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• I need my music bot to play on multiple servers at the same time

  12 mars 2024, par Ondosh

  I'm writing my music bot in Python and I want it to be able to play music on multiple servers at once. Now the attempt to play on several servers looks like this : I start music on the first server, go to the second and start other music there, but the one that was requested on the first one is playing.
Know my code is :

  


  from nextcord.ext import commands
import nextcord, random
import yt_dlp
import datetime
import lazy_queue as lq

FFMPEG_OPTIONS = {
    "before_options": "-reconnect 1 -reconnect_streamed 1 -reconnect_delay_max 5",
    'options': '-vn'}

YDL_OPTIONS = {
    'format': 'bestaudio/best',
    'extractaudio': True,
    'noplaylist': True,
    'keepvideo': False,
    'postprocessors': [{
        'key': 'FFmpegExtractAudio',
        'preferredcodec': 'mp3',
        'preferredquality': '320'
    }]
}

bot.remove_command("help")
songs_queue = lq.Queue()
loop_flag = False

@bot.command()
async def add(ctx, *url):
    url = ' '.join(url)
    with yt_dlp.YoutubeDL(YDL_OPTIONS) as ydl:
        try:
            info = ydl.extract_info(url, download=False)
        except:
            info = ydl.extract_info(f"ytsearch:{url}",
                                    download=False)['entries'][0]
    URL = info['url']
    name = info['title']
    time = str(datetime.timedelta(seconds=info['duration']))
    songs_queue.q_add([name, time, URL])
    embed = nextcord.Embed(description=f'Записываю [{name}]({url}) в очередь 📝',
                           colour=nextcord.Colour.red())
    await ctx.message.reply(embed=embed)


def step_and_remove(voice_client):
    if loop_flag:
        songs_queue.q_add(songs_queue.get_value()[0])
    songs_queue.q_remove()
    audio_player_task(voice_client)


def audio_player_task(voice_client):
    if not voice_client.is_playing() and songs_queue.get_value():
        voice_client.play(nextcord.FFmpegPCMAudio(
            executable="ffmpeg\\bin\\ffmpeg.exe",
            source=songs_queue.get_value()[0][2],
            **FFMPEG_OPTIONS),
            after=lambda e: step_and_remove(voice_client))


@bot.command()
async def play(ctx, *url):
    await join(ctx)
    await add(ctx, ' '.join(url))
    await ctx.message.add_reaction(emoji='🎸')
    voice_client = ctx.guild.voice_client
    audio_player_task(voice_client)
async def queue(ctx):
    if len(songs_queue.get_value()) > 0:
        only_names_and_time_queue = []
        for i in songs_queue.get_value():
            name = i[0]
            if len(i[0]) > 30:
                name = i[0][:30] + '...'
            only_names_and_time_queue.append(f'📀 `{name:<33}   {i[1]:>20}`\n')
        c = 0
        queue_of_queues = []
        while c < len(only_names_and_time_queue):
            queue_of_queues.append(only_names_and_time_queue[c:c + 10])
            c += 10
        embed = nextcord.Embed(title=f'ОЧЕРЕДЬ [LOOP: {loop_flag}]',
                               description=''.join(queue_of_queues[0]),
                               colour=nextcord.Colour.red())
        await ctx.send(embed=embed)
        for i in range(1, len(queue_of_queues)):
            embed = nextcord.Embed(description=''.join(queue_of_queues[i]),
                                   colour=nextcord.Colour.red())
            await ctx.send(embed=embed)
    else:
        await ctx.send('Очередь пуста')


  


  As far as I understand, I need to create variables that will be associated with the server ID, but I do not know how to do this.

  


• when ffpmeg drops frames some things aren't played back in real time

  8 février 2024, par Alex028502

  I am trying to run a bunch of ffpmeg processes that act as simulators for cameras, and something funny is happening when I the processor can't keep up with the configured frame rate.

  


  I have replaced the rtsp stream with an output file, and managed to reproduce the issue, so will just show that to keep it simple.

  


  First here is a makefile that creates my source movie :

  


  clock.mp4: Makefile
    rm -f $@
    ffmpeg -f lavfi -i color=c=black:s=4096x2160:r=25 -vf \
"drawtext=fontfile=/usr/share/fonts/truetype/dejavu/DejaVuSans-Bold.ttf:fontsize=72:fontcolor=white:x=(w-text_w)/2:y=(h-text_h)/2: \
text='%{eif\:trunc(n/25)\:d}':start_number=0:rate=25" \
-t 60 -r 25 $@


  


  that gives me a one minute long movie that prints the second number to the screen. I have tested it out and the seconds are close enough. I put a lot of pixels to make it easier to jam up my CPU.

  


  Here is the script that creates a processes similar to the one I am trying to debug (called experiment.sh)

  


  I am actually using H.264, but H.265 is easier to overwhelm the processor with

  


  #! /usr/bin/env bash

set -e

echo starting > message$1.txt

rm -f superclock$1.mp4
ffmpeg -re -stream_loop -1 -i clock.mp4 \
       -an -vcodec libx265 -preset ultrafast -sc_threshold -1 -x265-params repeat-headers=1 \
       -vf "drawtext=fontfile=/usr/share/fonts/truetype/dejavu/DejaVuSans-Bold.ttf:fontsize=24:fontcolor=white:x=10:y=10:text='%{localtime\:%X}', \
            drawtext=fontfile=/usr/share/fonts/truetype/dejavu/DejaVuSans-Bold.ttf:fontsize=24:fontcolor=white:x=10:y=(h-text_h-10):textfile=message$1.txt:reload=1, \
            scale=1920x1080,fps=25" \
       -b:v 3M -minrate 3M -maxrate 3M \
       -bufsize 6M -g 25 superclock$1.mp4 &
pid=$!

for x in $(seq 0 10)
do
    echo $x > message$1.txt
    sleep 10
done

kill -INT $pid || true


  


  It should

  


  

  • put the second in the middle of the screen - 'cause it gets it from the source video

  


  • put the approximate sixth of minute in the lower left corner
(only approximate because of the sleep but close enough)

  


  • put the the wall clock time in the upper left corner

  


  


  and it works

  


  make clock.mpg
./experiment.sh 0
vlc superclock0.mp4


  


  shows something like this


  


  Now here is the interesting part

  


  If I run the script in four different terminals at the same time

  


  ./experiment.sh 1
./experiment.sh 2
./experiment.sh 3
./experiment.sh 4


  


  It can't keep up with the frame rate, and I see this in the output :

  


  frame= 1515 fps= 16 q=0.0 size=     768kB time=00:00:59.96 bitrate= 104.9k


  


  I was hoping the end result would all look ok when I watch it except with fewer frames, but the timestamps of the frames would make it all work as expected

  


  However...

  


  

  • The time in the middle, that is inherited from the source video, the seconds in the middle of the screen, stays in sync with VLC's clock.

  


  • the wall clock in the upper left seems play at 150% speed

  


  • the every ten seconds incrementor in the lower left seems to increment every 7 seconds

  


  • the video is only 1:25 long even though it was recording for at least 1:40 according to sleeps

  


  • the wall clock in the upper right hand corner makes it more than 1'40" and then counter in the lower left makes it to 10.

  


  


  

  


  Here are four states to compare

  


  |                  | Start    | 30" in   | end      |
|------------------+----------+----------|----------|
| Video Time       | 00:00    | 00:30    | 01:24    |
| Wall Clock Time  | 15:05:50 | 15:06:39 | 15:07:39 |
| sixth of minute  | 0        | 4        | 10       |
| seconds counter  | 0        | 30       | 24       |


  


  

  


  So you can see the vlc clock keeps pace with the original clock from the source movie.. even when it is only able to produce frames at 2/3 of the rate. However, the it is taking 50% long to get through the whole source movie I guess ?

  


  I am having trouble coming up with a theory that can explain exactly how this happens.

  


  Does anybody know how I can "correct" this ? (make it so that the movie is played at the same rate that it is recorded)

  


  I am thinking of using a lower frame rate and size as the input video.. but it would be nice to have something that will always work as expected, just with a lower frame rate, no matter how busy the processor is.

  


• Use ffmpeg to record from RTSP stream to two outputs at different resolutions while segmenting the video based on time

  7 juin 2023, par Eric Hansen

  I need to record from an RTSP at two different resolutions (original and something lower). I also need to have the files broken up according to clock time on every even 30 minutes, ex. 1:00, 1:30, 2:00, etc.

  


  I imagine the command would look like this.

  


  ffmpeg -rtsp_transport tcp -i rtsp://<rtsp url="url"> \&#xA;  -filter_complex "[0:v]split=2[v1][v2];[v1]scale=1920:1080[out1];[v2]scale=448:252[out2]" \&#xA;  -map "[out1]" -c:v -f segment -segment_format mp4 -segment_time 00:30:00 -strftime 1 \ &#xA;  /mnt/data/original-%Y-%m-%d-%H.%M.%S.mp4 \&#xA;  -map "[out2]" -c:v -f segment -segment_format mp4 -segment_time 00:30:00 -strftime 1 \ &#xA;  /mnt/data/low-reso-%Y-%m-%d-%H.%M.%S.mp4 \&#xA;</rtsp>

  


  Of course, this doesn't work. The command above gives me this error.

  


  [NULL @ 0x556c999c2140] Unable to find a suitable output format for 'segment'
segment: Invalid argument


  


  I've tried so many perturbations of this command to get it to work without any luck. Does anyone know how to do this ?