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gerrit-public.fairphone.software / platform / external / webrtc / 14665ff7d4024d07e58622f498b23fd980001871
  1. 14665ff Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 9 years ago
  2. 792f1a1 Break out allocation from BitrateController into a BitrateAllocator. by stefan@webrtc.org · 9 years ago
  3. 61c22ac Eliminate AcmGenericCodec::Add10MsData by henrik.lundin@webrtc.org · 9 years ago
  4. f82109c Initialize memory in I420VideoFrame unittest by magjed@webrtc.org · 9 years ago
  5. 487afc7 Always define RTC_NOTREACHED, not just in non-chromium builds by magjed@webrtc.org · 9 years ago
  6. 9cd7c26 Rename NOTREACHED to RTC_NOTREACHED to avoid name conflict with Chromium by magjed@webrtc.org · 9 years ago
  7. 6dab6d7 Let Chromium declare the mips_dsp_rev build variable. by kjellander@webrtc.org · 9 years ago
  8. 1d25c87 Reland r8577 "Collapse AudioEncoderDecoderIsacRed into ..." by henrik.lundin@webrtc.org · 9 years ago
  9. 058b1f1 Remove GetReceiveBandwidthEstimatorStats. by pbos@webrtc.org · 9 years ago
  10. 7572d85 rtc_unittests on Android by kjellander@webrtc.org · 9 years ago
  11. c98f6f3 PRESUBMIT: Exclude overrides paths from source above GYP check. by kjellander@webrtc.org · 9 years ago
  12. fc2f146 Revert "Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%." by guoweis@webrtc.org · 9 years ago
  13. 7bea1ff Expose negotiated ciphers through stats API. by pthatcher@webrtc.org · 9 years ago
  14. be77872 Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background." by jiayl@webrtc.org · 9 years ago
  15. bbbdeed Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%. by guoweis@webrtc.org · 9 years ago
  16. 369f682 Create a in-memory DTLS identity store that keeps a free identity generated in the background. by jiayl@webrtc.org · 9 years ago
  17. c8895aa Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame by magjed@webrtc.org · 9 years ago
  18. 8ad9660 Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background." by jiayl@webrtc.org · 9 years ago
  19. bcef431 Revert r8577 "Collapse AudioEncoderDecoderIsacRed into ..." by henrik.lundin@webrtc.org · 9 years ago
  20. 1fc28f2 Collapse AudioEncoderDecoderIsacRed into AudioEncoderDecoderIsac by henrik.lundin@webrtc.org · 9 years ago
  21. df512cc Create a in-memory DTLS identity store that keeps a free identity generated in the background. by jiayl@webrtc.org · 9 years ago
  22. 982cd2a Filter receiver-side DataCountersUpdated on SSRC. by pbos@webrtc.org · 9 years ago
  23. b144b4b Fixed bug in SendTimeHistory, where deleting packets via the getter by sprang@webrtc.org · 9 years ago
  24. 0561716 Adding Opus DTX support in ACM. by minyue@webrtc.org · 9 years ago
  25. a1c9803 Fix crash in setPictureSize on Galaxy Nexus. by perkj@webrtc.org · 9 years ago
  26. be00e3c Make sure VideoFrameFactory handles rotated frames when scaling. by perkj@webrtc.org · 9 years ago
  27. 9e5f941 Remove webrtc/system_wrappers/interface/scoped_ptr.h by kwiberg@webrtc.org · 9 years ago
  28. 1f914ec Remove suppression for WebRtcVideoFrameTest::TestInit by perkj@webrtc.org · 9 years ago
  29. db93b68 Removing NetEq's direct dependencies on Opus headers. by minyue@webrtc.org · 9 years ago
  30. cb04aa4 WebRtcVideoFrameTest: Initialize memory to fix DrMemory error by magjed@webrtc.org · 9 years ago
  31. 909f494 Roll chromium_revision 2c3ffb2..e144d30 (317530:318658) by kjellander@webrtc.org · 9 years ago
  32. 1d82813 Reland "Fix CVO in androidvideocapturer". by perkj@webrtc.org · 9 years ago
  33. c9ce07e Add Config option to enable 48kHz support in AudioProcessing by aluebs@webrtc.org · 9 years ago
  34. 0482d01 Implement TraceCallback in a nested class of WebRtcVideoEngine. by tommi@webrtc.org · 9 years ago
  35. 97ed2a4 I420VideoFrame: Remove function ResetSize by magjed@webrtc.org · 9 years ago
  36. 43f4a47 Add more Android peer connection client unit tests: by glaznev@webrtc.org · 9 years ago
  37. 976c0f3 audio_processing/aec: NEON code should not be invoked if it is detectable, but is not NEON by bjornv@webrtc.org · 9 years ago
  38. 48ac226 Add support for writing h264 decoder input to file and parsing interleaved length/packet RTP dumps. by stefan@webrtc.org · 9 years ago
  39. 3fe17d1 Adjust a few thresholds for VP9 tests. by marpan@webrtc.org · 9 years ago
  40. fd33293 I420VideoFrame: Remove functions set_width and set_height by magjed@webrtc.org · 9 years ago
  41. f1f0d9a Remove WebRtcVideoEngine::SetVoiceEngine. by pbos@webrtc.org · 9 years ago
  42. e8f50df Remove avi recorder and corresponding enable_video flags. by andresp@webrtc.org · 9 years ago
  43. f56c162 Remove AudioCodingModule::Process() by henrik.lundin@webrtc.org · 9 years ago
  44. 25dd1db Fixed bug in test frame generator, causing incorrect reuse of frame by sprang@webrtc.org · 9 years ago
  45. 60f9d6f Revert "Add default implementation to VideoSourceInterface." by perkj@webrtc.org · 9 years ago
  46. afa6d16 Add a ToString() method to StatsReport::Value. by tommi@webrtc.org · 9 years ago
  47. 50b2295 cricket::VideoFrameFactory: Don't overwrite frames in use by magjed@webrtc.org · 9 years ago
  48. 24485eb Remove last pieces of libjingle_unittest by kjellander@webrtc.org · 9 years ago
  49. 5cd6828 Remove stale isolate files. by kjellander@webrtc.org · 9 years ago
  50. f35e4bc Introduce a send time history class, keeping track of packet send times. by sprang@webrtc.org · 9 years ago
  51. 59ae5ff Filter logic for ip leak misses ::ffff:0.0.0.0 by guoweis@webrtc.org · 9 years ago
  52. 2f6ae0d audio_coding/codec/ilbc: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT by bjornv@webrtc.org · 9 years ago
  53. e1b84a0 Fix a race reported by tsan. by tommi@webrtc.org · 9 years ago
  54. d68fa65 Improve cleaning for Android demo applications by kjellander@webrtc.org · 9 years ago
  55. f7bb6e7 Use new API from BoringSSL to get RFC name of cipher. by pthatcher@webrtc.org · 9 years ago
  56. d312505 Test to try to track down the alignment problem on Mac 10.9. by tommi@webrtc.org · 9 years ago
  57. 73acc15 Revert 8538 "Reland "Fix CVO in androidvideocapturer.""" by aluebs@webrtc.org · 9 years ago
  58. 3a93e33 Reland "Fix CVO in androidvideocapturer."" by perkj@webrtc.org · 9 years ago
  59. b8bcf8c Revert "Fix CVO in androidvideocapturer." by perkj@webrtc.org · 9 years ago
  60. 02ed57b Fix CVO in androidvideocapturer. by perkj@webrtc.org · 9 years ago
  61. 41d8fda VideoCapturerAndroid allocates direct buffers so that the frame buffers can be used in C++ without a copy. However byte[] array = ByteBuffer.array() seems to point to the beginning of the underlaying buffer and that is what the camera fills. But it turns out that ByteBuffer.arrayOffset() returns an offset and it seems like the pointer returned by jni->GetDirectBufferAddress(j_frame). This cl reverts back to pass the byte[] to c++ and use jni->GetByteArrayElements to get the address of the buffer. by perkj@webrtc.org · 9 years ago
  62. 07dcf60 Revert 8532 "Ensure only temporary IPv6 address is selected as t..." by aluebs@webrtc.org · 9 years ago
  63. 21ad375 Ensure we set the right attrib for correct shader by guoweis@webrtc.org · 9 years ago
  64. 385a7ce Ensure only temporary IPv6 address is selected as the best IP. by guoweis@webrtc.org · 9 years ago
  65. fbef5c6 Remove lock in ViEFrameProviderBase::IsFrameCallbackRegistered. by tommi@webrtc.org · 9 years ago
  66. 7400e0b Revert "I420VideoFrame: Remove functions set_width, set_height, and ResetSize" by magjed@webrtc.org · 9 years ago
  67. 4b3618c Remove TraceImpl logging thread. by pbos@webrtc.org · 9 years ago
  68. 6c2e506 Workaround Mac align bug for observer_ and crit_. by pbos@webrtc.org · 9 years ago
  69. 3985f01 ProcessThread improvements. by tommi@webrtc.org · 9 years ago
  70. f296859 PeerConnectionClient.createPeerConnectionClient was calling new PeerConnectionParameters and PeerConnectionClient.createPeerConnectionFactory, .createPeerConnection with invalid arguments. by hbos@webrtc.org · 9 years ago
  71. c68e0c9 Fix cpplint warning in the previous cl to peerconnection client example. by braveyao@webrtc.org · 9 years ago
  72. abbdd52 AudioEncoder: documentation fix by jmarusic@webrtc.org · 9 years ago
  73. ea89495 Remove {Is,Set}BlackOutput from VideoAdapter. by pbos@webrtc.org · 9 years ago
  74. 3aca0b0 Add 48kHz support to Beamformer by aluebs@webrtc.org · 9 years ago
  75. 9650ab4 Fix case sensitivity of AppRTCDemo include dirs by tkchin@webrtc.org · 9 years ago
  76. 2a72c65 Keep feedback params in SetDefaultEncoderConfig. by pbos@webrtc.org · 9 years ago
  77. b1f0de3 AudioEncoder: change Encode and EncodeInternal return type to void by jmarusic@webrtc.org · 9 years ago
  78. 00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 9 years ago
  79. ac2d27d Fix style violations in common_types.h and config.h by kwiberg@webrtc.org · 9 years ago
  80. 891d483 Wire up target_media_bitrate in VideoSendStream. by pbos@webrtc.org · 9 years ago
  81. 9dd0ebc Remove the default RTP module. by mflodman@webrtc.org · 9 years ago
  82. 3e6e271 Implement CpuOveruseMetrics as callbacks. by pbos@webrtc.org · 9 years ago
  83. 60f295f Remove lsan suppression.txt by kjellander@webrtc.org · 9 years ago
  84. e723728 Add p2p.gyp to rtc_base presubmit check exclusion. by kjellander@webrtc.org · 9 years ago
  85. 9a4410e Implement adaptation stats in WebRtcVideoEngine2. by pbos@webrtc.org · 9 years ago
  86. 38d9cc5 Add back return statement after FATAL() by henrik.lundin@webrtc.org · 9 years ago
  87. b5e60b6 Remove non necessary check from WebSocket send function. by glaznev@webrtc.org · 9 years ago
  88. f09e7b8 WebRtcVideoFrame: DCHECK exclusive ownership for non-const pixel access by magjed@webrtc.org · 9 years ago
  89. 6c66163 Fix TestScaler PSNR tests by magjed@webrtc.org · 9 years ago
  90. 96abda0 Removing FEC functionality from the default RTP module. by mflodman@webrtc.org · 9 years ago
  91. 9b969e1 AudioEncoderCopyRed: CHECK that encode call doesn't fail by jmarusic@webrtc.org · 9 years ago
  92. 749c602 Moved gypi to avoid presubmit warning about '..' when touching the files. by andresp@webrtc.org · 9 years ago
  93. 5c928eb Let first packet through to avoid getting key frame requests (and no nacks) for EndToEndTest.ReceivedFecPacketsNotNacked. by asapersson@webrtc.org · 9 years ago
  94. 09c77b9 Add decoder-timing stats to VideoReceiveStream. by pbos@webrtc.org · 9 years ago
  95. c5558b7 Remove AudioCodingModule's dependency on the Module interface by henrik.lundin@webrtc.org · 9 years ago
  96. af82f75 Let Add10MsData method do the encoding work as well by henrik.lundin@webrtc.org · 9 years ago
  97. 4aef5fe Add thread checks to the CaptureManager. by hbos@webrtc.org · 9 years ago
  98. 8d350d4 Add new AcmGenericCodecTest and verify output from Encode function by henrik.lundin@webrtc.org · 9 years ago
  99. 1eda4e3 Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call" by henrik.lundin@webrtc.org · 9 years ago
  100. 1e64263 Thread-safe ChannelManager.GetSupportedFormats, used by VideoSource by hbos@webrtc.org · 9 years ago