I'm developing on the Samsung Instinct, so I'm not certain that this question is relevant across other J2ME platforms. On the Instinct, when I start recording with a RecordControl, the receiving output stream remains empty until I call commit(). Is there a way to purge the so-far recorded data from a RecordControl to the output stream without calling commit()?
Call reset()
http://java.sun.com/javame/reference/apis/jsr135/javax/microedition/media/control/RecordControl.html#reset()
Related
I want to create a speech jammer. It is essentially something that repeats back to you what you just said, but it is continuous. I was trying to use the sounddevice library and record what I am saying while also playing it back. Then I changed it to originally record what I was saying, then play it back while also recording something new. However it is not functioning as I would like it. Any suggestions for other libraries? Or if someone sees a suggestion for the code I already have.
Instead of constantly playing back to me, it is starting and stopping. It does this at intervals of the duration specified. So it will record for 500 ms, then play that back for 500 ms and then start recording again. Wanted behavior would be - recording for 500ms while playing back the audio it is recording at some ms delay.
import sounddevice as sd
import numpy as np
fs = 44100
sd.default.samplerate = fs
sd.default.channels = 2
#the above is to avoid having to specify arguments in every function call
duration = .5
myarray = sd.rec(int(duration*fs))
while(True):
sd.wait()
myarray = sd.playrec(myarray)
sd.wait()
Paraphrasing my own answer from https://stackoverflow.com/a/54569667:
The functions sd.play(), sd.rec() and sd.playrec() are not meant to be used repeatedly in rapid succession. Internally, they each time create an sd.OutputStream, sd.InputStream or sd.Stream (respectively), play/record the audio data and close the stream again. Because of opening and closing the stream, gaps will occur. This is expected.
For continuous playback you can use the so-called "blocking mode" by creating a single stream and calling the read() and/or write() methods on it.
Or, what I normally prefer, you can use the so-called "non-blocking mode" by creating a custom "callback" function and passing it to the stream on creation.
In this callback function, you can e.g. write the input data to a queue.Queue and read the output data from the same queue. By filling the queue by a certain amount of zeros beforehand, you can specify how long the delay between input and output shall be.
You can have a look at the examples to see how callback functions and queues are used.
Let me know if you need more help, then I can try to come up with a concrete code example.
I'm seeing a potential problem here of you trying to use myarray as both the input and the output of the .playrec() function. I would recommend having two arrays, one for recording the live audio, and one for playing back the recorded audio.
Instead of using the .playrec() command, you could just rapidly alternate between the use of .record() and .play() with a small delay between within your while-loop.
For example, the following code should record for one millisecond, wait a millisecond, and then playback the one millisecond of audio:
duration = 0.001
while(True):
myarray= sd.rec(int(duration*fs))
sd.wait()
sd.play(myarray, (int(duration*fs)))
There is no millisecond delay after the playback because you want to go right back to recording the next millisecond straight away. It should be noted, however, that this does not keep a recording of your audio for more than one millisecond! You would have to add your own code that adds to the array of a specified size and fills it up over time.
In my application i am running a timer in background for every 8 seconds to play a custom sound,it works fine ,but it get stops at later sometime,so how can i play the sound continuously in background?
Currently i am using the below code to play the sound in background
SystemSoundID soundID;
AudioServicesCreateSystemSoundID((__bridge CFURLRef)filePath, &soundID);
AudioServicesPlaySystemSound(soundID);
let me know the good solution to play the sound continuously in background
Short answer: your app is simply suspended.
Long answer: You are missing key parts of the background savvy implementation.
You need to tell the iOS that you are an Audio App, and that you are requesting extra cycles when suspended.
The UIBackgroundModes is subject to approval
From the documentation:
Background modes for apps:
UIBackgroundModes value = audio
The app plays audible content to the user or records audio while in the background. (This content includes streaming audio or video content using AirPlay.)
If your app does not fall into any of the categories below, then your only option to extend backgrounding beyond the typical 5 seconds is to invoke -beginBackgroundTaskWithName:expirationHandler:. You will likely be suspend within 30 seconds or so.
audio
location
voip
newsstand-content
external-accessory
bluetooth-central & bluetooth-peripheral
fetch
remote-notification
When I call for the first time to Manager.CreatePlayer() its freezes my application for a split second and it's a problem for me because I'm writing a game and it's noticeable, what can I do to fix it ?
As far as I know, the common logics for game are :
Display loading screen
Here's all heavy operations are prepared/preloaded and cached, so the game can run smoothly later.
Methods that usually called here are Manager.createPlayer and Player.prefetch().
All the image & sound is prepared first, and can be used quickly when game started.
Start the game (loop)
As the resource have been prepared/preloaded, now you can use (draw/play) them here.
Use the Player instances that has been created & prefetched (from loading screen).
You can call Player.start() method here to play the sound.
You can read about the Player state (especially about prefetch) HERE.
Notice that you can reuse the Player instance and call start() method multiple times for playing the same sound. No need to call createPlayer again.
How do you go about stopping the playback of a video? I only see a method for pausing.
Can I maybe force the load of a null video or something like that?
There will be a version of CastCompanionLibrary that will be published soon which exposes stop and some more new features. In most cases, you don't really need to call stop since that unloads the video as well; you can simply pause. Are you dealing with live streams? What is your use case for calling stop?
I have built a standalone app version of a project that until now was just a VST/audiounit. I am providing audio support via rtaudio.
I would like to add MIDI support using rtmidi but it's not clear to me how to synchronise the audio and MIDI parts.
In VST/audiounit land, I am used to MIDI events that have a timestamp indicating their offset in samples from the start of the audio block.
rtmidi provides a delta time in seconds since the previous event, but I am not sure how I should grab those events and how I can work out their time in relation to the current sample in the audio thread.
How do plugin hosts do this?
I can understand how events can be sample accurate on playback, but it's not clear how they could be sample accurate when using realtime input.
rtaudio gives me a callback function. I will run at a low block size (32 samples). I guess I will pass a pointer to an rtmidi instance as the userdata part of the callback and then call midiin->getMessage( &message ); inside the audio callback, but I am not sure if this is thread-sensible.
Many thanks for any tips you can give me
In your case, you don't need to worry about it. Your program should send the MIDI events to the plugin with a timestamp of zero as soon as they arrive. I think you have perhaps misunderstood the idea behind what it means to be "sample accurate".
As #Brad noted in his comment to your question, MIDI is indeed very slow. But that's only part of the problem... when you are working in a block-based environment, incoming MIDI events cannot be processed by the plugin until the start of a block. When computers were slower and block sizes of 512 (or god forbid, >1024) were common, this introduced a non-trivial amount of latency which results in the arrangement not sounding as "tight". Therefore sequencers came up with a clever way to get around this problem. Since the MIDI events are already known ahead of time, these events can be sent to the instrument one block early with an offset in sample frames. The plugin then receives these events at the start of the block, and knows not to start actually processing them until N samples have passed. This is what "sample accurate" means in sequencers.
However, if you are dealing with live input from a keyboard or some sort of other MIDI device, there is no way to "schedule" these events. In fact, by the time you receive them, the clock is already ticking! Therefore these events should just be sent to the plugin at the start of the very next block with an offset of 0. Sequencers such as Ableton Live, which allow a plugin to simultaneously receive both pre-sequenced and live events, simply send any live events with an offset of 0 frames.
Since you are using a very small block size, the worst-case scenario is a latency of .7ms, which isn't too bad at all. In the case of rtmidi, the timestamp does not represent an offset which you need to schedule around, but rather the time which the event was captured. But since you only intend to receive live events (you aren't writing a sequencer, are you?), you can simply pass any incoming MIDI to the plugin right away.