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• Music bot returning errors at core.py and bot.py

  13 septembre 2021, par Loganox

  The code for the music bot I'm trying to write in python keeps returning an error with the following code :

  


  @bot.command(name='play_song', help='To play song')
async def play(ctx,url):
    
    #if not ctx.message.author.name=="Rohan Krishna" :
    #     await ctx.send('NOT AUTHORISED!')
    #     return
    if 1==1:#try :
        server = ctx.message.guild
        voice_channel = server.voice_client
        print("try 1 was a success")
        
        async with ctx.typing():
            filename = await YTDLSource.from_url(url, loop=bot.loop)
            #voice_channel.play(discord.FFmpegPCMAudio(executable="ffmpeg.exe", source=filename)) #ISSUE HERE
            print("try 2 was a success")
            voice = get(ctx.bot.voice_clients, guild=ctx.guild)
    
            #voice.play(discord.FFmpegPCMAudio('test.mp3'), after=your_check)
            #voice.source = discord.PCMVolumeTransformer(voice.source)
            #voice.source.volume = 0.5
        await ctx.send('**Now playing:** {}'.format(filename))
    if 1==0:#except:
        await ctx.send("The bot is not connected to a voice channel.")
        print("try 3 was unfortunately a success")


  


  It normally has a try/except function but I replaced it with if true and false statements to force it to run instead of breaking to determine the exact line of it breaking. The issue supposedly lies in

  


  voice_channel.play(discord.FFmpegPCMAudio(executable="ffmpeg.exe", source=filename)) #ISSUE HERE


  


  The error I get returned is :

  


  Traceback (most recent call last):
  File "C:\Users\heyin\anaconda3\lib\site-packages\discord\ext\commands\core.py", line 85, in wrapped
    ret = await coro(*args, **kwargs)
  File "app2.py", line 75, in play
    voice.play(discord.FFmpegPCMAudio('test.mp3'), after=your_check)
  File "C:\Users\heyin\anaconda3\lib\site-packages\discord\player.py", line 225, in __init__
    super().__init__(source, executable=executable, args=args, **subprocess_kwargs)
  File "C:\Users\heyin\anaconda3\lib\site-packages\discord\player.py", line 138, in __init__
    self._process = self._spawn_process(args, **kwargs)
  File "C:\Users\heyin\anaconda3\lib\site-packages\discord\player.py", line 147, in _spawn_process
    raise ClientException(executable + ' was not found.') from None
discord.errors.ClientException: ffmpeg was not found.

The above exception was the direct cause of the following exception:

Traceback (most recent call last):
  File "C:\Users\heyin\anaconda3\lib\site-packages\discord\ext\commands\bot.py", line 939, in invoke
    await ctx.command.invoke(ctx)
  File "C:\Users\heyin\anaconda3\lib\site-packages\discord\ext\commands\core.py", line 863, in invoke
    await injected(*ctx.args, **ctx.kwargs)
  File "C:\Users\heyin\anaconda3\lib\site-packages\discord\ext\commands\core.py", line 94, in wrapped
    raise CommandInvokeError(exc) from exc
discord.ext.commands.errors.CommandInvokeError: Command raised an exception: ClientException: ffmpeg was not found.


  


  I added the print statements to determine which lines were working so those can obviously be ignored. It seems to be working fine except for this, other commands in the bot are working appropriately.

  


• Cannot use pydub.AudioSegment on audio files downloaded by youtube-dl library

  26 octobre 2018, par Han M

  After I download a audio clip from a youtube link by the youtube-dl library, I cannot use pydub.AudioSegment to slice the audio file. The following is the information about my code.

  import youtube_dl
  
  options = {
  
          'format': 'bestaudio/best',
  
          'extractaudio': True,  # only keep the audio
  
          'audio-format': "wav",  # convert to wav
  
          'outtmpl': whole_path,  # name the file the ID of the video
  
          'noplaylist': True,  # only download single song, not playlist
  
          'audioquality': 1
  
  }
  
  with youtube_dl.YoutubeDL(options) as ydl:
  
      print('url is:', web_url)
  
      ydl.download([web_url])
  
  
  
  from pydub import AudioSegment
  
  audio_data = AudioSegment.from_wav(path_to_downloaded_file)

  Then, I got an error as follows first on Mac and second on Linux :

  b'avconv version 12.3, Copyright (c) 2000-2018 the Libav developers\n  built on Jul 26 2018 18:08:50 with Apple LLVM version 9.1.0 (clang-902.0.39.2)\n/Users/***/random17560390.wav: Invalid data found when processing input\n'
  
  
  
  b'ffmpeg version 2.8.15-0ubuntu0.16.04.1 Copyright (c) 2000-2018 the FFmpeg developers\n  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.10) 20160609\n  configuration: --prefix=/usr --e
  
  xtra-version=0ubuntu0.16.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-
  
  shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-li
  
  bbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmod
  
  plug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-li
  
  bspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --e
  
  nable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv\n  libavutil      54. 31.100 / 54.
  
   31.100\n  libavcodec     56. 60.100 / 56. 60.100\n  libavformat    56. 40.101 / 56. 40.101\n  libavdevice    56.  4.100 / 56.  4.100\n  libavfilter     5. 40.101 /  5. 40.101\n  libavresa
  
  mple   2.  1.  0 /  2.  1.  0\n  libswscale      3.  1.101 /  3.  1.101\n  libswresample   1.  2.101 /  1.  2.101\n  libpostproc    53.  3.100 / 53.  3.100\n[wav @ 0x20854c0] invalid start
  
   code [0][0][0][28] in RIFF header\n/home/users/test_download/random10485395.wav: Invalid data found when processing input\n'

  I think the problem might be about the encoding on the downloaded files by youtube-dl. I can use pydub.AudioSegment on other normal .wav files. After checking the details of the downloaded file by youtube-dl, I found the following feature :

  Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 125 kb/s (default)

  It is different from other normal .wav files, which have the following :

  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s

• varying RTP stream result from custom SIP implementation

  1er février, par Nik Hendricks

  I am in the process of creating my own SIP implementation in Node.js. As well as a b2bua as a learning project.

  


  Finding people wise in the ways of SIP has proved to be difficult elsewhere but here I have had good results

  


  this is the GitHub of my library so far node.js-sip

  


  this is the GitHub of my PBX so far FlowPBX

  


  Currently, everything is working as I expect. Although I really have some questions on possible errors in my implementation.

  


  My main issue is with RTP streams. Currently I am utilizing ffmpeg.

  


  my function goes as follows

  


  start_stream(call_id, sdp){
        console.log('Starting Stream')
        let port = sdp.match(/m=audio (\d+) RTP/)[1];
        let ip = sdp.match(/c=IN IP4 (\d+\.\d+\.\d+\.\d+)/)[1];
        let codec_ids = sdp.match(/m=audio \d+ RTP\/AVP (.+)/)[1].split(' ');
        let ffmpeg_codec_map = {
            'opus': 'libopus',
            'PCMU': 'pcm_mulaw',
            'PCMA': 'pcm_alaw',
            'telephone-event': 'pcm_mulaw',
            'speex': 'speex',
            'G722': 'g722',
            'G729': 'g729',
            'GSM': 'gsm',
            'AMR': 'amr',
            'AMR-WB': 'amr_wb',
            'iLBC': 'ilbc',
            'iSAC': 'isac',
        }

        let codecs = [];
        sdp.split('\n').forEach(line => {
            if(line.includes('a=rtpmap')){
                let codec = line.match(/a=rtpmap:(\d+) (.+)/)[2];
                let c_id = line.match(/a=rtpmap:(\d+) (.+)/)[1];
                codecs.push({                    
                    name: codec.split('/')[0],
                    rate: codec.split('/')[1],
                    channels: codec.split('/')[2] !== undefined ? codec.split('/')[2] : 1,
                    id: c_id
                })
            }
        })

        console.log('codecs')
        console.log(codecs)

        let selected_codec = codecs[0]
        if(selected_codec.name == 'telephone-event'){
            selected_codec = codecs[1]
            console.log(selected_codec)
        }

        //see if opus is available
        codecs.forEach(codec => {
            if(codec.name == 'opus'){
                selected_codec = codec;
            }
        })

        if(selected_codec.name != 'opus'){
            //check if g729 is available
            codecs.forEach(codec => {
                if(codec.name == 'G729'){
                    selected_codec = codec;
                }
            })
        }

        console.log('selected_codec')
        console.log(selected_codec)

        let spawn = require('child_process').spawn;
        let ffmpegArgs = [
            '-re',
            '-i', 'song.mp3',
            '-acodec', ffmpeg_codec_map[selected_codec.name],
            '-ar', selected_codec.rate,
            '-ac', selected_codec.channels,
            '-payload_type', selected_codec.id,
            '-f', 'rtp', `rtp://${ip}:${port}`
        ];

        let ffmpeg = spawn('ffmpeg', ffmpegArgs);

        ffmpeg.stdout.on('data', (data) => {
            console.log(`stdout: ${data}`);
        });
        ffmpeg.stderr.on('data', (data) => {
            console.error(`stderr: ${data}`);
        });




}


  


  When using zoiper to test it works great. I have seen the mobile version negotiate speex
and the desktop version negotiate opus mostly for the codec.

  


  today I tried to register a grandstream phone to my pbx and the rtp stream is blank audio.
opus is available and I have tried to prefer that in my stream but still even when selecting that I cannot get audio to the grandstream phone. This is the same case for a yealink phone. I can only get zoiper to work so far.

  


  what could be causing this behavior ? there is a clear path of communication between everything just like the zoiper client's I have used.

  


  Additionally in my sip implementation,
how important is the concept of a dialog ? currently, I just match messages by Call-ID

  


  and then choose what to send based on the method or response. is there any other underlying dialog functionality that I may need to implement ?

  


  It would just be awesome to get someone who really knows what they are talking about eyes on some of my code to direct this large codebase in the right direction but I realize that a big ask lol.