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  • How can i merge 2 videos with audio using ffmpeg ?

    4 juin 2020, par Easy Tricks For Android

    I tried this command
ffmpeg -f concat -i mylist.txt -c copy output.mp4
but output.mp4 has no audio. please help me.

    



    the mylist.txt contains these words : file '1.mp4'
file '2.mp4'

    
this is the log

    



    


    C :\Users\Admin\OneDrive\Desktop\New folder>ffmpeg -n -i 1.mp4 -i 2.mp4
 ffmpeg version git-2020-05-23-26b4509 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. 48.100 / 56. 48.100
 libavcodec 58. 87.101 / 58. 87.101
 libavformat 58. 43.100 / 58. 43.100
 libavdevice 58. 9.103 / 58. 9.103
 libavfilter 7. 83.100 / 7. 83.100
 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 '1.mp4' :
 Metadata :
 major_brand : mp42
 minor_version : 0
 compatible_brands : mp41isom
 creation_time : 2020-05-31T08:12:56.000000Z
 Duration : 00:00:05.93, start : 0.000000, bitrate : 186 kb/s
 Stream #0:0(und) : Video : h264 (Main) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 184 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc (default)
 Metadata :
 creation_time : 2020-05-31T08:21:56.000000Z
 handler_name : VideoHandler
 encoder : AVC Coding
 Input #1, mov,mp4,m4a,3gp,3g2,mj2, from '2.mp4' :
 Metadata :
 major_brand : mp42
 minor_version : 0
 compatible_brands : mp41isom
 creation_time : 2020-06-01T03:32:58.000000Z
 Duration : 00:00:12.23, start : 0.000000, bitrate : 1909 kb/s
 Stream #1:0(und) : Video : h264 (Main) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 1708 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc (default)
 Metadata :
 creation_time : 2020-06-01T04:17:17.000000Z
 handler_name : VideoHandler
 encoder : AVC Coding
 Stream #1:1(und) : Audio : aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 196 kb/s (default)
 Metadata :
 creation_time : 2020-06-01T04:17:17.000000Z
 handler_name : SoundHandler
 At least one output file must be specified

    


    


  • I am not able to create an RTMP stream using ffmpeg

    11 juin 2020, par Navneet Raju

    I have set up an rtmp server using Nginx and that seems to be working fine. But when I try to create a RTMP stream using FFmpeg, it gets stuck at a frame and streaming doesn't work(just keeps buffering).

    



    The FFmpeg command run :

    



    ffmpeg -re -i CD_Final_compress.mp4 -vcodec copy -loop -1 -c:a aac -b:a 160k -ar 44100 -strict -2 -f flv rtmp://127.0.0.1/live/bbb


    



    Output :

    



    ffmpeg version N-98112-g7aa7d68 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9 (Ubuntu 9.3.0-10ubuntu2)
  configuration: --prefix=/home/navneet/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/navneet/ffmpeg_build/include --extra-ldflags=-L/home/navneet/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/navneet/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree
  libavutil      56. 53.100 / 56. 53.100
  libavcodec     58. 92.100 / 58. 92.100
  libavformat    58. 46.101 / 58. 46.101
  libavdevice    58. 11.100 / 58. 11.100
  libavfilter     7. 86.100 /  7. 86.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'CD_Final_compress.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf58.33.100
  Duration: 00:17:33.85, start: 0.000000, bitrate: 521 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 386 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 126 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (aac (native) -> aac (native))
Press [q] to stop, [?] for help
Output #0, flv, to 'rtmp://127.0.0.1/live/bbb':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf58.46.101
    Stream #0:0(und): Video: h264 (High) ([7][0][0][0] / 0x0007), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 386 kb/s, 30 fps, 30 tbr, 1k tbn, 15360 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) ([10][0][0][0] / 0x000A), 44100 Hz, stereo, fltp, 160 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
      encoder         : Lavc58.92.100 aac
frame= 1652 fps= 30 q=-1.0 size=    4895kB time=00:00:54.96 bitrate= 729.5kbits/s speed=0.999x


    



    Note : I am running this on WSL 1 Ubuntu on Windows

    


  • 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


    


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