On 06.04.2016 15:55, Tanu Kaskinen wrote:
On Tue, 2016-04-05 at 21:54 +0200, Georg Chini wrote:
On 05.04.2016 17:42, Tanu Kaskinen wrote:
On Sat, 2016-04-02 at 18:13 +0200, Georg Chini wrote:
On 02.04.2016 16:10, Tanu Kaskinen wrote:
On Wed, 2016-03-30 at 22:31 +0200, Georg Chini wrote:
On 30.03.2016 18:06, Tanu Kaskinen wrote:
On Tue, 2016-03-29 at 20:29 +0200, Georg Chini wrote:
5) The pulseaudio sink code takes the first 10ms of audio out of the
loopback buffer,
writes it to the alsa buffer and calls snd_pcm_start().
If the sink takes something from the loopback buffer, this means that
the first pop() call has been made. Assuming no time has passed since
the previous step, the USB bus is still full, and so is the ring
buffer. Expected delay: 20 ms.
Reported delay is exactly the amount of audio that was written to
the buffer.
That's the bug that I think should be fixed in alsa if possible (and if
it's impossible, I don't see how it could be fixed in pulseaudio
either).
It can be fixed (or at least be worked around). If you take a time stamp
at the moment when snd_pcm_start() is called and another when
the first audio has definitely been played (delay < write_count), then
the difference between the time stamps corrected by the amount
of audio that has already been played, gives you exactly that
missing bit of latency.
I can't follow that line of reasoning. In the beginning the ring buffer
is filled to max, and once you call snd_pcm_start(), data starts to
move from the ring buffer to other buffers (I'll call the other buffers
the "not-ring-buffer"). Apparently the driver "sees" the not-ring-
buffer only partially, since it reports a larger latency than just the
ring buffer fill level, but it still doesn't report the full latency.
The time between snd_pcm_start() and the point where the reported delay
does not any more equal the written amount tells the size of the
visible part of the not-ring-buffer - it's the time it took for the
first sample to travel from the ring buffer to the invisible part of
the not-ring-buffer. I don't understand how the time could say anything
about the size of the invisible part of the not-ring-buffer. Your logic
"works" only if the visible and invisible parts happen to be of the
same size.
You should get the same results by calculating
adjusted delay = ring buffer fill level + 2 * (reported delay - ring buffer
fill level)
That formula doesn't make sense, but that's how I understand your logic
works, with the difference that your fix is based on one measurement
only, so it's constant over time, while my formula recalculates the
adjustment every time the delay is queried, so the adjustment size
varies somewhat depending on the granularity at which audio moves to
and from the visible part of the not-ring-buffer.
In any case, even if your logic actually makes sense and I'm just
misunderstanding something, I don't see why the correction should be
done in pulseaudio instead of the alsa driver.
Well, now I don't understand what you mean. The logic is very simple:
If there is a not reported delay between the time snd_pcm_start() is
called and the time when the first sample is delivered to the DAC, then
this delay will persist and become part of the continuous latency.
That's all, what causes the delay is completely irrelevant.
The code can't know when the first sample hits the DAC. The delay
reported by alsa is supposed to tell that, but if the reported delay is
wrong, I don't think you have any way to know the real delay.
Yes, the code can know when the first sample hits the DAC. I explained it
already. Before the first sample hits the DAC, the delay is growing and
larger or equal than the number of samples you have written to the
buffer.
At the moment the delay is smaller than the write count, you can be
sure that at least some audio has been delivered. Since the delay is
decreased by the amount of audio that has been delivered to the DAC,
you can work back in time to the moment when the first sample has been
played.
Maybe what I said above was not complete. At the point in time when
the first audio is played, there are two delays: First the one that is
reported
by alsa and the other is the difference between the time stamps minus
the played audio. If these two delays don't match, then there is an
"extra delay" that has to be taken into account.
The difference between the time stamps is not related to how big the
invisible part of the buffer is. I'll try to illustrate:
In the beginning, pulseaudio has written 10 ms of audio to the ring
buffer, and snd_pcm_start() hasn't been called:
DAC <- ssssssssss|sss|dddddddddd <- pulseaudio
Here "ssssssssss|sss|ddddddddd" is the whole buffer between the DAC and
pulseaudio. It's divided into three parts; the pipe characters separate
the different parts. Each letter represents 1 ms of data. "s" stands
for silence and "d" stands for data. The first part of the buffer is
the invisible part that is not included in the delay reports. I've put
10 ms of data there, but it's unknown to the driver how big the
invisible part is. The middle part of the buffer is the "send buffer"
that the driver maintains, its size is 3 ms in this example. It's
filled with silence in the beginning. The third part is the ring
buffer, containing 10 ms of data from pulseaudio.
At this point the driver reports 10 ms latency. It knows it has 3 ms of
silence buffered too, which it should include in its latency report,
but it's stupid, so it only reports the data in the ring buffer. The
driver has no idea how big the invisible part is, so it doesn't include
it in the report.
Now pulseaudio calls snd_pcm_start(), which causes data to start moving
from the ring buffer to the send buffer. After 1 ms the situation looks
like this:
DAC <- ssssssssss|ssd|ddddddddd <- pulseaudio
There's 2 ms of silence in the send buffer and 1 ms of data. The driver
again ignores the silence in the send buffer, and reports that the
delay is 10 ms, which consists of 1 ms of data in the send buffer and 9
ms of data in the ring buffer.
After 2 ms:
DAC <- ssssssssss|sdd|dddddddd <- pulseaudio
Reported delay: 10 ms
After 3 ms:
DAC <- ssssssssss|ddd|ddddddd <- pulseaudio
Reported delay: 10 ms
Let's say pulseaudio refills the ring buffer now.
DAC <- ssssssssss|ddd|dddddddddd <- pulseaudio
Reported delay: 13 ms
After 4 ms:
DAC <- sssssssssd|ddd|ddddddddd <- pulseaudio
The first data chunk has now entered the invisible part of the buffer,
but it will still take 9 ms before it hits the DAC. At this point
pulseaudio has written 13 ms of audio, and the reported delay is 12 ms.
According to your logic, the adjusted delay is 12 + (4 - 1) = 15 ms,
while in reality the latency is 22 ms.
At this point, no audio has been played yet. You still have silence in the
buffer, so alsa would not report back, that samples have been played.
I choose the point where the first d hits the DAC and that is reported
back by alsa. (see above) I've tried put it all together in a document.
I hope I can finish the part that deals with the smoother code today.
If so, I will send it to you privately because the part about
module-loopback
is still missing.
Anyway, even if you think it is wrong I am still measuring the correct
end-to-end latency with my code, so something I am doing must be
right ...
I don't know how well this model reflects the reality of how the usb
audio driver works, but this model seems like a plausible explanation
for why the driver reports delays equalling the amount of written data
in the beginning, and why the real latency is higher than the reported
latency at later times.
I hope this also clarifies why I don't buy your argument that the time
stamp difference is somehow related to the unreported latency.
No, in fact it doesn't.
Trying to fix up that delay on every iteration does not make any sense
at all, it is there from the start and it is constant.
Commenting on "it is constant": The playback latency is the sum of data
in various buffers. The DAC consumes one sample at a time from the very
last buffer, but I presume that all other places move data in bigger
chunks than one sample. The unreported delay can only be constant if
data moves to the invisible part of the buffering in one sample chunks.
Otherwise the latency goes down every time the DAC reads a sample, and
then when the buffer is refilled at the other end, the latency jumps up
by the refill amount.
I only said the "extra latency" is constant, not the latency as such.
See your own example above that your argument is wrong. Even
if the audio is moved in chunks through your invisible buffer part,
that part still has the same length all the time. When one "d" is
moved forward another one will replace it.
That was what my original question was about - what should I do with
this extra latency? Currently I am just adding it as an offset to the
"normal" latency. This however means, that if you configure let's say
10ms, you will get in fact around 22ms. (You would get 22ms anyway,
but the reports would show 10ms with the old code.)
For HDA the reported delay is even slightly negative, probably because
the card already starts during the preparation step. Negative delays
are truncated by my code, no real audio should have been played
before snd_pcm_start().
A negative delay indicates an underrun according to
http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___status.html#ga1fdce3985e64f66385a5805da1110f18
This is not a negative delay reported by alsa, but my "extra latency"
is getting negative, which means playback must have started
before snd_pcm_start().
According to Raymond Yau playback seems in fact to be started
before snd_pcm_start() for HDA devices, at least if I read his last
mail on that topic right. Then the negative delays would even make
sense, since data is written to the buffer before snd_pcm_start().
I had a look at the code to verify the claim that we configure alsa to
start playback already before we call snd_pcm_start(). If we really do
that intentionally, then it doesn't make sense to call snd_pcm_start()
explicitly.
This is what we do:
snd_pcm_sw_params_set_start_threshold(pcm, swparams, (snd_pcm_uframes_t) -1)
Note the casting of -1 to an unsigned integer. It seems that the
intention is to set as high threshold as possible to avoid automatic
starting. However, alsa-lib casts the threshold back to a signed value
when it's used, and I believe the end result is indeed that playback
starts immediately after the first write. I don't know if that matters,
since we do the manual snd_pcm_start() call immediately after the first
write anyway, but it seems like a bug in any case.
OK, this it why I measure an "extra latency" of -60 to -20 usec.
So again, if I can measure it and even detect a bug that way,
don't you think there must be some truth in what I'm saying?
BTW, do you think the debug output of module-loopback is
better now?
Yes, although if it's logged twice a second, it might be better to
print the status only if explicitly requested via a module argument.
The status is printed once every adjust time and only when debug
logging is enabled. 500ms seems to be a good value for the adjust
time, it is currently my default. If you prefer an additional argument
to enable logging, I can add it.
Yes, I'd prefer that. Continous logging annoys me when it's about
something that I'm not currently interested in. module-loopback isn't
the only thing generating annoying logs, but decreasing the interval
from 10 seconds to 0.5 seconds makes the problem that much worse.
OK, could we do something similar for those "memblock pool full"
messages? Sometimes I get several thousand messages per second
(which are suppressed by the rate limiting) and I cannot see any
relevance.
Have you investigated why there are so many memblocks active that the
pool gets full? It seems fishy to me. The limit could be triggered e.g.
by thousand 1 ms blocks in a 1 second buffer, but why would you do
that?
I have no idea at all. It always happens when you run a sink at
very low latencies and I have observed this also with the code
from git. It is not something that has been introduced by my
changes, so I did not investigate further and just accepted that
there seems to be a lower limit to the sink latency at around 2.3ms
for HDA. Now that USB devices also use timer based scheduling
I am seeing it there as well at around 8ms configured latency.
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