Is it possible to record voice and stream in J2ME,like I record and use commit() to get the byte array,but the commit() takes at least 250ms ,even the record length is 10 ms,and it irrationally takes 280ms if the record length is 10s etc,
the device I tested is Nokia 6300 s40 device.
How can I prevent the gap?
Actually I need to record voice in a packets of the time intervals as small as possible like 100ms,200ms,etc. but each time the commit() takes at least 250ms.
AFAIK its not possible. Because you have to commit() after recording the audio. So it should takes sometime to commit().
Related
I am trying to build an Android app that interfaces with the ESP32 using BLE. I am using the RxBluetoothKotlin library from Vincent Masselis for the Android side. For the ESP32 side, I am using the default Kolban libraries that are included in the Arduino IDE. My phone is a OnePlus 5T and my ESP32 is a MH ET Live ESP32DevKIT. My Android app can be found here, and my ESP32 program here.
The whole system works pretty much perfectly for me in terms of pure functionality. That is to say, every button does what it's supposed to do, and I get the exact behaviour I had expected to get. However, the communication itself is very slow. Around 200 bytes/second. My test button in the Android app requests a bunch of text data from the ESP32, and displays this in a dialog. It also lists a number which represents the time between request and reception in milliseconds. Using this, I get around 2 seconds for 440 bytes of data. When I send less data, the time decreases approximately linearly with data size. 40 bytes of data will take around 200ms, and 20 bytes or under typically takes less than 100ms.
This seems rather slow to me. From what I understand, I should be able to at least get a few kilobytes per second. I have tried to check the speed using nRF Connect, but I get the same 2 seconds timespan for my data transfer. This suggests that the problem is not in my app, since I also have it with a completely different app. I also put the code in my main loop inside of callbacks instead (which I probably should have done in the first place), but this didn't change things at all. I have tried taking the microcontroller and my phone to a few different locations, hoping to eliminate interference. I have tried to mess with BLEDevice::setPower and BLEDevice::setMTU, as well as setting RxBluetoothGatt.requestMtu(500) on the Android side. Everything so far seems to have had little to no effect. The only thing that did anything, was adding the line "pServer->updatePeerMTU(0,500);" in my loop during the connection phase. This caused the first 23 bytes of data to be repeated whenever I pressed the test button in my app, and made the data transfer take about 3 seconds. If I'm lucky, I can get maybe a bit under 1.8 seconds for 440 bytes, but this is a very small change when I'm expecting an order of magnitude of difference, and might even be down to pure chance rather than anything I did.
Does anyone have an idea of how to increase my transfer speed?
The data transmission speed is mainly influenced by the Bluetooth LE connection interval (between 7.5 ms and 4 seconds) and is negotiated between the master (central unit) and the peripheral device. The master establishes a connection with a parameter set and the peripheral can propose to change this parameter set. In the end, however, the central unit decides which parameter set is to be used.
But the Bluetooth connection interval cannot be changed by an Android applications directly, which normally act as the central role. Instead it can request a connection priority which is known to have an influence on the connection interval.
I want to built a SoundWave sampling an audio stream.
I read that a good method is to get amplitude of the audio stream and represent it with a Polygon. But, suppose we have and AudioGraph with just a DeviceInputNode and a FileOutpuNode (a simple recorder).
How can I get the amplitude from a node of the AudioGraph?
What is the best way to periodize this sampling? Is a DispatcherTimer good enough?
Any help will be appreciated.
First, everything you care about is kind of here:
uwp AudioGraph audio processing
But since you have a different starting point, I'll explain some more core things.
An AudioGraph node is already periodized for you -- it's generally how audio works. I think Win10 defaults to periods of 10ms and/or 20ms, but this can be set (theoretically) via the AudioGraphSettings.DesiredSamplesPerQuantum setting, with the AudioGraphSettings.QuantumSizeSelectionMode = QuantumSizeSelectionMode.ClosestToDesired; I believe the success of this functionality actually depends on your audio hardware and not the OS specifically. My PC can only do 480 and 960. This number is how many samples of the audio signal to accumulate per channel (mono is one channel, stereo is two channels, etc...), and this number will also set the callback timing as a by-product.
Win10 and most devices default to 48000Hz sample rate, which means they are measuring/output data that many times per second. So with my QuantumSize of 480 for every frame of audio, i am getting 48000/480 or 100 frames every second, which means i'm getting them every 10 milliseconds by default. If you set your quantum to 960 samples per frame, you would get 50 frames every second, or a frame every 20ms.
To get a callback into that frame of audio every quantum, you need to register an event into the AudioGraph.QuantumProcessed handler. You can directly reference the link above for how to do that.
So by default, a frame of data is stored in an array of 480 floats from [-1,+1]. And to get the amplitude, you just average the absolute value of this data.
This part, including handling multiple channels of audio, is explained more thoroughly in my other post.
Have fun!
I would like to track a large number of beacons (~500) at once within a 50-100 m radius via an app on an iPhone (5s). I've had a look at the spec and online and I can't see if there is any limit on the number of beacons you can track at once using BLE. Does anyone know if there is limitation on the number of beacons you can track exists or if an iPhone 5s would be up to the task of tracking that many beacons?
You used the word track, but iOS has two different methods: monitoring and ranging.
You can set a maximum of 20 regions to monitor. (Found in documentation for the startMonitoringForRegion: method.) Region limits mostly come into play if your app is in the background. The OS will alert your app when you enter or leave a region that you're monitoring (give or take a few minutes). The OS will even launch your app just to let it know what happened (although only for a short time).
The other method is ranging, which is to find all the beacons within the Bluetooth range of the device (typically around 100 feet give or take). If your beacons are spread out over 100 miles, then you probably won't run into any practical limit here. I have not found any documentation for this, and I have only four beacons that I'm testing with, and four at a time works.
Here's one way to handle your situation. Make all your 500 beacons use the same UUID, and make a beacon region using initWithProximityUUID:identifier: method. (Identifier is just for you -- it doesn't affect anything). Starting monitoring for that beacon region. That way, your app will be notified whenever one of your 500 beacons are found (give or take a few minutes). Once notified, you can use startRangingBeaconsInRegion: to find all the beacons around that area, then use the major and minor values to figure out which beacons the user is near.
I'll add to Tim Tisdall's answer, which sets out the right framework. I can't speak to the specific capabilities of the iPhone 5s, or iOS in general, but I don't see any reason why it wouldn't return every ADV_IND packet (i.e. beacon transmission) that it receives.
The question is, will the 500 beacons be able to transmit their ADV_IND packets without collisions?
It takes about 0.128ms to transmit an ADV_IND packet. The time between advertising transmissions is configurable between 20ms and 10240ms (at intervals of 0.625ms), so the probability of collisions depends on the configuration of the beacons.
Based on the Poisson distribution, the probability of a collision for any given ADV_IND packet is 1-exp(-2*N*(0.128/AI)), where N is the number of beacons within range, AI is the time in milliseconds of the advertising interval (assuming all the beacons are configured the same), and the 0.128 is the time in milliseconds it takes to send the ADV_IND packet. (See http://www3.cs.stonybrook.edu/~jgao/CSE590-fall09/aloha-analysis.pdf if you want an explanation.)
For 500 beacons with the maximum advertising interval of about 10 seconds, there will be a collision about once every 81 packets (or about 6 out of 500). If you're willing to wait for a couple intervals (i.e. 30 seconds), there's a good chance you'll be able to receive all 500 ADV_IND packets.
On the other hand, if the advertising interval is smaller, say 500ms, you'll have a collision about 23% of the time (or 113 out of 500). You'd have to wait for several more intervals to improve the probability that you'd see the broadcasts from all the beacons.
The other way to look at it is that the more beacons you have, the longer you have to wait to make sure you receive all their packets. (The math to calculate the delay to receive the packets with a certain probability from the number of beacons and the advertising interval is too much for me today.)
One caveat: if you want to connect to these beacons, as opposed to just receiving the ADV_IND packet, that requires an exchange of two more packets on the advertising channels, and the probability of a collision in the advertising channels goes up a bit.
If I am reading your question right, you want to put all 500 iBeacons within 100 meters of each other, meaning their transmissions will overlap. You will probably run into radio congestion problems long before you run into any limitations of iOS7 or your phone.
I have successfully tested 20 iBeacons in close proximity without problems, but 500 iBeacons is an extreme density. this discussion on the hardware issue suggests you may run into trouble.
At a minimum, the collisions of the transmissions of 500 iBEacons will make it take longer for your iOS device to see each iBeacon. Normally, iOS7 provides a ranging update once per second for each iOS device, but you may find that you get updates much less often. It all depends on your application whether or not less frequent updates are acceptable.
Even if delays are acceptable, I would absolutely test this before counting on it working at all. Unfortunately, that means getting your hands on lots of iBeacons.
I don't agree. It is true that ble beacons only transmit advertising data, but the transmission of such data last about 3ms (considering three advertising channels).
Having 500 beacons, WITHOUT considering any collision, the scanner will takes 1.5s to see them all.
But, if all beacons are configured in same way (same advertising interval) it is inevitable to have collisions which lead to have undiscovered beacons. Even if the advertising interval is different between beacons collisions occur. To avoid collision probability one should use longer advertising interval, but this lead to longer discovery latency.
This reasoning is very raw, it doesn't take care of many effects, but is just an order of magnitude calculation.
By the way, the question is not easy, there are many parameters which play role, some are known some are unknown. But I'm working with ble since one year about and, to me, 500 is a huge number and there is the possibility that you don't see the majority of nodes because of collisions.
I was doing some research into iBeacon's because of this question (I had no idea what it was about).
It seems that on the "beacon" side of things all that happens is general advertising packets are sent out. It's similar to how a device advertises that you can connect to it. However, you don't actually connect to iBeacon's, it just reads those advertising packets. There's no built-in limitation on how many advertising packets a device can receive.
So, it wouldn't surprise me if 500 iBeacon's would run with no issues. The advertising packets are small and are spaced out (time wise, they are repeated every X ms). There's no communication going from the phone to the iBeacon, the phone is simply receiving the packets it hears. If there's interference on one packet it'll likely manage to get the next one.
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.
I am trying to record what it is just playing out to the speaker using following ALSA APIs:
snd_pcm_mmap_writei()
snd_pcm_mmap_readi()
Both functions are called one to next in the same thread. The writei() function returns quickly (I believe it returns once playback buffer is available), while the readi() returns until designated samples are captured. But the samples captured are not what is has just played out. I am guessing that ALSA is not in a duplex mode, i.e., it has to finish playback first, then start to record, which records nothing meaningful but just clicks. The speaker still plays out the sound correctly.
All HW/SW parameters are setup correctly. If I do audio capture only, I will get a good sound wave.
The PCM handles are opened with normal mode (not non-block, not async).
Anybody has suggestions how to make this work?
You do not need to use the mmap functions; the normal writei/readi calls suffice.
To handle two PCM streams at the same time, run them in separate threads, or use non-blocking mode so that the same event loop can handle both devices.
You need to fill the playback buffer before the data is played, and capture data can be read only after the capture buffer has been filled, so the overall latency is the playback buffer size plus the capture period size plus any hardware delays and sound propagation delays.