import assemblyai as aai
import discord
from discord.ext import commands, voice_recv
import logging
from discord.opus import Decoder as OpusDecoder
import wave

# Replace with your API key
aai.settings.api_key = "aa9962f0088a449a9c4ab2361e96cc08"
discord.opus._load_default()

# URL of the file to transcribe
FILE_URL = "https://github.com/AssemblyAI-Community/audio-examples/raw/main/20230607_me_canadian_wildfires.mp3"

# You can also transcribe a local file by passing in a file path
# FILE_URL = './path/to/file.mp3'

logger = logging.getLogger("discord")


class SRBuffer(voice_recv.AudioSink):
    """Endpoint AudioSink that generates a wav file.
    Best used in conjunction with a silence generating sink. (TBD)
    """

    CHANNELS = OpusDecoder.CHANNELS
    SAMPLE_WIDTH = OpusDecoder.SAMPLE_SIZE // OpusDecoder.CHANNELS
    SAMPLING_RATE = OpusDecoder.SAMPLING_RATE

    ##on member join - 5 plików buforowych na człowieka otwarty jest tylko aktualny i następny
    #on member disconnect - dropujemy go


    #        event: BasicSinkWriteCB,
    #        rtcp_event: Optional[BasicSinkWriteRTCPCB] = None,
    #        self.cb = event
    #        self.cb_rtcp = rtcp_event
    #   def write(self, user: Optional[User], data: VoiceData) -> None:
    #       self.cb(user, data)
    #   @AudioSink.listener()
    #   def on_rtcp_packet(self, packet: RTCPPacket, guild: discord.Guild) -> None:
    #       self.cb_rtcp(packet) if self.cb_rtcp else None



    def __init__(self, destination):
        super().__init__()

        self._file: wave.Wave_write = wave.open(destination, 'wb')
        self._file.setnchannels(self.CHANNELS)
        self._file.setsampwidth(self.SAMPLE_WIDTH)
        self._file.setframerate(self.SAMPLING_RATE)
        self.user_list = []

    def wants_opus(self) -> bool:
        return False

    def write(self, user, data) -> None:
        logger.info("writing to file %s", self._file)
        logger.info("user: %s", user)
        self._file.writeframes(data.pcm)

    def cleanup(self) -> None:
        try:
            self._file.close()
        except Exception:
            logger.warning("WaveSink got error closing file on cleanup", exc_info=True)



class Transcriber(commands.Cog):
    def __init__(self, bot):
        def callback(user, data):
            logger.info(f"Got packet from {user}")

        self.bot = bot
        self._last_member = None
        self.wsink = voice_recv.SRBuffer(destination="/home/pi/Conjurer/wav.wav")

    async def transcribe():
        config = aai.TranscriptionConfig(speaker_labels=True)

        transcriber = aai.Transcriber()
        transcript = transcriber.transcribe(
        FILE_URL,
        config=config
        )
        for utterance in transcript.utterances:
          print(f"Speaker {utterance.speaker}: {utterance.text}")
    @commands.hybrid_command(name="transcribe")
    async def test(self, ctx):
        logger.info("Attempt transcribe")
        vc = await ctx.author.voice.channel.connect(cls=voice_recv.VoiceRecvClient)
        logger.info("Connected")
        vc.listen(self.wsink)


    @commands.command(name="stop_transcribe")
    async def stop(self, ctx):
        await ctx.voice_client.disconnect()

async def setup(bot):
    await bot.add_cog(Transcriber(bot))


#1. zapisuj kwestie człowieka do pliku w którym będzie można stwierdzić kto co powiedział (callback z basicaudio + write z wavesinka). Zamykaj plik i wysyłaj do transkrypcji w momencie ciszy dłuższej niż 0.5s. Jeśli człowiek nadaje cały czas dawaj sygnał że nie można go transkrybować - i wyłaczaj zapis. 
#2. Transkrypcja to abstrakt - w zależności od tego która metoda jest włączona wysyła do odpowiedniego silnika, silniki offline odpalam na activcomie. Zaczynamy od assemblyai bo najlepiej wspiera polski. Potem zobaczymy co dalej.