I am attempting to create a online radio application using FFMPEG - an audio only DASH stream.
I have a directory of mp3 clips (all of the same bitrate and sample size) which I am encoding to the AAC format and outputting to a mpd.
This is the current command I am working with to stream a single mp3 file:
ffmpeg -re -i <input>.mp3 -c:a aac -use_timeline 1 -use_template 1 -window_size 5 -f dash <out>.mpd
(Input and output paths have been substituted for < input >.mp3 and < output >.mpd in this snippet)
I am running a web server and have made the mpd accessible on it. I am testing the stream using VLC player at the moment.
The problem:
Well, the command works, but it will only work for one clip at a time. Once the next command is run immediately proceeding the completion of the first, VLC player will halt and I need to refresh the player to continue.
I'm aiming for an uninterrupted stream wherein the clips play in sequence.
I imagine the problem is that a new mpd is being created with no reference to the previous one, and what I ought to be doing is appending segments to the existing mpd - but I don't know how to do that using FFMPEG.
The question: Is there such a command to append segments to a previously existing mpd file in FFMPEG? or am I coming at this problem all wrong? Perhaps I should be using FFMPEG to format the clips into these segments, but then adjusting the mpd file manually.
Any help or suggestions would be very much appreciated!
Related
My problem basically comes from me having 2 different streams for videoplayback and having to mux them realtime in memory. One for video, and another for audio.
My goal is to create a proxy which can mux 2 different webm streams from their URLs, while supporting range requests (requires knowing the encoded file size). Would this be possible?
This is how I mux the audio and video streams manually using ffmpeg:
ffmpeg -i video.webm -i audio.webm -c copy output.webm
But, this requires me to download the video fully to process it, which I don't want to do unfortunately.
Thanks in advance!
If you are looking for this to work in go you can look into
github.com/at-wat/ebml-go/webm
This provides a BlockWriter interface to write to webm file using buffers; You can see the test file to checkout how to use it
https://github.com/at-wat/ebml-go
Checkout ffmpeg pipes.
Also since you have tagged go - i'm assuming you will use os/exec - in which case also checkout Cmd.ExtraFiles. This lets you use additional pipes(files) beyond just the standard 0, 1 and 2.
So let's say you have a stream for video and one for audio piping to 3 and 4 respectively. The ffmpeg bit of your command becomes:
ffmpeg -i pipe:3 -i pipe:4 -c copy output.webm
I use ffmpeg to record rtsp stream, it work good but the output file got some proble, when I use use K-Lite Codec Pack to open the output (avi) file the video cant be seek, forward, backward and dont display video time. It lock like i am viewing streaming.
here is the command i used
ffmpeg -i rtsp://27.74.xxx.xxx:55/ufirststream -acodec copy -vcodec copy abc.avi
video playing error with K-Lite Codec Park image
Looks like header file not been updated on output file. It's recommended to close output file by pressing "q" button while ffmpeg reads input stream to properly finalize output file.
I transcoding via FFmpeg (video codec - h264, container - MPEG-TS) output writing to local file (out.mpg). When FFmpeg dropped I restarted it with output to the same file (out.mpg). After this my video player shows incorrect file duration due to new FFmpeg process start counting PCR from 0.
Can I set starting pcr value at start FFmpeg ?
Run FFmpeg with key -copyts solved this issue.
How to transfer metadata using FFMPEG or other tools with CMD ?
I'm trying to encode video/audio and since they already have metadata inside obviously i want to preserve them into my new file
btw since i'm using mediamonkey as main player, there's also some Custom metadata. this is the one who wont transfer
for Video output file using mp4/mkv (using x264)
for Audio output file using m4a (using neroAac)
Thank You!
ps. which container is best for neroAac and x264? since i can't seem to edit mkv metadata (when i remove from mediamonkey playlist, they're all gone), mp4 is fine though and i can't seem to play AAC, although it's fine when muxed into video
Copy all custom and global metadata tag information using the following command:
ffmpeg <inputfile> -movflags use_metadata_tags -c copy <outputfile>
Recently I had a task to use ffmpeg as a transcoding as well a streaming tool. The task was to convert the file from a given format to MP4 and immediately stream it, by capturing it from stdout. So far so good. The streaming works well with the native player of android tabs as well as the VLC player. The issue is with the flash player. It gives the following error:
NetStream.Play.FileStructureInvalid : Adobe Flash cannot import files that have invalid file structures.
ffmpeg flags used are
$ ffmpeg -loglevel quiet -i somefile.avi -vbsf h264_mp4toannexb -vcodec libx264 \
-acodec aac -f MP4 -movflags frag_keyframe+empty_moov -re - 2>&1
As noted in the docs for -movflags
The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4 file has all the metadata about all packets stored in one location (written at the end of the file, it can be moved to the start for better playback using the qt-faststart tool). A fragmented file consists of a number of fragments, where packets and metadata about these packets are stored together. Writing a fragmented file has the advantage that the file is decodable even if the writing is interrupted (while a normal MOV/MP4 is undecodable if it is not properly finished), and it requires less memory when writing very long files (since writing normal MOV/MP4 files stores info about every single packet in memory until the file is closed). The downside is that it is less compatible with other applications.
Either switch to a flash player that can handle fragmented MP4 files, or use a different container format that supports streaming better.
Also, -re is an input-only option, so it would make more sense to specify it before the input, instead of before the output.