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• ffmpeg doesn't work, when starting up it doesn't play music, it gives errors

  14 août 2024, par Оля Михеева

  import discord
from discord import FFmpegPCMAudio
import os
import random
from gtts import gTTS
import asyncio

TOKEN="***"
VOICE_CHANNEL_ID=11122224444

class Voice_Bot(discord.Client):
    def __init__(self):
        intents = discord.Intents.default()
        intents.message_content = True
        intents.voice_states=True
        super().__init__(intents=intents)
        print(os.getcwd())
        self.sounds_hello = os.listdir(os.path.join('sounds','hello'))
        self.sounds_bye = os.listdir('sounds\\bye')
        self.sounds = os.listdir('sounds\\nature')

    async def on_ready(self):        
        self.voice_channel = self.get_channel(VOICE_CHANNEL_ID) 
        if self.voice_channel == None:
            print('Не удалось подключиться к голосовому каналу.')
            return
        self.voice_client = await self.voice_channel.connect()
        print('Бот подключен к голосовому каналу')
        await self.text_to_speech("Lets play Guess the Tune")
        
    async def on_message(self,message):
        if message.author==self.user:
            return
        if message.content.startswith("game"):
            await self.text_to_speech("let's start the game guess the melody")
            music=os.listdir("sounds\\music")
            self.melody=random.choice(music)
            await self.play_sound(f"sounds\\music\\{self.melody}")
        elif message.content==self.melody[0:len(self.melody)-4]:
            if (self.voice_client.is_playing()):
                self.voice_client.stop()
            await self.text_to_speech(f"Congratulations, {message.author.name} answered correctly! To continue, type game")
        else:
            if (self.voice_client.is_playing()):
                self.voice_client.stop()
            await self.text_to_speech(f"Unfortunately, {message.author.name} did not guess. To continue, write game")


    async def on_voice_state_update(self,member,before,after):
        if member.id ==self.user.id:
            print('Someone entered or left the voice channel.')
        else:
            try:
                if before.channel == None:
                    print(f'{member.name} entered the voice channel {after.channel}.')
                    await self.play_sound(f'sounds\\hello\\{random.choice(self.sounds_hello)}')
                elif after.channel == None:
                    print(f'{member.name} left the voice channel {before.channel}.')
                    await self.play_sound(f'sounds\\bye\\{random.choice(self.sounds_bye)}')
            except Exception as e:
                print(f"Error in on_voise_state_update: {e}")

    async def text_to_speech(self,text):
        try:
            tts = gTTS(text=text, lang ="en")
            tts.save("text.mp3")
        except Exception as e:
            print(f"Error e: {e}")
        await self.voice_channel.send(text)
        await self.play_sound("text.mp3")

    def play_sound(self,path):
        print(path)
        source=discord.FFmpegPCMAudio(source=path, executable="ffmpeg\\bin\\ffmpeg.exe")
        if (self.voice_client.is_playing()):
            self.voice_client.stop()
        self.voice_client.play(source)                

client = Voice_Bot()
client.run(TOKEN)


  


  [enter image description here](https://i.sstatic.net/ys8Xza0w.jpg)

  


• ffmpeg nvenc_h264 streaming uhd2160 rawvideo. Youtube buffering

  23 janvier 2019, par Alex Deroza

  I use this batch file for starting my stream :

  echo starting transocding from native uhd2160 to fhd format.. Host: YouTube
  
  ffmpeg ^
  
      -loglevel -8 ^
  
      -f rawvideo -s:v 1920x1080 -r 60 -pix_fmt nv12 ^
  
      -f dshow -i video="Game Capture 4K60 Pro Video 01":audio="Game Capture 4K60 Pro Audio 01" ^
  
      -c:v h264_nvenc ^
  
      -level:v 4.2 ^
  
      -profile:v high ^
  
      -preset:v hq ^
  
      -b:v 8.8M ^
  
      -color_range 2 ^
  
      -colorspace bt709 ^
  
      -bf 0 ^
  
      -g 60 ^
  
      -2pass 1 ^
  
      -rc:v cbr ^
  
      -coder cabac ^
  
      -acodec aac ^
  
      -ab 128k ^
  
      -movflags +faststart ^
  
      -f flv rtmp://a.rtmp.youtube.com/live2/xxxx-xxxx-xxxx-xxxx && pause

  Stream on YouTube starts fine, and plays with good quality. But stream stops after when "Buffer Health" emptied. And in this situation Youtube says that "Video output low", "YouTube is not receiving enough video to maintain smooth streaming. As such, viewers will experience buffering."

  I can’t understand why it happens. Because I’ve a good ISP, and over 30 mbps inet to YT services within 90ms latency to them.

  YouTube is not receiving enough video to maintain smooth streaming. As
  such, viewers will experience buffering.

  Your encoder is sending data faster than realtime (multipleseconds of
  video each second). You must rate limit your livevideo upload to
  approximately 1 second of video each second.

  The stream’s current bitrate (5730.00 Kbps) is lower than the
  recommended bitrate. We recommend that you use a stream bitrate of
  4500 Kbps.

• How to extract frames in sequence as PNG images from ffmpeg stream ?

  7 janvier, par JamesJGoodwin

  I'm trying to create a program that would capture my screen (a game to be precise) using ffmpeg and stream frames to NodeJS for live processing. So, if the game runs at 60 fps then I expect ffmpeg to send 60 images per second down to stdout. I've written a code for that

  


      import { spawn as spawnChildProcess } from 'child_process';

    const videoRecordingProcess = spawnChildProcess(
      ffmpegPath,
      [
        '-init_hw_device',
        'd3d11va',
        '-filter_complex',
        'ddagrab=0,hwdownload,format=bgra',
        '-c:v',
        'png',
        '-f',
        'image2pipe',
        '-loglevel',
        'error',
        '-hide_banner',
        'pipe:',
      ],
      {
        stdio: 'pipe',
      },
    );

    videoRecordingProcess.stderr.on('data', (data) => console.error(data.toString()));

    videoRecordingProcess.stdout.on('data', (data) => {
      fs.promises.writeFile(`/home/goodwin/genshin-repertoire-autoplay/imgs/${Date.now()}.bmp`, data);
    });


  


  Currently I'm streaming those images onto disk for debugging and it's almost working except that the image is cropped. Here's what's going on. I get 4 images saved on disk :

  


  

  1. Valid image that is 2560x1440, but only 1/4 or even 1/5 of the screen is present at the top, the remaining part of the image is empty (transparent)

  


  2. Broken image that won't open

  


  3. Broken image that won't open

  


  4. Broken image that won't open

  


  


  This pattern is nearly consistent. Sometimes it's 3, sometimes 4 or 5 images between valid images. What did I do wrong and how do I fix it ? My guess is that ffmpeg is streaming images in chunks, each chunk represents a part of the frame that was already processed by progressive scan. Though I'm not entirely sure if I should try and process it manually. There's gotta be a way to get fully rendered frames in one piece sequentially.