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AgeCommit message (Expand)Author
2014-11-04Fix problem with late packets in NetEqhenrik.lundin@webrtc.org
2014-11-04Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16kwiberg@webrtc.org
2014-11-04Remove the useless dummy state parameter to WebRtcG711_*kwiberg@webrtc.org
2014-11-04Remove the codec_type_ member from AudioDecoderkwiberg@webrtc.org
2014-11-04Improving error message from neteq_rtpplayhenrik.lundin@webrtc.org
2014-11-03Roll chromium_revision: 28d1981..d3db2ffmarpan@webrtc.org
2014-11-03Add Opus support to neteq_rtpplayhenrik.lundin@webrtc.org
2014-10-31Update all .isolate files for the new format.kjellander@webrtc.org
2014-10-31Build fix for MIPS32R6.andrew@webrtc.org
2014-10-29AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacketkwiberg@webrtc.org
2014-10-29Make an AudioEncoder subclass for Opuskwiberg@webrtc.org
2014-10-28audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>"bjornv@webrtc.org
2014-10-28common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16bjornv@webrtc.org
2014-10-28Use neteq_unittest_tools in audio_decoder_unittestshenrik.lundin@webrtc.org
2014-10-27Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFilehenrik.lundin@webrtc.org
2014-10-27isacfix: Refactor big-endian reading and writingkwiberg@webrtc.org
2014-10-23Add macros and APIs for webrtc histograms.asapersson@webrtc.org
2014-10-21Implement AudioEncoderPcmU/A classes and convert AudioDecoder testshenrik.lundin@webrtc.org
2014-10-21audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>bjornv@webrtc.org
2014-10-21Fix for glitches in ACM when switching desired output sample ratehenrik.lundin@webrtc.org
2014-10-20audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>bjornv@webrtc.org
2014-10-20audio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>bjornv@webrtc.org
2014-10-16Add encoded_timestamp to AudioEncoder base classhenrik.lundin@webrtc.org
2014-10-16New interface class AudioEncoderhenrik.lundin@webrtc.org
2014-10-15Extend AcmSwitchingOutputFrequencyOldApi with more frequencieshenrik.lundin@webrtc.org
2014-10-14New ACM test to trigger audio glitch when switching output sample ratehenrik.lundin@webrtc.org
2014-10-14Workarounds for a bug in VS2013.3 linker when PGO is turned on.kwiberg@webrtc.org
2014-10-13iSAC tests: Type buffers as uint8_t[] to avoid castskwiberg@webrtc.org
2014-10-13WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[]kwiberg@webrtc.org
2014-10-13WebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16kwiberg@webrtc.org
2014-10-13Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[]kwiberg@webrtc.org
2014-10-09Opus wrapper: Use const for inputs and uint8[] for byte streamskwiberg@webrtc.org
2014-10-09Removing useless packets when inserting them (NetEq)minyue@webrtc.org
2014-10-09common_audio: Removed macro WEBRTC_SPL_LSHIFT_W16bjornv@webrtc.org
2014-10-08audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>bjornv@webrtc.org
2014-10-08Change name of a NetEq internal member variablehenrik.lundin@webrtc.org
2014-10-07Fix neteq_rtpplay so that empty SSRC is validhenrik.lundin@webrtc.org
2014-10-07Set NetEq playout mode through the Config structhenrik.lundin@webrtc.org
2014-10-07Add an SSRC filter to neteq_rtpplayhenrik.lundin@webrtc.org
2014-10-07Prevent reading outside iSAC bitstream, if the stream is corrupted.turaj@webrtc.org
2014-10-02Let RtpFileSource use RtpFileReaderhenrik.lundin@webrtc.org
2014-10-01Fix parallelizability in modules_tests.pbos@webrtc.org
2014-09-30GN: Enable libvpx, add link target and convert some test targetskjellander@webrtc.org
2014-09-29common_audio refactoring: Removed macro WEBRTC_SPL_LSHIFT_U32bjornv@webrtc.org
2014-09-28GN: Add common configs to all targets.kjellander@webrtc.org
2014-09-25Removing error triggered for disabling FEC on non-opusminyue@webrtc.org
2014-09-25Revert r7049/r7123, which added unnecessary "u"s to "return 0"s.henrik.lundin@webrtc.org
2014-09-24WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_tkwiberg@webrtc.org
2014-09-24Move thread_annotations.h to webrtc/base/.pbos@webrtc.org
2014-09-23Reland "Converting five tests to use new AudioCoding interface" (r7258)henrik.lundin@webrtc.org
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