Instead of using the time on the first callback to Call::OnSentPacket, use the time when the first packet is sent from the pacer (to make sure this packet corresponds to an audio/video RTP packet).
BUG=webrtc:6244
Review-Url: https://codereview.webrtc.org/2825333002
Cr-Commit-Position: refs/heads/master@{#17777}
AudioDecoder and AudioDecoderFactory are in webrtc/api/ now, so move
their mocks to someplace central where tests from all over WebRTC are
allowed to #include them.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2798063004
Cr-Commit-Position: refs/heads/master@{#17619}
These are invalid since:
- An allocated bitrate of 0 means that the stream should be disabled. Changing the behavior to send audio at min bitrate even though the allocator asks for the stream to be disabled.
- It should be OK to set a min bitrate equal to the start bitrate.
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2806163003
Cr-Commit-Position: refs/heads/master@{#17613}
This provides a better default for audio-only tests.
BUG=None
Review-Url: https://codereview.webrtc.org/2794193003
Cr-Commit-Position: refs/heads/master@{#17536}
Reason for revert:
Looks like this has caused multiple Android webrtc perf build bot failures in RampUpTest.UpDownUpTransportSequenceNumberRtx
Original issue's description:
> Enable the bayesian bitrate estimator by default.
>
> BUG=webrtc:6566, webrtc:7415
>
> Review-Url: https://codereview.webrtc.org/2749803002
> Cr-Commit-Position: refs/heads/master@{#17475}
> Committed: c53a17f28eTBR=terelius@webrtc.org,magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6566, webrtc:7415
Review-Url: https://codereview.webrtc.org/2786913003
Cr-Commit-Position: refs/heads/master@{#17476}
Deletes left-over includes of trace.h and critical_section_wrapper.h.
BUG=webrtc:7035
Review-Url: https://codereview.webrtc.org/2784873002
Cr-Commit-Position: refs/heads/master@{#17460}
Reason for revert:
Reverting since this seems to break multiple WebRTC Perf buildbots
Original issue's description:
> Don't hardcode MediaType::ANY in FakeNetworkPipe.
>
> Instead let each test set the appropriate media type. This simplifies
> demuxing in Call and later in RtpTransportController.
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2774463003
> Cr-Commit-Position: refs/heads/master@{#17418}
> Committed: 9c47b00e24TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2784543002
Cr-Commit-Position: refs/heads/master@{#17427}
Instead let each test set the appropriate media type. This simplifies
demuxing in Call and later in RtpTransportController.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2774463003
Cr-Commit-Position: refs/heads/master@{#17418}
Implementation owned by call, and passed to VideoSendStream and
AudioSendStream.
BUG=webrtc:6847, webrtc:7135
Review-Url: https://codereview.webrtc.org/2685673003
Cr-Commit-Position: refs/heads/master@{#17389}
Reason for revert:
Makes perf and Chromium FYI bots unhappy.
Original issue's description:
> WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
>
> This removes one more place where we were unable to handle codecs not
> in the built-in set.
>
> BUG=webrtc:5805
>
> Review-Url: https://codereview.webrtc.org/2686043006
> Cr-Commit-Position: refs/heads/master@{#17370}
> Committed: 1724cfbdbaTBR=ossu@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2772043002
Cr-Commit-Position: refs/heads/master@{#17374}
This removes one more place where we were unable to handle codecs not
in the built-in set.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2686043006
Cr-Commit-Position: refs/heads/master@{#17370}
In ViEEncoder, try to reduce framerate instead of resolution if the
current degradation preference is maintain-resolution rather than
balanced.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2716643002
Cr-Commit-Position: refs/heads/master@{#17327}
New class ReceiveSideCongestionController, extracted from CongestionController, and responsible for the
OnReceivedPacket processing.
Rest of the CongestionController moved to a new class
SendSideCongestionController.
To avoid breaking applications, CongestionController is redefined
as a union of these two classes, with no intended change in behavior.
With one exception: CongestionController::SetBweBitrates used to call
remote_bitrate_estimator_.SetMinBitrate, but after remote_bitrate_estimator_ was moved to ReceiveSideCongestionController,
it no longer does this.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2752233002
Cr-Commit-Position: refs/heads/master@{#17321}
This is one step towards separation of send-side and receive-side
processing.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2740163002
Cr-Commit-Position: refs/heads/master@{#17306}
Fix reduced frame-rate on Mac and Android.
Also enable huge full-stack test Largeroom_50thumbs on Windows.
BUG=webrtc:7301
Review-Url: https://codereview.webrtc.org/2760583003
Cr-Commit-Position: refs/heads/master@{#17288}
Reason for revert:
Changes to frame-generator resulted in reduced fps on android and Mac on all tests.
Original issue's description:
> Reland of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2748643002/ )
>
> Reason for revert:
> Reland with fixes to the failing perf tests.
>
> Original issue's description:
> > Revert of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #2 id:90001 of https://codereview.webrtc.org/2744003002/ )
> >
> > Reason for revert:
> > CallPerfTest.ReceivesCpuOveruseAndUnderuse perf test fails due to this CL. It requires very accurate frame rate, which may not be so accurate now.
> >
> > Original issue's description:
> > > Reland of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2743993002/ )
> > >
> > > And enable large full-stack test depending on that change (Reland of https://codereview.webrtc.org/2741823003/)
> > > TBR=stefan@webrtc.org,tommi@webrtc.org
> > > BUG=webrtc:7301,webrtc:7325
> > >
> > > Review-Url: https://codereview.webrtc.org/2744003002
> > > Cr-Commit-Position: refs/heads/master@{#17196}
> > > Committed: 8c0a5896d1
> >
> > TBR=stefan@webrtc.org,tommi@webrtc.org,sprang@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7301,webrtc:7325
> >
> > Review-Url: https://codereview.webrtc.org/2748643002
> > Cr-Commit-Position: refs/heads/master@{#17198}
> > Committed: 382a72a0d3
>
> BUG=webrtc:7301,webrtc:7325
>
> Review-Url: https://codereview.webrtc.org/2750473002
> Cr-Commit-Position: refs/heads/master@{#17253}
> Committed: 2549ad4fefTBR=sprang@webrtc.org,tommi@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7301,webrtc:7325
Review-Url: https://codereview.webrtc.org/2751063005
Cr-Commit-Position: refs/heads/master@{#17276}
Reason for revert:
Causes problems with TSAN: https://bugs.chromium.org/p/webrtc/issues/detail?id=7325
Original issue's description:
> Reland of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper.
>
> Fix CallPerfTest.ReceivesCpuOveruseAndUnderuse to not fail on Android with new FrameGeneratorCapturer.
>
> BUG=webrtc:7301
>
> Review-Url: https://codereview.webrtc.org/2745583006
> Cr-Commit-Position: refs/heads/master@{#17168}
> Committed: b00742508aTBR=stefan@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7301
Review-Url: https://codereview.webrtc.org/2743993002
Cr-Commit-Position: refs/heads/master@{#17173}
Fix CallPerfTest.ReceivesCpuOveruseAndUnderuse to not fail on Android with new FrameGeneratorCapturer.
BUG=webrtc:7301
Review-Url: https://codereview.webrtc.org/2745583006
Cr-Commit-Position: refs/heads/master@{#17168}
Use of FlexFEC is known when streams are created in
WebRtcVideoChannel2, so this replaces the code in Call to infer
FlexFEC config of video streams from the configuration of the FlexFEC
stream(s). This also allows us to switch to a more logical creation
order, where media streams are created before the FlexFEC stream.
This is done in preparation for a larger refactoring of the RTP
demuxing done in Call.
BUG=None
Review-Url: https://codereview.webrtc.org/2712683002
Cr-Commit-Position: refs/heads/master@{#17143}
This makes a few things a lot clearer when looking at perf trace data:
* What module instances (where they were created) are called
* On what thread
* How frequently
* For how long
ProcessThread will be replaced by TaskQueue moving forward and this is a step towards understanding the behavior of the affected code.
BUG=webrtc:7219
Review-Url: https://codereview.webrtc.org/2729053002
Cr-Commit-Position: refs/heads/master@{#16998}
Patchset 1 is patchset #5 id:80001 of https://codereview.webrtc.org/2717983003/
Patchset 2 fix call_perf_test dep on fake_audio_device.
This reverts commit 985371bda999c6db51286586c5850d2ff58f3511.
Original cl description:
Move fake_audio_device to its own target.
The purpose is to make it usefull for test targets that does not need or can use test_common.
For some reason this also triggered override issues in rtp module tests that are fixed in the same cl.
BUG=none
TBR=kjellander@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2718363002
Cr-Commit-Position: refs/heads/master@{#16922}
