Also rename it from pacer_thread_ to congestion_controller_thread_.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2637783003
Cr-Commit-Position: refs/heads/master@{#16134}
Introduing IceTransportInternal2 is a temporary fix to switch the base
class of P2PTransportChannel to IceTransportInternal without breaking
Chromium. It is removed in this CL.
Reland this CL after Chromium doesn't depend on this.
BUG=webrtc:6951
Review-Url: https://codereview.webrtc.org/2632563002
Cr-Commit-Position: refs/heads/master@{#16131}
For low bitrates actual encoder output bitrate may differ from target
by 3-4 times.
Also adjust adaptation speed based on bitrate variation from the target.
BUG=b/34233384
R=wzh@webrtc.org
Review-Url: https://codereview.webrtc.org/2640543003 .
Cr-Commit-Position: refs/heads/master@{#16128}
Previously BirateProber was calculating delay between probes based on
the size of the previous probe. Because of that the actual sent bitrate
can deviate greatly from the target value. With this change it uses
total number of bytes in the cluster to estimate delay before each
probe.
BUG=webrtc:6952
Review-Url: https://codereview.webrtc.org/2613543003
Cr-Original-Commit-Position: refs/heads/master@{#15971}
Committed: 599c5011e7
Review-Url: https://codereview.webrtc.org/2613543003
Cr-Commit-Position: refs/heads/master@{#16127}
Previously CallActivity would be leaked on every call. This CL fixes the
problem.
BUG=webrtc:6684
Review-Url: https://codereview.webrtc.org/2640433003
Cr-Commit-Position: refs/heads/master@{#16123}
Reason for revert:
nisse landed a change that always disable adaptation in these tests.
Original issue's description:
> Fix flaky WebRtcVideoChannel2BaseTest.GetStats T
> This cl allows width and height of the produced encoded stream to be smaller than the configured camera resolution. This is since quality and cpu adapters may request a scaled input frame.
>
> BUG=webrtc:6990
>
> Review-Url: https://codereview.webrtc.org/2634273002
> Cr-Commit-Position: refs/heads/master@{#16118}
> Committed: 311a64ccf5TBR=sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6990
Review-Url: https://codereview.webrtc.org/2639573002
Cr-Commit-Position: refs/heads/master@{#16120}
This cl allows width and height of the produced encoded stream to be smaller than the configured camera resolution. This is since quality and cpu adapters may request a scaled input frame.
BUG=webrtc:6990
Review-Url: https://codereview.webrtc.org/2634273002
Cr-Commit-Position: refs/heads/master@{#16118}
Reason for revert:
Breaks android bots.
Original issue's description:
> Make the new jitter buffer the default jitter buffer.
>
> This CL contains only the changes necessary to make the switch to the new jitter
> buffer, clean up will be done in follow up CLs.
>
> In this CL:
> - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
> new video jitter buffer the default one.
> - Moved WebRTC.Video.KeyFramesReceivedInPermille and
> WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
>
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2627463004
> Cr-Commit-Position: refs/heads/master@{#16114}
> Committed: 0f0763d86dTBR=stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2632123005
Cr-Commit-Position: refs/heads/master@{#16117}
Move file capturer/renderer tests from the AppRTCMobile tests directory
to the WebRTC tests directory. These tests do not test AppRTCMobile but
rather WebRTC functionality. Therefore, they belong in WebRTC tests
directory.
BUG=webrtc:6545
Review-Url: https://codereview.webrtc.org/2632233002
Cr-Commit-Position: refs/heads/master@{#16115}
This CL contains only the changes necessary to make the switch to the new jitter
buffer, clean up will be done in follow up CLs.
In this CL:
- Removed the WebRTC-NewVideoJitterBuffer experiment and made the
new video jitter buffer the default one.
- Moved WebRTC.Video.KeyFramesReceivedInPermille and
WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2627463004
Cr-Commit-Position: refs/heads/master@{#16114}
This is a first step towards it. I plan to refactor modules_unittests before continuing with this.
BUG=webrtc:6626
NOTRY=True
Review-Url: https://codereview.webrtc.org/2626163004
Cr-Commit-Position: refs/heads/master@{#16109}
The code previously allowed ipv6 addresses with less than eight sections even without all-zero sections being compacted by a ::.
BUG=webrtc:1028
Review-Url: https://codereview.webrtc.org/2606383003
Cr-Commit-Position: refs/heads/master@{#16108}
This CL adds an RTP module to FlexfecReceiveStreamImpl, and wires it up
to send RTCP RRs. It further makes some methods take const refs instead
of values, to make it more clear where packet copies are made. This
change reduces the number of copies by one, for the case when media
packets are added to the FlexFEC receiver.
The end-to-end test is modified to check for RTCP RRs being sent.
Part of this modification involves some indentation changes, and the
diff thus looks bigger than it logically is.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2625633003
Cr-Commit-Position: refs/heads/master@{#16106}
Calling event handlers when the camera is closing is safe because
CameraCapturer checks if the errors are coming from the current session.
Calling onFailure after camera has already started might lead to strange
behavior.
BUG=b/34112992
Review-Url: https://codereview.webrtc.org/2634973002
Cr-Commit-Position: refs/heads/master@{#16101}
The existence of FlexfecConfig is due to a naive design. Now when it
is not used on the receiving side (see https://codereview.webrtc.org/2542413002),
it is time to remove it from the sending side as well.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2621573002
Cr-Commit-Position: refs/heads/master@{#16097}
Currently, parameters are periodically updated, but the TargetBitrate
message is only sent if a new bitrate is set. It should be sent
periodically to indicate the signaled bitrate is valid and to prevent
stale values due to loss of RTCP packets.
BUG=webrtc:6897
Review-Url: https://codereview.webrtc.org/2616393003
Cr-Commit-Position: refs/heads/master@{#16096}
Based on the mapping between [Audio/Video]TrackInterface and
[Voice/Video][Sender/Receiver]Info.
The IDs of RTCMediaStreamTrackStats are updated to distinguish between
local and remote cases. Previously, if local and remote cases had the
same label only one of them would be included in the report (bug).
BUG=webrtc:6758, chromium:657854, chromium:657855, chromium:657856, chromium:627816
Review-Url: https://codereview.webrtc.org/2610843003
Cr-Commit-Position: refs/heads/master@{#16095}
This information is already available in another member, and
storing it in more places only creates more opportunities
for bugs.
BUG=None
Review-Url: https://codereview.webrtc.org/2613713002
Cr-Commit-Position: refs/heads/master@{#16094}
The MultithreadedFakeH264Encoder is a derived class from FakeEncoder
and FakeH264Encoder, and these should thus also be thread safe.
TESTED=Ran "out/Tsan/video_engine_tests --gtest_filter="*Multithreaded*" --gtest_repeat=100" with is_debug=false, dcheck_always_on=true, is_tsan=true.
BUG=webrtc:6943
Review-Url: https://codereview.webrtc.org/2604403003
Cr-Commit-Position: refs/heads/master@{#16093}
Removed const_cast while creating rtcp packet.
This way manually created packet is as good as parsed packet and can be used in tests directly.
To archive this, changed the way class stores deltas and their sizes:
encoded chunks are stored directly for all but last chunk simplifying rtcp packet creation.
deltas stored together with sequence_number that would allow to simplify reading them from the parsed packet.
Fixed test for maximum received packets.
BUG=None
Review-Url: https://codereview.webrtc.org/2616343003
Cr-Commit-Position: refs/heads/master@{#16091}
This maps, in both directions, [Audio/Video]TrackInterface with
[Voice/Video][Sender/Receiver]Info.
This mapping is necessary for RTCStatsCollector to know the relationship
between RTCMediaStreamTrackStats and RTC[In/Out]boundRTPStreamStats, and
to be able to collect several RTCMediaStreamTrackStats stats.
BUG=webrtc:6757, chromium:659137, chromium:657854, chromium:627816
Review-Url: https://codereview.webrtc.org/2611983002
Cr-Commit-Position: refs/heads/master@{#16090}
The test case is put inside a new test target. That test target will be started from a test script to asses video quality.
BUG=webrtc:6545
Review-Url: https://codereview.webrtc.org/2585813002
Cr-Commit-Position: refs/heads/master@{#16088}
