问题
I'm trying to get the input from my guitar to be played through my computer using the portaudio library and the ASIO sdk.
I have been following some of the tutorials on the official website to get the basics set up. Currently I got it working so that portaudio is listening to the right input and output device and I have the callback setup to just output the input and do nothing with it like this:
static int paTestCallback(const void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo* timeInfo, PaStreamCallbackFlags statusFlags, void *userData)
{
float *out = (float*)outputBuffer;
float* in = (float*)inputBuffer;
for (int i = 0; i<framesPerBuffer; i++)
{
*out++ = *in++; /* left */
*out++ = *in++; /* right */
}
return 0;
}
This callback is setup by calling this:
PaError error = Pa_OpenDefaultStream(&stream, 2, 2, paFloat32, 44100, paFramesPerBufferUnspecified, paTestCallback, &data);
Pa_StartStream(stream);
Now, this does work but I have a lot of delay (about 0.5s) when I strike a string on my guitar and when I hear it through the monitors.
Is there a way to solve this delay? Do I need to rewrite the callback method?
EDIT:
So, I got the delay to be a lot better using this code instead of the basic Pa_OpenDefaultStream()
int defaultIn = Pa_GetDefaultInputDevice();
int defaultOut = Pa_GetDefaultOutputDevice();
PaStreamParameters *inParam = new PaStreamParameters();
inParam->channelCount = 2;
inParam->device = defaultIn;
inParam->sampleFormat = paFloat32;
inParam->suggestedLatency = 0.05;
PaStreamParameters *outParam = new PaStreamParameters();
outParam->channelCount = 2;
outParam->device = defaultOut;
outParam->sampleFormat = paFloat32;
outParam->suggestedLatency = 0;
error = Pa_OpenStream(&stream, inParam, outParam, 44100, paFramesPerBufferUnspecified, paNoFlag, paTestCallback, &data);
if (error != paNoError) {
Logger::log("[PortAudioManager] Could not open default stream. Exiting function...");
return;
}
Pa_StartStream(stream);
There is still a little bit of delay though, mostly noticeable when playing more then just a single note.
EDIT:
I figured out with the help of Ross-Bencina that the windows default input device and output device doesn't change anything to the index of the host api's in PortAudio. I seemed to be using MME all this time. I did the following to get the right index for the ASIO device:
int hostNr = Pa_GetHostApiCount();
std::vector<const PaHostApiInfo*> infoVertex;
for (int t = 0; t < hostNr; ++t) {
infoVertex.push_back(Pa_GetHostApiInfo(t));
}
Then I just checked which is the one with ASIO and set the suggestedLatency in both PaStreamParameters
to 0 and the delay is now gone and sound is good (although it's mono for now).
回答1:
You are on the right track using paFramesPerBufferUnspecified
.
The ASIO latency behavior depends on the driver. There are two possibilities:
The ASIO driver lets the code (i.e. PortAudio) request a latency (possibly with some constraints). PortAudio finds to best match between a supported driver buffer size and the latency that you request.
The other possibility is that your audio interface does not provide programmatic control over latency settings. Instead, latency is only selectable from the driver's ASIO control panel UI (and the driver will force a fixed buffer size on PortAudio). In this case, you should investigate the driver control panel UI to set the lowest workable latency.
In either case, your approach with Pa_OpenStream
is close to optimal, but you should request zero latency for both input and output (in your edit you're requesting 50ms input latency, zero output latency). The end result will be that PortAudio selects the lowest available ASIO buffer size. If this turns out to be unstable (audio glitches) then you'll need to increase the requested latency.
include/pa_asio.h
exposes a host-API-specific interface for querying the ASIO buffer sizes allowed by the driver (be aware that this can change if you change settings in the control panel). It also provides a function to display the driver's control panel UI.
EDIT: Note that Pa_GetDefaultInputDevice()
and Pa_GetDefaultOutputDevice()
will only return ASIO devices if you built PortAudio for ASIO only. If you included any other more common APIs in the build (e.g. WMME or DirectSound) they will be given priority as the (lowest common denominator) default device. You could add a check that you are actually accessing the ASIO device:
assert(Pa_GetHostApiInfo(Pa_GetDeviceInfo(Pa_GetDefaultOutputDevice())->hostApi)->type == paASIO);
If PortAudio is compiled with support for multiple host APIs: To get the default ASIO device: enumerate host APIs using Pa_GetHostApiCount
and Pa_GetHostApiInfo
to find the ASIO host API. Then pull the default ASIO device indices from the returned PaHostApiInfo
struct.
来源:https://stackoverflow.com/questions/33171525/input-delay-with-portaudio-callback-and-asio-sdk