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webrtc / src.git / 09c77b95bb62566be64da662f0b3b6a838ec6553 / webrtc
  1. 09c77b9 Add decoder-timing stats to VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  2. c5558b7 Remove AudioCodingModule's dependency on the Module interface by henrik.lundin@webrtc.org · 10 years ago
  3. af82f75 Let Add10MsData method do the encoding work as well by henrik.lundin@webrtc.org · 10 years ago
  4. 8d350d4 Add new AcmGenericCodecTest and verify output from Encode function by henrik.lundin@webrtc.org · 10 years ago
  5. 1eda4e3 Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call" by henrik.lundin@webrtc.org · 10 years ago
  6. 0a3ff79 New AudioTrack implementation now works on pre-Lollipop devices. by henrika@webrtc.org · 10 years ago
  7. d4dfba8 iSAC Decode: Prevent Memcheck from complaining about uninitialized value by kwiberg@webrtc.org · 10 years ago
  8. 87a592d Fix dependencies of media_file module and move gypi into the right dir to by andresp@webrtc.org · 10 years ago
  9. 49096de DCHECK send DataCountersUpdated for valid SSRCs. by pbos@webrtc.org · 10 years ago
  10. 903182b Revert r8476 "Set decoder output frequency in AudioDecoder::Decode call" by henrik.lundin@webrtc.org · 10 years ago
  11. b9c18d5 Set decoder output frequency in AudioDecoder::Decode call by henrik.lundin@webrtc.org · 10 years ago
  12. f88791d AudioEncoderCng: CHECK that encode calls don't fail by jmarusic@webrtc.org · 10 years ago
  13. 5e3fea1 Fixing WebRTC engine demo JNI symbol export. by phoglund@webrtc.org · 10 years ago
  14. db8e605 Break out BWE test models to separate files by sprang@webrtc.org · 10 years ago
  15. ccd7c7c Remove more unused code in ACM by henrik.lundin@webrtc.org · 10 years ago
  16. 13ca5f6 AudioEncoderOpus: CHECK that encode call doesn't fail by jmarusic@webrtc.org · 10 years ago
  17. d324546 Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ : by pkasting@chromium.org · 10 years ago
  18. 7227391 Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530) by kjellander@webrtc.org · 10 years ago
  19. 829a6f4 Merge ACMGenericCodec and ACMGenericCodecWrapper by henrik.lundin@webrtc.org · 10 years ago
  20. f3a306b g722: Enhanced documentation. Added CHECK. by jmarusic@webrtc.org · 10 years ago
  21. 2acec4c Enhanced documentation. Replaced DCHECK with CHECK. by jmarusic@webrtc.org · 10 years ago
  22. 962c624 Refactoring WebRTC Java/JNI audio track in C++ and Java. by henrika@webrtc.org · 10 years ago
  23. 8278c07 Enable NACK under SendsAndReceivesH264. by pbos@webrtc.org · 10 years ago
  24. fa58745 Delete all codec-specific subclasses of ACMGenericCodec by henrik.lundin@webrtc.org · 10 years ago
  25. 2a5cfc2 Replaced unnecessary check with an explicit CHECK. by jmarusic@webrtc.org · 10 years ago
  26. 343096a Fix incorrect rtx config in full_stack tests. by sprang@webrtc.org · 10 years ago
  27. 1467421 Fix for flaky test: VideoSendStreamTest.RtcpSenderReportContainsMediaBytesSent. by asapersson@webrtc.org · 10 years ago
  28. 50e2816 Move SetTargetSendBitrates logic from default module to payload router. by mflodman@webrtc.org · 10 years ago
  29. a43fce6 Add functions rtc::AtomicOps::Load and rtc::RefCountedObject::HasOneRef by magjed@webrtc.org · 10 years ago
  30. 2af3057 Revert "When clearing the priority message queue, don't copy an item to itself." by decurtis@webrtc.org · 10 years ago
  31. 2bffc3c When clearing the priority message queue, don't copy an item to itself. by decurtis@webrtc.org · 10 years ago
  32. 7ac374a Fix shutdown race for ViEEncoder when there is a frame in the encoder. by mflodman@webrtc.org · 10 years ago
  33. dc77d74 Disable FullStackTest.ForemanCifPlr5 temporarily while investigating flakiness. by sprang@webrtc.org · 10 years ago
  34. 804eb46 Change default from GICE to ICE5245 for SDP offers by jlmiller@webrtc.org · 10 years ago
  35. d3d3baa Copy SetThreadName from webrtc/base/thread.cc into thread_win.cc by tommi@webrtc.org · 10 years ago
  36. 661af50 Small Beamformer optimization by aluebs@webrtc.org · 10 years ago
  37. e07710c Make SendCodec() lock-free. by tommi@webrtc.org · 10 years ago
  38. be29b3b I420VideoFrame: Remove functions set_width, set_height, and ResetSize by magjed@webrtc.org · 10 years ago
  39. be96bfb Re-land "Switch to using AudioEncoderIsac instead of ACMISAC" by kwiberg@webrtc.org · 10 years ago
  40. 2877552 Fix a problem with reading uninitialized memory in ACM by henrik.lundin@webrtc.org · 10 years ago
  41. 1d0fa5d Add RtcpPacketTypeCounter stats to new API. by pbos@webrtc.org · 10 years ago
  42. 5060412 Method WebRtc_g722_encode that is eventually called always returns non-negative integer (internal counter) by jmarusic@webrtc.org · 10 years ago
  43. 47d657b Remove Set/Get sending status from the default RTP module. by mflodman@webrtc.org · 10 years ago
  44. 32c784c ViEExternalRendererImpl: Remove dependency to webrtc::VideoFrame by magjed@webrtc.org · 10 years ago
  45. 30540fe Initialize RTPVideoHeader fields to correctly set simulcastIdx for non VP8 codecs. by glaznev@webrtc.org · 10 years ago
  46. 9dfe7aa Fix WebRTC IP leaks. by guoweis@webrtc.org · 10 years ago
  47. 931e0cf Fix WebRTC IP leaks. by guoweis@webrtc.org · 10 years ago
  48. f358aea Fix WebRTC IP leaks. by guoweis@webrtc.org · 10 years ago
  49. 88828e7 Fix I420VideoFrame unittests by magjed@webrtc.org · 10 years ago
  50. c0bd7be Adding two new stats to VoiceReceiverInfo by minyue@webrtc.org · 10 years ago
  51. b255865 The PCM codecs can never fail, so we don't need to check the return value by jmarusic@webrtc.org · 10 years ago
  52. 78619e2 Revert of r8378 "Switch to using AudioEncoderIsac instead of ACMISAC" by henrik.lundin@webrtc.org · 10 years ago
  53. 635838bd Re-implementing AcmOpusTest as AcmGenericCodecOpusTest by henrik.lundin@webrtc.org · 10 years ago
  54. f68e186 Remove EnableMirroring and MirrorRenderStream by magjed@webrtc.org · 10 years ago
  55. 131bea8 Offline screenshare quality test, plus loopback. by sprang@webrtc.org · 10 years ago
  56. 0521127 AudioEncoder: Rename virtual accessors to CamelCase by kwiberg@webrtc.org · 10 years ago
  57. cc483b7 Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737) by kjellander@webrtc.org · 10 years ago
  58. 7d721ee Adding speech_expand_rate to NetEQ Network Statistics. by minyue@webrtc.org · 10 years ago
  59. 97aaf68 Bump to version 42. by jansson@webrtc.org · 10 years ago
  60. bfa3c72 Don't call g_thread_init on glib >=2.31.0 by decurtis@webrtc.org · 10 years ago
  61. e9facf8 Add range checks in a variety of places where the values will subsequently be by pkasting@chromium.org · 10 years ago
  62. 27669f3 Apply good settings to Beamformer by aluebs@webrtc.org · 10 years ago
  63. b08f404 Fix issue 4061. by guoweis@webrtc.org · 10 years ago
  64. 0abc601 Remove SetCaptureDelay from the RTP module. by mflodman@webrtc.org · 10 years ago
  65. 7663684 Implement the Nada rmcat proposal within the simulation framework. by stefan@webrtc.org · 10 years ago
  66. 71b35a4 iLBC: Use uint8_t[] for byte arrays by jmarusic@webrtc.org · 10 years ago
  67. 640313c WebRtcVideoCapturer: Remove dead code |OnIncomingCapturedEncodedFrame| by magjed@webrtc.org · 10 years ago
  68. 7a91acb ViECapturer: Remove unimplemented function declaration |DeliverCodedFrame| by magjed@webrtc.org · 10 years ago
  69. a28a91d Fix data race for RTCPReceiver stats callback. by pbos@webrtc.org · 10 years ago
  70. 959dac7 VideoCaptureImpl: Remove unused member variable |_capture_encoded_frame| by magjed@webrtc.org · 10 years ago
  71. 4dd40d6 Signal threads for faster receiver destruction. by pbos@webrtc.org · 10 years ago
  72. 0a7d4ee Remove usage of BitrateController in VoiceEngine. by mflodman@webrtc.org · 10 years ago
  73. f9b5c1b Removing CELT. by minyue@webrtc.org · 10 years ago
  74. 2c1bcf2 Adding decoded_fec_rate to NetEQ Network Statistics. by minyue@webrtc.org · 10 years ago
  75. 290cb56 Remove TimeToSendPacket and TimeToSendPadding from the default module. by mflodman@webrtc.org · 10 years ago
  76. 86196c4 Setup encoders inexpensively before first frame. by pbos@webrtc.org · 10 years ago
  77. 34509d9 Fix an issue with comfort noise in ACMGenericCodecWrapper by henrik.lundin@webrtc.org · 10 years ago
  78. e9f0f59 Enable bitrate probing by default in PacedSender. by stefan@webrtc.org · 10 years ago
  79. fbc347f Re-land r8342 "Switch to using AudioEncoderIsac instead of ACMISAC"" by henrik.lundin@webrtc.org · 10 years ago
  80. ce22f13 GN: Changes for vp9, opus and direct trace by kjellander@webrtc.org · 10 years ago
  81. e35fa96 Move isacfix.gypi and isac.gypi by kjellander@webrtc.org · 10 years ago
  82. 0200f70 Set webrtc_rtp category to be disabled by default. by sprang@webrtc.org · 10 years ago
  83. 14b0279 Break out code from bloated files in the BWE simulator. by stefan@webrtc.org · 10 years ago
  84. 0f7f161 Add audio_coding module OWNERS file. by kjellander@webrtc.org · 10 years ago
  85. 4dc0003 Revert r8342 "Switch to using AudioEncoderIsac instead of ACMISAC" by henrik.lundin@webrtc.org · 10 years ago
  86. 30142bb Add arraysize to overrides to avoid macros redefinitions in Chromium by aluebs@webrtc.org · 10 years ago
  87. d3b453b Remove the incremental IP address behavior from virtualsocketserver by guoweis@webrtc.org · 10 years ago
  88. 92a19bc Simplify mask calculation by aluebs@webrtc.org · 10 years ago
  89. 56cb0ea Add support for bi-directional simulations by having both an uplink and a downlink. by stefan@webrtc.org · 10 years ago
  90. d5ce2e6 Remove EventWrapper::Reset(). by pbos@webrtc.org · 10 years ago
  91. 5a7dc39 This is a code clean up. No functional change intended. by guoweis@webrtc.org · 10 years ago
  92. a8cc344 Allowing RED decoding for Opus. by minyue@webrtc.org · 10 years ago
  93. 8db5854 Fix potential flakiness in voe_auto_test. by solenberg@webrtc.org · 10 years ago
  94. 2adf4c4 Re-enable BWE tests using baseline files. by solenberg@webrtc.org · 10 years ago
  95. 58f6f01 WebRTC now compiles for enable_android_opensl=1. by henrika@webrtc.org · 10 years ago
  96. ba97ea6 audio_coding/codec/ilbc: Removed usage of macro WEBRTC_SPL_MUL_16_16 by bjornv@webrtc.org · 10 years ago
  97. 2bd299a Remove call to RtpRtcp::RegisterSendPayload for the default RTP module. by mflodman@webrtc.org · 10 years ago
  98. 40367f9 Remove default video encoders for new video API. by pbos@webrtc.org · 10 years ago
  99. 94eb9a6 Whitespace change to test gsubtreed. by kjellander@webrtc.org · 10 years ago
  100. bb1219e Add a unit test for callbacks with empty frames and fix bug in code by henrik.lundin@webrtc.org · 10 years ago