I have a single ts file and created a single-file m3u8 using ffmpeg. It looks like the following
#EXTM3U
#EXT-X-VERSION:4
#EXT-X-TARGETDURATION:1
#EXT-X-MEDIA-SEQUENCE:0
#EXTINF:0.000000,
#EXT-X-BYTERANGE:22184#0
video.ts
#EXTINF:1.000667,
#EXT-X-BYTERANGE:713836#22184
video.ts
#EXTINF:1.000667,
#EXT-X-BYTERANGE:549336#736020
video.ts
#EXTINF:1.000667,
#EXT-X-BYTERANGE:568324#1285356
video.ts
#EXTINF:1.000667,
#EXT-X-BYTERANGE:569264#1853680
video.ts
...
The m3u8 file works perfectly but in its creation, ffmpeg re-creates the ts file. I wanted to avoid this and thought I could simply create the m3u8 file myself. I used the following command to get the byte offset of keyframes. However, none of the keyframe locations agrees with the offsets in the m3u8 file.
ffprobe -loglevel error -skip_frame nokey -select_streams v:0 -show_entries frame=pkt_pos -of compact video.m3u8
frame|pkt_pos=22560
frame|pkt_pos=736396
frame|pkt_pos=1285732
...
All of the offsets disagree by 376 bytes. That number is twice the mpeg-ts package size (which is 188). Both locations contains the ASCII character "G" which is the package header for MPEG-TS.
How can I get the correct offset positions using ffprobe that I can use to create an HLS playlist? Does Ffmpeg just subtract 2 packages for safety, is it important?
ffmpeg re-creates the ts file (which is a byte-wise copy of the original)
No, it’s not a bytewise copy. Ffmpeg still parses and repackages the file. You you need an exact copy, downlod with curl, or wget.
The package the offset points to is a PAT (Program Association Table), followed by a PMT (Program Mapping Table). Each packet takes 188 bytes which makes a total of 376. After this meta data, the actual keyframe starts.
In simple cases the m3u8 offset can point to the keyframe directly and the file will play correctly. However, in general cases, it makes sense for the decoder to be given the list of programs right away when seeking into the middle of a transport stream.
Related
I'm new here.
I have a set of TIF frames that equal 1 minute and 25 seconds of a video.
I'm attempting to copy the frames without re-encoding using the "-c:v copy" function to avoid visible quality loss for a process I'm doing on my side. The command is as follows:
ffmpeg -r 23.977 -i %06d.tif -c:v copy out.mkv
However for some reason, the timing does not seem to be accurate and the video is slightly desynced from the original, ending at 1 minute and 22 seconds instead.
When I use the following command:
ffmpeg -r 23.977 -i %06d.tif out.mkv
It comes out with the proper timing at 1 minute and 25 seconds, however, I did not appreciate the quality loss that came with it.
Is there a workaround to this or is there something I'm missing?
I used both Command Line and Windows Terminal.
In general, it would make sense to transcode when you go from tiff to video format. (I'm surprised it actually works.) You can set encoding quality to your own liking. See [this FFmpeg Wiki article[(https://trac.ffmpeg.org/wiki/Encode/H.264).
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 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!
I am trying to add chapters to a ogg file containing vorbis audio.
From this link I copied the following ffmpeg command.
ffmpeg -threads auto -y -i in.ogg -i metadata_OGG.txt -map_metadata 1 -codec copy out_METADATA.ogg
My metadata_OGG.txt file is as given below.
CHAPTER00=00:00:00.000
CHAPTER00NAME=Chapter 01
CHAPTER01=00:00:05.000
CHAPTER01NAME=Chapter 02
CHAPTER02=00:00:10.000
CHAPTER02NAME=Chapter 03
I am getting the following error.
[ogg # 00000000006d6900] Unsupported codec id in stream 0
Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
But if i change -codec copy to -acodec copy there is no error in ffmpeg but the text file is converted to video. i.e. the output file will have a static video frame with the text of metadata_OGG.txt in it. Also, I observe the following log message during conversion.
Stream #1:0 -> #0:0 (ansi (native) -> theora (libtheora))
Stream #0:0 -> #0:1 (copy)
Anybody please tell me what is going wrong here?
Also, I would like to know what is the right way to add chapters to ogg. I searched for some tools also. I did not get any.
Here is what worked for me using ffmpeg 4.3.1.
I have a metadata file which almost respects ffmpeg's metadata file format:
;FFMETADATA1
title=Evolution theory
[CHAPTER]
TIMEBASE=1/1000
START=0
END=
title=Darwin's point of view
[CHAPTER]
TIMEBASE=1/1000
START=78880
END=
title=Genghis Khan's children
Notice the file format requires an END time, but leaving it empty didn't bother in my case.
Now I add chapter information to my opus/ogg file:
ffmpeg -i darwin.opus.ogg -i darwin_chapters.txt -map_metadata 1 -c copy darwin_withchapters.opus.ogg
Note: if you want to overwrite existing chapter information from the file, you may need to add a -map_chapters 1 parameter in the ffmpeg command line above.
That creates the file darwin_withchapters.opus.ogg. I check if chapter info has really been added to the file:
opusinfo darwin_withchapters.opus.ogg
You would use ogginfo for Ogg/Vorbis files.
And here is the result (I removed a few irrelevant lines):
ENCODER=opusenc from opus-tools 0.1.10
ENCODER_OPTIONS=--bitrate 112
title=Evolution theory
CHAPTER000=00:00:00.000
CHAPTER000NAME=Darwin's point of view
CHAPTER001=00:01:19.880
CHAPTER001NAME=Genghis Khan's children
[...]
Here you go. ffmpeg did the conversion between its metadata file format to the vorbis tag/comment chapter format.
You could also directly write metadata in the Vorbis Chapter Extension format, and use the classic vorbiscomment tool, or other tools which allow editing of opus/ogg in-file tags.
Opus has been mentioned here. I was trying to make opusenc from opus-tools add chapters when encoding and couldn’t find a command line example anywhere. Thanks to the hints in this thread I managed to figure it out, perhaps someone may find it helpful.
opusenc --comment "CHAPTER000=00:00:00.000" --comment "CHAPTER000NAME=Hello" --comment "CHAPTER001=01:23:45.678" --comment "CHAPTER001NAME=World" input.wav output.opus
The chapter key/value scheme is the aforementioned Ogg/Matroska one. Of course, more metadata options like --title, --artist etc. can be added.
Using ffmpeg to add the chapters resulted in two problems for me: The artwork image in the ogg/opus input file was missing in the output file, and ffmpeg rejected empty END chapter times.
I did this on Windows 10 using
opusenc opus-tools 0.2-3-gf5f571b (using libopus 1.3)
ffmpeg version 4.4.1-essentials_build-www.gyan.dev
opusinfo, MPC-HC (64-bit) v1.7.11 and VLC Media Player 3.0.14 Vetinari to confirm.
I found the issue.
For ffmpeg to work, the metadata file should have the following header.
;FFMETADATA1
I followed the steps given in ffmpeg documentation for metadata.
But the issue is not resolved completely.
With the above steps I am able to add metadata to mp4, mkv and other container files but not to ogg files. I am not sure whether ffmpeg supports adding chapters to ogg files.
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.