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
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I am now working on a Audio algorithm which need 256 samples of audio from a micro phone and i need to process this 256 samples and result should get played on a speaker. I have done it using two wave file which is already on the file, now i need to do it in the real time.
Need a solution for this
I am working on a project on the stm32f769i-disco board, which has to run a video game made by me. The graphical part is solved, where I have problems is when I need to play two or more wav files, I do not know which is the correct way to add them in a single stream.
What is the algorithm that I must follow for more than one stream?
PD: I know manage a single stream which is played through dma with double buffer.
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 have a DirectShow application that I built with Delphi 6 using the DSPACK component library. For two days I have been trying to solve a problem with audio playback. When I run the filter graph I create I hear repetitive clicks in the playback. What was really confusing was that the audio file I created simultaneously with my filter graph had clean continuous audio, not gaps. So I knew that the audio buffers were being delivered properly but something I was doing was "jamming up" the "live" playback. Or so I thought. I spent two days diagnosing the problem looking for semaphores being held too long (locks) or perhaps timestamp problems, which I documented in this other Stack Overflow post:
Getting stuttering during rendering of my DirectShow filter despite output file being "smooth"
A few minutes ago I decided to try a test with the Graph Edit utility. I created a dead simple graph consisting of just the capture device I was using (VOIP phone microphone), and the renderer device I was using (HD ATI Rear Audio output to headphones). Two filters total. Much to my surprise I heard the same clicking. So here was a case that did not involve my code at all and I heard clicking.
Then I changed the audio renderer in the Graph Edit created filter graph to the VOIP phone ear piece. The clicking went away.
Now I know there's a way to get smooth audio on ut the ATI Rear Audio device since its the preferred audio output device and everything from videos I play on my PC to wave files I play on it sound flawless. So are the other software programs doing something different than just connecting filters? I am wondering if perhaps the default mode for the HD ATI Rear Audio is without double-buffering and perhaps those other software programs know how to enable that feature? Or are they doing something else, perhaps using another DirectShow or DirectSound filter or technique for example, to make the audio play smoothly on the HD ATI Rear Audio renderer?
What you possibly having (depends on actual stuttering though) is that when you are using capture and playback devices backed by different hardware, their sampling rates slightly differ. For example, you capture 22050 Hz at actual rate of (22050 - 2%) Hz and you play it back with hardware consuming bytes at (22050 + 2%) Hz.
Now obviously this won't work out smooth: eventually playback will experience data underlow... If you save into file and play back from file, it will go smooth as the file will be able to supply data at the rate of playback device. If capture and playback devices are the same hardware, they are likely to use shared "hardware" clock and rates match.
The problem is known as "rate matcing" and is discussed on MSDN in Live Sources section.
I have a very complicated audio setup for a project. Here's what we have:
3 applications playing sound
2 applications recording sound
2 sound cards
I really don't really have the code to any of these applications. All I want to do is monitor and control the audio streams. Here are a few examples of operations I'd like to do while the applications are running:
Mute one of the incoming audio streams.
Have one of the incoming audio streams do a "solo" (be the only stream that can "talk").
Get a graph (about 30 seconds worth) of the audio that each stream produced.
Send one of the audio streams to soundcard #1, but all three audio streams to soundcard #2.
I would likely switch audio streams every 2 minutes or so with one of the operations listed above. A GUI would be preferred. I started looking at the sound systems in Linux and it gets extremely complex and I feel like there have been many new advances in the past few years. I see jack, pulseaudio, artsd, and several other packages. They all have some promise but where should I start? Is there something someone already built that can help?
PulseAudio should be able to let you do all that. You'll need to configure a custom pipeline for splitting the app's audio for task 4, and I'm not exactly certain how you'd accomplish task 3, but I do know that it's capable of all sorts of audio stream handling via its volume control (pavucontrol).
I use Jack, which is quite simple to install and use, even if it
requires more efforts to configure with Flash and Firefox ...
You can try the latest Ubuntu Studio distribution and see if it solves your
problem (for the GUI, look at "patchage").