The usage of MetricsObserverInterface to log metrics has been replaced
by RTC_HISTOGRAM_* macros in WebRTC.
Bug: webrtc:9409
Change-Id: I67df74a18942ac7ea4227e4affdf84f06258a287
Reviewed-on: https://webrtc-review.googlesource.com/86780
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24048}
Generated using gmock_gen.py with some editing.
This mock doesn't seem to be used by unittest in webrtc, but we need to use it in downstream unittests.
Bug: None
Change-Id: Ia7904ffdd22f3d16fe5fd515fa68833817b44481
Reviewed-on: https://webrtc-review.googlesource.com/85780
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23900}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This removes the SessionStats object and replaces it with two
methods on PeerConnection: GetTransportNamesByMid and
GetTransportStatsByNames for use by the stats collectors. These
methods are more flexible and can cover cases where there are more
than one video/audio channel.
Bug: webrtc:8764
Change-Id: Id400cc548fc43675462ff6175a7fa9c9f4fd5948
Reviewed-on: https://webrtc-review.googlesource.com/47244
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21921}
This provides an intermediate class for defining default, null
implementations of all the PeerConnectionInterface/
PeerConnectionInternal methods. Specific fake PeerConnections then can
inherit from this and only override the methods pertaining to the
scenarios it will be used in.
Bug: webrtc:8764
Change-Id: I7614303b5673747244053b54b839e58aada43d10
Reviewed-on: https://webrtc-review.googlesource.com/43245
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21752}
Moves ownership of the RtpSenders/RtpReceivers/BaseChannels to
RtpTransceiver objects. For now, there can only be one
RtpTransceiver for audio and one for video. Future work to
implement Unified Plan will relax this restriction.
Bug: webrtc:7600
Change-Id: I9dfe324de61e2b363948858da72624396e27fc1a
Reviewed-on: https://webrtc-review.googlesource.com/21461
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20802}
This literally copies & pastes the code from WebRtcSession into
PeerConnection as private methods. The only other changes were to
inline the WebRtcSession construction/initialization/destruction
into PeerConnection and fix issues using rtc::Bind on the
reference-counted PeerConnection.
Bug: webrtc:8323
Change-Id: Ib3f071ac10d18566a21a3b04813b1d4ec691ef3c
Reviewed-on: https://webrtc-review.googlesource.com/15160
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20574}
This commit prepares WebRtcSession so that it can be cleanly
copy & pasted into PeerConnection in the next commit. To accomplish
this, the following was done:
1. Added a pointer to the owning PeerConnection to WebRtcSession.
2. Replace WebRtcSession state enum with signaling state.
3. All signals/observers only observed by PeerConnection were
replaced with direct calls to PeerConnection methods.
4. All duplicated fields were moved to PeerConnection.
5. The remaining tests that still use WebRtcSession for mocks were
updated to minimize dependence on WebRtcSession construction.
Bug: webrtc:8323
Change-Id: Ifc1a4ee819dcc9beca5363291012f7d5563ff7b1
Reviewed-on: https://webrtc-review.googlesource.com/9020
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20573}
Enable cpplint check in the PRESUBMIT for pc/ and fix all existing
warnings.
Bug: webrtc:5583
Change-Id: If39994692ab6f6f3c83c74f23850f02fdfe810e8
Reviewed-on: https://webrtc-review.googlesource.com/16540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20482}
This change is part of the work to merge WebRtcSession into
PeerConnection. To make that work easier, this moves all clients
of WebRtcSession to use shims added to PeerConnection. That way
when the classes are merged they won't need to be modified.
Bug: webrtc:8183
Change-Id: I5758a5954b91d235faf810c8bf4bf9f6f31d83c1
Reviewed-on: https://webrtc-review.googlesource.com/5040
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20090}
This reverts commit 3dc4d4a21f80cdf44c508412d784b254957696eb.
Reason for revert: breaks internal project
Original change's description:
> Move clients of WebRtcSession to use PeerConnection
>
> This change is part of the work to merge WebRtcSession into
> PeerConnection. To make that work easier, this moves all clients
> of WebRtcSession to use shims added to PeerConnection. That way
> when the classes are merged they won't need to be modified.
>
> Bug: webrtc:8183
> Change-Id: I43de7acf7e38c9fcf2dbf55d50eb05e73767c251
> Reviewed-on: https://webrtc-review.googlesource.com/4320
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20030}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org
Change-Id: I13f335b24c26753429cd08a4ca3e295eed5660ff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8183
Reviewed-on: https://webrtc-review.googlesource.com/4700
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20035}
This change is part of the work to merge WebRtcSession into
PeerConnection. To make that work easier, this moves all clients
of WebRtcSession to use shims added to PeerConnection. That way
when the classes are merged they won't need to be modified.
Bug: webrtc:8183
Change-Id: I43de7acf7e38c9fcf2dbf55d50eb05e73767c251
Reviewed-on: https://webrtc-review.googlesource.com/4320
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20030}
The video codec factories should be owned by the video engine instead
of by the PeerConnectionFactory.
Bug: None
Change-Id: If63d47cef565138d51377af3fc9ea973950c9390
Reviewed-on: https://webrtc-review.googlesource.com/1601
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20002}
We currently pass in a lot of audio parameters to PeerConnectionFactory
which we never use. This CL removes them.
All these parameters are reference counted, so they are not needed for
lifetime management (unless we do something crazy). Even if we want to
switch from reference counting to std::unique_ptrs in the future, the
voice engine is a more suitable owner than PeerConnectionFactory. The
PeerConnectionFactory already owns a MediaEngine which in turn owns a
VoiceEngine.
Bug: webrtc:7613
Change-Id: I393cf0d29ffa762a3a13475f6fbe00b8565f4c07
Reviewed-on: https://webrtc-review.googlesource.com/1600
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19931}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
