问题
I'm having a debate with some developers on another forum about accurately generating MIDI events (Note On messages and so forth). The human ear is pretty sensitive to slight timing inaccuracies, and I think their main problem comes from their use of relatively low-resolution timers which quantize their events around 15 millisecond intervals (which is large enough to cause perceptible inaccuracies).
About 10 years ago, I wrote a sample application (Visual Basic 5 on Windows 95) that was a combined software synthesizer and MIDI player. The basic premise was a leapfrog-buffer playback system with each buffer being the duration of a sixteenth note (example: with 120 quarter-notes per minute, each quarter-note was 500 ms and thus each sixteenth-note was 125 ms, so each buffer is 5513 samples). Each buffer was played via the waveOutWrite method, and the callback function from this method was used to queue up the next buffer and also to send MIDI messages. This kept the WAV-based audio and the MIDI audio synchronized.
To my ear, this method worked perfectly - the MIDI notes did not sound even slightly out of step (whereas if you use an ordinary timer accurate to 15 ms to play MIDI notes, they will sound noticeably out of step).
In theory, this method would produce MIDI timing accurate to the sample, or 0.0227 milliseconds (since there are 44.1 samples per millisecond). I doubt that this is the true latency of this approach, since there is presumably some slight delay between when a buffer finishes and when the waveOutWrite callback is notified. Does anyone know how big this delay would actually be?
回答1:
The Windows scheduler runs at either 10ms or 16ms intervals by default depending on the processor. If you use the timeBeginPeriod() API you can change this interval (at a fairly significant power consumption cost).
In Windows XP and Windows 7, the wave APIs run with a latency of about 30ms, for Windows Vista the wave APIs have a latency of about 50ms. You then need to add in the audio engine latency.
Unfortunately I don't have numbers for the engine latency in one direction, but we do have some numbers regarding engine latency - we ran a test that played a tone looped back through a USB audio device and measured the round-trip latency (render to capture). On Vista the round trip latency was about 80ms with a variation of about 10ms. On Win7 the round trip latency was about 40ms with a variation of about 5ms. YMMV however since the amount of latency introduced by the audio hardware is different for each piece of hardware.
I have absolutely no idea what the latency was for the XP audio engine or the Win9x audio stack.
回答2:
At the very basic level, Windows is a multi threaded OS. And it schedules threads with 100ms time slices. Which means that, if there is no CPU contention, the delay between the end of the buffer and the waveOutWrite callback could be arbitrailly short. Or, if there are other busy threads, you have to wait up to 100ms per thread. In the best case however... CPU speeds clock in at the GHz now. Which puts an absolute lower bound on how fast the callback can be called in the 0.000,000,000,1 second order of magnitude.
Unless you can figure out the maximum number of waveOutWrite callbacks you can process in a single second, which could imply the latency of each call, I think that really, the latency is going to be orders of magnitude below preception most of the time, unless there are too many busy threads, in which case its going to go horribly, horribly wrong.
回答3:
To add to great answers above.
Your question is about the latency Windows neither promised not cared of. And as such, it might be quite different depending on OS version, hardware and other factors. WaveOut API, and DirectSound too (not sure about WASAPI, but I guess it is also true for this latest Vista+ audio API) are all set for buffered audio output. Specific callback accuracy is not required as long as your are on time queuing next buffer while current is still being played.
When you start audio playback, you have a few assumptions such as no underflows during playback and all output is continuous, and audio clock rate is exactly as you expect is, such as 44,100 Hz precisely. Then you do simple math to schedule your wave output in time, converting time to samples and then to bytes.
Sadly, effective playback rate is not precise, e.g. imagine real hardware sampling rate may be 44,100 Hz -3%, and in long run the time-to-byte math might be letting you down. There has been attempt to compensate for this effect, such as making audio hardware the playback clock and synchronizing video to it (this is how players work), and rate matching technique to match incoming data rate to actual playback rate on hardware. Both these make absolute time measurements and latency in question quite a speculative knowledge.
More to this, the API latencies 20 ms, 30 ms, 50 ms and so on. Since long ago waveOut API is a layer on top of other APIs. This means that some processing takes place before data actually reach the hardware and this processing requires that you put your hands off the queued data well in advance, or the data won't reach the hardware. Let's say if you attempt to queue your data in 10 ms buffers right before playback time, the API will accept this data but it will be late itself passing this data downstream, and there will be silence or comfort noise on the speakers.
Now this is also related to callbacks that you receive. You could say that you don't care about latency of buffers and what is important to you is precise callback time. However since the API is layered, you receive callback at the accuracy of inner layer synchronization, such second inner layer notifies on free buffer, and first inner layer updates its records and checks if it can release your buffer too (hey, those buffers don't have to match too). This makes callback accuracy expectations really weak and unreliable.
Provided that I have not been touching waveOut API for quite some time, if such question of synchronization accuracy would come up, I would probably first of all thought of two things:
Windows provides access to audio hardware clock (I am aware of IReferenceClock interface available through DirectShow, and it probably comes from another lower level thing which is also accessible) and having that available I would try to synchronize with it
Latest audio API from Microsoft, WASAPI, provides special support for low latency audio with new cool stuff there like better media thread scheduling, exclusive mode streams and <10 ms latency for PCM - this is where better sync is to be looked at
来源：https://stackoverflow.com/questions/1303811/what-is-the-latency-or-delay-time-for-callbacks-from-the-waveoutwrite-api-meth