webrtc/api
Alessio Bazzica b768e8800f Reland "Isolating APM API build target: making :api an actual target."
This reverts commit 61c6e5643e.

Reason for revert: downstream projects prepared for this change

Original change's description:
> Revert "Isolating APM API build target: making :api an actual target."
> 
> This reverts commit a7f77a7c05.
> 
> Reason for revert: breaking downstream
> 
> Original change's description:
> > Isolating APM API build target: making :api an actual target.
> > 
> > This CL is part of a refactoring work to unblock other CLs
> > that would generate a circular dependency when including
> > modules/audio_processing. It will also allow to easily move
> > the APM interface part under //api.
> > 
> > More in detail, this change moves the APM interface files from
> > the build target modules/audio_processing to
> > modules/audio_processing:api. It also adds :api as dependency
> > where needed.
> > 
> > Bug: webrtc:9535
> > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25539}
> 
> TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
> 
> Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9535
> Reviewed-on: https://webrtc-review.googlesource.com/c/109820
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25540}

TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org

Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109884
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:30:06 +00:00
..
audio AEC3: Decrease latency until the delay has been detected 2018-10-31 07:29:48 +00:00
audio_codecs [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
call [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
crypto [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
ortc Reland "Delete leftover includes and declarations for MediaConstraintsInterface" 2018-09-03 09:00:01 +00:00
stats Export symbols needed by the Chromium component build (part 3). 2018-10-16 12:57:04 +00:00
test Update MockVideoEncoder with correct methods. 2018-11-07 12:19:54 +00:00
transport Removes deprecated GetSentPacket from PacketResult. 2018-11-06 19:03:13 +00:00
units [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
video Add HdrMetadata to VideoFrame 2018-11-06 09:29:54 +00:00
video_codecs Remove non-default VideoEncoder::EncoderInfo() ctor 2018-11-07 12:20:59 +00:00
array_view.h Reformat the WebRTC code base 2018-06-19 14:00:39 +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
asyncresolverfactory.h Support domain name ICE candidates 2018-08-24 04:54:43 +00:00
audio_options.cc Remove AECM comfort noise setting from API 2018-10-16 09:42:16 +00:00
audio_options.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +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 Reland "Isolating APM API build target: making :api an actual target." 2018-11-07 14:30:06 +00:00
candidate.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
candidate.h Export symbols needed by the Chromium component build (part 2). 2018-10-15 19:52:31 +00:00
cryptoparams.h Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" 2018-10-11 23:09:07 +00:00
datachannelinterface.cc Enabling clang::find_bad_constructs for libjingle_peerconnection_api. 2018-07-19 09:17:10 +00:00
datachannelinterface.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
DEPS Add interfaces for using MediaTransport as the transport for data channels. 2018-10-31 18:44:31 +00:00
dtmfsenderinterface.h Add "tones remaining" argument to DTMF ontonechange callback 2018-09-07 17:29:37 +00:00
fec_controller.h Revert "Revert "Enables PeerConnectionFactory using external fec controller"" 2018-02-20 12:41:55 +00:00
jsep.cc Move SdpType from/to string definition close to declaration. 2018-10-12 09:59:40 +00:00
jsep.h Export symbols needed by the Chromium component build (part 6). 2018-10-23 06:48:51 +00:00
jsepicecandidate.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
jsepicecandidate.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
jsepsessiondescription.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
media_transport_interface.cc Delete the default implementations of MediaTransportInterface methods. 2018-11-01 00:15:52 +00:00
media_transport_interface.h Delete the default implementations of MediaTransportInterface methods. 2018-11-01 00:15:52 +00:00
mediaconstraintsinterface.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
mediaconstraintsinterface.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
mediastreaminterface.cc Remove deprecated AudioProcessing::GetStatistics function 2018-11-01 11:21:15 +00:00
mediastreaminterface.h Remove deprecated AudioProcessing::GetStatistics function 2018-11-01 11:21:15 +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
OWNERS Use checkdeps to ensure API headers don't include internal headers. 2018-10-24 12:55:01 +00:00
peerconnectionfactoryproxy.h Delete almost all use of MediaConstraintsInterface in the PeerConnection API 2018-08-23 07:14:37 +00:00
peerconnectioninterface.cc Compute RTCConnectionState and RTCIceConnectionState. 2018-10-22 11:33:17 +00:00
peerconnectioninterface.h Add a configuration parameter for using the media transport for data channels. 2018-11-05 21:05:22 +00:00
peerconnectionproxy.h Delete almost all use of MediaConstraintsInterface in the PeerConnection API 2018-08-23 07:14:37 +00:00
proxy.cc Add default constructor for rtc::Event 2018-11-07 08:57:50 +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.cc Remove all remaining non-test uses of std::stringstream. 2018-09-13 08:52:05 +00:00
rtcerror.h Remove all remaining non-test uses of std::stringstream. 2018-09-13 08:52:05 +00:00
rtcerror_unittest.cc Remove all remaining non-test uses of std::stringstream. 2018-09-13 08:52:05 +00:00
rtceventlogoutput.h Move RtcEventLogOutput to api/ 2017-10-06 13:58:14 +00:00
rtp_headers.cc Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
rtp_headers.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
rtpparameters.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
rtpparameters.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
rtpparameters_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtpreceiverinterface.cc Injects FrameEncryptorInterface into RtpSender. 2018-08-30 00:33:54 +00:00
rtpreceiverinterface.h Injects FrameEncryptorInterface into RtpSender. 2018-08-30 00:33:54 +00:00
rtpsenderinterface.cc Add support for send_encodings parameters in addTransceiver 2018-10-01 22:56:30 +00:00
rtpsenderinterface.h Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
rtptransceiverinterface.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
rtptransceiverinterface.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
setremotedescriptionobserverinterface.h Reland "SetRemoteDescriptionObserverInterface added." 2017-11-23 19:59:48 +00:00
statstypes.cc Reimplement rtc::ToString and rtc::FromString without streams. 2018-08-16 16:14:01 +00:00
statstypes.h Revert "Add framesRendered to StatsReport" 2018-07-27 14:53:07 +00:00
turncustomizer.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
umametrics.h Remove MetricsObserverInterface. 2018-07-19 23:00:20 +00:00
videosourceproxy.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00