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webrtc / src.git / a826006132b3606b7325befcbffd608df6714f6c
  1. a826006 Add NACK and RPSI packet types to RTCP packet builder. by asapersson@webrtc.org · 11 years ago
  2. 2db9f45 Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size by minyue@webrtc.org · 11 years ago
  3. 1732a59 Add a UIView for rendering a video track. by tkchin@webrtc.org · 11 years ago
  4. 7ca1edb Remove IOKit linkage from iOS builds. by tkchin@webrtc.org · 11 years ago
  5. 40bc777 talk_base: remove lock inversion between MessageQueue and MessageQueueManager. by fischman@webrtc.org · 11 years ago
  6. cb711f77 Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 11 years ago
  7. ebb467f Avoid NACK-list flush error on keyframe packets. by pbos@webrtc.org · 11 years ago
  8. 64339a7 Don't crash if a frame returned from the decoder is too old. by stefan@webrtc.org · 11 years ago
  9. 725e582 Use the new gyp_var_prefix local variable set by gyp instead of the by michaelbai@google.com · 11 years ago
  10. 14abcc7 libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict. by henrike@webrtc.org · 11 years ago
  11. a3b5673 common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16 by bjornv@webrtc.org · 11 years ago
  12. 1e019d1 Fix delivery error-checking missed in r6151. by pbos@webrtc.org · 11 years ago
  13. 57e0602 Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*. by solenberg@webrtc.org · 11 years ago
  14. 60015d2 Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main} by andresp@webrtc.org · 11 years ago
  15. 1b21a57 common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16 by bjornv@webrtc.org · 11 years ago
  16. d83d607 common_audio/signal_processing: Removed macro WEBRTC_SPL_MAX_SEED_USED by bjornv@webrtc.org · 11 years ago
  17. 75718cf * Implement WindowsRealTimeClock::CurrentTimeVal with GetSystemTimeAsFileTime as it supposes to return a POSIX gettimeofday, so that later it can be converted to NTP timee correctly. by wu@webrtc.org · 11 years ago
  18. bf58a75 removed webrtc_base_tests_utils from merge libs as it was breaking some builds. by henrike@webrtc.org · 11 years ago
  19. 508795f Made the presubmit script accept license headers back to 2003 by henrike@webrtc.org · 11 years ago
  20. cfdf420 Rebase webrtc/base 6129:6163 (svn diff -r 6129:6163 http://webrtc.googlecode.com/svn/trunk/talk/base apply diff manually) by henrike@webrtc.org · 11 years ago
  21. 6bfd619 (Auto)update libjingle 67052073-> 67134648 by buildbot@webrtc.org · 11 years ago
  22. 6aeeac9 Fix Windows debug compile of overrides/ logging. by pbos@webrtc.org · 11 years ago
  23. d5da250 Revert "Revert "Audio processing: Feed each processing step its choice by mflodman@webrtc.org · 11 years ago
  24. 024e4d5 Fix Win VideoSendStream::...::ToString() compiles. by pbos@webrtc.org · 11 years ago
  25. 1e92b0a Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 11 years ago
  26. 1aae6bf common_audio: Removes unused macros by bjornv@webrtc.org · 11 years ago
  27. b4e80e0 Re-enable almost all NetEqDecodingTests for Android by henrik.lundin@webrtc.org · 11 years ago
  28. 7cb4752 WebRTCDemo: couldn't run a second time. The reason is voe could register/unregister for each run, but vie would expect initialization only once per process. by braveyao@webrtc.org · 11 years ago
  29. 54231f0 Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log. by wu@webrtc.org · 11 years ago
  30. bb6201a TCP remote socket address should have both server hostname and IP address. by mallinath@webrtc.org · 11 years ago
  31. a150bc9 PeerConnection(android): allow initializing either (or neither) of {Voice,Video}Engine. by fischman@webrtc.org · 11 years ago
  32. ef5a752 (Auto)update libjingle 67043374-> 67044055 by buildbot@webrtc.org · 11 years ago
  33. 3e92468 (Auto)update libjingle 67037200-> 67043374 by buildbot@webrtc.org · 11 years ago
  34. 4f58014 Drop the DataChannel message if it's received when the channel is not open. by jiayl@webrtc.org · 11 years ago
  35. 372701a (Auto)update libjingle 67023528-> 67036361 by buildbot@webrtc.org · 11 years ago
  36. 21299d4 Remove the use of AudioFrame::energy_ from AudioProcessing and VoE. by andrew@webrtc.org · 11 years ago
  37. 688ed69 (Auto)update libjingle 67017551-> 67023528 by buildbot@webrtc.org · 11 years ago
  38. c50bf7c Added namespace rtc to some base classes and functions. It was causing linker error in the FYI bots: http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Android%20Builder%20%28dbg%29/builds/1808/steps/compile/logs/stdio but also, not doing it pollutes the global namespace. by henrike@webrtc.org · 11 years ago
  39. 3147b97 LSan suppressions for libjingle tests (fix) by kjellander@webrtc.org · 11 years ago
  40. 7c0f6e1 LSan suppressions for libjingle tests (more) by kjellander@webrtc.org · 11 years ago
  41. 2c98af7 PeerConnection(Java): auto-WrapCurrentThread() when creating PeerConnectionFactory. by fischman@webrtc.org · 11 years ago
  42. a70dff4 LSan suppressions for libjingle tests. by kjellander@webrtc.org · 11 years ago
  43. 88abf11 Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe. by wu@webrtc.org · 11 years ago
  44. 4e545cc Update webrtcvideoengine2.cc to use DeliveryStatus. by pbos@webrtc.org · 11 years ago
  45. caba2d2 Add DeliveryStatus enum to DeliverPacket(). by pbos@webrtc.org · 11 years ago
  46. 581e217 Fix libjingle to provide a field_trial implementation. by andresp@webrtc.org · 11 years ago
  47. 01edf2e Updating LSan third party suppressions. by kjellander@webrtc.org · 11 years ago
  48. a36ad69 Add webrtc field trials API. by andresp@webrtc.org · 11 years ago
  49. 9f27735 Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
  50. f383a1b Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
  51. 2fa1701 Re-enable NetEqExternalDecoderTest for Android by henrik.lundin@webrtc.org · 11 years ago
  52. bf93fb3 Re-enable NetEQ DecoderDatabase test for Android by henrik.lundin@webrtc.org · 11 years ago
  53. b1a66d1 Revert "Audio processing: Feed each processing step its choice of int or float data" by mflodman@webrtc.org · 11 years ago
  54. db60434 Re-enable the BitrateEstimatorTest cases for the Call API. by solenberg@webrtc.org · 11 years ago
  55. 5c49c64 Remove all use of AudioFrame::energy_ from AudioCodingModule by henrik.lundin@webrtc.org · 11 years ago
  56. 06c1d6f VoEVolumeTest: Adds error return tests. by bjornv@webrtc.org · 11 years ago
  57. 934a265 Audio processing: Feed each processing step its choice of int or float data by kwiberg@webrtc.org · 11 years ago
  58. 3d5cb33 Remove WEBRTC_TRACE use in video_capture/ by pbos@webrtc.org · 11 years ago
  59. 4e2806d Remove WEBRTC_TRACE uses in video_engine/ by pbos@webrtc.org · 11 years ago
  60. 98c76a1 Make vie/voe_auto_test accept non-supported flags without error. by kjellander@webrtc.org · 11 years ago
  61. cd846dd (Auto)update libjingle 66924241-> 66927231 by buildbot@webrtc.org · 11 years ago
  62. da510c5 (Auto)update libjingle 66923202-> 66924241 by buildbot@webrtc.org · 11 years ago
  63. d8af5b5 Deallocate the result of mach_host_self() when done with it, fixing a port leak. by fischman@webrtc.org · 11 years ago
  64. c14f521 (Auto)update libjingle 66887616-> 66900106 by buildbot@webrtc.org · 11 years ago
  65. f048872 Adds a modified copy of talk/base to webrtc/base. It is the first step in by henrike@webrtc.org · 11 years ago
  66. 3e01e0b (Auto)update libjingle 66867790-> 66887616 by buildbot@webrtc.org · 11 years ago
  67. c156174 Suppressing all tests for WebRtcVideoEngine2 for Win DrMemory Full. by henrike@webrtc.org · 11 years ago
  68. 8d63d0e Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255 by bjornv@webrtc.org · 11 years ago
  69. 93ec9c5 Revert "FieldTrial implementation for webrtc." (rev 6089) by andresp@webrtc.org · 11 years ago
  70. e41dbee Reduced kMaxSampleDiffMs (limit to 22fps). by asapersson@webrtc.org · 11 years ago
  71. 023b101 Move gflags usage to video_loopback. by pbos@webrtc.org · 11 years ago
  72. b5a22b1 Revert r6110 and r6109. by pbos@webrtc.org · 11 years ago
  73. c3e8abd Deleting all NetEq3 files by henrik.lundin@webrtc.org · 11 years ago
  74. 4d363ae The webrtc::AudioFrame struct contains a variable energy_. Since the energy isn't always calculated when the frame is created, this change makes the CalculateEnergy method in Audio Conference Mixer always calculate the energy. by henrik.lundin@webrtc.org · 11 years ago
  75. e9a604a Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..." by perkj@webrtc.org · 11 years ago
  76. 3a58259 Deleting all ACM1 files by henrik.lundin@webrtc.org · 11 years ago
  77. 46e636a Fix failing test introduced with r6111. by stefan@webrtc.org · 11 years ago
  78. eaf2bd9 (Auto)update libjingle 66813165-> 66836233 by buildbot@webrtc.org · 11 years ago
  79. d37bcfa Changed enums to less generic names. by mallinath@webrtc.org · 11 years ago
  80. 72885d1 Fixes log spam introduced with r6041. by stefan@webrtc.org · 11 years ago
  81. 17911dc (Auto)update libjingle 66798415-> 66813165 by buildbot@webrtc.org · 11 years ago
  82. 0df2ea0 Rollback of r6108 by henrike@webrtc.org · 11 years ago
  83. a7f70a4 Initialize bitrates in ValidateCodecFormat. by pbos@webrtc.org · 11 years ago
  84. 2c7d1b3 Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base. by henrike@webrtc.org · 11 years ago
  85. f3a5e6a Suppression for WebRtcVideoChannel2BaseTest.SetSendSsrc. by henrike@webrtc.org · 11 years ago
  86. d886e4a Suppression for test failing on dr memory (in waterfall). by henrike@webrtc.org · 11 years ago
  87. d266a20 Initial wiring of new webrtc API in libjingle. by pbos@webrtc.org · 11 years ago
  88. 6b02eea Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
  89. 1cec395 Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
  90. 924e81f Echo cancellation functions docs: Follow style guide w.r.t. placement of * by kwiberg@webrtc.org · 11 years ago
  91. 66021e0 Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
  92. b9863ce One of the NetEq methods needs to be virtual. by turaj@webrtc.org · 11 years ago
  93. e14ffaa Update DEPS to pull r6096 changes to third_party/openmax_dl/dl/dl.gyp by turaj@webrtc.org · 11 years ago
  94. 0f2a22b Removed sending metrics from PeerConnection about IPv4 and IPv6. by mallinath@webrtc.org · 11 years ago
  95. 8a54844 (Auto)update libjingle 66624678-> 66643715 by buildbot@webrtc.org · 11 years ago
  96. 17bf9a2 Modifying neteq.gyp by turaj@webrtc.org · 11 years ago
  97. 1cd14a4 (Auto)update libjingle 66556498-> 66624678 by buildbot@webrtc.org · 11 years ago
  98. 3b76627 Removes parts of the webrtc::VoEHardware sub API (relanding) by henrika@webrtc.org · 11 years ago
  99. 3106b70 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." by henrika@webrtc.org · 11 years ago
  100. 9de3d84 Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago