6 Commits

Author SHA1 Message Date
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