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• FFmpeg ; combining two multi-channel streams into one

  30 mars 2023, par Christiaan Roselaar

  Hi to all you FFmpeg Guru's.

  


  I'm doing my best to combine two incoming webstreams on my server. One of them containing two (2) mono audio-channels, the other five (5). The objective is to create one stream containing seven (7) channels, to be routed directly to the hardware.

  


  At the two transmission-origins, host_a and host_b, the streams are generated as follows :

  


  host_a : $ ffmpeg -hide_banner -loglevel error -f jack -channels 1 -i Tonmeister_mic1 -f jack -channels 1 -i Tonmeister_mic2 -nostdin -filter_complex "[0:a][1:a]amerge=inputs=2[a]" -map "[a]" -codec:a libopus -b:a 128k -vbr on -packet_loss 10 -fec on -compression_level 10 -frame_duration 20 -application audio -f mpegts udp ://X.X.X.X:yyyy &

  


  host_b : $ ffmpeg -hide_banner -loglevel error -f jack -channels 1 -i Tonmeister_mic1 -f jack -channels 1 -i Tonmeister_mic2 -f jack -channels 1 -i Tonmeister_mic3 -f jack -channels 1 -i Tonmeister_mic4 -f jack -channels 1 -i Tonmeister_mic5 -nostdin -filter_complex "[0:a][1:a][2:a][3:a][4:a]amerge=inputs=5[a]" -map "[a]" -codec:a libopus -b:a 128k -vbr on -packet_loss 10 -fec on -compression_level 10 -frame_duration 20 -application audio -f mpegts udp ://X.X.X.X:zzzz &

  


  The streams do seem to get in allright :

  


  cjr-macbook : CJR$ ffprobe -hide_banner -i udp ://127.0.0.1:yyyy
Input #0, mpegts, from 'udp ://127.0.0.1:yyyy' :
Duration : N/A, start : 4759.783900, bitrate : N/A
Program 1
Metadata :
service_name : Service01
service_provider : FFmpeg
Stream #0:0[0x100] : Audio : opus (Opus / 0x7375704F), 48000 Hz, stereo, fltp

  


  cjr-macbook : CJR$ ffprobe -hide_banner -i udp ://127.0.0.1:zzzz
Input #0, mpegts, from 'udp ://127.0.0.1:zzzz' :
Duration : N/A, start : 4856.559000, bitrate : N/A
Program 1
Metadata :
service_name : Service01
service_provider : FFmpeg
Stream #0:0[0x100] : Audio : opus (Opus / 0x7375704F), 48000 Hz, 5.0, fltp

  


  However, I just cannot figure out how to combine the seven (7) channels present in these two incoming streams into one seven-channel stream.

  


  At the receiving end I tried various commands, amongst others

  


  cjr-macbook : CJR$ ffmpeg -hide_banner -loglevel verbose -i udp ://127.0.0.1:yyyy -i udp ://127.0.0.1:zzzz -filter_complex "[0:a]channelsplit=channel_layout=stereo[b]" ;"[1:a]channelsplit=channel_layout=5.0[c]" ;"[b][c]amerge=inputs=7[d]" -map "[d]" sevenchannels.wav

  


  Any help appreciated !

  


• How to add or generate a missing keyframe at the beginning of converted video in FFMPEG ?

  21 décembre 2022, par Jindrich Vavruska

  There is a MP4 file representing a downloaded stream transmission from which I want to extract a program that starts after the beginning and ends before end of this MP4 file. I cut the unwanted parts off by two simple ffmpeg 'conversions'. In the first step, I cut off the tail :

  


  ffmpeg -i stream_record.mp4 -c copy -t 1:26:33.15 cut_record.mp4


  


  The second step is to cut off the unwanted beginning, but here I run into problems. E.g. after

  


  ffmpeg -i cut_record.mp4 -c copy -ss 02:36 final_record.mp4


  


  I quite often get a file with no video in the beginning (up to 6 seconds), only audio is playing. This is because after skipping the time specified in -ss option there is no keyframe in the video stream and it takes up to those 6 seconds to find the first one. Unfortunately, my desired part of the downloaded stream quite seldom starts with a keyframe, and I must keep some part of the video preceding my program to maintain video integrity in the beginning of the file.

  


  Is there a method in FFMPEG to add something (e.g. black rectangle) as the first keyframe right at the start of the copied video stream ? I mean, other than completely recoding the video stream (I would like to keep the -c copy parameter in the process) ?

  


  I could probably achieve the desired result using Openshot Editor - but it recodes the video and audio, takes relatively long time and consumes much CPU and memory resources.

  


• Does FFmpeg not support G723_1 transmit at 5.3k bitrate？

  8 octobre 2022, par Zih

  Recently I needed to implement the 5.3k bitrate transmission of the G723.1 protocol on an VoIP phone, but I found that in the g723_1enc.c language file under the ffmpeg's libavcodec folder, there is a description of the picture below, the image is from Github.enter image description here

  


  This code is in the "g723_1_encode_init AVCodecContext *avctx", does this mean that ffmpeg does not support the G723.1 protocol to transmit at 5.3 bps ? What kind of changes might I be able to make it support ? Because I want it to be a choice between 6.3k and 5.3k. Is this achievable ? I would appreciate any advice.

  


  p.s.I'm guessing that a drastic revision of the code will be needed, I just graduated from college and have recently entered the field, please give me some advice