Some frames are already marked as 'timing frames' via video-timing RTP header extension. Timestamps along full WebRTC pipeline are gathered for these frames. This CL implements reporting of these timestamps for a single
timing frame since the last GetStats(). The frame with the longest end-to-end delay between two consecutive GetStats calls is reported.
The purpose of this timing information is not to provide a realtime statistics but to provide debugging information as it will help identify problematic places in video pipeline for outliers (frames which took longest to process).
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2946413002
Cr-Commit-Position: refs/heads/master@{#18909}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
This CL finalizes the support for allowing an external
audio processing module to be used in a peerconnection.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2965703002
Cr-Commit-Position: refs/heads/master@{#18864}
Can be enabled by setting "enable_encrypted_rtp_header_extensions" in
"crypto_options" of "PeerConnectionFactoryInterface::Options" and will
only be used if both peers support it.
BUG=webrtc:3411
Review-Url: https://codereview.webrtc.org/2761143002
Cr-Commit-Position: refs/heads/master@{#18842}
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
Was only working when the nonstandard "renomination" extension to ICE
is enabled, which chromium doesn't use.
BUG=chromium:734094
Review-Url: https://codereview.webrtc.org/2957303002
Cr-Commit-Position: refs/heads/master@{#18814}
This is in preparation of upgrading to libsrtp 2.1.0.
BUG=webrtc:7856
Review-Url: https://codereview.webrtc.org/2958123002
Cr-Commit-Position: refs/heads/master@{#18805}
There is a dependency chain from Chromium windows main_dll to Opus
which should never exist. We used to rely on rtc_static_library
to break this chain. So this CL replaced some rtc_source_set
with rtc_static_library.
libvpx fix (https://chromium-review.googlesource.com/c/544107/) for
ios-simulator linking issue is landed and this CL can be sumbitted once the new
Chromium is rolled into WebRTC.
BUG=chromium:734631
Review-Url: https://codereview.webrtc.org/2947273002
Cr-Commit-Position: refs/heads/master@{#18709}
Reason for revert:
Relanding the orginal CL. The breakage would be a flakey build.
Original issue's description:
> Revert of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2945233002/ )
>
> Reason for revert:
> The Android 32 (more config) bot is broken.
>
> Original issue's description:
> > Try to fix the binary size increase issue on Chromium.
> >
> > The target common_video used to depend on rtc_media_base which introduces
> > the dependency on p2p. This probably causes the binary size increase on Win
> > Chromium. Some chromium targets like src/media/gpu:gpu depends on common_video directly.
> >
> > BUG=chromium:734631
> >
> > Review-Url: https://codereview.webrtc.org/2945233002
> > Cr-Commit-Position: refs/heads/master@{#18693}
> > Committed: 9ed1609737
>
> TBR=kjellander@webrtc.org,deadbeef@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:734631
>
> Review-Url: https://codereview.webrtc.org/2949953003
> Cr-Commit-Position: refs/heads/master@{#18694}
> Committed: c2e208a924TBR=kjellander@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:734631
Review-Url: https://codereview.webrtc.org/2949883003
Cr-Commit-Position: refs/heads/master@{#18695}
Reason for revert:
The Android 32 (more config) bot is broken.
Original issue's description:
> Try to fix the binary size increase issue on Chromium.
>
> The target common_video used to depend on rtc_media_base which introduces
> the dependency on p2p. This probably causes the binary size increase on Win
> Chromium. Some chromium targets like src/media/gpu:gpu depends on common_video directly.
>
> BUG=chromium:734631
>
> Review-Url: https://codereview.webrtc.org/2945233002
> Cr-Commit-Position: refs/heads/master@{#18693}
> Committed: 9ed1609737TBR=kjellander@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:734631
Review-Url: https://codereview.webrtc.org/2949953003
Cr-Commit-Position: refs/heads/master@{#18694}
The target common_video used to depend on rtc_media_base which introduces
the dependency on p2p. This probably causes the binary size increase on Win
Chromium. Some chromium targets like src/media/gpu:gpu depends on common_video directly.
BUG=chromium:734631
Review-Url: https://codereview.webrtc.org/2945233002
Cr-Commit-Position: refs/heads/master@{#18693}
In this case it wasn't an issue, because only one result would be found
by remove_if, but might as well fix it just in case.
BUG=None
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2945723002
Cr-Commit-Position: refs/heads/master@{#18641}
This CL makes the WebRTC more modular and allows the users to build
WebRTC without audio and video(DataChannel only).
The BUILD files in call/, logging/, media/ and pc/ are modified to
support modular WebRTC.
The dependencies on Call and RtcEventLog are removed from the
PeerConnection. Instead of being created internally, they would be
passed in by the PeerConnectionFactory.
Add the CreateModularPeerConnectionFactory function which allow the
users to create a PeerConnectionFactory with the modules they need.
If the users want to build WebRTC without audio and video, they can
pass in null pointers for modules they don't need. (MediaEngine,
VideoEncoderFactory etc.)
BUG=webrtc:7613
Review-Url: https://codereview.webrtc.org/2854123003
Cr-Commit-Position: refs/heads/master@{#18617}
This eliminates a thread hop in PeerConnectionFactory initialization,
and will allow some code to be simplified.
BUG=None
Review-Url: https://codereview.webrtc.org/2934103002
Cr-Commit-Position: refs/heads/master@{#18613}
PeerConnection::SetBitrate calls PeerConnectionFactory::worker_thread
from multiple threads, so it was triggering the DCHECK. However, the
worker thread never changes after construction, so worker_thread should
be safe to call from multiple threads.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2923953004
Cr-Commit-Position: refs/heads/master@{#18576}
WebRtcVideoChannel and and WebRtcVideoEngine seem to have been removed, and only WebRtcVideoChannel2 and WebRtcVideoEngine2 remain, which removes the need for the "2" postfix.
BUG=None
Review-Url: https://codereview.webrtc.org/2932073002
Cr-Commit-Position: refs/heads/master@{#18531}
Reason for revert:
Broken downstream project.
Original issue's description:
> Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
>
> BUG=webrtc:7395
>
> Review-Url: https://codereview.webrtc.org/2888303005
> Cr-Commit-Position: refs/heads/master@{#18417}
> Committed: 9641c13327TBR=deadbeef@webrtc.org,stefan@webrtc.org,kwiberg@webrtc.org,solenberg@webrtc.org,holmer@google.com,zstein@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7395
Review-Url: https://codereview.webrtc.org/2914413002
Cr-Commit-Position: refs/heads/master@{#18420}
Reason for revert:
Broken downstream projects
Original issue's description:
> Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls.
>
> Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
>
> BUG=webrtc:5079
> R=deadbeef@webrtc.org, hbos@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2863123002 .
> Cr-Commit-Position: refs/heads/master@{#18384}
> Committed: e80f4c91d0TBR=hbos@webrtc.org,deadbeef@webrtc.org,holmer@google.com,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2916793003
Cr-Commit-Position: refs/heads/master@{#18386}
Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
BUG=webrtc:5079
R=deadbeef@webrtc.org, hbos@webrtc.org
Review-Url: https://codereview.webrtc.org/2863123002 .
Cr-Commit-Position: refs/heads/master@{#18384}
The change is now compatible with the old JVM::Initialize API. The
context is passed to the ContextUtils class when calling its deprecated
signature.
BUG=webrtc:7665
NOTRY=True # Only comment changes since the last patchset.
Review-Url: https://codereview.webrtc.org/2903253004
Cr-Commit-Position: refs/heads/master@{#18268}
This CL doesn't yet offer these protos; it just accepts them if they're
seen in a remote offer. It also doesn't verify that the ICE candidate
protocol matches the m= section protocol (UDP vs. TCP), since we don't
do this elsewhere and don't really have a reason to care.
This CL also adds an integration test that receives a spec-compliant
SCTP offer and attempts to send data bidirectionally.
BUG=webrtc:7706
Review-Url: https://codereview.webrtc.org/2902213002
Cr-Commit-Position: refs/heads/master@{#18265}
The media related stats wasn't working for unsignaled stream because there
is no mapping between the receiver_info and unsignaled tracks.
This CL fixes the issue by adding some special logic to the TrackMediaInfoMap
which would create the mapping.
BUG=b/37836881
BUG=webrtc:7685
TBR=deadbeef@webrtc.org
Review-Url: https://codereview.webrtc.org/2883943003
Cr-Commit-Position: refs/heads/master@{#18217}
Add the include to the files where it is actually used instead.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2869863003
Cr-Commit-Position: refs/heads/master@{#18176}
The only thing the physical socket server was used for was
"Wait"/"WakeUp", but it could be replaced by a simple rtc::Event.
So, removing this dependency makes things less confusing; the fact that
VirtualSocketServer takes a PhysicalSocketServer may lead someone to
think it uses real sockets internally, when it doesn't.
BUG=None
Review-Url: https://codereview.webrtc.org/2883313003
Cr-Commit-Position: refs/heads/master@{#18172}
For every failing test, the solution was either to do a "has IPv4" check
before the test is run, or avoid depending on real network interfaces
altogether.
This specifically fixes rtc_unittests, peerconnection_unittests, and
webrtc_nonparallel_tests.
BUG=None
Review-Url: https://codereview.webrtc.org/2881973002
Cr-Commit-Position: refs/heads/master@{#18155}
Instead, make the pointer to the associated socket server a
construction time const, and delete its lock.
Introduces a helper class AutoSocketServerThread for code
(mainly tests) which need a socket server associated with
the current thread.
BUG=webrtc:7501
Review-Url: https://codereview.webrtc.org/2828223002
Cr-Commit-Position: refs/heads/master@{#18047}
The test attempted to call InsertDtmf before verifying that the DTLS
handshake was complete. Unlike a data channel, the DTMF sender doesn't
do any buffering, so this isn't reliable.
BUG=webrtc:7547
NOTRY=True
Review-Url: https://codereview.webrtc.org/2855573004
Cr-Commit-Position: refs/heads/master@{#18041}
The test contained an invalid IPv6 address. It should have ":" instead of "::" as separation.
BUG=webrtc:7565
Review-Url: https://codereview.webrtc.org/2868453002
Cr-Commit-Position: refs/heads/master@{#18035}
Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2799033006
Cr-Commit-Position: refs/heads/master@{#17977}
Specifically, the tests that only use data channels shouldn't need any
audio codec support; by using an audio decoder factory that supports
no codecs, we ensure that this is the case.
For completeness, I tried doing the same to the two tests that
actually use audio and video; as expected, they fail, with messages
like this:
[000:032] (webrtcsession.cc:334): Failed to set remote sdp: Session
error code: ERROR_CONTENT. Session error description: Failed to set
local audio description recv parameters..
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2848563002
Cr-Commit-Position: refs/heads/master@{#17907}
