| 9fea80f50daab46f20d4a6fc67b0144fbbbf56cd |
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07-Jan-2016 |
Stefan Holmer <stefan@webrtc.org> |
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}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
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| ff483617a4fdf282bb82d7f4ce15af3dbe305a4a |
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21-Dec-2015 |
stefan <stefan@webrtc.org> |
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}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
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| 5811a39f14fd77ebc0793ee93d03ee15a669bd8f |
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10-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
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| 7c704b82893bbe7fc206b004fb9dfe6e69a986ef |
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04-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
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| 521af4e344678ce9dcf996341af6ba8056e1e147 |
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27-Nov-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
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| 3a94154035fa16e4efd91125311f076b547c38b9 |
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16-Nov-2015 |
solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
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| 566ef247b9779f6c9d0e7ec9eea6b037f4682c53 |
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07-Nov-2015 |
solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
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| 0ccae135562ac180da053fcecda91a0365621f14 |
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03-Nov-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
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| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
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28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
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}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
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| f116bd0d7a3cdad20bb638d5a87427bd920c8904 |
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27-Oct-2015 |
stefan <stefan@webrtc.org> |
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}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
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| 4f4ec0a9270a8cefadfa12e9fa3b979b58b15392 |
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22-Oct-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Re-Land: Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
BUG=webrtc:4690
Committed: https://chromium.googlesource.com/external/webrtc/+/a457752f4afc496ed7f4d6b584b08d8635f18cc0
Review URL: https://codereview.webrtc.org/1390753002 .
Cr-Commit-Position: refs/heads/master@{#10369}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
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| 43e83d44f01683fbd304e37d47d2f6db0d52660d |
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20-Oct-2015 |
solenberg <solenberg@webrtc.org> |
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: https://chromium.googlesource.com/external/webrtc/+/a457752f4afc496ed7f4d6b584b08d8635f18cc0
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}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
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| a457752f4afc496ed7f4d6b584b08d8635f18cc0 |
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20-Oct-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
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| 5c389d3e09646c0e2ed76d5ccb37a3419a09eb6a |
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25-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/call/bitrate_estimator_tests.cc
|