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.
Related
We are trying to use a full-duplex stream from two different devices on two different clock domains. PortAudio reports flags periodically, which I've put down to the interfaces being on different clock domains.
I have read and understood the behaviour documented in http://www.portaudio.com/docs/proposals/001-UnderflowOverflowHandling.html, and it is consistent with my observations. These flags are of course accompanied by clicks. I can reproduce this behaviour easily in the pa_fuzz example, so I believe it to be expected behaviour.
What I'm unsure of exactly what I should do with this information? The flags tell me that an overflow/underflow has already happened; how could I feasibly implement any resampling with this?
Am I going about this completely wrong? What is the typical usage pattern for PortAudio full-duplex using two different devices?
I'm using Core Audio. Thanks!
I don't have a proper answer for handling it at the PortAudio level, but there is a platform-specific solution that you could consider.
On OS X, you can create an aggregate device that combines the two devices, and enable drift correction to have CoreAudio perform the resampling. The resulting aggregate will have the inputs and outputs for both devices, so it may take additional logic to select the input and output channels of interest.
You could try it first by creating an aggregate device using Audio MIDI Setup.app. Apple does have some support notes on it, like
Create an Aggregate Device to combine multiple audio interfaces, and Set aggregate device settings in Audio MIDI Setup on Mac (which may have conflicting guidance on selecting drift correction)
I'm not sure it's formally documented other than headers like AudioHardware.h, but it is possible to create an aggregate device programatically as well. I have only experimented with it myself, and recall having some intermittent issues configuring the device (perhaps an asynchronous behavior with some operations? ).
Of course, if you're using PortAudio and trying to find a portable solution, this won't solve the issue generally.
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
This one makes me crazy:
On a Vista+ computer dedicated to this sound playing/recording application, I need my application to make sure the (default) microphone level is pushed to the max. How do I that?
I found the Core Audio lib, found how to get an IMMDevice to the default microphone. Now what?
Docs lead me to think that I need an ISimpleAudioVolume interface pointer from my IMMDevice, but how do I do that?
Note that I'm interested in any programmatic way to set this micro level (whether Core Audio or anything else). Ideally system-wide, but application-wide is ok.
TIA,
The trick is that in Core Audio, recording (aka capture) and rendering devices are not considered different (as long as you don't dive too deep of course), as opposed to former APIs such as waveXXX where there are different APIs for input and output devices.
Therefore, this full example by Larry Osterman that sets the speaker volume can be modified to set the microphone volume by simply changing eRender to eCapture in the enumerator call that returns the default device.
Thanks Larry!
The VideoPlayer (possibly VideoDisplay also) component is capable of somehow automatically picking the best quality video on the list it's given. An example is here:
http://help.adobe.com/en_US/FlashPlatform/beta/reference/actionscript/3/spark/components/mediaClasses/DynamicStreamingVideoItem.html#includeExamplesSummary
I cannot find the answers to below questions.
Assuming that the server that streams recorded videos is capable of switching across same videos with different bit rates and streaming them from any point within their timelines:
Is the bandwidth test/calculation within this component only done before the video starts playing, at which point it picks the best video source and never uses the other ones? Or, does it continuously or periodically execute its bandwidth tests and does it accordingly switch between video sources during the playback?
Does it support setting the video source through code and can its automatic switching between video sources be turned off (in case I want to provide this functionality to the user in the form of some button/dropdown or similar)? I know that the preferred video source can be set, but this only means that that video source will be tested/attempted first.
What other media servers can be used with this component, besides the one provided by Adobe, to achieve automated and manual switching between different quality of same video?
Obviously, I'd like to create a player that is smart enough to automatically switch between different quality of videos, and that will support manual instructions related to which source to play - both without interrupting the playback, or at least without restarting it (minor interruptions acceptable). Also, the playback needs to be able to start at any given point within the video, after enough data has been buffered (of course), but most importantly, I want to be able to start the playback beyond what's buffered. A note or two about fast-forwarding is not going to hurt, if anyone knows anything.
Thank you for your time.
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.