f"GStreamer warning: {error.message}")
            gst_logger.debug(
                f"Got WARNING bus message: error={error!r} debug={debug!r}"
            )

        def on_async_done(self):
            gst_logger.debug("Got ASYNC_DONE bus message.")

        def on_tag(self, taglist):
            tags = tags_lib.convert_taglist(taglist)
            gst_logger.debug(
                f"Got TAG bus message: tags={tags_lib.repr_tags(tags)}"
            )

            # Postpone emitting tags until stream start.
            if self._audio._pending_tags is not None:
                self._audio._pending_tags.update(tags)
                return

            # TODO: Add proper tests for only emitting changed tags.
            unique = object()
            changed = []
            for key, value in tags.items():
                # Update any tags that changed, and store changed keys.
                if self._audio._tags.get(key, unique) != value:
                    self._audio._tags[key] = value
                    changed.append(key)

            if changed:
                logger.debug("Audio event: tags_changed(tags=%r)", changed)
                AudioListener.send("tags_changed", tags=changed)

        def on_missing_plugin(self, msg):
            desc = GstPbutils.missing_plugin_message_get_description(msg)
            debug = GstPbutils.missing_plugin_message_get_installer_detail(msg)
            gst_logger.debug("Got missing-plugin bus message: description=%r", desc)
            logger.warning("Could not find a %s to handle media.", desc)
            if GstPbutils.install_plugins_supported():
                logger.info(
                    "You might be able to fix this by running: "
                    'gst-installer "%s"',
                    debug,
                )
            # TODO: store the missing plugins installer info in a file so we can
            # can provide a 'mopidy install-missing-plugins' if the system has the
            # required helper installed?

        def on_stream_start(self):
            gst_logger.debug("Got STREAM_START bus message")
            uri = self._audio._pending_uri
            logger.debug("Audio event: stream_changed(uri=%r)", uri)
            AudioListener.send("stream_changed", uri=uri)

            # Emit any postponed tags that we got after about-to-finish.
            tags, self._audio._pending_tags = self._audio._pending_tags, None
            self._audio._tags = tags or {}

            if tags:
                logger.debug("Audio event: tags_changed(tags=%r)", tags.keys())
                AudioListener.send("tags_changed", tags=tags.keys())

            if self._audio._pending_metadata:
                self._audio._playbin.send_event(self._audio._pending_metadata)
                self._audio._pending_metadata = None

        def on_segment(self, segment):
            gst_logger.debug(
                "Got SEGMENT pad event: "
                "rate=%(rate)s format=%(format)s start=%(start)s stop=%(stop)s "
                "position=%(position)s",
                {
                    "rate": segment.rate,
                    "format": Gst.Format.get_name(segment.format),
                    "start": segment.start,
                    "stop": segment.stop,
                    "position": segment.position,
                },
            )
            position_ms = segment.position // Gst.MSECOND
            logger.debug("Audio event: position_changed(position=%r)", position_ms)
            AudioListener.send("position_changed", position=position_ms)

    # TODO: create a player class which replaces the actors internals
    class Audio(pykka.ThreadingActor):

        """
        Audio output through `GStreamer <https://gstreamer.freedesktop.org/>`_.
        """

        #: The GStreamer state mapped to :class:`mopidy.audio.PlaybackState`
        state = PlaybackState.STOPPED

        #: The software mixing interface :class:`mopidy.audio.actor.SoftwareMixer`
        mixer = None

        def __init__(self, config, mixer):
            super().__init__()

            self._config = config
            self._target_state = Gst.State.NULL
            self._buffering = False
            self._live_stream = False
            self._tags = {}
            self._pending_uri = None
            self._pending_tags = None
            self._pending_metadata = None

            self._playbin = None
            self._outputs = None
            self._queue = None
            self._about_to_finish_callback = None
            self._source_setup_callback = None

            self._handler = _Handler(self)
            self._appsrc = _Appsrc()
            self._signals = utils.Signals()

            if mixer and self._config["audio"]["mixer"] == "software":
                self.mixer = pykka.traversable(SoftwareMixer(mixer))

        def on_start(self):
            self._thread = threading.current_thread()
            try:
                self._setup_preferences()
                self._setup_playbin()
                self._setup_outputs()
                self._setup_audio_sink()
            except GLib.GError as ex:
                logger.exception(ex)
                process.exit_process()

        def on_stop(self):
            self._teardown_mixer()
            self._teardown_playbin()

        def _setup_preferences(self):
            # TODO: move out of audio actor?
            # Fix for https://github.com/mopidy/mopidy/issues/604
            registry = Gst.Registry.get()
            jacksink = registry.find_feature("jackaudiosink", Gst.ElementFactory)
            if jacksink:
                jacksink.set_rank(Gst.Rank.SECONDARY)

        def _setup_playbin(self):
            playbin = Gst.ElementFactory.make("playbin")
            playbin.set_property("flags", _GST_PLAY_FLAGS_AUDIO)

            # TODO: turn into config values...
            playbin.set_property("buffer-size", 5 << 20)  # 5MB
            playbin.set_property("buffer-duration", 5 * Gst.SECOND)

            self._signals.connect(playbin, "source-setup", self._on_source_setup)
            self._signals.connect(
                playbin, "about-to-finish", self._on_about_to_finish
            )

            self._playbin = playbin
            self._handler.setup_message_handling(playbin)

        def _teardown_playbin(self):
            self._handler.teardown_message_handling()
            self._handler.teardown_event_handling()
            self._signals.disconnect(self._playbin, "about-to-finish")
            self._signals.disconnect(self._playbin, "source-setup")
            self._playbin.set_state(Gst.State.NULL)

        def _setup_outputs(self):
            # We don't want to use outputs for regular testing, so just install
            # an unsynced fakesink when someone asks for a 'testoutput'.
            if self._config["audio"]["output"] == "testoutput":
                self._outputs = Gst.ElementFactory.make("fakesink")
            else:
                self._outputs = _Outputs()
                try:
                    self._outputs.add_output(self._config["audio"]["output"])
                except exceptions.AudioException:
                    process.exit_process()  # TODO: move this up the chain

            self._handler.setup_event_handling(self._outputs.get_static_pad("sink"))

        def _setup_audio_sink(self):
            audio_sink = Gst.ElementFactory.make("bin", "audio-sink")
            queue = Gst.ElementFactory.make("queue")
            volume = Gst.ElementFactory.make("volume")

            # Queue element to buy us time between the about-to-finish event and
            # the actual switch, i.e. about to switch can block for longer thanks
            # to this queue.

            # TODO: See if settings should be set to minimize latency. Previous
            # setting breaks appsrc, and settings before that broke on a few
            # systems. So leave the default to play it safe.
            buffer_time = self._config["audio"]["buffer_time"]
            if buffer_time is not None and buffer_time > 0:
                queue.set_property("max-size-time", buffer_time * Gst.MSECOND)

            audio_sink.add(queue)
            audio_sink.add(self._outputs)
            audio_sink.add(volume)

            queue.link(volume)
            volume.link(self._outputs)

            if self.mixer:
                self.mixer.setup(volume, self.actor_ref.proxy().mixer)

            ghost_pad = Gst.GhostPad.new("sink", queue.get_static_pad("sink"))
            audio_sink.add_pad(ghost_pad)

            self._playbin.set_property("audio-sink", audio_sink)
            self._queue = queue

        def _teardown_mixer(self):
            if self.mixer:
                self.mixer.teardown()

        def _on_about_to_finish(self, element):
            if self._thread == threading.current_thread():
                logger.error(
                    "about-to-finish in actor, aborting to avoid deadlock."
                )
                return

            gst_logger.debug("Got about-to-finish event.")
            if self._about_to_finish_callback:
                logger.debug("Running about-to-finish callback.")
                self._about_to_finish_callback()

        def _on_source_setup(self, element, source):
            gst_logger.debug(
                "Got source-setup signal: element=%s", source.__class__.__name__
            )

            if source.get_factory().get_name() == "appsrc":
                self._appsrc.configure(source)
            else:
                self._appsrc.reset()

            if self._source_setup_callback:
                logger.debug("Running source-setup callback")
                self._source_setup_callback(source)

            if self._live_stream and hasattr(source.props, "is_live"):
                gst_logger.debug("Enabling live stream mode")
                source.set_live(True)

            utils.setup_proxy(source, self._config["proxy"])

        def set_uri(self, uri, live_stream=False, download=False):
            """
            Set URI of audio to be played.

            You *MUST* call :meth:`prepare_change` before calling this method.

            :param uri: the URI to play
            :type uri: string
            :param live_stream: disables buffering, reducing latency for stream,
                and discarding data when paused
            :type live_stream: bool
            :param download: enables "download" buffering mode
            :type download: bool
            """

            # XXX: Hack to workaround issue on Mac OS X where volume level
            # does not persist between track changes. mopidy/mopidy#886
            if self.mixer is not None:
                current_volume = self.mixer.get_volume()
            else:
                current_volume = None

            flags = _GST_PLAY_FLAGS_AUDIO
            if download:
                flags |= _GST_PLAY_FLAGS_DOWNLOAD

            logger.debug(f"Flags: {flags}")
            if live_stream and download:
                logger.warning(
                    "Ambiguous buffering flags: "
                    "'live_stream' and 'download' should not both be set."
                )

            self._pending_uri = uri
            self._pending_tags = {}
            self._live_stream = live_stream
            self._playbin.set_property("flags", flags)
            self._playbin.set_property("uri", uri)

            if self.mixer is not None and current_volume is not None:
                self.mixer.set_volume(current_volume)

        def set_appsrc(
            self, caps, need_data=None, enough_data=None, seek_data=None
        ):
            """
            Switch to using appsrc for getting audio to be played.

            You *MUST* call :meth:`prepare_change` before calling this method.

            :param caps: GStreamer caps string describing the audio format to
                expect
            :type caps: string
            :param need_data: callback for when appsrc needs data
            :type need_data: callable which takes data length hint in ms
            :param enough_data: callback for when appsrc has enough data
            :type enough_data: callable
            :param seek_data: callback for when data from a new position is needed
                to continue playback
            :type seek_data: callable which takes time position in ms
            """
            self._appsrc.prepare(
                Gst.Caps.from_string(caps), need_data, enough_data, seek_data
            )
            uri = "appsrc://"
            self._pending_uri = uri
            self._live_stream = False
            self._playbin.set_property("flags", _GST_PLAY_FLAGS_AUDIO)
            self._playbin.set_property("uri", uri)

        def emit_data(self, buffer_):
            """
            Call this to deliver raw audio data to be played.

            If the buffer is :class:`None`, the end-of-stream token is put on the
            playbin. We will get a GStreamer message when the stream playback
            reaches the token, and can then do any end-of-stream related tasks.

            Note that the URI must be set to ``appsrc://`` for this to work.

            Returns :class:`True` if data was delivered.

            :param buffer_: buffer to pass to appsrc
            :type buffer_: :class:`Gst.Buffer` or :class:`None`
            :rtype: boolean
            """
            return self._appsrc.push(buffer_)

        def set_source_setup_callback(self, callback):
            """
            Configure audio to use a source-setup callback.

            This should be used to modify source-specific properties such as login
            details.

            :param callable callback: Callback to run when we setup the source.
            """
            self._source_setup_callback = callback

        def set_about_to_finish_callback(self, callback):
            """
            Configure audio to use an about-to-finish callback.

            This should be used to achieve gapless playback. For this to work the
            callback *MUST* call :meth:`set_uri` with the new URI to play and
            block until this call has been made. :meth:`prepare_change` is not
            needed before :meth:`set_uri` in this one special case.

            :param callable callback: Callback to run when we need the next URI.
            """
            self._about_to_finish_callback = callback

        def get_position(self):
            """
            Get position in milliseconds.

            :rtype: int
            """
            success, position = self._playbin.query_position(Gst.Format.TIME)

            if not success:
                # TODO: take state into account for this and possibly also return
                # None as the unknown value instead of zero?
                logger.debug("Position query failed")
                return 0

            return utils.clocktime_to_millisecond(position)

        def set_position(self, position):
            """
            Set position in milliseconds.

            :param position: the position in milliseconds
            :type position: int
            :rtype: :class:`True` if successful, else :class:`False`
            """
            # TODO: double check seek flags in use.
            gst_position = utils.millisecond_to_clocktime(position)
            gst_logger.debug("Sending flushing seek: position=%r", gst_position)
            # Send seek event to the queue not the playbin. The default behavior
            # for bins is to forward this event to all sinks. Which results in
            # duplicate seek events making it to appsrc. Since elements are not
            # allowed to act on the seek event, only modify it, this should be safe
            # to do.
            result = self._queue.seek_simple(
                Gst.Format.TIME, Gst.SeekFlags.FLUSH, gst_position
            )
            return result

        def start_playback(self):
            """
            Notify GStreamer that it should start playback.

            :rtype: :class:`True` if successfull, else :class:`False`
            """
            return self._set_state(Gst.State.PLAYING)

        def pause_playback(self):
            """
            Notify GStreamer that it should pause playback.

            :rtype: :class:`True` if successfull, else :class:`False`
            """
            return self._set_state(Gst.State.PAUSED)

        def prepare_change(self):
            """
            Notify GStreamer that we are about to change state of playback.

            This function *MUST* be called before changing URIs or doing
            changes like updating data that is being pushed. The reason for this
            is that GStreamer will reset all its state when it changes to
            :attr:`Gst.State.READY`.
            """
            return self._set_state(Gst.State.READY)

        def stop_playback(self):
            """
            Notify GStreamer that is should stop playback.

            :rtype: :class:`True` if successfull, else :class:`False`
            """
            return self._set_state(Gst.State.NULL)

        def wait_for_state_change(self):
            """Block until any pending state changes are complete.

            Should only be used by tests.
            """
            self._playbin.get_state(timeout=Gst.CLOCK_TIME_NONE)

        def enable_sync_handler(self):
            """Enable manual processing of messages from bus.

            Should only be used by tests.
            """

            def sync_handler(bus, message):
                self._handler.on_message(bus, message)
                return Gst.BusSyncReply.DROP

            bus = self._playbin.get_bus()
            bus.set_sync_handler(sync_handler)

        def _set_state(self, state):
            """
            Internal method for setting the raw GStreamer state.

            .. digraph:: gst_state_transitions

                graph [rankdir="LR"];
                node [fontsize=10];

                "NULL" -> "READY"
                "PAUSED" -> "PLAYING"
                "PAUSED" -> "READY"
                "PLAYING" -> "PAUSED"
                "READY" -> "NULL"
                "READY" -> "PAUSED"

            :param state: State to set playbin to. One of: `Gst.State.NULL`,
                `Gst.State.READY`, `Gst.State.PAUSED` and `Gst.State.PLAYING`.
            :type state: :class:`Gst.State`
            :rtype: :class:`True` if successfull, else :class:`False`
            """
            if state < Gst.State.PAUSED:
                self._buffering = False

            self._target_state = state
            result = self._playbin.set_state(state)
            gst_logger.debug(
                "Changing state to %s: result=%s",
                state.value_name,
                result.value_name,
            )

            if result == Gst.StateChangeReturn.FAILURE:
                logger.warning(
                    "Setting GStreamer state to %s failed", state.value_name
                )
                return False
            # TODO: at this point we could already emit stopped event instead
            # of faking it in the message handling when result=OK
            return True

        # TODO: bake this into setup appsrc perhaps?
        def set_metadata(self, track):
            """
            Set track metadata for currently playing song.

            Only needs to be called by sources such as ``appsrc`` which do not
            already inject tags in playbin, e.g. when using :meth:`emit_data` to
            deliver raw audio data to GStreamer.

            :param track: the current track
            :type track: :class:`mopidy.models.Track`
            """
            taglist = Gst.TagList.new_empty()
            artists = [a for a in (track.artists or []) if a.name]

            def set_value(tag, value):
                gobject_value = GObject.Value()
                gobject_value.init(GObject.TYPE_STRING)
                gobject_value.set_string(value)
                taglist.add_value(Gst.TagMergeMode.REPLACE, tag, gobject_value)

            # Default to blank data to trick shoutcast into clearing any previous
            # values it might have.
            # TODO: Verify if this works at all, likely it doesn't.
            set_value(Gst.TAG_ARTIST, " ")
            set_value(Gst.TAG_TITLE, " ")
            set_value(Gst.TAG_ALBUM, " ")

            if artists:
                set_value(Gst.TAG_ARTIST, ", ".join(a.name for a in artists))

            if track.name:
                set_value(Gst.TAG_TITLE, track.name)

            if track.album and track.album.name:
                set_value(Gst.TAG_ALBUM, track.album.name)

            gst_logger.debug(
                "Sending TAG event for track %r: %r", track.uri, taglist.to_string()
            )
            event = Gst.Event.new_tag(taglist)
            if self._pending_uri:
                self._pending_metadata = event
            else:
                self._playbin.send_event(event)

        def get_current_tags(self):
            """
            Get the currently playing media's tags.

            If no tags have been found, or nothing is playing this returns an empty
            dictionary. For each set of tags we collect a tags_changed event is
            emitted with the keys of the changes tags. After such calls users may
            call this function to get the updated values.

            :rtype: {key: [values]} dict for the current media.
            """
            # TODO: should this be a (deep) copy? most likely yes
            # TODO: should we return None when stopped?
            # TODO: support only fetching keys we care about?
            return self._tags