This patch estimates the connection RTT using
EventBasedExponentialMovingAverage. The half time is
set to 500 but can be modified using field trials.
This new metric is currently unused, but will
be used for exploration of of whether it can be used
instead of the existing metric.
Bug: webrtc:11140
Change-Id: I9db93e9b9eb932e3cd18935cd4ce0d90fc1cb293
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161000
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29944}
There are edge cases where the caching of encoder info will cause
issues. For instance if a sub-encoder fails en Encode call and falls
back to some other implementation, or if the fps targets shift due to
SetRates() triggering new layers to be enabled.
This CL forces a complete rebuild on every call to GetEncoderInfo().
It also adds new logging of when the info changes, as debugging issues
can be very time consuming if we can't tell that happened.
Bug: webrtc:11000
Change-Id: I7ec7962a589ccba0e188e60a11f851c9de874fab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29938}
Removed old code, which took care of VP8 case, but actually interferes with
VP9 case.
Now regardless of codec bw_limited_resolution is calculated based on signals
from the bitrate allocator.
Bug: webrtc:11015
Change-Id: Ic99dbb504ab2d1a4b5f15ca93193a1af05ae5924
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160651
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29937}
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 patch makes Connection::port() protected
and add explicit methods for the use cases instead
- network() - port()->Network()
- generation() - port()->generation()
This is done to easier mock a Connection.
BUG=webrtc:10647
Change-Id: I5b35477ed9f81d57cd871072874262d0a8af2d4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160784
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29929}
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}
Many WebRTC users need only Opus, and no other audio codecs. This
makes it convenient for them to do the right thing.
To prove that the new factories work, use them in
PeerConnectionEndToEndTest.
Bug: webrtc:11130
Change-Id: I2c2450ba0fb33ef3b50da8f6cd325cad6b1e59a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160648
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29921}
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 change implements the methods in VideoTrackSourceInterface
that are related to encoded output.
Bug: chromium:1013590
Change-Id: Id9ddbc00a7098e9b44cee1517c69002865a5fb33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159926
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29912}
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}
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}
RtpVideoSenderTest used a SimulatedClock but the task queue factor still
looked at the real-time clock when posting delayed tasks.
This CL changes that so everything is using simulated time, which makes
test faster and should avoid flakiness.
In particular, fixing this timing issue exposed flaws in
DoesNotRetrasmitAckedPackets, which was likely the root case of bug
10873, so let's re-enable on ios again.
Bug: webrtc:10873,webrtc:10809
Change-Id: If8a0c244b1a34f7427543deaa2431ab1e9f124a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160404
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29897}
