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116 Commits

Author SHA1 Message Date
Erik Språng
4ab61cb9b4 Optionally allows TaskQueuePacedSender to coalesce send events. 
With an optional parameter this allows the task-queue based paced
sender to mimic the old behavior and coalesce sending of packets in
order to reduce thread wakeups and provide opportunity for batching.
This is done by simply overriding the minimum time the thread should
sleep. The pacing controller will already handle the "late wakup" case
and send any packets as if it had been woken at the optimal time.

Bug: webrtc:10809
Change-Id: Iceea00693a4e87d39b0e0ee8bdabca081dff2cba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175648
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31328}
 2020-05-19 17:23:30 +00:00
Mirko Bonadei
06d3559b79 Replace std::string::find() == 0 with absl::StartsWith (part 2). 
This CL has been generated using clang-tidy [1] except for changes to
BUILD.gn files.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/abseil-string-find-startswith.html

Bug: None
Change-Id: Ibf75601065a53bde28623b8eef57bec067235640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172586
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30984}
 2020-04-02 14:38:30 +00:00
Christoffer Rodbro
6404cddefb Allow setting a bandwidth cap for relayed connections. 
For now the capping is experimental and applied via a field trial.

Bug: webrtc:11434
Change-Id: Id8e6e9b948f099a0940974a9a431b5b0a43c32f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171226
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30909}
 2020-03-26 20:41:46 +00:00
Christoffer Rodbro
b0ca519c40 Handle extended route information in TransportFeedbackAdapter. 
Instead of passing only the local- and remote network IDs the whole
NetworkRoute is forwarded to TransportFeedbackAdapter that can then
use more detailed information to distinguish routes.

Bug: webrtc:11434
Change-Id: I48f36aa1177822d20c2b556dcc2275f6145ed845
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171581
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30895}
 2020-03-26 09:39:34 +00:00
Jonas Oreland
5b6a4d8908 Only print route if it has changed 
This is a follow up change to https://webrtc-review.googlesource.com/c/src/+/170628
and modifies code to only LOG if the route really has changed.

Existing code will LOG like this, which is slightly annoying. Notice that the same route change is LOG:ed twice.
03-23 13:28:49.281 17986 18850 I rtp_transport_controller_send.cc: [1183:590] [18850] (line 253): Network route changed on transport audio: new_route = [ connected: 1 local: [ 2/4 Wifi turn: 0 ] remote: [ 2/3 Wifi turn: 0 ] packet_overhead_bytes: 32 ]
03-23 13:28:49.281 17986 18850 I rtp_transport_controller_send.cc: [1183:590] [18850] (line 278): old_route = [ connected: 1 local: [ 2/4 Wifi turn: 1 ] remote: [ 2/3 Wifi turn: 0 ] packet_overhead_bytes: 28 ]
03-23 13:28:49.281 17986 18850 I rtp_transport_controller_send.cc: [1183:591] [18850] (line 253): Network route changed on transport audio: new_route = [ connected: 1 local: [ 2/4 Wifi turn: 0 ] remote: [ 2/3 Wifi turn: 0 ] packet_overhead_bytes: 32 ]
03-23 13:28:49.282 17986 18850 I rtp_transport_controller_send.cc: [1183:591] [18850] (line 278): old_route = [ connected: 1 local: [ 2/4 Wifi turn: 0 ] remote: [ 2/3 Wifi turn: 0 ] packet_overhead_bytes: 32 ]

The way this method is called twice with same argument is out of scope
for this change.

BUG: webrtc:11434
Change-Id: I052d089c59714513a09cbaed49f24c8f1300af58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171460
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30865}
 2020-03-24 11:48:42 +00:00
Jonas Oreland
71fda3613c Extend NetworkRoute with more info about local/remote endpoints 
This patch extends the NetworkRoute struct with more information
about local/remote endpoints. It adds
- adapter type
- adapter id
- relay

(previously it was "only" network_id)

The patch leaves the {local/remote}_network_id fields
around and populated since downstream projects depend
on them. They will be removed once they have migrated.

OWNER: srte@ call/ test/
OWNER: asapersson@ video/
OWNER: hta@ p2p/ pc/ rtc_base/

BUG: webrtc:11434
Change-Id: I9bcec385b40d707db385fef40b2c7a315dd35dd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170628
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30848}
 2020-03-20 16:55:38 +00:00
Marina Ciocea
e77912ba8c Insert frame transformer between Encoded and Packetizer. 
Add a new API in RTPSenderInterface, to be called from the browser side
to insert a frame transformer between the Encoded and the Packetizer.

The frame transformer is passed from RTPSenderInterface through the
library to be eventually set in RTPSenderVideo, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169128.

Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: I46cd0d8a798c2736c837e90cbf90d8901c7d27fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169127
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30642}
 2020-02-28 07:43:13 +00:00
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
Jakob Ivarsson
b4cdd62bb1 Update transport overhead the first time the network connects. 
This fixes a bug where transport_overhead_bytes_per_packet_ is sometimes
not set and therefore not included in the BWE.

Bug: webrtc:11359
Change-Id: Id3593299c6bcd7ce3c44a7b148c202240b3f1981
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168525
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30522}
 2020-02-14 10:24:01 +00:00
Danil Chapovalov
0c626afcf3 Use newer version of TimeDelta and TimeStamp factories in webrtc 
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30491}
 2020-02-10 12:21:17 +00:00
Mirko Bonadei
e7bc3a3477 Reland "Adds trial to use correct overhead calculation in pacer." 
This reverts commit 7affd9bcbb7a778408942d8afa4fe3ce29a8fc0b.

Reason for revert: The perf issue has been addressed in the reland (https://webrtc-review.googlesource.com/c/src/+/167883).

Original change's description:
> Revert "Adds trial to use correct overhead calculation in pacer."
> 
> This reverts commit 71a77c4b3b314a5e3b4e6b2f12d4886cff1b60d7.
> 
> Reason for revert: https://webrtc-review.googlesource.com/c/src/+/167524 needs to be reverted and this CL causes a merge conflict.
> 
> Original change's description:
> > Adds trial to use correct overhead calculation in pacer.
> > 
> > Bug: webrtc:9883
> > Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30399}
> 
> TBR=sprang@webrtc.org,srte@webrtc.org
> 
> Change-Id: I7d3efa29f70aa0363311766980acae6d88bbcaaa
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9883
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167880
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30409}

TBR=mbonadei@webrtc.org,sprang@webrtc.org,srte@webrtc.org

Change-Id: Iafdef81d08078000dc368e001f67bee660e2f5bc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167861
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30414}
 2020-01-29 18:45:16 +00:00
Sebastian Jansson
c3eb9fd49f Reland "Reland "Only include overhead if using send side bandwidth estimation."" 
This is a reland of 086055d0fd9b9b9efe8bcf85884324a019e9bd33

ANA was accitendly disabled even when transport sequence numbers were
negotiated due to a bug in how the audio send stream is configured. To
solve this we simply continue to always allow enabling ANA and leave it
up to the application to ensure that it's not used together with receive
side estimation.

Original change's description:
> Reland "Only include overhead if using send side bandwidth estimation."
>
> This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e
>
> Original change's description:
> > Only include overhead if using send side bandwidth estimation.
> >
> > Bug: webrtc:11298
> > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30382}
>
> Bug: webrtc:11298
> Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30390}

Bug: webrtc:11298
Change-Id: If2ad91e17ebfc85dc51edcd9607996e18c5d1f13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30413}
 2020-01-29 18:42:34 +00:00
Mirko Bonadei
4356490b7b Revert "Reland "Only include overhead if using send side bandwidth estimation."" 
This reverts commit 086055d0fd9b9b9efe8bcf85884324a019e9bd33.

Reason for revert: Causes some perf regressions.

Original change's description:
> Reland "Only include overhead if using send side bandwidth estimation."
> 
> This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e
> 
> Original change's description:
> > Only include overhead if using send side bandwidth estimation.
> > 
> > Bug: webrtc:11298
> > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30382}
> 
> Bug: webrtc:11298
> Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30390}

TBR=saza@webrtc.org,ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,alito@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11298
Change-Id: Id38de92ac25a1ce9a1360f0e37f65747d4cfb31b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30411}
 2020-01-29 16:38:57 +00:00
Mirko Bonadei
7affd9bcbb Revert "Adds trial to use correct overhead calculation in pacer." 
This reverts commit 71a77c4b3b314a5e3b4e6b2f12d4886cff1b60d7.

Reason for revert: https://webrtc-review.googlesource.com/c/src/+/167524 needs to be reverted and this CL causes a merge conflict.

Original change's description:
> Adds trial to use correct overhead calculation in pacer.
> 
> Bug: webrtc:9883
> Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30399}

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

Change-Id: I7d3efa29f70aa0363311766980acae6d88bbcaaa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167880
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30409}
 2020-01-29 15:29:04 +00:00
Sebastian Jansson
71a77c4b3b Adds trial to use correct overhead calculation in pacer. 
Bug: webrtc:9883
Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30399}
 2020-01-29 09:39:40 +00:00
Sebastian Jansson
086055d0fd Reland "Only include overhead if using send side bandwidth estimation." 
This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e

Original change's description:
> Only include overhead if using send side bandwidth estimation.
> 
> Bug: webrtc:11298
> Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30382}

Bug: webrtc:11298
Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30390}
 2020-01-28 10:36:39 +00:00
Sebastian Jansson
c709412c76 Revert "Only include overhead if using send side bandwidth estimation." 
This reverts commit 8c79c6e1af354c526497082c79ccbe12af03a33e.

Reason for revert: Introduced a Bug that can happen if the include overhead state changes between pushing and poping a packet from the pacer packet queue.

Original change's description:
> Only include overhead if using send side bandwidth estimation.
> 
> Bug: webrtc:11298
> Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30382}

TBR=saza@webrtc.org,ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,alito@webrtc.org

Change-Id: I0cacbc26408b7bec5bc3855a628e62407c081117
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11298
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167523
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30383}
 2020-01-27 15:09:49 +00:00
Sebastian Jansson
8c79c6e1af Only include overhead if using send side bandwidth estimation. 
Bug: webrtc:11298
Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30382}
 2020-01-27 14:19:54 +00:00
Sebastian Jansson
658f1814da Reland "Moves TransportFeedbackAdapter to TaskQueue." 
This is a reland of 62d01cde6f6ec1fa91b1e5234a7922ad1a4ce036

Original change's description:
> Moves TransportFeedbackAdapter to TaskQueue.
> 
> Bug: webrtc:9883
> Change-Id: Id87e281751d98043f4470df5a71d458f4cd654c1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158793
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30037}

Bug: webrtc:9883
Change-Id: Icc63883903b283d490e9d4ed455e0eca69ed2074
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162000
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30285}
 2020-01-16 16:41:53 +00:00
JT Teh
ea992f8771 Reland "Reland "Extracts ssrc based feedback tracking from feedback adapter."" 
This reverts commit d2d7a47247187236ce62e3c842963f6e4e9f0f1f.

Reason for revert: This revert is not needed. Failure was not due to webrtc.

Original change's description:
> Revert "Reland "Extracts ssrc based feedback tracking from feedback adapter.""
> 
> This reverts commit d61338fa6ed957dd992f25da4844db34b14f89c7.
> 
> Reason for revert: Causing a build break:
> webrtc/call/BUILD:300:1: Undeclared inclusion(s) in rule 'webrtc/call:rtp_sender':
> this rule is missing dependency declarations for the following files included by 'call/rtp_transport_controller_send.cc':
>   'webrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h'
> 
> 
> 
> Original change's description:
> > Reland "Extracts ssrc based feedback tracking from feedback adapter."
> > 
> > This is a reland of 08c46adc1e9f9a8d74357fe132a68906ae6e6974
> > 
> > Original change's description:
> > > Extracts ssrc based feedback tracking from feedback adapter.
> > > 
> > > This prepares for moving TransportFeedbackAdapter to TaskQueue.
> > > 
> > > Bug: webrtc:9883
> > > Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
> > > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30076}
> > 
> > Bug: webrtc:9883
> > Change-Id: Ia74a3b1fba4d83eece9b0eb6475d6e6aecb65700
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162201
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30266}
> 
> TBR=sprang@webrtc.org,srte@webrtc.org
> 
> Change-Id: I7f3f872c7ff863a37ad8dca08051fe1e04671bfb
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9883
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166182
> Reviewed-by: JT Teh <jtteh@webrtc.org>
> Commit-Queue: JT Teh <jtteh@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30270}

TBR=sprang@webrtc.org,srte@webrtc.org,jtteh@webrtc.org

Change-Id: Idd1073ebfef77b2154d7123b47dacb479537c550
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166200
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30271}
 2020-01-15 18:24:32 +00:00
JT Teh
d2d7a47247 Revert "Reland "Extracts ssrc based feedback tracking from feedback adapter."" 
This reverts commit d61338fa6ed957dd992f25da4844db34b14f89c7.

Reason for revert: Causing a build break:
webrtc/call/BUILD:300:1: Undeclared inclusion(s) in rule 'webrtc/call:rtp_sender':
this rule is missing dependency declarations for the following files included by 'call/rtp_transport_controller_send.cc':
  'webrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h'



Original change's description:
> Reland "Extracts ssrc based feedback tracking from feedback adapter."
> 
> This is a reland of 08c46adc1e9f9a8d74357fe132a68906ae6e6974
> 
> Original change's description:
> > Extracts ssrc based feedback tracking from feedback adapter.
> > 
> > This prepares for moving TransportFeedbackAdapter to TaskQueue.
> > 
> > Bug: webrtc:9883
> > Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30076}
> 
> Bug: webrtc:9883
> Change-Id: Ia74a3b1fba4d83eece9b0eb6475d6e6aecb65700
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162201
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30266}

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

Change-Id: I7f3f872c7ff863a37ad8dca08051fe1e04671bfb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166182
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30270}
 2020-01-15 17:44:42 +00:00
Sebastian Jansson
d61338fa6e Reland "Extracts ssrc based feedback tracking from feedback adapter." 
This is a reland of 08c46adc1e9f9a8d74357fe132a68906ae6e6974

Original change's description:
> Extracts ssrc based feedback tracking from feedback adapter.
> 
> This prepares for moving TransportFeedbackAdapter to TaskQueue.
> 
> Bug: webrtc:9883
> Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30076}

Bug: webrtc:9883
Change-Id: Ia74a3b1fba4d83eece9b0eb6475d6e6aecb65700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162201
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30266}
 2020-01-15 12:51:16 +00:00
Jonas Olsson
b2b2031457 Concatenate string literals at compile time. 
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format

After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.

This primary benefit of this change is a small reduction in binary size.

Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
 2020-01-14 14:47:48 +00:00
Sebastian Jansson
41466b7bef Revert "Extracts ssrc based feedback tracking from feedback adapter." 
This reverts commit 08c46adc1e9f9a8d74357fe132a68906ae6e6974.

Reason for revert: Incomplete.

Original change's description:
> Extracts ssrc based feedback tracking from feedback adapter.
> 
> This prepares for moving TransportFeedbackAdapter to TaskQueue.
> 
> Bug: webrtc:9883
> Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30076}

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

Change-Id: I6a79e7627f9de2d8c876d6a13ca36f3ac06fde7f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162200
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30087}
 2019-12-13 14:47:48 +00:00
Harald Alvestrand
977b265702 Reduce some logging at INFO level by moving log statements 
from LS_INFO to LS_VERBOSE.

By default, unit tests run with logging at info level.
A random run today produced more than 70.000 lines of
output. This CL would reduce that by approximately 15.000.

Bug: none
Change-Id: Ie62708cebf109510a2443aa5ab5c4e645ffc6707
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161950
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30077}
 2019-12-12 21:54:06 +00:00
Sebastian Jansson
08c46adc1e Extracts ssrc based feedback tracking from feedback adapter. 
This prepares for moving TransportFeedbackAdapter to TaskQueue.

Bug: webrtc:9883
Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30076}
 2019-12-12 18:25:25 +00:00
Mirko Bonadei
f18f9206e5 Revert "Moves TransportFeedbackAdapter to TaskQueue." 
This reverts commit 62d01cde6f6ec1fa91b1e5234a7922ad1a4ce036.

Reason for revert: Causes SIGSEGV in webrtc::RTPSender::BuildRtxPacket.

Original change's description:
> Moves TransportFeedbackAdapter to TaskQueue.
>
> Bug: webrtc:9883
> Change-Id: Id87e281751d98043f4470df5a71d458f4cd654c1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158793
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30037}

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

# Not skipping CQ checks because original CL landed > 1 day ago.

No-Try: True
Bug: webrtc:9883
Change-Id: If54bdb8694144fae3fafbabd72d1ac1198e51aa6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161726
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30052}
 2019-12-10 13:51:29 +00:00
Sebastian Jansson
62d01cde6f Moves TransportFeedbackAdapter to TaskQueue. 
Bug: webrtc:9883
Change-Id: Id87e281751d98043f4470df5a71d458f4cd654c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158793
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30037}
 2019-12-09 10:38:54 +00:00
Erik Språng
014dd3c9f7 Trials should always be populated in call config. 
The trials are always set when a Call instead is created by a
CallFactory, but a lot of unit tests creates instances directly.
This CL updates those call site. There is still a fallback in place
in RtpTransportControllerSend, since there are down-stream usages that
need to be clean up. After that, we'll remove the fallback.

Bug: webrtc:10809
Change-Id: I0aacf0473317bcd64252dd43d93c42de730e2ffa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160408
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29978}
 2019-12-03 10:34:55 +00:00
Erik Språng
4314a494cf Implements a task-queue based PacedSender, wires it up for field trials 
Bug: webrtc:10809
Change-Id: Ia181c16559f4598f32dd399c24802d0a289e250b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150942
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29946}
 2019-11-28 12:13:53 +00:00
Erik Språng
662678dbf7 Adds injectable trials from peerconnection down to transport controller. 
This will be immediately useful to guarantee consistent state across
components referencing the pacer, but will be a net benefit overall
imo.

Bug: webrtc:10809
Change-Id: I49630696f757a832ccf2e4c8597193bf087ce53b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159885
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29859}
 2019-11-21 12:41:45 +00:00
Sebastian Jansson
f298855981 Cleanup of feedback observer interface 
Removes all unused features, reducing the exposed interface surface.
This makes refactoring and maintenance simpler as we can change
TransportFeedbackAdapter without making corresponding changes
to RtpVideoSender.

Bug: webrtc:9883
Change-Id: If372a868e0765e94df52b4de52d3bb619ce11471
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156943
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29649}
 2019-10-30 07:50:29 +00:00
Danil Chapovalov
e34fb878b9 Clarify NetworkControl interface: result of each function must be used 
Bug: None
Change-Id: Iff93513d36ed60d2c1bcbabb4dd5f8716e40d183
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157860
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29558}
 2019-10-21 12:35:07 +00:00
Sebastian Jansson
f34116e356 Replacing bandwidth adaptation trial with stable target in Opus encoder. 
This also means that the NetworkEstimate::bandwidth can be deprecated
as it's currently just a copy of the target_rate.

Bug: webrtc:10981
Change-Id: I1bc57b98480bd77ce052736b19d630c775428546
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153669
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29288}
 2019-09-24 16:35:02 +00:00
Sebastian Jansson
0a5ed896e2 Adds remote estimates to rtc event log. 
Bug: webrtc:10742
Change-Id: I0db998a05492603fcdeedca780d9ee3d64aa00d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151651
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29233}
 2019-09-19 09:22:37 +00:00
Sebastian Jansson
93b1ea2168 Using struct for bitrate allocation limits. 
Bug: webrtc:9883
Change-Id: I855c29808ffa14626d78842491fdf81cd00589e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153344
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29227}
 2019-09-18 17:03:59 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique. 
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
 2019-09-17 15:47:29 +00:00
Tommi
55dd72c54b Remove lock for process thread pointer from PacedSender. 
Also adding code in preparation of hiding the Module
implementation in PacedSender. The implementation details of
how the PacedSender+ProcessThread interaction works, has now
been moved into PacedSender (and out of RtpTransportControllerSend).

Instead of adding a "GetModuleImplementationForTesting" method
to the PacedSender class (which would have been the lazy way
out), I incorporated MockedProcessThread in the PacedSender tests.
This means more boilerplate code but the Module functionality
can be tested separately from the PacedSender and down the line
I think it would be a good idea to start using a separate thread
in the test, which is how the class under test is really used
in production.

Bug: none
Change-Id: Iec1b7c97cb0b363b331143ca70545e6ebafe2cd4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149176
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29011}
 2019-08-29 17:08:24 +00:00
Erik Språng
425d6aaa4c Add RtpPacketPacer interface for pacer control 
The PacedSender is being reworked and will need an interface so we can
inject different implementations of it.

This CL introduces a new RtpPacketPacer interface inside the pacing
module. This interface handles the details of _how_ packets should be
paced, such as pacing rates/account for audio/max queue length etc.

The RtpPacketSender interface exposed from the rtp_rtcp module handles
only the actual sending of packets.

Some minor cleanups are included here.

Bug: webrtc:10809
Change-Id: I150b1a6262306d99e3f9d5f0b4afdb16a50e5ad8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145212
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28699}
 2019-07-29 15:37:39 +00:00
Erik Språng
aa59eca891 Move RtpPacketSender and merge it with RtpPacketPacer. 
This interface is intended to only handle packet-sending parts of the
paced sender.

See https://webrtc-review.googlesource.com/c/src/+/145212 for context

Bug: webrtc:10809
Change-Id: I93f0b40e1865665c2d436db67021350a0ed0687b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145216
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28662}
 2019-07-24 13:28:21 +00:00
Sebastian Jansson
e1795f4158 Adds remote estimate RTCP packet. 
This adds the RemoteEstimate rtcp packet and wires it up to GoogCC where
it's used to improve congestion controller behavior.

The functionality is negotiated using SDP.

It's added with a field trial that allow disabling the functionality in
case there's any issues.

Bug: webrtc:10742
Change-Id: I1ea8e4216a27cd2b00505c99b42d1e38726256c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146602
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28654}
 2019-07-24 10:17:26 +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
Erik Språng
59b8654045 Switch from RtpPacketSender to RtpPacketPacer interface usage. 
RtpPacketSender interface will be removed when downstream projects have
been updated.

Bug: webrtc:10633
Change-Id: Ie127b9814f39bd213d00ded0f7b98380f2f01084
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143175
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28350}
 2019-06-24 10:46:06 +00:00
Sebastian Jansson
607a6f1c55 Moves conversion to ReceivedPacket from RtpPacketReceived to Call. 
This moves the conversion from RtpPacketReceived to ReceivedPacket to
Call rather than RtpTransportController. This prepares for reusing the
struct for receive side network state estimation.

Bug: webrtc:10742
Change-Id: I9581438bc912ef4bb635a5d9a6dea488cf871d48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141872
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28284}
 2019-06-14 12:19:49 +00:00
Ying Wang
8b27910cbc Include downlink delay into congestion window size. 
Change-Id: I33db0c8134b6b3181a7b3abcf32a622a89ff3ab4

Bug: webrtc:10688
Change-Id: I33db0c8134b6b3181a7b3abcf32a622a89ff3ab4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138275
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28079}
 2019-05-27 16:07:19 +00:00
Sebastian Jansson
166b45db26 Adds route changes in event logs. 
Bug: webrtc:10614
Change-Id: Ifd859c977fc66cb606914ddb38a3fb3618e3ad90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135952
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27924}
 2019-05-13 10:41:40 +00:00
Erik Språng
30a276b5d7 Add RTP sequence number to TransportFeedbackObserver::AddPacket() 
With this change, both the normal RTP and the transport-wide sequence
numbers are propagated with with AddPacket() call via a new
RtpPacketSendInfo struct, replacing the previous set of parameters.

The intent with this is that SendTimeHistory can hold a mapping from
transport-wide to rtp sequence numbers, and then via callbacks let the
RTP modules know when packets have been received by the remote end.

Bug: webrtc:8975
Change-Id: I6a24fc6282cbb041393752d39593c2867b242192
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133021
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27708}
 2019-04-23 11:02:56 +00:00
Sebastian Jansson
df88cc014a Allow injection of network estimator into GoogCC. 
Bug: webrtc:10498
Change-Id: Ie9225411db201dfcfa0a37a3c40992acbdc215bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132402
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@{#27624}
 2019-04-15 14:12:08 +00:00
Ying Wang
0810a7c25a Add base class NetworkPredictor and NetworkPredictorFactory and wire up. 
Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe.

Bug: webrtc:10492
Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27543}
 2019-04-10 12:38:58 +00:00
Oleh Prypin
e8964903a9 Revert "Fix target bitrate RTCP messages behavior for SVC streams" 
This reverts commit ab65d8aab5fe63619033371fca1ce2711c2c2137.

Reason for revert: Fails video_engine_tests ExtendedReportsEndToEndTest.TestExtendedReportsCanSignalZeroTargetBitrate
https://ci.chromium.org/p/webrtc/builders/ci/Linux%20MSan/18366

Original change's description:
> Fix target bitrate RTCP messages behavior for SVC streams
>
> Before this CL for SVC streams (e.g VP9) still 3 separate RTP_RTCP senders
> were created. The RTCP target bitrate messages were treated as simulcast
> and were split and send for each separate spatial layer in a separate SSRC.
>
> To fix that an svc flag is now wired to VideoSendStream config
> and filled based on the encoder config in WebrtcVideoEngine. This flag is
> used to differentiate between simulcast and SVC mode in RtpVideoSender.
>
> Bug: webrtc:10485
> Change-Id: Ifa01d12a7d4f01fcbe448ad11e0cc39ab2d1df55
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129929
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27345}

TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org

Change-Id: I184f87289d9dccc67de165038d76a5690158a3b5
No-Tree-Checks: True
No-Try: True
Bug: webrtc:10485
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130467
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27355}
 2019-03-29 15:52:11 +00:00