1. 39b8eb3 Fix Chromium GN build problem introduced in 608c3cfe by Karl Wiberg · 9 years ago
  2. 4e14f09 Add support for external decoders in ACM by kwiberg · 9 years ago
  3. e7cdc7f No-op CL to test if video engine core problem solved. by phoglund · 9 years ago
  4. d8ee4f9 Use RtcpPacket to send BYE in RtcpSender by sprang · 9 years ago
  5. 608c3cf iSAC: Make separate AudioEncoder and AudioDecoder objects by kwiberg · 9 years ago
  6. 2159b89 Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots. by Peter Thatcher · 9 years ago
  7. 9deaa86 Fix initialization/termination of AudioDeviceTemplate by kaorimatz · 9 years ago
  8. 7612f17 Fix accidental redeclaration. by Andrew MacDonald · 9 years ago
  9. c0775c0 Fix accessing uninitialized variables when not processing a reverse stream. by Andrew MacDonald · 9 years ago
  10. ea1012b address comments from https://codereview.webrtc.org/1277263002/ by guoweis · 9 years ago
  11. 5bdafd4 Revert "Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."" by minyuel · 9 years ago
  12. 81a3e60 Use RtcpPacket to send TMMBR in RtcpSender by sprang · 9 years ago
  13. dd4edc5 Reland of Use RtcpPacket to send REMB in RtcpSender (patchset #1 id:1 of https://codereview.webrtc.org/1300863002/ ) by sprang · 9 years ago
  14. c232096 Remove cricket::VideoProcessor and AddVideoProcessor() functionality by Magnus Jedvert · 9 years ago
  15. 9d15c66 Include webrtc/base/json.h rather than from jsoncpp directly. by phoglund · 9 years ago
  16. 22ff75a Add unit tests for more packet types in rtcp_sender_unittest. by asapersson · 9 years ago
  17. bfab5cb Fix some minor errors with the voice engine caused by the refactor CL https://codereview.webrtc.org/1229283003/. by Peter Thatcher · 9 years ago
  18. a5b273a Fixing problems with RTP extension ID conflict resolution by deadbeef · 9 years ago
  19. 874ca3a Don't do reconfiguration if recv codec order/preference changes by deadbeef · 9 years ago
  20. 5a3acd8 First step of passive aggressive nomination. by honghaiz · 9 years ago
  21. fe3bc9d Relanding "Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied." by Guo-wei Shieh · 9 years ago
  22. a1f590f Add helper class GuardedAsyncInvoker to protect against thread dying by Magnus Jedvert · 9 years ago
  23. d3de9c5 rtc::Bind: Capture method objects as scoped_refptr if they are ref counted by Magnus Jedvert · 9 years ago
  24. efefda6 Move SystemInfo to rtc_base_approved and delete unused code. by tommi · 9 years ago
  25. ff020c0 Android: Move common functions from VideoRendererGui to new RendererCommon file by Magnus Jedvert · 9 years ago
  26. 41b3a38 Adds RTCCertificate, a reference counted object indirectly owning an SSLCertificate (by owning the SSLIdentity). by Henrik Boström · 9 years ago
  27. 9e260f1 Prevent TimeUntilNextProcess log spam. by pbos · 9 years ago
  28. d476b95 Android EglBase: Add helper functions to query the surface size by Magnus Jedvert · 9 years ago
  29. 081f34b Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots." by Peter Thatcher · 9 years ago
  30. 3d564c1 Add instrumentation to track the IceEndpointType. by Guo-wei Shieh · 9 years ago
  31. 86cb923 In the past, P2PPortAllocator.enable_multiple_routes is the indicator whether we should bind to the any address. It's easy to translate that into a port allocator flag in P2PPortAllocator's ctor. Going forward, we have to depend on an asynchronous permission check to determine whether gathering local address is allowed or not, hence the current way of passing it through constructor approach won't work any more. The asynchronous check will trigger SignalNetowrksChanged so we could only check that inside DoAllocate. by Guo-wei Shieh · 9 years ago
  32. 47872ec In the past, P2PPortAllocator.enable_multiple_routes is the indicator whether we should bind to the any address. It's easy to translate that into a port allocator flag in P2PPortAllocator's ctor. Going forward, we have to depend on an asynchronous permission check to determine whether gathering local address is allowed or not, hence the current way of passing it through constructor approach won't work any more. The asynchronous check will trigger SignalNetowrksChanged so we could only check that inside DoAllocate. by Guo-wei Shieh · 9 years ago
  33. 5d69648 Disabling TestUdpIPv6 on Linux by minyue · 9 years ago
  34. 048e80c Revert of Revert "Remove CpuMonitor and related, unused, code." (patchset #1 id:1 of https://codereview.webrtc.org/1287913004/ ) by tommi · 9 years ago
  35. c844ca4 Move scoped_ptr.h to rtc_base_approved. by Tommi · 9 years ago
  36. 1f4ffe0 NetEq: Implement two UMA stats for delay adaptation. by Henrik Lundin · 9 years ago
  37. a472e96 Revert "Remove CpuMonitor and related, unused, code." by Guo-wei Shieh · 9 years ago
  38. 370c884 Revert "Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied." by Guo-wei Shieh · 9 years ago
  39. ba9ab4c In the past, P2PPortAllocator.enable_multiple_routes is the indicator whether we should bind to the any address. It's easy to translate that into a port allocator flag in P2PPortAllocator's ctor. Going forward, we have to depend on an asynchronous permission check to determine whether gathering local address is allowed or not, hence the current way of passing it through constructor approach won't work any more. The asynchronous check will trigger SignalNetowrksChanged so we could only check that inside DoAllocate. by Guo-wei Shieh · 9 years ago
  40. 1a24012 Remove CpuMonitor and related, unused, code. by Tommi · 9 years ago
  41. 0a2955f Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied. by Guo-wei Shieh · 9 years ago
  42. bef77e2 NetEq: Implement logging of Delayed Packet Outage Events by Henrik Lundin · 9 years ago
  43. d84dcbd rtpAnalyze matlab tool: filter out RTCP packets by henrik.lundin · 9 years ago
  44. 141c595 Revert of Use RtcpPacket to send REMB in RtcpSender (patchset #1 id:1 of https://codereview.webrtc.org/1290573004/ ) by sprang · 9 years ago
  45. 35ab4ba Use RtcpPacket to send REMB in RtcpSender by Erik Språng · 9 years ago
  46. 7b3de4b Re-enable LLVM LTO on Neon targets. by Peter Collingbourne · 9 years ago
  47. 3260133 Fix -Wreorder compile error after https://codereview.webrtc.org/1189583002/ by Nico Weber · 9 years ago
  48. dbe5bd9 Delete unused function SetSessionError. by Nico Weber · 9 years ago
  49. b6d4ec4 Support generation of EC keys using P256 curve and support ECDSA certs. by Torbjorn Granlund · 9 years ago
  50. 1147702 WebRTC Bug 4865 by Guo-wei Shieh · 9 years ago
  51. 805d8fb Remove WebRtcIsac_Highpass_float(). by pkasting · 9 years ago
  52. 55e9a7d Add Android VideoRendererGui events. by Alex Glaznev · 9 years ago
  53. d332580 Add stats overlay to iOS AppRTCDemo. by Zeke Chin · 9 years ago
  54. 60d9b33 Integrate Intelligibility with APM by ekmeyerson · 9 years ago
  55. 03bb7c7 Add LoudestFilter in ConferenceTransport by minyue · 9 years ago
  56. 4c530dc Delete dummy dtlsidentityservice.[cc,h] files. by hbos · 9 years ago
  57. d5031fc Android VideoRendererGui: Add dispose function by magjed · 9 years ago
  58. af5c035 VideoCapturerAndroid: Release queued camera frames when stopCapture() is called by magjed · 9 years ago
  59. 38f8893 WebRTC Bug 4865 by Guo-wei Shieh · 9 years ago
  60. ee8c6d3 In PeerConnectionTestWrapper, put audio input on a separate thread. by deadbeef · 9 years ago
  61. 7437588 Adding locking to webrtc::voe::Channel to fix race conditions by deadbeef · 9 years ago
  62. c558af8 Removing DtlsIdentityService[Interface] which has been replaced by DtlsIdentityStore[Interface/Impl]. by hbos · 9 years ago
  63. cf7f54d Use RtcpPacket to send RPSI in RtcpSender by sprang · 9 years ago
  64. e2a8be1 Revert of AppRTCDemo: Render each video in a separate SurfaceView (patchset #4 id:120001 of https://codereview.webrtc.org/1257043004/ ) by magjed · 9 years ago
  65. d941b76 Fix distortions of remote stream with odd size dimensions by budnyjj · 9 years ago
  66. 8a2cd3d Revert H.264 HW encoder setting to CBR mode. by Alex Glaznev · 9 years ago
  67. d6b243f Enabling screensharing perf test. by ivica · 9 years ago
  68. 05bfbe4 AppRTCDemo: Render each video in a separate SurfaceView by magjed · 9 years ago
  69. fa30180 Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots. by pthatcher · 9 years ago
  70. cc4ebad Empty dtlsidentityservice.h/cc files added, to be removed once chromium gyp files don't reference it. by Henrik Boström · 9 years ago
  71. 5e56c59 DtlsIdentityStoreInterface added and the implementation is called DtlsIdentityStoreImpl (previously named without the -Impl bit and without an interface). by Henrik Boström · 9 years ago
  72. 0365a27 Use RtcpPacket to send SLI in RtcpSender by sprang · 9 years ago
  73. 4bc66fc Fix data race in AMP. by Michael Graczyk · 9 years ago
  74. 4de6622 Fix a bug in computing audio delay on ios device. Converts seconds to by Jiawei Ou · 9 years ago
  75. 3449faa Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever). by Peter Thatcher · 9 years ago
  76. 4cee419 Separating voice activity flag from audio level in RtpHeaderExtension. by Minyue · 9 years ago
  77. c2ee2c8 Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well. by Peter Thatcher · 9 years ago
  78. eb04d68 Moved project configs to infra/config branch by nodir · 9 years ago
  79. 25c96d0 Add thread checker to StatsCollection. by jbauch · 9 years ago
  80. 2328a94 Add average rtt to CallStatsObserver and an average rtt histogram. by stefan · 9 years ago
  81. 0482dcc Enable HW H.264 decoding on Intel platforms. by Alex Glaznev · 9 years ago
  82. 8381b37 Removed bjornv from OWNERS and added two new owners by peah · 9 years ago
  83. 2e1d8bb Suppress a race in libjingle_peerconnection_unittest by henrik.lundin · 9 years ago
  84. fcf8ece AndroidVideoCapturer: Return frames that have been dropped by magjed · 9 years ago
  85. c937139 Regenerate bind.h using pump.py BUG=webrtc:4690 R=pthatcher@webrtc.org by Fredrik Solenberg · 9 years ago
  86. a873644 Move all the examples from the talk directory into the webrtc examples directory. by Donald E Curtis · 9 years ago
  87. 5b4ce33 DtlsIdentityStoreInterface added. by Henrik Boström · 9 years ago
  88. 0c02264 Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it. by Fredrik Solenberg · 9 years ago
  89. bd10ee8 Tiny cleanups. by Fredrik Solenberg · 9 years ago
  90. 62dae19 Use RtcpPacket to send FIR in RtcpSender by sprang · 9 years ago
  91. ef7228c Selectable number of TL screenshare loopback test. Also contains some tweaks to make a single TL perform better. by sprang · 9 years ago
  92. 907dcfd Increase packet limit in jitter buffer. by sprang · 9 years ago
  93. 37ec733 VideoCapturerAndroid: Check if data is null in onPreviewFrame() by magjed · 9 years ago
  94. 0c85020 Add list of devices with HW H.264 encoder non suitable for WebRTC. by Alex Glaznev · 9 years ago
  95. 8d62971 Fix race condition in EndToEndTest.AssignsTransportSequenceNumbers by Erik Språng · 9 years ago
  96. b19eba3 Fix Turn TCP port issue. by honghaiz · 9 years ago
  97. 867fb52 Add support for transport wide sequence numbers by sprang · 9 years ago
  98. d67a219 Switch to base/logging.h in neteq_impl.cc by Henrik Lundin · 9 years ago
  99. 62cde2c Disabling VP9 perf test by ivica · 9 years ago
  100. 503726c Fix the generation mismatch assertion error. by honghaiz · 9 years ago