summaryrefslogtreecommitdiff
path: root/modules
AgeCommit message (Expand)Author
2013-04-05Resolves TSan v2 reports data races in voe_auto_test.henrika@webrtc.org
2013-04-05Permit arbitrary payload names for kVideoCodecGeneric.pbos@webrtc.org
2013-04-04Adds event traces and counters for WebRTC receive side.edjee@google.com
2013-04-04TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGChenrika@webrtc.org
2013-04-02Webrtc_Word32 => int32_t in video_coding/main/pbos@webrtc.org
2013-04-02Revert of r3747.henrike@webrtc.org
2013-04-02Two more sleep calls converted to use SleepMs().hta@webrtc.org
2013-04-02Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizerhenrika@webrtc.org
2013-04-01Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higherjustinlin@chromium.org
2013-04-01For VGA (640x360), currently 1 thread is used. This change increases it to 2...fbarchard@google.com
2013-03-28Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549marpan@webrtc.org
2013-03-28Removed CPU APIs from VoEHardware. Code is now only used by test applications.henrike@webrtc.org
2013-03-27G722-stereo has been missing when creating AudioDecoder.turaj@webrtc.org
2013-03-27NetEq4 fails if the first packets inserted in are out-of-band DTMFs.turaj@webrtc.org
2013-03-27Fix flakiness in network up/down event tests when running under memcheck.stefan@webrtc.org
2013-03-27WebRTCDemo: remove unnecessary stop & start during orientation change which i...fischman@webrtc.org
2013-03-27Add interface to signal a network down event.stefan@webrtc.org
2013-03-26Fix potential buffer overrun when checking if a packet is RTCP. Also makes va...solenberg@webrtc.org
2013-03-25Restart Android capture after orientation change.fischman@webrtc.org
2013-03-25Add some VoE and AudioProcessing mocks.andrew@webrtc.org
2013-03-22Introduced pause and resume to the pacerpwestin@webrtc.org
2013-03-21WebRtc_Word -> stdint in audio_coding/g711/pbos@webrtc.org
2013-03-21Remove incorrect asserts.stefan@webrtc.org
2013-03-21WebRtc_Word -> stdint in audio_coding/cng/pbos@webrtc.org
2013-03-20Fix -Wstring-conversion warnings.wu@webrtc.org
2013-03-20Thread safety issue fix in incoming_video_stream.cc. See issue 1465.vikasmarwaha@webrtc.org
2013-03-19Account for header inside I420Encoder::InitEncode.pbos@webrtc.org
2013-03-19Follow-up fix for r3681.stefan@webrtc.org
2013-03-19Fixed initialization of SPL in echo_control_mobile.kma@webrtc.org
2013-03-18Fix framerate sent to account for actually sent frames.stefan@webrtc.org
2013-03-18Change VCM interface to take target bitrate in bits per second.stefan@webrtc.org
2013-03-18Generic video-codec support.pbos@webrtc.org
2013-03-18Revert the deletion of test_api_nack.cc in r3674.stefan@webrtc.org
2013-03-18Truncated delay quality to avoid negative return valuesbjornv@webrtc.org
2013-03-15Adding RTX on sourcemikhal@webrtc.org
2013-03-15Adding Opus frame length testtina.legrand@webrtc.org
2013-03-14Fixed a crash issue in NSX module.kma@webrtc.org
2013-03-13Revert r3667 and r3665pwestin@webrtc.org
2013-03-13Removed the engine API:s related to transport such as SetSendDestination, the...pwestin@webrtc.org
2013-03-13Refactor webrtc specific Event implementation to an EventFactory.stefan@webrtc.org
2013-03-12Remove DTMF detection. Talk team has been in the loop and there is no need forturaj@webrtc.org
2013-03-12Change intrinsic code in isac fix to let it pass chrome clang compiler.kma@webrtc.org
2013-03-12Removed redundant VP8 width/height and made sure the generic width/height is ...stefan@webrtc.org
2013-03-12Revert "Internal clean up: removing unused include line."dwkang@webrtc.org
2013-03-12Internal clean up: removing unused include line.dwkang@webrtc.org
2013-03-12Fixed issue 1497 in iSAC fixed point.kma@webrtc.org
2013-03-09Optimized EstCodeLpcCoef() for iSAC with intrinsics in Android-Neon platform.kma@webrtc.org
2013-03-08Splitting out video_coding_test executable again.kjellander@webrtc.org
2013-03-07Fixed an assembly code error in AECM for ARMv7.kma@webrtc.org
2013-03-07Disable frame dropper for screenshare mode.stefan@webrtc.org
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-22 20:54:13 +0000