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Mirko Bonadei 94c94205f7 Remove cricket::BundleFilter. 
This code is never built by GN, and the header is never included.

Bug: webrtc:9855
Change-Id: I16e6a54cc95629917d454f91d9bcc99fc55d8a00
Reviewed-on: https://webrtc-review.googlesource.com/c/111754
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25753}
2018-11-22 14:04:33 +00:00
..
test
Move webrtc::CreatePeerConnectionFactory definition next to decl.
2018-11-22 09:07:51 +00:00
audiotrack.cc
Remove "using rtc::scoped_ptr" from audiotrack.cc.
2017-11-28 21:29:18 +00:00
audiotrack.h
…
BUILD.gn
Move webrtc::CreatePeerConnectionFactory definition next to decl.
2018-11-22 09:07:51 +00:00
channel_unittest.cc
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
2018-10-30 08:06:49 +00:00
channel.cc
Implement TargetBitrate, NetworkRoute and overhead features of media transport interface.
2018-11-16 19:31:11 +00:00
channel.h
Implement TargetBitrate, NetworkRoute and overhead features of media transport interface.
2018-11-16 19:31:11 +00:00
channelinterface.h
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
channelmanager_unittest.cc
Reland "Reland "Propagate media transport to media channel.""
2018-10-17 20:54:06 +00:00
channelmanager.cc
Exposing audio and video engines directly.
2018-11-16 15:40:45 +00:00
channelmanager.h
Reland "Reland "Propagate media transport to media channel.""
2018-10-17 20:54:06 +00:00
datachannel_unittest.cc
Removing warning suppression flags from pc/.
2018-07-04 10:35:27 +00:00
datachannel.cc
Implement data channels over media transport.
2018-11-09 00:40:32 +00:00
datachannel.h
Implement data channels over media transport.
2018-11-09 00:40:32 +00:00
DEPS
Allows injection of network controller factory into peer connection factory.
2018-05-18 17:07:16 +00:00
dtlssrtptransport_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
dtlssrtptransport.cc
Compute RTCConnectionState and RTCIceConnectionState.
2018-10-22 11:33:17 +00:00
dtlssrtptransport.h
Compute RTCConnectionState and RTCIceConnectionState.
2018-10-22 11:33:17 +00:00
dtmfsender_unittest.cc
Add "tones remaining" argument to DTMF ontonechange callback
2018-09-07 17:29:37 +00:00
dtmfsender.cc
Add "tones remaining" argument to DTMF ontonechange callback
2018-09-07 17:29:37 +00:00
dtmfsender.h
Use AsyncInvoker in DtmfSender instead of MessageHandler
2018-08-31 22:49:26 +00:00
externalhmac.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
externalhmac.h
[CFI] Fix cfi-icall failures
2017-12-12 09:27:07 +00:00
iceserverparsing_unittest.cc
Revert "Add SSLConfig object to IceServer."
2018-09-12 10:46:04 +00:00
iceserverparsing.cc
Revert "Add SSLConfig object to IceServer."
2018-09-12 10:46:04 +00:00
iceserverparsing.h
…
jsepicecandidate.cc
Delete unused utf8 conversion utilities
2018-10-11 16:16:33 +00:00
jsepsessiondescription_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
jsepsessiondescription.cc
Move SdpType from/to string definition close to declaration.
2018-10-12 09:59:40 +00:00
jseptransport_unittest.cc
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
jseptransport.cc
Plug-in media transport state listener
2018-11-01 15:52:56 +00:00
jseptransport.h
Plug-in media transport state listener
2018-11-01 15:52:56 +00:00
jseptransportcontroller_unittest.cc
Pass the media transport from JsepTransportController to Call.
2018-11-15 17:36:48 +00:00
jseptransportcontroller.cc
Create 'MaybeCreateMediaTransport' function
2018-11-21 16:22:36 +00:00
jseptransportcontroller.h
Create 'MaybeCreateMediaTransport' function
2018-11-21 16:22:36 +00:00
localaudiosource_unittest.cc
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
localaudiosource.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
localaudiosource.h
AudioSource allows implementations to return settings
2018-11-13 16:30:09 +00:00
mediaconstraintsinterface_unittest.cc
Rename a few MediaConfig::Video flags for consistency.
2018-01-22 17:32:58 +00:00
mediasession_unittest.cc
Rename too long variable name to extmap_allow_mixed
2018-10-23 09:25:19 +00:00
mediasession.cc
Add offer_extmap_allow_mixed to RTCConfiguration
2018-11-12 12:35:45 +00:00
mediasession.h
Add offer_extmap_allow_mixed to RTCConfiguration
2018-11-12 12:35:45 +00:00
mediastream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastreamobserver.cc
…
mediastreamobserver.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
mediastreamtrack.h
…
OWNERS
Adding shampson (me) as an owner to pc/ & api/.
2018-07-03 20:39:17 +00:00
peerconnection_bundle_unittest.cc
Move webrtc::CreatePeerConnectionFactory definition next to decl.
2018-11-22 09:07:51 +00:00
peerconnection_crypto_unittest.cc
Move webrtc::CreatePeerConnectionFactory definition next to decl.
2018-11-22 09:07:51 +00:00
peerconnection_datachannel_unittest.cc
Implement data channels over media transport.
2018-11-09 00:40:32 +00:00
peerconnection_histogram_unittest.cc
Properly setup MockPeerConnectionObserver in tests (continued).
2018-11-01 09:35:07 +00:00
peerconnection_ice_unittest.cc
Move webrtc::CreatePeerConnectionFactory definition next to decl.
2018-11-22 09:07:51 +00:00
peerconnection_integrationtest.cc
Add test PeerConnectionIntegrationTest.MediaTransportBidirectionalAudio
2018-11-21 07:59:34 +00:00
peerconnection_jsep_unittest.cc
Properly setup MockPeerConnectionObserver in tests (continued).
2018-11-01 09:35:07 +00:00
peerconnection_media_unittest.cc
Exposing audio and video engines directly.
2018-11-16 15:40:45 +00:00
peerconnection_rampup_tests.cc
Move webrtc::CreatePeerConnectionFactory definition next to decl.
2018-11-22 09:07:51 +00:00
peerconnection_rtp_unittest.cc
Move webrtc::CreatePeerConnectionFactory definition next to decl.
2018-11-22 09:07:51 +00:00
peerconnection_signaling_unittest.cc
Move webrtc::CreatePeerConnectionFactory definition next to decl.
2018-11-22 09:07:51 +00:00
peerconnection.cc
Add check for media transport and bundle policy
2018-11-21 16:36:39 +00:00
peerconnection.h
Pass the media transport from JsepTransportController to Call.
2018-11-15 17:36:48 +00:00
peerconnectionendtoend_unittest.cc
Run some peer connection end-to-end tests with an empty audio encoder factory
2018-11-13 16:14:50 +00:00
peerconnectionfactory_unittest.cc
Move webrtc::CreatePeerConnectionFactory definition next to decl.
2018-11-22 09:07:51 +00:00
peerconnectionfactory.cc
Exposing audio and video engines directly.
2018-11-16 15:40:45 +00:00
peerconnectionfactory.h
Pass MediaTransportFactory to PeerConnectionFactory.
2018-10-08 18:11:06 +00:00
peerconnectioninterface_unittest.cc
Move webrtc::CreatePeerConnectionFactory definition next to decl.
2018-11-22 09:07:51 +00:00
peerconnectioninternal.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
peerconnectionwrapper.cc
Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
2018-07-12 21:35:47 +00:00
peerconnectionwrapper.h
Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
2018-07-12 21:35:47 +00:00
proxy_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
remoteaudiosource.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
remoteaudiosource.h
Configure media flow correctly with Unified Plan
2018-01-18 19:01:38 +00:00
rtcpmuxfilter_unittest.cc
Delete unused code from media/base/testutils.{cc,h}
2018-10-18 16:06:33 +00:00
rtcpmuxfilter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcpmuxfilter.h
Move sessiondescription.h/cc from p2p/base to pc/
2017-12-20 00:21:52 +00:00
rtcstats_integrationtest.cc
Disable RTCStatsIntegrationTest.GetsStatsWhileDestroyingPeerConnection
2018-11-16 17:13:40 +00:00
rtcstatscollector_unittest.cc
Refactor: Renaming ssl_cert_chain to GetSSLCertificateChain()
2018-10-25 15:52:06 +00:00
rtcstatscollector.cc
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtcstatscollector.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
rtcstatstraversal_unittest.cc
Stats traversal algorithm added.
2018-03-12 10:54:09 +00:00
rtcstatstraversal.cc
RTCStatsCollector::GetStatsReport() with optional selector argument.
2018-03-19 15:32:16 +00:00
rtcstatstraversal.h
Stats traversal algorithm added.
2018-03-12 10:54:09 +00:00
rtpmediautils_unittest.cc
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
rtpmediautils.cc
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
rtpmediautils.h
Reland "Reland "Remove our stream << overloads from non-test build targets.""
2018-05-03 10:41:41 +00:00
rtpparametersconversion_unittest.cc
Remove multiple RTX codec entries in GetRtpReceiver/SenderCapabilities
2018-11-07 15:35:56 +00:00
rtpparametersconversion.cc
Remove multiple RTX codec entries in GetRtpReceiver/SenderCapabilities
2018-11-07 15:35:56 +00:00
rtpparametersconversion.h
Add PeerConnection GetRtpSender/ReceiverCapabilities
2018-06-28 12:40:07 +00:00
rtpreceiver.cc
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtpreceiver.h
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtpsender.cc
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtpsender.h
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtpsenderreceiver_unittest.cc
Add RTCP and simulcast support for RTCRtpReceiver::getParameters()
2018-11-20 14:03:18 +00:00
rtptransceiver_unittest.cc
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtptransceiver.cc
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtptransceiver.h
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtptransport_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransport.cc
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
rtptransport.h
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
rtptransportinternal.h
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
rtptransportinternaladapter.h
Remove MetricsObserverInterface.
2018-07-19 23:00:20 +00:00
rtptransporttestutil.h
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
sctputils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sctputils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sctputils.h
…
sdputils.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
sdputils.h
Move sessiondescription.h/cc from p2p/base to pc/
2017-12-20 00:21:52 +00:00
sessiondescription_unittest.cc
Rename too long variable name to extmap_allow_mixed
2018-10-23 09:25:19 +00:00
sessiondescription.cc
Rename too long variable name to extmap_allow_mixed
2018-10-23 09:25:19 +00:00
sessiondescription.h
Add offer_extmap_allow_mixed to RTCConfiguration
2018-11-12 12:35:45 +00:00
srtpfilter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
srtpfilter.cc
Delete unused utf8 conversion utilities
2018-10-11 16:16:33 +00:00
srtpfilter.h
When SDES is used, pass pre-shared key to media transport.
2018-10-31 16:04:16 +00:00
srtpsession_unittest.cc
Remove deprecated field_trial_default and metrics_default.
2018-09-28 07:21:07 +00:00
srtpsession.cc
Throttle the RTP decryption error messages in the SrtpSession and SrtpTransport
2018-10-09 12:10:55 +00:00
srtpsession.h
Throttle the RTP decryption error messages in the SrtpSession and SrtpTransport
2018-10-09 12:10:55 +00:00
srtptestutil.h
…
srtptransport_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
srtptransport.cc
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
srtptransport.h
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
statscollector_unittest.cc
Remove deprecated AudioProcessing::GetStatistics function
2018-11-01 11:21:15 +00:00
statscollector.cc
Refactor: Renaming ssl_cert_chain to GetSSLCertificateChain()
2018-10-25 15:52:06 +00:00
statscollector.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamcollection.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
trackmediainfomap_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
trackmediainfomap.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
trackmediainfomap.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
transportstats.cc
Create the JsepTransportController and JsepTransport2.
2018-02-23 00:13:45 +00:00
transportstats.h
Create the JsepTransportController and JsepTransport2.
2018-02-23 00:13:45 +00:00
videocapturertracksource_unittest.cc
Add default constructor for rtc::Event
2018-11-07 08:57:50 +00:00
videocapturertracksource.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
videocapturertracksource.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
videotrack_unittest.cc
Refactor VideoTrackSource, without raw pointer injection.
2018-05-23 15:42:10 +00:00
videotrack.cc
Enable cpplint in pc/
2017-10-30 18:08:29 +00:00
videotrack.h
Reland "[cleanup] Remove useless includes."
2018-10-08 07:44:19 +00:00
videotracksource.cc
Delete deprecated VideoTrackSource constructor.
2018-06-11 19:15:10 +00:00
videotracksource.h
Export symbols needed by the Chromium component build (part 2).
2018-10-15 19:52:31 +00:00
webrtcsdp_unittest.cc
Rename too long variable name to extmap_allow_mixed
2018-10-23 09:25:19 +00:00
webrtcsdp.cc
Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
2018-10-25 14:13:44 +00:00
webrtcsdp.h
Export symbols needed by the Chromium component build (part 7).
2018-10-25 11:41:16 +00:00
webrtcsessiondescriptionfactory.cc
Remove obsolete comment (WebRtcSessionDescriptionFactory ctor)
2018-11-12 15:24:58 +00:00
webrtcsessiondescriptionfactory.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00