I am using AVAudioPlayer.
I have a short sound in my application.
Randomly it does not play , Play API does not return any error,
Volume is also proper that moment. I am not getting what is going wrong.
Interesting thing is If I play some other sound file, It does play properly every time.
Then I did check properties of both the sound files.
The both the sound files have same bit rate, only difference is
The one which does not play randomly has 00.02 duration.
and the one which play properly has 00.00 duration.
Can anybody give me idea to fix this issue?.
Related
I have coded two versions of video player based on QMediaPlayer and on Vlc-qt. In both cases I have the incorrect value for the full time video. The player show me that total time is 7 seconds, but in fact the time approx. 5 minutes. And of course, the slider of position shows not correct.
I was confused, so that maybe I did something wrong. But this video file was tested with MS video player, and I see the same problem.
Video for testing can be found at https://1drv.ms/u/s!AgCzZ90Ttbz65jqiluS2NS95Id0U
My guess is that the file contains the wrong information about the time. Or maybe not codec provides such information in not correct way.
Can anybody clarify to me what the reason of the problem and how it should be fixed.
My application must read one video track and several audio tracks, and be able to specify one section of the file and play it in loop. I have created a setup with Media Foundation, using the sequencer source and creating several topologies with the start and end point of the section I want to loop. It works, except for the fact that there is a 0.5 to 1 sec time of stabilization of the playback just when it goes back to the starting point.
First, I made it with individual audio files and one video file. This was quite bad for some files, sometimes all the files were completely out of sync, sometimes the video was frozen for several seconds, then went very fast to catch with the audio.
I had a good improvement using only one file, that includes the video and the multiple audio tracks. However, for most files, there is still a problem about the smoothness of the transition.
With a poor quality video AVI file, I could make it work smoothly, which would mean that the method I use is correct. I have noticed that the quality of the loop smoothness is strongly related to the CPU used on a file when simply playing it.
I use the "SetTopology" on the session, using a series of topologies, so normally it should preroll the next one during the playback of the current one, right ? Or am I missing something there ?
My app works also on Mac, where I have used a similar setup with AVFoundation, and it works fine with the same media files I use on Windows.
What can I do to have the looping work smoothly with better quality video on Windows ? Is there something to do about it ?
When I play the media file without looping, I notice that when I preroll it to some point, then when I hit the START button, the media starts instantly and with no glitch. Could it work better if I was using two independent simple playback setups, start the first, preroll the second, then stop the first and start the second programmatically at the looping point ?
I have a DirectShow webcam application. I make use of Sample Grabber to get the buffer callbacks and IVideoWindow to control the display co-ordinates for the Preview. I have Preview and Capture Streams which I run as below.
g_pBuild->RenderStream(&PIN_CATEGORY_CAPTURE, &MEDIATYPE_Video,cam,g_pGrabberF,pNullRenderer2); g_pBuild->RenderStream(&PIN_CATEGORY_PREVIEW, &MEDIATYPE_Video,cam,NULL,NULL);
On certain On board cameras, IMediaControl::Run followed by IMediaControl::Stop followed by IMediaCOntrol::Run doesn't switch on the camera.
Extenal USB cameras work properly here. How can I diagnose more on this? Any pointers, please help.
Maybe its specific to a certain hardware issue in the unit.
Do a quick test by adding sleep of 1 sec between calls.
If it does help than you need to find a way to know when to unit state in idle or not.
There are two important parts of the question which you did not provide:
Filter graph topologies
HRESULTs of the method calls
A problem you might be having is that one of the filters in the topology does not handle well state transitions and fails somewhere between states. Supposedly your second Run meets it still trying to complete Stop. You might get a HRESULT there which indicates the issue (better for you) or the filter fails silently.
The filter graph's is the unlikely source of the bug itself. Chances are high that it does everything flawlessly, however since internally it distributes the calls between filters, one of the filter is letting you down.
In MXML-based apps, you set the target framerate for the app and I believe this is a core part of Flash as well.
Two questions...
In many games you want the game to run as fast as it can for graphical smoothness, with some upper cap like 50-100Hz. How can you have variable framerates in Flash, or is it really not how things work?
What happens when an app cannot run at the target framerate? Do updates 'stack up' leading to other problems, or does Flash discard them?
I've a simple case where I'm moving a sprite from left to right at 100px/s, at 20fps it looks un-smooth. It's not jerky or anything, but you can clearly see the stepped motion, the artwork is black/white which accentuates it I think. I reckon a higher FPS is needed ideally, but on slower systems it might be too much and I don't want to run into nasty issues where I try to drive it too fast.
If the ActionScript takes too much time and the player can't maintain the frame rate that was specified, the frame rate just drops. There's nothing to stack up, you just get frames generated less often. For this reason when frame rate is important it's critical to make sure the AS code is not taking more time than you have available given the desired frame rate. Also make sure all calculates for movement are based on time and not frames.
As far as animation as 20fps, yeah, it will not look smooth. Bump up the frame rate. :-)
http://www.morearty.com/blog/2006/07/17/flex-tip-a-higher-frame-rate-even-makes-text-entry-look-better/
If Flash is unable to execute at the desired framerate, it will begin dropping frames. You can read the details of this here.
If you are dealing with framerates in Flash a lot, it can be helpful to understand the 'elastic racetrack' model that Flash uses. You can see details about it here, but the basic idea is that the amount of time spend executing code or rendering a frame can vary on a frame by frame basis.
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.
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