Hi everybody!
Although I looked at quite a bit of source code and documentation I have to admit that I am confused about one question:
How should alsa playback be stopped in a way that _guarantees_ that no artifacts remaining in the drivers ringbuffer are played back again?
Let´s assume we play stereo 16bit audio data using the rme96 driver, period size set to 0x800 frames.
At the end of the penultimate frame playing will not be stopped even if there is only one frame of audio data remaining. Obviously that means 0x7ff frames of already played data is played again after that one valid frame, plus possibly a few that are played during the short time from interrupt request to the real stop of the playback hardware (about 5 frames on an old Pentium-M 1.2GHz).
Now that could be fixed in different ways:
A: Always ensure that there is a bit of silence (at least hardware buffer size plus a few frames) at the end of all audio data.
Easy for the user , but only a workaround.
B: At the rme96 driver level that condition can be detected and the relevant part of the buffer can be filled with zeros.
I did that in an experimental change. It does work, but it´s definitely only a hack as it fixes the problem for only one device.
C: The application playing audio data could pad all audio data to the period size with zeros.
That would prevent playback of that max 0x7ff frames of garbage, but the few samples that are played during the time from interrupt request to the execution of the stop trigger do escape
D: Don´t try to drain the buffer but pad with zeros and set a stop threshold.
Workaround for applications should also work with older kernels.
E: The alsa core could detect that condition and solve it equivalent to solution D
I think that´s the proper solution.
Did I miss something or do you agree?
cu, Knut