This CL removes a check in RoundRobinPacketQueue::FinalizePop. This
check will trigger if a the pause state is changed in PacedSender while
a packet is sent. This is a rare occurrence but would yield flaky
behavior. The check should not be required for the code to function
since the paused state is not read in FinalizePop other than for this
check.
Bug: webrtc:9586
Change-Id: Ib9476168eb637dc2f9710d0592bed92c4b03dacb
Reviewed-on: https://webrtc-review.googlesource.com/92090
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24197}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I118156a4f9b00d8c4c4f199a5af50c494e31c34a
Reviewed-on: https://webrtc-review.googlesource.com/89343
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24020}
This replaces the old AlrDetector used by the pacer with the one in
GoogCC. This reduces the risk of accidentally changing only one version.
Note that the pacer instance will be removed when moving over to the
task queue based send side congestion controller.
Bug: webrtc:8415
Change-Id: Id4b2000ee5a04b94565092c29a84572a7750d2f5
Reviewed-on: https://webrtc-review.googlesource.com/85363
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23791}
This CL adds a field trial that disables the feature that the pacer will
ignore the pacing rate and send extra fast to drain the queues if the
pacer queue starts to fill up. BBR assumes that the pacing rate will be
respected and sending more increase the risk of overestimating the
bandwidth.
Bug: webrtc:8415
Change-Id: Ibba315360dafef1c317d14a83199172f9f8cc6aa
Reviewed-on: https://webrtc-review.googlesource.com/80964
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23773}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script from modules with parameters
'pacing video_coding congestion_controller remote_bitrate_estimator':
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118
Reviewed-on: https://webrtc-review.googlesource.com/83900
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23640}
This CL adds a field trial to ensure that audio packets are only blocked
if they are also accounted for. Without the field trial active, audio
packets are blocked due to full congestion windows and media budget
overuse caused by video packets, even it the audio is not accounted for.
Bug: webrtc:8415
Change-Id: I64c3507fcc6e91e6b0759e5f97b34d7f99492658
Reviewed-on: https://webrtc-review.googlesource.com/81187
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23635}
This can be used to avoid getting stuck in a state where the encoder
is paused due to low bandwidth estimate which means no additional
feedback is received to update the bandwidth estimate. This could
happen if feedback packets are lost.
Bug: webrtc:8415
Change-Id: I59cd60c0277e8b31a6b911b25e8e488af9008fc2
Reviewed-on: https://webrtc-review.googlesource.com/80880
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23632}
This fixes an issue where the media budget in the pacer was allowed to
increase more than the process interval when congested.
Bug: webrtc:8415
Change-Id: I79bf965b6a72ed88313074cdae4746fcaff63340
Reviewed-on: https://webrtc-review.googlesource.com/80121
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23531}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
Currently we prefer the last added rtp module that supports rtx, and
assume this is the HD stream.
If we suffer a network degradation and stop sending HD, the current
behavior will trigger RTX padding on an inactive stream, which is not
very useful.
With this change, we will prefer the rtp module that last sent media,
which will spread the load a bit across active media streams, but will
be biased toward the one with highest packet rate.
Bug: webrtc:8975
Change-Id: Id52865ccd5263722c66d327b8c80457f63b90385
Reviewed-on: https://webrtc-review.googlesource.com/77360
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23281}
This removes the redundant type and replaces all usages. A slight change
in behavior is that we no longer get nanosecond resolution. This should
not matter since no current code requires nanosecond resolution.
Bug: webrtc:9155
Change-Id: I04334e08c686d95731621a6c8a7e40400d0ae3b2
Reviewed-on: https://webrtc-review.googlesource.com/71163
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23174}
The paced sender did not update the time out clock before the first
packet was send in paused state. This caused it to incorrectly log
warnings about elapsed time. This CL fixes this.
Bug: None
Change-Id: I240d169464a708c12eb580d57bc385330b8dd6b1
Reviewed-on: https://webrtc-review.googlesource.com/69561
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22852}
This adds the ability to the pacer to apply a congestion window by
tracking sent data. This makes it more reliable when the congestion
window is small enough to be filled at a high rate as there are less
thread context switches that might affect the timing and performance.
Outstanding data is not reduced by the pacer as it has no information
about acknowledged packet feedback. This is by design as the pacer would
also need to keep track of on which connection packets were sent or
received, requiring a larger, more complex, change to the pacer.
Bug: webrtc:8415
Change-Id: I4ecd303e835552ced042cd21186da910288a8258
Reviewed-on: https://webrtc-review.googlesource.com/51764
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22371}
This CL activates unit tests for the task queue based send side
congestion controller that will replace the current one in the future.
To be able to have the tests run side by side with the tests of the old
congestion controller, the old tests have been prefixed with "Legacy".
This CL also contains some minor fixes to the new congestion controller
code.
Bug: webrtc:8415
Change-Id: I5e7474d42f17fcbfef402e26f638846fa3424695
Reviewed-on: https://webrtc-review.googlesource.com/55381
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22229}
Added sanity checks to interval budget to protect against integer
overflow. To avoid tests failing due to initializing paced sender
in another time domain, the elapsed time that is input into the
interval budget is limited to max 2 seconds.
Bug: webrtc:8942
Change-Id: I9ed32f059e65df7898c37bb34a008189ce79dc60
Reviewed-on: https://webrtc-review.googlesource.com/58087
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22220}
This is a reland of 18cf4b67dd.
Original change's description:
> Base pacer padding in pause state on time since last send.
>
> This clarifies the logic behind the pacer packet interval
> in paused state and prepares for future congestion window
> functionality.
>
> Bug: None
> Change-Id: Ibf6e23f73523b43742830353915b2b94d09a6fc9
> Reviewed-on: https://webrtc-review.googlesource.com/52060
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22004}
Bug: None
Change-Id: I19fc02bc226ad59cb4cbd2a6ef8ac6f47212f834
Reviewed-on: https://webrtc-review.googlesource.com/53080
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22138}
This cl prepares for a later CL introducing a new send side congestion
controller that will run on a task queue. It mostly consists of minor
fixes but adds some new interfaces that are unused in practice.
Bug: webrtc:8415
Change-Id: I1b58d0180a18eb15320d18733dac0dfe2e0f902a
Reviewed-on: https://webrtc-review.googlesource.com/53321
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22099}
This reverts commit 4e849f6925.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Reland "Moved congestion controller to task queue.""
>
> This reverts commit 57daeb7ac7.
>
> Reason for revert: Cause increased congestion and deadlocks in downstream project
>
> Original change's description:
> > Reland "Moved congestion controller to task queue."
> >
> > This is a reland of 0cbcba7ea0.
> >
> > Original change's description:
> > > Moved congestion controller to task queue.
> > >
> > > The goal of this work is to make it easier to experiment with the
> > > bandwidth estimation implementation. For this reason network control
> > > functionality is moved from SendSideCongestionController(SSCC),
> > > PacedSender and BitrateController to the newly created
> > > GoogCcNetworkController which implements the newly created
> > > NetworkControllerInterface. This allows the implementation to be
> > > replaced at runtime in the future.
> > >
> > > This is the first part of a split of a larger CL, see:
> > > https://webrtc-review.googlesource.com/c/src/+/39788/8
> > > For further explanations.
> > >
> > > Bug: webrtc:8415
> > > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> > > Reviewed-on: https://webrtc-review.googlesource.com/43840
> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21868}
> >
> > Bug: webrtc:8415
> > Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da
> > Reviewed-on: https://webrtc-review.googlesource.com/48000
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21899}
>
> TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:8415
> Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83
> Reviewed-on: https://webrtc-review.googlesource.com/52980
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22017}
TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I3393b74370c4f4d0955f50728005b2b925be169b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/53262
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22023}
This reverts commit 57daeb7ac7.
Reason for revert: Cause increased congestion and deadlocks in downstream project
Original change's description:
> Reland "Moved congestion controller to task queue."
>
> This is a reland of 0cbcba7ea0.
>
> Original change's description:
> > Moved congestion controller to task queue.
> >
> > The goal of this work is to make it easier to experiment with the
> > bandwidth estimation implementation. For this reason network control
> > functionality is moved from SendSideCongestionController(SSCC),
> > PacedSender and BitrateController to the newly created
> > GoogCcNetworkController which implements the newly created
> > NetworkControllerInterface. This allows the implementation to be
> > replaced at runtime in the future.
> >
> > This is the first part of a split of a larger CL, see:
> > https://webrtc-review.googlesource.com/c/src/+/39788/8
> > For further explanations.
> >
> > Bug: webrtc:8415
> > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> > Reviewed-on: https://webrtc-review.googlesource.com/43840
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21868}
>
> Bug: webrtc:8415
> Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da
> Reviewed-on: https://webrtc-review.googlesource.com/48000
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21899}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8415
Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83
Reviewed-on: https://webrtc-review.googlesource.com/52980
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22017}
This reverts commit 9af0a672c2.
Reason for revert: Conflicts with reverting https://webrtc-review.googlesource.com/c/src/+/48000
Original change's description:
> Renamed constants in unit tests for paced sender.
>
> Since paced sender no longer has an internal multiplier, the constants
> used for the unit tests were changed from supplying kTargetBitrateBps *
> kPaceMultiplier at each usage to simply using the new constant
> kPacingRateBps, simplifying the test code.
>
> The function PacketsSentPerInterval was introduced as the value was
> computed and explained several times over in the different tests.
>
> Bug: None
> Change-Id: Ib1cf9b40194272b1529abb02d49cae6b8732d1e6
> Reviewed-on: https://webrtc-review.googlesource.com/50443
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21999}
TBR=stefan@webrtc.org,srte@webrtc.org
Change-Id: Icb19b0796199a5fc30e531339d5fb13bfec9f329
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/53064
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22015}
This reverts commit 18cf4b67dd.
Reason for revert: Conflicts with reverting https://webrtc-review.googlesource.com/c/src/+/48000
Original change's description:
> Base pacer padding in pause state on time since last send.
>
> This clarifies the logic behind the pacer packet interval
> in paused state and prepares for future congestion window
> functionality.
>
> Bug: None
> Change-Id: Ibf6e23f73523b43742830353915b2b94d09a6fc9
> Reviewed-on: https://webrtc-review.googlesource.com/52060
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22004}
TBR=stefan@webrtc.org,srte@webrtc.org
Change-Id: I670d6f24bb600444d1b3d947795c59955d7b2d77
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/53061
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22014}
This clarifies the logic behind the pacer packet interval
in paused state and prepares for future congestion window
functionality.
Bug: None
Change-Id: Ibf6e23f73523b43742830353915b2b94d09a6fc9
Reviewed-on: https://webrtc-review.googlesource.com/52060
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22004}
Since paced sender no longer has an internal multiplier, the constants
used for the unit tests were changed from supplying kTargetBitrateBps *
kPaceMultiplier at each usage to simply using the new constant
kPacingRateBps, simplifying the test code.
The function PacketsSentPerInterval was introduced as the value was
computed and explained several times over in the different tests.
Bug: None
Change-Id: Ib1cf9b40194272b1529abb02d49cae6b8732d1e6
Reviewed-on: https://webrtc-review.googlesource.com/50443
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21999}
The previous name packet queue 2 had no indication on what the
difference was compared to the regular packet queue. This rename makes
it easier to understand the codebase.
Additionally the PacketQueueInterface class was introduced to make the
class hierarchy easier to follow. The round robin packet queue did not
extend the packet queue so there was no reason for inheriting from the
specific implementation.
Bug: None
Change-Id: Idbce081c751fbacd927632f5e71220887d0b5991
Reviewed-on: https://webrtc-review.googlesource.com/49120
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21931}
This is a reland of 0cbcba7ea0.
Original change's description:
> Moved congestion controller to task queue.
>
> The goal of this work is to make it easier to experiment with the
> bandwidth estimation implementation. For this reason network control
> functionality is moved from SendSideCongestionController(SSCC),
> PacedSender and BitrateController to the newly created
> GoogCcNetworkController which implements the newly created
> NetworkControllerInterface. This allows the implementation to be
> replaced at runtime in the future.
>
> This is the first part of a split of a larger CL, see:
> https://webrtc-review.googlesource.com/c/src/+/39788/8
> For further explanations.
>
> Bug: webrtc:8415
> Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> Reviewed-on: https://webrtc-review.googlesource.com/43840
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21868}
Bug: webrtc:8415
Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da
Reviewed-on: https://webrtc-review.googlesource.com/48000
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21899}
This reverts commit 0cbcba7ea0.
Reason for revert: Major regressions on perf bots.
Original change's description:
> Moved congestion controller to task queue.
>
> The goal of this work is to make it easier to experiment with the
> bandwidth estimation implementation. For this reason network control
> functionality is moved from SendSideCongestionController(SSCC),
> PacedSender and BitrateController to the newly created
> GoogCcNetworkController which implements the newly created
> NetworkControllerInterface. This allows the implementation to be
> replaced at runtime in the future.
>
> This is the first part of a split of a larger CL, see:
> https://webrtc-review.googlesource.com/c/src/+/39788/8
> For further explanations.
>
> Bug: webrtc:8415
> Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> Reviewed-on: https://webrtc-review.googlesource.com/43840
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21868}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: Ia8a273eb9e92b7d0d960c49658c228208170962d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/47560
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21877}
The goal of this work is to make it easier to experiment with the
bandwidth estimation implementation. For this reason network control
functionality is moved from SendSideCongestionController(SSCC),
PacedSender and BitrateController to the newly created
GoogCcNetworkController which implements the newly created
NetworkControllerInterface. This allows the implementation to be
replaced at runtime in the future.
This is the first part of a split of a larger CL, see:
https://webrtc-review.googlesource.com/c/src/+/39788/8
For further explanations.
Bug: webrtc:8415
Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
Reviewed-on: https://webrtc-review.googlesource.com/43840
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21868}
This is a reland of 001546da95
Original change's description:
> Break up rtc_event_log_api to solve circular dependencies.
>
> The original rtc_event_log_api is refactored to a pure API target plus
> multiple targets coupled with WebRTC implementations.
>
> Bug: None
> Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
> Reviewed-on: https://webrtc-review.googlesource.com/43247
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#21811}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org
Bug: None
Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647
Reviewed-on: https://webrtc-review.googlesource.com/46900
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21862}
The original rtc_event_log_api is refactored to a pure API target plus
multiple targets coupled with WebRTC implementations.
Bug: None
Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
Reviewed-on: https://webrtc-review.googlesource.com/43247
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#21811}
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.
Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).
This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.
Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.
TBR=phoglund@webrtc.org
Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
Various files in webrtc codebase use scanf/sscanf function without
including stdio.h header file which is supposed to define it. This
somehow works when using glibc, but fails with uClibc.
Bug: webrtc:8641
Change-Id: Ie4ae17af32b32ed8cea567166b6b0e5193966995
Reviewed-on: https://webrtc-review.googlesource.com/32261
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21775}
GetPacingFactor exposed internal details that should not be relied upon.
In a later CL theese won't be available any more, this CL is in
preparation for that change.
The only usage was in video send stream tests. To keep the tests
working, they now access the internal video send stream directly. The
test code retrieves an optional that indicates whether the send stream
has overridden the pacing factor. This means the implementation
dependency between video send stream and video send stream tests is
increased. This is an improvement compared to depending on the paced
sender implementation.
Bug: webrtc:8415
Change-Id: Id357553692b3ff3283fa3b64da1b1ebb3c97f04d
Reviewed-on: https://webrtc-review.googlesource.com/39265
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21675}
This replaces most of the existing dependencies on the application
limited region(ALR) detector. This is to achieve a greater separation of
concerns and will make further refactoring regarding the ALR Detector
less invasive on other parts of the code base.
Bug: webrtc:8415
Change-Id: I92912254c6d02285cce6a88f6789f0ac94794c88
Reviewed-on: https://webrtc-review.googlesource.com/37560
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21598}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
Bug: None
Change-Id: Ic83404f3b10ef7a9725ae1f4cf55e1b295cc9c5d
Reviewed-on: https://webrtc-review.googlesource.com/23570
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21571}
