Re: [PATCH 09/10] ALSA: pcm: Add snd_pcm_ops for snd_pcm_link()

From: Timo Wischer
Date: Tue Mar 26 2019 - 11:17:07 EST

On 3/26/19 15:23, Takashi Iwai wrote:
On Tue, 26 Mar 2019 12:25:37 +0100,
Timo Wischer wrote:
On 3/26/19 09:35, Takashi Iwai wrote:

On Tue, 26 Mar 2019 08:49:33 +0100,
<twischer@xxxxxxxxxxxxxx> wrote:
From: Timo Wischer <twischer@xxxxxxxxxxxxxx>
snd_pcm_link() can be called by the user as long as the device is not
yet started. Therefore currently a driver which wants to iterate over
the linked substreams has to do this at the start trigger. But the start
trigger should not block for a long time. Therefore there is no callback
which can be used to iterate over the linked substreams without delaying
the start trigger.
This patch introduces a new callback function which will be called after
the linked substream list was updated by snd_pcm_link(). This callback
function is allowed to block for a longer time without interfering the
synchronized start up of linked substreams.
Signed-off-by: Timo Wischer <twischer@xxxxxxxxxxxxxx>
Well, the idea appears interesting, but I'm afraid that the
implementation is still racy. The place you're calling the new
callback isn't protected, hence the stream can be triggered while
calling it. That is, even during operating your loopback link_changed
callback, another thread is able to start the stream.
Hi Takashi,

As far as I got you mean the following scenario:

* snd_pcm_link() is called for a HW sound card
+ loopback_snd_timer_link_changed()
The start may happen at this point.

In this case the last link status will be used and aloop will print a warning "Another sound timer was requested but at least one device is already running...".

Without this patch set a similar issue already exists. When calling snd_pcm_start() before snd_pcm_link() was done the additional device linked by the snd_pcm_link() will not be started.
Therefore the application has already to take care about the order of the calls.

+ loopback_snd_timer_open()
+ spin_lock_irqsave(&dpcm->cable->lock, flags);
* snd_pcm_start() called for aloop sound card
+ loopback_trigger()
+ spin_lock(&cable->lock) -> has to wait till loopback_snd_timer_open()
calls spin_unlock_irqrestore()

So far snd_pcm_start() has to wait for loopback_snd_timer_open().

* loopback_snd_timer_open() will continue with
+ dpcm->cable->snd_timer.instance = NULL;
+ spin_unlock_irqrestore()
* loopback_trigger() can enter the lock
+ loopback_snd_timer_start() will fail with -EINVAL due to
(loopback_trigger == NULL)

At least this will not result into memory corruption due to race or any other
wired behavior.
I don't expect the memory corruption, but my point is that dealing
with linked streams is still tricky. It was considered for the
lightweight coupled start/stop operation, and something intensively
depending on the linked status was out of the original design...

But my expectation is that snd_pcm_link(hw, aloop) or snd_pcm_link(aloop, hw)
is only called by the application calling snd_pcm_start(aloop)
because the same aloop device cannot be opened by multiple applications at the
same time.

Do you see an use case where one application would call snd_pcm_start() in
parallel with snd_pcm_link() (somehow configuring the device)?
It's not about the actual application usages but rather against the
malicious attacks. Especially aloop is a virtual device that is
available allover the places, it may be deployed / attacked easily.
The attack we are identifying here can only be done by the application opening the aloop device.
An application allowed to open the aloop device is anyway able to manipulate the audio streaming.
Or do you see an attack which would influence any other device/stream not opened by this application?

May be we should add an additional synchronization mechanism in pcm_native.c
to avoid call of snd_pcm_link() in parallel with snd_pcm_start().
If it really matters... Honestly speaking, I'm not fully convinced
whether we want to deal with this using the PCM link mechanism.

What's the motivation for using the linked streams at the first place?
That's one of the biggest missing piece in the whole picture.
In general when the user uses snd_pcm_link() it expects that the linked devices are somehow synchronized.
Any applications already using snd_pcm_link() do not need to be adapted to use the new feature of aloop (for example JACK or ALSA multi plugin)

But when linking a HW sound card and aloop without this patch set, both devices will be started in sync but
the snd_pcm_period_eleapsed() calls of the different devices will drift. To avoid this the aloop plugin can automatically use the right timer.
If this feature is not implemented the user has to use snd_pcm_link() to trigger snd_pcm_start() in sync but also has to configure the aloop plugin to use the right sound timer.
May be the linked cards can change during runtime of the system. Without this feature the aloop kernel driver has to be loaded with different kernel parameters
to select the right timer.

ALSA controls cannot be used easily. Selecting the sound timer by the card number could be error prone because the card ID could change between system starts.
Therefore an ALSA control supporting the card name should be used. This could be for example done via an ALSA enum control. But in this case the names of all sound cards of the system has to be available
on aloop probe() call. But at this point in time the sound cards probed after aloop are not available. Therefore only the sound timers of the sound cards probed before aloop are selectable.