I have an MFC based project that decodes some data and generates 16 bit 48000 Hz raw wav audio data
The program continuously generates wav audio data in real time
Are there any functions in MFC that will let me play back the audio data in the sound card? I have been googling around for a while and the consensus seems to be that MFC doesn't have this feature. I have also found this tutorial that shows how to playback a wav file using PlaySound() function, but it looks like it is only for wav files and even if it plays audio data in memory, that data has to be prepared in the form of a full wav file with all the header information, while I need to play back raw wav data generated in real time
I have also seen people suggest using Direct X, but I feel like something like this should be possible using basic windows library functions without having to use any other extra libraries. I also found this tutorial for creating and reading wav files in an MFC based project, but it's not really clear how to use it to play raw wav data in memory. This tutorial uses waveOutOpen() function to playbakc the wav file, and it looks like this is probably what I need, but I cannot find a simple tutorial that shows how to use it.
How do I playback raw wav audio in memory in an MFC Dialog based project? I am looking for something where I can specify pointer to the wav data, number of samples, bits and sampling frequency and the function would playback the wav data for me. A basic working example such as generating a sinewave and playing it back will be appreciated. If directx is the only way to do this then that's fine as well.
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I would like to make an app (Target pc windows) that let you modify the micro input in real time, like introducing sound effects or even modulating your voice.
I searched over the internet and only found people telling that it would not be possible without using a virtual audio cable.
However I know some apps with similar behavior (voicemod, resonance) not using a virtual audio cable so I would like some help about how can be done (just the name of a library capable would be enough) or where to start.
Firstly, you can use professional ready-made software for that - Digital audio workstation (DAW) in combination with a huge number of plugins for that.
See 5 steps to real-time process your instrument in the DAW.
And What is (audio) direct monitoring?
If you are sure you have to write your own, you can use libraries for real-time audio processing (as far as I know, C++ is better for this than C#).
These libraries really works. They are specially designed for realtime.
https://github.com/thestk/rtaudio
http://www.portaudio.com/
See also https://en.wikipedia.org/wiki/Csound
If you don't have a professional sound interface yet, but want to minimize a latency, read about Asio4All
The linked tutorial worked for me. In it, a sound is recorded and saved to a .wav.
The key to having this stream to a speaker would be opening a SourceDataLine and outputting to that instead of writing to a wav file. So, instead of outputting on line 59 to AudioSystem.write, output to a SourceDataLine write method.
IDK if there will be a feedback issue. Probably good to output to headphones and not your speakers!
To add an effect, the AudioInputLine has to be accessed and processed in segments. In each segment the following needs to happen:
obtain the byte array from the AudioInputLine
convert the audio bytes to PCM
apply your audio effect to the PCM (if the effect is a volume change over time, this could be done by progressively altering a volume factor between 0 to 1, multiplying the factor against the PCM)
convert back to audio bytes
write to the SourceDataLine
All these steps have been covered in StackOverflow posts.
The link tutorial does some simplification in how file locations, threads, and the stopping and starting are handled. But most importantly, it shows a working, live audio line from the microphone.
Basically my question is: if I have some raw LPCM data (NOT AN AUDIO FILE) that I want to play how would I go about writing that data to playable audio buffers and then beginning playback in C/C++? I realize that C might not be possible but my understanding is that the whole OS X native audio system is based on C++ so I don't think I should have to use objective C. I have dealt with AudioQueue services but I have not been able to find a method to directly write audio data to an AudioQueueBuffer. Any help is appreciated.
I am downloading various sound files with my own c++ http client (i.e. mp3's, aiff's etc.). Now I want to parse them using Core Audio's AudioToolbox, to get linear PCM data for playback with i.e. OpenAL. According to this document: https://developer.apple.com/library/mac/#documentation/MusicAudio/Conceptual/CoreAudioOverview/ARoadmaptoCommonTasks/ARoadmaptoCommonTasks.html , it should be possible to also create an audio file from memory. Unfortunately I didn't find any way of doing this when browsing the API, so what is the common way to do this? Please don't say that I should save the file to my hard drive first.
Thank you!
I have done this using an input memory buffer, avoiding any files, in my case I started with AAC audio format and used apple's api : AudioConverterFillComplexBuffer to do the hardware decompress into LPCM. The trick is you have to define a callback function to supply each packet of input data. That api call does the format conversion on a per packet basis. In my case I had to write code to parse the compressed AAC data to identify packet starts (0xfff) then use the callback to spoon feed each packet into the api call. I am also using OpenAL for audio rendering which has its own challenges to avoid using input files.
I am struggling to work out how to pass the data from buffer to an array to allow me to display what is in the buffer. Is there an example of code somewhere that is a simple record audio and read buffer? Simpler the better.
I am trying to do something in real time and not read in data from a saved file. I could paste some code that I have tried with no success.
The classic example of writing and reading to audio buffers using AudioQueue is Apple's SpeakHere sample project.
You can find tons of stuff on this and on the web. Just search on "speakhere". One standout page is Matt Gallagher's articles on Streaming and playing an MP3 stream.
Check out my personal answers also. I have some quite in depth posts on audio buffers, e.g.
iOS Stream Audio from one iOS Device to Another
I have the need to play multiple soundeffects at once in my WP7 app.
I currently have it working with wav files that takes around 5 megabyte, instead of 500kb when coded in wma/mp3.
Current part of the code:
Stream stream = TitleContainer.OpenStream(String.Format("/location/{0}.wav", value)
SoundEffect effect = SoundEffect.FromStream(stream);
effect.Play();
This works great in a loop, preparing all effects, and then playing them.
However, I would really like to use mp3/wma/whatever-codec to slim my xap file down.
I tried to use MediaElement, but it appears that you also can't use that to play multiple files. Also the XNA MediaPlayer can't be instantiated, and as far as I experienced can't be made to play multiple files at once.
The only solution I see left is that I somehow decode the mp3 to wav and feed that Stream to SoundEffect.
Any ideas on how to accomplish the multiple playback? Or suggestions on how to decode mp3 to wav?
On the conversion... sorry - but I don't think there's any api currently available for WMA or MP3 decoding.
Also, I don't think there are any implementations of MP3, WMA or Ogg decoders which are available in pure c# code - all of them I've seen use DirectShow or PInvoke - e.g. see C# Audio Library.
I personally do expect audio/video compression/decompression to be available at some point in the near future in the WP7 APIs - but I can't guess when!
For some simple compression you can try things like shipping mono instead of stereo files, or shipping 8 bit rather than 16 bit audio files - these are easy to convert back to 16 bit (with obvious loss of resolution) on the phone.
Using compression like zip might also help for some sound effects... but I wouldn't expect it to be hugely successful.