Alr state is now logged by the pacer. To avoid confusion,
loopback tools will now create two separate rtc event
logs for sender and receiver calls.
Bug: webrtc:8287, webrtc:8588
Change-Id: Ib3e47d109c3a65a7ed069b9a613e6a08fe6a2f30
Reviewed-on: https://webrtc-review.googlesource.com/26880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21084}
Usage should now be removed and this change can be relanded.
It was reverted here: https://webrtc-review.googlesource.com/c/src/+/27200
NOTRY=TRUE
TBR=solenberg
Bug: webrtc:7306
Change-Id: I5191263e6cfd48952b59ff8f9af2e59c3e9eadef
Reviewed-on: https://webrtc-review.googlesource.com/29682
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21080}
CL https://webrtc-review.googlesource.com/c/src/+/28120 removed a
public dependency from rtc_tools:video_quality_analysis on
common_video:common_video.
This was keeping the MSVC64(dbg) build green because was giving the
linker the opportunity to find api:optional symbols.
This CL tries to fix and adds a TODO to remove the synthetic
dependency. The dependency on api:optional should be added to
rtc_base:rtc_base_approved_generic but this triggers another
dependency cycle.
TBR=tommi@webrtc.org
Bug: webrtc:6828
Change-Id: I4e28b49fdb3ee6484a253ca7b1f1a8aafa20e915
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/29683
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21079}
There's no downstream use of kFileFormatCompressedFile,
kFileFormatPreencodedFile or kFileFormatPcm48kHzFile.
Bug: None
Change-Id: I66cbe71151472d6348515a2432a280acbc3bbf85
Reviewed-on: https://webrtc-review.googlesource.com/28040
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21078}
After changing from TEST_F to TEST_P the MAYBE_InitialProbing macro was not
expanded as expected, causing the test to be enabled on all bots with the name
MAYBE_InitialProbing.
Bug: None
Change-Id: Icfb0c4b381510c1b73295f017ebb68d43b7d9809
Reviewed-on: https://webrtc-review.googlesource.com/29640
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21074}
This rewrites UpdateSessionState to better show the logic common
to all description types and the logic specific to
offers/answers/etc. Separating these will allow more code to be
reused with the Unified Plan implementation.
Bug: webrtc:8587
Change-Id: I56e0370dcb8bb4b59af2a5209edcad4606480e1c
Reviewed-on: https://webrtc-review.googlesource.com/27322
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@{#21065}
PeerConnection had an Action enum as a holdover from the
WebRtcSession merge with the same members as
cricket::ContentAction. Since ContentAction is used in more places
outside of PeerConnection, this change removes the Action enum and
replaces its use with cricket::ContentAction.
Bug: webrtc:8587
Change-Id: I3e825fe285dbaf6b3f128eccde0f38864171af13
Reviewed-on: https://webrtc-review.googlesource.com/27321
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@{#21063}
Calls to SetLocalDescription and SetRemoteDescription in
PeerConnection delegate to many different internal helper methods
which can fail. The error ultimately needs to propagate to the
caller and cause the SetXXXDescription to fail. Right now these
methods signal errors by returning false and copying the error
message into an out parameter. This changes these methods to
return RTCError instead and avoid the use of the out parameter.
Bug: webrtc:8587
Change-Id: Ib1d31622be742718b74780110c1bbe273d66444e
Reviewed-on: https://webrtc-review.googlesource.com/27241
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@{#21061}
Also renames methods for interacting with the session error. This
clarifies the scope of this error type and lets methods have a
local variable named |error| without confusing it with the
|error()| getter.
Bug: webrtc:8587
Change-Id: I90e6eed24d961abbce15e56a76a8793ff1a806ea
Reviewed-on: https://webrtc-review.googlesource.com/27124
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@{#21060}
Eventually we want BaseChannel to depend on the RtpTransportInternal
instead of DtlsTransportInternal and share RtpTransport when bundling.
This CL is the first step.
Add SetRtpTransport and Init_w(RtptransportInternal*) to BaseChannel.
These two methods would replace the existing SetTransports and Init_w
methods.
Add new CreateVoice/VideoChannel methods to the ChannelManager which
take RtpTransportInternal instead of Dtls/PacketTransportInternal.
|cotnent_name| is removed from the SrtpTransport to simplify to code
since it is only used for debugging.
InitNetwork_n is removed from BaseChannel in CL as well.
Bug: webrtc:7013
Change-Id: I35b1565958548bd4896854c49e61d3ee160b7634
Reviewed-on: https://webrtc-review.googlesource.com/27840
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21057}
Removing the check as they were causing some tests to fail.
Bug: none
Change-Id: I42878d93a3239b18e3807a77bffc597794b65bf1
TBR: ossu@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/29300
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21055}
When the initial offer side uses the ICE lite implementation, and
initiates a peer connection with an endpoint with the full
implementation, the offer side assumes the controlled ICE role per
RFC5245 and the remote endpoint MUST take the controlling role.
This logic was partially implemented in SetRemoteTransportDescription in
reflection where the endpoint switches its role to the controlling after
receiving the offer. The bug was caused by the following
SetLocalDescription at the remote endpoint after creating the answer,
which overrides the role to the controlled since it has no initial offer
and the role is not reflected in SetLocalTransportDescription. This
results in no nomination of candidate pairs and timeout of establishing
the peer connection.
The fix adds reflection on one's ICE role in SetLocalTransportDescription.
This fix also takes into account the case when both sides use the lite
implementation of ICE and the initial offer side MUST take the controlling
role per RFC5245 in this case, which is the default behavior in the
current implementation.
Bug: webrtc:8531
Change-Id: I65edd296c155bff51fcdb28709975e6837f302d5
Reviewed-on: https://webrtc-review.googlesource.com/26780
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#21053}
This refactoring reduces code duplication in PeerConnection and
will make it easier to use these methods with the Unified Plan
implementation.
Bug: webrtc:8587
Change-Id: I6afd44fff702290903555cbe7703198b6b091da6
Reviewed-on: https://webrtc-review.googlesource.com/26822
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@{#21052}
This changes the CreateVoiceChannel/CreateVideoChannel helper
methods in PeerConnection to return the created channel instead of
setting it directly. That allows the Unified Plan version of
SetLocalDescription to use the same factory methods without the
assumption that there is at most one voice and one video channel.
Also simplifies and deduplicates the logic for determining the
transport name for a given channel in the presence of BUNDLE.
Bug: webrtc:8587
Change-Id: I1f156f45309ce2d08d6d5d5ed3c6e01fbf094b36
Reviewed-on: https://webrtc-review.googlesource.com/26821
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@{#21050}
The problem with using an Event for the 'running_' flag is that a Wait() is necessary to check the value of the flag and calling Wait() means a synchronous system call. In chromium that can furthermore trigger checks on threads where IO/blocking operations aren't allowed.
Luckily, we don't really need this variable. Instead, I'm adding thread checks to make sure that the Thread class is used correctly and ensure that locking isn't needed for modifying state (no locks are there now). As long as we follow those rules, we only need to check if a thread_ variable has been set when we want to know if the thread is running or not.
Along the way, fixed some places where variables weren't being set or reset correctly.
Bug: webrtc:8596
Change-Id: I1467542416bc2ffbfefe276c460e76078a759bd9
Reviewed-on: https://webrtc-review.googlesource.com/27720
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21042}
A lot of WebRTC targets were depending on //third_party/libyuv using
public_deps instead of deps. This causes issues because a the
inclusion of libyuv headers is not declared to the build system and
this creates hidden dependencies that put the modularity of the project
at risk.
Bug: webrtc:8603
Change-Id: Ide0ceb84eb5640ae664dc782f3a722b55c3b601a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28120
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21039}
In particular, PlayoutStereoData and StartPlayingAudioFile. This also
eliminates the dependency on system_wrappers FileWrapper.
Bug: None
Change-Id: I61df1eea1ad5f5035e36c8229febbf3668808f65
Reviewed-on: https://webrtc-review.googlesource.com/28121
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21038}
