Now I am working on a musical project in which i need accurate timings.I already used NSTimer,NSdate,but iam geting delay while playing the beats(beats i.e tik tok)So i have decided to use Audio Queue API to play my sound file present in the main bundle that is of .wav format, Its been 2 weeks i am struggling with this, Can anybody please help me out of this problem.
Use uncompressed sounds and a single Audio Queue, continuously running. Don't stop it between beats. Instead mix in raw PCM samples of each Tock sound (etc.) starting some exact number of samples apart. Play silence (zeros) in between. That will produce sub-millisecond accurate timing.
Related
I need to play 4 audios through a browser web.
These audios last 150ms, 300ms, 450ms and 600ms.
I don't care about latency (if an audio is played 100 ms after it's not that important for my purpose).
But I do care about the duration of these audios: is the 150ms audio last exactly 150ms or there is an error due to the audio board or other components?
I know for sure that there is an error (I see a test using a Mac).
My question is: can anyone show me a paper, an article or anything that talks about the duration and test different setting or tell me if this error is always (Windows, Mac, old device, new device) very small (less than 10ms for example).
In other words: if I play an audio of 100ms how long does it really last (100ms? more? less?)?
In what manner is the sound not lasting the correct amount of time?
Does the beginning or the end get cut off?
Does the sound play back slower or faster than it should?
In my experience, I've never heard an error with playback rates caused by the browser or sound boards. But I have come across situations where a sound is played back with a different audio format than which is was encoded. For example, a sound encoded at 48000 fps played back at 44100 fps will take longer to execute, but will be very close to the original in pitch (maybe about a 1/2 step lower). I recommend as a diagnostic step to confirm the audio format used at each end. How to do so will depend on the systems being used.
We are using PortAudio for recording audio in our electron application. As a node wrapper, we use naudiodon.
The application needs to record both audio and video, but using different sources. Audio, as said, is being recorded with Port Audio, with additional app logic on top. Video, on the other hand, is being recorded with standard MediaRecorder API, with its own formats, properties, and codecs.
We use event 'onstart' to track actual video start and in order to sync audio and video, we must also know the exact audio start time.
Problem is: We are not able to detect that exact timestamp of audio start. What should be the correct way of doing it?
Here is what we tried:
1. The first option is to listen to portaudio.AudioIO events, such as 'data' and 'readable'. Those are called as soon as PortAudio has new data chunk, so tracking the very first chunk minus its length in milliseconds would result in approximate audio start.
2. The second option is to add Writable pipe to AudioIO, and do pretty much the same thing as with events.
The issue is, that by doing any of those options, calculated start doesn't always result in the actual timestamp of audio start. While playing around with port audio it was known, that calculated timestamp is higher than it should be, as though some chunks are being buffered before actually released.
Actual audio start and first chunk release can be different, in a range of around 50 - 500 ms with chunk length ~50ms. So chunks might buffer sometimes, and sometimes they don't. Is there any way to track the actual start time of the first chunk? I wasn't able to find any relevant info in checking port audio docs.
Maybe there are any other ways to keep using PortAudio and record video separately, but finally achieve the same desired feature, of synching them together?
PortAudio 19.5, Naudiodon 2.1.0, Electron 6.0.9, Node.js 12.4.0
I have a windows phone 8 app which plays audio streams from a remote location or local files using the BackgroundAudioPlayer. I now want to be able to add audio effects, for example, reverb or echo, etc...
Please could you advise me on how to do this? I haven't been able to find a way of hooking extra audio processing code into the pipeline of audio processing even through I've read much about WASAPI, XAudio2 and looked at many code examples.
Note that the app is written in C# but, from my previous experience with writing audio processing code, I know that I should be writing the audio code in native C++. Roughly speaking, I need to find a point at which there is an audio buffer containing raw PCM data which I can use as an input for my audio processing code which will then write either back to the same buffer or to another buffer which is read by the next stage of audio processing. There need to be ways of synchronizing what happens in my code with the rest of the phone's audio processing mechanisms and, of course, the process needs to be very fast so as not to cause audio glitches. Or something like that; I'm used to how VST works, not how such things might work in the Windows Phone world.
Looking forward to seeing what you suggest...
Kind regards,
Matt Daley
I need to find a point at which there is an audio buffer containing
raw PCM data
AFAIK there's no such point. This MSDN page hints that audio/video decoding is performed not by the OS, but by the Qualcomm chip itself.
You can use something like Mp3Sharp for decoding. This way the mp3 will be decoded on the CPU by your managed code, you can interfere / process however you like, then feed the PCM into the media stream source. Main downside - battery life: the hardware-provided codecs should be much more power-efficient.
I'm working on a project that requires me to sync an audio playback(preferably an mp3 file) with my program.
My program reads a motion file from a txt file and output's it onto the serial port at a particular rate. At the same time an audio file has to be played back on the speaker. This audio file has to be in sync with the data..that is to say after say transmittin 100 bytes of data, the audio mustve played back to a predefined time.
What would be the tools used to play and control audio like this?
a tutorial would be great!
Thanks!!
In general, when working with audio, you want to synchronize other sources to audio. This is for several reasons, but most important is that audio runs on a clock running on its own hardware. You'll have to get timing information from that clock. There is a guide here written for using portaudio, but the principles apply to other situations:
http://www.portaudio.com/docs/portaudio_sync_acmc2003.pdf
I have an application that plays back video frame by frame. This is all working. However it needs to have playback Audio too, when Audio and Video running simultaneously it seems, Video lagging behind audio,
Logic i am using to display the video frame as follows
ProcessVideoThread(){
// Read the data from socket,
// decode it : this is going to be in side libvpx library, after decoding i am getting raw
// bitmap data
// After getting raw bitmap data, use some mechanism to update the image,
// here i tried runOnUIThread, handler but
}
Now what is happening, it seems UI thread is getting very late chance to update the image, i.e. libvpx is taking approx 30 ms to decode the image and through runOnUIThread, its taking 40 more ms to update the image, but inside UI thread i am updating it.
Can anyone advise me, how can i reduce the delay to update the image on the UI thread.
Looks interesting. If i was in your situation, i would examine the following methods.
1) Try using Synchronization between Audio An Video Threads.
2) Try reducing the Video Frames where audio is lagging and reduce the audio frequency where Video is lagging.
You can do the same in the following way.
int i;
if(i%2==0)
ShowFrame();
else
i++
What this will do is that it will straight away reduce the Video Speed from 24 12. So the Audio will now match with video. But yaa quality will be at stake as i already mentioned to you. This method is called frequency scaling. Widely used method to sync audio and Video.
Refer to the following for clear understanding and ways in which you can sync audio and video. This is in relation to the FFMPEG. I dont know how much of it you will be able to use, but definitely it will help you to get some idea.
http://dranger.com/ffmpeg/tutorial05.html
All the Best..