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webrtc_m130/webrtc/call
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Steve Anton b3329179a7 Rename RsidResolutionObserver to SsrcBindingObserver. 
This rename prepares SsrcBindingObserver to be used for observing all
the ways a sink can be bound to SSRCs (e.g., MID, payload types).

Bug: webrtc:4050
Change-Id: I16b68481d01f921c326a33f5f2baf79d8b3f12e2
Reviewed-on: https://chromium-review.googlesource.com/590762
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19396}
2017-08-17 23:01:59 +00:00
..
test
Only one implementation of MockRtpPacketSink once
2017-07-28 17:05:45 +00:00
audio_receive_stream.h
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
audio_send_stream.cc
Injectable audio encoders: WebRtcVoiceEngine and company
2017-04-27 09:08:52 +00:00
audio_send_stream.h
Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
2017-07-26 09:09:44 +00:00
audio_state.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
bitrate_allocator_unittest.cc
Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ )
2017-02-21 15:28:31 +00:00
bitrate_allocator.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
bitrate_allocator.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
bitrate_estimator_tests.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
BUILD.gn
Rename RsidResolutionObserver to SsrcBindingObserver.
2017-08-17 23:01:59 +00:00
call_perf_tests.cc
Removed an unused variable from CallPerfTest::TestAudioVideoSync()
2017-08-11 09:08:33 +00:00
call_unittest.cc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
call.cc
Reland of Add functionality which limits the number of bytes on the network. (patchset #1 id:1 of https://codereview.webrtc.org/3001653002/ )
2017-08-16 15:16:25 +00:00
call.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
callfactory.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
callfactory.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
callfactoryinterface.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
DEPS
Passed AudioMixer to AudioState::Config.
2016-11-17 14:48:56 +00:00
fake_rtp_transport_controller_send.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
flexfec_receive_stream_impl.cc
Delete unneeded Start and Stop methods on FlexfecReceiveStream.
2017-08-16 09:41:27 +00:00
flexfec_receive_stream_impl.h
Delete unneeded Start and Stop methods on FlexfecReceiveStream.
2017-08-16 09:41:27 +00:00
flexfec_receive_stream_unittest.cc
Delete unneeded Start and Stop methods on FlexfecReceiveStream.
2017-08-16 09:41:27 +00:00
flexfec_receive_stream.h
Delete unneeded Start and Stop methods on FlexfecReceiveStream.
2017-08-16 09:41:27 +00:00
OWNERS
Remove all references to GYP
2016-11-16 19:11:38 +00:00
rampup_tests.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rampup_tests.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcp_demuxer_unittest.cc
Rename RsidResolutionObserver to SsrcBindingObserver.
2017-08-17 23:01:59 +00:00
rtcp_demuxer.cc
Rename RsidResolutionObserver to SsrcBindingObserver.
2017-08-17 23:01:59 +00:00
rtcp_demuxer.h
Rename RsidResolutionObserver to SsrcBindingObserver.
2017-08-17 23:01:59 +00:00
rtcp_packet_sink_interface.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_demuxer_unittest.cc
Rename RsidResolutionObserver to SsrcBindingObserver.
2017-08-17 23:01:59 +00:00
rtp_demuxer.cc
Rename RsidResolutionObserver to SsrcBindingObserver.
2017-08-17 23:01:59 +00:00
rtp_demuxer.h
Rename RsidResolutionObserver to SsrcBindingObserver.
2017-08-17 23:01:59 +00:00
rtp_packet_sink_interface.h
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
2017-06-26 18:23:54 +00:00
rtp_rtcp_demuxer_helper_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_rtcp_demuxer_helper.cc
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
2017-06-26 18:23:54 +00:00
rtp_rtcp_demuxer_helper.h
Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )
2017-08-04 13:34:54 +00:00
rtp_stream_receiver_controller_interface.h
Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )
2017-08-04 13:34:54 +00:00
rtp_stream_receiver_controller.cc
Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )
2017-08-04 13:34:54 +00:00
rtp_stream_receiver_controller.h
Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )
2017-08-04 13:34:54 +00:00
rtp_transport_controller_send_interface.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
rtp_transport_controller_send.cc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
rtp_transport_controller_send.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
rtx_receive_stream_unittest.cc
Only one implementation of MockRtpPacketSink once
2017-07-28 17:05:45 +00:00
rtx_receive_stream.cc
Address some violations of chromium-style.
2017-05-31 09:24:52 +00:00
rtx_receive_stream.h
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
2017-06-01 11:02:35 +00:00
ssrc_binding_observer.h
Rename RsidResolutionObserver to SsrcBindingObserver.
2017-08-17 23:01:59 +00:00
syncable.cc
Stop using VoEVideoSync in Call/VideoReceiveStream.
2017-01-31 11:58:40 +00:00
syncable.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00