Custom player using NDK/C++/MediaCodec - starvation/buffering in decoder - android-ndk

I have a very interesting problem.
I am running custom movie player based on NDK/C++/CMake toolchain that opens streaming URL (mp4, H.264 & stereo audio). In order to restart from given position, player opens stream, buffers frames to some length and then seeks to new position and start decoding and playing. This works fine all the times except if we power-cycle the device and follow the same steps.
This was reproduced on few version of the software (plugin build against android-22..26) and hardware (LG G6, G5 and LeEco). This issue does not happen if you keep app open for 10 mins.
I am looking for possible areas of concern. I have played with decode logic (it is based on the approach described as synchronous processing using buffers).
Edit - More Information (4/23)
I modified player to pick a stream and then played only video instead of video+audio. This resulted in constant starvation resulting in buffering. This appears to have changed across android version (no fix data here). I do believe that I am running into decoder starvation. Previously, I had set timeouts of 0 for both AMediaCodec_dequeueInputBuffer and AMediaCodec_dequeueOutputBuffer, which I changed on input side to 1000 and 10000 but does not make much difference.
My player is based on NDK/C++ interface to MediaCodec, CMake build passes -DANDROID_ABI="armeabi-v7a with NEON" and -DANDROID_NATIVE_API_LEVEL="android-22" \ and C++_static.
Anyone can share what timeouts they have used and found success with it or anything that would help avoid starvation or resulting buffering?

This is solved for now. Starvation was not caused from decoding perspective but images were consumed in faster pace as clock value returned were not in sync. I was using clock_gettime method with CLOCK_MONOTONIC clock id, which is recommended way but it was always faster for first 5-10 mins of restarting device. This device only had Wi-Fi connection. Changing clock id to CLOCK_REALTIME ensures correct presentation of images and no starvation.

Related

Programmatic access to a sound played through OpenAL

I am working with an application that uses OpenAL API quite extensively. In particular, there are multiple sound sources, non-trivial listener filters, etc.
I want to be able to run this application significantly faster than real-time. At the same time, the sound must be saved for later postprocessing. Is there a way to access the OpenAL output programmatically (virtually) without ever playing the sound on the real playback device?
Ideally, I'd like to have access that would be played during every tick of the main loop of my application. Normally one tick corresponds to one rendered frame (e.g. 1/30th of a second). But in this case we would be running the app as fast as possible.
We ended up using OpenAL Soft to do this. Example:
#include "alext.h"
LPALCLOOPBACKOPENDEVICESOFT alcLoopbackOpenDeviceSOFT;
alcLoopbackOpenDeviceSOFT = alcGetProcAddress(NULL,"alcLoopbackOpenDeviceSOFT");
replace your default device with this device
ALCcontext *context = alcCreateContext(device, attrs);
Set the attrs as you would for your default device
Then in the main loop use:
LPALCRENDERSAMPLESSOFT alcRenderSamplesSOFT;
alcRenderSamplesSOFT = alcGetProcAddress(NULL, "alcRenderSamplesSOFT");
alcRenderSamplesSOFT(device, buffer, 1024);
Here the buffer will store 1024 samples. This code runs faster than real-time, therefore you can sample frames every tick
Are you able to do your required functions with the audio data prior to its being shipped to OpenAL? I've done a lot with javax.sound.sampled when it is untethered by the blocking write() method in SourceDataLine, especially when saving to file rather than playing back.
From what little I know about OpenAL, there is also a blocking process occurs when data is shipped, with a queue of arrays that are managed. I've been meaning to look into this further...
(Probably not being very helpful here. Apologies.)

Sync two soundcards

I have a program written in C++ that uses RtAudio ( Directsound ) to capture and playback audio at 48kHz samplerate.
The input capture uses a callback option. The callback writes data to a ringbuffer.
The output is a blocking write function in a separate thread that reads from the ringbuffer.
If the input and output devices are the same the audio loops thru perfectly.
Now I want to get audio from device 1 and playback on device 2. Each device has its own sampleclock set to 48kHz but are not in sync. After a couple of seconds the input and output are out of sync.
Is it possible to sync two independent oudio devices?
There are two challenges you face:
getting the two devices to start at the same time.
getting the two devices to stay in sync.
Both of these tasks are difficult. In the pro audio world, #2 is accomplished with special hardware to sync the word-clocks of multiple devices. It can also be done with a high quality video signal. I believe it can also be done with firewire devices, but I'm not sure how that works. In practice, I have used devices with no sync ("wild") and gotten very reasonable sync for up to an hour or two. Depending on what you are trying to do, the sync should not drift more than a few milliseconds over the course of a few minutes. If it does, you can consider your hardware broken (of course, cheap hardware is often broken).
As for #1, I'm not sure this is possible in any reliable sense with directsound. To the extent that it's possible with any audio API, it is difficult at best: both cards have streams that require some time to setup, open and start playing. In general, the solution is to use an API where this time is super low (ASIO, for example). This works reasonably well for applications like video, but I don't know if it really solves the problem in general.
If you really need to solve this problem, you could open both cards, starting to play silence, and use the timing information generated by the cards to establish the delay between putting data into the card and its eventual playback (this will be different for each card and probably each time you run) and use that data to calculate when to start actual playback. I don't know if RTAudio supplies the necessary timing information, but PortAudio does. This document may help.

How can I determine the length of time since the last screen refresh on X11?

I'm trying to debug a laggy machine vision camera by writing text timestamps to a terminal window and then observing how long it takes for the camera to 'detect' the screen change. My monitor has a 60hz refresh rate, so the screen is updated every ~17ms. Is there a way to determine at what point within that 17ms window the refresh timer currently is for an X11 application.
EDIT: After wrestling with the problem for nearly a day, I think the real question I should have asked was how to generate a visual signal that was sufficiently fast to test the camera images. My working hypothesis was that the camera was buffering frames before transmitting them, as the video stream seemed to lag behind other synchronised digital events (in this case, output signals to a robotic controller)
'xrefresh' is a tool which can trigger a refresh event on an X server. It does this by painting a global window of a specified color and then removing it, causing all subsequent windows to repaint. Even with this, I was still getting very inconsistent results when trying to correlate the captured frames against the monitor output, no matter what I tried to do, the video stream seemed to lag behind what I expected the monitor state to be. This could mean that either the camera was slow to capture or the monitor was slow to update. Fortunately, I eventually hit upon the idea of using the keyboard leds to verify the synchronicity of the camera frames. ('xset led' and 'xset -led'). This showed me immediately that in fact my computer monitor was slow to update, instead of the camera lagging behind.

low latency sounds on key presses

I am trying to write an application(I'm a gui first timer) for my son, he has autism. There is a video player in the top half and a text entry area in the bottom. When letters are typed sounds are produced to mimic the words in the video.
There have been other posts on this site in regard to playing sounds on key presses, using gstreamer as a system call. I have also tried libcanberra but both seem to have significant delays between sounds. I can write the app in python or C but will likely do at least some of it in C.
I also want to mention that the video portion is being played by gstreamer. I tried to create two instances of gstreamer, to avoid expensive system calls but the audio instance seemed to kill the app when called.
If anyone has any tips on creating faster responding sounds I would really appreciate it.
You can upload a raw audio sample directly to PulseAudio so there will be no decoding and (perhaps save) extra switches by using the following function from Canberra:
http://developer.gnome.org/libcanberra/unstable/libcanberra-canberra.html#ca-context-cache
The next ca_context_play() will use it.
However, the biggest problem you'll encounter with this scenario (with simultaneous video playback) is that the audio device might be configured with large latency with PulseAudio (up to 1/2s or more for normal playback). It may be reasonable to file a bug to libcanberra to support a LOW_LATENCY flag, as it currently doesn't attempt to minimize delay for sound events afaik. That would be great to have.
GStreamer pulsesink could probably get low latency too (it has some properties for that), but I am afraid it won't be as lightweight as libcanberra, and you won't be able to cache a sample for instance. Ideally, GStreamer could also learn to cache samples, or pre-fill PulseAudio...

Linux: Screen desktop video capture over network, and VNC framerate

Sorry for the wall of text - TL;DR:
What is the framerate of VNC connection (in frames/sec) - or rather, who determines it: client or server?
Any other suggestions for desktop screen capture - but "correctly timecoded"/ with unjittered framerate (with a stable period); and with possibility to obtain it as uncompressed (or lossless) image sequence?
Briefly - I have a typical problem that I am faced with: I sometimes develop hardware, and want to record a video that shows both commands entered on the PC ('desktop capture'), and responses of the hardware ('live video'). A chunk of an intro follows, before I get to the specific detail(s).
Intro/Context
My strategy, for now, is to use a video camera to record the process of hardware testing (as 'live' video) - and do a desktop capture at the same time. The video camera produces a 29.97 (30) FPS MPEG-2 .AVI video; and I want to get the desktop capture as an image sequence of PNGs at the same frame rate as the video. The idea, then, would be: if the frame rate of the two videos is the same; then I could simply
align the time of start of the desktop capture, with the matching point in the 'live' video
Set up a picture-in-picture, where a scaled down version of the desktop capture is put - as overlay - on top of the 'live' video
(where a portion of the screen on the 'live' video, serves as a visual sync source with the 'desktop capture' overlay)
Export a 'final' combined video, compressed appropriately for the Internet
In principle, I guess one could use a command line tool like ffmpeg for this process; however I would prefer to use a GUI for finding the alignment start point for the two videos.
Eventually, what I also want to achieve, is to preserve maximum quality when exporting the 'final' video: the 'live' video is already compressed when out of the camera, which means additional degradation when it passes through the Theora .ogv codec - which is why I'd like to keep the original videos, and use something like a command line to generate a 'final' video anew, if a different compression/resolution is required. This is also why I like to have the 'desktop capture' video as a PNG sequence (although I guess any uncompressed format would do): I take measures to 'adjust' the desktop, so there aren't many gradients, and lossless encoding (i.e. PNG) would be appropriate.
Desktop capture options
Well, there are many troubles in this process under Ubuntu Lucid, which I currently use (and you can read about some of my ordeals in 10.04: Video overlay/composite editing with Theora ogv - Ubuntu Forums). However, one of the crucial problems is the assumption, that the frame rate of the two incoming videos is equal - in reality, usually the desktop capture is of a lower framerate; and even worse, very often frames are out of sync.
This, then, requires the hassle of sitting in front of a video editor, and manually cutting and editing less-than-a-second clips on frame level - requiring hours of work for what will be in the end a 5 minute video. On the other hand, if the two videos ('live' and 'capture') did have the same framerate and sync: in principle, you wouldn't need more than a couple of minutes for finding the start sync point in a video editor - and the rest of the 'merged' video processing could be handled by a single command line. Which is why, in this post, I would like to focus on the desktop capture part.
As far as I can see, there are only few viable (as opposed to 5 Ways to Screencast Your Linux Desktop) alternatives for desktop capture in Linux / Ubuntu (note, I typically use a laptop as target for desktop capturing):
Have your target PC (laptop) clone the desktop on its VGA output; use a VGA-to-composite or VGA-to-S-video hardware to obtain a video signal from VGA; use video capture card on a different PC to grab video
Use recordMyDesktop on the target PC
Set up a VNC server (vino on Ubuntu; or vncserver) on the target PC to be captured; use VNC capture software (such as vncrec) on a different PC to grab/record the VNC stream (which can, subsequently, be converted to video).
Use ffmpeg with x11grab option
*(use some tool on the target PC, that would do a DMA transfer of a desktop image frame directly - from the graphics card frame buffer memory, to the network adapter memory)
Please note that the usefulness of the above approaches are limited by my context of use: the target PC that I want to capture, typically runs software (utilizing the tested hardware) that moves around massive ammounts of data; best you could say about describing such a system is "barely stable" :) I'd guess this is similar to problems gamers face, when wanting to obtain a video capture of a demanding game. And as soon as I start using something like recordMyDesktop, which also uses quite a bit of resources and wants to capture on the local hard disk - I immediately get severe kernel crashes (often with no vmcore generated).
So, in my context, I typically do assume involvement of a second computer - to run the capture and recording of the 'target' PC desktop. Other than that, the pros and cons I can see so far with the above options, are included below.
(Desktop preparation)
For all of the methods discussed below, I tend to "prepare" the desktop beforehand:
Remove desktop backgrounds and icons
Set the resolution down to 800x600 via System/Preferences/Monitors (gnome-desktop-properties)
Change color depth down to 16 bpp (using xdpyinfo | grep "of root" to check)
... in order to minimize the load on desktop capture software. Note that changing color depth on Ubuntu requires changes to xorg.conf; however, "No xorg.conf (is) found in /etc/X11 (Ubuntu 10.04)" - so you may need to run sudo Xorg -configure first.
In order to keep graphics resource use low, also I usually had compiz disabled - or rather, I'd have 'System/Preferences/Appearance/Visual Effects' set to "None". However, after I tried enabling compiz by setting 'Visual Effects' to "Normal" (which doesn't get saved), I can notice windows on the LCD screen are redrawn much faster; so I keep it like this, also for desktop capture. I find this a bit strange: how could more effects cause a faster screen refresh? It doesn't look like it's due to a proprietary driver (the card is "Intel Corporation N10 Family Integrated Graphics Controller", and no proprietary driver option is given by Ubuntu upon switch to compiz) - although, it could be that all the blurring and effects just cheat my eyes :) ).
Cloning VGA
Well, this is the most expencive option (as it requires additional purchase of not just one, but two pieces of hardware: VGA converter, and video capture card); and applicable mostly to laptops (which have both a screen + additional VGA output - for desktops one may also have to invest in an additional graphics card, or a VGA cloning hardware).
However, it is also the only option that requires no additional software of the target PC whatsoever (and thus uses 0% processing power of the target CPU) - AND also the only one that will give a video with a true, unjittered framerate of 30 fps (as it is performed by separate hardware - although, with the assumption that clock domains misalignment, present between individual hardware pieces, is negligible).
Actually, as I already own something like a capture card, I have already invested in a VGA converter - in expectation that it will eventually allow me to produce final "merged" videos with only 5 mins of looking for alignment point, and a single command line; but I am yet to see whether this process will work as intended. I'm also wandering how possible it will be to capture desktop as uncompressed video # 800x600, 30 fps.
recordMyDesktop
Well, if you run recordMyDesktop without any arguments - it starts first with capturing (what looks like) raw image data, in a folder like /tmp/rMD-session-7247; and after you press Ctrl-C to interrupt it, it will encode this raw image data into an .ogv. Obviously, grabbing large image data on the same hard disk as my test software (which also moves large ammounts of data), is usually a cause for an instacrash :)
Hence, what I tried doing is to setup Samba to share a drive on the network; then on the target PC, I'd connect to this drive - and instruct recordMyDesktop to use this network drive (via gvfs) as its temporary files location:
recordmydesktop --workdir /home/user/.gvfs/test\ on\ 192.168.1.100/capture/ --no-sound --quick-subsampling --fps 30 --overwrite -o capture.ogv
Note that, while this command will use the network location for temporary files (and thus makes it possible for recordMyDesktop to run in parallel with my software) - as soon as you hit Ctrl-C, it will start encoding and saving capture.ogv directly on the local hard drive of the target (though, at that point, I don't really care :) )
First of my nags with recordMyDesktop is that you cannot instruct it to keep the temporary files, and avoid encoding them, on end: you can use Ctrl+Alt+p for pause - or you can hit Ctrl-C quickly after the first one, to cause it to crash; which will then leave the temporary files (if you don't hit Ctrl-C quickly enough the second time, the program will "Cleanning up cache..."). You can then run, say:
recordmydesktop --rescue /home/user/.gvfs/test\ on\ 192.168.1.100/capture/rMD-session-7247/
... in order to convert the raw temporary data. However, more often than not, recordMyDesktop will itself segfault in the midst of performing this "rescue". Although, the reason why I want to keep the temp files, is to have the uncompressed source for the picture-in-picture montage. Note that the "--on-the-fly-encoding" will avoid using temp files altogether - at the expence of using more CPU processing power (which, for me, again is cause for crashes.)
Then, there is the framerate - obviously, you can set requested framerate using the '--fps N' option; however, that is no guarantee that you will actually obtain that framerate; for instance, I'd get:
recordmydesktop --fps 25
...
Saved 2983 frames in a total of 6023 requests
...
... for a capture with my test software running; which means that the actually achieved rate is more like 25*2983/6032 = 12.3632 fps!
Obviously, frames are dropped - and mostly that shows as video playback is too fast. However, if I lower the requested fps to 12 - then according to saved/total reports, I achieve something like 11 fps; and in this case, video playback doesn't look 'sped up'. And I still haven't tried aligning such a capture with a live video - so I have no idea if those frames that actually have been saved, also have an accurate timestamp.
VNC capture
The VNC capture, for me, consists of running a VNC server on the 'target' PC, and running vncrec (twibright edition) on the 'recorder' PC. As VNC server, I use vino, which is "System/Preferences/Remote Desktop (Preferences)". And apparently, even if vino configuration may not be the easiest thing to manage, vino as a server seems not too taxing to the 'target' PC; as I haven't experienced crashes when it runs in parallel with my test software.
On the other hand, when vncrec is capturing on the 'recorder' PC, it also raises a window showing you the 'target' desktop as it is seen in 'realtime'; when there are large updates (i.e. whole windows moving) on the 'target' - one can, quite visibly, see problems with the update/refresh rate on the 'recorder'. But, for only small updates (i.e. just a cursor moving on a static background), things seem OK.
This makes me wonder about one of my primary questions with this post - what is it, that sets the framerate in a VNC connection?
I haven't found a clear answer to this, but from bits and pieces of info (see refs below), I gather that:
The VNC server simply sends changes (screen changes + clicks etc) as fast as it can, when it receives them ; limited by the max network bandwidth that is available to the server
The VNC client receives those change events delayed and jittered by the network connection, and attempts to reconstruct the desktop "video" stream, again as fast as it can
... which means, one cannot state anything in terms of a stable, periodic frame rate (as in video).
As far as vncrec as a client goes, the end videos I get usually are declared as 10 fps, although frames can be rather displaced/jittered (which then requires the cutting in video editors). Note that the vncrec-twibright/README states: "The sample rate of the movie is 10 by default or overriden by VNCREC_MOVIE_FRAMERATE environment variable, or 10 if not specified."; however, the manpage also states "VNCREC_MOVIE_FRAMERATE - Specifies frame rate of the output movie. Has an effect only in -movie mode. Defaults to 10. Try 24 when your transcoder vomits from 10.". And if one looks into "vncrec/sockets.c" source, one can see:
void print_movie_frames_up_to_time(struct timeval tv)
{
static double framerate;
....
memcpy(out, bufoutptr, buffered);
if (appData.record)
{
writeLogHeader (); /* Writes the timestamp */
fwrite (bufoutptr, 1, buffered, vncLog);
}
... which shows that some timestamps are written - but whether those timestamps originate from the "original" 'target' PC, or the 'recorder' one, I cannot tell.
EDIT: thanks to the answer by #kanaka, I checked through vncrec/sockets.c again, and can see that it is the writeLogHeader function itself calling gettimeofday; so the timestamps it writes are local - that is, they originate from the 'recorder' PC (and hence, these timestamps do not accurately describe when the frames originated on the 'target' PC).
In any case, it still seems to me, that the server sends - and vncrec as client receives - whenever; and it is only in the process of encoding a video file from the raw capture afterwards, that some form of a frame rate is set/interpolated.
I'd also like to state that on my 'target' laptop, the wired network connection is broken; so the wireless is my only option to get access to the router and the local network - at far lower speed than the 100MB/s that the router could handle from wired connections. However, if the jitter in captured frames is caused by wrong timestamps due to load on the 'target' PC, I don't think good network bandwidth will help too much.
Finally, as far as VNC goes, there could be other alternatives to try - such as VNCast server (promising, but requires some time to build from source, and is in "early experimental version"); or MultiVNC (although, it just seems like a client/viewer, without options for recording).
ffmpeg with x11grab
Haven't played with this much, but, I've tried it in connection with netcat; this:
# 'target'
ffmpeg -f x11grab -b 8000k -r 30 -s 800x600 -i :0.0 -f rawvideo - | nc 192.168.1.100 5678
# 'recorder'
nc -l 0.0.0.0 5678 > raw.video #
... does capture a file, but ffplay cannot read the captured file properly; while:
# 'target'
ffmpeg -f x11grab -b 500k -r 30 -s 800x600 -i :0.0 -f yuv4mpegpipe -pix_fmt yuv444p - | nc 192.168.1.100 5678
# 'recorder'
nc -l 0.0.0.0 5678 | ffmpeg -i - /path/to/samplimg%03d.png
does produce .png images - but with compression artifacts (result of the compression involved with yuv4mpegpipe, I guess).
Thus, I'm not liking ffmpeg+x11grab too much currently - but maybe I simply don't know how to set it up for my needs.
*( graphics card -> DMA -> network )
I am, admittedly, not sure something like this exists - in fact, I would wager it doesn't :) And I'm no expert here, but I speculate:
if DMA memory transfer can be initiated from the graphics card (or its buffer that keeps the current desktop bitmap) as source, and the network adapter as destination - then in principle, it should be possible to obtain an uncompressed desktop capture with a correct (and decent) framerate. The point in using DMA transfer would be, of course, to relieve the processor from the task of copying the desktop image to the network interface (and thus, reduce the influence the capturing software can have on the processes running on the 'target' PC - especially those dealing with RAM or hard-disk).
A suggestion like this, of course, assumes that: there are massive ammounts of network bandwidth (for 800x600, 30 fps at least 800*600*3*30 = 43200000 bps = 42 MiB/s, which should be OK for local 100 MB/s networks); plenty of hard disk on the other PC that does the 'recording' - and finally, software that can afterwards read that raw data, and generate image sequences or videos based on it :)
The bandwidth and hard disk demands I could live with - as long as there is guarantee both for a stable framerate and uncompressed data; which is why I'd love to hear if something like this already exists.
-- -- -- -- --
Well, I guess that was it - as brief as I could put it :) Any suggestions for tools - or process(es), that can result with a desktop capture
in uncompressed format (ultimately convertible to uncompressed/lossless PNG image sequence), and
with a "correctly timecoded", stable framerate
..., that will ultimately lend itself to 'easy', single command-line processing for generating 'picture-in-picture' overlay videos - will be greatly appreciated!
Thanks in advance for any comments,
Cheers!
References
Experiences Producing a Screencast on Linux for CryptoTE - idlebox.net
The VideoLAN Forums • View topic - VNC Client input support (like screen://)
VNCServer throttles user inpt for slow client - Kyprianou, Mark - com.realvnc.vnc-list - MarkMail
Linux FAQ - X Windows: How do I Display and Control a Remote Desktop using VNC
How much bandwidth does VNC require? RealVNC - Frequently asked questions
x11vnc: a VNC server for real X displays
HowtoRecordVNC (an X11 session) - Debian Wiki
Alternative To gtk-RecordMyDesktop in Ubuntu
(Ffmpeg-user) How do I use pipes in ffmpeg
(ffmpeg-devel) (PATCH) Fix segfault in x11grab when drawing Cursor on Xservers that don't support the XFixes extension
You should get a badge for such a long well though out question. ;-)
In answer to your primary question, VNC uses the RFB protocol which is a remote frame buffer protocol (thus the acronym) not a streaming video protocol. The VNC client sends a FrameBufferUpdateRequest message to the server which contains a viewport region that the client is interested in and an incremental flag. If the incremental flag is not set then the server will respond with a FrameBufferUpdate message that contains the content of the region requested. If the incremental flag is set then the server may respond with a FrameBufferUpdate message that contains whatever parts of the region requested that have changed since the last time the client was sent that region.
The definition of how requests and updates interact is not crisply defined. The server won't necessarily respond to every request with an update if nothing has changed. If the server has multiple requests queued from the client it is also allowed to send a single update in response. In addition, the client really needs to be able to respond to an asynchronous update message from the server (not in response to a request) otherwise the client will fall out of sync (because RFB is not a framed protocol).
Often clients are simply implemented to send incremental update requests for the entire frame buffer viewport at a periodic interval and handle any server update messages as they arrive (i.e. no attempt is made to tie requests and updates together).
Here is a description of FrameBufferUpdateRequest messages.

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