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14 Commits

Author SHA1 Message Date
Danil Chapovalov
cad3e0e2fa Replace DataSize and DataRate factories with newer versions 
This is search and replace change:
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::Bytes<\(.*\)>()/DataSize::Bytes(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::bytes/DataSize::Bytes/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BitsPerSec<\(.*\)>()/DataRate::BitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BytesPerSec<\(.*\)>()/DataRate::BytesPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::KilobitsPerSec<\(.*\)>()/DataRate::KilobitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::bps/DataRate::BitsPerSec/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::kbps/DataRate::KilobitsPerSec/g"
git cl format

Bug: webrtc:9709
Change-Id: I65aaca69474ba038c1fe2dd8dc30d3f8e7b94c29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168647
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30545}
 2020-02-18 16:09:50 +00:00
Jonas Olsson
a4d873786f Format almost everything. 
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
 2019-07-08 13:45:15 +00:00
Sebastian Jansson
470a5eae93 Introduces common AudioAllocationSettings class. 
This class collects the field trial based configuration of audio
allocation and bandwidth in one place. This makes it easier
overview and prepares for future cleanup of the trials.

Bug: webrtc:9718
Change-Id: I34a441c0165b423f1e2ee63894337484684146ac
Reviewed-on: https://webrtc-review.googlesource.com/c/118282
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26370}
 2019-01-23 12:13:29 +00:00
Sebastian Jansson
5f00995964 Using unit classes in AimdRateControl. 
Bug: webrtc:9718
Change-Id: I1efed4e55c9d1ccec3c32ed012cb3cd82d7f4ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/110788
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25705}
 2018-11-20 08:04:11 +00:00
Sebastian Jansson
7c1744d3dc Reland "Reland "Using units in SendSideBandwidthEstimation."" 
This reverts commit a4de9c8b04ce9f7b1c4be639edd6cc4d3a9f07a1.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Revert "Reland "Using units in SendSideBandwidthEstimation.""
> 
> This reverts commit e2cb26cb4fa2a3ce7c12636225ba9c720d7c7e56.
> 
> Reason for revert: <INSERT REASONING HERE>
> 
> Original change's description:
> > Reland "Using units in SendSideBandwidthEstimation."
> > 
> > This reverts commit 917e5967a597fa8d6e6cae9ffccb21e3d35d553b.
> > 
> > Reason for revert: Handling downstream use case.
> > 
> > Original change's description:
> > > Revert "Using units in SendSideBandwidthEstimation."
> > > 
> > > This reverts commit 35b5e5f3b0dc409bf571b3609860ad5bb8e00c29.
> > > 
> > > Reason for revert: Breaks downstream project
> > > 
> > > Original change's description:
> > > > Using units in SendSideBandwidthEstimation.
> > > >
> > > > This CL moves SendSideBandwidthEstimation to use the unit types
> > > > DataRate, TimeDelta and Timestamp. This prepares for upcoming changes.
> > > >
> > > > Bug: webrtc:9718
> > > > Change-Id: If10e329920dda037b53055ff3352ae7f8d7e32b8
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/104021
> > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#25029}
> > > 
> > > TBR=terelius@webrtc.org,srte@webrtc.org
> > > 
> > > No-Try: True
> > > Bug: webrtc:9718
> > > Change-Id: Iaf470f1eec9911ee6fc7c1b4f5db9675d89d3780
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/104480
> > > Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> > > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#25035}
> > 
> > TBR=oprypin@webrtc.org,terelius@webrtc.org,srte@webrtc.org
> > 
> > Change-Id: I0940791fcd1e196598b0f0a2ec779c49931ee5df
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:9718
> > Reviewed-on: https://webrtc-review.googlesource.com/c/104520
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25036}
> 
> TBR=oprypin@webrtc.org,terelius@webrtc.org,srte@webrtc.org
> 
> Change-Id: I6628771c79fc78dfd856649ae92232e95df63495
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9718
> Reviewed-on: https://webrtc-review.googlesource.com/c/104540
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25037}

TBR=oprypin@webrtc.org,terelius@webrtc.org,srte@webrtc.org

Change-Id: If5473859cea725420afce11b6683fa0c70a29b0a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9718
Reviewed-on: https://webrtc-review.googlesource.com/c/104501
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25039}
 2018-10-08 09:02:03 +00:00
Sebastian Jansson
a4de9c8b04 Revert "Reland "Using units in SendSideBandwidthEstimation."" 
This reverts commit e2cb26cb4fa2a3ce7c12636225ba9c720d7c7e56.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Reland "Using units in SendSideBandwidthEstimation."
> 
> This reverts commit 917e5967a597fa8d6e6cae9ffccb21e3d35d553b.
> 
> Reason for revert: Handling downstream use case.
> 
> Original change's description:
> > Revert "Using units in SendSideBandwidthEstimation."
> > 
> > This reverts commit 35b5e5f3b0dc409bf571b3609860ad5bb8e00c29.
> > 
> > Reason for revert: Breaks downstream project
> > 
> > Original change's description:
> > > Using units in SendSideBandwidthEstimation.
> > >
> > > This CL moves SendSideBandwidthEstimation to use the unit types
> > > DataRate, TimeDelta and Timestamp. This prepares for upcoming changes.
> > >
> > > Bug: webrtc:9718
> > > Change-Id: If10e329920dda037b53055ff3352ae7f8d7e32b8
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/104021
> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#25029}
> > 
> > TBR=terelius@webrtc.org,srte@webrtc.org
> > 
> > No-Try: True
> > Bug: webrtc:9718
> > Change-Id: Iaf470f1eec9911ee6fc7c1b4f5db9675d89d3780
> > Reviewed-on: https://webrtc-review.googlesource.com/c/104480
> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25035}
> 
> TBR=oprypin@webrtc.org,terelius@webrtc.org,srte@webrtc.org
> 
> Change-Id: I0940791fcd1e196598b0f0a2ec779c49931ee5df
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9718
> Reviewed-on: https://webrtc-review.googlesource.com/c/104520
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25036}

TBR=oprypin@webrtc.org,terelius@webrtc.org,srte@webrtc.org

Change-Id: I6628771c79fc78dfd856649ae92232e95df63495
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9718
Reviewed-on: https://webrtc-review.googlesource.com/c/104540
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25037}
 2018-10-08 08:27:29 +00:00
Sebastian Jansson
e2cb26cb4f Reland "Using units in SendSideBandwidthEstimation." 
This reverts commit 917e5967a597fa8d6e6cae9ffccb21e3d35d553b.

Reason for revert: Handling downstream use case.

Original change's description:
> Revert "Using units in SendSideBandwidthEstimation."
> 
> This reverts commit 35b5e5f3b0dc409bf571b3609860ad5bb8e00c29.
> 
> Reason for revert: Breaks downstream project
> 
> Original change's description:
> > Using units in SendSideBandwidthEstimation.
> >
> > This CL moves SendSideBandwidthEstimation to use the unit types
> > DataRate, TimeDelta and Timestamp. This prepares for upcoming changes.
> >
> > Bug: webrtc:9718
> > Change-Id: If10e329920dda037b53055ff3352ae7f8d7e32b8
> > Reviewed-on: https://webrtc-review.googlesource.com/c/104021
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25029}
> 
> TBR=terelius@webrtc.org,srte@webrtc.org
> 
> No-Try: True
> Bug: webrtc:9718
> Change-Id: Iaf470f1eec9911ee6fc7c1b4f5db9675d89d3780
> Reviewed-on: https://webrtc-review.googlesource.com/c/104480
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25035}

TBR=oprypin@webrtc.org,terelius@webrtc.org,srte@webrtc.org

Change-Id: I0940791fcd1e196598b0f0a2ec779c49931ee5df
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9718
Reviewed-on: https://webrtc-review.googlesource.com/c/104520
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25036}
 2018-10-08 08:26:09 +00:00
Oleh Prypin
917e5967a5 Revert "Using units in SendSideBandwidthEstimation." 
This reverts commit 35b5e5f3b0dc409bf571b3609860ad5bb8e00c29.

Reason for revert: Breaks downstream project

Original change's description:
> Using units in SendSideBandwidthEstimation.
>
> This CL moves SendSideBandwidthEstimation to use the unit types
> DataRate, TimeDelta and Timestamp. This prepares for upcoming changes.
>
> Bug: webrtc:9718
> Change-Id: If10e329920dda037b53055ff3352ae7f8d7e32b8
> Reviewed-on: https://webrtc-review.googlesource.com/c/104021
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25029}

TBR=terelius@webrtc.org,srte@webrtc.org

No-Try: True
Bug: webrtc:9718
Change-Id: Iaf470f1eec9911ee6fc7c1b4f5db9675d89d3780
Reviewed-on: https://webrtc-review.googlesource.com/c/104480
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25035}
 2018-10-08 08:21:18 +00:00
Sebastian Jansson
35b5e5f3b0 Using units in SendSideBandwidthEstimation. 
This CL moves SendSideBandwidthEstimation to use the unit types
DataRate, TimeDelta and Timestamp. This prepares for upcoming changes.

Bug: webrtc:9718
Change-Id: If10e329920dda037b53055ff3352ae7f8d7e32b8
Reviewed-on: https://webrtc-review.googlesource.com/c/104021
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25029}
 2018-10-05 18:37:13 +00:00
Bjorn Terelius
43d0b98fe5 Clean up RateControlInput struct, used by bandwidth estimation. 
Remove unused member noise_var from RateControlInput struct.

Rename incoming_bitrate to estimated_throughput_bps to reflect
that the AimdRateControl may be running on either the send side
or the receive side.

Bug: webrtc:9411
Change-Id: Ie1ae0c29dc9559ef389993144e69fcd41684f1a4
Reviewed-on: https://webrtc-review.googlesource.com/83728
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Anastasia Koloskova <koloskova@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23783}
 2018-06-29 10:47:37 +00:00
Danil Chapovalov
0040b66ad3 Replace rtc::Optional with absl::optional 
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}
 2018-06-18 10:24:48 +00:00
Alex Narest
853715c9a9 Min BWE default is 10kbps but for audio send side BWE it was overridden to 5kbps. Now audio send side BWE is used for video calls too and should set min to 10kbps in case of video call. 
Bug: webrtc:9019
Change-Id: I3896bc8a014e918600d41b305afa5bceca550ee8
Reviewed-on: https://webrtc-review.googlesource.com/61963
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22446}
 2018-03-15 11:58:11 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/ 
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
 2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/. 
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
 2017-09-15 04:25:06 +00:00