1844181 - Poor quality in WebRTC audio capture on Linux

Status: RESOLVED FIXED

(Core :: WebRTC: Audio/Video, defect, P3)

Description

[jim3692]

User Agent: Mozilla/5.0 (X11; Linux x86_64; rv:109.0) Gecko/20100101 Firefox/115.0

Steps to reproduce:

We are developing this extension, that allows screensharing with audio on Linux: https://github.com/IceDBorn/pipewire-screenaudio

Actual results:

When getUserMedia creates a new Pipewire Node, it's locked to F32LE format. We suspect that this causes audio pitching in calls, mainly when music is present. When the audio transmitted only contains voices, or sound effects, the quality is perfect. We checked the format on Chromium and it is set to S16LE. They also have different latency values. Firefox has 1200/48000 whereas Chromium has 480/48000.

Expected results:

Both browsers should have transmitted audio with similar quality, while Chromium has crystal clear audio, even with music, Firefox has pitching issues.

Comment 1

[jim3692]

The only place I have seen a discussion related to this topic is this issue on Github: https://github.com/edisionnano/Screenshare-with-audio-on-Discord-with-Linux/issues/12

Comment 2

[BugBot [:suhaib / :marco/ :calixte]]

The Bugbug bot thinks this bug should belong to the 'Core::WebRTC: Audio/Video' component, and is moving the bug to that component. Please correct in case you think the bot is wrong.

Comment 3

[Jan-Ivar Bruaroey [:jib] (needinfo? me)]

We are developing this extension, that allows screensharing with audio on Linux ...

When getUserMedia creates a new Pipewire Node, it's locked to F32LE format. We suspect that this causes audio pitching in calls, mainly when music is present.

Do you mean getDisplayMedia? getUserMedia is for camera and microphone.

Comment 4

[github.envenomed]

He meant getDisplayMedia, I'm another developer of the project and I can confirm by looking at the source code.

Comment 5

[jim3692]

(In reply to Jan-Ivar Bruaroey [:jib] (needinfo? me) from comment #3)

We are developing this extension, that allows screensharing with audio on Linux ...

When getUserMedia creates a new Pipewire Node, it's locked to F32LE format. We suspect that this causes audio pitching in calls, mainly when music is present.

Do you mean getDisplayMedia? getUserMedia is for camera and microphone.

Sorry for the confusion. We override the getDisplayMedia method in order to inject the audio from the pipewire-screenaudio node. To do that, we use getUserMedia to find that node.

Comment 6

[Paul Adenot (:padenot)]

Can someone get us ways to reproduce or audio examples of this happening? Should we install the extension or something?

Comment 7

[kowalski]

(In reply to Paul Adenot (:padenot) from comment #6)

Can someone get us ways to reproduce or audio examples of this happening? Should we install the extension or something?

I have samples from when the bug was discovered (plus samples of the original music for comparison). https://mega.nz/file/dK4mXQqL#quxhbB_sS6ukf5Qqyi6iGHobcCepKg2ycBYxV7RDthE

Comment 8

[jim3692]

(In reply to Paul Adenot (:padenot) from comment #6)

Can someone get us ways to reproduce or audio examples of this happening? Should we install the extension or something?

You will need to install the extension from Addons, and optimally give it access to all websites, in case the call is in an iframe: https://addons.mozilla.org/en-US/firefox/addon/pipewire-screenaudio/

For its native part, which controls Pipewire you can either:

1. Install our AUR package on Arch: https://aur.archlinux.org/packages/pipewire-screenaudio
   And choose the audio source to stream from within the extension

2. Or you can manually create the capturing node with this command:
   pw-cli create-node adapter "{ factory.name=support.null-audio-sink node.name=pipewire-screenaudio media.class=Audio/Source/Virtual object.linger=1 audio.position=[FL,FR] }"
   And connect the inputs of the " Pipewire Screenaudio" node to a media player using Helvum

After selecting or connecting your media, you can start your screen share.

You can try using Spotify for the test. We saw that setting its quality to High, made the pitching much more noticeable than when it was set to Normal. But you can also try YouTube videos with and without music.

Comment 9

[Paul Adenot (:padenot)]

This is a bug in our drift compensator. An easier way to repro:

Run Helvum, make a new device like instructed:

pw-cli create-node adapter "{ factory.name=support.null-audio-sink node.name=pipewire-screenaudio media.class=Audio/Source/Virtual object.linger=1 audio.position=[FL,FR] }"

Play something in a media player, connect it to this new device using Helvum.

Open:
https://paul.cx/public/testing-multi-gum.html

Open a regular mic, and then open the new device, that should play what the media player plays.

Essentially what happens is that the device buffer size isn't the same as the graph, and everything breaks.

Comment 10

[Paul Adenot (:padenot)]

Attachment: Bug 1844181 - Move implementation of AudioDriftCorrection into a cpp file, create a dedicated log module, add some logging. r?#media-playback-reviewers

This is not a functional change, only a small refactoring.

Comment 11

[github.envenomed]

Is there a workaround we can use on our extension until this bug gets fixed?

Comment 12

[Paul Adenot (:padenot)]

(In reply to github.envenomed from comment #11)

Is there a workaround we can use on our extension until this bug gets fixed?

I'm not sure. Opening the mic in second instead of first should do the trick, but maybe it's not practical. We'll get this fixed.

Comment 13

[github.envenomed]

Opening the mic in second instead of first should do the trick

What do you mean by that?

Comment 14

[Paul Adenot (:padenot)]

Do the capture first, then the microphone.

But we have new code for this that is getting ready, there should be something to try in a few days.

Comment 15

[Paul Adenot (:padenot)]

Do the capture first, then the microphone.

But we have new code for this that is getting ready, there should be something to try in a few days.

Comment 16

[github.envenomed]

Watching a stream using Firefox 117.0b3 seems to fix the issue. We tried watching the stream with Firefox ESR and the audio drift bug was present. Did you fix the bug in 117?

Comment 17

[Andreas Pehrson [:pehrsons]]

No. Maybe you end up with different latencies on 117?

This try build is of my WIP work. You could try the linux x64 shippable build to see how it fares.

Comment 18

[github.envenomed]

It works, the audio pitch is gone!

Comment 19

[BugBot [:suhaib / :marco/ :calixte]]

There is an r+ patch which didn't land and no activity in this bug for 2 weeks.
:pehrsons, could you have a look please?
If you still have some work to do, you can add an action "Plan Changes" in Phabricator.
For more information, please visit BugBot documentation.

Comment 20

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Add gtest for transmitter block size that is larger than the receiver block size. r?padenot

Comment 21

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Add gtests for block sizes that are larger than the desired buffer size. r?padenot

Comment 22

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Add gtest for changing transmitter block size dynamically. r?padenot

Comment 23

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Allow to differentiate different sources of audio callbacks when tracing. r?padenot

Enables differentiating audio callback sources when tracing a two-mic use case
together with https://github.com/padenot/fx-profiler-audio-cb/pull/3.

Comment 24

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Trace audio callbacks in AudioInputSource. r?padenot

Comment 25

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Move drift control classes to a dedicated dir. r?padenot

This will help keep things together when in a later patch we add a plotting
script and instructions for using it.

Comment 26

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Split driftcontrol files into .h and .cpp, dedicated per-class. r?padenot

Comment 27

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Add support for plotting data from ClockDrift. r?padenot

Comment 28

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Add WavDumper to DynamicResampler as a diagnosis tool. r?padenot

Comment 29

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - In DynamicResampler include first audio input segment in pre-buffer. r?padenot

The pre-buffer has the same size as the target buffer size we aim towards
keeping constant. If the first input segment is large the total amount buffered
is significantly higher than the target buffer size, which leads to us applying
the minimum drift correction (0.9) to shrink it back to the target buffer size
eventually. This can be avoided by accounting for the first input segment when
adding pre-buffering silence.

Comment 30

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Allow to drain DynamicResampler of all remaining input. r?padenot

Comment 31

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Update clock states prior to calculating correction. r?padenot

In order to do the correction calculations on fresh data.

Comment 32

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Calculate drift correction on a callback where there's input data present. r?padenot

This guarantees that our measurement of the current buffering level is fairly
accurate, since the reading of the input buffering level is recent.

This is especially important in situations where the input stream latency
(callback buffer size) is significantly higher than the output stream latency
which drives the drift correction.

Comment 33

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Add some step response gtests for AudioDriftCorrection. r?padenot

These help verify the functionality of the drift correction. Especially useful
together with the data logging and plotting, see comments on the test cases.

Comment 34

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Undef some media LOG macros properly. r?padenot

Comment 35

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Use a logging macro in ClockDrift. r?padenot

Comment 36

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Measure and plot input and output latency. r?padenot

Comment 37

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Use a PID-controller to correct drift. r?padenot

PID-controllers are a proven way to control a system by only measuring an
indirect output of them.

This PID-controller is tuned by hand by looking at step responses
(see TestDriftController.cpp) and applying, in order, the P, D and I terms.

The tuning was done on top of later patches in this bug, so may not be ideal
when looking at this patch in isolation.

Comment 38

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Implement AudioRingBuffer::PrependSilence. r?padenot

This can be handy when AudioDriftCorrection resamples the first time.

In order to calculate how many frames to prepend in order to get the buffer to
contain exactly the number of desired pre-buffer frames after resampling, so
the DriftController starts out with an error = 0.

Comment 39

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Prepend silence to the pre-buffer on first resample. r?padenot

This allows the drift correction code to start out with an error of 0, avoiding
starting out with a significant correction.

Comment 40

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Make it possible to resize AudioRingBuffer on the fly. r?padenot

This lets AudioDriftCorrection handle situations where it's about to overflow
the buffer. Be it because of unexpectedly high drift, very high input latency or
an increased target buffering level.

Comment 41

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Allocate DynamicResampler's internal in-buffer dynamically. r?padenot

Comment 42

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Drive drift corrections by target clock only. r?padenot

Comment 43

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Track/plot/test AudioDriftCorrection's current buffer size. r?padenot

Comment 44

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Increase drift correction's desired buffering level dynamically. r?padenot

This patch allows setting the desired buffering level on AudioDriftCorrection
dynamically. If the AudioResampler's pre-buffer has not yet been set, it will
be populated to match the new desired buffering level on the next Resample().

Comment 45

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Seed the desired buffering level with input stream latency. r?padenot

Note that this patch forwards the latency from CubebInputStream which can be
queried prior to starting, and we are therefore guaranteed to be able to set it
on the drift corrector prior to the first input callback.

CrossGraphPort is more complex with an entire MediaTrackGraph and its
GraphDriver behind it, and is left out at this time.

Comment 46

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Handle drift correction underruns. r?padenot

This patch handles an underrun to the drift correction input buffer by:

• Resetting the pre-buffer in the AudioResampler, so on the next Resample(),
  it will prepend silence to match the desired buffering level.
• Doubling the desired buffering level, since apparently it was not enough.

Overall this will make the perceived discontinuity longer than if we just output
the underrun duration worth of silence. There should also be fewer
discontinuities because of this, likely just one.

Comment 47

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Impose a cap on the integral term's steps. r?padenot

This is so that the integral term does not grow unboundedly while the control
signal is hitting its cap, as it would lead to increasing overshoot and
instability.

Comment 48

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Apply hysteresis to the drift controller. r?padenot

Having a hysteresis threshold on the error signal significantly reduces the
frequency at which we reconfigure the AudioResampler as a response to changes
in the control signal.

Comment 49

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Remove comments about threading from AudioDriftCorrection since it's single threaded. r?padenot

Comment 50

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Add a CeilingPolicy to TimeUnit. r?padenot

For allowing rounding upwards.

Comment 51

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Use TimeUnit across drift control classes. r?padenot

For safer conversions between time bases, of which drift control has plenty.

Comment 52

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Remove commented-out variables from TestAudioResampler.cpp. r?padenot

Comment 53

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Fix an assertion in AudioBufferUtils. r?padenot

This prevents iterating MediaTrackGraph 0 frames.

Comment 54

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Fix a log message in CrossGraphTransmitter. r?padenot

CrossGraphTransmitter does not have an mSegment so its duration is meaningless.

Comment 55

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Improve some fallback driver log messages. r?padenot

There are not many statements identifying fallback wrappers in MediaTrackGraph:5
logs. Log the owning AudioCallbackDriver pointer instead.

Comment 56

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - In MockCubeb add a manual mode. r?padenot

Manual mode means it is up to the test to manually call the DataCallback.
With manual mode there will be no background thread spawned to run callbacks.
This can help make unittests that use MockCubeb more deterministic.

Comment 57

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - In AudioVerifier allow a discontinuity to stretch across multiple samples. r?padenot

When a sudden (one high sample to a zero sample) discontinuity is fed to the
speex resampler, this will typically be smoothed out over multiple samples in
the output. Before this patch AudioVerifier only cares about the difference
between two adjacent samples when counting discontinuity.

This patch makes AudioVerifier account for several discontinuous samples in a
single discontinuity. I.e., once a discontinuity is detected, AudioVerifier
waits for a handful of samples to pass or for the discontinuity to go away,
whichever comes first, before being eligible for detecting another
discontinuity.

Comment 58

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Use MockCubeb's manual mode and tighten expectations in TestCrossGraphPort. r?padenot

Comment 59

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Improve comparison of arrays in TestAudioInputSource. r?padenot

Comment 60

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Reduce the desired buffering level automatically. r?padenot

Comment 61

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Fix channel asserts in DynamicResampler. r?padenot

Comment 62

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Remove helper for rudimentary logging of audio frames in TestAudioDriftCorrection. r?padenot

Comment 63

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Make testAudioCorrection and testMonoToStereoInput more accurate on amount of drift generated. r?padenot

This also makes testMonoToStereoInput more similar to testAudioCorrection, since
they are essentially two variants of the same test, and fixes a test expectation.

Comment 64

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Fix DynamicResampler's handling of channel count changes. r?padenot

It must not store more data than was consumed by the resampler in the tail
buffer.

Comment 65

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Merge testMonoToStereoInput into testAudioCorrection. r?padenot

Comment 66

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Remove the media.clockdrift.buffering pref. r?padenot

Comment 67

[Andreas Pehrson [:pehrsons]]

This is ready, except for a data race found by tsan (roughly 20% failure rate) that we don't yet understand.

Comment 68

[Andreas Pehrson [:pehrsons]]

Attachment: Bug 1844181 - Add memory synchronization to SPSCQueue's thread id reset methods. r?padenot!, r?decoder!

Without this, the necessary synchronization must be provided externally.

This fixes the memory order in the following case of changing producer thread:

• Thread A does SPSCQueue::Enqueue
  • non-atomic write into the ring buffer, at memory location X
  • mWriteIndex.load(relaxed)
  • mWriteIndex.store(release)
• Producer thread is switched to B, no external memory order synchronization is
  provided, but thread B is guaranteed to run after thread A has finished its
  Enqueue task.
• Thread B does SPSCQueue::Enqueue
  • mWriteIndex.load(relaxed)
  • mWriteIndex.store(release)
• Thread C does SPSCQueue::Dequeue
  • mWriteIndex.load(acquire)
  • non-atomic read from the ring buffer, at memory location X

In this scenario, there is no memory synchronization between threads A and B,
and therefore the non-atomic read on C is a data race, and flagged as such by
TSAN.

A similar scenario can be applied to changing the consumer thread, if first A
enqueues, then B dequeues, then C dequeues. However, since Dequeue doesn't
necessarily (MoveOrCopy) do non-atomic writes to the ring buffer, and more
importantly, since Enqueue doesn't do non-atomic reads from the ring buffer,
this is less of a problem.

Comment 69

[Pulsebot]

Pushed by [e-mail]:
https://hg.mozilla.org/integration/autoland/rev/f3d52a166497
Add gtest for transmitter block size that is larger than the receiver block size. r=padenot
https://hg.mozilla.org/integration/autoland/rev/07f593daa504
Add gtests for block sizes that are larger than the desired buffer size. r=padenot
https://hg.mozilla.org/integration/autoland/rev/b3ab4f7079a6
Add gtest for changing transmitter block size dynamically. r=padenot
https://hg.mozilla.org/integration/autoland/rev/3995617f816a
Allow to differentiate different sources of audio callbacks when tracing. r=padenot
https://hg.mozilla.org/integration/autoland/rev/c1ab79a9ac72
Trace audio callbacks in AudioInputSource. r=padenot
https://hg.mozilla.org/integration/autoland/rev/30f59a0c95f2
Move drift control classes to a dedicated dir. r=padenot
https://hg.mozilla.org/integration/autoland/rev/0bce9535589c
Split driftcontrol files into .h and .cpp, dedicated per-class. r=padenot
https://hg.mozilla.org/integration/autoland/rev/6f25e0f2a000
Add support for plotting data from ClockDrift. r=padenot
https://hg.mozilla.org/integration/autoland/rev/dfc424bf89e4
Add WavDumper to DynamicResampler as a diagnosis tool. r=padenot
https://hg.mozilla.org/integration/autoland/rev/d5b79e973899
In DynamicResampler include first audio input segment in pre-buffer. r=padenot
https://hg.mozilla.org/integration/autoland/rev/42f435429f25
Allow to drain DynamicResampler of all remaining input. r=padenot
https://hg.mozilla.org/integration/autoland/rev/6197b1172cdc
Update clock states prior to calculating correction. r=padenot
https://hg.mozilla.org/integration/autoland/rev/1538a5bea488
Calculate drift correction on a callback where there's input data present. r=padenot
https://hg.mozilla.org/integration/autoland/rev/60880eedb9de
Add some step response gtests for AudioDriftCorrection. r=padenot
https://hg.mozilla.org/integration/autoland/rev/39e5ea5b7c9a
Undef some media LOG macros properly. r=padenot
https://hg.mozilla.org/integration/autoland/rev/ed07e8de5754
Use a logging macro in ClockDrift. r=padenot
https://hg.mozilla.org/integration/autoland/rev/541cfe3da283
Measure and plot input and output latency. r=padenot
https://hg.mozilla.org/integration/autoland/rev/907c917b52fe
Use a PID-controller to correct drift. r=padenot
https://hg.mozilla.org/integration/autoland/rev/c4f57f87e34d
Implement AudioRingBuffer::PrependSilence. r=padenot
https://hg.mozilla.org/integration/autoland/rev/15af59e306c6
Prepend silence to the pre-buffer on first resample. r=padenot
https://hg.mozilla.org/integration/autoland/rev/68d05ed50d8a
Make it possible to resize AudioRingBuffer on the fly. r=padenot
https://hg.mozilla.org/integration/autoland/rev/d448783c33e4
Allocate DynamicResampler's internal in-buffer dynamically. r=padenot
https://hg.mozilla.org/integration/autoland/rev/5cdcf42ebc69
Drive drift corrections by target clock only. r=padenot
https://hg.mozilla.org/integration/autoland/rev/89c0108a7c3a
Track/plot/test AudioDriftCorrection's current buffer size. r=padenot
https://hg.mozilla.org/integration/autoland/rev/2e257850dc13
Increase drift correction's desired buffering level dynamically. r=padenot
https://hg.mozilla.org/integration/autoland/rev/1e1f3831e261
Add memory synchronization to SPSCQueue's thread id reset methods. r=decoder,padenot,handyman
https://hg.mozilla.org/integration/autoland/rev/d0497551c605
Seed the desired buffering level with input stream latency. r=padenot
https://hg.mozilla.org/integration/autoland/rev/f4195ac81452
Handle drift correction underruns. r=padenot
https://hg.mozilla.org/integration/autoland/rev/8484b3a81291
Impose a cap on the integral term's steps. r=padenot
https://hg.mozilla.org/integration/autoland/rev/b06a6636ac90
Apply hysteresis to the drift controller. r=padenot
https://hg.mozilla.org/integration/autoland/rev/7b797b4dd354
Remove comments about threading from AudioDriftCorrection since it's single threaded. r=padenot
https://hg.mozilla.org/integration/autoland/rev/fbbfe1de3c65
Add a CeilingPolicy to TimeUnit. r=padenot
https://hg.mozilla.org/integration/autoland/rev/25757a9dd4ca
Use TimeUnit across drift control classes. r=padenot
https://hg.mozilla.org/integration/autoland/rev/1aa8fb4ee69c
Remove commented-out variables from TestAudioResampler.cpp. r=padenot
https://hg.mozilla.org/integration/autoland/rev/706c4902fd36
Fix an assertion in AudioBufferUtils. r=padenot
https://hg.mozilla.org/integration/autoland/rev/5c577d284800
Fix a log message in CrossGraphTransmitter. r=padenot
https://hg.mozilla.org/integration/autoland/rev/e0ab2775ca26
Improve some fallback driver log messages. r=padenot
https://hg.mozilla.org/integration/autoland/rev/333b7208d6d7
In MockCubeb add a manual mode. r=padenot
https://hg.mozilla.org/integration/autoland/rev/e2e5c96f32e4
In AudioVerifier allow a discontinuity to stretch across multiple samples. r=padenot
https://hg.mozilla.org/integration/autoland/rev/630a12c1de0e
Use MockCubeb's manual mode and tighten expectations in TestCrossGraphPort. r=padenot
https://hg.mozilla.org/integration/autoland/rev/18c8184118db
Improve comparison of arrays in TestAudioInputSource. r=padenot
https://hg.mozilla.org/integration/autoland/rev/da112e1234e5
Reduce the desired buffering level automatically. r=padenot
https://hg.mozilla.org/integration/autoland/rev/dd7b8d38431e
Fix channel asserts in DynamicResampler. r=padenot
https://hg.mozilla.org/integration/autoland/rev/f103c687dfe1
Remove helper for rudimentary logging of audio frames in TestAudioDriftCorrection. r=padenot
https://hg.mozilla.org/integration/autoland/rev/61d2120b3d48
Make testAudioCorrection and testMonoToStereoInput more accurate on amount of drift generated. r=padenot
https://hg.mozilla.org/integration/autoland/rev/c86d8d07927a
Fix DynamicResampler's handling of channel count changes. r=padenot
https://hg.mozilla.org/integration/autoland/rev/e7b747380a92
Merge testMonoToStereoInput into testAudioCorrection. r=padenot
https://hg.mozilla.org/integration/autoland/rev/c406e2e8f277
Remove the media.clockdrift.buffering pref. r=padenot

Comment 70

[Norisz Fay [:noriszfay]]

https://hg.mozilla.org/mozilla-central/rev/f3d52a166497
https://hg.mozilla.org/mozilla-central/rev/07f593daa504
https://hg.mozilla.org/mozilla-central/rev/b3ab4f7079a6
https://hg.mozilla.org/mozilla-central/rev/3995617f816a
https://hg.mozilla.org/mozilla-central/rev/c1ab79a9ac72
https://hg.mozilla.org/mozilla-central/rev/30f59a0c95f2
https://hg.mozilla.org/mozilla-central/rev/0bce9535589c
https://hg.mozilla.org/mozilla-central/rev/6f25e0f2a000
https://hg.mozilla.org/mozilla-central/rev/dfc424bf89e4
https://hg.mozilla.org/mozilla-central/rev/d5b79e973899
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https://hg.mozilla.org/mozilla-central/rev/ed07e8de5754
https://hg.mozilla.org/mozilla-central/rev/541cfe3da283
https://hg.mozilla.org/mozilla-central/rev/907c917b52fe
https://hg.mozilla.org/mozilla-central/rev/c4f57f87e34d
https://hg.mozilla.org/mozilla-central/rev/15af59e306c6
https://hg.mozilla.org/mozilla-central/rev/68d05ed50d8a
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https://hg.mozilla.org/mozilla-central/rev/dd7b8d38431e
https://hg.mozilla.org/mozilla-central/rev/f103c687dfe1
https://hg.mozilla.org/mozilla-central/rev/61d2120b3d48
https://hg.mozilla.org/mozilla-central/rev/c86d8d07927a
https://hg.mozilla.org/mozilla-central/rev/e7b747380a92
https://hg.mozilla.org/mozilla-central/rev/c406e2e8f277

Comment 71

[Paul Adenot (:padenot)]

Reporter, do you mind testing your add-on again in a Firefox Nightly? Please make sure that it's up-to-date since this just got merged, it's only on the very last version.

This is set to release to the general public in December 5th.

Comment 72

[jim3692]

Hello, we have tested with Firefox Nightly and it looks like it is fixed.
Thank you very much for your time and support.