Together with RtpDepacketizer refactoring that would reduce
number of memcpy while handling an rtp packet
Bug: webrtc:11152
Change-Id: I6f4e09c93af5e2a9314967a15eac8ced57ec712e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161087
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29985}
By using the top level VideoCodec maxFramerate, the FrameBufferController
would sometimes not use the intended value for each simulcast layer.
In the case of "conference mode", top level maxFramerate was set to 5,
which matches the lower layer but is different from the overall maximum
maxFramerate which would be 60.
Bug: webrtc:11117
Change-Id: I4e1e68184d32675b083cd8e4e73a5291dc8fa620
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161096
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29982}
This reverts commit 3a77f93589f77aabfdfa75cd7202f09b4315ff7c.
Reason for revert: The change is breaking downstream tests.
Original change's description:
> AEC3: Ensure that the high-pass filter effect is on when AEC3 is active
>
> This CL ensures that the high-pass filter is on whenever the echo
> controller is on. This is important as the echo controller code assumes
> that the external high-pass filter is active.
>
> The CL also corrects the ToggleAec unit test (which started failing
> after this code change).
>
> Bug: webrtc:11159,chromium:1030179
> Change-Id: Ie29db74bf3de6279a08564398d32d67d5e1569db
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161222
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29979}
TBR=saza@webrtc.org,peah@webrtc.org
Change-Id: I78b4e397555f50898ca42c4b32fb39cf06a2541a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11159, chromium:1030179
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161226
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29981}
This CL adds code for doing signal-dependent downmixing
before the delay estimation in the multichannel case.
As part of the CL, the unittests of the render delay
controller are corrected. However, as that caused some of
them to fail, the CL (for now) as well disables the failing
test.
Bug: webrtc:11153,chromium:1029740, webrtc:11161
Change-Id: I0b765c28fa5e547aabd6dfbd24b626ff9a16346f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161045
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29980}
This CL ensures that the high-pass filter is on whenever the echo
controller is on. This is important as the echo controller code assumes
that the external high-pass filter is active.
The CL also corrects the ToggleAec unit test (which started failing
after this code change).
Bug: webrtc:11159,chromium:1030179
Change-Id: Ie29db74bf3de6279a08564398d32d67d5e1569db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161222
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29979}
There is currently a bug in NetEq that causes audio to leak from the
first channel to all others during loss concealment. This CL fixes the
problem and also adds a unit test to verify.
Bug: webrtc:11145
Change-Id: Ia6c4a234ff7f78e9a6080f1cb17eb80af671c3dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161091
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29974}
This adds functionality to AEC3 to produce setup-specific
default configurations that are tailored to work well for the
number of channels at hand.
The tunings are only used for the case when no echo control factory
has been provided.
Bug: webrtc:11151,chromium:1029717
Change-Id: I1bd2d10327300c7b0f3169a52bf66700b781fd6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161086
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29972}
This CL decouples NetEqFactory and AudioDecoderFactory.
AudioDecoderFactory is used in more places than just inside of NetEq, so
decoupling these makes sense.
Bug: webrtc:11005
Change-Id: I78dd856e4248e398e69a65816b062ef30555b055
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161005
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29961}
instead of guessing based on presence of the sequence header OBU.
Bug: webrtc:11042
Change-Id: I9a0674249feceebb07299ea965c62e87499f6baf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161013
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29958}
This CL changes the downmixing of the input to the delay estimation
for surround/stereo signals to be off by default.
A kill-switch is also added for enforcing the downmix to be on.
Bug: webrtc:10913
Change-Id: I1030fef593ba56416deeb13b80d2f3812bffb9ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161012
Commit-Queue: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29951}
This reverts commit 15be5282e91ba38894e6ad51fe9a35a38a6b7f29.
Reason for revert: crbug.com/1028937
Original change's description:
> Add support for RtpEncodingParameters::max_framerate
>
> This adds the framework support for the max_framerate parameter.
> It doesn't implement it in any encoder yet.
>
> Bug: webrtc:11117
> Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29907}
TBR=steveanton@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,orphis@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11117
Change-Id: Ic44dd36bea66561f0c46e73db89d451cb3e22773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160941
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29935}
Some ErrorProne warnings have been enabled by [1], that broke the
Chromium Roll into WebRTC, this CL should have taken care of all the
problems.
[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1935889
Bug: None
Change-Id: I2670e948c320984a122fdb774b891c98e05f582e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160862
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29933}
This CL corrects the analog AGC code so that the levels are properly
aggregated and not only the level of the first channel is chosen.
It also adds a kill-switch to allow the aggrated level to be the maximum
level rather than the minimum level.
Bug: webrtc:10859
Change-Id: Ibf4fecb53cfaf0dc064c334112105bf26401f78d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160708
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29931}
This is a reland of 72e6cb0b3f548900fd3b548b4b6966e3f5ee854f
Was not the cause of perf alert, relanding.
TBR=ilnik@webrtc.org
Original change's description:
> Fixes dynamic mode pacing issues.
>
> This CL fixes a few issues in the (default-disabled) dynamic pacing
> mode:
> * Slight update to sleep timing to avoid short spin loops
> * Removed support for early execution as that lead to time-travel
> contradictions that were difficult to solve.
> * Makes sure we schedule a process call when a packet is due to be
> drained even if the queue is empty, so that padding will start at
> the correct time.
> * While paused or empty, sleep relative last send time if we send
> padding while silent - otherwise just relative to last process
> time.
> * If target send time shifts so far back that packet should have
> been sent prior to the last process, make sure we don't let the
> buffer level remain.
> * Update the PacedSender test to _actually_ use dynamic processing
> when the param says so.
>
> Bug: webrtc:10809
> Change-Id: Iebfde9769647d2390fd192a40bbe2d5bf1f6cc62
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160407
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29911}
Bug: webrtc:10809
Change-Id: Ie7b307e574c2057bb05af87b6718a132d639a416
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160786
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29928}
This CL removes the experimental status of the multi-channel processing
in APM, and accordingly updates the variable naming.
It also splits the activation of multi-channel processing to be separate
for render and capture.
Bug: webrtc:10859
Change-Id: I0e5d04dcb94b6637c33d97146231b8ddddbaea39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160707
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29926}
This CL corrects the re-initialization behavior of the analog
AGC to work correctly when the AGC is reinitialized.
Bug: webrtc:11131
Change-Id: Ie455ba3db1aa3936cbcbb2fab023528124853284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160650
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29924}
This reverts commit 72e6cb0b3f548900fd3b548b4b6966e3f5ee854f.
Reason for revert: Speculative revert due to perf change
Original change's description:
> Fixes dynamic mode pacing issues.
>
> This CL fixes a few issues in the (default-disabled) dynamic pacing
> mode:
> * Slight update to sleep timing to avoid short spin loops
> * Removed support for early execution as that lead to time-travel
> contradictions that were difficult to solve.
> * Makes sure we schedule a process call when a packet is due to be
> drained even if the queue is empty, so that padding will start at
> the correct time.
> * While paused or empty, sleep relative last send time if we send
> padding while silent - otherwise just relative to last process
> time.
> * If target send time shifts so far back that packet should have
> been sent prior to the last process, make sure we don't let the
> buffer level remain.
> * Update the PacedSender test to _actually_ use dynamic processing
> when the param says so.
>
> Bug: webrtc:10809
> Change-Id: Iebfde9769647d2390fd192a40bbe2d5bf1f6cc62
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160407
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29911}
TBR=ilnik@webrtc.org,sprang@webrtc.org
Change-Id: I5d1532d2e041e60a7f1bfeb8185f7760c9789711
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10809
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160701
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29920}
The NetEqFactory is currently expected to wrap the AudioDecoderFactory,
but this turns out not to be a good idea. Instead, it makes more sense
to pass the AudioDecoderFactory through the CreateNetEq method.
Bug: webrtc:11005
Change-Id: I8027ff6593f40c92072e7e88157631dcf329a984
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160644
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29918}
This CL adds a DCHECK for the deprecated 8 kHz rate in APM.
It also updates the agc fuzzer code to properly do band-split on
the signals, and not send 8 kHz signals into the AGC.
Bug: chromium:1028092,chromium:1028172
Change-Id: I1e7c8d721834310e94b0e21efea07f75da837cab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160600
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29914}
This CL fixes a few issues in the (default-disabled) dynamic pacing
mode:
* Slight update to sleep timing to avoid short spin loops
* Removed support for early execution as that lead to time-travel
contradictions that were difficult to solve.
* Makes sure we schedule a process call when a packet is due to be
drained even if the queue is empty, so that padding will start at
the correct time.
* While paused or empty, sleep relative last send time if we send
padding while silent - otherwise just relative to last process
time.
* If target send time shifts so far back that packet should have
been sent prior to the last process, make sure we don't let the
buffer level remain.
* Update the PacedSender test to _actually_ use dynamic processing
when the param says so.
Bug: webrtc:10809
Change-Id: Iebfde9769647d2390fd192a40bbe2d5bf1f6cc62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160407
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29911}
This adds the framework support for the max_framerate parameter.
It doesn't implement it in any encoder yet.
Bug: webrtc:11117
Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29907}
Implements an alternative to the dominant nearend detector.
Bug: b/130016532
Change-Id: If4867d58aad036ccf4e456ef81689b8db0284f7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159865
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29906}
In the unlikely event that the decoded audio is really short, the
downsampling would read outside of the decoded audio vector. This CL
fixes that, and adds a unit test that verifies the fix (when running
with ASan).
Bug: chromium:1016506
Change-Id: Ifb8071ce0550111cd66e7f7c1bed7f17b33f93c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160304
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29898}
Beyond making the digital AGC1 code properly support
multichannel, this CL also
-Removes deprecated debug logging code.
-Converts the gain application to be fully in floating point
which
--Is less computationally complex.
--Does not quantize the samples to 16 bit before applying the
gains.
Bug: webrtc:10859
Change-Id: I6020ba8ae7e311dfc93a72783a2bb68d935f90c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159861
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29886}
The changes have been shown to be bitexact on a large dataset.
Bug: webrtc:10859
Change-Id: Iedc0e9e944ebfabb717dd7fb4d2682c695da883e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159694
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29883}
AECM only supports up to two capture channels, this CL extends it to arbitrary channel counts.
Bug: webrtc:10859
Change-Id: Id56ca633cd9de706fa1254bfa8153de88de0ef70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160340
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29880}
This CL adds proper multi-channel support to the analog AGC.
Beyond that, it prepares adding multi-channel support to the digital
AGC by removing the tight dependency between the analog and digital
AGC codes.
Bug: webrtc:10859
Change-Id: I4414ccbc3db5dbb5ae069fdf426cbd038375ca7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159480
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29878}
Summary of changes/improvements and fixes:
Changes container for list of devices from std::vector to std:deque to
allow fast insertion and deletion at both its beginning and its end. This
approach makes it easier to first build a list of all available devices
and then check the size of the list. If size > 0 => two more devices are
added at the front (Default and Default Communication). The old solution
contained a risk of adding invalid Default and Default Communication
devices in cases where not physical device could be found.
Adds usage of |device_index_| in CoreAudioBase to ensure that the selected
device is unique. The previous version used only an ID but that ID is not
unique when e.g. only one device exists since it can have up to three
different roles.
Improves logging and comments.
No-Try: True
Tbr: thaloun@chromium.org
Bug: webrtc:11107
Change-Id: I9a09f7716ed8d8858dcc6a5354b038fc06496166
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160050
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29874}
Fix signed/unsigned mismatch.
Protect against NumberOfEnumeratedDevices and Get[In|Out]putDeviceNames returning inconsistent results.
It's possible for an device to be counted but getting its name fails, in which case the utility function returns true but would continue from its loop filling the AudioDeviceNames vector, leading to a smaller output than the later code expects.
Bug: b/144382120
Change-Id: Iab008c28f03023c830011d229b1f1c7e3e7bb5ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160226
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29871}
