I'm trying to work with an old version of DirectX (8.1) and I'm finding the documentation more than a little confusing. It feels like the IDirectMusicPerformance8 interface is for MIDI playback, as it has various MIDI-related methods on it, but various parts of the documentation suggest that it can be used to play back WAV data as well. For instance, the tutorial says that you call InitAudio on the performance, and it states "Wave files require only a single performance channel, and MIDI files require up to 16".
So is it for MIDI or can it be used for WAV?
If it can be used for WAV, I'm even more stumped, because I'm initializing it, running performance->PlaySegment(segment, 0, 0, NULL), getting a success response, but the audio is silent.
OK, I figured it out. Apparently I had been using IDirectMusicPerformance instead of IDirectMusicPerformance8, which I guess is some kind of old compatibility thing that is missing various new DirectX 8.1 methods. Once I switched to that (along with the associated '8' versions of the loader and segment interfaces), used their new InitAudio and LoadObjectFromFile methods, called segment->Download(performance) before playing, then played with the new PlaySegmentEx method, it worked and played the WAV.
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Looking through all the API documentation, I can see how one could create procedural audio, but once an audio file is created, I want it to play on an object, but from I can tell, I believe I need it to play using the function calls PlayEventAtLocation in the UE4 plugin, which means I need to get the sound into an event.
I used to have my setup in Unity 4.x. I want to dynamically construct a wav file in game and play it back. The idea was to have silent audio all over the map that would loop, but play muted. The player when in range would capture audio from this muted audio source at their discretion.
The idea is that you have wav file that plays in game and at any given time I can start grabbing data from where the buffer is at currently until I decide to stop. I take all the data that I created in this new buffer and create a new wav file with it.
Example, like a 20 second file, but I would grab the a 7 second audio clip starting 5 seconds in. So my new audio file would be from 5 to 12. I would think you could do similar things in FMOD because I’ve looked at the recording examples and gapless playback examples, etc. and it does seem to have that same functionality and access to seek the files.
Now I need to migrate this new file that will made in game to something UE4 would use. In FMOD, looking through the .h and .cpp files in the plugin files, I see accept Fmod events only to attach to a UObject. Since I've not made an event in FMOD Studio, I can't use these. What is the sound that createSound makes? is it a wav? and fsb? I just a have a sound, and don't know what format it is.
I can’t use designer to make this sound because its dependent on the player at any given time during play.
In Unity, what I did was access the buffer of an audio file, pull data from the buffer for any given time, and place in a new buffer that I then turned into a file. While pulling data, I would check buffer size and frequency of sound files to make sure I had a gapless playback. (Not perfect, but pretty darn close), I’d use the audio functions in Unity to convert my array of data into a useable audioclip and run it through a sound emitter. It was pretty nice. Because I would keep the original wav file muted, but looping. So the player never knew what they captured. It was really nice.
Since UE4 doesn’t allow access to uncompressed PCM data, I can't do this low level data manipulation in UE4. I had to use FMOD, but its proving to be just as difficult because either its not documented, or lacks the functionality I want. I need help please.
If the data that is created in createsound is just normal pcm wav file data, then I can use a standard AudioComponent, and just save it to a file, and pull it in from UE4. If it is, then I need to turn it into an event so I can use FMODPlayEventAttached from the FMOD plugin library.
I've made a few other posts in various locations that have all been silent. Any comment would be appreciated. I know I've been reading a lot of documentation these last few days on FMOD, but I still may have missed something if people want to point me in a better direction, or if you have something to add, feel free.
Thanks all. I hope I was descriptive enough.
So, I've been working on a project that is going to require the playback of Midi. I have already done this with a little bit of code that looks like this:
mciSendString(L"play C:\\aha.mid", NULL, 0, NULL);
And that works perfectly fine for playing a Midi file.
But, what I also need to do is to get the note events from the playback of the midi file itself. I've seen a great deal of libraries that can help with playback or with reading the raw data of the midifile, but not both, and not both at the same time. I've tried Midifile, did some searching around the Juce library, but nothing seems to get the functionality that I need.
Is there something out there that can solve my problem or am I just thinking about this problem the wrong way? Maybe there's some way to get the current notes being played on the Microsoft wavetable (would be much easier if that's possible).
I'm trying to develop a little application in which you can load a mp3 file and play it in variable speeds! (I know it already exists :-) )
I'm using Qt and C++. I already have the basic player but I'm stuck with the rate thing, because I want to change the rate smoothly (like in Mixxx) without stopping the playback! The QMediaPlayer always stops if I change the value and creates a gap in the sound. Also I don't want the pitch to change!
I already found something called "SoundTouch" but now I'm completely clueless what to do with it, how to process my mp3 data and how to get it to the player! The "SoundTouch" Library is capable of doing what I want, i got that from the samples on the homepage.
How do I have to import the mp3 file, so I can process it with the SoundTouch functions
How can I play the output from the SoundTouch function? (Perhaps QMediaPlayer can do the job?)
How is that stuff done live? I have to do some kind of stream I guess? So I can change the speed during play and keep on playing without gaps. Graphicaly in my head it has to be something that sits between the data and the player, where all data has to go through live, with a small buffer (20-50 ms or so) behind to avoid gaps during processing future data.
Any help appreciated! I'm also open to any another solution then "SoundTouch" as long as I can stay with Qt/C++!
(Second thing: I want to view a waveform overview aswell as moving part of it (around actual position of the song), so I could also use hints on how to get the waveform data)
Thanks in advance!
As of now (Qt 5.5) this is impossible to do with QMediaPlayer only. You need to do the following:
Decode the audio using GStreamer, FFMpeg or (new) QAudioDecoder: http://doc.qt.io/qt-5/qaudiodecoder.html - this will give you raw PCM stream;
Apply SoundTouch or some other library to this raw data to change the pitch. If GPL is ok, take a look at http://nsound.sourceforge.net/examples/index.html, if you develop proprietary stuff, STK might be a better choice: https://ccrma.stanford.edu/software/stk/
Output the modified data into audio device by using QAudioOutput.
This strategy uses Qt as much as possible, and brings you the best platform coverage (you still lose Android though as it does not support QAudioOutput)
I've stumbled through some code to enumerate my microphone devices (with some help), and am able to grab the "friendly name" and "clsid" information from each device.
I've done some tinkering with GraphEd.exe to try and figure out how I can take audio from directshow and write it to a file (I'm not currently concerned about the format, wav should be fine), and can't seem to find the right combination.
One of the articles I've read linked to this Windows SDK sample, but when I examined the code, I ended up getting pretty confused at how to use that code, ie. setting the output file, or specifying which audio capture device to use.
I also came across a codeguru article that has a nicely featured audio recorder, but it does not have an interface for selecting the audio device, and I can't seem to find where it statically picks which recording device to use.
I think I'd be most interested in figuring out how to use the Windows SDK sample, but any explanation on either of the two approaches would be fantastic.
Edit: I should mention my knowledge and ability as a win32 COM programmer is very low on the scale, so if this is easy, just explain it to me like I'm five, please.
Recording audio into file with DirectShow needs you to build the right filter graph, as you should have figured out already. The parts include:
The device itself, which you instantiate via moniker (not CLSID!), it is typically PCM format
Multiplexer component that converts streams into container format
File Writer Filter that takes file-compatible stream and writes into a file
The tricky moment is #2 since there is not standard component available. Windows SDK samples however contains the missing part - WavDest Filter Sample. Building it and making it ready for use, you can build a graph that records from device into .WAV file.
Your graph will look like this, and it's built easily programmatically as well:
I noticed that I have a variation of WavDest installed with Google Earth - for the case you have troubles building it yourself and you will be looking for prebuilt binary.
You can instruct ffmpeg to record from a directshow device, and output to a file.
I'm writing a cross-platform Qt-based program that from time to time needs to play back audio supplied externally (outside my control) as raw PCM. The exact format is 16 bit little-endian PCM at various common sample rates.
My first obvious idea was to use Qt's own Phonon for audio playback, but there are two problems with this approach:
As far as I can see, Phonon does not support headerless PCM data. I would have to hack around this and fake a WAV header every time playback starts. Not a showstopper, though.
More importantly: There doesn't seem to be any way to control how Phonon (and its backends such as xine, PulseAudio, DirectX, whatever) prebuffers. Its default behaviour seems to be something like 5 seconds of prebuffering, which is way too much for me. I'd prefer about 1 second, and I'd definitely like to be able to control this!
I'm currently looking at Gstreamer, FFMPEG and libvlc. Any thoughts? Since my audio is in a very simple format and I don't need to do fancy mixing stuff (just volume control), I'd like a simple, free (as in freedom), cross-platform and widely available library.
Qt 4.6 has the new QtMultimedia module.
https://doc.qt.io/archives/4.6/qtmultimedia.html
The QAudioOutput class would seem to do what you want - it just plays raw PCM data.
ffmpeg, libvlc and gstreamer have abilities beyond raw pcm, such as codec support.
For your purposes, SDL (example 1, example 2), OpenAL, QAudioOutput are sufficient. SDL is probably the most popular option.
Also, why do you want to control buffering? Buffering a lot means less interrupts and lower power consumption.
Have you looked at OpenAL?