When reading about the Mac OS X audio API, they seem to list quite a few functions, but not (that I can find) how to use them.
For instance, they list a subclass, IOAudioLevelControl. Should I just create an object with this, and assume that it controls the audio level? And if so, what should I assume? That it has a constructor that sets the audio level, so I should declare them IOAudioLevelControl levelControl(5);?
I'm very confused here.
Edit: My goal is to make a program that detects the audio level of all the running programs(like Chrome when you play music from Youtube etc) and lower the level if it goes over a certain level, set by the user.
IOKit is way too low level - it's the user-space proxy for device drivers on the system. MacOSX provides several layers that sit about this. Most likely, you can use these APIs directly without being root.
Core Audio provides the next layer up in the stack. The diagram on this page shows the architecture.
Setting CoreAudio are higher level APIs such as Audio Units and AVFoundation. It is likely that are what you want.
Related
I'm right now reading the microsoft documentation about drivers and core audio apis. At the moment I'm still confuse which way to go to achieve what I need.
I have an audio application which is Standalone and coded with framework JUCE in C++. And I need to build a Windows solution that would capture the audio stream that is going to an audio endpoint device to use it as an input of my audio application.
This stream must have an unaltered volume: always 1.0 (no matter if the hardware volume is changed or muted).
I must be able to choose between the different endpoint devices, for exemple if I have an external soundcard that is plugged, my audio application should be able to intercept and copy the stream that is going to that external soundcard, or do the same for the stream that is going to the built-in speakers.
The idea is to capture the output streams before they are modified by hardware volume modifications, and make a copy of them routed to my application without changing the output routing and behaviour.
The microsoft documentation is very furnished, but even if the WASAPI provides a lot of ways to capture and stream from audio endpoint devices, I'm not sure it is possible to get an unaltered volume, as it will always capture what's exactly coming out of the speakers.
This is why I don't know If I can implement a feature directly in my audio application that will get the streams I want with WASAPIs or if I have to code a proper Audio Driver that would make a copy of the streams I want for my application to be able to use these streams.
The documentations I refer to:
Audio Drivers design guide
Core Audio APIs / WASAPI
Thanks for the help,
Best,
Maxime
Sometimes the volume control is implemented in software, and sometimes it is implemented in hardware. You can call IAudioEndpointVolume::QueryHardwareSupport to see if the volume control for the audio endpoint you're working with is implemented in hardware or software.
Sometimes the audio loopback is implemented in software, and sometimes it is implemented in hardware. There is no API to tell which.
If the audio loopback is implemented in software, and the volume control is implemented in hardware, then you will get back the data you want.
If the audio loopback is implemented in hardware, or the volume control is implemented in software, the the audio data you get back has already had the volume adjustment applied.
What does your application do with the audio data it receives? The primary use case for audio loopback data is echo cancelation, where you usually WANT the volume to be applied.
I am trying to write a pro music/audio processing application, and I would like to be able to interact with the audio inputs/outputs at a very low level - ideally something allowing me to apply effects to the audio inputs and output this in real-time, similar to programs like Logic, Ableton etc.
I have written a pretty basic program that detects audio endpoint devices and can change their volumes using the MMDevice interface, but this is nowhere near the functionality I would like.
I have learned from the Microsoft docs that the four core-audio APIs are:
MMDevice
WASAPI
DeviceTopology
EndpointVolume
but it doesn't seem like any of these have the capabilities that I need. I'm thinking that I will need to be able to interact with the speakers at the level of setting the position of the membrane at a given time.
Is this even possible? If so, what can I use to do this?
The Windows Audio Session API (WASAPI) is the best bet for this purpose. It allows interaction with audio endpoints and setting up audio streams (which are streams of data that you can send or receive in real time). A good example is here.
I was wondering if it is possible to capture a copy of the audio output in Qt so I can process it. Here they said it's possible to monitor the playback, but I think it's only possible if you use a self made music player, which I don't want. I want to capture the signal from no matter where it is player (youtube, spotify, facebook, etc.). Is there a way to analyze this data with Qt? Is it possible to set my output of my soundcard as a QMediaSource?
Thank you in advance.
In general, no, that isn't possible, simply because your process (and therefore the Qt library that is loaded into your process) does not have access to that information. (I believe this lack of access is deliberate; since if it did have access like that, there might be security and/or privacy implications, i.e. app A could use it to spy on the audio output of app B, etc)
There may be an OS-specific mechanism that you can use; for example, if you are running your program under MacOS/X, you can install the SoundFlower audio driver that can function as a loopback device, allowing programs to read audio from its "audio input" that was previously routed to its "audio output". But without that kind of external support, it's not currently possible to record the computer's audio output via Qt.
I learnt that c++ has many low level access to hardware so we write drivers in c++. But when we write a normal desktop application in c++, will it be able to access hardware directly?
I'm asking this because desktop applications runs on the application layer. OS layer and OEM layer are there Between App layer and Hardware. then how can our application written in c++ access hardware directly?
please somebody explain me how exactly c++ desktop apps works.
thanks.
In general, desktop applications have to go through the OS to access any resources, from extra memory to hard drives and sound chips.
In an embedded system, the hardware can be accessed directly, usually through a pointer. So to write to a UART register, one would assign the address of the register to a pointer and dereference the pointer.
Many compiler libraries provide support for various platforms and embedded systems. I'm using an IAR compiler for an ARM embedded system and programming with C++. We don't use streams since we don't have terminal I/O.
Edit 1: cout example
For example, to print to the console, via cout or printf, the compiler provides a library that calls Operating System functions to display the text.
The Operating System function may send the text directly to the console or may pop up a "console window" and send the text to that.
The Console functions convert the text to bitmaps and send the bitmaps to the Graphics Controller.
The Graphics Controller displays the bitmap on the screen.
I am trying to fix an Audacity bug that revolves around portmixer. The output/input level is settable using the mac version of portmixer, but not always in windows. I am debugging portmixer's window code to try to make it work there.
Using IAudioEndpointVolume::SetMasterVolumeLevelScalar to set the master volume works fine for onboard sound, but using pro external USB or firewire interfaces like the RME Fireface 400, the output volume won't change, although it is reflected in Window's sound control panel for that device, and also in the system mixer.
Also, outside of our program, changing the master slider for the system mixer (in the taskbar) there is no effect - the soundcard outputs the same (full) level regardless of the level the system says it is at. The only way to change the output level is using the custom app that the hardware developers give with the card.
The IAudioEndpointVolume::QueryHardwareSupport function gives back ENDPOINT_HARDWARE_SUPPORT_VOLUME so it should be able to do this.
This behavior exists for both input and output on many devices.
Is this possibly a Window's bug?
It is possible to workaround this by emulating (scaling) the output, but this is not preferred as it is not functionally identical - better to let the audio interface do the scaling (esp. for input if it involves a preamp).
The cards you talk about -like the RME- ones simply do not support setting the master or any other level through software, and there is not much you can do about it. This is not a Windows bug. One could argue that giving back ENDPOINT_HARDWARE_SUPPORT_VOLUME is a bug though, but that likely originates from the driver level, not Windows itself.
The only solution I found so far is hooking up a debugger (or adding a dll hook) to the vendor supplied software and looking at the DeviceIOControl calls it makes (those are the ones used to talk to the hardware) while setting the volume in the vendor software. Pretty hard to do this for every single card, but probably worth doing for a couple of pro cards. Especially for Audacity, for open source audio software it's actually not that bad so I can imagine some people being really happy if the volume on their card could be set by it. (at the time we were exclusively using an RME Multiface I spent quite some time in figuring out the DeviceIOControl calls, but in the end it was definitely worth it as I could set the volume in dB for any point in the matrix)