On 11.04.2016 21:13, Tanu Kaskinen wrote:
On Fri, 2016-04-08 at 20:23 +0200, Georg Chini wrote:
On 08.04.2016 18:01, Tanu Kaskinen wrote:
On Wed, 2016-04-06 at 21:17 +0200, Georg Chini wrote:
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:
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.
Yes, you explained that already, but you didn't give a convincing
explanation of why the point in time when the delay stops growing would
indicate the point when the first sample hit the DAC.
See below. The precondition for my thoughts naturally is that no
samples vanish from the latency reports, maybe that is where
we are thinking differently.
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.
But the reported delay stopped growing! That's the point where you
claim the first sample hits the DAC, but as my example illustrates,
that doesn't seem to be true.
In your example it is not true, that's right. But for the USB devices it is.
They only start decreasing the delay when real audio has been played,
and they would increase the delay when you write to the buffer,
I have checked that in the code.
And I think any driver that makes samples vanish is so severely screwed,
that we can't do anything about it. If the driver reports complete moonshine
numbers, you can't fix it, I agree with you in that respect.
But that is not the case with USB. There is only some missing latency
that is not reported - call it transport delay or whatever and I suspect a
similar delay can be found in other alsa drivers. There is no need to figure
out the reason for it, it just takes some time after snd_pcm_start() was
called until the first sample is played - without making samples vanish.
And in that case the delay can be detected and used by the code.
I don't understand how it could be possible to have a "transport delay"
that doesn't get reported, if the delay is in the part of the pipeline
that the driver can see. If there's no extra latency reporting
mechanism in the USB audio spec, the driver can see up to the point
where URBs get released. If the driver counts the submitted-but-not-
released URBs, which it appears to do if releasing the first URB
triggers the first decrease in the previously-growing delay reports,
then any transport delay before that point should be automatically
visible in the reported latency. If there's a 10 ms "transport delay",
the driver has to submit at least 10 ms worth of URBs before the first
URB gets released.
I can't imagine a system where the full transport delay doesn't get
included in the delay report, if at the same time the release of the
first URB is the trigger for stopping the delay growth. Either the full
URB queue is taken into account, or none of it is. The transport delay
can't be larger than the URB queue, otherwise there are holes in the
stream.
You make very confident-sounding claims of the URB release notification
happening only after the data in the URB has hit the DAC. But have you
considered that the notification might actually happen immediately
after the USB host controller has sent the data packet to the USB
device? The USB device may then have an additional buffer before the
DAC, so the received audio gets queued and doesn't immediately hit the
DAC. That would explain the permanently too small delay reports, since
the buffer in the USB device isn't included in the reports. In this
case there's no way to know the USB device buffer size without a
separate latency reporting mechanism in the USB audio spec.
You still don't understand. I do not care at all, where the delay comes
from.
It may be for example from pulseaudio starting and stopping the sink because
of sample rate change or any other real-time event that delays playback
after snd_pcm_start() is called.
The point is, that the reported delay does not get lower than than the
write count
before the sample is either played or vanishes into your invisible
buffer. So with my
code you can at least catch any delays that have happened up to that time.
Your invisible buffer would by the way match my "small but constant
remaining
latency offset", so in your initial example I would see an additional
latency offset
of 9ms that my code does not account for, but would still have a better
latency
estimate than the current code delivers.
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 ...
From what I can tell, that's a coincidence.
No, it definitely isn't. If you accept the precondition, that samples
not simply vanish from the latency reports, it's physics.
I'm not sure if you understood what I meant, if you talk about
"vanishing samples" as something crazy that can be assumed to not
happen.
At the end of the pipeline samples always vanish - I already said that there
is a point where your access ends, every delay that happens behind
that point can only be corrected manually as a latency offset. Again
it does not matter where they vanish to - if to the DAC or to an invisible
buffer, at least we can try to look as far ahead as possible.
I tried to illustrate that the driver may see only up to some
point in the full audio pipeline, so before the DAC there may be some
buffer that the driver has no idea of. Samples don't vanish, they just
move from the visible part to the invisible part. A clear example of
this kind of situation is the bluetooth implementation in pulseaudio:
we don't have visibility to the audio pipeline after the socket to
which we write data. We can't query the amount of audio buffered
between the socket and the bluetooth device's DAC (or if there's a
mechanism for that, I'm not aware of it), and we don't get any
notifications of played chunks of audio. Therefore, we can't report
bluetooth latency accurately.
I would tend to agree that I have overlooked something, if the "extra
delay" would be the same every time and if I could not write down
the math for it.
But it isn't completely constant (just in the same range) and I can
write down the math and it matches my measurements. So I am
fairly sure that I am right. Did you have a look at my document?
I've only glanced it so far. It's a big undertaking to understand all
the math, and I haven't had time for that yet.
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.
No, the invisible part is not constant, even though my presentation
didn't show the variance. The DAC consumes data from the invisible
buffer one sample at a time, and each time it does that, the extra
latency decreases by one sample. Data moves from the visible part of
the buffer to the invisible part in bigger chunks. I didn't specify the
chunk size, but if we assume 1 ms chunks, the extra latency grows by 1
ms every time a chunk is transferred from the visible part to the
invisible part.
Then take any part of the buffer but the last or the first bit. All the
chunks are always full, so it's constant. The moving bit is dealt with
elsewhere, (in the smoother) but there is a lot of buffer that is always
full.
And when you take USB, the driver sees only chunks. The sample
by sample consuming of the DAC is never seen by the driver, it gets
the notification from USB that a chunk has been played.
I'm not sure how it is with HDA, but probably similar.
You are probably not talking about my model where there's an invisible
buffer at the end of the pipeline, and since I don't understand what
model you have in mind (see the text earlier in this mail), I can't
really comment here, other than to point out that even when talking
about a buffer that deals with equally sized chunks at both read and
write ends, the buffered amount varies if the reads and writes don't
happen exactly at the same time (each read reduces the buffered amount
and each write increases the buffered amount).
Surely the amount in the buffers vary, that's why I said leave out
the first and the last bit. Everything in between is always full and
makes a constant contribution to the latency.
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.
Not very important, but I'll clarify one thing: I had another look, and
I'm not any more sure that the code where I saw the casting back to a
signed integer is actually used by pulseaudio. The function
is snd_pcm_write_areas(), but pulseaudio doesn't call that at least
directly, and I did some searching in alsa-lib too, and I didn't find a
call path that would cause snd_pcm_write_areas() to be used by
pulseaudio. Even if snd_pcm_write_areas() isn't used, though, it's
entirely possible that there's some other code that does a similar
cast. I don't know the code is that triggers the snd_pcm_start() call
when the ring buffer fill level exceeds the configured threshold. It
might be in the kernel.
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?
Do I understand correctly that your "extra latency" is affected by
whether snd_pcm_start() is called implicitly in mmap_write() or
explicitly after mmap_write()? The time when mmap_write() is called
doesn't affect the latency in the long term.
It does. It isn't much, but if playback starts earlier, the delay
will be exactly that amount less even after 10 hours of playback.
Let's assume you have 10ms of audio to write to the buffer.
During the time, when you write, samples are coming in.
Let's say it takes 100 usec to write the buffer. If you start
playback after the write, this will be 100 usec additional delay.
5 samples have accumulated.
If you start playback immediately after the first bit of data is
written this might take much less time, say 20 usec.
So your delay is four samples less and it will remain that way
until the sink is stopped. There is nothing that would take away
the delay.
Those 5 samples come from a source and accumulate in module-loopback's
memblockq, right? That has no effect on the sink latency, and module-
loopback is able to get rid of excess stuff in the memblockq, so the
initial accumulation has no long-term effects on latency.
That is exactly my point. If the fact, that playback started earlier
than expected would be correctly reported, then module-loopback
could react to it. But with the current code it isn't, because alsa
calculates the delay relative to the start of the playback, but
playback was in fact started earlier than pulseaudio assumes.
The same is valid in the other direction, if the start of the playback
is delayed for any reason.
The smoother will produce
wrong values if it's not started at the same time as snd_pcm_start() is
called, but I presume the smoother is able to fix such inaccuracies
over time, so it doesn't matter that much when the snd_pcm_start() is
called. So isn't it a bad thing if your "extra latency" permanently
includes something that doesn't have any real effect after some time?
Yes, it is affected by it and it should be, because the "extra delay"
is the time between snd_pcm_start() and the first sample being
played. So if the first samples are played before snd_pcm_start()
the "extra latency" will become negative. And as explained above,
it has permanent effect. Somehow you seem to be of the opinion
that all delays that are not controlled by the pulseaudio code
vanish magically, but they don't.
A permanent difference in delay means that there's some buffer
somewhere whose fill level is permanently higher or lower. I don't
think there's any place in the alsa driver where that could happen. The
loopback memblockq could be a place where this could happen, if module-
loopback didn't do adaptive resampling to handle the case where the
memblockq has too much or too little audio buffered.
I don't understand your comment, sorry.
For the reported latency, it just means, that it will become slightly
smaller. As I said, the smoother does not use the "extra delay"
for anything, it is only calculated once when the origin for the
smoother is set and added later as an offset, when get_latency()
is called.
If you ever make the extra delay negative, that means that you think
that alsa reports playback latencies that are higher than reality. Do
you really think that it's possible for the delay values that alsa
reports to be permanently higher than reality?
As explained above, the "higher than reality" or "lower than reality"
latencies may not be caused by wrong delay reports of alsa, but by the
assumption that pulseaudio makes regarding the time when playback
was started. And that is what my code tries to do - get the best possible
value for the "real" start time.
It does not claim to cover all possible latencies anywhere between source
and sink and would not catch an "invisible buffer" but it is much better
than to rely blindly on the reported values like the current code does.
Just one more point regarding those negative latencies - if I allow
the "extra latency" to become negative, the resulting end-to end latency
for HDA -> HDA will be exactly the same if you try it a couple of times,
while if I truncate, the resulting latency will vary by 20 - 60 usec.
I can't believe that is coincidence ...
I can only point you again to my document - I sent you the full version
including the loopback part earlier today and math is unambiguous.
_______________________________________________
pulseaudio-discuss mailing list
[email protected]
https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
