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• Making my Discord Bot automatically play music from WAV on loop

  5 décembre 2022, par Mativ9

  So I was trying to make a Discord Bot in Python, which would atomatically join a voice channel and play my own music from a list in a loop. So far it's joining the channel, shuffling the list so the music is on random, but when I try to write a code so after one song it will play the next one it crushes and doesn't play anything (tho it's joining the channel)

  


  import discord
import random
from discord.ext import commands
from discord import FFmpegPCMAudio

#playlist as a list
queue = [FFmpegPCMAudio('Iceland1.wav'), FFmpegPCMAudio('Iceland2.wav'), FFmpegPCMAudio('Iceland3.wav'), FFmpegPCMAudio('Iceland4.wav'),
         FFmpegPCMAudio('Iceland5.wav'), FFmpegPCMAudio('Iceland6.wav'), FFmpegPCMAudio('Iceland7.wav'), FFmpegPCMAudio('Iceland8.wav'),
         FFmpegPCMAudio('Iceland9.wav'), FFmpegPCMAudio('Iceland10.wav'), FFmpegPCMAudio('Norway1.wav'), FFmpegPCMAudio('Norway2.wav'),
         FFmpegPCMAudio('Norway3.wav'), FFmpegPCMAudio('Norway4.wav'), FFmpegPCMAudio('Norway5.wav'), FFmpegPCMAudio('Norway6.wav'),
         FFmpegPCMAudio('Norway7.wav'), FFmpegPCMAudio('Norway8.wav'), FFmpegPCMAudio('Norway9.wav'), FFmpegPCMAudio('Norway10.wav'),
         FFmpegPCMAudio('Norway11.wav'), FFmpegPCMAudio('Presents1.wav'), FFmpegPCMAudio('Presents2.wav'), FFmpegPCMAudio('Presents3.wav'),
         FFmpegPCMAudio('Presents4.wav'), FFmpegPCMAudio('Presents5.wav'), FFmpegPCMAudio('Presents6.wav'), FFmpegPCMAudio('Presents7.wav'),
         FFmpegPCMAudio('Presents8.wav'), FFmpegPCMAudio('Presents9.wav'), FFmpegPCMAudio('Presents10.wav'), FFmpegPCMAudio('Autumn1.wav'),
         FFmpegPCMAudio('Autumn2.wav'), FFmpegPCMAudio('Autumn3.wav'), FFmpegPCMAudio('Autumn4.wav'), FFmpegPCMAudio('Autumn5.wav'),
         FFmpegPCMAudio('Autumn6.wav'), FFmpegPCMAudio('Autumn7.wav'), FFmpegPCMAudio('Autumn8.wav'), FFmpegPCMAudio('Covers1.wav'),
         FFmpegPCMAudio('Covers2.wav'), FFmpegPCMAudio('Covers3.wav'), FFmpegPCMAudio('Covers4.wav'), FFmpegPCMAudio('Covers5.wav'),
         FFmpegPCMAudio('Covers6.wav'), FFmpegPCMAudio('Covers7.wav'), FFmpegPCMAudio('Covers8.wav'), FFmpegPCMAudio('Covers9.wav'),
         FFmpegPCMAudio('Covers10.wav'), FFmpegPCMAudio('Covers11.wav'), FFmpegPCMAudio('Covers12.wav')]

intents = discord.Intents.default()
intents.message_content = True
client = commands.Bot(command_prefix='>', intents=intents)

@client.event
async def on_ready():
    global voice
    print("The Matt Bot is ready")
    print("--------------------------")
    await client.change_presence(activity=discord.Game('Matt Krupa')) #makes my bot play Matt Krupa
    channel = client.get_channel(thechannelid) #geting channel ID
    voice = await channel.connect() #connecting to channel
    random.shuffle(queue) #randomazing the playlist
    def after_song(): #moving the first song to the end so its on loop, and playling the next one
        queue.append(queue[0])
        del queue[0]
        player = await voice.play(queue[0], after=await after_song())
    player = await voice.play(queue[0], after=await after_song()) #plays song from the playlist, after the song doing the after_song() function

client.run(mytokenidontwanttoshowitsry)


  


  I wanted it to play all the songs on the infinite loop, i can't find how to correctly detect the end of a song...

  


• TypeError : parameter 'url' is missing a type annotation in callback 'play' + ffmpeg

  10 décembre 2022, par morichh

  I can't understand why Python interpreter doesn't like the fact that I use the url in the last line

  


  @client.tree.command()
async def play(interaction: Interaction, url = str):
    global voice
    channel = interaction.user.voice.channel
    voice = get(client.voice_clients, Guild=interaction.guild)
    if voice and voice.is_connected():
        await voice.move_to(channel)
        await interaction.response.send_message(f"*1*")
    else:
        voice = await channel.connect()
        await interaction.response.send_message(f"*2*")
    voice.play(discord.FFmpegPCMAudio(executable="C:/ffmpeg/bin/ffmpeg.exe", source = url))    



  


  By the way, ffmpeg also does not work.

  


• SDL2 won't play with more than 6 audio channels

  13 juin 2020, par Hiko Haieto

  I am trying to stream (raw) video and audio from a capture device as part of my home media setup (with my pc acting similarly to a receiver in a typical home theatre setup), but the biggest problem I haven't been able to get past is that I haven't been able to get ffplay (using SDL2 as its audio backend) to work with all 8 channels in 7.1 streams - two simply get dropped, despite it recognising 8 channel input or me specifying a 7.1 layout.

  



  I have been able to confirm that all 8 channels are present in the source by first using ffmpeg to save the output of a speaker test to a file and playing that back with both mplayer (which works) and ffplay (which doesn't). I also wrote some minimal code to play the audio directly through SDL's API with the same result, so it's not the fault of ffplay. I might simply use mplayer if it weren't for the fact that piping output from ffmpeg adds too much latency for real-time use. I am using libSDL version 2.0.12 and ffplay 4.2.3, both of which are the latest at the time of writing and are ostensibly supposed to support 7.1 audio.

  



  Using output recorded from speaker-test -c 8, I am using the following to play it back in mplayer :

  



  mplayer -channels 8 -rawaudio channels=8 -format s16le -demuxer rawaudio speaker-test.pcm


  



  and the following to play it back in ffplay :

  



  ffplay -f s16le -ac 8 -af 'channelmap=channel_layout=7.1' speaker-test.pcm


  



  No matter what I try, the two side channels get dropped. I couldn't figure out how to play raw pcm in SDL, so I repeated the same tests with wav output and used the following code to play it back :

  



  #include <sdl2></sdl2>SDL.h>&#xA;&#xA;int main(int argc, char **argv) {&#xA;    SDL_Init(SDL_INIT_AUDIO);&#xA;    SDL_AudioSpec wavSpec;&#xA;    Uint32 wavLength;&#xA;    Uint8 *wavBuffer;&#xA;    SDL_LoadWAV("speaker-test.wav", &amp;wavSpec, &amp;wavBuffer, &amp;wavLength);&#xA;    SDL_AudioDeviceID deviceID = SDL_OpenAudioDevice(NULL, 0, &amp;wavSpec, NULL, 0);&#xA;    SDL_QueueAudio(deviceID, wavBuffer, wavLength);&#xA;    SDL_PauseAudioDevice(deviceID, 0);&#xA;    SDL_Delay(30000);&#xA;    SDL_CloseAudioDevice(deviceID);&#xA;    SDL_FreeWAV(wavBuffer);&#xA;    SDL_Quit();&#xA;    return 0;&#xA;}&#xA;

  



  The above code exhibits the same behaviour of dropping the two additional side channels, despite it being the latest version of SDL that should have supported 7.1 for many releases now. Why might this be happening, and how might I fix it ?