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platform-external-webrtc/api
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Åsa Persson bdee46d275 Add functionality to set min bitrate for single stream through RtpEncodingParameters. 
Bug: webrtc:9341
Change-Id: Ia1e819bd61dbef9c93d0ba84907faeeebf925db2
Reviewed-on: https://webrtc-review.googlesource.com/80261
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23722}
2018-06-25 10:10:48 +00:00
..
audio
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
audio_codecs
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
call
Pass packet retransmission information in PacketOptions.
2018-05-29 10:12:04 +00:00
ortc
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
stats
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
test
Generalize SimulcastEncoderAdapter, use for H264 & VP8.
2018-06-21 15:57:43 +00:00
transport
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
units
Add conversions to and from double for units.
2018-05-30 14:34:02 +00:00
video
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
video_codecs
Add functionality to set min bitrate for single stream through RtpEncodingParameters.
2018-06-25 10:10:48 +00:00
array_view_unittest.cc
ArrayView, adding ctor for fixed-size views of const(expr) std::array.
2018-05-15 13:49:02 +00:00
array_view.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_options.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
audio_options.h
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
bitrate_constraints.h
Convert video quality test from a TEST_F to a TEST fixture.
2018-06-21 15:49:43 +00:00
BUILD.gn
Convert video quality test from a TEST_F to a TEST fixture.
2018-06-21 15:49:43 +00:00
candidate.cc
This CL removes all usages of our custom ostream << overloads.
2018-04-03 12:51:00 +00:00
candidate.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
cryptoparams.h
Fix ortc_api circular deps.
2017-11-15 13:31:51 +00:00
datachannelinterface.h
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
DEPS
Make rtc_software_fallback_wrappers target visible.
2018-06-12 12:51:34 +00:00
dtmfsenderinterface.h
Remove unused/deprecated DTMF methods
2018-06-20 21:00:10 +00:00
fakemetricsobserver.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakemetricsobserver.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fec_controller.h
Revert "Revert "Enables PeerConnectionFactory using external fec controller""
2018-02-20 12:41:55 +00:00
jsep.cc
Fix clang style errors in api/jsep.h
2017-12-06 18:12:06 +00:00
jsep.h
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
jsepicecandidate.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
jsepsessiondescription.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediaconstraintsinterface.cc
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
mediaconstraintsinterface.h
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
mediastreaminterface.cc
Final name changing of MediaStreamInterface.label() to id().
2018-03-14 20:30:52 +00:00
mediastreaminterface.h
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
mediastreamproxy.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastreamtrackproxy.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediatypes.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediatypes.h
Convert video quality test from a TEST_F to a TEST fixture.
2018-06-21 15:49:43 +00:00
notifier.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
optional_unittest.cc
Reland "Use absl::optional instead or rtc::Optional"
2018-06-12 19:13:21 +00:00
optional.h
Reland "Use absl::optional instead or rtc::Optional"
2018-06-12 19:13:21 +00:00
OWNERS
Make hbos@webrtc.org OWNER of peerconnection*.
2017-11-13 12:27:29 +00:00
peerconnectionfactoryproxy.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
peerconnectioninterface.h
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
peerconnectionproxy.h
Remove unused/deprecated DTMF methods
2018-06-20 21:00:10 +00:00
proxy.cc
Enable the clang style plugin in primary p2p/ target
2017-11-01 00:19:05 +00:00
proxy.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
refcountedbase.h
New classes RefCounter and RefCountedBase.
2017-10-23 11:46:47 +00:00
rtcerror_unittest.cc
This CL removes all usages of our custom ostream << overloads.
2018-04-03 12:51:00 +00:00
rtcerror.cc
Reland "Reland "Remove our stream << overloads from non-test build targets.""
2018-05-03 10:41:41 +00:00
rtcerror.h
Reland "Reland "Remove our stream << overloads from non-test build targets.""
2018-05-03 10:41:41 +00:00
rtceventlogoutput.h
Move RtcEventLogOutput to api/
2017-10-06 13:58:14 +00:00
rtp_headers.cc
Fix circular dependencies in webrtc_common.
2017-12-15 14:33:26 +00:00
rtp_headers.h
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
rtpparameters_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtpparameters.cc
Add Rtcp parameters for PeerConnection senders
2018-05-28 09:28:59 +00:00
rtpparameters.h
Add functionality to set min bitrate for single stream through RtpEncodingParameters.
2018-06-25 10:10:48 +00:00
rtpreceiverinterface.cc
Reland "Update RTCStatsCollector to work with RtpTransceivers"
2018-02-17 00:01:39 +00:00
rtpreceiverinterface.h
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
rtpsenderinterface.h
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
rtptransceiverinterface.h
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00
setremotedescriptionobserverinterface.h
Reland "SetRemoteDescriptionObserverInterface added."
2017-11-23 19:59:48 +00:00
statstypes.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
statstypes.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
turncustomizer.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
umametrics.cc
Enable the clang style plugin in primary p2p/ target
2017-11-01 00:19:05 +00:00
umametrics.h
Add collection of usage signatures on PeerConnections
2018-05-31 13:07:09 +00:00
videosourceinterface.h
Move VideoStreamEncoderInterface to api/.
2018-05-21 19:50:37 +00:00
videosourceproxy.h
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00