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This change is based on the techniques described in [1] and [2]. The input audio stream from Spice is not synchronised to the audio playback device. While the input and output may be both nominally running at 48 kHz, when compared against each other, they will differ by a tiny fraction of a percent. Given enough time (typically on the order of a few hours), this will result in the ring buffer becoming completely full or completely empty. It will stay in this state permanently, periodically resulting in glitches as the buffer repeatedly underruns or overruns. To address this, adjust the speed of the received data to match the rate at which it is being consumed by the audio device. This will result in a slight pitch shift, but the changes should be small and smooth enough that this is unnoticeable to the user. The process works roughly as follows: 1. Every time audio data is received from Spice, or consumed by the audio device, sample the current time. These are fed into a pair of delay locked loops to produce smoothed approximations of the two clocks. 2. Compute the difference between the two clocks and compare this against the target latency to produce an error value. This error value will be quite stable during normal operation, but can change quite rapidly due to external factors, particularly at the start of playback. To smooth out any sudden changes in playback speed, which would be noticeable to the user, this value is also filtered through another delay locked loop. 3. Feed this error value into a PI controller to produce a ratio value. This is the target playback speed in order to bring the error value towards zero. 4. Resample the input audio using the computed ratio to apply the speed change. The output of the resampler is what is ultimately inserted into the ring buffer for consumption by the audio device. Since this process targets a specific latency value, rather than simply trying to rate match the input and output, it also has the effect of 'correcting' latency issues. If a high latency application (such as a media player) is already running, the time between requesting the start of playback and the audio device actually starting to consume samples can be very high, easily in the hundreds of milliseconds. The changes here will automatically adjust the playback speed over the course of a few minutes to bring the latency back down to the target value. [1] https://kokkinizita.linuxaudio.org/papers/adapt-resamp.pdf [2] https://kokkinizita.linuxaudio.org/papers/usingdll.pdf |
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| .. | ||
| audiodev.h | ||
| displayserver.h | ||
| font.h | ||
| overlay.h | ||
| renderer.h | ||