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Patrik Höglund b6b29e0718 Convert video quality test from a TEST_F to a TEST fixture. 
The purpose is to make the fixture reusable in downstream
projects. The CL adds the following things to API:

- api/test/video_quality_test_fixture.h
- api/test/create_video_quality_test_fixture.h

The following things are moved to API:

- call/bitrate_constraints.h (api/bitrate_constraints.h)
- call/simulated_network.h (api/test/simulated_network.h)
- call/media_type.h (api/mediatypes.h)

These are required by the params struct passed to the
fixture. I didn't attempt to split the params struct into
an internal-only and public version in this CL, and as
a result we need to pull in the above things. They are
quite harmless though, so I think it's worth it in order
to avoid splitting up the test config struct.

This CL doesn't solve all the problems we need to
implement downstream tests; we probably need to upstream
tracing variants of FakeNetworkPipe for instance, but
that will come later. This puts in place the basic
structure for now.

Bug: None
Change-Id: I35e26ed126fad27bc7b2a465400291084f6ac911
Reviewed-on: https://webrtc-review.googlesource.com/69601
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23714}
2018-06-21 15:49:43 +00:00
..
audio Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
audio_codecs Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
call Pass packet retransmission information in PacketOptions. 2018-05-29 10:12:04 +00:00
ortc Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
stats Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
test Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
transport Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
units Add conversions to and from double for units. 2018-05-30 14:34:02 +00:00
video Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
video_codecs Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8." 2018-06-21 13:41:14 +00:00
array_view.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
array_view_unittest.cc ArrayView, adding ctor for fixed-size views of const(expr) std::array. 2018-05-15 13:49:02 +00:00
audio_options.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_options.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
bitrate_constraints.h Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
BUILD.gn Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
candidate.cc This CL removes all usages of our custom ostream << overloads. 2018-04-03 12:51:00 +00:00
candidate.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
cryptoparams.h Fix ortc_api circular deps. 2017-11-15 13:31:51 +00:00
datachannelinterface.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
DEPS Make rtc_software_fallback_wrappers target visible. 2018-06-12 12:51:34 +00:00
dtmfsenderinterface.h Remove unused/deprecated DTMF methods 2018-06-20 21:00:10 +00:00
fakemetricsobserver.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fakemetricsobserver.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fec_controller.h Revert "Revert "Enables PeerConnectionFactory using external fec controller"" 2018-02-20 12:41:55 +00:00
jsep.cc Fix clang style errors in api/jsep.h 2017-12-06 18:12:06 +00:00
jsep.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
jsepicecandidate.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
jsepsessiondescription.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mediaconstraintsinterface.cc Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
mediaconstraintsinterface.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
mediastreaminterface.cc Final name changing of MediaStreamInterface.label() to id(). 2018-03-14 20:30:52 +00:00
mediastreaminterface.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
mediastreamproxy.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mediastreamtrackproxy.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mediatypes.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mediatypes.h Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
notifier.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
optional.h Reland "Use absl::optional instead or rtc::Optional" 2018-06-12 19:13:21 +00:00
optional_unittest.cc Reland "Use absl::optional instead or rtc::Optional" 2018-06-12 19:13:21 +00:00
OWNERS Make hbos@webrtc.org OWNER of peerconnection*. 2017-11-13 12:27:29 +00:00
peerconnectionfactoryproxy.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
peerconnectioninterface.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
peerconnectionproxy.h Remove unused/deprecated DTMF methods 2018-06-20 21:00:10 +00:00
proxy.cc Enable the clang style plugin in primary p2p/ target 2017-11-01 00:19:05 +00:00
proxy.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
refcountedbase.h New classes RefCounter and RefCountedBase. 2017-10-23 11:46:47 +00:00
rtcerror.cc Reland "Reland "Remove our stream << overloads from non-test build targets."" 2018-05-03 10:41:41 +00:00
rtcerror.h Reland "Reland "Remove our stream << overloads from non-test build targets."" 2018-05-03 10:41:41 +00:00
rtcerror_unittest.cc This CL removes all usages of our custom ostream << overloads. 2018-04-03 12:51:00 +00:00
rtceventlogoutput.h Move RtcEventLogOutput to api/ 2017-10-06 13:58:14 +00:00
rtp_headers.cc Fix circular dependencies in webrtc_common. 2017-12-15 14:33:26 +00:00
rtp_headers.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
rtpparameters.cc Add Rtcp parameters for PeerConnection senders 2018-05-28 09:28:59 +00:00
rtpparameters.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
rtpparameters_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtpreceiverinterface.cc Reland "Update RTCStatsCollector to work with RtpTransceivers" 2018-02-17 00:01:39 +00:00
rtpreceiverinterface.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
rtpsenderinterface.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
rtptransceiverinterface.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
setremotedescriptionobserverinterface.h Reland "SetRemoteDescriptionObserverInterface added." 2017-11-23 19:59:48 +00:00
statstypes.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
statstypes.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
turncustomizer.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
umametrics.cc Enable the clang style plugin in primary p2p/ target 2017-11-01 00:19:05 +00:00
umametrics.h Add collection of usage signatures on PeerConnections 2018-05-31 13:07:09 +00:00
videosourceinterface.h Move VideoStreamEncoderInterface to api/. 2018-05-21 19:50:37 +00:00
videosourceproxy.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00