I can download http://www.w6rz.net/adv8dvbt23.ts.
And there are many samples for dvbt sample ts files.
But, I want to convert my video file to TS file for dvbt.
First, I checked on google, but I cannot find any answer.
I think, this does not make sense, or, the way of thinking may have been wrong.
FFmpeg can used for this?
but, there is no any parmameter for Transmit mode, QAM / 64QAB, guard interval.
FFmpeg can used for this? but, there is no any parmameter for Transmit mode, QAM / 64QAB, guard interval.
As I explained already:
ffmpeg doesn't know anything about RF things like Constellation type; it is just a tool to transcode between different video formats. .ts is for "transport stream", and it's the video container format that DVB uses. The GNU Radio transmit flowgraphs on the other hand know nothing about video things – all they do is take the bits from a file. So that file needs to be in a format that the receiver would understand, and that's why I instructed you to use FFMPEG with the parameters you need. Since I don't know which bitrate you're planning on transmitting, I can't help you with how to use ffmpeg
So, you need to generate video data that your DVB-T receiver understands, but more importantly even, you need to put them in a container that ensures constant bitrate.
As pointed out in a different comment to your ham.stackexchange.com question about the topic, your prime source of examples would be GNU Radio's own gr-dtv module; when you look into gnuradio/gr-dtv/examples/README.dvbt, you'll find a link to https://github.com/drmpeg/dtv-utils , W6RZ's own tooling :)
There you'll find the tools necessary to calculate the exact stream bitrate you need your MPEG transport stream to have. Remember, a DVB-T transmitter has to transmit at a constant bits per second, so your video container must be constant-bitrate. That's why a transport stream pads the video data to achieve constant rate.
Then, you'll use ffmpeg to transcode your video and put into the transport stream container:
ffmpeg -re -i inputvideo.mpeg \
-vcodec mpeg2video \
-s 720x576 #resolution; this is a good choice, since most TVs will deal with it \
-r 25 #frames per second, use 25\
-flags cgop+ilme -sc_threshold 1000000000 #MPEG codec options\
-b:v 2M #Video *codec data* bit rate (defines video quality). Must be lower than stream bit rate, so < muxrate-(audio bitrate)\
-minrate:v 2M -maxrate:v 2M #enforce constant video bit rate\
-acodec mp2 -ac 2 -b:a 192k #audio codec, quality and bitrate\
-muxrate ${RATE FROM TOOL}
-f mpegts #specify you want a MPEG Transport Stream container as output\
outputfile.ts
Related
I am creating streamable audio by using ffmpeg like this:
ffmpeg -i test.mp3 -vn -ac 2 -acodec aac -f segment -segment_format mpegts -segment_time 10 -segment_list playlist.m3u8 playlist-%05d.ts
My goal was to create content in the HLS format and it seems to work (it is playable by hls.js). I just wonder about the segment_format which is mpegts, is that correct? By reading the answer here mpegts seems to be a different format than HLS. Which is it?
The ffmpeg script above is generating .ts-files for the chunks and an .m3u8-file for the playlist. Is it possible to play this playlist on all devices that supports m3u-playlist? What is the difference between old Winamp-m3u-playlists and the hls format?
Is the .ts files still in the same content type as the mp3?
mpegts seems to be a different format than HLS. Which is it?
It’s both. HLS is not a file type, it a collect of technologies used together to encode and deliver media over Http. One of the technologies is TS
Is it possible to play this playlist on all devices that supports m3u-playlist?
All devices? Probably not. Nobody can guarantee something to work on all devices. You need to test the devices you care about.
Is the .ts files still in the same content type as the mp3?
What do you mean by “content type”? Your ffmpeg command has a mp3 an input and a ts wrapped aac stream out. So no, I guess. Unless by content type you mean audio, then yes.
Can someone tell me what server-side technology (perhaps ffmpeg), one could use in order to:
1) display this full-screen live-streaming video:
http://aolhdshls-lh.akamaihd.net/i/gould_1#134793/master.m3u8
2) and overlay it in the lower-right corner with a live video coming from a webRTC video-chat stream?
3) and send that combined stream into a new m3u8 live-stream
4) Note that it needs to be a server-side solution - - - cannot launch multiple video players in this case (needs to pass the resulting stream to SmartTV's which only have one video-decoder at a time)
The closest example I've found so far is this article:
https://trac.ffmpeg.org/wiki/Create%20a%20mosaic%20out%20of%20several%20input%20videos
Which isn't really live, nor is it really doing overlays.
any advice is greatly appreciated.
Let me clear what you want in this case :
Input video is HLS streaming from webRTC : What about delay? is dealy important thing in your work?
Overlay image into video : This will need that decoding input video, and filtering it, encoding again. so it needs a lot of cpu resource and even more if input video is 1080p.
Re-struct new HLS format : You must put it lot of encoding option to make sure that ts fragment works well. most important thing is GOP size and ts duration.
You need a web server to provide m3u8 index file. you can use nginx, apache.
What i tell you now in this answer is that ffmpeg command line, which making overlay from input HLS streaming and re-make ts segments.
Following command-line will do what you want in step 1 to step 3 :
ffmpeg \
-re -i "http://aolhdshls-lh.akamaihd.net/i/gould_1#134793/master.m3u8" \
-i "[OVERLAY_IMAGE].png" \
-filter_complex "[0:v][1:v]overlay=main_w:main_h[output]" \
-map [output] -0:a -c:v libx264 -c:a aac -strict -2 \
-f ssegment -segment_list out.list out%03d.ts
This is basic command line that overlay image from your input HLS streaming then creating ts segment and index file.
I don't have any further experience with HLS so it can be done without any tuning option but maybe you should tune it for you work. and also you should search a little bit for web server to provide m3u8 but it won't be hard.
GOP size(-g) and its duration(segment_tim), as i said, will be key point of your tuning.
I need convert all videos to my video player (in website) when file type is other than flv/mp4/webm.
When I use: ffmpeg -i filename.mkv -sameq -ar 22050 filename.mp4 :
[h264 # 0x645ee0] error while decoding MB 22 1, bytestream (8786)
My point is, what I should do, when I need convert file type: .mkv and other(not supported by jwplayer) to flv/mp4 without quality loss.
Instead of -sameq (removed by FFMpeg), use -qscale 0 : the file size will increase but it will preserve the quality.
Do not use -sameq, it does not mean "same quality"
This option has been removed from FFmpeg a while ago. This means you are using an outdated build.
Use the -crf option instead when encoding with libx264. This is the H.264 video encoder used by ffmepg and, if available, is the default encoder for MP4 output. See the FFmpeg H.264 Video Encoding Guide for more info on that.
Get a recent ffmpeg
Go to the FFmpeg Download page and get a build there. There are options for Linux, OS X, and Windows. Or you can follow one of the FFmpeg Compile Guides. Because FFmpeg development is so active it is always recommended that you use the newest version that is practical for you to use.
You're going to have to accept some quality loss
You can produce a lossless output with libx264, but that will likely create absolutely huge files and may not be decodeable by the browser and/or be supported by JW Player (I've never tried).
The good news is that you can create a video that is roughly visually lossless. Again, the files may be somewhat large, but you need to make a choice between quality and file size.
With -crf choose a value between 18 to around 29. Choose the highest number that still gives an acceptable quality. Use that value for your videos.
Other things
Add -movflags +faststart. This will relocate the moov atom from the end of the file to the beginning. This will allow the video to begin playback while it is still being downloaded. Otherwise the whole video must be completely downloaded before it can begin playing.
Add -pix_fmt yuv420p. This will ensure a chroma subsampling that is compatible for all players. Otherwise, ffmpeg, by default and depending on several factors, will attempt to minimize or avoid chroma subsampling and the result is often not playable by non-FFmpeg based players.
convert all mkv to mp4 without quality loss (actually it is only re-packaging):
for %a in ("*.mkv") do ffmpeg.exe -i "%a" -vcodec copy -acodec copy -scodec mov_text "%~na.mp4"
For me that was the best way to convert it.
ffmpeg -i {input} -vcodec copy {output}
I am writing a script in python that appends multiple .webm files to one .mp4. It was taking me 10 to 20 seconds to convert one chunk of 5 seconds using:
ffmpeg -i {input} -qscale 0 copy {output}
There's some folders with more than 500 chunks.
Now it takes less than a second per chunk. It took me 5 minutes to convert a 1:20:00 long video.
For MP3, the best is to use -q:a 0 (same as '-qscale 0'), but MP3 has always loss quality.
To have less loss quality, use FLAC
See this documentation link
I'm on Windows 7 and i have many .MP4 video that i want to convert on .flv. I have try ffmpeg and Free FLV converter, but each time the results are not what i'm looking for.
I want a video of same quality (or almost, looking good) and a more little size for the video, because right now, each time i have made a try, the video result is pretty bad and the video size just increase.
How can i have a good looking video, less in size and in .FLV ?
Thanks a lot !
First, see slhck's blog post on superuser for a good FFmpeg tutorial. FLV is a container format and can support several different video formats such as H.264 and audio formats such as AAC and MP3. The MP4 container can also support H.264 and AAC, so if your input uses these formats then you can simply "copy and paste" the video and audio from the mp4 to the flv. This will preserve the quality because there is no re-encoding. These two examples do the same thing, which is copying video and audio from the mp4 to the flv, but the ffmpeg syntax varies depending on your ffmpeg version. If one doesn't work then try the other:
ffmpeg -i input.mp4 -c copy output.flv
ffmpeg -i input.mp4 -vcodec copy -acodec copy output.flv
However, you did not supply any information about your input, so these examples may not work for you. To reduce the file size you will need to re-encode. The link I provided shows how to do that. Pay special attention to the Constant Rate Factor section.
I use ffmpeg to encode a video file to an mpeg transport stream (.ts), which is subsequently sent over network. If there is any network bandwidth fluctuation, I want to dynamically change the stream's bitrate.
My current solution involves restarting ffmpeg with a different bitrate as below
`ffmpeg -i input.avi -ss <resume point> -b:v <new bitrate> output.ts
Unfortunately, for certain i/p file formats, glitches get introduced in the video stream using the above approach. So I am looking for a solution where ffmpeg's output bitrate can be changed dynamically, possibly using signals