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2234121cfb1f96e6067873688c0b0d06cbfb4260
platform-external-webrtc/call
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Sebastian Jansson cd8f382557 Removes unnecessary destructor checks in tests. 
Removes checks that are not relevant to the particular tests. The checks
create dependencies on the CallTest base class. This prepares for
further refactoring in CallTest.

Bug: webrtc:9510
Change-Id: Ie6b0093a8fcb8a152ca58f421727c4b085b60a87
Reviewed-on: https://webrtc-review.googlesource.com/87845
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23930}
2018-07-11 11:03:46 +00:00
..
test
Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
2018-06-27 10:33:40 +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_unittest.cc
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_estimator_tests.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
BUILD.gn
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
call_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
call.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
call.h
Convert video quality test from a TEST_F to a TEST fixture.
2018-06-21 15:49:43 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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_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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
OWNERS
Added nisse@webrtc.org as owner in call.
2018-02-20 09:39:51 +00:00
rampup_tests.cc
Removes unnecessary destructor checks in tests.
2018-07-11 11:03:46 +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_unittest.cc
Added receive time calculator under field trial.
2018-03-21 15:40:39 +00:00
receive_time_calculator.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
rtcp_demuxer_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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_packet_sink_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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_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_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_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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_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_unittest.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +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_stream_receiver_controller_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_stream_receiver_controller.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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_transport_controller_send_interface.h
Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
2018-06-27 10:33:40 +00:00
rtp_transport_controller_send.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
rtp_transport_controller_send.h
Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
2018-06-27 10:33:40 +00:00
rtx_receive_stream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +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
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