#!/usr/bin/env python3

# Libervia WebRTC implementation
# Copyright (C) 2009-2023 Jérôme Poisson (goffi@goffi.org)

# This program is free software: you can redistribute it and/or modify
# it under the terms of the GNU Affero General Public License as published by
# the Free Software Foundation, either version 3 of the License, or
# (at your option) any later version.

# This program 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 Affero General Public License for more details.

# You should have received a copy of the GNU Affero General Public License
# along with this program.  If not, see <http://www.gnu.org/licenses/>.

import gi
gi.require_versions({
    "Gst": "1.0",
    "GstWebRTC": "1.0"
})
from gi.repository import Gst, GstWebRTC, GstSdp

try:
    from gi.overrides import Gst as _
except ImportError:
    print(
        "no GStreamer python overrides available, please install relevant pacakges on "
        "your system."
    )
import asyncio
from dataclasses import dataclass
from datetime import datetime
import logging
import re
from typing import Callable
from urllib.parse import quote_plus

from libervia.backend.core import exceptions
from libervia.backend.tools.common import data_format
from libervia.frontends.tools import aio


log = logging.getLogger(__name__)

Gst.init(None)

SOURCES_AUTO = "auto"
SOURCES_TEST = "test"
SINKS_APP = "app"
SINKS_AUTO = "auto"
SINKS_TEST = "test"


@dataclass
class AppSinkData:
    local_video_cb: Callable
    remote_video_cb: Callable


class WebRTC:
    """GSTreamer based WebRTC implementation for audio and video communication.

    This class encapsulates the WebRTC functionalities required for initiating and
    handling audio and video calls.
    """

    def __init__(
        self,
        bridge,
        profile: str,
        sources: str = SOURCES_AUTO,
        sinks: str = SINKS_AUTO,
        appsink_data: AppSinkData | None = None,
        reset_cb: Callable | None = None,
    ) -> None:
        self.main_loop = asyncio.get_event_loop()
        self.bridge = bridge
        self.profile = profile
        self.pipeline = None
        self._audio_muted = False
        self._video_muted = False
        self.sources = sources
        self.sinks = sinks
        if sinks == SINKS_APP:
            self.appsink_data = appsink_data
        elif appsink_data is not None:
            raise exceptions.InternalError(
                "appsink_data can only be used for SINKS_APP sinks"
            )
        self.reset_cb = reset_cb
        self.reset_instance()

    @property
    def audio_muted(self):
        return self._audio_muted

    @audio_muted.setter
    def audio_muted(self, muted: bool) -> None:
        if muted != self._audio_muted:
            self._audio_muted = muted
            self.on_audio_mute(muted)

    @property
    def video_muted(self):
        return self._video_muted

    @video_muted.setter
    def video_muted(self, muted: bool) -> None:
        if muted != self._video_muted:
            self._video_muted = muted
            self.on_video_mute(muted)

    @property
    def sdp_set(self):
        return self._sdp_set

    @sdp_set.setter
    def sdp_set(self, is_set: bool):
        self._sdp_set = is_set
        if is_set:
            self.on_ice_candidates_new(self.remote_candidates_buffer)
            for data in self.remote_candidates_buffer.values():
                data["candidates"].clear()

    @property
    def media_types(self):
        if self._media_types is None:
            raise Exception("self._media_types should not be None!")
        return self._media_types

    @media_types.setter
    def media_types(self, new_media_types: dict) -> None:
        self._media_types = new_media_types
        self._media_types_inv = {v: k for k, v in new_media_types.items()}

    @property
    def media_types_inv(self) -> dict:
        if self._media_types_inv is None:
            raise Exception("self._media_types_inv should not be None!")
        return self._media_types_inv

    def generate_dot_file(
        self,
        filename: str = "pipeline",
        details: Gst.DebugGraphDetails = Gst.DebugGraphDetails.ALL,
        with_timestamp: bool = True,
        bin_: Gst.Bin|None = None,
    ) -> None:
        """Generate Dot File for debugging

        ``GST_DEBUG_DUMP_DOT_DIR`` environment variable must be set to destination dir.
        ``dot -Tpng -o <filename>.png <filename>.dot`` can be use to convert to a PNG file.
        See
        https://gstreamer.freedesktop.org/documentation/gstreamer/debugutils.html?gi-language=python#GstDebugGraphDetails
        for details.

        @param filename: name of the generated file
        @param details: which details to print
        @param with_timestamp: if True, add a timestamp to filename
        @param bin_: which bin to output. By default, the whole pipeline
            (``self.pipeline``) will be used.
        """
        if bin_ is None:
            bin_ = self.pipeline
        if with_timestamp:
            timestamp = datetime.now().isoformat(timespec='milliseconds')
            filename = f"{timestamp}_filename"

        Gst.debug_bin_to_dot_file(bin_, details, filename)

    def get_sdp_mline_index(self, media_type: str) -> int:
        """Gets the sdpMLineIndex for a given media type.

        @param media_type: The type of the media.
        """
        for index, m_type in self.media_types.items():
            if m_type == media_type:
                return index
        raise ValueError(f"Media type '{media_type}' not found")

    def _set_media_types(self, offer_sdp: str) -> None:
        """Sets media types from offer SDP

        @param offer: RTC session description containing the offer
        """
        sdp_lines = offer_sdp.splitlines()
        media_types = {}
        mline_index = 0

        for line in sdp_lines:
            if line.startswith("m="):
                media_types[mline_index] = line[2 : line.find(" ")]
                mline_index += 1

        self.media_types = media_types

    def _a_call(self, method, *args, **kwargs):
        """Call an async method in main thread"""
        aio.run_from_thread(method, *args, **kwargs, loop=self.main_loop)

    def get_payload_types(
        self, sdpmsg, video_encoding: str, audio_encoding: str
    ) -> dict[str, int | None]:
        """Find the payload types for the specified video and audio encoding.

        Very simplistically finds the first payload type matching the encoding
        name. More complex applications will want to match caps on
        profile-level-id, packetization-mode, etc.
        """
        # method coming from gstreamer example (Matthew Waters, Nirbheek Chauhan) at
        # subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
        video_pt = None
        audio_pt = None
        for i in range(0, sdpmsg.medias_len()):
            media = sdpmsg.get_media(i)
            for j in range(0, media.formats_len()):
                fmt = media.get_format(j)
                if fmt == "webrtc-datachannel":
                    continue
                pt = int(fmt)
                caps = media.get_caps_from_media(pt)
                s = caps.get_structure(0)
                encoding_name = s["encoding-name"]
                if video_pt is None and encoding_name == video_encoding:
                    video_pt = pt
                elif audio_pt is None and encoding_name == audio_encoding:
                    audio_pt = pt
        return {video_encoding: video_pt, audio_encoding: audio_pt}

    def parse_ice_candidate(self, candidate_string):
        """Parses the ice candidate string.

        @param candidate_string: The ice candidate string to be parsed.
        """
        pattern = re.compile(
            r"candidate:(?P<foundation>\S+) (?P<component_id>\d+) (?P<transport>\S+) "
            r"(?P<priority>\d+) (?P<address>\S+) (?P<port>\d+) typ "
            r"(?P<type>\S+)(?: raddr (?P<rel_addr>\S+) rport "
            r"(?P<rel_port>\d+))?(?: generation (?P<generation>\d+))?"
        )
        match = pattern.match(candidate_string)
        if match:
            candidate_dict = match.groupdict()

            # Apply the correct types to the dictionary values
            candidate_dict["component_id"] = int(candidate_dict["component_id"])
            candidate_dict["priority"] = int(candidate_dict["priority"])
            candidate_dict["port"] = int(candidate_dict["port"])

            if candidate_dict["rel_port"]:
                candidate_dict["rel_port"] = int(candidate_dict["rel_port"])

            if candidate_dict["generation"]:
                candidate_dict["generation"] = candidate_dict["generation"]

            # Remove None values
            return {k: v for k, v in candidate_dict.items() if v is not None}
        else:
            log.warning(f"can't parse candidate: {candidate_string!r}")
            return None

    def build_ice_candidate(self, parsed_candidate):
        """Builds ICE candidate

        @param parsed_candidate: Dictionary containing parsed ICE candidate
        """
        base_format = (
            "candidate:{foundation} {component_id} {transport} {priority} "
            "{address} {port} typ {type}"
        )

        if parsed_candidate.get("rel_addr") and parsed_candidate.get("rel_port"):
            base_format += " raddr {rel_addr} rport {rel_port}"

        if parsed_candidate.get("generation"):
            base_format += " generation {generation}"

        return base_format.format(**parsed_candidate)

    def extract_ufrag_pwd(self, sdp: str) -> tuple[str, str]:
        """Retrieves ICE password and user fragment for SDP offer.

        @param sdp: The Session Description Protocol offer string.
        @return: ufrag and pwd
        @raise ValueError: Can't extract ufrag and password
        """
        ufrag_line = re.search(r"ice-ufrag:(\S+)", sdp)
        pwd_line = re.search(r"ice-pwd:(\S+)", sdp)

        if ufrag_line and pwd_line:
            ufrag = self.ufrag = ufrag_line.group(1)
            pwd = self.pwd = pwd_line.group(1)
            return ufrag, pwd
        else:
            log.error(f"SDP with missing ice-ufrag or ice-pwd:\n{sdp}")
            raise ValueError("Can't extract ice-ufrag and ice-pwd from SDP")

    def reset_instance(self):
        """Inits or resets the instance variables to their default state."""
        self.role: str | None = None
        if self.pipeline is not None:
            self.pipeline.set_state(Gst.State.NULL)
        self.pipeline = None
        self._remote_video_pad = None
        self.sid: str | None = None
        self.offer: str | None = None
        self.local_candidates_buffer = {}
        self.ufrag: str | None = None
        self.pwd: str | None = None
        self.callee: str | None = None
        self._media_types = None
        self._media_types_inv = None
        self._sdp_set: bool = False
        self.remote_candidates_buffer: dict[str, dict[str, list]] = {
            "audio": {"candidates": []},
            "video": {"candidates": []},
        }
        self._media_types = None
        self._media_types_inv = None
        self.audio_valve = None
        self.video_valve = None
        if self.reset_cb is not None:
            self.reset_cb()


    async def setup_call(
        self,
        role: str,
        audio_pt: int | None = 96,
        video_pt: int | None = 97,
    ) -> None:
        """Sets up the call.

        This method establishes the Gstreamer pipeline for audio and video communication.
        The method also manages STUN and TURN server configurations, signal watchers, and
        various connection handlers for the webrtcbin.

        @param role: The role for the call, either 'initiator' or 'responder'.
        @param audio_pt: The payload type for the audio stream.
        @param video_pt: The payload type for the video stream

        @raises NotImplementedError: If audio_pt or video_pt is set to None.
        @raises AssertionError: If the role is not 'initiator' or 'responder'.
        """
        assert role in ("initiator", "responder")
        self.role = role
        if audio_pt is None or video_pt is None:
            raise NotImplementedError("None value is not handled yet")

        if self.sources == SOURCES_AUTO:
            video_source_elt = "v4l2src"
            audio_source_elt = "pulsesrc"
        elif self.sources == SOURCES_TEST:
            video_source_elt = "videotestsrc is-live=true pattern=ball"
            audio_source_elt = "audiotestsrc"
        else:
            raise exceptions.InternalError(f'Unknown "sources" value: {self.sources!r}')

        extra_elt = ""

        if self.sinks == SINKS_APP:
            local_video_sink_elt = (
                "appsink name=local_video_sink emit-signals=true drop=true max-buffers=1 "
                "sync=True"
            )
        elif self.sinks == SINKS_AUTO:
            extra_elt = "compositor name=compositor ! autovideosink"
            local_video_sink_elt = """compositor.sink_1"""
        else:
            raise exceptions.InternalError(f"Unknown sinks value {self.sinks!r}")

        self.gst_pipe_desc = f"""
        webrtcbin latency=100 name=sendrecv bundle-policy=max-bundle

        input-selector name=video_selector
        ! videorate
        ! video/x-raw,framerate=30/1
        ! tee name=t

        {extra_elt}

        {video_source_elt} name=video_src ! queue leaky=downstream ! video_selector.
        videotestsrc is-live=true pattern=black ! queue leaky=downstream ! video_selector.

        t.
        ! queue max-size-buffers=5 max-size-time=0 max-size-bytes=0 leaky=downstream
        ! videoconvert
        ! vp8enc deadline=1 keyframe-max-dist=60
        ! rtpvp8pay picture-id-mode=15-bit
        ! application/x-rtp,media=video,encoding-name=VP8,payload={video_pt}
        ! sendrecv.

        t.
        ! queue max-size-buffers=5 max-size-time=0 max-size-bytes=0 leaky=downstream
        ! videoconvert
        ! {local_video_sink_elt}

        {audio_source_elt} name=audio_src
        ! valve
        ! queue max-size-buffers=10 max-size-time=0 max-size-bytes=0 leaky=downstream
        ! audioconvert
        ! audioresample
        ! opusenc audio-type=voice
        ! rtpopuspay
        ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt}
        ! sendrecv.
        """

        log.debug(f"Gstreamer pipeline: {self.gst_pipe_desc}")

        # Create the pipeline
        self.pipeline = Gst.parse_launch(self.gst_pipe_desc)
        if not self.pipeline:
            raise exceptions.InternalError("Failed to create Gstreamer pipeline.")

        self.webrtcbin = self.pipeline.get_by_name("sendrecv")
        self.video_src = self.pipeline.get_by_name("video_src")
        self.video_selector = self.pipeline.get_by_name("video_selector")
        self.audio_valve = self.pipeline.get_by_name("audio_valve")

        if self.video_muted:
            self.on_video_mute(True)
        if self.audio_muted:
            self.on_audio_mute(True)

        # set STUN and TURN servers
        external_disco = data_format.deserialise(
            await self.bridge.external_disco_get("", self.profile), type_check=list
        )

        for server in external_disco:
            if server["type"] == "stun":
                if server["transport"] == "tcp":
                    log.info(
                        "ignoring TCP STUN server, GStreamer only support one STUN server"
                    )
                url = f"stun://{server['host']}:{server['port']}"
                log.debug(f"adding stun server: {url}")
                self.webrtcbin.set_property("stun-server", url)
            elif server["type"] == "turn":
                url = "{scheme}://{username}:{password}@{host}:{port}".format(
                    scheme="turns" if server["transport"] == "tcp" else "turn",
                    username=quote_plus(server["username"]),
                    password=quote_plus(server["password"]),
                    host=server["host"],
                    port=server["port"],
                )
                log.debug(f"adding turn server: {url}")

                if not self.webrtcbin.emit("add-turn-server", url):
                    log.warning(f"Erreur while adding TURN server {url}")

        # local video feedback
        if self.sinks == SINKS_APP:
            assert self.appsink_data is not None
            local_video_sink = self.pipeline.get_by_name("local_video_sink")
            local_video_sink.set_property("emit-signals", True)
            local_video_sink.connect("new-sample", self.appsink_data.local_video_cb)
            local_video_sink_caps = Gst.Caps.from_string(f"video/x-raw,format=RGB")
            local_video_sink.set_property("caps", local_video_sink_caps)

        # Create bus and associate signal watchers
        self.bus = self.pipeline.get_bus()
        if not self.bus:
            log.error("Failed to get bus from pipeline.")
            return

        self.bus.add_signal_watch()
        self.webrtcbin.connect("pad-added", self.on_pad_added)
        self.bus.connect("message::error", self.on_bus_error)
        self.bus.connect("message::eos", self.on_bus_eos)
        self.webrtcbin.connect("on-negotiation-needed", self.on_negotiation_needed)
        self.webrtcbin.connect("on-ice-candidate", self.on_ice_candidate)
        self.webrtcbin.connect(
            "notify::ice-gathering-state", self.on_ice_gathering_state_change
        )
        self.webrtcbin.connect(
            "notify::ice-connection-state", self.on_ice_connection_state
        )

    def start_pipeline(self) -> None:
        """Starts the GStreamer pipeline."""
        log.debug("starting the pipeline")
        self.pipeline.set_state(Gst.State.PLAYING)

    def on_negotiation_needed(self, webrtc):
        """Initiate SDP offer when negotiation is needed."""
        log.debug("Negotiation needed.")
        if self.role == "initiator":
            log.debug("Creating offer…")
            promise = Gst.Promise.new_with_change_func(self.on_offer_created)
            self.webrtcbin.emit("create-offer", None, promise)

    def on_offer_created(self, promise):
        """Callback for when SDP offer is created."""
        log.info("on_offer_created called")
        assert promise.wait() == Gst.PromiseResult.REPLIED
        reply = promise.get_reply()
        if reply is None:
            log.error("Promise reply is None. Offer creation might have failed.")
            return
        offer = reply["offer"]
        self.offer = offer.sdp.as_text()
        log.info(f"SDP offer created: \n{self.offer}")
        self._set_media_types(self.offer)
        promise = Gst.Promise.new()
        self.webrtcbin.emit("set-local-description", offer, promise)
        promise.interrupt()
        self._a_call(self._start_call)

    def on_answer_set(self, promise):
        assert promise.wait() == Gst.PromiseResult.REPLIED

    def on_answer_created(self, promise, _, __):
        """Callback for when SDP answer is created."""
        assert promise.wait() == Gst.PromiseResult.REPLIED
        reply = promise.get_reply()
        answer = reply["answer"]
        promise = Gst.Promise.new()
        self.webrtcbin.emit("set-local-description", answer, promise)
        promise.interrupt()
        answer_sdp = answer.sdp.as_text()
        log.info(f"SDP answer set: \n{answer_sdp}")
        self.sdp_set = True
        self._a_call(self.bridge.call_answer_sdp, self.sid, answer_sdp, self.profile)

    def on_offer_set(self, promise):
        assert promise.wait() == Gst.PromiseResult.REPLIED
        promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None)
        self.webrtcbin.emit("create-answer", None, promise)

    def link_element_or_pad(
        self, source: Gst.Element, dest: Gst.Element | Gst.Pad
    ) -> bool:
        """Check if dest is a pad or an element, and link appropriately"""
        src_pad = source.get_static_pad("src")

        if isinstance(dest, Gst.Pad):
            # If the dest is a pad, link directly
            if not src_pad.link(dest) == Gst.PadLinkReturn.OK:
                log.error(
                    "Failed to link 'conv' to the compositor's newly requested pad!"
                )
                return False
        elif isinstance(dest, Gst.Element):
            return source.link(dest)
        else:
            log.error(f"Unexpected type for dest: {type(sink)}")
            return False

        return True

    def scaled_dimensions(
        self, original_width: int, original_height: int, max_width: int, max_height: int
    ) -> tuple[int, int]:
        """Calculates the scaled dimensions preserving aspect ratio.

        @param original_width: Original width of the video stream.
        @param original_height: Original height of the video stream.
        @param max_width: Maximum desired width for the scaled video.
        @param max_height: Maximum desired height for the scaled video.
        @return: The width and height of the scaled video.
        """
        aspect_ratio = original_width / original_height
        new_width = int(max_height * aspect_ratio)

        if new_width <= max_width:
            return new_width, max_height

        new_height = int(max_width / aspect_ratio)
        return max_width, new_height

    def on_remote_decodebin_stream(self, _, pad: Gst.Pad) -> None:
        """Handle the stream from the remote decodebin.

        This method processes the incoming stream from the remote decodebin, determining
        whether it's video or audio. It then sets up the appropriate GStreamer elements
        for video/audio processing and adds them to the pipeline.

        @param pad: The Gst.Pad from the remote decodebin producing the stream.
        """
        assert self.pipeline is not None
        if not pad.has_current_caps():
            log.error(f"{pad} has no caps, ignoring")
            return

        caps = pad.get_current_caps()
        assert len(caps)
        s = caps[0]
        name = s.get_name()
        log.debug(f"====> NAME START: {name}")

        q = Gst.ElementFactory.make("queue")

        if name.startswith("video"):
            log.debug("===> VIDEO OK")

            self._remote_video_pad = pad

            # Check and log the original size of the video
            width = s.get_int("width").value
            height = s.get_int("height").value
            log.info(f"Original video size: {width}x{height}")

            # This is a fix for an issue found with Movim on desktop: a non standard
            # resolution is used (990x557) resulting in bad alignement and no color in
            # rendered image
            adjust_resolution = width % 4 != 0 or height % 4 != 0
            if adjust_resolution:
                log.info("non standard resolution, we need to adjust size")
                width = (width + 3) // 4 * 4
                height = (height + 3) // 4 * 4
                log.info(f"Adjusted video size: {width}x{height}")

            conv = Gst.ElementFactory.make("videoconvert")
            if self.sinks == SINKS_APP:
                assert self.appsink_data is not None
                remote_video_sink = Gst.ElementFactory.make("appsink")

                appsink_caps = Gst.Caps.from_string("video/x-raw,format=RGB")
                remote_video_sink.set_property("caps", appsink_caps)

                remote_video_sink.set_property("emit-signals", True)
                remote_video_sink.set_property("drop", True)
                remote_video_sink.set_property("max-buffers", 1)
                remote_video_sink.set_property("sync", True)
                remote_video_sink.connect("new-sample", self.appsink_data.remote_video_cb)
                self.pipeline.add(remote_video_sink)
            if self.sinks == SINKS_AUTO:
                compositor = self.pipeline.get_by_name("compositor")

                sink1_pad = compositor.get_static_pad("sink_1")

                local_width, local_height = self.scaled_dimensions(
                    sink1_pad.get_property("width"),
                    sink1_pad.get_property("height"),
                    width // 3,
                    height // 3,
                )

                sink1_pad.set_property("xpos", width - local_width)
                sink1_pad.set_property("ypos", height - local_height)
                sink1_pad.set_property("width", local_width)
                sink1_pad.set_property("height", local_height)
                sink1_pad.set_property("zorder", 1)

                # Request a new pad for the remote stream
                sink_pad_template = compositor.get_pad_template("sink_%u")
                remote_video_sink = compositor.request_pad(sink_pad_template, None, None)
                remote_video_sink.set_property("zorder", 0)

            else:
                raise exceptions.InternalError(f'Unhandled "sinks" value: {self.sinks!r}')

            if adjust_resolution:
                videoscale = Gst.ElementFactory.make("videoscale")
                adjusted_caps = Gst.Caps.from_string(
                    f"video/x-raw,width={width},height={height}"
                )
                capsfilter = Gst.ElementFactory.make("capsfilter")
                capsfilter.set_property("caps", adjusted_caps)

                self.pipeline.add(q, conv, videoscale, capsfilter)


                self.pipeline.sync_children_states()
                ret = pad.link(q.get_static_pad("sink"))
                if ret != Gst.PadLinkReturn.OK:
                    log.error(f"Error linking pad: {ret}")
                q.link(conv)
                conv.link(videoscale)
                videoscale.link(capsfilter)
                self.link_element_or_pad(capsfilter.link, remote_video_sink)

            else:
                self.pipeline.add(q, conv)

                self.pipeline.sync_children_states()
                ret = pad.link(q.get_static_pad("sink"))
                if ret != Gst.PadLinkReturn.OK:
                    log.error(f"Error linking pad: {ret}")
                q.link(conv)
                self.link_element_or_pad(conv, remote_video_sink)

        elif name.startswith("audio"):
            log.debug("===> Audio OK")
            conv = Gst.ElementFactory.make("audioconvert")
            resample = Gst.ElementFactory.make("audioresample")
            remote_audio_sink = Gst.ElementFactory.make("autoaudiosink")
            self.pipeline.add(q, conv, resample, remote_audio_sink)
            self.pipeline.sync_children_states()
            ret = pad.link(q.get_static_pad("sink"))
            if ret != Gst.PadLinkReturn.OK:
                log.error(f"Error linking pad: {ret}")
            q.link(conv)
            conv.link(resample)
            resample.link(remote_audio_sink)

        else:
            log.warning(f"unmanaged name: {name!r}")

    def on_pad_added(self, __, pad: Gst.Pad) -> None:
        """Handle the addition of a new pad to the element.

        When a new source pad is added to the element, this method creates a decodebin,
        connects it to handle the stream, and links the pad to the decodebin.

        @param __: Placeholder for the signal source. Not used in this method.
        @param pad: The newly added pad.
        """
        log.debug("on_pad_added")
        if pad.direction != Gst.PadDirection.SRC:
            return

        decodebin = Gst.ElementFactory.make("decodebin")
        decodebin.connect("pad-added", self.on_remote_decodebin_stream)
        self.pipeline.add(decodebin)
        decodebin.sync_state_with_parent()
        pad.link(decodebin.get_static_pad("sink"))

    async def _start_call(self) -> None:
        """Initiate the call.

        Initiates a call with the callee using the stored offer. If there are any buffered
        local ICE candidates, they are sent as part of the initiation.
        """
        assert self.callee
        self.sid = await self.bridge.call_start(
            str(self.callee), data_format.serialise({"sdp": self.offer}), self.profile
        )
        if self.local_candidates_buffer:
            log.debug(
                f"sending buffered local ICE candidates: {self.local_candidates_buffer}"
            )
            if self.pwd is None:
                sdp = self.webrtcbin.props.local_description.sdp.as_text()
                self.extract_ufrag_pwd(sdp)
            ice_data = {}
            for media_type, candidates in self.local_candidates_buffer.items():
                ice_data[media_type] = {
                    "ufrag": self.ufrag,
                    "pwd": self.pwd,
                    "candidates": candidates,
                }
            await self.bridge.ice_candidates_add(
                self.sid, data_format.serialise(ice_data), self.profile
            )
            self.local_candidates_buffer.clear()

    def _remote_sdp_set(self, promise) -> None:
        assert promise.wait() == Gst.PromiseResult.REPLIED
        self.sdp_set = True

    def on_accepted_call(self, sdp: str, profile: str) -> None:
        """Outgoing call has been accepted.

        @param sdp: The SDP answer string received from the other party.
        @param profile: Profile used for the call.
        """
        log.debug(f"SDP answer received: \n{sdp}")

        __, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp)
        answer = GstWebRTC.WebRTCSessionDescription.new(
            GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg
        )
        promise = Gst.Promise.new_with_change_func(self._remote_sdp_set)
        self.webrtcbin.emit("set-remote-description", answer, promise)

    async def answer_call(self, sdp: str, profile: str) -> None:
        """Answer an incoming call

        @param sdp: The SDP offer string received from the initiator.
        @param profile: Profile used for the call.

        @raise AssertionError: Raised when either "VP8" or "OPUS" is not present in
            payload types.
        """
        log.debug(f"SDP offer received: \n{sdp}")
        self._set_media_types(sdp)
        __, offer_sdp_msg = GstSdp.SDPMessage.new_from_text(sdp)
        payload_types = self.get_payload_types(
            offer_sdp_msg, video_encoding="VP8", audio_encoding="OPUS"
        )
        assert "VP8" in payload_types
        assert "OPUS" in payload_types
        await self.setup_call(
            "responder", audio_pt=payload_types["OPUS"], video_pt=payload_types["VP8"]
        )
        self.start_pipeline()
        offer = GstWebRTC.WebRTCSessionDescription.new(
            GstWebRTC.WebRTCSDPType.OFFER, offer_sdp_msg
        )
        promise = Gst.Promise.new_with_change_func(self.on_offer_set)
        self.webrtcbin.emit("set-remote-description", offer, promise)

    def on_ice_candidate(self, webrtc, mline_index, candidate_sdp):
        """Handles the on-ice-candidate signal of webrtcbin.

        @param webrtc: The webrtcbin element.
        @param mlineindex: The mline index.
        @param candidate: The ICE candidate.
        """
        log.debug(
            f"Local ICE candidate. MLine Index: {mline_index}, Candidate: {candidate_sdp}"
        )
        parsed_candidate = self.parse_ice_candidate(candidate_sdp)
        try:
            media_type = self.media_types[mline_index]
        except KeyError:
            raise exceptions.InternalError("can't find media type")

        if self.sid is None:
            log.debug("buffering local ICE candidate")
            self.local_candidates_buffer.setdefault(media_type, []).append(
                parsed_candidate
            )
        else:
            sdp = self.webrtcbin.props.local_description.sdp.as_text()
            assert sdp is not None
            ufrag, pwd = self.extract_ufrag_pwd(sdp)
            ice_data = {"ufrag": ufrag, "pwd": pwd, "candidates": [parsed_candidate]}
            self._a_call(
                self.bridge.ice_candidates_add,
                self.sid,
                data_format.serialise({media_type: ice_data}),
                self.profile,
            )

    def on_ice_candidates_new(self, candidates: dict) -> None:
        """Handle new ICE candidates.

        @param candidates: A dictionary containing media types ("audio" or "video") as
            keys and corresponding ICE data as values.

        @raise exceptions.InternalError: Raised when sdp mline index is not found.
        """
        if not self.sdp_set:
            log.debug("buffering remote ICE candidate")
            for media_type in ("audio", "video"):
                media_candidates = candidates.get(media_type)
                if media_candidates:
                    buffer = self.remote_candidates_buffer[media_type]
                    buffer["candidates"].extend(media_candidates["candidates"])
            return
        for media_type, ice_data in candidates.items():
            for candidate in ice_data["candidates"]:
                candidate_sdp = self.build_ice_candidate(candidate)
                try:
                    mline_index = self.get_sdp_mline_index(media_type)
                except Exception as e:
                    raise exceptions.InternalError(f"Can't find sdp mline index: {e}")
                self.webrtcbin.emit("add-ice-candidate", mline_index, candidate_sdp)
                log.debug(
                    f"Remote ICE candidate added. MLine Index: {mline_index}, "
                    f"{candidate_sdp}"
                )

    def on_ice_gathering_state_change(self, pspec, __):
        state = self.webrtcbin.get_property("ice-gathering-state")
        log.debug(f"ICE gathering state changed to {state}")

    def on_ice_connection_state(self, pspec, __):
        state = self.webrtcbin.props.ice_connection_state
        if state == GstWebRTC.WebRTCICEConnectionState.FAILED:
            log.error("ICE connection failed")
        log.info(f"ICE connection state changed to {state}")

    def on_bus_error(self, bus: Gst.Bus, message: Gst.Message) -> None:
        """Handles the GStreamer bus error messages.

        @param bus: The GStreamer bus.
        @param message: The error message.
        """
        err, debug = message.parse_error()
        log.error(f"Error from {message.src.get_name()}: {err.message}")
        log.error(f"Debugging info: {debug}")

    def on_bus_eos(self, bus: Gst.Bus, message: Gst.Message) -> None:
        """Handles the GStreamer bus eos messages.

        @param bus: The GStreamer bus.
        @param message: The eos message.
        """
        log.info("End of stream")

    def on_audio_mute(self, muted: bool) -> None:
        if self.audio_valve is not None:
            self.audio_valve.set_property("drop", muted)
            state = "muted" if muted else "unmuted"
            log.info(f"audio is now {state}")

    def on_video_mute(self, muted: bool) -> None:
        if self.video_selector is not None:
            # when muted, we switch to a black image and deactivate the camera
            if not muted:
                self.video_src.set_state(Gst.State.PLAYING)
            pad = self.video_selector.get_static_pad("sink_1" if muted else "sink_0")
            self.video_selector.props.active_pad = pad
            if muted:
                self.video_src.set_state(Gst.State.NULL)
            state = "muted" if muted else "unmuted"
            log.info(f"video is now {state}")

    async def end_call(self) -> None:
        """Stop streaming and clean instance"""
        self.reset_instance()