I'm using Directshow SampleGrabber in callback mode to capture video frame from source file and do some processing. Also I would like to maintain the current playback rate of video and need to support both random, forward and backward seeking. For this I'm also doing some local buffering in a different thread.
I'm running graph with syn source set to NULL, so as to get maximum speed. However when I pause the graph after fixed amount of buffering. The SampleGrabber callback is getting called spuriously even when graph is paused. This is affecting my frame indexing and tracking. I want to resume the graph exactly from the same position at which it was paused. However if I run the graph with default clock it works fine but then my playback get affected. I want buffering thread to finish as soon as possible.
How can I make sure that callback is not called when graph is paused? Any thoughts or suggestion would be of great help.
Thanks in advance
Pradeep
Paused graph typically has all the same streaming internally (active state) with the exception that renderers are blocking streaming, esp. as soon as enough data is received for a preview banner. Since you removed clock from the graph, your renderer is likely to not block execution because it does not hold any clock to pause against. In your case this is the problem coming out of your intent to reuse the same graph for quick parsing through file and playback. Separate graph design looks having more chances to do better.
Related
I am using the excellent GMFBridge directshow family of filters to great effect, allowing me to close a video recording graph and open a new one, with no data-loss.
My original source graph was capturing live video from standard video and audio inputs.
There is an undocumented method on the GMFBridgeController filter named SetLiveTiming(). From the name, I figured that this should be set to true if we are capturing from a Live graph (not from a file) as is my case. I set this value to true and everything worked as expected
The same capture hardware allows me to capture live TV signals (ATSC in my case), so I created a new version of the graph using the BDA architecture filters, for tuning purposes. Once the data flows out from the MPEG demuxer, the rest of the graph is virtually the same as my original graph.
However, on this ocassion my muxing graph (on the other side of the bridge) was not working. Data flowed from the BridgeSource filter (video and audio) and reached an MP4 muxer filter, however no data was flowing from the muxer output feeding a FileWriter filter.
After several hours I traced the problem to the SetLiveTiming() setting. I turned it off and everything began working as expected. and the muxer filter began producing an output file, however, the audio was not synchronized to the video.
Can someone enlighten me on the real purpose of the SetLiveTiming() setting and perhaps, why one graph works with the setting enabled, while the other fails?
UPDATE
I managed to compile the GMFBridge Project, and it seems that the filter is dropping every received sample because of a negative timestamp computation. However I am completely baffled at the results I am seeing after enabling the filter log.
UPDATE 2: The dropped samples were introduced by the way I launched the secondary (muxer) graph. I inspected a sample using a SampleGrabber (thus inside a streaming thread) as a trigger-point and used a Task.Run() .NET call to instantiate the muxer graph. This somehow messed up the clocks and I ended having a 'reference start point' in the future - when the bridge attempted to fix the timestamp by subtracting the reference start point, it produced a negative timestamp - once I corrected this and spawned the graph from the application thread (by posting a graph event), the problem was fixed.
Unfortunately, my multiplexed video (regardless of the SetLiveTiming() setting) is still out of sync.
I read that the GMFBridge filter can have trouble when the InfTee filter is being used, however, I think that my graph shouldn't have this problem, as no instance of the InfTee filter is directly connected to the bridge sink.
Here is my current source graph:
-->[TIF]
|
[NetworkProvider]-->[DigitalTuner]-->[DigitalCapture]-->[demux]--|-->[Mpeg Tables]
|
|-->[lavAudioDec]-->[tee]-->[audioConvert]-->[sampleGrabber]-->[NULL]
| |
| |
| ->[aacEncoder]----------------
| |--->[*Bridge Sink*]
-->[VideoDecoder]-->[sampleGrabber]-->[x264Enc]--------
Here is my muxer graph:
video
... |bridge source|-------->[MP4 muxer]--->[fileWriter]
| ^
| audio |
---------------------
All the sample grabbers in the graph are read-only. If I mux the output file without bridging (by placing the muxer on the capture graph), the output file remains in sync, (this ended being not true, the out-of-sync problem was introduced by a latency setting in the H264 encoder) but then I can't avoid losing some seconds between releasing the current capture graph, and running the new one (with the updated file name)
UPDATE 3:
The out of sync problem was inadvertently introduced by me several days ago, when I switched off a "Zero-latency" setting in the x264vfw encoder. I hadn't noticed that this setting had desynchronized my already-working graphs too and I was blaming the bridge filter.
In summary, I screwed up things by:
Launching the muxer graph from a thread other than the Application
thread (the thread processing the graph's event loop).
A latency switch in an upstream filter that was probably delaying
things too much for the muxer to be able to keep-up.
Author's comment:
// using this option, you can share a common clock
// and avoid any time mapping (essential if audio is in mux graph)
[id(13), helpstring("Live Timing option")]
HRESULT SetLiveTiming([in] BOOL bIsLiveTiming);
The method enables a special mode of operation which addresses live data. In this mode sample times are converted between the graphs as relative to respective clock start times. Otherwise, the default mode is to expect reset of time stamps to zero with graph changes.
I'm trying to use a customized filter to receive video and audio data from a RTSP stream, and deliver samples downstream the graph.
It seems like that this filter was modified from the SDK source.cpp sample (CSource), and implemented two output pins for audio and video.
When the filter is directly connected to an avi mux filter with INTERLEAVE_NONE mode, it works fine.
However, when the interleave mode of avi mux is set to INTERLEAVE_CAPTURE,
the video output pin will hang on the GetDeliveryBuffer method (in DoBufferProcessingLoop) of this filter after several samples have sent,
while the audio output pin still works well.
Moreover, when I inserted an infinite pin tee filter into one of the paths between the avi mux and this source filter,
the graph arbitrarily turned into stop state after some samples had been sent (one to three samples or the kind).
And when I put a filter that is just an empty trans-in-place filter which does nothing after the infinite tee,
the graph went back to the first case: never turns to stop state, but hang on the GetDeliveryBuffer.
(Here is an image that shows the connections I've mentioned like)
So here are my questions:
1: What could be the reasons that the video output pin hanged on the GetDeliveryBuffer ?
In my guess it looks like the avi mux caught these sample buffers and did not release them until they are enough for interleaving,
but even when I set the amount of video buffers to 30 in DecideBufferSize it will still hang. If the reason is indeed like that, so how do I decide the buffer size of the pin for a downstream avi muxer ?
Likely a creation of more than 50 buffers of a video pin is not guaranteed to work because the memory size cannot be promised. :(
2: Why does the graph goes to stop state when the infinite pin tee is inserted ? And why could a no-operation filter overcomes it ?
Any answer or suggestion is appreciated. Or hope someone just give me some directions. Thanks.
Blocked GetDeliveryBuffer means the allocator, you are requesting a buffer from, does not [yet] have anything for you. All media samples are outstanding and are not yet returned back to the allocator.
An obvious work around is to request more buffers at pin connection and memory allocator negotiation stage. This however just postpones the issue, which can very much similarly appear later for the same reason.
A typical issue with a topology in question is related to threading. Multiplexer filter which has two inputs will have to match input streams to produce a joint file. Quite so often on runtime it will be holding media samples on one leg while expecting more media samples to come on the other leg on another thread. It is assumes that upstream branches providing media samples are running independently so that a lock on one leg is not locking the other. This is why multiplexer can freely both block IMemInputPin::Receive methods and/old hold media samples inside. In the topology above it is not clear how exactly source filter is doing threading. The fact that it has two pins make me assume it might have threading issues and it is not taking into account that there might be a lock downstream on multiplexer.
Supposedly source filter is yours and you have source code for it. You are interested in making sure audio pin is sending media samples on a separate thread, such as through asynchronous queue.
I was using a callback mechanism to grab the webcam frames in my media application. It worked, but was slow due to certain additional buffer functions that were performed within the callback itself.
Now I am trying the other way to get frames. That is, call a method and grab the frame (instead of callback). I used a sample in CodeProject which makes use of IVMRWindowlessControl9::GetCurrentImage.
I encountered the following issues.
In a Microsoft webcam, the Preview didn't render (only black screen) on Windows 7. But the same camera rendered Preview on XP.
Here my doubt is, will the VMR specific functionalities be dependent on camera drivers on different platforms? Otherwise, how could this difference happen?
Wherever the sample application worked, I observed that the biBitCount member of the resulting BITMAPINFOHEADER structure is 32.
Is this a value set by application or a driver setting for VMR operations? How is this configured?
Finally, which is the best method to grab the webcam frames? A callback approach? Or a Direct approach?
Thanks in advance,
IVMRWindowlessControl9::GetCurrentImage is intended for occasional snapshots, not for regular image grabbing.
Quote from MSDN:
This method can be called at any time, no matter what state the filter
is in, whether running, stopped or paused. However, frequent calls to
this method will degrade video playback performance.
This methods reads back from video memory which is slow in first place. This methods does conversion (that is, slow again) to RGB color space because this format is most suitable for for non-streaming apps and gives less compatibility issues.
All in all, you can use it for periodic image grabbing, however this is not what you are supposed to do. To capture at streaming rate you need you use a filter in the pipeline, or Sample Grabber with callback.
I'm using a simple DirectShow graph to convert some videos to WMV format, which is working fine. I'm now trying to use a filter based on the Synth Filter sample to supply a silent audio track to the videos and I'm running into some problems.
Essentially, I don't know how to stop the graph when this filter (the synth filter) is connected. I guess because it just provides samples forever until somebody tells it to stop, the usual approach of calling IMediaEvent::WaitForCompletion on the filter graph doesn't work (the graph never stops). What I want it to do of course is stop as soon as the video source filter is finished.
I've tried tracking the position of the graph with IMediaSeeking::GetPositions and then manually stopping the graph when this exceeds the duration of the source file, but the accuracy of the stop time with this approach isn't great.
Can anyone think of a better way to do this? Do I need to have another filter that monitors the output from the video source and also has a pointer to the audio source so it can stop it as soon as the video source delivers EndOfStream? Is there no way to accomplish this from purely application-side code?
I've done something not too different myself in the past. I added support for IMediaSeeking to the silence generator filter, and then you need to make sure that you set start and stop times for the conversion (even if it's just 0 and duration), so that the silence generator can generate the right amount of audio and then send EOS.
G
I am attempting to use DirectShow to play two AVI files consecutively (one after the other) so that there is no interruption in the audio or video when the player transitions from one file to the next.
I have two custom controls on my form. Each one is pre-loaded with an AVI file, and before playback begins I set up all the DirectShow interfaces, set the video windows and resize them, call IMediaControl.Run(), then IMediaControl.Pause(), then IMediaSeeking.SetPositions to reset to frame 0, on both controls. On the form, you can see that both files are paused at their initial frames.
I then call IMediaControl.Run() on the first control, and wait for it to complete before calling Run() on the second control. Initially, I hooked into the first video's EC_COMPLETE notification message, and used this to start the second. Thinking that this event might be slow to arrive (turns out it is, but for a weird reason), I tried two other approaches:
Check the first video's current position inside a timer that goes off every second or so (using IMediaPosition.get_CurrentPosition). When the current position is within a second of the video's stop time (known in advance from IMediaPosition.get_StopTime), I go into a tight while loop and wait for the current position to equal the stop time, and then call Run() on the second video.
Same as the first, except I replace the while loop with a call to timeSetEvent from winmm.dll, with a delay set so that it fires right when the first file is supposed to end. I use the callback to Run() the second file.
Either of these two methods substantially cuts down the delay between the end of the first file and the beginning of the second, indicating that the EC_COMPLETE message doesn't arrive immediately after the file is complete (I also tried hooking the EC_SEGMENT_COMPLETE message, which is supposed to be used for looping within a file, but apparently nobody supports this - it never occurs on my machine, at least).
Doing all of the above has cut the transition delay from as much as a second, down to a barely perceptible glitch; about a third of the time the files transition with no interruption at all, which suggests there's no fundamental reason I can't get this to work all the time.
The slight delay is still unacceptable, unfortunately. I assume (and I could easily be wrong) that the remaining delay is due to a slight variable delay between the call to IMediaControl.Run() and when the video actually starts playing.
Does anybody know anything I can do to eliminate this little lag? It would also help to be told this is fundamentally impossible for whatever reason, which wouldn't surprise me. I've never encountered a video player in Windows that doesn't have this problem, so it may not be doable.
More info: the AVI files I'm playing are completely uncompressed (video and audio are uncompressed), so I don't think the lag is due to DirectShow's having to uncompress the video ahead of play start, although it may still buffer ahead as matter of course (and this may be the source of the problem). I would have though that starting play, pausing and then rewinding to the beginning would fix this.
Also, the way I'm handling the transition is to actually have the second control underneath the first; when the first completes playing, I start the second and then call BringToFront on it, creating the appearance of a single video transitioning between the two originals. I don't think the glitch is due to this, because it works perfectly some of the time, and even if this were problematic, it wouldn't explain the matching audio glitch.
Even more: I just tried starting the second video 30-50 milliseconds "early" and that seemed to eliminate even more of the gap, so I'm guessing that the lag in Run() is about that long. It appears to be variable, though, so this is still not where I need it to be.
Still more: perhaps I could eliminate this delay by loading the AVIs from memory rather than from a file. Unfortunately, I have no idea how to do this. IMediaControl only has a RenderFile() method, not something like a RenderStream or RenderMemory method.
If you call IMediaControl::Run on a stopped graph, the graph manager will post the call to a worker thread (so there's some variability). On the worker thread, the graph will be paused. Render filters only complete a pause transition once they have received data, so once GetState() returns S_OK, the graph manager knows that the graph is fully cued. At this point, it picks a time roughly 10ms into the future, and calls Run on each filter with that time as the start point. Since it takes time to tell each filter to Run, the dshow Run method has a parameter which is the refclock time at which a sample timestamped zero should be played -- i.e. the time at which the actual transition to run mode should take place.
To synchronise this with another graph, you first have to ensure that both graphs have the same clock. Query the graph (not the filter) for IMediaFilter, and call GetSyncSource on one graph and SetSyncSource on the other. Then you need to pause the second graph, so that it is cued and ready. When you want to start it, call IMediaFilter::Run instead of IMediaControl::Run, and you can pass your own start time. This still has to be a few milliseconds into the future, so the best thing might be to set the start time of the second graph to be the first graph's start time plus its duration (for an indexed container of uncompressed streams, the duration should be accurate).
Another approach is to use multiple graphs. Separating source from rendering would allow you to switch seamlessly between sources since they feed into a common render graph. There is sample source code for this approach at www.gdcl.co.uk/gmfbridge.
G