On Donnerstag, 6. Januar 2022 10:21:47 CET Volker Rümelin wrote:
This patch series reduces the playback latency for audio backends,
in some cases significantly. For PulseAudio, the audio buffer is
also moved from the QEMU side to the PulseAudio server side. This
improves the drop-out safety for PulseAudio.
I actually measured the latency reduction with the PulseAudio
backend. For the test I used my Linux host configured to play
audio with PulseAudio. The guest was a Linux guest, also
configured to use PulseAudio.
I haven't reviewed all the patches yet, but from what I read so far, does that
mean the additional 3rd buffer is solely for PulseAudio, so for JACK and other
backends these changes would overall be a degradation, wouldn't they?
No, nothing changes for JACK and it's an improvement for all the other
backends where I added a buffer_get_free function. The important changes
are in [PATCH 10/15] audio: restore mixing-engine playback buffer size.
That patch tries to keep the mixing-engine buffer empty at the end of
audio_run_out().
I couldn't reduce the playback latency for JACK, because the JACK audio
buffers are already very small and any further reduction introduces
playback glitches on my system.
For PulseAudio there is no additional buffer. I only increased the size
of the server side buffer from 15ms to 46,4ms and added a
buffer_get_free function. Before this patch series a few ten ms after
playback started the mixing-engine buffer was full which added 2 *
46,4ms to the playback latency. With these patches the mixing-engine
buffer is empty. This looks like the buffer in use was moved from the
mixing-engine to the PulseAudio server side.
Measuring audio latencies is difficult. I played a sine tone in
the guest with Audacity and measured the time from releasing the
left mouse button until the tone can be heard. A few seconds
before the measurement I started playback of an audio file with
10 minutes of silence to fill the audio buffers. The over-all
latency can't be used to estimate the playback latency, but it
can be used to calculate the playback latency reduction.
The measured over-all latency with PulseAudio is around 200ms
without these patches and around 135ms with these patches. The
difference of 65ms agrees well with the expected value of
46.4ms * 2 + 15ms - 46.4ms = 61.4ms. 46.4ms * 2 is the size of
the mixing-engine buffer ("[PATCH 14/15] paaudio: fix samples vs.
frames mix-up" explains the factor 2), 15ms is the server side
PulseAudio buffer size used before these patches and 46.4ms is
the new server side PulseAudio buffer size.
Volker Rümelin (15):
audio: replace open-coded buffer arithmetic
audio: move function audio_pcm_hw_clip_out()
audio: add function audio_pcm_hw_conv_in()
audio: inline function audio_pcm_sw_get_rpos_in()
paaudio: increase default latency to 46ms
jackaudio: use more jack audio buffers
audio: copy playback stream in sequential order
audio: add pcm_ops function table for capture backend
audio: revert tests for pcm_ops table
audio: restore mixing-engine playback buffer size
paaudio: reduce effective playback buffer size
dsoundaudio: reduce effective playback buffer size
ossaudio: reduce effective playback buffer size
paaudio: fix samples vs. frames mix-up
sdlaudio: fix samples vs. frames mix-up
audio/alsaaudio.c | 1 +
audio/audio.c | 194 ++++++++++++++++++++++++--------------------
audio/audio_int.h | 9 +-
audio/coreaudio.c | 13 +--
audio/dsoundaudio.c | 30 ++++---
audio/jackaudio.c | 5 +-
audio/noaudio.c | 1 +
audio/ossaudio.c | 17 +++-
audio/paaudio.c | 49 ++++++-----
audio/sdlaudio.c | 21 +++--
audio/wavaudio.c | 1 +
11 files changed, 199 insertions(+), 142 deletions(-)
