I know blackberry audio player has internal buffer which has to be filled before it starts playing. This causes about 2 seconds delay before the player starts.
Can i eliminate the delay and start playing audio as soon as possible.
Is there any way to reduce the internal buffer size.
If yes can any one tell me how to do it....
I am using blackberry os version 5.
Thanks and regards
Uttam
It turns out that the buffer size cannot be eliminated. This buffer is introduced somewhere in mid os 5 version. In previous versions you only have initial buffer you have to fill.
(Answer of the first related topic, try to use the search function next time;))
Related
I'm implementing an application which records and analyzes audio in real time (or at least as close to real time as possible), using the JDK Version 8 Update 201. While performing a test which simulates typical use cases of the application, I noticed that after several hours of recording audio continuously, a sudden delay of somewhere between one and two seconds was introduced. Up until this point there was no noticeable delay. It was only after this critical point of recording for several hours when this delay started to occur.
What I've tried so far
To check if my code for timing the recording of the audio samples is wrong, I commented out everything related to timing. This left me essentially with this update loop which fetches audio samples as soon as they are ready (Note: Kotlin code):
while (!isInterrupted) {
val audioData = read(sampleSize, false)
listener.audioFrameCaptured(audioData)
}
This is my read method:
fun read(samples: Int, buffered: Boolean = true): AudioData {
//Allocate a byte array in which the read audio samples will be stored.
val bytesToRead = samples * format.frameSize
val data = ByteArray(bytesToRead)
//Calculate the maximum amount of bytes to read during each iteration.
val bufferSize = (line.bufferSize / BUFFER_SIZE_DIVIDEND / format.frameSize).roundToInt() * format.frameSize
val maxBytesPerCycle = if (buffered) bufferSize else bytesToRead
//Read the audio data in one or multiple iterations.
var bytesRead = 0
while (bytesRead < bytesToRead) {
bytesRead += (line as TargetDataLine).read(data, bytesRead, min(maxBytesPerCycle, bytesToRead - bytesRead))
}
return AudioData(data, format)
}
However, even without any timing from my side the problem was not resolved. Therefore, I went on to experiment a bit and let the application run using different audio formats, which lead to very confusing results (I'm going to use a PCM signed 16 bit stereo audio format with little endian and a sample rate of 44100.0 Hz as default, unless specified otherwise):
The critical amount of time that has to pass before the delay appears seems to be different depending on the machine used. On my Windows 10 desktop PC it is somewhere between 6.5 and 7 hours. On my laptop (also using Windows 10) however, it is somewhere between 4 and 5 hours for the same audio format.
The amount of audio channels used seems to have an effect. If I change the amount of channels from stereo to mono, the time before the delay starts to appear is doubled to somewhere between 13 and 13.5 hours on my desktop.
Decreasing the sample size from 16 bits to 8 bits also results in a doubling of the time before the delay starts to appear. Somewhere between 13 and 13.5 hours on my desktop.
Changing the byte order from little endian to big endian has no effect.
Switching from stereomix to a physical microphone has no effect either.
I tried opening the line using different buffer sizes (1024, 2048 and 3072 sample frames) as well as its default buffer size. This also didn't change anything.
Flushing the TargetDataLine after the delay has started to occur results in all bytes being zero for approximately one to two seconds. After this I get non-zero values again. The delay, however, is still there. If I flush the line before the critical point, I don't get those zero-bytes.
Stopping and restarting the TargetDataLine after the delay appeared also does not change anything.
Closing and reopening the TargetDataLine, however, does get rid of the delay until it reappears after several hours from there on.
Automatically flushing the TargetDataLines internal buffer every ten minutes does not help to resolve the issue. Therefore, a buffer overflow in the internal buffer does not seem to be the cause.
Using a parallel garbage collector to avoid application freezes also does not help.
The used sample rate seems to be important. If I double the sample rate to 88200 Hertz, the delay starts occurring somewhere between 3 and 3.5 hours of runtime.
If I let it run under Linux using my "default" audio format, it still runs fine after about 9 hours of runtime.
Conclusions that I've drawn:
These results let me come to the conclusion that the time for which I can record audio before this issue starts to happen is dependent on the machine on which the application is run and dependent on the byte rate (i.e. frame size and sample rate) of the audio format. This seems to hold true (although I can't completely confirm this as of now) because if I combine the changes made in 2 and 3, I would assume that I can record audio samples for four times as long (which would be somewhere between 26 and 27 hours) as when using my "default" audio format before the delay starts to appear. As I didn't find the time to let the application run for this long yet, I can only tell that it did run fine for about 15 hours before I had to stop it due to time constraints on my side. So, this hypothesis is still to be confirmed or denied.
According to the result of bullet point 13, it seems like the whole issue only appears when using Windows. Therefore, I think that it might be a bug in the platform specific parts of the javax.sound.sampled API.
Even though I think I might have found a way to change when this issue starts to happen, I'm not satisfied with the result. I could periodically close and reopen the line to avoid the problem from starting to appear at all. However, doing this would result in some arbitrary small amount of time where I wouldn't be able to capture audio samples. Furthermore, the Javadoc states that some lines can't be reopened at all after being closed. Therefore, this is not a good solution in my case.
Ideally, this whole issue shouldn't be happening at all. Is there something I am completely missing or am I experiencing limitations of what is possible with the javax.sound.sampled API? How can I get rid of this issue at all?
Edit: By suggestion of Xtreme Biker and gidds I created a small example application. You can find it inside this Github repository.
I have (a rather) vast experience with Java audio interfacing.
Here are a few points that may be useful in guiding you towards a proper solution:
It's not a matter of JVM version - the java audio system have barely been upgraded since Java 1.3 or 1.5
The java audio system is a poor-man's wrapper around whatever audio interface API the operating system has to offer. In linux it's the Pulseaudio library, For windows, it's the direct show audio API (if I'm not mistaken about the latter).
Again, the audio system API is kind of a legacy API - some of the features are not working or not implemented, other behaviors are straight out weird, as they are dependent on an obsolete design (I can provide examples, if required).
It's not a matter of Garbage Collection - If your definition of "delay" is what I understand it to be (audio data is delayed by 1-2 seconds, meaning you start hearing stuff 1-2 seconds later), well, the garbage collector cannot cause blank data to magically be captured by the target data line and then append data as usual in an 2 seconds worth byte offset.
What's most likely happening here is either the hardware or driver providing you with 2 seconds worth of garbled data at some point, and then, streams the rest of the data as usual, resulting in the "delay" you are experiencing.
The fact that it works perfectly on linux means it's not a hardware issue, but rather a driver related one.
To affirm that suspicion, you can try capturing audio via FFmpeg for the same duration and see if the issue is reproduced.
If you are using specialized audio capturing hardware, better approach your hardware manufacturer and inquire him about the issue you are facing on windows.
In any case, when writing an audio capturing application from scratch I'd strongly suggest keeping away from the Java audio-system if possible. It's nice for POCs, but it's an un-maintained legacy API. JNA is always a viable option (I've used it in Linux with ALSA/Pulse-audio to control audio hardware attributes the Java audio system could not alter), so you could look for audio capturing examples in C++ for windows and translate them to Java. It'll give you fine grain control over audio capture devices, much more than what the JVM provide OOTB. If you want to have a look at a living/breathing usable JNA example, check out my JNA AAC encoder project.
Again, if you use special capturing harwdare, there's a good chance the manufacturer already provides it's own low-level C api for interfacing with the hardware, and you should consider having a look at it as well.
If that's not the case, maybe you and your company/client should
consider using specialized capturing hardware (doesn't have to be
that expensive).
Clip clip = AudioSystem.getClip();
AudioInputStream a = AudioSystem.getAudioInputStream(new File(path));
clip.open(a);
This is the code that I'm using to play audio in my program. From Java profiling I can see that on average the clip.open() call takes less than 1ms. However, occasionally at random times it will block for a couple of seconds causing lag.
The screenshot below shows my Java profiler. As you can see, the exact same method is called 316 times with no issue. But one time it hangs for 2.4 seconds on Clip.open()
Notice how Clip.open doesn't even show in the bottom one because the time spent is less than 0.1ms.
The clips that I'm playing are all around 100KB in size, I don't understand why it works fine 316 calls but then one time it hangs.
I've also tried not closing the clips but leaving them all open, even then the problem still occurs.
Usually programmers .open() a Clip well in advance of when they want to play it. The moment of playback should only involve a .play() command and nothing else. If you both "open" and "play" the clip in consecutive commands, the play() can be delayed considerably because the file has to be loaded into memory in its entirety before the play() command will execute. For this reason, if you can't afford the memory for a Clip, then a SourceDataLine will execute more quickly as it only needs to load a buffer's worth into memory before the play() will execute.
Perhaps you already know about that aspect of Clips and that wasn't the issue. (You mention playing the Clips without having closed them.) Well another fact of Java is that there are no real-time guarantees. The system does do a good job of keeping a file or clip playing, but controlling the exact starting point is hard. This is because of several factors, one of which is the time spent juggling multiple threads and processes. For example, if a garbage collection command gets the call right before your sound call, the sound will have to wait until that segment is done and the processor gives the sound thread the priority.
There are other factors impacting real-time performance as well, which are well laid out in the following paper: Real Time Low Latency Processing in Java
Depending on what your goal is, there are ways to take advantage of the sound thread to improve timing accuracy, by "counting sound frames" and running events when a specific sound frame is up for processing, from the operation that is doing that processing. But in general, communications between the audio and other threads is going to be subject to some jitter.
I'm building a client-server android application. One app is installed in Google Glass and sends video frames captured by the camera and sent via Bluetooth connection. Another application is installed in an Android device and reads the video frames. When reading from the stream len = mmInStream.read(buffer, 0, bufferSize); it seems like the maximum byte count read is 990. It doesn't matter if I set bufferSize to an extremely large number.
I'm sending a 320x240 image frame with 4 channels. So this is a total
of 307200 bytes. So when reading the entire image, it is being read
in chunks of 990 bytes which I think affects the speed of my app. It
takes 1-3 seconds to read all data. Is there a way to change the
maximum bytes read? Is this an application setting or controlled by
the Android OS? I'm not sure if reading all data at once would affect
performance but I am just curious.
UPDATE:
I notice the same thing when sending the data from Google Glass using OutputStream. It takes about 2 seconds to write to the OutputStream. Is this normal performance for Bluetooth connection? Is there a better way to transmit camera capture frames between 2 devices?
UPDATE 2:
I think the delay is in the write speed. Writing the data to the stream takes about 2 seconds. When the other app is trying to read the data, it probably waits for the complete data to be written to the stream. Still not sure if this is as expected or can still be improved.
I'm making a game in Java. I want for there to be about 100 different samples and at any given time, 10 samples could be playing. However, for each of these 10 samples, I want to be able to manipulate their volume and pan.
As of right now, I request a line as follows: new DataLine.Info(Clip.class, format);
I do not specify the controls that I need for this line, but it appears that Clips always have MASTER_GAIN and BALANCE controls.
Is this correct?
Could I just create an array of 100 clips and preload all of the samples? I don't quite understand if Java's lines correspond with physical lines into a physical mixer or if they are virtualized.
If I am limited, then how can I swap samples in and out of lines? Is there a way to do this so that all of my say 100 samples are preloaded? Or, does preloading only help when you already have a line designated?
Again, if I am limited, is this the wrong approach? Should I either:
a. use a different programming language, and/or
b. combine audio streams manually and put them all through the same line.
Wow, that's a lot of questions. I didn't find answers in the documentation and I really hope that you guys can help. Please number your answers 1 to 4. Thank you very much!
1) I do NOT think it is safe to assume there will always be a BALANCE or even a MASTER_GAIN. Maybe there is. My experience with Java Controls for audio was vexing and short. I quickly decided to write my own mixer, and have done so. I'm willing to share this code. It includes basic provisions for handling volume and panning.
Even when they work, the Java Controls have a granularity that is limited by the buffer size being used, and this severely limits how fast you can fade in or out without creating clicks, if you trying to do fades. Setting and holding a single volume is no problem, though.
Another java library (bare bones but vetted by several game programmers at java-gaming.org) is "TinySound" which is available via github. I've looked it over but not used it myself. It also mixes all sounds down to a single output SourceDataLine. I can't recall how volume or panning is handled. He included provisions for ogg/vorbis files.
2) I'm not sure how you are envisioning using Clips work when you mention "Samples". Yes, you can preload an array of 100 Clips. And you would directly play one or another of these Clips on it's own thread (assuming using raw Java instead of an audio-mixing library), then reset them back to frame 0, then play them again. But you can only have one thread playing a given Clip at a time: they do not accommodate concurrent playback. (You can "retrigger" though by stopping a given playback and moving the position back to frame #0 then replaying.)
How long are the Clips? 100 of them could be a LOT of memory. If each is a second long, 100 seconds * 44100 frames per second * 4 bytes per frame = 17,640,000 bytes (almost 18MB just dedicated to RAM for sound!).
I guess, if you know you'll only need a few at a time and you can predict which ones will be needed, you can pre-load those and reuse them. But don't fall into the trap of thinking that Clips are meant to be loaded at the time of playback. If you are doing that, you should use SourceDataLines instead. They start playing back quicker since they don't have to wait until the entire external file has been put into memory (as Clips do). I'd recommend only using a Clip if you plan to reset it to the 0th frame and replay it (or loop it)!
3) Once it is loaded as a Clip, it is basically ready to go, there really isn't an additional stage. There really isn't any intermediate stage between an external file and a Clip in memory that I can think of that might be helpful.
Ah, another thought: You might want to create a thread pool ( = max number of concurrent sounds) and manage that. I don't know at what point the scaling justifies the extra management.
4) It IS possible to run concurrent SourceDataLines in many contexts, which relieves the need for holding the entire file in RAM. In that case, the only thing you can preload are the Strings for the File locations, I think. I may be wrong and you can preload the Files as well, but maybe not. Definitely can't reuse an AudioInputLine! On the plus side, an SDL kicks off pretty quick compared to an UNLOADED Clip.
HOWEVER! There are systems (e.g., some Linux OS) that limit you to a single output, which might be either a Clip or a SourceDataLine. That was the clincher for me when I decided to build my own mixer.
I think if only 8 or 10 tones are playing at one time, you will probably be okay as long as the graphics are not too ambitious (not counting the above mentioned Linux OS situation). You'll have to test it.
I don't know what alternative languages you are considering. Some flavor of C is the only alternative I know of. Most everything else that I know of, except Java, is not low-level or fast enough to handle that much audio processing. But I am only modestly experienced, and do not have a sound engineering background but am self-taught.
I want to play a certain parts of a wav file. Like playing the first ten seconds and then playing it from 50th-60th seconds and so on. I know how to play a entire wave file in Java using the start method of SourceDataLine class. Could anybody give me some pointers as to how I can seek a particular time position for audio and play it?
Find the length of a frame, in bytes, from the AudioFormat
Find the length in bytes of a second, by multiplying the frame size by the frame rate.
skip() that amount of bytes.
Play until the 2nd number of bytes calculated using the same formula.
As far as I can see, nothing happens when you just call start. You are responsible for pushing the bytes of your choice into the line. So open a RandomAccessFile, seek to the appropriate offset, and execute a loop that transports the file data to the SourceDataLine.