(OSX/Cocoa)
I need to be able to take any sound (or sounds) from a collection of sound files and play it (or them) through an arbitrary sound output channel, preferably with per-sound level control and more-or-less automatic channel mixing. (meaning that multiple sounds can play simultaneously in the same channel)
The sound must be capable of being looped cleanly.
The available output channels would be in addition to the default built-in audio, added via USB audio devices. For the purposes of this question a "channel" is equivalent to a USB Audio adapter.
In other words, I need to find/write something like:
-(BOOL) [someObject playContentsOfURL: (NSURL) soundURL
viaChannel: (id) channelspec
atVolumeLevel: (double) volume
loop: (BOOL) loop
];
AVAudioPlayer is very close, but it appears to assume the use of the built-in stereo audio. I cannot figure out how to route the output to a particular output device. It loops great, however.
NSSound's playbackDeviceIdentifier does the output device routing part, but NSSound's looping is terrible. If NSSounds looping wasn't so lousy, it would be a winner.
So if you have an answer for making NSSound loop cleanly, that would answer this question very satisfactorily.
If you can specify how to provide output routing to AVAudioPlayer, that would be equally nifty.
OR, if you can point to or provide CoreAudio or other code that will accomplish this, that could work also.
Related
In the newer XAudio2 API's for Windows 8 and 10, an AUDIO_STREAM_CATEGORY is passed to IXAudio2::CreateMasteringVoice.
The documentation goes on to say how these should be used for different types of audio. However an IXAudio2 is only allowed one master voice. To do this is completely separate IXAudio2 instances along with all associated interfaces required, or can categories be specified elsewhere in the audio graph by some means?
Games should categorize their music streams as AudioCategory_GameMedia so that game music mutes automatically if another application plays music in the background. Music or video applications should categorize their streams as AudioCategory_Media or AudioCategory_Movie so they will take priority over AudioCategory_GameMedia streams. Game audio for in-game cinematics or cutscenes, when the audio is premixed or for creative reasons should take priority over background audio, should also be categorized as Media or Movie.
You can create more than one IXAudio2 instance in a process so each will have it's own master voice. If you want to output more than one category of audio from a process, you need to create more than one IXAudio2 instance.
Generally you can get away with just one and always use AudioCategory_GameMedia.
I know this design is a bit of a kludge but the category is set on the WASAPI output voice, which is where XAudio2 sends it's mastering voice stuff to. Any other design would have required annotating category data within the internal XAudio audio graph which would have been quite complicated to implement for not a lot of value. We choose instead to just let applications have more than one audio-graph active at once each with it's own mastering voice and therefore it's own category.
How you choose to you support the audio category feature of WASAPI is up to you, and of course the best user experience depends on what exactly your application actually does.
OK, the first issue. I am trying to write a virtual soundboard that will output to multiple devices at once. I would prefer OpenAL for this, but if I have to switch over to MS libs (I'm writing this initially on Windows 7) I will.
Anyway, the idea is that you have a bunch of sound files loaded up and ready to play. You're on Skype, and someone fails in a major way, so you hit the play button on the Price is Right fail ditty. Both you and your friends hear this sound at the same time, and have a good laugh about it.
I've gotten OAL to the point where I can play on the default device, and selecting a device at this point seems rather trivial. However, from what I understand, each OAL device needs its context to be current in order for the buffer to populate/propagate properly. Which means, in a standard program, the sound would play on one device, and then the device would be switched and the sound buffered then played on the second device.
Is this possible at all, with any audio library? Would threads be involved, and would those be safe?
Then, the next problem is, in order for it to integrate seamlessly with end-user setups, it would need to be able to either output to the default recording device, or intercept the recording device, mix it with the sound, and output it as another playback device. Is either of these possible, and if both are, which is more feasible? I think it would be preferable to be able to output to the recording device itself, as then the program wouldn't have to be running in order to have the microphone still work for calls.
If I understood well there are two questions here, mainly.
Is it possible to play a sound on two or more audio output devices simultaneously, and how to achieve this?
Is it possible to loop back data through a audio input (recording) device so that is is played on the respective monitor i.e for example sent through the audio stream of Skype to your partner, in your respective case.
Answer to 1: This is absolutely feasable, all independent audio outputs of your system can play sounds simultaneously. For example some professional audio interfaces (for music production) have 8, 16, 64 independent outputs of which all can be played sound simultaneously. That means that each output device maintains its own buffer that it consumes independently (apart from concurrency on eventual shared memory to feed the buffer).
How?
Most audio frameworks / systems provide functions to get a "device handle" which will need you to pass a callback for feeding the buffer with samples (so does Open AL for example). This will be called independently and asynchroneously by the framework / system (ultimately the audio device driver(s)).
Since this all works asynchroneously you dont necessarily need multi-threading here. All you need to do in principle is maintaining two (or more) audio output device handles, each with a seperate buffer consuming callback, to feed the two (or more) seperate devices.
Note You can also play several sounds on one single device. Most devices / systems allow this kind of "resources sharing". Actually, that is one purpose for which sound cards are actually made for. To mix together all the sounds produced by the various programs (and hence take off that heavy burden from the CPU). When you use one (physical) device to play several sounds, the concept is the same as with multiple device. For each sound you get a logical device handle. Only that those handle refers to several "channels" of one physical device.
What should you use?
Open AL seems a little like using heavy artillery for this simple task I would say (since you dont want that much portability, and probably dont plan to implement your own codec and effects ;) )
I would recommend you to use Qt here. It is highly portable (Win/Mac/Linux) and it has a very handy class that will do the job for you: http://qt-project.org/doc/qt-5.0/qtmultimedia/qaudiooutput.html
Check the example in the documentation to see how to play a WAV file, with a couple of lines of code. To play several WAV files simultaneously you simply have to open several QAudioOutput (basically put the code from the example in a function and call it as often as you want). Note that you have to close / stop the QAudioOutput in order for the sound to stop playing.
Answer to 2: What you want to do is called a loopback. Only a very limited number of sound cards i.e audio devices provide a so called loopback input device, which would permit for recording what is currently output by the main output mix of the soundcard for example. However, even this kind of device provided, it will not permit you to loop back anything into the microphone input device. The microphone input device only takes data from the microphone D/A converter. This is deep in the H/W, you can not mix in anything on your level there.
This said, it will be very very hard (IMHO practicably impossible) to have Skype send your sound with a standard setup to your conversation partner. Only thing I can think of would be having an audio device with loopback capabilities (or simply have a physical cable connection a possible monitor line out to any recording line in), and have then Skype set up to use this looped back device as an input. However, Skype will not pick up from your microphone anymore, hence, you wont have a conversation ;)
Note: When saying "simultaneous" playback here, we are talking about synchronizing the playback of two sounds as concerned by real-time perception (in the range of 10-20ms). We are not looking at actual synchronization on a sample level, and the related clock jitter and phase shifting issues that come into play when sending sound onto two physical devices with two independent (free running) clocks. Thus, when the application demands in phase signal generation on independent devices, clock recovery mechanisms are necessary, which may be provided by the drivers or OS.
Note: Virtual audio device software such as Virtual Audio Cable will provide virtual devices to achieve loopback functionnality in Windows. Frameworks such as Jack Audio may achieve the same in UX environment.
There is a very easy way to output audio on two devices at the same time:
For Realtek devices you can use the Audio-mixer "trick" (but this will give you a delay / echo);
For everything else (and without echo) you can use Voicemeeter (which is totaly free).
I have explained BOTH solutions in this video: https://youtu.be/lpvae_2WOSQ
Best Regards
I am trying to find out which output formats are supported by a specific audio device in exclusive mode.
To do this, I am using IAudioClient->IsFormatSupported(), which according to the documentation should be usable for this.
Unfortunately, it returns AUDCLNT_E_UNSUPPORTED_FORMAT for almost every format I try to pass, except for default 2-channel, 44.1khz audio.
If I actually try to initialize the audioclient, there are however formats that succeed, but which failed in IsFormatSupported().
Just trying to Initialize every format is not an option because this could result in stopping the audio from other applications.
Has anyone else seen this behavior or know if there is another way to find which formats are supported by a specific audio device?
I have seen this behavior as well. It seems like IsFormatSupported will only accept what is marked as 'supported' in the playback device settings in Windows, but Initialize seems to actually end up asking the drivers if it's indeed possible.
In my specific situation, I have a Xoxar HDAV1.3 setup to use HDMI as output. Two playback devices are always available: Speakers and S/PDIF Pass-through Device. If I try, for example, to request 6 channels for the S/PDIF playback device, IsFormatSupported will reject it (in theory, S/PDIF only supports 2, and that's all I can see in the settings), but calling Initialize will succeed and work (it goes out HDMI after all, for which 6 channels is supported). Talk about misleading device names!
I'm afraid there's no real practical way to work around this issue.
I've pretty much finished work on a white noise feature for one of my applications using NSSound to play a loop of 10 second AAC-encoded pre-recorded white noise.
[sound setLoops: YES]
should be all that's required, right?
It works like a charm but I've noticed that there is an audible pause between the sound file finishing and restarting.. a sort of "plop" sound. This isn't present when looping the original sound files and after an hour or so of trying to figure this out, I've come to the conclusion that NSSound sucks and that the audible pause is an artefact of the synchronisation of the private background thread playing the sound. It seems to be dependent on the main run loop somehow and this causes the audible gap between the end and restarting of the sound.
I know very little about sound stuff and this is a very minor feature, so I don't want to get into the depths of CoreAudio just to play a looping 10s sound fragment.. so I went chasing after a nice alternative, but nothing seems to quite fit:
Core Audio: total overkill, but at least a standard framework
AudioQueue: complicated, with C++ sample code!?
MusicKit/ SndKit: also huge learning curve, based on lots of open source stuff, etc.
I saw that AVFoundation on iOS 4 would be a nice way to play sounds, but that's only scheduled for Mac OS X 10.7..
Is there any easy-to-use way of reliably looping sound on Mac OS X 10.5+?
Is there any sample code for AudioQueue or Core Audio that takes the pain out of using them from an Objective-C application?
Any help would be very much appreciated..
Best regards,
Frank
Use QTKit. Create a QTMovie for the sound, set it to loop, and leave it playing.
Just for the sake of the archives.
QTKit also suffers from a gap between the end of one play through and start of the next one. It seems to be linked with re-initializing the data (perhaps re-reading it from disk?) in some way. It's a lot more noticeable when using the much smaller but highly compressed m4a format than when playing uncompressed aiff files but it's still there even so.
The solution I've found is to use Audio Queue Services:
http://developer.apple.com/mac/library/documentation/MusicAudio/Conceptual/AudioQueueProgrammingGuide/AQPlayback/PlayingAudio.html#//apple_ref/doc/uid/TP40005343-CH3-SW1
and
http://developer.apple.com/mac/library/samplecode/AudioQueueTools/Listings/aqplay_cpp.html#//apple_ref/doc/uid/DTS10004380-aqplay_cpp-DontLinkElementID_4
The Audio Queue calls a callback function which prepares and enqueues the next buffer, so when you reach the end of the current file you need to start again from the beginning. This gives completely gapless playback.
There's two gotchas in the sample code in the documentation.
The first is an actual bug (I'll contact DTS about this so they can correct it). Before allocating and priming the audio buffers, the custom structure must switch on playback otherwise the audio buffer never get primed and nothing is played:
aqData.mIsRunning = 1;
The second gotcha is that the code doesn't run in Cocoa but as a standalone tool, so the code connects the audio queue to a new run loop and actually implements the run loop itself as the last step of the program.
Instead of passing CFRunLoopGetCurrent(), just pass NULL which causes the AudioQueue to run in its own run loop.
result = AudioQueueNewOutput ( // 1
&aqData.mDataFormat, // 2
HandleOutputBuffer, // 3
&aqData, // 4
NULL, //CFRunLoopGetCurrent (), // 5
kCFRunLoopCommonModes, // 6
0, // 7
&aqData.mQueue // 8
);
I hope this can save the poor wretches trying to do this same thing in the future a bit of time :-)
Sadly, there is a lot of pain when developing audio applications on OS X. The learning curve is very steep because the documentation is fairly sparse.
If you don't mind Objective-C++ I've written a framework for this kind of thing: SFBAudioEngine. If you wanted to play a sound with my code here is how you could do it:
DSPAudioPlayer *player = new DSPAudioPlayer();
player->Enqueue((CFURLRef)audioURL);
player->Play();
Looping is also possible.
Is it possible to use the NSSpeechRecognizer with an pre-recorded audio file instead of direct microphone input?
Or is there any other speech-to-text framework for Objective-C/Cocoa available?
Added:
Rather than using voice at the machine that is running the application external devices (e.g. iPhone) could be used for sending just an recorded audio stream to that desktop application. The desktop Cocoa app then would process and do whatever it's supposed to do using the assigned commands.
Thanks.
I don't see any obvious way to switch the input programmatically, though the "Speech" companion guide's first paragraph in the "Recognizing Speech" section seems to imply other inputs can be used. I think this is meant to be set via System Preferences, though. I'm guessing it uses the primary audio input device selected there.
I suspect, though, you're looking for open-ended speech recognition, which NSSpeechRecognizer is not. If you're looking to transform any pre-recorded audio into text (ie, make a transcript of a recording), you're completely out of luck with NSSpeechRecognizer, as you must give it an array of "commands" to listen for.
Theoretically, you could feed it the whole dictionary, but I don't think that would work since you usually have to give it clear, distinct commands. Its performance would suffer, I would guess, if you gave it a bunch of stuff to analyze for (in real time).
Your best bet is to look at third-party open source solutions. There are a few generalized packages out there (none specifically for Cocoa/Objective-C), but this poses another question: What kind of recognition are you looking for? The two main forms of speech recognition ('trained' is more accurate but less flexible for different voices and the recording environment, whereas 'open' is generally much less accurate).
It'd probably be best if you stated exactly what you're trying to accomplish.