How to use the aiortc.rtcrtpparameters.RTCRtpParameters function in aiortc

queue = asyncio.Queue()

        async def mock_send_rtp(data):
            if not is_rtcp(data):
                await queue.put(RtpPacket.parse(data))

        self.local_transport._send_rtp = mock_send_rtp

        sender = RTCRtpSender(VideoStreamTrack(), self.local_transport)
        sender._ssrc = 1234
        sender._rtx_ssrc = 2345
        self.assertEqual(sender.kind, "video")

        run(
            sender.send(
                RTCRtpParameters(
                    codecs=[
                        VP8_CODEC,
                        RTCRtpCodecParameters(
                            mimeType="video/rtx",
                            clockRate=90000,
                            payloadType=101,
                            parameters={"apt": 100},
                        ),
                    ]
                )
            )
        )

        # wait for one packet to be transmitted, and ask to retransmit
        packet = run(queue.get())
        run(sender._retransmit(packet.sequence_number))

"""
        Ask for an RTP packet retransmission.
        """
        queue = asyncio.Queue()

        async def mock_send_rtp(data):
            if not is_rtcp(data):
                await queue.put(RtpPacket.parse(data))

        self.local_transport._send_rtp = mock_send_rtp

        sender = RTCRtpSender(VideoStreamTrack(), self.local_transport)
        sender._ssrc = 1234
        self.assertEqual(sender.kind, "video")

        run(sender.send(RTCRtpParameters(codecs=[VP8_CODEC])))

        # wait for one packet to be transmitted, and ask to retransmit
        packet = run(queue.get())
        run(sender._retransmit(packet.sequence_number))

        # wait for packet to be retransmitted, then shutdown
        run(asyncio.sleep(0.1))
        run(sender.stop())

        # check packet was retransmitted
        found_rtx = None
        while not queue.empty():
            queue_packet = queue.get_nowait()
            if queue_packet.sequence_number == packet.sequence_number:
                found_rtx = queue_packet
                break

def test_handle_rtcp_rr(self):
        sender = RTCRtpSender(VideoStreamTrack(), self.local_transport)
        self.assertEqual(sender.kind, "video")

        run(sender.send(RTCRtpParameters(codecs=[VP8_CODEC])))

        # receive RTCP RR
        packet = RtcpRrPacket(
            ssrc=1234,
            reports=[
                RtcpReceiverInfo(
                    ssrc=sender._ssrc,
                    fraction_lost=0,
                    packets_lost=0,
                    highest_sequence=630,
                    jitter=1906,
                    lsr=0,
                    dlsr=0,
                )
            ],
        )

def test_handle_rtcp_pli(self):
        sender = RTCRtpSender(VideoStreamTrack(), self.local_transport)
        self.assertEqual(sender.kind, "video")

        run(sender.send(RTCRtpParameters(codecs=[VP8_CODEC])))

        # receive RTCP feedback NACK
        packet = RtcpPsfbPacket(fmt=RTCP_PSFB_PLI, ssrc=1234, media_ssrc=sender._ssrc)
        run(sender._handle_rtcp_packet(packet))

        # clean shutdown
        run(sender.stop())

def test_track_ended(self):
        track = AudioStreamTrack()
        sender = RTCRtpSender(track, self.local_transport)
        run(sender.send(RTCRtpParameters(codecs=[PCMU_CODEC])))

        # stop track and wait for RTP loop to exit
        track.stop()
        run(asyncio.sleep(0.1))

0x74,
                0x78,  # RtpStreamId
                0xD5,
                0x73,
                0x74,
                0x72,
                0x65,
                0x61,
                0x6D,  # RepairedRtpStreamId
                0x00,
                0x00,  # Padding to 32bit boundary.
            ]
        )
        extensions_map = rtp.HeaderExtensionsMap()
        extensions_map.configure(
            RTCRtpParameters(
                headerExtensions=[
                    RTCRtpHeaderExtensionParameters(
                        id=2, uri="urn:ietf:params:rtp-hdrext:toffset"
                    ),
                    RTCRtpHeaderExtensionParameters(
                        id=4, uri="urn:ietf:params:rtp-hdrext:ssrc-audio-level"
                    ),
                    RTCRtpHeaderExtensionParameters(
                        id=6,
                        uri="http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
                    ),
                    RTCRtpHeaderExtensionParameters(
                        id=8,
                        uri="http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
                    ),
                    RTCRtpHeaderExtensionParameters(

def test_handle_rtcp_remb(self):
        sender = RTCRtpSender(VideoStreamTrack(), self.local_transport)
        self.assertEqual(sender.kind, "video")

        run(sender.send(RTCRtpParameters(codecs=[VP8_CODEC])))

        # receive RTCP feedback REMB
        packet = RtcpPsfbPacket(
            fmt=RTCP_PSFB_APP,
            ssrc=1234,
            media_ssrc=0,
            fci=pack_remb_fci(4160000, [sender._ssrc]),
        )
        run(sender._handle_rtcp_packet(packet))

        # receive RTCP feedback REMB (malformed)
        packet = RtcpPsfbPacket(fmt=RTCP_PSFB_APP, ssrc=1234, media_ssrc=0, fci=b"JUNK")
        run(sender._handle_rtcp_packet(packet))

        # clean shutdown
        run(sender.stop())

def test_handle_rtcp_nack(self):
        sender = RTCRtpSender(VideoStreamTrack(), self.local_transport)
        self.assertEqual(sender.kind, "video")

        run(sender.send(RTCRtpParameters(codecs=[VP8_CODEC])))

        # receive RTCP feedback NACK
        packet = RtcpRtpfbPacket(
            fmt=RTCP_RTPFB_NACK, ssrc=1234, media_ssrc=sender._ssrc
        )
        packet.lost.append(7654)
        run(sender._handle_rtcp_packet(packet))

        # clean shutdown
        run(sender.stop())

def test_padding_only_with_header_extensions(self):
        extensions_map = rtp.HeaderExtensionsMap()
        extensions_map.configure(
            RTCRtpParameters(
                headerExtensions=[
                    RTCRtpHeaderExtensionParameters(
                        id=2,
                        uri="http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
                    )
                ]
            )
        )

        data = load("rtp_only_padding_with_header_extensions.bin")
        packet = RtpPacket.parse(data, extensions_map)
        self.assertEqual(packet.version, 2)
        self.assertEqual(packet.marker, 0)
        self.assertEqual(packet.payload_type, 98)
        self.assertEqual(packet.sequence_number, 22138)
        self.assertEqual(packet.timestamp, 3171065731)

def test_send_keyframe(self):
        """
        Ask for a keyframe.
        """
        queue = asyncio.Queue()

        async def mock_send_rtp(data):
            if not is_rtcp(data):
                await queue.put(RtpPacket.parse(data))

        self.local_transport._send_rtp = mock_send_rtp

        sender = RTCRtpSender(VideoStreamTrack(), self.local_transport)
        self.assertEqual(sender.kind, "video")

        run(sender.send(RTCRtpParameters(codecs=[VP8_CODEC])))

        # wait for one packet to be transmitted, and ask for keyframe
        run(queue.get())
        sender._send_keyframe()

        # wait for packet to be transmitted, then shutdown
        run(asyncio.sleep(0.1))
        run(sender.stop())