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platform-external-webrtc/api
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Emircan Uysaler bc609eaab1 Revert "Adds support for multiple or no media stream ids." 
This reverts commit 1550292efe680ac79a18004705c908b1cdca54cb.

Reason for revert: 

webkit_layout_tests:fast/peerconnection/RTCPeerConnection-sdpSemantics.html is broken, see below. WebRTC roll isn't going through because of it. This CL looks the most suspicious within the 5 in the range. 

https://chromium-review.googlesource.com/c/chromium/src/+/981899
https://webrtc.googlesource.com/src.git/+log/bb50ce5bb6d5..27f3bf512827
https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_rel_ng/54616

Original change's description:
> Adds support for multiple or no media stream ids.
> 
> With Unified Plan SDP semantics, this adds support for specifying
> either no media stream ids or multiple media stream ids for a
> transceiver/sender/receiver. This includes serializing/deserializing
> SDPs with multiple a=msid lines in a m section, or an "a=msid:-
> <appdata>" line to indicate the no stream case. Note that this does
> not synchronize between multiple streams, this is still just supported
> based upon the first media stream id.
> 
> Bug: webrtc:7932, webrtc:7933
> Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
> Reviewed-on: https://webrtc-review.googlesource.com/61341
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22611}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org,shampson@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:7932, webrtc:7933
Change-Id: I1d4e4332b518ec32b305c8af07679650059d02bb
Reviewed-on: https://webrtc-review.googlesource.com/65000
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22634}
2018-03-27 23:01:55 +00:00
..
audio
Robustification of the echo suppression behavior during headset usage.
2018-03-22 00:23:23 +00:00
audio_codecs
Tests: Pass codec ID argument to audio codecs
2018-03-12 13:25:29 +00:00
call
Add application_data field(s) to RtpPacketToSend and PacketOptions.
2018-02-23 17:20:46 +00:00
ortc
Delete ortc methods using cricket::VideoCapturer.
2018-03-22 08:55:24 +00:00
stats
PeerConnectionInterface::GetStats() with selector argument added.
2018-03-26 12:08:20 +00:00
test
Moved audioproc_f interface into api directory.
2018-03-15 12:31:37 +00:00
video
Fix chromium warnings for SdpVideoFormat.
2018-03-27 08:11:21 +00:00
video_codecs
Fix chromium warnings for SdpVideoFormat.
2018-03-27 08:11:21 +00:00
array_view_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
array_view.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
audio_options.h
Reland "Deprecate the adaptive level controller"
2018-03-09 09:42:13 +00:00
BUILD.gn
Fix chromium warnings for SdpVideoFormat.
2018-03-27 08:11:21 +00:00
candidate.cc
Fix clang style warnings in api/candidate.h
2017-10-26 23:22:18 +00:00
candidate.h
Fix clang style warnings in api/candidate.h
2017-10-26 23:22:18 +00:00
cryptoparams.h
Fix ortc_api circular deps.
2017-11-15 13:31:51 +00:00
datachannelinterface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
DEPS
Move aligned memory utilities to rtc_base/memory/
2018-03-22 14:13:24 +00:00
dtmfsenderinterface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
fakemetricsobserver.cc
Report UMA metrics for received SDP format
2018-03-06 21:22:51 +00:00
fakemetricsobserver.h
Report UMA metrics for received SDP format
2018-03-06 21:22:51 +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
Change error handlers for Set*Description to use RTCError
2018-03-09 15:37:34 +00:00
jsepicecandidate.h
Reland "Clean up libjingle API dependencies."
2017-10-05 13:51:21 +00:00
jsepsessiondescription.h
Introduce webrtc::SdpType, the chosen enum for offer/pranswer/answer
2017-12-06 02:27:32 +00:00
mediaconstraintsinterface.cc
Reland "Deprecate the adaptive level controller"
2018-03-09 09:42:13 +00:00
mediaconstraintsinterface.h
Reland "Deprecate the adaptive level controller"
2018-03-09 09:42:13 +00:00
mediastreaminterface.cc
Final name changing of MediaStreamInterface.label() to id().
2018-03-14 20:30:52 +00:00
mediastreaminterface.h
Final name changing of MediaStreamInterface.label() to id().
2018-03-14 20:30:52 +00:00
mediastreamproxy.h
Final name changing of MediaStreamInterface.label() to id().
2018-03-14 20:30:52 +00:00
mediastreamtrackproxy.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediatypes.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediatypes.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
notifier.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
optional_unittest.cc
Added nullopt and implicit construction to rtc::Optional
2017-11-15 15:50:11 +00:00
optional.cc
Added nullopt and implicit construction to rtc::Optional
2017-11-15 15:50:11 +00:00
optional.h
Added nullopt and implicit construction to rtc::Optional
2017-11-15 15:50:11 +00:00
OWNERS
Make hbos@webrtc.org OWNER of peerconnection*.
2017-11-13 12:27:29 +00:00
peerconnectionfactoryproxy.h
Remove dead version of StartRtcEventLog
2017-10-06 15:18:24 +00:00
peerconnectioninterface.h
Revert "Adds support for multiple or no media stream ids."
2018-03-27 23:01:55 +00:00
peerconnectionproxy.h
PeerConnectionInterface::GetStats() with selector argument added.
2018-03-26 12:08:20 +00:00
proxy.cc
Enable the clang style plugin in primary p2p/ target
2017-11-01 00:19:05 +00:00
proxy.h
Enable the clang style plugin in primary p2p/ target
2017-11-01 00:19:05 +00:00
refcountedbase.h
New classes RefCounter and RefCountedBase.
2017-10-23 11:46:47 +00:00
rtcerror_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtcerror.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtcerror.h
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +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
Drop dependency of common_video on api:libjingle_peerconnection_api.
2018-02-19 13:20:24 +00:00
rtpparameters_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtpparameters.cc
Wire up MID send value to the PeerConnection API
2018-03-26 18:14:30 +00:00
rtpparameters.h
Wire up MID send value to the PeerConnection API
2018-03-26 18:14:30 +00:00
rtpreceiverinterface.cc
Reland "Update RTCStatsCollector to work with RtpTransceivers"
2018-02-17 00:01:39 +00:00
rtpreceiverinterface.h
Reland "Update RTCStatsCollector to work with RtpTransceivers"
2018-02-17 00:01:39 +00:00
rtpsenderinterface.h
Revert "Adds support for multiple or no media stream ids."
2018-03-27 23:01:55 +00:00
rtptransceiverinterface.h
Revert "Adds support for multiple or no media stream ids."
2018-03-27 23:01:55 +00:00
setremotedescriptionobserverinterface.h
Reland "SetRemoteDescriptionObserverInterface added."
2017-11-23 19:59:48 +00:00
statstypes.cc
Reland "Add hugeFramesSent GetStats metric"
2018-03-06 13:38:11 +00:00
statstypes.h
Reland "Add hugeFramesSent GetStats metric"
2018-03-06 13:38:11 +00:00
turncustomizer.h
TurnCustomizer - an interface for modifying stun messages sent by TurnPort
2017-10-11 07:45:29 +00:00
umametrics.cc
Enable the clang style plugin in primary p2p/ target
2017-11-01 00:19:05 +00:00
umametrics.h
Report SRTP error codes to UMA
2018-03-20 18:37:49 +00:00
videosinkinterface.h
Move videosinkinterface.h to common_video to solve a circular dep.
2018-01-04 13:19:49 +00:00
videosourceinterface.cc
Move videosourceinterface to api.
2018-01-05 09:14:19 +00:00
videosourceinterface.h
Move videosourceinterface to api.
2018-01-05 09:14:19 +00:00
videosourceproxy.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00