Recherche avancée

Sur d’autres sites (14007)

• ffmpeg stops reading rtsp livestream after some seconds

  26 décembre 2020, par user3019668

  I'm trying to save locally a rtsp livestream with ffmpeg but it stops after a few seconds. It works smooth, but suddently it just stops, more or less always after 10-15 seconds.

  



  This is the command :

  



  ffmpeg -rtsp_transport tcp -i rtsp://xxx.xxx.xxx.xxx:554/test.sdp -c copy test.ts


  



  And this is the log. I tried with previous ffmpeg versions with the same result :

  



  ffmpeg version git-2020-04-26-1128aa8 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200328
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  libavutil      56. 43.100 / 56. 43.100
  libavcodec     58. 82.100 / 58. 82.100
  libavformat    58. 42.101 / 58. 42.101
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 79.100 /  7. 79.100
  libswscale      5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
Input #0, rtsp, from 'rtsp://xxx.xxx.xxx.xxx:554/test.sdp':
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: h264 (Main), yuv420p(top first), 1920x1080, 25 fps, 50 tbr, 90k tbn, 50 tbc
    Stream #0:1: Audio: aac (LC), 48000 Hz, 4.0, fltp
Output #0, mpegts, to 'test.ts':
  Metadata:
    encoder         : Lavf58.42.101
    Stream #0:0: Video: h264 (Main), yuv420p(top first), 1920x1080, q=2-31, 25 fps, 50 tbr, 90k tbn, 90k tbc
    Stream #0:1: Audio: aac (LC), 48000 Hz, 4.0, fltp
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame=  387 fps= 21 q=-1.0 Lsize=   18405kB time=00:00:16.91 bitrate=8912.2kbits/s speed=0.923x
video:17295kB audio:575kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.993069%


  



  Debug shows also "No more output streams to write to, finishing" when it stops.

  



  Could you help me, please ? I don't know what else to try... Thank you.

  


• FFmpeg doesn't play audio, yet no error shown

  4 août 2023, par Kristupas

  So i'm learning Python and don't know much about FFmpeg. I am following a tutorial, which explains everything very clearly. Everything is working, with one exception. Whenever I try to get it to play a sound, it won't.. Here's what it is saying :

  


  INFO     discord.player ffmpeg process 2540 successfully terminated with return code of 1.

  


  And here's my code (forgive me for all of the childish things in there, i'm just trying out different features) :

  


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


Token = "No token for you :)"
client = commands.Bot(command_prefix = '!', intents=discord.Intents.all())


@client.event
async def on_ready():
    print(f"we're rolling as {client.user} \n")
    channel = client.get_channel(1022535885851459727)
    await channel.send("Tremble before my might hoomans😤😤")

#Member events:

@client.event
async def on_member_join(member):
    await member.send("Ok comrade, welcome to bot lab, pls not leave. Anways here rules \n1. No swearing \n2. No cursing \n3. No bullying, the owner is a crybaby \n4. No following the rules (u get banned if this one is broken)")
    channel = client.get_channel(1136658873688801302)
    jokes = [f"A failure known as {member} has joined this chat!", 
             f"Another {member} has joined the channel", 
             f"A {member} spawned", 
             f'cout << "{member} has joined teh chat" << endl;', 
             f"OUR great {member} has come to save us" ]
    await channel.send(jokes[random.randint(0,len(jokes))])

@client.event 
async def on_member_remove(member):
    await member.send("Bye our dear comrade")
    channel = client.get_channel(1136663317738442752)
    await channel.send(f"{member} has left the chat :(.)")

#Client commands:
    
@client.command()
async def hello(ctx):
    await ctx.send("Hello, I am pro bot")

@client.command()
async def byelol(ctx):
    await ctx.send("Bye, I am pro bot")
@client.command()
async def ping(ctx):
    await ctx.send(f"**pong** {ctx.message.author.mention}")


@client.event
async def on_message(message):
    message.content = message.content.lower()
    await client.process_commands(message)


#voice channel commands:

@client.command(pass_context = True)
async def micup(ctx):
    if (ctx.author.voice):
        await ctx.send(f"Joining on {ctx.message.author}'s command")
        channel = ctx.message.author.voice.channel
        voice = await channel.connect()
        source = FFmpegPCMAudio('Bluetooth.wav')
        player = voice.play(source)
        
        
        
    else:
        await ctx.send("No.")



@client.command(pass_Context = True)
async def leave(ctx):
    if (ctx.voice_client):
        await ctx.send(f"Leaving on {ctx.message.author}'s command")
        await ctx.guild.voice_client.disconnect()
    else:
        await ctx.send("Nyet. Im not in voice chat u stoopid hooman")


@client.command(pass_Context = True)
async def pause(ctx):
    voice = discord.utils.get(client.voice_clients, guild = ctx.guild)
    if voice.is_playing():
        await ctx.send("Pausing..⏸")
        voice.pause()
    else:
        await ctx.send("I don't think I will.")

@client.command(pass_Context = True)
async def resume(ctx):
    voice = discord.utils.get(client.voice_clients, guild = ctx.guild)
    if voice.is_paused():
        await ctx.send("My ears are bleeding")
        voice.resume()
    else:
        await ctx.send("ALREADY BLASTING MUSIC")

@client.command(pass_Context = True)
async def stop(ctx):
    voice = discord.utils.get(client.voice_clients, guild = ctx.guild)
    await ctx.send("You can stop the song, but you can't stop me!")
    voice.stop()

@client.command(pass_Context = True)
async def play(ctx, arg):
    await ctx.send("Playing..")
    voice = ctx.guild.voice_client
    source = FFmpegPCMAudio(arg)
    player = voice.play(source)

if '__main__' == __name__:
    client.run(Token)


  


  I tried installing different versions of ffmpeg, still nothing. I tried to run the code outside of my venv, but still nothing (i doubt that it's the problem). I changed the path to it to different folders, still nothing.
The only time it DID work is when i entered a full path, but then when you want to play something, you wouldn't want to say " !play D:_Python\DiscordBot\Bluetooth.wav". From what i've seen, it's possible to play it by just saying " !play Bluetooth.wav".

  


  So long story short : I want to make it so that the path i have to specify is just the file name. And when I do, it doesn't play the sound.
(sorry if this is a dupe question, i just couldn't find anything understandable for my amateur brain)

  


• Dealing with problems in FLAC audio files with ffmpeg

  15 janvier 2020, par Seamus

  I have gotten a set of FLAC (audio) files from a friend. I copied them to my Sonos music library, and got set to enjoy a nice album. Unfortunately, Sonos would not play the files. As a result I have been getting to know ffmpeg.

  Sonos’ complaint with the FLAC files was that it was "encoded at an unsupported sample rate". With rolling eyes and shaking head, I note that the free VLC media player happily plays these files, but the product I’ve paid for (Sonos) - does not. But I digress...

  ffprobe revealed that the FLAC files contain both an Audio channel and a Video channel :

  $ ffprobe -hide_banner  -show_streams "/path/to/Myaudio.flac"
  
  
  
  
  
      Duration: 00:02:23.17, start: 0.000000, bitrate: 6176 kb/s  
  
      Stream #0:0: Audio: flac, 176400 Hz, stereo, s32 (24 bit)  
  
      Stream #0:1: Video: mjpeg (Progressive), yuvj444p(pc, bt470bg/unknown/unknown), 450x446 [SAR 72:72 DAR 225:223], 90k tbr, 90k tbn, 90k tbc (attached pic)  
  
      Metadata:  
  
        comment         : Cover (front)  

  Cool ! I guess this is how some audio players are able to display the ’album artwork’ when they play a song ? Note also that the Audio stream is reported at 176400 Hz ! Apparently I’m out of touch ; I thought that 44.1khz sampling rate effectively removed all of the ’sampling artifacts’ we could hear. Anyway, I learned that Sonos would support a max of 48kHz sampling rate, and this (the 176.4kHz rate) is what Sonos was unhappy about. I used ffmpeg to ’dumb it down’ for them :

  $ ffmpeg -i "/path/to/Myaudio.flac" -sample_fmt s32 -ar 48000 "/path/to/Myaudio48K.flac"

  This seemed to work - at least I got a FLAC file that Sonos would play. However, I also got what looks like a warning of some sort :

  [swscaler @ 0x108e0d000] deprecated pixel format used, make sure you did set range correctly
  [flac @ 0x7feefd812a00] Frame rate very high for a muxer not efficiently supporting it.
  Please consider specifying a lower framerate, a different muxer or -vsync 2

  A bit more research turned up this answer which I don’t quite understand, and then in a comment says, "not to worry" - at least wrt the swscaler part of the warning.

  And that (finally) brings me to my questions :

  1.a. What framerate, muxer & other specifications make a graphic compatible with a majority of programs that use the graphic ?

  1.b. How should I use ffmpeg to modify the Video channel to set these specifications (ref. Q 1.a.) ?

  2.a. How do I remove the Video channel from the .flac audio file ?

  2.b. How do I add a Video channel into a .flac file ?

  EDIT :

  I asked the above (4) questions after failing to accomplish a ’direct’ conversion (a single ffmpeg command) from FLAC at 176.4 kHz to ALAC (.m4a) at 48 kHz (max supported by Sonos). I reasoned that an ’incremental’ approach through a series of conversions might get me there. With the advantage of hindsight, I now see I should have posted my original failed direct conversion incantation... we live and learn.

  That said, the accepted answer below meets my final objective to convert a FLAC file encoded at 176.4kHz to an ALAC (.m4a) at 48kHz, and preserve the cover art/video channel.