Commit graph

32 commits

Author SHA1 Message Date
Per K
30f1cb318b Remove dependency from rtp_rtcp module to remote_bitrate_estimator
This depenency is not needed and may lead to a circular dependency. The cl removes old unused functionaliy to log BWE related statistics using compile time flags.

Bug: webrtc:42225697
Change-Id: I6cc01b367c0c48ab30f34c12a10afc58d1e7822f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352142
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42386}
2024-05-27 15:49:28 +00:00
philipel
5ace0710bf Remove unused PacketOptions::additional_data.
Bug: none
Change-Id: I642ad5fde070d7c9c708d99ec9a91b28e294d11e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341960
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41863}
2024-03-06 11:17:52 +00:00
Danil Chapovalov
3aa951a7c6 Delete SendDelayObserver interface
send delay is now measured through  SendPacketObserver interface

Bug: None
Change-Id: I0dc3de1522e2824d9431d7e3a3dc524588687dda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319500
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40755}
2023-09-15 14:59:23 +00:00
Harald Alvestrand
34d82df2ba Use ArrayView versions of SendRtp and SendRtcp
This CL adds [[deprecated]] to the old signatures, and uses the new
signatures throughout.

Bug: webrtc:14870
Change-Id: Ic9a8198ac0a2f954e1b2e7d05a55dbe04342f958
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314962
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40517}
2023-08-07 08:28:48 +00:00
Danil Chapovalov
950e231b63 In RtpRtcp use BitrateTracker instead of RateStatistics to measure bitrate
BitrateTracker uses RateStatistics underneath, thus algorithm is the same,
but it provides Timestamp/TimeDelta friendly interface

Bug: webrtc:13757
Change-Id: I9f2fcb3d498b2a137b531b94b660d15aa273c4bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40465}
2023-07-24 14:57:29 +00:00
Danil Chapovalov
8beb6314ef Pass and process capture time through SendPacketObserver with Timestamp type
Bug: webrtc:13757
Change-Id: Icc9f650590640f402ca9004171bbddaf918c78d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308682
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40339}
2023-06-22 17:16:41 +00:00
Danil Chapovalov
0f1a2c5d97 Change StreamDataCounters to use Timestamp instead of int64_t
Bug: webrtc:13757
Change-Id: I11151682a07a2d95389f81cbd7f47f26ad8e67ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306700
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40155}
2023-05-26 11:35:57 +00:00
Danil Chapovalov
7bb9322e9e Drop RtpRtcp unittest dependency on global field trial string
FieldTrialBasedConfig reads config from the global field trial string
ScopedKeyValueConfig adjust the global field trial string
test::ExplicitKeyValueConfig doesnt touch the global field trial string

Bug: webrtc:11926
Change-Id: I8883634fdc7e1bdb63eec9bf38114a3031103839
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299062
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39683}
2023-03-27 09:53:19 +00:00
Erik Språng
1b11b58b56 Remove pending packets from the pacer when an RTP module is removed.
This CL adds functionality to remove packets matching a given SSRC from
the pacer queue, and calls that with any SSRCs used by an RTP module
when that module is removed.

Bug: chromium:1395081
Change-Id: I13c0285ddca600e784ad04a806727a508ede6dcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287124
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38880}
2022-12-13 11:32:58 +00:00
Per Kjellander
e0b4cab69c Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
Bug: webrtc:6762
Change-Id: I520188a13ee5f50c441226574ccb3df54f842835
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285300
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38783}
2022-11-30 20:19:36 +00:00
Henrik Boström
d81992197c [Stats] Update totalPacketSendDelay to only cover time in pacer queue.
This metric was always supposed to be the spec's answer to
googBucketDelay, and is defined as "The total number of seconds that
packets have spent buffered locally before being transmitted onto the
network." But our implementation measured the time between capture and
send, including encode time. This is incorrect and yields a much larger
value than expected.

This CL updated the metric to do what the spec says. Implementation-wise
we measure the time between pushing and popping each packet from the
queue (in modules/pacing/prioritized_packet_queue.cc).

The spec says to increment the delay counter at the same time as we
increment the packet counter in order for the app to be able to do
"delta totalPacketSendDelay / delta packetSent". For this reason,
`total_packet_delay` is added to RtpPacketCounter. (Previously, the
two counters were incremented on different threads and observers.)

Running Google Meet on a good network, I could observe a 2-3 ms average
send delay per packet with this implementation compared to 20-30 ms
with the old implementation. See b/137014977#comment170 for comparison
with googBucketDelay which is a little bit different by design -
totalPacketSendDelay is clearly better than googBucketDelay.

Since none of this depend on the media kind, we can wire up this metric
for audio as well in a follow-up:
https://webrtc-review.googlesource.com/c/src/+/280523

Bug: webrtc:14593
Change-Id: If8fcd82fee74030d0923ee5df2c2aea2264600d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280443
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38480}
2022-10-26 21:29:20 +00:00
Danil Chapovalov
e58f1991dc Add Timestamp -> AbsoluteSendTime conversion function
instead of ms -> AbsoluteSendTime helper

Bug: webrtc:13757
Change-Id: I57389a66a43b4f4838023f9c224a985f2cd57107
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266024
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37350}
2022-06-28 09:26:42 +00:00
Jonas Oreland
e62c2f2c77 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
rename WebRtcKeyValueConfig to FieldTrialsView

Bug: webrtc:10335
Change-Id: If725bd498c4c3daf144bee638230fa089fdde833
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 10:14:00 +00:00
Danil Chapovalov
a2ee9234b4 Migrate to Timestamp and TimeDelta types in RtpPacketHistory
Bug: webrtc:13757
Change-Id: Ie542fca50b97fe9dc450e45da40f05e2b66c7da5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252981
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36132}
2022-03-04 15:02:58 +00:00
Danil Chapovalov
9af4aa7cf4 Reland "Represent RtpPacketToSend::capture_time with Timestamp"
This reverts commit 56db8d0952.

Reason for revert: downstream problem addressed

Original change's description:
> Revert "Represent RtpPacketToSend::capture_time with Timestamp"
>
> This reverts commit 385eb9714d.
>
> Reason for revert: Causes problems downstream:
>
> #
> # Fatal error in: rtc_base/units/unit_base.h, line 122
> # last system error: 0
> # Check failed: value >= 0 (-234 vs. 0)
>
> Original change's description:
> > Represent RtpPacketToSend::capture_time with Timestamp
> >
> > Bug: webrtc:13757
> > Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#36083}
>
> Bug: webrtc:13757
> Change-Id: I8442abd438be8726cf671d0f372d50ecfac6847e
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252720
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36087}

Bug: webrtc:13757
Change-Id: I1fa852757480116f35deb2b6c3c27800bdf5e197
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252781
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36093}
2022-02-28 10:04:37 +00:00
Tomas Gunnarsson
56db8d0952 Revert "Represent RtpPacketToSend::capture_time with Timestamp"
This reverts commit 385eb9714d.

Reason for revert: Causes problems downstream:

#
# Fatal error in: rtc_base/units/unit_base.h, line 122
# last system error: 0
# Check failed: value >= 0 (-234 vs. 0)

Original change's description:
> Represent RtpPacketToSend::capture_time with Timestamp
>
> Bug: webrtc:13757
> Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36083}

Bug: webrtc:13757
Change-Id: I8442abd438be8726cf671d0f372d50ecfac6847e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252720
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36087}
2022-02-26 10:35:13 +00:00
Danil Chapovalov
385eb9714d Represent RtpPacketToSend::capture_time with Timestamp
Bug: webrtc:13757
Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36083}
2022-02-25 16:44:07 +00:00
Danil Chapovalov
4ad09762da Delete legacy RtpPacketSendInfo::ssrc field
Bug: webrtc:12713
Change-Id: I8785eaabbb8725fb119067f5897ef245e6187996
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239642
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35460}
2021-12-02 11:59:37 +00:00
Erik Språng
5f1d406cc9 Move legacy RtpRtcpModule to deferred sequencing.
Bug: webrtc:11340
Change-Id: I45ba6c37fe7fec8de756dee1eb914aceebbeae93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228437
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34734}
2021-08-12 10:22:59 +00:00
Artem Titov
913cfa76ec Use backticks not vertical bars to denote variables in comments for /modules/rtp_rtcp
Bug: webrtc:12338
Change-Id: I52eb3b6675c4705e22f51b70799ed6139a3b46bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227164
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34686}
2021-08-09 15:51:03 +00:00
Erik Språng
6a0a55907b Reland "Correctly handle retransmissions/padding in early loss detection."
This is a reland of e9ae4729e0

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

Original change's description:
> Correctly handle retransmissions/padding in early loss detection.
>
> This CL makes sure we don't cull packets from the history based on
> incorrect ack mapping, just like it's predecessor:
> https://webrtc-review.googlesource.com/c/src/+/218000
>
> It also changes the logic to make sure retransmits counts towards
> history pruning - and properly ignores padding/fec.
>
> Bug: webrtc:12713
> Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34293}

Bug: webrtc:12713
Change-Id: Iec123d71edafea98fe289acde007b57e212681f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222640
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34297}
2021-06-16 08:14:27 +00:00
Erik Språng
d6957c2eed Revert "Correctly handle retransmissions/padding in early loss detection."
This reverts commit e9ae4729e0.

Reason for revert: Internal test failure

Original change's description:
> Correctly handle retransmissions/padding in early loss detection.
>
> This CL makes sure we don't cull packets from the history based on
> incorrect ack mapping, just like it's predecessor:
> https://webrtc-review.googlesource.com/c/src/+/218000
>
> It also changes the logic to make sure retransmits counts towards
> history pruning - and properly ignores padding/fec.
>
> Bug: webrtc:12713
> Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34293}

TBR=danilchap@webrtc.org,terelius@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Iaca6dc7739d953e97add5f5d516139b4819e43ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222601
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34294}
2021-06-15 15:59:10 +00:00
Erik Språng
e9ae4729e0 Correctly handle retransmissions/padding in early loss detection.
This CL makes sure we don't cull packets from the history based on
incorrect ack mapping, just like it's predecessor:
https://webrtc-review.googlesource.com/c/src/+/218000

It also changes the logic to make sure retransmits counts towards
history pruning - and properly ignores padding/fec.

Bug: webrtc:12713
Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34293}
2021-06-15 15:39:19 +00:00
Danil Chapovalov
eee0e9e9d4 Remove passing rtp packet metadata through webrtc as array of bytes
Instead metadata is now passed via refcounted class.

Bug: b/178094662
Change-Id: I9591fb12990282b60310ca01aea2a7b73d92487a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204060
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33134}
2021-02-02 12:22:57 +00:00
Danil Chapovalov
5312a8f532 Add option to attach custom object to an rtp packet
As an alternative to attaching custom array of bytes.

Bug: b/178094662
Change-Id: I92dcbf04998d8206091125febc520ebfcc4bcebf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203264
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33069}
2021-01-25 18:31:34 +00:00
Jakob Ivarsson
36274f9158 Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
This is a reland of 1dbe30c7e8

Original change's description:
> Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
>
> This is a reland of 87c1950841
>
> Original change's description:
> > Default enable WebRTC-SendSideBwe-WithOverhead.
> >
> > Bug: webrtc:6762
> > Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32472}
>
> Bug: webrtc:6762
> Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32492}

Bug: webrtc:6762
Change-Id: I6d79894a213fc42d2338409e7513247725881b1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191221
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32534}
2020-11-02 11:05:56 +00:00
Björn Terelius
d546186b89 Revert "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
This reverts commit 1dbe30c7e8.

Reason for revert: Speculative revert due to failing tests.

Original change's description:
> Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
>
> This is a reland of 87c1950841
>
> Original change's description:
> > Default enable WebRTC-SendSideBwe-WithOverhead.
> >
> > Bug: webrtc:6762
> > Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32472}
>
> Bug: webrtc:6762
> Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32492}

TBR=stefan@webrtc.org,jakobi@webrtc.org,alito@webrtc.org

Change-Id: I7e0378788576236059627cf8c3bad58cd70aff7e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32504}
2020-10-27 10:51:46 +00:00
Jakob Ivarsson
1dbe30c7e8 Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
This is a reland of 87c1950841

Original change's description:
> Default enable WebRTC-SendSideBwe-WithOverhead.
>
> Bug: webrtc:6762
> Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32472}

Bug: webrtc:6762
Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32492}
2020-10-26 12:35:47 +00:00
Jakob Ivarsson
27af3c4c24 Revert "Default enable WebRTC-SendSideBwe-WithOverhead."
This reverts commit 87c1950841.

Reason for revert: breaks downstream tests

Original change's description:
> Default enable WebRTC-SendSideBwe-WithOverhead.
>
> Bug: webrtc:6762
> Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32472}

TBR=stefan@webrtc.org,jakobi@webrtc.org,alito@webrtc.org

Change-Id: If59fd41dcd8f6db76ea297c34c25fe19ae2ae973
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189973
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32474}
2020-10-22 16:57:18 +00:00
Jakob Ivarsson
87c1950841 Default enable WebRTC-SendSideBwe-WithOverhead.
Bug: webrtc:6762
Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32472}
2020-10-22 13:37:18 +00:00
Markus Handell
e7c015e112 Migrate modules/rtp_rtcp to webrtc::Mutex.
Bug: webrtc:11567
Change-Id: I4c71f3a28ef875af2c232b1b553840d6e21649d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178804
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31645}
2020-07-07 12:13:47 +00:00
Tomas Gunnarsson
593e6a4238 Fork RtpSenderEgress.
The forked and deprecated implementation is used by the
deprecated ModuleRtpRtcpImpl implementation.

Change-Id: If67ca1181f40969791cf9c8903c0e49679c86834
Bug: webrtc:11581, webrtc:11611
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176566
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31459}
2020-06-08 08:02:50 +00:00