10 Commits

Author SHA1 Message Date
skvlad
11a9cbfa50 Refactoring: move ownership of RtcEventLog from Call to PeerConnection
This CL is a pure refactoring which should not result in any functinal
changes. It moves ownership of the RtcEventLog from webrtc::Call to the
webrtc::PeerConnection object.

This is done so that we can add RtcEventLog support for ICE events -
which will require the TransportController to have a pointer to the
RtcEventLog. PeerConnection is the closest common owner of both Call and
TransportController (through WebRtcSession).

BUG=webrtc:6393

Review-Url: https://codereview.webrtc.org/2353033005
Cr-Commit-Position: refs/heads/master@{#14578}
2016-10-07 18:53:15 +00:00
perkj
fa10b557d9 Releand of Let ViEEncoder handle resolution changes.
The original landed cl is in patchset 1.
The following patchset fix VideoQualityTest as well as fix the case where max_bitrate is set in the SendParams. A unit test is added for that as well.

Original cl description:
Let ViEEncoder handle resolution changes.

This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.

With this change, many variables in WebRtcVideoSendStream no longer need to be locked.

BUG=webrtc:5687, webrtc:6371, webrtc:5332

Review-Url: https://codereview.webrtc.org/2386573002
Cr-Commit-Position: refs/heads/master@{#14467}
2016-10-03 06:45:33 +00:00
perkj
3b703ede8b Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ )
Reason for revert:
Fails on a content_browsertest (and also webrtc_perf?)

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Tester/builds/34336

https://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/9091/steps/webrtc_perf_tests/logs/stdio
[  FAILED  ] FullStackTest.ParisQcifWithoutPacketLoss (59436 ms)

Original issue's description:
> Let ViEEncoder handle resolution changes.
>
> This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.
>
> With this change, many variables in WebRtcVideoSendStream no longer need to be locked.
>
> BUG=webrtc:5687, webrtc:6371, webrtc:5332
>
> Committed: https://crrev.com/26105b41b4f97642ee30cb067dc786c2737709ad
> Cr-Commit-Position: refs/heads/master@{#14445}

TBR=sprang@webrtc.org,mflodman@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:5687, webrtc:6371, webrtc:5332

Review-Url: https://codereview.webrtc.org/2383493005
Cr-Commit-Position: refs/heads/master@{#14447}
2016-09-30 06:25:46 +00:00
perkj
26105b41b4 Let ViEEncoder handle resolution changes.
This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.

With this change, many variables in WebRtcVideoSendStream no longer need to be locked.

BUG=webrtc:5687, webrtc:6371, webrtc:5332

Review-Url: https://codereview.webrtc.org/2351633002
Cr-Commit-Position: refs/heads/master@{#14445}
2016-09-30 05:39:15 +00:00
mflodman
86cc6ffc7c Variable audio bitrate.
This is a first CL wiring up AudioSendStream to BitrateAllocator. This
is still experimental and there is a test added for the audio only case,
combined audio video variable bitrate test cases will be added as a
follow up.

BUG=5079

Review-Url: https://codereview.webrtc.org/2165743003
Cr-Commit-Position: refs/heads/master@{#13527}
2016-07-26 11:44:12 +00:00
kwiberg
b25345ee3f Replace scoped_ptr with unique_ptr in webrtc/call/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1789903003

Cr-Commit-Position: refs/heads/master@{#11970}
2016-03-12 14:10:53 +00:00
Stefan Holmer
ff2a6351e0 Add ramp-up tests for transport sequence number with and w/o audio.
Also add a perf metric tracking the average network latency.

The audio stream test is disabled for now since audio isn't included in bitrate allocation.

BUG=webrtc:5263
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1582833002 .

Cr-Commit-Position: refs/heads/master@{#11244}
2016-01-14 09:00:34 +00:00
Stefan Holmer
d20e651327 Fix test bug introduced in r11101.
BUG=chromium:572995
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1578223002 .

Cr-Commit-Position: refs/heads/master@{#11224}
2016-01-12 14:51:28 +00:00
stefan
e74eef19bd Add CreateSend/ReceiveTransport() methods to CallTest.
This allows the test to create its own transports if it, for instance, needs to do demuxing.

BUG=webrtc:5416

Review URL: https://codereview.webrtc.org/1573453002

Cr-Commit-Position: refs/heads/master@{#11187}
2016-01-08 14:47:21 +00:00
stefan
ff483617a4 Step 1 to prepare call_test.* for combined audio/video tests.
Also move (and clean up includes) rampup_tests.* to webrtc/call in preparation for combined audio/video ramp-up tests.

No functional changes.

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1537273003

Cr-Commit-Position: refs/heads/master@{#11101}
2015-12-21 11:14:05 +00:00