- 0e61fdd Use backticks not vertical bars to denote variables in comments for /api by Artem Titov · 3 years, 8 months ago
- e93fe6c Enable Chromium to stop including api/proxy.h indirectly. by Markus Handell · 3 years, 11 months ago
- 4ccdf932 VideoRtpReceiver & AudioRtpReceiver threading fixes. by Tommi · 3 years, 11 months ago
- 9d9b8de Reland "Rename SIGNALING and WORKER to PRIMARY and SECONDARY" by Mirko Bonadei · 4 years, 1 month ago
- 07a01d0 Revert "Rename SIGNALING and WORKER to PRIMARY and SECONDARY" by Mirko Bonadei · 4 years, 1 month ago
- a37f2bd Rename SIGNALING and WORKER to PRIMARY and SECONDARY by Harald Alvestrand · 4 years, 1 month ago
- e904161 Replace RTC_DEPRECATED with ABSL_DEPRECATED by Danil Chapovalov · 4 years, 1 month ago
- fc83cdc Avoid proxy thread hops for reading const properties. by Tomas Gunnarsson · 4 years, 7 months ago
- 412a31b Insert frame transformer between Depacketizer and Decoder. by Marina Ciocea · 5 years ago
- 934afc6 Deprecate RtpReceiver's SetParameters method by Saurav Das · 5 years ago
- 35214fc Add missing RTC_EXPORT for the component build. by Mirko Bonadei · 5 years ago
- a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 6 years ago
- 015c3cb Remove deprecated constructors of RtpSource by Johannes Kron · 6 years ago
- b5d9183 Add RTP timestamp to contributing sources by Johannes Kron · 6 years ago
- 428dcb2 Remove SetLatency/GetLatency from MediaSourceInterface API level by Ruslan Burakov · 6 years ago
- 4bac79e Add SetJitterBufferMinimumDelay method to RtpReceiverInterface. by Ruslan Burakov · 6 years ago
- 22f9925 webrtc: Remove semicolons. by Nico Weber · 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/rtpreceiverinterface.h]
- 5b14778 [Unified Plan] If "a=msid" is missing, create default stream. by Henrik Boström · 6 years ago
- d81ac95 Injects FrameEncryptorInterface into RtpSender. by Benjamin Wright · 7 years ago
- 199e27b RtpReceiverInterface::stream_ids() added. by Henrik Boström · 7 years ago
- 0bc58cf Replace rtc::Optional with absl::optional in api by Danil Chapovalov · 7 years ago
- 665174f Reformat the WebRTC code base by Yves Gerey · 7 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
- 2a5ce2b Fix clang style errors in rtp_rtcp and dependant targets by Danil Chapovalov · 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
- 9e6fd2b Add streams() to RtpReceiverInterface and implementations. by Henrik Boström · 7 years ago
- 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
- bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/api/rtpreceiverinterface.h]
- 2b70634 Add audio_level member to RtpSource and set it from RtpReceiverImpl::IncomingRtpPacket. by zstein · 8 years ago
- c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
- 0426222 Modified the rtp_receiver_unittests. by zhihuang · 8 years ago
- 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
- fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
- 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
- b10f32f Adding more comments to every header file in api/ subdirectory. 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
- ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
- 3784b4a Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 9 years ago
- 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
- 1a7162d Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 9 years ago
- bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
- 184a3fd Forward the SignalFirstPacketReceived to RtpReceiver. by zhihuang · 9 years ago
- a601f5c Separating internal and external methods of RtpSender/RtpReceiver. by deadbeef · 9 years ago
- db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 9 years ago
- 72c8d2b Rename BEGIN_PROXY_MAP --> BEGIN_SIGNALLING_PROXY_MAP. by nisse · 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 (90%) from talk/app/webrtc/rtpreceiverinterface.h]
- a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
- 70ab1a1 Exposing RtpSenders and RtpReceivers from PeerConnection. by deadbeef · 10 years ago
- 6979b02 Adding stub files for RtpSender/RtpReceiver. by deadbeef · 10 years ago
