Alex Narest d0e196bd26 Adding two tests: ...

1. Bitrate allocation strategy unit test 
2. Perf test determining minimal supported bitrate with and without strategy

Bug: webrtc:8243
Change-Id: Idf675fbadddb66c77b2582052d6497971eb99ad6
Reviewed-on: https://webrtc-review.googlesource.com/4880
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20886}

2017-11-27 10:38:22 +00:00

..

test

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

audio_receive_stream.h

Added RTCMediaStreamTrackStats.jitterBufferDelay for audio

2017-10-02 10:47:00 +00:00

audio_send_stream.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

audio_send_stream.h

Use the new AudioProcessing statistics everywhere.

2017-11-24 18:17:39 +00:00

audio_state.h

Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine.

2017-11-21 10:51:02 +00:00

bitrate_allocator_unittest.cc

Extended the bitrate allocator to allow allocation to tracks based upon priorities which are planned to be defined as a relative bitrate in the RTCRtpEncodingParameters.

2017-11-14 22:20:09 +00:00

bitrate_allocator.cc

Add kTransmissionMaxBitrateMultiplier logic to audio priority bitrate allocation strategy similarly to default bitrate allocation logic.

2017-11-16 20:23:56 +00:00

bitrate_allocator.h

Extended the bitrate allocator to allow allocation to tracks based upon priorities which are planned to be defined as a relative bitrate in the RTCRtpEncodingParameters.

2017-11-14 22:20:09 +00:00

bitrate_estimator_tests.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

BUILD.gn

Use the new AudioProcessing statistics everywhere.

2017-11-24 18:17:39 +00:00

call_perf_tests.cc

Adding two tests:

2017-11-27 10:38:22 +00:00

call_unittest.cc

Move ADM initialization into WebRtcVoiceEngine

2017-11-21 20:48:07 +00:00

call.cc

Deprecated the Get BitrateController method

2017-11-24 15:07:44 +00:00

call.h

Reland of BWE allocation strategy

2017-10-20 10:16:15 +00:00

callfactory.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

callfactory.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

callfactoryinterface.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

DEPS

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

fake_rtp_transport_controller_send.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

flexfec_receive_stream_impl.cc

Stop using LOG macros in favor of RTC_ prefixed macros.

2017-11-09 11:56:32 +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

flexfec_receive_stream.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

OWNERS

Remove pbos@webrtc.org from all OWNERS.

2017-11-01 08:03:46 +00:00

rampup_tests.cc

Reland of BWE allocation strategy

2017-10-20 10:16:15 +00:00

rampup_tests.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtcp_demuxer_unittest.cc

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

rtcp_demuxer.cc

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

rtcp_demuxer.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +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_config.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +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

Move some numeric utility code from rtc_base/ to rtc_base/numerics/

2017-11-22 11:21:47 +00:00

rtp_demuxer.cc

Stop using LOG macros in favor of RTC_ prefixed macros.

2017-11-09 11:56:32 +00:00

rtp_demuxer.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +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

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_rtcp_demuxer_helper.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_rtcp_demuxer_helper.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_stream_receiver_controller.cc

Stop using LOG macros in favor of RTC_ prefixed macros.

2017-11-09 11:56:32 +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

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_transport_controller_send.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_transport_controller_send.h

Delete deprecated constructor of SendSideCongestionController.

2017-11-06 15:02:36 +00:00

rtx_receive_stream_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtx_receive_stream.cc

Stop using LOG macros in favor of RTC_ prefixed macros.

2017-11-09 11:56:32 +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

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

syncable.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

syncable.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

video_config.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

video_config.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

video_receive_stream.cc

Delete member VideoReceiveStream::Config::Rtp::ulpfec.

2017-09-26 09:49:21 +00:00

video_receive_stream.h

Delete member VideoReceiveStream::Config::Rtp::ulpfec.

2017-09-26 09:49:21 +00:00

video_send_stream.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

video_send_stream.h

Add stats googHasEnteredLowResolution.

2017-11-17 13:02:07 +00:00