After pausing icecast radio player app with just audio and audio service packages from Ryan Heise, I wait a little and tap play again, the stream continues to play from saved position with data in buffers, then suddenly (I think when buffers are empty) changes to live position in stream. It is possible for me to clear buffers on pause/stop in my Audio Handler? Or should I use another approach?
I tried player.seek(null), but it needed more work, so I decided to not use it. We haven't encountered this behaviour anymore since a month or so. It works fine now. Thanks!
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I have my own MediaSink in Windows Media Foundation with one stream. In the OnClockStart method, I instruct the stream to queue (i) MEStreamStarted and (ii) MEStreamSinkRequestSample on itself. For implementing the queue, I use the IMFMediaEventQueue, and using the mtrace tool, I can also see that someone dequeues the event.
The problem is that ProcessSample of my stream is actually never called. This also has the effect that no further samples are requested, because this is done after processing a sample like in https://github.com/Microsoft/Windows-classic-samples/tree/master/Samples/DX11VideoRenderer.
Is the described approach the right way to implement the sink? If not, what would be the right way? If so, where could I search for the problem?
Some background info: The sink is an RTSP sink based on live555. Since the latter is also sink-driven, I thought it would be a good idea queuing a MEStreamSinkRequestSample whenever live555 requests more data from me. This is working as intended.
However, the solution has the problem that new samples are only requested as long as a client is connected to live555. If I now add a tee before the sink, eg to show a local preview, the system gets out of control, because the tee accumulates samples on the output of my sink which are never fetched. I then started playing around with discardable samples (cf. https://social.msdn.microsoft.com/Forums/sharepoint/en-US/5065a7cd-3c63-43e8-8f70-be777c89b38e/mixing-rate-sink-and-rateless-sink-on-a-tee-node?forum=mediafoundationdevelopment), but the problem is either that the stream does not start, queues are growing or the frame rate of the faster sink is artificially limited depending on which side is discardable.
Therefore, the next idea was rewriting my sink such that it always requests a new sample when it has processed the current one and puts all samples in a ring buffer for live555 such that whenever clients are connected, they can retrieve their data from there, and otherwise, the samples are just discarded. This does not work at all. Now, my sink does not get anything even without the tee.
The observation is: if I just request a lot of samples (as in the original approach), at some point, I get data. However, if I request only one (I also tried moderately larger numbers up to 5), ProcessSample is just not called, so no subsequent requests can be generated. I send MeStreamStarted once the clock is started or restarted exactly as described on https://msdn.microsoft.com/en-us/library/windows/desktop/ms701626, and after that, I request the first sample. In my understanding, MEStreamSinkRequestSample should not get lost, so I should get something even on a single request. Is that a misunderstanding? Should I request until I get something?
Currently using the lib's from FFPMEG to stream some MPEG2 TS (h264 encoded) video. The streaming is done via UDP multicast.
The issue I am having currently is two main things. There is a long initial connection time / getting the video to show (the stream also contains metadata, and that stream is detected by my media tool immediately).
Once the video gets going things are fine but it is always delayed by that initial connection time.
I am trying to get as near to LIVE streaming as possible.
Currently using the av_dict_set(&dict, "tune", "zerolatency", 0) and "profile" -> "baseline" options.
GOP size = 12;
At first I thought the issue was an i frame issue, but the initial delay is there if gopsize is 12 or default 250. Sometimes the video will connect quickly, but it is immediately dropped, the delay occurs, then it starts back up and is good from that point on.
According to documentation the zero latency option should be sending many i frames, to limit initial syncing delays.
I am starting to think its a buffering type issue, as when I close the application and leave the media player up, it then fast forwards through the delay till it hits basically where the file stops streaming.
So while I don't completely understand what was wrong, I at least fixed the problem I was having.
The issue came from using the av_write_interleaved_frame() vs. the regular av_write_frame()(this one works for live streaming), when writing out the video frames. Ill have to dig into the differences a bit more to fully understand it, but its funny sometimes how you figure out the problem you are having on a total whim after bashing your face for a few days.
I can get pretty good live ish video streaming with the tune "zerolatency" option set.
I am trying to rewind a video file with "-1" rate parameter.
It rewinds for a small duration and then the playback stops. Finally the player gets killed.
However the fast forward for the same video file works fine. I tested it with "2x" and "4x" speed. If I just seek backwards with certain duration (rate is "1.0" ), it goes to that timestamp and starts the playback as expected.
From what I understand, Seek event is handled in the Demuxer element of the pipeline, wherein:
It flushes the currently queued stream data
Creates a new-segment with updated values from the seek event.
Once the new segment is ready with the new stream data, playback starts.
From here on playback will be started,based on the new parameters set in new segment.
For the reverse playback, I'm not able to figure out where the pipeline is actually getting blocked.
I'm able to see the demuxer element is fetching the data and pushing it on the new segment.
Can anyone suggest or point where the issue could be?
Reverse playback might not be properly implemented here. Please file a bug, give as much details about the format (e.g. using gst-discoverer) and if possible link to the file.
I receive audio packets from net (4 packets per second, 250ms each) and video - 15fps. Everything goes with my own timestamps. How should I sync them? I've seen the source code of one of our developers but he did syncing VIDEO according to audio. I.e. audio is always played immediately and video can be dropped or buffered. I don't think it is correct because audio can overrun video for a second or two - in that case we will not have actual video frames at all.
I'd like to know some basics in sync stuff. What should be buffered? Should audio and video in sync mode be played in separate thread(s)? Any clues would be regardful!
Thanks a lot!
I needed in smth like that - http://www.freepatentsonline.com/7680153.html
Pretty difficult to understand but I think this patent explains the basics of sync.
I am writing an application which is a kinda video streamer.The client is receiving a video stream using udp socket.Now as I am receiving the stream I want to play it simultaneous.It is different from playing local video file lying in your hard disk in which case it can be as simple as running the file using system("vlc filename").But here many issues are involved like there can be delay in receiving and player will have to wait for the incoming data.I have come to know about using vlc to run a video stream.Can you please elaborate the step for playing the stream using vlc.I am implementing my application in c++.
EDIT: Can somebody give me some idea regarding VLC API which can be used to stream a given video to particular destination and receive that stream at other end play it.
with regards,
Mawia
Well you can always take a look at VideoLan's own homepage
Other than that, streaming is quite straightforward:
Decide on a video codec that supports streaming. (ok obvious and probably already done)
Choose appropriate packet size.
Choose appropriate video quality.
At the client side: pre-buffer at least 2 secs of video and audio.
Number 2 and 3 sound strange, but they are worth thinking about:
If you have a broadband connection, you can afford to pump big packets over to the client. Note: Packets here means consistent units of data that the client needs to have completely to decode the next bit of video. If you send big packets, say 4 secs of video, you risk lag due to waiting for the complete data unit of, well, full 4 seconds, whilst small 0.5 sec packets would get you laggy but still recognizable and relatively fluent video on a bad connection.
Same goes for quality. Pixelated and artifact ridden videos are bad, stuttering video/sound desyncing videos are worse. Rather switch down to a lower quality/higher compression setting.
If your question is purely about the getting it done part, well, points 1 and 4 should do for you.
You might ask:
"If I want to do real time live video?"
All of the advice above still applies, but all of it has to be done smarter. First things first: You cannot do realtime over bad connections. It's a reality thing. If your connection is fat enough you can reach almost real time, just pump each image and a small sound sample out without much processing or any buffering at all. It is possible to get a good client experience from that, but connections like that are highly unlikely. The trick here usually is, transmit a video quality slightly lower than the connection would allow in theory and still wiggle caching and packet reordering in there... have fun. It is hard.
Unfortunately really the only API vlc has is the command line or equivalent of the command line (you can start player instances, passing them essentially what you would have on the command line). You can use libvlc if you need multiple instances or callbacks but it's pretty opaque still...