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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
..
test Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
audio_receive_stream.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_receive_stream.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
audio_send_stream.cc replace stringstream in call/ 2018-04-04 16:09:15 +00:00
audio_send_stream.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_state.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_state.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
bitrate_allocator.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
bitrate_allocator.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
bitrate_allocator_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
bitrate_estimator_tests.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
BUILD.gn Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
call.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
call.h Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
call_config.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
call_config.h Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
call_perf_tests.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
call_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
callfactory.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
callfactory.h Rename Call::Config to CallConfig, keep old name as alias. 2018-02-14 15:14:39 +00:00
degraded_call.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
degraded_call.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
DEPS Make fec controller plug-able. 2018-01-22 11:48:16 +00:00
fake_network_pipe.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
fake_network_pipe.h Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
flexfec_receive_stream.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
flexfec_receive_stream.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
flexfec_receive_stream_impl.cc replace stringstream in call/ 2018-04-04 16:09:15 +00:00
flexfec_receive_stream_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
flexfec_receive_stream_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Added nisse@webrtc.org as owner in call. 2018-02-20 09:39:51 +00:00
rampup_tests.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rampup_tests.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
receive_time_calculator.cc Fix of ReceiveTimeCalculator field trial parsing. 2018-03-24 12:36:17 +00:00
receive_time_calculator.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
receive_time_calculator_unittest.cc Added receive time calculator under field trial. 2018-03-21 15:40:39 +00:00
rtcp_demuxer.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtcp_demuxer.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
rtcp_demuxer_unittest.cc Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
rtcp_packet_sink_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_bitrate_configurator.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_bitrate_configurator.h Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
rtp_bitrate_configurator_unittest.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_config.cc replace stringstream in call/ 2018-04-04 16:09:15 +00:00
rtp_config.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_demuxer.cc Don't check MIDs when demuxing RTP packets in Call 2018-03-29 20:36:08 +00:00
rtp_demuxer.h Don't check MIDs when demuxing RTP packets in Call 2018-03-29 20:36:08 +00:00
rtp_demuxer_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_packet_sink_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_rtcp_demuxer_helper.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_rtcp_demuxer_helper.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_rtcp_demuxer_helper_unittest.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_stream_receiver_controller.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_stream_receiver_controller.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_stream_receiver_controller_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_transport_controller_send.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_transport_controller_send.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_transport_controller_send_interface.h Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
rtx_receive_stream.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtx_receive_stream.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtx_receive_stream_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
ssrc_binding_observer.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
syncable.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
syncable.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
video_config.h Move VideoEncoderConfig from call/ to api/. 2018-05-18 12:58:16 +00:00
video_receive_stream.cc Delete pre_decode_callback. 2018-06-20 07:04:09 +00:00
video_receive_stream.h Delete pre_decode_callback. 2018-06-20 07:04:09 +00:00
video_send_stream.cc Delete unused stats for preferred_bitrate. 2018-06-07 08:11:07 +00:00
video_send_stream.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00