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webrtc_m130/webrtc/api
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andersc 063f0c0d3a Reland of Prepare for injectable SW decoders (patchset #1 id:1 of https://codereview.webrtc.org/3010953002/ ) 
Reason for revert:
Fix bug introduced by keeping the allocated decoders in a map.

Original issue's description:
> Revert of Prepare for injectable SW decoders (patchset #3 id:40001 of https://codereview.webrtc.org/3009973002/ )
>
> Reason for revert:
> Tentative revert since it seems to cause problems in Chrome, MAC.
>
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Tester/builds/42684
>
>
>
> Original issue's description:
> > Prepare for injectable SW decoders
> >
> > Pretty much mirrors the work done on the encoding side in CLs:
> >
> > "Clean up ownership of webrtc::VideoEncoder"
> > https://codereview.webrtc.org/3007643002/
> >
> > "Let VideoEncoderSoftwareFallbackWrapper own the wrapped encoder"
> > https://codereview.webrtc.org/3007683002/
> >
> > "WebRtcVideoEngine: Encapsulate logic for unifying internal and external video codecs"
> > https://codereview.webrtc.org/3006713002/
> >
> > BUG=webrtc:7925
> >
> > Review-Url: https://codereview.webrtc.org/3009973002
> > Cr-Commit-Position: refs/heads/master@{#19641}
> > Committed: 084c55a63a
>
> TBR=magjed@webrtc.org,andersc@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7925
>
> Review-Url: https://codereview.webrtc.org/3010953002
> Cr-Commit-Position: refs/heads/master@{#19647}
> Committed: 1f88531038

TBR=magjed@webrtc.org,perkj@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7925

Review-Url: https://codereview.webrtc.org/3005363002
Cr-Commit-Position: refs/heads/master@{#19782}
2017-09-11 18:50:51 +00:00
..
audio
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
audio_codecs
Add and modify a few ANA stats.
2017-09-08 20:24:21 +00:00
call
Reland of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2877013002/ )
2017-05-12 05:07:37 +00:00
ortc
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
stats
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
test
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
video
Fix code formating in api/video/video_content_type.*
2017-09-11 09:20:45 +00:00
video_codecs
Reland of Prepare for injectable SW decoders (patchset #1 id:1 of https://codereview.webrtc.org/3010953002/ )
2017-09-11 18:50:51 +00:00
array_view_unittest.cc
Move array_view.h to webrtc/api/
2017-09-04 12:43:17 +00:00
array_view.h
Move array_view.h to webrtc/api/
2017-09-04 12:43:17 +00:00
BUILD.gn
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
datachannel.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
datachannelinterface.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
DEPS
Recently we moved webrtc/base to webrtc/rtc_base, so these
2017-08-30 11:45:58 +00:00
dtmfsenderinterface.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fakemetricsobserver.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fakemetricsobserver.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
jsep.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
jsepicecandidate.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
jsepsessiondescription.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediaconstraintsinterface.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediaconstraintsinterface.h
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
mediastream.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreaminterface.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreaminterface.h
Change RtpSender to have multiple stream_ids
2017-09-07 17:03:24 +00:00
mediastreamproxy.h
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
mediastreamtrack.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamtrackproxy.h
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
mediatypes.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
mediatypes.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
notifier.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
optional_unittest.cc
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
optional.cc
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
optional.h
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
peerconnectionfactoryproxy.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
peerconnectioninterface.h
Add support for changing the TLS elliptic curve set.
2017-09-08 20:02:03 +00:00
peerconnectionproxy.h
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
2017-06-02 21:37:37 +00:00
proxy.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcerror_unittest.cc
Move rtc_api_unittests into rtc_unittests.
2017-02-24 09:17:43 +00:00
rtcerror.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcerror.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtpparameters_unittest.cc
Move RtpExtension to api/ directory and config.h/.cc to call/.
2017-09-01 13:29:30 +00:00
rtpparameters.cc
Move RtpExtension to api/ directory and config.h/.cc to call/.
2017-09-01 13:29:30 +00:00
rtpparameters.h
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
rtpreceiverinterface.h
Add audio_level member to RtpSource and set it from RtpReceiverImpl::IncomingRtpPacket.
2017-08-24 21:52:17 +00:00
rtpsender.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
rtpsenderinterface.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
statstypes.cc
Add and modify a few ANA stats.
2017-09-08 20:24:21 +00:00
statstypes.h
Update thread annotiation macros to use RTC_ prefix
2017-09-09 11:17:22 +00:00
streamcollection.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
umametrics.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
videosourceproxy.h
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
videotracksource.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
webrtcsdp.h
Added api/webrtcsdp.h forwarding header to work around upstream projects.
2017-01-24 15:13:59 +00:00