Log - 976faae028c100ad66dbe865c954960b6d38c2f8 - platform/external/webrtc

e77912b Insert frame transformer between Encoded and Packetizer. by Marina Ciocea · 4 years, 6 months ago
35214fc Add missing RTC_EXPORT for the component build. by Mirko Bonadei · 5 years ago
1ff16c8 Add RtpSenderInterface.SetStreams by Guido Urdaneta · 5 years ago
cc18917 Revert "Improve spec compliance of SetStreamIDs in RtpSenderInterface" by Henrik Andreassson · 5 years ago
df5731e Improve spec compliance of SetStreamIDs in RtpSenderInterface by Guido Urdaneta · 5 years ago
e1e789b Removing non-const RtpSenderInterface::GetParameters(). by Amit Hilbuch · 6 years ago
22f9925 webrtc: Remove semicolons. by Nico Weber · 6 years ago
2297d33 Rejected simulcast layers will no longer appear in GetParameters(). by Amit Hilbuch · 6 years ago
d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
4a7b3ac Add DTLSTransport info into sender/receiver state. by Harald Alvestrand · 6 years ago
10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from api/rtpsenderinterface.h]
2e00abc Reland "[cleanup] Remove useless includes." by Yves Gerey · 6 years ago
96a0f61 Revert "[cleanup] Remove useless includes." by Oleh Prypin · 6 years ago
be8b534 [cleanup] Remove useless includes. by Yves Gerey · 6 years ago
892acf0 Add support for send_encodings parameters in addTransceiver by Florent Castelli · 6 years ago
d81ac95 Injects FrameEncryptorInterface into RtpSender. by Benjamin Wright · 6 years ago
79eb4dd Enabling clang::find_bad_constructs for libjingle_peerconnection_api. by Mirko Bonadei · 6 years ago
0bc58cf Replace rtc::Optional with absl::optional in api by Danil Chapovalov · 6 years ago
5565981 Add functionality to set min/max bitrate per simulcast layer through RtpEncodingParameters. by Åsa Persson · 6 years ago
665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
4c6390a Remove deprecated RtpSenderInterface::GetParameters() const method by Florent Castelli · 6 years ago
cebf50f Reland "Implement RtpParameters.transaction_id for PC RtpSenderInterface" by Florent Castelli · 6 years ago
909338b Revert "Implement RtpParameters.transaction_id for PC RtpSenderInterface" by Max Morin · 6 years ago
5faf36e Implement RtpParameters.transaction_id for PC RtpSenderInterface by Florent Castelli · 6 years ago
5b4f075 Reland "Reland "Adds support for multiple or no media stream ids."" by Seth Hampson · 6 years ago
191bf5c Revert "Reland "Adds support for multiple or no media stream ids."" by Tomas Gunnarsson · 6 years ago
f351c34 Reland "Adds support for multiple or no media stream ids." by Seth Hampson · 6 years ago
bc609ea Revert "Adds support for multiple or no media stream ids." by Emircan Uysaler · 6 years ago
1550292 Adds support for multiple or no media stream ids. by Seth Hampson · 6 years ago
57858b3 Reland "Update RTCStatsCollector to work with RtpTransceivers" by Steve Anton · 7 years ago
ee2388f Revert "Update RTCStatsCollector to work with RtpTransceivers" by Guido Urdaneta · 7 years ago
56bae8d Update RTCStatsCollector to work with RtpTransceivers by Steve Anton · 7 years ago
ba37b4b Change return type of RtpSenderInterface::SetParameters from bool to RTCError by Zach Stein · 7 years ago
c72af93 Reland "Move stats ID generation from SSRC to local ID" by Harald Alvestrand · 7 years ago
c0092c3 Revert "Move stats ID generation from SSRC to local ID" by Erik Språng · 7 years ago
e357a4d Move stats ID generation from SSRC to local ID by Harald Alvestrand · 7 years ago
92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/api/rtpsenderinterface.h]
c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
b10f32f Adding more comments to every header file in api/ subdirectory. by deadbeef · 8 years ago
20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 8 years ago
7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago
d99a200 Adding some features to proxy.h, and restructuring the macros. by deadbeef · 8 years ago
a601f5c Separating internal and external methods of RtpSender/RtpReceiver. by deadbeef · 8 years ago
72c8d2b Rename BEGIN_PROXY_MAP --> BEGIN_SIGNALLING_PROXY_MAP. by nisse · 8 years ago
dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 8 years ago
9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago
b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (93%) from talk/app/webrtc/rtpsenderinterface.h]
a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
fac0655 Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
5def7b9 Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ ) by deadbeef · 9 years ago
6834fa1 Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) by deadbeef · 9 years ago
8f46c63 Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) by deadbeef · 9 years ago
ac9d92c Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
70ab1a1 Exposing RtpSenders and RtpReceivers from PeerConnection. by deadbeef · 9 years ago
6979b02 Adding stub files for RtpSender/RtpReceiver. by deadbeef · 9 years ago