Commit Graph

157 Commits

Author SHA1 Message Date
6e436d1cc0 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo
This is part of implementing RTCRemoteInboundRtpStreamStats. The CL was
split up into smaller pieces for reviewability. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats

In [1], ReportBlockData was implemented and tested.
This CL adds the plumbing to make it available in MediaSenderInfo for
audio streams, but the code is not wired up to make use of these stats.

In follow-up CL [2], ReportBlockData will be used to implement
RTCRemoteInboundRtpStreamStats. The follow-up CL will add integration
tests exercising the full code path.

[1] https://webrtc-review.googlesource.com/c/src/+/136584
[2] https://webrtc-review.googlesource.com/c/src/+/138067

Bug: webrtc:10455
Change-Id: Id8940090cc9c121e8f06ccdb068a22ce24c07092
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138066
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28072}
2019-05-27 12:40:22 +00:00
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
d8b9ed77cf Promote RtcEventLogOutputFile to api/
Preparation for deleting PeerConnectionInterface::StartRtcEventLog
method with a PlatformFile argument.

Bug: webrtc:6463
Change-Id: Ia9fa1d99a3d87f3bf193e73382690b782ffea65c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135285
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27879}
2019-05-08 12:29:42 +00:00
ce9281794f Split test:test_common source set
To remove dependency vp9_replay_fuzzer -> test/call_test -> DefaultTaskQueueFactory
that blocks chromium import

Bug: None
Change-Id: Iab843eaa789b234d8842074d46fb3198ba67075e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134109
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27751}
2019-04-25 07:35:49 +00:00
d3ba236686 Stop using GlobalTaskQueueFactory in video unittests
instead use DefaultTaskQueueFactory directly

Bug: webrtc:10284
Change-Id: I58ae120cf185553d0145d7feb365deca90a93bc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132401
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27610}
2019-04-15 09:24:18 +00:00
304ea5f7b0 Add RtcEventLogFactory factory with explicit TaskQueueFactory
remove RtcEventLog factory function that relies on GlobalTaskQueueFactory,
move that default behaviour up to RtcEventLogFactory level.

Bug: webrtc:10284
Change-Id: I512d8a13e6a2f320000dd08e6355c0a7e9de8561
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132542
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27573}
2019-04-11 16:05:09 +00:00
898395d181 RTPSenderVideo::GetSentRtpPacketInfo() over a set of sequence numbers
Add a version of RTPSenderVideo::GetSentRtpPacketInfo() that operates
over a set of numbers, so as to only grab the lock once.

Bug: webrtc:10501
Change-Id: I9453b0cb44dcd6e2ce196390b2c5c9a7dd6d800a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132014
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27544}
2019-04-10 14:32:00 +00:00
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
b55015e4e1 Replacing SequencedTaskChecker with SequenceChecker.
Bug: webrtc:9883
Change-Id: I5e3189da2a46e6f4ed1a3c5a5dfd2f7d75a16b5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130961
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27518}
2019-04-09 12:28:04 +00:00
9d8eaac4ee Delete unneeded direct includes of common_types.h
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:

api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/

There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.

Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
2019-04-01 07:18:13 +00:00
94b57c044e Cleanup BUILD.gn files from imports like foo:foo
Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn
files.

Done with Sublime regex replace.
Find: \b([-a-zA-Z0-9_]+):+\1\b
In: *.gn
Replace with: \1

Bug: None
Change-Id: I40aba1b14face687a595b852ffe443cb20197611
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:05:28 +00:00
ef1052a134 Reland "Move api/rtp_headers.h to its own build target."
This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e

Original change's description:
> Move api/rtp_headers.h to its own build target.
>
> Reduces dependencies on the libjingle_peerconnection_api target from
> lower-level code.
>
> Bug: None
> Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27213}

Bug: None
Tbr: kwiberg@webrtc.org
Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-21 09:17:07 +00:00
2baef3509f Revert "Move api/rtp_headers.h to its own build target."
This reverts commit a67050debcb5a3461a452a7928d7aaea1562747e.

Reason for revert: breaks downstream projects

Original change's description:
> Move api/rtp_headers.h to its own build target.
> 
> Reduces dependencies on the libjingle_peerconnection_api target from
> lower-level code.
> 
> Bug: None
> Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27213}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org

Change-Id: I8cccaa8be1700ca8db141db7252eb6ce588ba2e0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128645
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27215}
2019-03-20 16:47:30 +00:00
a67050debc Move api/rtp_headers.h to its own build target.
Reduces dependencies on the libjingle_peerconnection_api target from
lower-level code.

Bug: None
Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27213}
2019-03-20 16:00:49 +00:00
5fe9510efb Move ownership of RTPSenderVideo one more level up, to RtpVideoSender
The idea is to let the RtpRtcp and RTPSender classes be responsible for
media-agnostic RTP transport, and move out the media-specific processing,
such as packetization and media-specific headers.

Bug: webrtc:7135
Change-Id: Ib0ce45bf06713b3eb6c06acd91c5168856874e4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123187
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26954}
2019-03-04 16:57:49 +00:00
ed50e6c759 Inject TaskQueueFactory in RtpTransportControllerSend.
Bug: webrtc:10365
Change-Id: I1656dcf774fb347afd8b28aa998acff8942cdd9f
Reviewed-on: https://webrtc-review.googlesource.com/c/125180
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26928}
2019-03-01 17:44:01 +00:00
74682c1191 Inject TaskQueueFactory to video streams.
Bug: webrtc:10365
Change-Id: Ib655d8eac4467926bcb86cf2cb3728eabf5342d8
Reviewed-on: https://webrtc-review.googlesource.com/c/125089
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26921}
2019-03-01 11:35:39 +00:00
fc52b912a3 Implicitly suppress //build/config/clang:find_bad_constructs.
Since there is no way to enable/disable these diagnostics at runtime,
this CL moves the suppression into the rtc_* templates in order to
remove the need to explicitly add the snippet of code needed to
suppress it (currently copy/pasted in 144 locations).

The diagnostic that causes the most problems is the one about "complex
class/struct explicit ctor/dtor" [1] because WebRTC doesn't find
it useful enough.

Other diagnostics are good (for example the one that warns about
using "virtual" instead of "override", but that will be covered by
this clang-tidy check [2]) while others are Chromium related so
they have never triggered.

[1] - https://cs.chromium.org/chromium/src/tools/clang/plugins/FindBadConstructsConsumer.cpp?l=147-167&rcl=b4bebe1aa15dba7ca5fcc6456a81a55665327c3a
[2] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html

Bug: webrtc:163
Change-Id: Icbf27efa5b369100a31e6a32df1a0913729b3b34
Reviewed-on: https://webrtc-review.googlesource.com/c/125088
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26918}
2019-03-01 10:18:17 +00:00
2b08e3188e Adds CoDel implementation to network simulation.
Adds an implementation of the CoDel active queue management algorithm
to the network simulation. It is loosely based on CoDel pseudocode
from ACMQueue: https://queue.acm.org/appendices/codel.html

Bug: webrtc:9510
Change-Id: Ice485be35a01dafa6169d697b51b5c1b33a49ba6
Reviewed-on: https://webrtc-review.googlesource.com/c/123581
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26834}
2019-02-25 09:54:03 +00:00
487c09bb73 Adds FakeNetworkPipeTest to rtc_unittests.
Also updating the tests so they pass. The previous target was not part
of the targets that runs on the try bots, so they started breaking
during a recent refactor. By adding it to the common target we'll avoid
regression like this in the future.

Bug: None
Change-Id: I55391d454bd52964875818be39cdaa853360ae81
Reviewed-on: https://webrtc-review.googlesource.com/c/123199
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26820}
2019-02-22 15:05:39 +00:00
836fee1e1a Calculate next process time in simulated network.
Currently there's an implicit requirement that users of
SimulatedNetwork should call it repeatedly, even if the return value
of NextDeliveryTimeUs is unset.

With this change, it will indicate that there might be a delivery in
5 ms at any time there are packets in queue. Which results in unchanged
behavior compared to current usage but allows new users to expect
robust behavior.

Bug: webrtc:9510
Change-Id: I45b8b5f1f0d3d13a8ec9b163d4011c5f01a53069
Reviewed-on: https://webrtc-review.googlesource.com/c/120402
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26617}
2019-02-08 19:33:17 +00:00
eceea31cc2 Reduces locking in SimulatedNetwork class.
Bug: webrtc:9883
Change-Id: I07c78fd2dbba5e7481194b0d1aabfb56809ff6fc
Reviewed-on: https://webrtc-review.googlesource.com/c/120612
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26491}
2019-01-31 12:03:43 +00:00
f380284035 (7) Rename files to snake_case: remove forwarding headers
Bug: webrtc:10159
Change-Id: I2ba899e0283b953538c7941c8790213e36d7c70e
Reviewed-on: https://webrtc-review.googlesource.com/c/118561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26417}
2019-01-26 00:33:46 +00:00
d970807e0c Remove rtc_base/scoped_ref_ptr.h.
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.

More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.

Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:29:58 +00:00
ecb6897ade Adds repeating task class.
This CL adds a single class to manage the use case of having a task
that repeats itself by a fixed or variable interval. It replaces the
repeating task previously locally defined for rtp transport controller
send as well as the cancelable periodic task. Furthermore, it is
introduced where one off repeating tasks were created before.

It provides the currently used functionality of the cancelable periodic
task, but not some of the unused features, such as allowing cancellation
of tasks before they are started and cancellation of a task after the
owning task queue has been destroyed.

Bug: webrtc:9883
Change-Id: Ifa7edee836c2a64fce16a7d0f682eb09c879eaca
Reviewed-on: https://webrtc-review.googlesource.com/c/116182
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26313}
2019-01-18 10:55:41 +00:00
aec15aa810 (5) Rename files to snake_case: install forwarding headers
Mechanically generated with this command:

tools_webrtc/do-renames.sh install all-renames.txt && git cl format

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: Ic8e99f71f2da62e5c99863c6d24a8cfe311466cd
Reviewed-on: https://webrtc-review.googlesource.com/c/115682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26227}
2019-01-11 17:13:36 +00:00
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
40d55331d7 Include absl/memory/memory.h if absl::make_unique is used
Tbr: kwiberg@webrtc.org
Bug: None
Change-Id: Iaf4533d2ce0e80b351a8a664ef8cf7ba0e5ec583
Reviewed-on: https://webrtc-review.googlesource.com/c/115746
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26168}
2019-01-08 20:08:32 +00:00
87609be863 Merges RtpTransportControllerSend with SendSideCongestionController.
Bug: webrtc:9586
Change-Id: I50332f2e128f107e40af7776be0ed530e20774d9
Reviewed-on: https://webrtc-review.googlesource.com/c/113183
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25922}
2018-12-06 13:38:39 +00:00
3e70781361 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.

Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 18:25:07 +00:00
6736df1778 Moves BitrateAllocationUpdate to api.
This way it can be forwarded to lower layers. This makes it easier to
add information without having to change signatures of intermediate
classes. This will be used in a later CL to use the link capacity in the
Opus decoder.

Bug: webrtc:9718
Change-Id: I4a4c9d104fedb0e4a0bb7f14d169475940edbf7e
Reviewed-on: https://webrtc-review.googlesource.com/c/111508
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25738}
2018-11-21 19:59:55 +00:00
13e5903626 Using unit classes in BitrateAllocationUpdate struct.
This prepares for moving BitrateAllocationUpdate to API.

Bug: webrtc:9718
Change-Id: Ib2bcedb6b68fde33b6a2466f40829e86438aa973
Reviewed-on: https://webrtc-review.googlesource.com/c/111507
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25737}
2018-11-21 19:56:15 +00:00
2222a80e79 Delete unneeded includes of common_types.h and gn deps on webrtc_common.
Bug: webrtc:5876
Change-Id: Iae14e5f1679067a5a5e0584ca830aee0870c8807
Reviewed-on: https://webrtc-review.googlesource.com/c/111463
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25715}
2018-11-20 16:28:39 +00:00
179a3923b9 Implement TargetBitrate, NetworkRoute and overhead features of media transport interface.
So far ANA was not available for media transport interface. With recent changes to media transport, we can now account for packet overhead, network route (ip/tcp/udp/turn overheads) and we can also use bandwidth estimate from the media transport.


Bug: webrtc:9719
Change-Id: I98c9a09dd418b763c339ee2ee05592e164cf9199
Reviewed-on: https://webrtc-review.googlesource.com/c/110367
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25677}
2018-11-16 19:31:11 +00:00
cc8e8bb73f Pass the media transport from JsepTransportController to Call.
Add TargetRateObservers for media transport in the call object.



Bug: webrtc:9719
Change-Id: I5448d05359cf09b8cd2a678b2ac876aa8f8970e7
Reviewed-on: https://webrtc-review.googlesource.com/c/110622
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25662}
2018-11-15 17:36:48 +00:00
c2ebe21ba9 Reland "Use the factory instead of using the builtin code path in VideoCodecInitializer"
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782

This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.

Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.

One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.

Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
2018-11-08 19:10:47 +00:00
b768e8800f Reland "Isolating APM API build target: making :api an actual target."
This reverts commit 61c6e5643e7ea058e653956980a90e033249c055.

Reason for revert: downstream projects prepared for this change

Original change's description:
> Revert "Isolating APM API build target: making :api an actual target."
> 
> This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.
> 
> Reason for revert: breaking downstream
> 
> Original change's description:
> > Isolating APM API build target: making :api an actual target.
> > 
> > This CL is part of a refactoring work to unblock other CLs
> > that would generate a circular dependency when including
> > modules/audio_processing. It will also allow to easily move
> > the APM interface part under //api.
> > 
> > More in detail, this change moves the APM interface files from
> > the build target modules/audio_processing to
> > modules/audio_processing:api. It also adds :api as dependency
> > where needed.
> > 
> > Bug: webrtc:9535
> > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25539}
> 
> TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
> 
> Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9535
> Reviewed-on: https://webrtc-review.googlesource.com/c/109820
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25540}

TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org

Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109884
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:30:06 +00:00
61c6e5643e Revert "Isolating APM API build target: making :api an actual target."
This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.

Reason for revert: breaking downstream

Original change's description:
> Isolating APM API build target: making :api an actual target.
> 
> This CL is part of a refactoring work to unblock other CLs
> that would generate a circular dependency when including
> modules/audio_processing. It will also allow to easily move
> the APM interface part under //api.
> 
> More in detail, this change moves the APM interface files from
> the build target modules/audio_processing to
> modules/audio_processing:api. It also adds :api as dependency
> where needed.
> 
> Bug: webrtc:9535
> Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25539}

TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org

Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109820
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25540}
2018-11-07 11:28:03 +00:00
a7f77a7c05 Isolating APM API build target: making :api an actual target.
This CL is part of a refactoring work to unblock other CLs
that would generate a circular dependency when including
modules/audio_processing. It will also allow to easily move
the APM interface part under //api.

More in detail, this change moves the APM interface files from
the build target modules/audio_processing to
modules/audio_processing:api. It also adds :api as dependency
where needed.

Bug: webrtc:9535
Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
Reviewed-on: https://webrtc-review.googlesource.com/c/109501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25539}
2018-11-07 10:34:51 +00:00
2cd3b4c819 Fixing bug in SimulatedNetwork where packets stop.
Bug: webrtc:9952
Change-Id: I68491f1d18fee317165999453776a35cea41e71f
Reviewed-on: https://webrtc-review.googlesource.com/c/109009
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25531}
2018-11-06 19:04:37 +00:00
59844ce57e Revert "Use the factory instead of using the builtin code path in VideoCodecInitializer."
This reverts commit be142178aaf6ab4089b4d81c88c3d59c12cca567.

Reason for revert: breaking internal projects

Original change's description:
> Use the factory instead of using the builtin code path in `VideoCodecInitializer`.
> 
> Bug: webrtc:9513
> Change-Id: Ia299ae1044a3ff4c91e208200938cba540bdcea6
> Reviewed-on: https://webrtc-review.googlesource.com/c/94782
> Commit-Queue: Jiawei Ou <ouj@fb.com>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25456}

TBR=brandtr@webrtc.org,magjed@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,tommi@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,tkchin@webrtc.org,shampson@webrtc.org,glaznev@webrtc.org,ouj@fb.com,qingsi@webrtc.org

Change-Id: I8040ccabe3ae6464d72c7696adb663c1dd275b63
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9513
Reviewed-on: https://webrtc-review.googlesource.com/c/108980
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25459}
2018-11-01 04:46:02 +00:00
be142178aa Use the factory instead of using the builtin code path in VideoCodecInitializer.
Bug: webrtc:9513
Change-Id: Ia299ae1044a3ff4c91e208200938cba540bdcea6
Reviewed-on: https://webrtc-review.googlesource.com/c/94782
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25456}
2018-10-31 22:47:02 +00:00
71822866c6 Allow FakeNetworkPipe to wake up its processing thread
Bug: webrtc:9630
Change-Id: I2b09593f175e3f3e1fe0d990515aa70c2481161b
Reviewed-on: https://webrtc-review.googlesource.com/c/95144
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25451}
2018-10-31 15:20:57 +00:00
69807e8871 Depend directly on destination targets.
Makes 'gn check' happy.
Followup to https://webrtc-review.googlesource.com/c/src/+/106820

Bug: webrtc:5876, webrtc:9855
Change-Id: I33fa2c31ba26dc10c9a9c17da0ffed255c1f4d5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108760
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25447}
2018-10-31 10:21:40 +00:00
21cddffd99 Harmonize paths to dependent targets.
This CL consistently use:
 * relative paths for WebRTC dependent targets (test_support)
 * absolute paths for shared dependent targets (abseil)
This is a necessary (but insufficient) step to build WebRTC tests
from Chromium tree (rtc_include_tests=true), since test/ doesn't
sit anymore in the top level directory.

We also make sure that target declarations and uses are
consistent in regard to build_with_chromium flag.

Bug: webrtc:9943
Bug: webrtc:9855
Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108720
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25445}
2018-10-31 10:04:59 +00:00
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00
f5e767dbbc Don't send max allocation probe unless allocation changed.
This changes the behavior to a probe only gets trigged if
the total max allocated bitrate  actually changed.

Also adding helpful log dump flag to ramp up tests that
was used to investigate the issue.

Bug: chromium:894434
Change-Id: I907675b8fd5a339f838b07d433ecf837e312def1
Reviewed-on: https://webrtc-review.googlesource.com/c/105981
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25212}
2018-10-16 15:13:57 +00:00
12985414b9 Removing unnecessary dependencies on socket.h.
Since rtc:SentPacket was removed to a separate header. Some usages of
socket.h can be replaced with sent_packet.h which defines a lot less
things, making future maintenance simpler.

Bug: webrtc:9586
Change-Id: If705edda293c389cf2a175117db52a6720a7be86
Reviewed-on: https://webrtc-review.googlesource.com/c/106144
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25201}
2018-10-16 10:24:51 +00:00
40a7a35eaa Extract functionality of test_main into separate library.
Extract functionality of test_main into separate library to be able to
reuse it if another main will be required.

Bug: webrtc:5996
Change-Id: I2925b4240bd0e4fb884b43bb16667ca2d6216bbd
Reviewed-on: https://webrtc-review.googlesource.com/c/105921
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25172}
2018-10-15 14:13:06 +00:00
76ad154eef New method for precise packet reception time measurement.
Bug: webrtc:9054
Change-Id: I43a32122e9af992b5e0ba8b187c9ad4f22aba80d
Reviewed-on: https://webrtc-review.googlesource.com/c/104503
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25167}
2018-10-15 11:25:26 +00:00