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webrtc_m130/call
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Erik Språng c12d41b747 Add field trial kill switch for packetization overhead subtraction. 
Just in case.
Also slightly update picture id test to make it more clear.
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/115410

Bug: webrtc:10155
Change-Id: I9a0239e474b79fe545738860983e1931e8b82eff
Reviewed-on: https://webrtc-review.googlesource.com/c/116661
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26173}
2019-01-09 10:19:22 +00:00
..
test
Include absl/memory/memory.h if absl::make_unique is used
2019-01-08 20:08:32 +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
Add PeerConnection option to configure minimum audio jitter buffer delay.
2018-11-27 19:49:48 +00:00
audio_send_stream.cc
Remove more unneeded things from ChannelSend
2018-11-21 09:04:07 +00:00
audio_send_stream.h
Simulcast work to enable RID mux.
2018-12-21 20:59:23 +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
Don't trigger too many probes when max allocated bitrate changes.
2018-12-11 16:19:53 +00:00
bitrate_allocator.cc
Signals start rate when registering to TargetTransferRateObserver.
2018-12-12 09:22:32 +00:00
bitrate_allocator.h
Signals start rate when registering to TargetTransferRateObserver.
2018-12-12 09:22:32 +00:00
bitrate_estimator_tests.cc
Include absl/memory/memory.h if absl::make_unique is used
2019-01-08 20:08:32 +00:00
BUILD.gn
Include absl/memory/memory.h if absl::make_unique is used
2019-01-08 20:08:32 +00:00
call_config.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
call_config.h
Delete VideoReceiveStream::EnableEncodedFrameRecording
2018-10-02 10:31:46 +00:00
call_perf_tests.cc
Disable flaky tests CallPerfTest.CaptureNtpTimeWithNetworkDelay on WIN.
2018-11-22 11:23:37 +00:00
call_unittest.cc
Remove Transport implementation from ChannelSend
2018-12-18 09:34:52 +00:00
call.cc
Register stat callbacks after rate observer is registered.
2018-12-19 21:19:01 +00:00
call.h
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
callfactory.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
callfactory.h
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
degraded_call.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
degraded_call.h
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
DEPS
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
fake_network_pipe.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
fake_network_pipe.h
Delete rtc::PacketTime (was an alias for int64_t)
2019-01-02 09:07:51 +00:00
flexfec_receive_stream_impl.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +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
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
flexfec_receive_stream.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
flexfec_receive_stream.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
OWNERS
Makes srte owner in call/ and test/
2018-07-13 08:39:41 +00:00
packet_receiver.h
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
rampup_tests.cc
Include absl/memory/memory.h if absl::make_unique is used
2019-01-08 20:08:32 +00:00
rampup_tests.h
Refactor bitrate configuration in CallTest
2018-11-13 16:03:00 +00:00
receive_time_calculator_unittest.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
receive_time_calculator.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
receive_time_calculator.h
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
rtcp_demuxer_unittest.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
rtcp_demuxer.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +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
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
rtp_config.cc
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
rtp_config.h
Simulcast work to enable RID mux.
2018-12-21 20:59:23 +00:00
rtp_demuxer_unittest.cc
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
rtp_demuxer.cc
Allow all "token" chars from RFC 4566 when checking for legal mid names.
2018-08-01 18:20:42 +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_payload_params_unittest.cc
Send and receive color space information if available
2018-12-03 21:07:45 +00:00
rtp_payload_params.cc
Send and receive color space information if available
2018-12-03 21:07:45 +00:00
rtp_payload_params.h
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
rtp_rtcp_demuxer_helper_unittest.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +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
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +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
Removing redundant argument for SSRCs from ctor of RtpVideoSender.
2018-12-21 23:33:39 +00:00
rtp_transport_controller_send.cc
Removing redundant argument for SSRCs from ctor of RtpVideoSender.
2018-12-21 23:33:39 +00:00
rtp_transport_controller_send.h
Removing redundant argument for SSRCs from ctor of RtpVideoSender.
2018-12-21 23:33:39 +00:00
rtp_video_sender_interface.h
Move FecController to RtpVideoSender.
2018-10-05 14:39:01 +00:00
rtp_video_sender_unittest.cc
Include absl/memory/memory.h if absl::make_unique is used
2019-01-08 20:08:32 +00:00
rtp_video_sender.cc
Add field trial kill switch for packetization overhead subtraction.
2019-01-09 10:19:22 +00:00
rtp_video_sender.h
Add field trial kill switch for packetization overhead subtraction.
2019-01-09 10:19:22 +00:00
rtx_receive_stream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtx_receive_stream.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
rtx_receive_stream.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
simulated_network.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
simulated_network.h
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
simulated_packet_receiver.h
Eliminate methods SetConfig() from DirectTransport and FakeNetworkPipe
2018-08-22 11:12:40 +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_receive_stream.cc
New statistic added to VideoReceiveStream to determine latency to first decode.
2018-12-10 18:49:34 +00:00
video_receive_stream.h
Remove unused VideoReceiveStream::Config::AddRtxBinding.
2019-01-09 10:16:12 +00:00
video_send_stream.cc
Delete pre_encode_callback from VideoSendStream::Config
2019-01-02 09:14:32 +00:00
video_send_stream.h
Delete pre_encode_callback from VideoSendStream::Config
2019-01-02 09:14:32 +00:00