21 Commits

Author SHA1 Message Date
deadbeef
1086ed6469 Disable SwitchesToASTThenBackToTOFForVideo test completely.
It's flaky not only for ASan/MSan, but regular Windows/Linux bots.

NOTRY=True
BUG=webrtc:5790

Review URL: https://codereview.webrtc.org/1908663002

Cr-Commit-Position: refs/heads/master@{#12475}
2016-04-22 17:52:44 +00:00
deadbeef
844f99351f Disabling SwitchesToASTThenBackToTOFForVideo test for MSan bot.
TBR=pbos@webrtc.org
NOTRY=True
BUG=webrtc:5790

Review URL: https://codereview.webrtc.org/1902603002

Cr-Commit-Position: refs/heads/master@{#12418}
2016-04-18 23:19:02 +00:00
minyuel
4aa438c915 Suppress a flaky test: SwitchesToASTThenBackToTOFForVideo.
BUG=webrtc:5790
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1883923005 .

Cr-Commit-Position: refs/heads/master@{#12378}
2016-04-15 13:34:54 +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
789ba92e14 Simplify CongestionController.
- Removes the dependency on CallStats.
- Implements Module interface so that we don't have to register
  each internal component to the process thread separately.

R=pbos@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1704983002 .

Cr-Commit-Position: refs/heads/master@{#11655}
2016-02-17 14:52:25 +00:00
Peter Boström
8c66a00a37 Initialize VideoSendStream members in constructor.
Removes scoped_ptrs and provides clearer lifetime between objects.

BUG=webrtc:5494
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1674663002 .

Cr-Commit-Position: refs/heads/master@{#11571}
2016-02-11 12:51:18 +00:00
stefan
ba4c0e45ff Add send-side BWE to WebRtcVoiceEngine under a finch experiment.
This adds negotiation of both transport sequence number and transport
feedback. Only offers transport seq num if the
WebRTC-Audio-SendSideBwe finch experiment is enabled.

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

Review URL: https://codereview.webrtc.org/1604563002

Cr-Commit-Position: refs/heads/master@{#11487}
2016-02-04 12:12:31 +00:00
Stefan Holmer
9fea80f50d Add audio streams to CallTest and a first A/V call test.
Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code.

R=pbos@webrtc.org
TBR=kjellander@webrtc.org

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1542653002 .

Cr-Commit-Position: refs/heads/master@{#11171}
2016-01-07 16:43:31 +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
Peter Boström
5811a39f14 Replace EventWrapper in video/, test/ and call/.
Makes use of rtc::Event which is simpler and can be used without
allocating additional objects on the heap.

Does not modify test/channel_transport/.

BUG=
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1487893004 .

Cr-Commit-Position: refs/heads/master@{#10968}
2015-12-10 12:03:00 +00:00
Peter Boström
7c704b8289 Use webrtc/base/logging.h in stefan@'s ownership.
Replaces system_wrappers' logging in call/, bitrate_controller/, pacing/
and remote_bitrate_estimator/.

BUG=webrtc:5118
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1484503002 .

Cr-Commit-Position: refs/heads/master@{#10896}
2015-12-04 15:13:12 +00:00
Peter Boström
521af4e344 Remove duplicate decoders in BitrateEstimatorTest.
Multiple decoders were used for the same payload type in this test case,
causing CHECK failures when configuring.

BUG=webrtc:5249
TBR=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1484443003 .

Cr-Commit-Position: refs/heads/master@{#10825}
2015-11-27 15:35:14 +00:00
solenberg
3a94154035 Move some send stream configuration into webrtc::AudioSendStream.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1418503010

Cr-Commit-Position: refs/heads/master@{#10652}
2015-11-16 15:34:59 +00:00
solenberg
566ef247b9 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1403363003

Cr-Commit-Position: refs/heads/master@{#10548}
2015-11-06 23:34:58 +00:00
Fredrik Solenberg
0ccae13556 Changed FakeVoiceEngine into a MockVoiceEngine.
BUG=webrtc:4690
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1402403008 .

Cr-Commit-Position: refs/heads/master@{#10491}
2015-11-03 09:15:59 +00:00
Henrik Kjellander
98f53510b2 system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
stefan
f116bd0d7a Call OnSentPacket for all packets sent in the test framework.
Required a bit of refactoring to make it possible to pass a Call to DirectTransport on construction. This also lead to me having to remove the shared lock between PacketTransport and RtpRtcpObserver. Now RtpRtcpObserver has a SetTransports method instead of a SetReceivers method.

BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1419193002

Cr-Commit-Position: refs/heads/master@{#10430}
2015-10-27 15:29:47 +00:00
Fredrik Solenberg
4f4ec0a927 Re-Land: Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
BUG=webrtc:4690

Committed: a457752f4a

Review URL: https://codereview.webrtc.org/1390753002 .

Cr-Commit-Position: refs/heads/master@{#10369}
2015-10-22 08:49:39 +00:00
solenberg
43e83d44f0 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
Reason for revert:
webrtc_perf_tests started failing on Win32 Release, Mac32 Release and Linux64 Release (all running large tests). These were not caught by try bots.

Original issue's description:
> Implement AudioReceiveStream::GetStats().
>
> R=tommi@webrtc.org
> TBR=hta@webrtc.org
> BUG=webrtc:4690
>
> Committed: a457752f4a

TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1411083006

Cr-Commit-Position: refs/heads/master@{#10340}
2015-10-20 13:41:06 +00:00
Fredrik Solenberg
a457752f4a Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
TBR=hta@webrtc.org
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1390753002 .

Cr-Commit-Position: refs/heads/master@{#10338}
2015-10-20 13:01:55 +00:00
Peter Boström
5c389d3e09 Split webrtc/video into webrtc/{audio,call,video}.
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.

BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1227923005 .

Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 11:58:39 +00:00