I have Live stream of HLS [https://82-80-192-30.vidnt.com/ipbc_IPBCchannel11LVMRepeat/definst/IPBCchannel11LVM_3.stream/playlist.m3u8] and I want to convert it to MPEG-DASH.
What is the best practice?
The stream is already h264 aac therefore I understand I do not need to reencode and I just need to transmux.
What should I use?
ffmpeg? mp4box?
Notes:
I used nginx-rtmp-module (https://github.com/ut0mt8/nginx-rtmp-module/) in order to create DASH from RTMP stream according to this tutorial: https://isrv.pw/html5-live-streaming-with-mpeg-dash
But nginx-rtmp-module can get as input just rtmp streams and it did not work for me with HLS stream.
I used ffmpeg in order to create dash from m3u8 as following:
ffmpeg -i https://82-80-192-30.vidnt.com/ipbc_IPBCchannel11LVMRepeat/_definst_/IPBCchannel11LVM_3.stream/playlist.m3u8 -strict -2 -min_seg_duration 2000 -window_size 5 -extra_window_size 5 -use_template 1 -use_timeline 1 -f dash out.mpd
But this is very limited. I can't control the segment duration.
The min_seg_duration parameter of ffmpeg does not work very well for me, and also it can set the minimum duration while I want to limit the maximum duration of each segment (the segment comes out with ~10 seconds, while I need it to be ~2-4 seconds as I'm playing live).
Firstly it is worth saying that if you can avoid doing this you will be saving yourself a whole lot of work!
Most devices and clients these days can play both HLS and DASH streams, so the usual approach is to add any extra functionality needed in your app or client.
If you do have to convert server side, then its worth being aware that while HLS streams typically used TS segments in the past, recently support for fragmented MP4 has become available within the HLS ecosystem.
If you have TS video streams then you will need to do a conversion along the lines you outline above with ffmpeg.
If you have fragmented MP4 then you should actually have the correct format already and may find you just have to create the manifest file so DASH can access the fragmented mp4 streams.
All the above assumes that your content is not encrypted or that you don't have to support encryption - if it is then you may not be able to convert the media, or you may have to also encrypt the media differently for some streams than others, as currently most deployed windows and chrome devices and browsers use a slightly different encryption approach (a different AES mode) than Apple devices.
Related
I have 1-5 input streams, each uploading on a slightly different time offset.
With rtmp and ffmpeg, I can reliably encode a single stream into an HLS playlist that plays seamlessly on iOS, my target delivery platform.
I know that you can accept multiple input streams into ffmpeg, and I want to switch between the input streams to create a consistent, single, seamless output.
So I want to switch between
rtmp://localhost/live/stream1 .. rtmp://localhost/live/stream5 on a regular interval. Sometimes there will be multiple streams, and sometimes there won't.
Is there any way for ffmpeg to rotate between input streams while generating an HLS playlist? My goal is to avoid running duplicate instances of ffmpeg for server cost reasons, and I think connecting disparately encoded input streams for playback would be difficult if not impossible.
Switching on each segment is the ideal behavior, but I also need to keep the streams in time sync. Is this possible?
Switching live stream inputs can cause delays due to the initial connection time and buffering (rtmp_buffer).
There's no straight-forward way to do it with ffmpeg. Being an open source project you can add the functionality yourself. It shouldn't be very complicated if all all your inputs share the same codecs, number of tracks, frame sizes etc.
Some people suggested using another software to do the switch such as MLT or using filters such as zmq (ZeroMQ) to make ffmpeg accept commands.
One way to do it would be to re-stream the source as mpgets on a local port and use the local address as input in the command that outputs the HLS:
Stream switcher (60s of each stream, one at a time) - you can make a script with your own logic, this is for illustrative purposes:
ffmpeg -re -i rtmp://.../stream1 -t 60 -f mpegts udp://127.0.0.1:10000
ffmpeg -re -i rtmp://.../stream2 -t 60 -f mpegts udp://127.0.0.1:10000
[...]
ffmpeg -re -i rtmp://.../stream5 -t 60 -f mpegts udp://127.0.0.1:10000
Use the local address as source for the HLS stream - it'll wait for input if there's none and fix your DTS/PTS but you will probably introduce some delays on switching:
ffmpeg -re -i udp://127.0.0.1:10000 /path/to/playlist.m3u8
I am starting with a high res video file and I would like to create 3 variants, low quality, mid quality, and high quality for mobile streaming. I want these mid/low/high variants to be segmented into ts pieces that the m3u8 file will be pointing that. Is there a way to do this in one line in ffmpeg?
I have successfully generated an m3u8 file and ts segments with ffmpeg, do I need to do this 3x and set specs for low/mid/high? If so, how do I get the singular m3u8 file to point to all variants as opposed to one for each variant?
This is the command I used to generate the m3u8 file along with the ts segments.
ffmpeg -i C:\Users\george\Desktop\video\hos.mp4 -strict -2 -acodec aac -vcodec libx264 -crf 25 C:\Users\user\Desktop\video\hos_Phone.m3u8
Yes, you need to encode all variants and generate the media playlists first (the playlists containing the segments).
If you want you can do it in one command since ffmepg supports multiple inputs/outputs. Eg:
ffmpeg -i input \
... [encoding parameters 1] ... output1 \
... [encoding parameters 2] ... output2 \
....[encoding parameters 3] ... output3
You must provide the variants in multiple qualities/bitrates but the aspect ratio should remain the same. Keeping the aspect ratio was initially mandatory but in the latest HLS authoring guide it's downgraded to a recommendation.
All variant streams must be keyframe aligned so set a GOP size using the -g option, disable scene-cut detection and use a segment duration hls_time which is a multiple of your keyframe interval.
Once you have all 3x m3u8 media playlist you can manually create the master playlist which points to each media playlist.
Example from the Apple HLS documentation, you must change the bandwidth, codecs, resolution and playlist filenames according to your own encoding options:
#EXTM3U
#EXT-X-VERSION:6
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=2855600,CODECS="avc1.4d001f,mp4a.40.2",RESOLUTION=960x540
medium.m3u8
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=5605600,CODECS="avc1.640028,mp4a.40.2",RESOLUTION=1280x720
high.m3u8
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=1755600,CODECS="avc1.42001f,mp4a.40.2",RESOLUTION=640x360
low.m3u8
The Aspect ratio Does not have to be the same, that makes no sense.
How could you know what the client can play?
Aspect ratios are 4:3 for non-HD, 16:9 for a HD variants.
You don't want to do all your variants in one ffmpeg command if you
need segment times to be consistent.
Also watch transcoding downward, if you go from 1080 to 360, there
might be issues. One that I often get is that the audio degrades and
sounds weird. I try to go down no more than half, if I want high
quality.
#DavidC That hex is the codec version number.
I am developing a player based on ffmpeg.
Now I try to decode hls video. The video stream has several programs (AVProgram) separated by quality. I want to select one specific program with desired quality. But ffmpeg reads packets from all programs (all streams).
How can I tell ffmpeg which streams to read?
Solved by using disard field in AVStream structure:
_stream->discard = AVDISCARD_ALL;
I am streaming short videos (4 or 5 seconds) encoded in H264 at 15 fps in VGA quality from different clients to a server using RTMP which produced an FLV file. I need to analyse the frames from the video as images as soon as possible so I need the frames to be written as PNG images as they are received.
Currently I use Wowza to receive the streams and I have tried using the transcoder API to access the individual frames and write them to PNGs. This partially works but there is about a second delay before the transcoder starts processing and when the stream ends Wowza flushes its buffers causing the last second not to get transcoded meaning I can lose the last 25% of the video frames. I have tried to find a workaround but Wowza say that it is not possible to prevent the buffer getting flushed. It is also not the ideal solution because there is a 1 second delay before I start getting frames and I have to re-encode the video when using the transcoder which is computationally expensive and unnecessarily for my needs.
I have also tried piping a video in real-time to FFmpeg and getting it to produce the PNG images but unfortunately it waits until it receives the entire video before producing the PNG frames.
How can I extract all of the frames from the stream as close to real-time as possible? I don’t mind what language or technology is used as long as it can run on a Linux server. I would be happy to use FFmpeg if I can find a way to get it to write the images while it is still receiving the video or even Wowza if I can find a way not to lose frames and not to re-encode.
Thanks for any help or suggestions.
Since you linked this question from the red5 user list, I'll add my two cents. You may certainly grab the video frames on the server side, but the issue you'll run into is transcoding from h.264 into PNG. The easiest was would be to use ffmpeg / avconv after getting the VideoData object. Here is a post that gives some details about getting the VideoData: http://red5.5842.n7.nabble.com/Snapshot-Image-from-VideoData-td44603.html
Another option is on the player side using one of Dan Rossi's FlowPlayer plugins: http://flowplayer.electroteque.org/snapshot
I finally found a way to do this with FFmpeg. The trick was to disable audio, use a different flv meta data analyser and to reduce the duration that FFmpeg waits for before processing. My FFmpeg command now starts like this:
ffmpeg -an -flv_metadata 1 -analyzeduration 1 ...
This starts producing frames within a second of receiving an input from a pipe so writes the streamed frames pretty close to real-time.
I want to stream video (no audio) from a server to a client. I will encode the video using libx264 and decode it with ffmpeg. I plan to use fixed settings (at the very least they will be known in advance by both the client and the server). I was wondering if I can avoid wrapping the compressed video in a container format (like mp4 or mkv).
Right now I am able to encode my frames using x264_encoder_encode. I get a compressed frame back, and I can do that for every frame. What extra information (if anything at all) do I need to send to the client so that ffmpeg can decode the compressed frames, and more importantly how can I obtain it with libx264. I assume I may need to generate NAL information (x264_nal_encode?). Having an idea of what is the minimum necessary to get the video across, and how to put the pieces together would be really helpful.
I found out that the minimum amount of information are the NAL units from each frame, this will give me a raw h264 stream. If I were to write this to a file, I could watchit using VLC if adding a .h264
I can also open such a file using ffmpeg, but if I want to stream it, then it makes more sense to use RTSP, and a good open source library for that is Live555: http://www.live555.com/liveMedia/
In their FAQ they mention how to send the output from your encoder to live555, and there is source for both a client and a server. I have yet to finish coding this, but it seems like a reasonable solution