Recherche avancée

Sur d’autres sites (6289)

• How to add a fake microphone to android emulator running on linux headless

  2 juin, par Red

  Trying to add a microphone to the android emulator running on Linux headless. Host as no microphone so I need to create fake one and simulate playing some random music for my android test.

  


  In emulator command there is a option to use the host audio as input, by passing
-allow-host-audio but it is not working on the phone.

  


  How to create virtual microphone
Start the pusleaudio

  


  pulseaudio -D --exit-idel-time=-1


  


  Create the fake mic

  


  pactl load-module module-null-sink sink_name=FakeSink


  


  pactl load-module module-remap-source master=FakeSink.monitor source_name=FakeMic

  


  And set to default

  


  pactl sets default FakeSink.monitor


  


  Microphone test

  


  $ffmpeg -f pulse -i default out.wav


$ sox out.wav -n stat

Samples read:            579654

Length (seconds):      6.038062

Scaled by:         2147483647.0

Maximum amplitude:     0.533081

Minimum amplitude:    -0.585297

Midline amplitude:    -0.026108

Mean    norm:          0.067096

Mean    amplitude:     0.003363

RMS     amplitude:     0.093545

Maximum delta:         0.603760

Minimum delta:         0.000000

Mean    delta:         0.073738

RMS     delta:         0.105326

Rough   frequency:         8601

Volume adjustment:        1.709


  


  Run the emulator

  


  emulator -avd and_1 -allow-host-audio -no_window


  


  There was no audio on the phone.

  


• ffmpeg live streaming for twitch has audio delay

  8 avril 2019, par ILoveCake

  I am relatively new to ffmpeg and live streaming.
  Trying to overlay some videos and an audio stream.
  One of the audio streams is delayed for about 2 seconds.

  Here are the details :

  stream 0 : video : a static background image

  stream 1 : video : an area of the screen recorded

  stream 2 : video : webcam video

  stream 3 : audio : webcam audio

  AREA_X=1024; AREA_Y=576; OFFSET_X=110; OFFSET_Y=145
  
  
  
  ffmpeg \
  
    -async 1 -vsync 1 \ 
  
    -loop 1 -i /home/helmi/Documents/Streaming.Chess.png \
  
    -thread_queue_size 512 -f x11grab -s ${AREA_X}x${AREA_Y} -framerate 25 -async 1 -vsync 1 -i :0.0+${OFFSET_X},${OFFSET_Y} \
  
    -thread_queue_size 512 -f v4l2 -framerate 25 -video_size 160x120 -i /dev/video0 \
  
    -thread_queue_size 512 -f pulse -ac 2 -ar 48000 -i default \
  
    -filter_complex \
  
    "color=0x336699cc:1024x64, drawtext=textfile=/home/helmi/Documents/Streaming.Chess.txt:fontfile=/home/helmi/.fonts/PersonalUse_Clipper_Script_fat.ttf:x=10:y=16:fontsize=40:fontcolor=white [bottom]; \
  
     [1:v]scale=960:-1,setpts=PTS-STARTPTS [a]; \
  
     [0:v]setpts=PTS-STARTPTS [0v]; \
  
     [0v][a]overlay=15:15 [b]; \
  
     [b][2:v]overlay=(W-w):0 [c]; \
  
     [c][bottom]overlay=0:H-64 [video]" \
  
     -map "[video]" -map "3:a" \
  
      -async 1 -vsync 1 \ 
  
      -c:v libx264 -b:v 500k -maxrate 500k -bufsize 1000k -framerate 25 -crf 17 -preset superfast -pix_fmt yuv420p -tune zerolatency \
  
        -force_key_frames "expr:gte(t,n_forced*2)" \
  
      -c:a aac -b:a 256k -ac 2 -af "aresample=async=1" \
  
    -f flv rtmp://live-vie.twitch.tv/app/...

  Am I missing something here ? Any help is appreciated.

• ffmpeg recording audio / video sync issues [closed]

  1er novembre 2022, par sling jones

  I'm using ffmpeg to record NTSC analog video on Linux Fedora 36 using a Blackmagic Intensity Pro 4K for video and a Scarlett 2i2 for audio. I'm using a TBC to avoid dropped frames and to ensure a constant S-Video Y/C framerate on the analog end.

  


  The problem I'm running into is that on playback the audio will start out relatively in sync with the video at the beginning of the captured file but will eventually run ahead of the video eventually becoming many seconds off.

  


  Nothing I do seems to change this or change the degree to which it happens. The audio and video stay in sync throughout the entire video as I'm monitoring the source so I don't understand how they can diverge so much once encoded into a digital file ?

  


  Here is the command I am using :

  


  ffmpeg -hwaccel cuda -fflags +igndts -format_code ntsc -f decklink -raw_format auto -vsync passthrough -rtbufsize 1500M -thread_queue_size 512 -i 'Intensity Pro 4K' -f pulse -rtbufsize 500M -thread_queue_size 512 -i 'Scarlett 2i2 Camera Analog Stereo' -c copy -map 0:1 -map 1:0 "/tmp/ffmpeg-raw/file-raw.avi"


  


  here's the ffprobe output from one of my files :

  


  Input #0, avi, from 'test2-raw.avi':
  Metadata:
    software        : Lavf59.33.100
  Duration: 00:11:42.87, start: 0.000000, bitrate: 169341 kb/s
  Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x486, 167801 kb/s, 59.94 fps, 59.94 tbr, 59.94 tbn
  Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s



  


  As you can see in my code snippet, I've been throwing things at the wall for a bit. I've tried different rtbufsizes, adding -copyts, and going through the different -vsync options. I've tried it with and without hardware acceleration ( I do have a NVIDIA card), +igdts did get rid of a warning but did not help with the sync, as did changing the thread queue sizes.

  


  OBS can do this, why can't I ?