Åsa Persson c3ed630560 Add stats googHasEnteredLowResolution. ...

Indicates if the forced sw fallback has had an effect (or would have had an effect if it had been
enabled).


Bug: webrtc:6634
Change-Id: I574b9001a2fae650fb894a1caa0d0f84257658e3
Reviewed-on: https://webrtc-review.googlesource.com/23300
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20729}

2017-11-17 13:02:07 +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

Media track ID visibility at BWE level

2017-10-13 13:47:07 +00:00

audio_state.h

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)

2017-11-01 11:04:26 +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

Fix deps of audio:audio_tests.

2017-11-14 08:20:47 +00:00

call_perf_tests.cc

Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)."

2017-09-29 13:48:29 +00:00

call_unittest.cc

Voice Engine: Require caller to supply an AudioDecoderFactory

2017-11-02 13:54:57 +00:00

call.cc

Stop using LOG macros in favor of RTC_ prefixed macros.

2017-11-09 11:56:32 +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

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +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