74 Commits

Author SHA1 Message Date
Sebastian Jansson
bae12756da Using unit types in TransportFeedbackAdapter.
Bug: webrtc:9883
Change-Id: I6d7d653079bb969fa3bc6f62fd35f2aa870edab6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158792
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29705}
2019-11-06 12:25:00 +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
Mirko Bonadei
86d053c2db Use source_sets in component builds and static_library in release builds.
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).

Source sets always pass all the object files to the linker.

On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.

See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set

Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:17:18 +00:00
Honghai Zhang
f8998cf8c4 Add a turn port prune policy to keep the first ready turn port.
Bug: webrtc:11026
Change-Id: I6222e9613ee4ce2dcfbb717e2430ea833c0dc373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155542
Commit-Queue: Honghai Zhang <honghaiz@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29470}
2019-10-14 19:08:23 +00:00
Sebastian Jansson
79f3287fcf Cleanup of simple TODO(srte) comments.
Just fixing some minor TODOs in my name. Not worth splitting into
separate CLs as the changes are minor.

Bug: webrtc:9883
Change-Id: I05c54b76507a1d51b92cad080ca4e2dfe8546bf1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155520
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29377}
2019-10-04 07:57:16 +00:00
Erik Språng
78c82a4040 Adds trial to always start probes with a small padding packet.
This will reduce bias caused by uncertainty in averaging window.

Bug: None
Change-Id: I5c4fe39ffe69fb4af87d86995196a54115d3e0b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144720
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29374}
2019-10-03 17:19:22 +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
Bjorn A Mellem
cd40de94ef Delete the deprecated GetTransportParametersOffer().
Transport parameters are no longer retreived using this method, and no
implementations currently override it.

Bug: webrtc:9719
Change-Id: Iba0e1c7a320266f199aab6f2add36c6a22b48458
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154004
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29272}
2019-09-23 20:52:18 +00:00
Bjorn A Mellem
37ad5ab940 Change DatagramTransportInterface methods to pure virtual.
These methods are implemented everywhere, and no longer need to provide
default implementations.

Bug: webrtc:9719
Change-Id: I2b33ace17696ec832a9936cf02a81c4973158046
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154003
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29263}
2019-09-20 20:11:24 +00:00
Bjorn A Mellem
88db835278 Change DataChannelTransportInterface/Sink methods to pure virtual.
These methods are implemented everywhere, so they no longer need to
provide default implementations.

Bug: webrtc:9719
Change-Id: Idf67a78010a55f545d882793d0d6edbccfae525b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154002
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29262}
2019-09-20 20:06:14 +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
53227ccba9 Remove webrtc::MinPositive from api/.
Follow-up of https://webrtc-review.googlesource.com/c/src/+/153220,
where during code review it was suggested to move webrtc::MinPositive
out of the api/ directory.

Bug: None
Change-Id: I0c3b87a9ffd1cd205a85dddd9f44cfd95eb02206
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153480
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29220}
2019-09-18 12:52:09 +00:00
Mirko Bonadei
738bfa7bab Remove api/bitrate_constraints.h.
Bug: webrtc:8733
Change-Id: Iaeb26e07d399f25dc18b0c4af38ed400577a5d3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29217}
2019-09-18 06:37:58 +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
Sebastian Jansson
e1b777717b Removing deprecated min_pacing_rate alias in StreamsConfig.
Bug: webrtc:9883
Change-Id: I8ca9f51b60b5fc24233f14404c13b411a5f2c253
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153343
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29206}
2019-09-17 11:20:34 +00:00
Niels Möller
65f17ca6b4 Move MediaTransportInterface out of the libjingle_peerconnection_api target
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.

Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
2019-09-13 10:49:56 +00:00
Per Kjellander
52f7ae7c89 Make NetworkStateEstimator injectable in RemoteBitrateEstimator
The NetworkStateEstimator is updated on every incoming RTP packet if available.

A rtcp::RemoteEstimate packet is sent every time a rtcp::TransportFeedback packet is sent.

BUG=webrtc:10742

Change-Id: I4cd8e9d85d35faf76aeefd2e26c2a9fe1a62ca3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152161
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29143}
2019-09-10 18:47:36 +00:00
Niels Möller
a837030f8f Split out RtpSource from libjingle_peerconnection_api
And moved declaration into a new api directory, as
api/transport/rtp/rtp_source.h.

Bug: webrtc:8733
Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29039}
2019-09-02 14:04:47 +00:00
Florent Castelli
4e615d590a Wire the stable target bitrate from GoogCC to the BitrateAllocator
Deprecated the field BitrateAllocationUpdate::link_capacity since it is only
used by the Opus codec in order to smooth the target bitrate, which is
equivalent to the stable_target_bitrate field.

The unused field trial WebRTC-Bwe-StableBandwidthEstimate is also removed.

Bug: webrtc:10126
Change-Id: Ic4a8a9ca4202136d011b91dc23c3a27cfd00d975
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149839
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28941}
2019-08-22 15:25:15 +00:00
Niels Möller
5297cf368d Delete unused class MockTargetTransferRateObserver
Bug: None
Change-Id: I60e9dc05450207dfd572ae17a42cf1adaed4c1b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148525
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28813}
2019-08-09 06:15:06 +00:00
Sebastian Jansson
e34d62caec Revert "Makes send side network estimation opt-in."
This reverts commit c36c8e114df8f21ec286adf1ffe76cf5fc495f53.

Reason for revert: Breaks downstream code.

Original change's description:
> Makes send side network estimation opt-in.
> 
> Bug: webrtc:10742
> Change-Id: I57159a3bf853ef8de32f5ba70ad4fa897ee5c528
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144046
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28631}

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

Change-Id: Ie9ce099d9154551fd123dbc1eca96ca2730fbc9f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10742
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146544
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28634}
2019-07-22 12:55:52 +00:00
Sebastian Jansson
c36c8e114d Makes send side network estimation opt-in.
Bug: webrtc:10742
Change-Id: I57159a3bf853ef8de32f5ba70ad4fa897ee5c528
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144046
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28631}
2019-07-22 12:17:29 +00:00
Sebastian Jansson
ea3dddf1d0 Use capacity bounds in AimdRateControl if available.
Bug: webrtc:10742
Change-Id: I139f7053d33092efe6430d82596b05c730855383
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145725
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28584}
2019-07-17 07:19:38 +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
Artem Titov
386802ef7c Move network emulation framework under test/network
Bug: webrtc:10138
Change-Id: I654bc124866241ceca65462937e2fad6294cc62b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144622
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28492}
2019-07-05 11:08:42 +00:00
Sebastian Jansson
dd4dc7a18e Adds additional fields to NetworkStateEstimate.
This is part of a series of CLs with the goal to make the network
state estimate fields more semantically useful.

Bug: webrtc:10742
Change-Id: I3185dc8c4574710a4dce44a4ffd6cefe0fb4e8cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144036
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28406}
2019-06-27 18:25:12 +00:00
Sebastian Jansson
49167de0be Adds interface for remote network estimates to NetworkControllerInterface.
Bug: webrtc:10742
Change-Id: I593fc17ce5d42c5dc17fd289f0621230319f9752
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144039
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28405}
2019-06-27 16:57:32 +00:00
Sebastian Jansson
88290ae358 Reland "Cleanup of RTP references in GoogCC implementation."
This is a reland of fa79081dca9faa8322943641352d9d2fd1b1b445

It crashed due to inability to handle small timestamps in probe
estimator. This was fixed by moving history window check to avoid
subtracting from the timestamp.

Original change's description:
> Cleanup of RTP references in GoogCC implementation.
>
> As the send time congestion controller now has been removed,
> we don't need the RTP related constructs anymore.
>
> Bug: webrtc:9510
> Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28330}

Bug: webrtc:9510
Change-Id: I3bf91222068e4fbb6aa159bfeb7a73e00bb6a0d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143165
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28347}
2019-06-24 09:10:52 +00:00
Sebastian Jansson
7953ad5dab Revert "Cleanup of RTP references in GoogCC implementation."
This reverts commit fa79081dca9faa8322943641352d9d2fd1b1b445.

Reason for revert: Breaks downstream project.

Original change's description:
> Cleanup of RTP references in GoogCC implementation.
> 
> As the send time congestion controller now has been removed,
> we don't need the RTP related constructs anymore.
> 
> Bug: webrtc:9510
> Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28330}

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

Change-Id: I562365fc5d1da68326d603338ccc6371114d7e12
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9510
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143164
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28331}
2019-06-20 10:21:51 +00:00
Sebastian Jansson
fa79081dca Cleanup of RTP references in GoogCC implementation.
As the send time congestion controller now has been removed,
we don't need the RTP related constructs anymore.

Bug: webrtc:9510
Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28330}
2019-06-20 10:08:29 +00:00
Sebastian Jansson
23026ee6c3 Adds SortedByReceiveTime to TransportPacketsFeedback.
This is part of a series of CLs removing RTP dependencies from GoogCC
implementation.

Bug: webrtc:10749
Change-Id: I4d8ec92643f1a0d4b5fdac2fbcb067766afc6548
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142222
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28300}
2019-06-17 17:43:27 +00:00
Sebastian Jansson
3d61ab12e6 Adds send time to ReceivedPacket struct.
Bug: webrtc:10742
Change-Id: I7e83d5ec2e23d1db38d02a0c883466ecdcd387c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141874
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28290}
2019-06-14 15:01:36 +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
3525f86c42 Adds feedback generator.
This is a useful tool to use for unittests of code that uses
TransportFeedback as input.

Bug: webrtc:10498
Change-Id: I171b22841eb9e16a5d5b785ff45ae9df5a6ccd7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137423
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27975}
2019-05-17 16:14:32 +00:00
Sebastian Jansson
e847481dc8 Adds debug printing of network estimate.
Bug: webrtc:10498
Change-Id: Idce952675ef079b5981f973ca58ca2cd7e5d5332
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134648
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27838}
2019-05-03 10:08:46 +00:00
Sebastian Jansson
2db5fc00c0 Deprecating injection of event log into GoogCC factory.
Bug: webrtc:9883
Change-Id: I6087b4a0a2c934e6a9ab435fffaf2eb1fc2a29e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134644
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27824}
2019-05-02 11:05:17 +00:00
Sebastian Jansson
7ccaf8969d Cleanup of network controller handling in Scenario tests.
Removing functionality to choose congestion controller implementation,
using injection instead. Also cleaning up some related functionality
that's no longer needed, such as the injection of event logs into the
factory.

Bug: webrtc:9883
Change-Id: Ia528005625430ae31a15bc88881e2d4ac6ad1d42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133890
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27768}
2019-04-25 12:40:00 +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
Mirko Bonadei
66e7679fb8 Export symbols needed by the Chromium component build (part 8).
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).

[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md

Bug: webrtc:9419
Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27410}
2019-04-02 10:13:36 +00:00
Sebastian Jansson
11e55ee90a Renaming min_pacing_rate to min_total_allocated_bitrate.
This prepares for upcoming CL using the value for more than
controlling pacing rates.

Bug: webrtc:9887
Change-Id: Id3891c3727865149b87f946b3e7c3095a6ac9f26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126001
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27004}
2019-03-06 18:09:16 +00:00
Sergey Silkin
e049eba27c Revert "Add Sender and Receiver interfaces for MediaTransport audio"
This reverts commit 0d8eed6ac77fadf7f9bcf70c671710d60b1ee62d.

Reason for revert: crashes of unit tests.

Original change's description:
> Add Sender and Receiver interfaces for MediaTransport audio
> 
> Implement in LoopbackMediaTransport.
> 
> Bug: webrtc:9719
> Change-Id: I429ac3f78d99b8ea4f9ac85b9a3600b215b61a55
> Reviewed-on: https://webrtc-review.googlesource.com/c/121957
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26731}

TBR=solenberg@webrtc.org,nisse@webrtc.org,mellem@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com

Change-Id: I02e409e1bbe2b2dea8a7b1aa08fa44d4146bda8f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/123232
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26733}
2019-02-18 09:52:40 +00:00
Niels Möller
0d8eed6ac7 Add Sender and Receiver interfaces for MediaTransport audio
Implement in LoopbackMediaTransport.

Bug: webrtc:9719
Change-Id: I429ac3f78d99b8ea4f9ac85b9a3600b215b61a55
Reviewed-on: https://webrtc-review.googlesource.com/c/121957
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26731}
2019-02-18 08:51:26 +00:00
Piotr (Peter) Slatala
c39f462b2d Move RtcEventProbeClusterCreated to the network controller.
Originally RtcEventProbeClusterCreated was logged in bitrate prober. This means that anyone who was using GoogCcNetworkControl wasn't logging it, and the NetworkControl wasn't self-contained.
This changes moves the responsibility for logging ProbeClusterCreated to ProbeController (where the probe is created), it also moves the responsibility for assigning probe ids to the probe controller.

Bug: None
Change-Id: If0433cc6d311b5483ea3980749b03ddbcd2bf041
Reviewed-on: https://webrtc-review.googlesource.com/c/122927
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26713}
2019-02-15 16:42:47 +00:00
Niels Möller
7e0e44f8b4 Move video-related MediaTransport interfaces to their own file and target
Bug: webrtc:9719
Change-Id: I2cf4a8520ce5c07c76ab0310cf7ab0ab285d9e0c
Reviewed-on: https://webrtc-review.googlesource.com/c/122504
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26702}
2019-02-15 08:17:20 +00:00
Niels Möller
ec3b9ffdb0 Move audio-related MediaTransport interfaces to their own file and target
Bug: webrtc:9719
Change-Id: I8bef979e4073d51be7cb93d38ee0e2ae22baef0e
Reviewed-on: https://webrtc-review.googlesource.com/c/121942
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26594}
2019-02-08 01:58:14 +00:00
Sebastian Jansson
79f0d4d0c7 Enables feature to account for unacknowledged data.
By enabling this trial, we can also remove reporting of packet
feedback status from send streams that was used before.

Bug: webrtc:9718
Change-Id: I3e7c4656b0ac6592a834617e044f23a072454181
Reviewed-on: https://webrtc-review.googlesource.com/c/118281
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26363}
2019-01-23 10:00:52 +00:00
Sebastian Jansson
95edb037a4 Adds WebRtcKeyValueConfig interface
The WebRtcKeyValueConfig interface allows providing custom key value
configurations that changes per instance of GoogCcNetworkController.

Bug: webrtc:10009
Change-Id: I520fff030d1c3c755455ec8f67896fe8a6b4d970
Reviewed-on: https://webrtc-review.googlesource.com/c/116989
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26312}
2019-01-18 08:45:08 +00:00
Per Kjellander
79a07cd9f6 Change type StreamsConfig::requests_alr_probing to abls::optional
That means it does not have to be set on every update of StreamsConfig.

BUG=webrtc:9586

Change-Id: I6a348160e209042857c4475323466e2aa92adef8
Reviewed-on: https://webrtc-review.googlesource.com/c/116690
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26184}
2019-01-10 06:12:05 +00:00
Christoffer Rodbro
c610e26be5 Include pacing buffer size in congestion window.
Bug: webrtc:10171
Change-Id: I9e21880a8b6f325415b62397081c301ee904f2ea
Reviewed-on: https://webrtc-review.googlesource.com/c/116068
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26175}
2019-01-09 11:16:58 +00:00