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• FFmpeg filter complex add audio file to video at specific points

  4 juin 2020, par steve

  atrim=0:2 starts fine when I play the video it plays the mp3 now how do I add a working end time say finish at 9 seconds ?
  

  



  Summary I want to start at 2 seconds and finish at 9 seconds.
  

  



  ffmpeg -y -i "C:\Users\test\Desktop\vidz\New folder (2)\target\vaastav song .mp4" -i "C:\Users\test\Desktop\vidz\New folder (2)\target\2.mp3" -filter_complex "[0]atrim=0:2[Apre];[0]atrim=5,asetpts=PTS-STARTPTS[Apost];[Apre][1][Apost]concat=n=3:v=0:a=1" -vcodec copy -y "C:\Users\test\Desktop\vidz\New folder (2)\target\output1.mp4"


  



  log file

  



  ffmpeg version git-2020-04-13-59e3a9a 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. 42.102 / 56. 42.102
  libavcodec     58. 78.102 / 58. 78.102
  libavformat    58. 42.100 / 58. 42.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 77.101 /  7. 77.101
  libswscale      5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\Users\test\Desktop\vidz\New folder (2)\target\vaastav song .mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: isommp42
    creation_time   : 2018-09-02T04:28:46.000000Z
  Duration: 00:05:08.80, start: 0.000000, bitrate: 2289 kb/s
    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 2160 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc (default)
    Metadata:
      creation_time   : 2018-09-02T04:28:46.000000Z
      handler_name    : ISO Media file produced by Google Inc. Created on: 09/01/2018.
    Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 125 kb/s (default)
    Metadata:
      creation_time   : 2018-09-02T04:28:46.000000Z
      handler_name    : ISO Media file produced by Google Inc. Created on: 09/01/2018.
Input #1, mp3, from 'C:\Users\test\Desktop\vidz\New folder (2)\target\2.mp3':
  Metadata:
    genre           : Electronic;Indie
    title           : A Distorted Noise With A Little Bit Of Sense
    artist          : Lenin Was A Zombie
    encoder         : Lavf56.19.100
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: isommp42
  Duration: 00:00:54.00, start: 0.025057, bitrate: 192 kb/s
    Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 192 kb/s
    Metadata:
      encoder         : Lavc56.21
Stream mapping:
  Stream #0:1 (aac) -> atrim
  Stream #0:1 (aac) -> atrim
  Stream #1:0 (mp3float) -> concat:in1:a0
  concat -> Stream #0:0 (aac)
  Stream #0:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
Output #0, mp4, to 'C:\Users\test\Desktop\vidz\New folder (2)\target\output1.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: isommp42
    encoder         : Lavf58.42.100
    Stream #0:0: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
    Metadata:
      encoder         : Lavc58.78.102 aac
    Stream #0:1(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 2160 kb/s, 25 fps, 25 tbr, 90k tbn, 90k tbc (default)
    Metadata:
      creation_time   : 2018-09-02T04:28:46.000000Z
      handler_name    : ISO Media file produced by Google Inc. Created on: 09/01/2018.
frame=  737 fps=0.0 q=-1.0 size=    7936kB time=00:00:29.44 bitrate=2208.1kbits/s speed=58.9x    
[out_0_0 @ 000000000312c900] 100 buffers queued in out_0_0, something may be wrong.
[out_0_0 @ 000000000312c900] 1000 buffers queued in out_0_0, something may be wrong.
frame= 1401 fps=876 q=-1.0 size=   16640kB time=00:01:46.85 bitrate=1275.7kbits/s speed=66.8x    
frame= 2281 fps=1086 q=-1.0 size=   26112kB time=00:02:22.01 bitrate=1506.3kbits/s speed=67.6x    
frame= 3152 fps=1212 q=-1.0 size=   35584kB time=00:02:56.58 bitrate=1650.8kbits/s speed=67.9x    
frame= 4011 fps=1294 q=-1.0 size=   45824kB time=00:03:31.32 bitrate=1776.4kbits/s speed=68.2x    
frame= 4882 fps=1356 q=-1.0 size=   56320kB time=00:04:06.03 bitrate=1875.2kbits/s speed=68.3x    
frame= 5753 fps=1403 q=-1.0 size=   66560kB time=00:04:40.79 bitrate=1941.8kbits/s speed=68.5x    
frame= 6622 fps=1440 q=-1.0 size=   77056kB time=00:05:15.37 bitrate=2001.6kbits/s speed=68.6x    
frame= 7482 fps=1467 q=-1.0 size=   85248kB time=00:05:50.20 bitrate=1994.1kbits/s speed=68.7x    
frame= 7720 fps=1479 q=-1.0 Lsize=   87248kB time=00:05:59.79 bitrate=1986.5kbits/s speed=68.9x    
video:81451kB audio:5634kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.186449%
[aac @ 0000000003017ac0] Qavg: 648.065


  



  I tried messing about with the line of code but no success so I turn to you for additional support

  


• FFMPEG Streaming to twitch low bitrate

  17 juillet 2020, par El_Presidente

  I have a python script that will produce frames for a video stream. To stream it to twitch I decided to use ffmpeg (as it is the only option I found). However, the bitrate of my stream is very low (70 KB), although in ffmpeg options it's set to 3000K.

  


  # This script copies the video frame by frame
import cv2
import subprocess as sp
twitch_stream_key = 'MY_TWITCH_STREAM_KEY'
input_file = 'video.mp4'

cap = cv2.VideoCapture(input_file)
ret, frame = cap.read()
height, width, ch = frame.shape

ffmpeg = 'FFMPEG'
dimension = '{}x{}'.format(width, height)

fps = cap.get(cv2.CAP_PROP_FPS)
command = []
command.extend([
    'FFMPEG',
    '-loglevel', 'verbose',
    '-y',  # overwrite previous file/stream
    '-analyzeduration', '1',
    '-f', 'rawvideo',
    '-r', '%d' % fps,  # set a fixed frame rate
    '-vcodec', 'rawvideo',
    # size of one frame
    '-s', '%dx%d' % (width, height),
    '-pix_fmt', 'rgb24',  # The input are raw bytes
    '-thread_queue_size', '1024',
    '-i', '-',  # The input comes from a pipe
])       
command.extend([
    '-ar', '8000',
    '-ac', '1',
    '-f', 's16le',
    '-i', 'work.mp3',
])
command.extend([
    # VIDEO CODEC PARAMETERS
    '-vcodec', 'libx264',
    '-r', '%d' % fps,
    '-b:v', '3000k',
    '-s', '%dx%d' % (width, height),
    '-preset', 'faster', '-tune', 'zerolatency',
    '-crf', '23',
    '-pix_fmt', 'yuv420p',

    '-minrate', '3000k', '-maxrate', '3000k',
    '-bufsize', '12000k',
    '-g', '60',  # key frame distance
    '-keyint_min', '1',

    # AUDIO CODEC PARAMETERS
    '-acodec', 'libmp3lame', '-ar', '44100', '-b:a', '160k',
    # '-bufsize', '8192k',
    '-ac', '1',
    '-map', '0:v', '-map', '1:a',

    '-threads', '2',
    # STREAM TO TWITCH
    '-f', 'flv', 'rtmp://live-hel.twitch.tv/app/%s' %
          twitch_stream_key
])
proc = sp.Popen(command, stdin=sp.PIPE, stderr=sp.PIPE)

while True:
    ret, frame = cap.read()        
    if not ret:        
        break    
    proc.stdin.write(frame.tostring())

cap.release()
proc.stdin.close()
proc.stderr.close()
proc.wait()


  


  How can I increase the bitrate ? Maybe you can point me towards some different solution on how I can stream python frames to twitch or any other rtmp server.

  


  Here is the complete log, the audio is also broken, it's just noise :

  


  ffmpeg version git-2020-06-01-dd76226 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200523
  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. 49.100 / 56. 49.100
  libavcodec     58. 90.100 / 58. 90.100
  libavformat    58. 44.100 / 58. 44.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 84.100 /  7. 84.100
  libswscale      5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
Input #0, rawvideo, from 'pipe:':
  Duration: N/A, start: 0.000000, bitrate: 1443225 kb/s
    Stream #0:0: Video: rawvideo, 1 reference frame (RGB[24] / 0x18424752), rgb24, 1920x1080, 1443225 kb/s, 29 tbr, 29 tbn, 29 tbc
[s16le @ 0000026d64eb5340] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for Input Stream #1.0 : mono
Input #1, s16le, from 'work.mp3':
  Metadata:
    encoded_by      : iTunes v7.0
  Duration: 00:09:36.13, bitrate: 128 kb/s
    Stream #1:0: Audio: pcm_s16le, 8000 Hz, mono, s16, 128 kb/s
[tcp @ 0000026d64ee34c0] Starting connection attempt to 99.181.64.78 port 1935
[tcp @ 0000026d64ee34c0] Successfully connected to 99.181.64.78 port 1935
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
  Stream #1:0 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
[graph 0 input from stream 0:0 @ 0000026d64f47c00] w:1920 h:1080 pixfmt:rgb24 tb:1/29 fr:29/1 sar:0/1
[scaler_out_0_0 @ 0000026d64f4c780] w:1920 h:1080 flags:'bicubic' interl:0
[scaler_out_0_0 @ 0000026d64f4c780] w:1920 h:1080 fmt:rgb24 sar:0/1 -> w:1920 h:1080 fmt:yuv420p sar:0/1 flags:0x4
[libx264 @ 0000026d64edf840] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0000026d64edf840] profile High, level 4.0, 4:2:0, 8-bit
[libx264 @ 0000026d64edf840] 264 - core 160 - H.264/MPEG-4 AVC codec - Copyleft 2003-2020 - http://www.videolan.org/x264.html - options: cabac=1 ref=2 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=4 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=2 lookahead_threads=2 sliced_threads=1 slices=2 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=1 keyint=60 keyint_min=1 scenecut=40 intra_refresh=0 rc_lookahead=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=3000 vbv_bufsize=12000 crf_max=0.0 nal_hrd=none filler=0 ip_ratio=1.40 aq=1:1.00
[graph_1_in_1_0 @ 0000026d651319c0] tb:1/8000 samplefmt:s16 samplerate:8000 chlayout:0x4
[format_out_0_1 @ 0000026d65132d80] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_1'
[auto_resampler_0 @ 0000026d651331c0] ch:1 chl:mono fmt:s16 r:8000Hz -> ch:1 chl:mono fmt:s16p r:44100Hz
Output #0, flv, to 'rtmp://live-hel.twitch.tv/app/live_*************':
  Metadata:
    encoder         : Lavf58.44.100
    Stream #0:0: Video: h264 (libx264), 1 reference frame ([7][0][0][0] / 0x0007), yuv420p(progressive), 1920x1080, q=-1--1, 3000 kb/s, 29 fps, 1k tbn, 29 tbc
    Metadata:
      encoder         : Lavc58.90.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 3000000/0/3000000 buffer size: 12000000 vbv_delay: N/A
    Stream #0:1: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, mono, s16p, delay 1105, 160 kb/s
    Metadata:
      encoder         : Lavc58.90.100 libmp3lame


  


• youtube stream from ffmpeg is buffering

  1er juin 2020, par Bartonsen

  I'm using ffmpeg running on a Raspberry Pi 3B with 1GB RAM to stream live video on youtube.
In the beginning the audio+video stream is excellent, but after some minutes I see error messages in YT studio, and video starts buffering.
After some more time (could be 30 mins or 1 hr) the youtube stream is gone, although ffmpeg is still running.

  



  ffmpeg is configured like this :

  



  ./configure --arch=armel --target-os=linux --enable-gpl --enable-libx264 --enable-nonfree --enable-omx-rpi


  



  Running ffmpeg :

  



  pi@raspberrypi:~ $ ffmpeg -thread_queue_size 512 -rtsp_transport udp -i "rtsp://10.x.x.x:554/user=user&password=password&channel=1&stream=0.sdp?real_stream" -c:v copy -c:a aac -f flv rtmp://a.rtmp.youtube.com/live2/mykey
ffmpeg version git-2020-05-01-3c740f2 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 6.3.0 (Raspbian 6.3.0-18+rpi1+deb9u1) 20170516
  configuration: --arch=armel --target-os=linux --enable-gpl --enable-libx264 --enable-nonfree --enable-omx-rpi
  libavutil      56. 43.100 / 56. 43.100
  libavcodec     58. 82.100 / 58. 82.100
  libavformat    58. 42.102 / 58. 42.102
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 80.100 /  7. 80.100
  libswscale      5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
[udp @ 0x3a8c370] attempted to set receive buffer to size 393216 but it only ended up set as 327680
[udp @ 0x3a9ea80] attempted to set receive buffer to size 393216 but it only ended up set as 327680
[udp @ 0x3a8c3e0] attempted to set receive buffer to size 393216 but it only ended up set as 327680
[udp @ 0x3abf4d0] attempted to set receive buffer to size 393216 but it only ended up set as 327680
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://10.x.x.x:554/user=user&password=password&channel=1&stream=0.sdp?real_stream':
  Metadata:
    title           : RTSP Session
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709, progressive), 1920x1080, 20 fps, 20 tbr, 90k tbn, 180k tbc
    Stream #0:1: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (pcm_alaw (native) -> aac (native))
Press [q] to stop, [?] for help
[aac @ 0x3ae9f00] Too many bits 8832.000000 > 6144 per frame requested, clamping to max
Output #0, flv, to 'rtmp://a.rtmp.youtube.com/live2/mykey':
  Metadata:
    title           : RTSP Session
    encoder         : Lavf58.42.102
    Stream #0:0: Video: h264 (Main) ([7][0][0][0] / 0x0007), yuvj420p(pc, bt709, progressive), 1920x1080, q=2-31, 20 fps, 20 tbr, 1k tbn, 90k tbc
    Stream #0:1: Audio: aac (LC) ([10][0][0][0] / 0x000A), 8000 Hz, mono, fltp, 48 kb/s
    Metadata:
      encoder         : Lavc58.82.100 aac


  



  In youtube studio I see this :

  



  19:36 Good transmission. The quality is excellent.
19:39 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:39 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:39 Warning: The current bit rate (1974.24 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.
19:41 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:41 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:41 Warning: The current bit rate (2151.41 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.
19:43 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:43 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:45 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:45 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:45 Warning: The current bit rate (1737.61 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.
...
19:54: Error: YouTube does not receive enough video to maintain consistent streaming. Viewers will therefore experience buffering.


  



  What is the problem ? How can I get rid of the buffering ?
I've also tried the below two commands, but found the output to be worse...

  



  ffmpeg -i rtsp://... -c:v libx264  -b:v 4000k -maxrate 4000k -bufsize 8000k -g 40 -preset ultrafast -vf format=yuv420p -c:a aac -f flv output
ffmpeg -i rtsp://... -c:v h264_omx -b:v 4000k -maxrate 4000k -bufsize 8000k -g 40 -preset ultrafast -vf format=yuv420p -c:a aac -f flv output