Google Git
Sign in
webrtc / src / b6541ca3a1431c7f0cd0fb7ad467aacb2ba3f97f / webrtc / voice_engine
  1. 60730cf Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
  2. 0062a6d Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  3. a7cfa67 Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  4. 54ae4ff Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  5. e1bc6c8 Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing. by turaj@webrtc.org · 11 years ago
  6. 167b6df Fix jitter buffer delay estimate. by turaj@webrtc.org · 11 years ago
  7. 24301a6 Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  8. 9ee75e9 Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency). by henrike@webrtc.org · 11 years ago
  9. f9bdbe3 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  10. 863b536 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
  11. 3054ba6 Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
  12. a750044 Fixes a crash in VoE when unregistering JNI hooks. by henrike@webrtc.org · 11 years ago
  13. 0b72f58 Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
  14. 03f3370 Inject config when creating channels to override the existing one. by turaj@webrtc.org · 11 years ago
  15. 48df381 Fix for making sure that the packet in order checks are done prior to updating the last received packet state. by stefan@webrtc.org · 11 years ago
  16. 886aef0 Fixing broken tests in voe_auto_test extended by tina.legrand@webrtc.org · 11 years ago
  17. 31628aa Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  18. fb648da Protect _transportPtr, which can be accessed by different threads in the case of external transport. This change avoid the potential use-after-free, e.g. the case in the reported bug. by wu@webrtc.org · 11 years ago
  19. 6342066 Fix tsan failures in channel.cc regarding to the volume settings. by wu@webrtc.org · 11 years ago
  20. 675e260 Check the number of playout channels instead of the send channels in StopPlayout() by xians@webrtc.org · 11 years ago
  21. 3555303 Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
  22. 3f9288f Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  23. 6c264cc Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
  24. 2a97317 Fix include of isolate.gypi by kjellander@webrtc.org · 12 years ago
  25. cc92e00 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 12 years ago
  26. 9b6eefc Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket by henrika@webrtc.org · 12 years ago
  27. eb524d9 Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 12 years ago
  28. 63fe8e1 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 12 years ago
  29. f3930e9 Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 12 years ago
  30. bf00740 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 12 years ago
  31. e509f94 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 12 years ago
  32. 82f014a OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 12 years ago
  33. 182d025 Remove include_dirs from voice_engine.gyp. by pbos@webrtc.org · 12 years ago
  34. eda189b Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 12 years ago
  35. 7bb8f02 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 12 years ago
  36. 9080518 Restore severity precondition to logging.h. by andrew@webrtc.org · 12 years ago
  37. 65abb6b Revert 4671 "Enable SetInitialPlayoutDelay on Android." by mflodman@webrtc.org · 12 years ago
  38. 310ac91 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 12 years ago
  39. b295a3f Update SSRC in RtpRtcp for audio channel so that it can have NTP values for further AV sync. by dwkang@webrtc.org · 12 years ago
  40. e141373 Add isolate configuration for Android for all tests. by kjellander@webrtc.org · 12 years ago
  41. 563910b Disables RtpRtcpTest.CanTransmitExtraRtpPacketsWithoutError as it flakily breaks the waterfall. See http://chromegw.corp.google.com/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/99/steps/voe_auto_test/logs/stdio the cl triggering it was a no-change (disabled some other broken tests). by henrike@webrtc.org · 12 years ago
  42. 3365422 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 12 years ago
  43. 286fe0b Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 12 years ago
  44. a0218a8 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 12 years ago
  45. 1a65d6c Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 12 years ago
  46. 822fbd8 Update talk to 50918584. by wu@webrtc.org · 12 years ago
  47. dde7d4c Roll chromium_revision 214260:217707 and gflags 45:84 by fischman@webrtc.org · 12 years ago
  48. 4298f73 Revert 4547 "Isolate GYP target and .isolate files for tests" by kjellander@webrtc.org · 12 years ago
  49. d7a4d23 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 12 years ago
  50. d659143 Disable CanTransmitExtraRtpPacketsWithoutError on Windows. by pbos@webrtc.org · 12 years ago
  51. 6cd9341 Hand over loopback packets to a network thread. by pbos@webrtc.org · 12 years ago
  52. 58d76cb Delete Channels without ChannelManager lock. by pbos@webrtc.org · 12 years ago
  53. 676ff1e Ref-counted rewrite of ChannelManager. by pbos@webrtc.org · 12 years ago
  54. 94aca5c Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer. by phoglund@webrtc.org · 12 years ago
  55. bd69d1b Disabled SsrcPropagatesCorrectly on Linux. by phoglund@webrtc.org · 12 years ago
  56. 12dc1a3 Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 12 years ago
  57. 89c6740 Adds all unittests to android NDK-APK framework. by henrike@webrtc.org · 12 years ago
  58. 09e8c47 Merge r4374 from stable to trunk. by xians@webrtc.org · 12 years ago
  59. 8fff1f0 Merge r4394 from stable to trunk. by xians@webrtc.org · 12 years ago
  60. 2f84afa Merge r4326 from stable to trunk. by xians@webrtc.org · 12 years ago
  61. a3f3014 Default constructor for RtcpAppHandler. by pbos@webrtc.org · 12 years ago
  62. 64e2cbf clean up incomplete revert in r4357 Also revert r4319, will follow up with pbos by tnakamura@webrtc.org · 12 years ago
  63. aa4d96a Revert r4301 by tnakamura@webrtc.org · 12 years ago
  64. 4888fd4 Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered" by elham@webrtc.org · 12 years ago
  65. b7eda43 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on by elham@webrtc.org · 12 years ago
  66. 5b10d8f Fix some voe_auto_test uninitialised-value errors. by pbos@webrtc.org · 12 years ago
  67. 717d147 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 12 years ago
  68. 9de89a6 Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered. by stefan@webrtc.org · 12 years ago
  69. 08933a5 Initialize payload-type frequency in channel.cc. by pbos@webrtc.org · 12 years ago
  70. 66b2e5c Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 12 years ago
  71. d900e8b Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 12 years ago
  72. 0851df8 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. by andrew@webrtc.org · 12 years ago
  73. 6c35e0b Reorganize test targets in WebRTC by kjellander@webrtc.org · 12 years ago
  74. 1064cf0 Fixed Rtp/Rtcp tests by pwestin@webrtc.org · 12 years ago
  75. da71044 Fix size_t to int conversion error on Win64. by andrew@webrtc.org · 12 years ago
  76. 8d80fa8 Fix for STL vector function data not available. by pwestin@webrtc.org · 12 years ago
  77. d30859e Connect ACM with RTP module for audio NACK. by pwestin@webrtc.org · 12 years ago
  78. db24995 Wire up Nack for Voe by pwestin@webrtc.org · 12 years ago
  79. 0a38432 Fix error in mixing test for supported sample rates. by andrew@webrtc.org · 12 years ago
  80. fa64a59 Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. by wu@webrtc.org · 12 years ago
  81. c1eb560 Replace the old resampler with SincResampler in the voice engine signal path. by andrew@webrtc.org · 12 years ago
  82. b35d2e3 Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 12 years ago
  83. 9aca5b3 Remove #pragma once by pbos@webrtc.org · 12 years ago
  84. a5cb98c Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 12 years ago
  85. f791b1c Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. by andrew@webrtc.org · 12 years ago
  86. e46c8d3 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. by turaj@webrtc.org · 12 years ago
  87. 956aa7e Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 12 years ago
  88. 8a025e2 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 12 years ago
  89. 9213521 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 12 years ago
  90. 3be565b Refactoring for typing detection by niklas.enbom@webrtc.org · 12 years ago
  91. ea83c6a Allow voe_cmd_test to select Opus mono (now the default). by andrew@webrtc.org · 12 years ago
  92. 8c845cb Relax VoE's max packet length threshold. by andrew@webrtc.org · 12 years ago
  93. 258f55e Disabled flaky test. by phoglund@webrtc.org · 12 years ago
  94. 28e82bf Replace Resampler with PushResampler in transmit_mixer. by andrew@webrtc.org · 12 years ago
  95. 3423537 Consolidate common_audio into a single target. by andrew@webrtc.org · 12 years ago
  96. 50b2efe Add a wrapper around PushSincResampler and the old Resampler. by andrew@webrtc.org · 12 years ago
  97. 4392d5f Fix for "RTP dynamic payload type 100 is reserved" by henrika@webrtc.org · 12 years ago
  98. 6e788df Remove vim/emacs modelines from .gypi files by pbos@webrtc.org · 12 years ago
  99. 1de0135 Adding playout buffer status to the voe video sync by pwestin@webrtc.org · 12 years ago
  100. 6141e13 WebRtc_Word32 -> int32_t in voice_engine/ by pbos@webrtc.org · 12 years ago