I would like to create a web page where users can listen to various public safety radio streams in an area. A typical setup involves running a traditional radio scanner to make one audio stream.
My goal is to have a page with many different streams, each one can be selected "on" or "off", and they can play simultaneously.
The naive approach would just be to scale up the traditional audio stream method with one stream per radio channel, however I am concerned about the bandwidth demands and reliability. Users report their streams dropping out and having to reconnect.
My next idea, is to essentially buffer the streams through an FFMPEG instance and record them to disk while also cutting out the silence, then monitor the output of that with some code to "push" the new audio clips out to the listeners.
I don't have any code written yet, I am looking for suggestions on an overall approach to take.
My solution seems overly-complex to me, does anyone know of an audio codec or streaming solution that is well suited to audio with long periods of silence? Or is my idea the best way to do this? Can you think of any improvements?
The naive approach would just be to scale up the traditional audio stream method with one stream per radio channel
This is what I would recommend doing. It's simple, fairly low latency, has no special requirements on the server or the client.
however I am concerned about the bandwidth demands and reliability
Bandwidth for this is minimal. I recommend using Opus for the best quality for the bandwidth. Also, consider using VBR for the encoding. You'll end up with very low bandwidth when there is silence, with more bandwidth used while there's actual content. This is similar to what you were considering doing, but already built into the codec.
Related
I am beginner of video codec. not an video codec expert
I just want to know base on the same criteria, Comparing H254 encoding/decoding which is more efficiency.
Thanks
Decoding is more efficient. To be useful, decoding must run in real time, where encoding does not (except in videophone / conferencing applications).
How much more efficient? An encoder can generate motion vectors. The more compute power used on generating those motion vectors, the more accurate they are. And, the more accurate they are, the more bandwidth is available for the difference frames, so the quality goes up.
So, the kind of encoding used to generate video for streaming or distribution on DVD or BD discs can run many times slower than real time on server farms. But decoding for that kind of program is useless unless it runs in real time.
Even in the case of real-time encoding it takes more power (actual milliwatts, compute cycles, etc) than decoding.
It's true of H.264, H.265, VP8, VP9, and other video codecs.
I'm currently reading part 2 of the "A trip through the Graphics Pipeline" blog series by Fabian "ryg" Giesen.
In this particular part, he talks about one interesting point. The commands that the GPU works on can either be read from the video memory which would reside on the GPU, or from the CPU side via the PCI Express bus.
I implicitly understood a few pros and cons. However, I'm quite hazy on them. I cannot say that I perfectly understand the trade-offs. Anyway, I'm going to try to present my thoughts coherently.
Storing the Data in the Video Memory:
Definitely faster.
However, it has a memory overhead (not sure about bandwidth).
Reading from the CPU:
Could be better in situations where storing the data in the video memory is a waste (maybe because it would never be used again?).
I'd think that storing data in the video memory would be useful when that data is reused since it would save bandwidth on the PCIe bus.
But this communication will be slower than directly reading from the video memory.
I have a feeling that there are many more intricacies and trade-offs between the two. Also, I'm not entirely sure that what I've said above is true. I'd like for someone to:
Illustrate some important trade-offs and possibly other intricacies just to better my understanding.
Verify whether whatever I've said above is true or not! (I'm not quite clear on this yet.)
Anyone know how configure the HLS media server for reduce a little bit the delay of live streaming video?
what types of parameters i need to change?
I had heard that you could do some tuning using parameters like this: HLSMediaFileDuration
Thanks in advance
A Http Live Streaming system typically has an encoder which produces segments of a certain number of seconds and a media server (web server) which serves playlists containing a list of URLs to these segments to player applications.
Media Files = Segments = .ts files = MPEG2-TS files (in HLS speak)
There are some ways to reduce the delay in HLS:
Reduce the encoded segment length from Apple's recommended 10 seconds to 5 seconds or less. Reducing segment length increases network overhead and load on the web server.
Use lower bitrates, larger .ts files take longer to upload and download. If you use multi-bitrate streams, make sure the first bitrate listed in the playlist is a little lower than the bitrate most of your users use. This will reduce the time to start playing back the stream
Get the segments from the encoder to the web server faster. Upload while still encoding if possible. Update the playlist as soon as the segment has finished uploading
Also remember that the higher the delay the better the quality of your stream (low delay = lower quality). With larger segments, there is less overhead so more space for video data. Taking a longer time to encode results in better quality. More buffering results in less chance of video streams stuttering on playback.
HLS is all about trading quality of playback for longer delay, so you will never be able to use HLS for things like video conferencing. Typical delay in HLS is 30-60 sec, minimum in practice is around 15 sec. If you want low delay use RTP for streaming, but good luck getting good quality on low or variable speed networks.
Please specify which media server you use. Generally speaking, yes - changing chunk size will definitely affect delay time. The less is the first chunk, the quicker the video will be shown in the player.
Actually, Apple recommend to divide your file to small chunks this equal length of file, integers.
In practice, there is huge difference between players. Some of them parse manifest changing this values.
Known practice is to pre-cache in memory first chunks in low & medium resolution (Or try to download them in background of app/page - Amazon does this, though their video is MSS)
I was having the same problem and the keys for me were:
Lower the segment length. I set it to 2s because I'm streaming on a local network. In other type of networks, you need to be careful with the overhead that a low segment length adds that can impact your playback quality.
In your manifest, make sure the #EXT-X-TARGETDURATION is accurate. From here:
The EXT-X-TARGETDURATION tag specifies the maximum Media Segment
duration. The EXTINF duration of each Media Segment in the Playlist
file, when rounded to the nearest integer, MUST be less than or equal
to the target duration; longer segments can trigger playback stalls
or other errors. It applies to the entire Playlist file.
For some reason, the #EXT-X-TARGETDURATION in my manifest was set to 5 and I was seeing a 16-20s delay. After changing that value to 2, which is the correct one according to my segments' length, I am now seeing delays of 6-10s.
In summary, you should expect a delay of at least 3X your #EXT-X-TARGETDURATION. So, lowering the segment length and the #EXT-X-TARGETDURATION value should help reducing the delay.
I am making racing game in Libgdx.My game apk size is 9.92 mb and I am using four texture packer of total size is 9.92 Mb. My game is running on desktop but its run on android device very slow. What is reason behind it?
There are few loopholes which we neglect while programming.
Desktop processors are way more powerful so the game may run smoothly on Desktop but may slow on mobile Device.
Here are some key notes which you should follow for optimum game flow:
No I/O operations in render method.
Avoid creating Objects in Render Method.
Objects must be reused (for instance if your game have 1000 platforms but on current screen you can display only 3, than instead of making 1000 objects make 5 or 6 and reuse them). You can use Pool class provided by LibGdx for object pooling.
Try to load only those assets which are necessary to show on current screen.
Try to check your logcat if the Garbage collector is called. If so than try to use finalize method of object class to find which class object are collected as garbage and try to improve on it.
Good luck.
I've got some additional tips for improving performance:
Try to minimize texture bindings (or generally bindings when you're making a 3D game for example) in you render loop. Use texture atlases and try to use one texture after binding as often as possible, before binding another texture unit.
Don't display things that are not in the frustum/viewport. Calculate first if the drawn object can even be seen by the active camera or not. If it's not seen, just don't load it onto your GPU when rendering!
Don't use spritebatch.begin() or spritebatch.end() too often in the render loop, because every time you begin/end it, it's flushed and loaded onto the GPU for rendering its stuff.
Do NOT load assets while rendering, except you're doing it once in another thread.
The latest versions of libgdx also provide a GLProfiler where you can measure how many draw calls, texture bindings, vertices, etc. you have per frame. I'd strongly recommend this since there always can be situations where you would not expect an overhead of memory/computational usage.
Use libgdx Poolable (interface) objects and Pool for pooling objects and minimizing the time for object creation, since the creation of objects might cause tiny but noticable stutterings in your game-render loop
By the way, without any additional information, no one's going to give you a good or precise answer. If you think it's not worth it to write enough text or information for your question, why should it be worth it to answer it?
To really understand why your game is running slow you need to profile your application.
There are free tools avaiable for this.
On Desktop you can use VisualVM.
On Android you can use Android Monitor.
With profiling you will find excatly which methods are taking up the most time.
A likely cause of slowdowns is texture binding. Do you switch between different pages of packed textures often? Try to draw everything from one page before switching to another page.
The answer is likely a little more that just "Computer fast; phone slow". Rather, it's important to note that your computer Java VM is likely Oracles very nicely optimized JVM while your phone's Java VM is likely Dalvik, which, to say nothing else of its performance, does not have the same optimizations for object creation and management.
As others have said, libGDX provides a Pool class for just this reason. Take a look here: https://github.com/libgdx/libgdx/wiki/Memory-management
One very important thing in LibGDX is that you should make sure that sometimes loading assets from the memory cannot go in the render() method. Make sure that you are loading the assets in the right times and they are not coming in the render method.
Another very important thing is that try to calculate your math and make it independent of the render in the sense that your next frame should not wait for calculations to happen...!
These are the major 2 things i encountered when I was making the Snake game Tutorial.
Thanks,
Abhijeet.
One thing I have found, is that drawing is laggy. This means that if you are drawing offscreen items, then it uses a lot of useless resources. If you just check if they are onscreen before drawing, then your performance improves by a lot surprisingly.
Points to ponder (From personal experience)
DO NOT keep calling a function,in render method, that updates something like time,score on HUD (Make these updates only when required eg when score increases ONLY then update score etc)
Make calls IF specific (Make updations on certain condition, not all the time)
eg. Calling/updating in render method at 60FPS - means you update time 60 times a sec when it just needs to be updated once per sec )
These points will effect hugely on performance (thumbs up)
You need to check the your Image size of the game.If your image size are more than decrease the size of images by using the following link "http://tinypng.org/".
It will be help you.
I'm looking for an example of code that samples signal from microphone and
plays it on speakers. I need a solution that has a resonably constant delay on
different platforms (PC, android, iphone). Delay around 1-2 s is ok for me, and I don't
mind if it varies everytime the application starts.
I tried using SampleDataEvent.SAMPLE_DATA event on Sound and Microphpne classess.
One event would put data into buffer the other would read data.
But it seems impossible to provide constant delay, either the delay grows constantly or
it gets lower to the point where I have less than 2048 samples to put out and Sound class stops
generating SampleDataEvent.SAMPLE_DATA events.
I wan't to process each incoming frame so using setLoopBack(true) is not an option.
ps This is more a problem on Android devices than on PC. Althought when I start to resize application
window on PC delay starts to grow also.
Please help.
Unfortunately, this won't be possible... at least not directly.
Some sound devices will use a different clock between recording and playback. This will be especially true for cell phones where what is running the microphone may very well be different hardware than the headphone audio output.
Basically, if you record at 44.1kHz and play back at 44.1kHz, but those clocks are not in sync, you may be recording at 44.099kHz and play back at 44.101kHz. Over time, this drift will mean that you won't have enough data in the buffer to send to the output.
Another complication (and more than likely your problem) is that your record and playback sample rates may be different. If you record from the microphone at 11kHz and playback at 48kHz, you will note that 11 doesn't evenly fit into 48. Software is often used to up-sample the recording. Sometimes this is done with a nice algorithm which is guaranteed to give you the necessary output. Other times, that 11kHz will get pushed to 44kHz and is deemed "close enough".
In short, you cannot rely on recording and playback devices being in sync, and will need to synchronize yourself. There are many algorithms out there for handling this. The easiest method is to add a sample here and there that averages the sample before and after it. If you do this with just a few samples, it will be inaudible. Depending on the kind of drift problem you are having, this may be sufficient.