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path: root/webrtc/modules/audio_coding/neteq
AgeCommit message (Expand)Author
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
2014-10-02Let RtpFileSource use RtpFileReaderhenrik.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
2014-09-23Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..."andrew@webrtc.org
2014-09-22WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_tkwiberg@webrtc.org
2014-09-22Ensure that NetEq recovers after a large timestamp jumphenrik.lundin@webrtc.org
2014-09-19Creating a test helper class TimestampJumpRtpGeneratorhenrik.lundin@webrtc.org
2014-09-17Modifying NetEqExternalDecoderTesthenrik.lundin@webrtc.org
2014-09-15Re-enable neteq_performance_unittest.cc for android.andresp@webrtc.org
2014-09-10Mark all virtual overrides in the hierarchy of AudioPacketizationCallback,henrike@webrtc.org
2014-09-04Rename Audio[Multi]Vector.CopyFrom to .CopyTohenrik.lundin@webrtc.org
2014-09-04Change gflags and gmock includes to be full paths.kjellander@webrtc.org
2014-09-04Add support for multi-channel DTMF tone generationhenrik.lundin@webrtc.org
2014-09-03Partial revert of r7014 (Android APK refactor)kjellander@webrtc.org
2014-09-02Disable video_engine_tests and webrtc_perf_tests on Android.kjellander@webrtc.org
2014-09-02Divide-by-zero problem in NetEq's Normal::Process fixedhenrik.lundin@webrtc.org
2014-09-01Fix audio_decoder_unittests.isolatekjellander@webrtc.org
2014-09-01Android APK tests built from a normal WebRTC checkout.kjellander@webrtc.org
2014-08-29Allow same src and dst in InputAudioFile::DuplicateInterleavedhenrik.lundin@webrtc.org
2014-08-21Use a deterministic input in NetEqBgnTesthenrik.lundin@webrtc.org
2014-08-13NetEq background noise generation off by defaulthenrik.lundin@webrtc.org
2014-08-11Merge NetEqDecodingTest.TestBitExactnesst and .TestNetworkStatisticshenrik.lundin@webrtc.org
2014-08-11Use test::Packet test::PacketSource classes in neteq_rtpplayhenrik.lundin@webrtc.org
2014-08-07Change how background noise mode in NetEq is sethenrik.lundin@webrtc.org
2014-08-05initialize packet len in NETEQTEST_DummyRTPpacket.cc and NETEQTEST_RTPpacket....fbarchard@google.com
2014-07-22The lastest commit on this file was inminyue@webrtc.org
2014-07-18This is to re-open an earlier CLminyue@webrtc.org
2014-06-26Receiver bit-exactness test for AudioCoding Modulehenrik.lundin@webrtc.org
2014-06-25This is to compare NetEq with various codecs under a shared packet loss pattern.minyue@webrtc.org
2014-06-25Remove payload duplication in AudioDecoderTesthenrik.lundin@webrtc.org
2014-06-21Roll chromium_revision 272489:277350 + fix sanitizer optionskjellander@webrtc.org
2014-06-19Adding an empty constructor implementation to the AudioSink classhenrik.lundin@webrtc.org
2014-06-19Adding test::AudioSink interface and derived classeshenrik.lundin@webrtc.org
2014-06-18Update PacketSource and RtpFileSourcehenrik.lundin@webrtc.org
2014-06-18Revert "Restore ptypes.txt file"henrik.lundin@webrtc.org
2014-06-17Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."minyue@webrtc.org
2014-06-17Restore ptypes.txt filehenrik.lundin@webrtc.org
2014-06-17Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-...minyue@webrtc.org
2014-06-10Add kjellander@webrtc.org as OWNER for *.isolatekjellander@webrtc.org
2014-06-09Rename neteq4 folder to neteqhenrik.lundin@webrtc.org
2014-05-28Revert 6257 "Rename neteq4 folder to neteq"henrik.lundin@webrtc.org
2014-05-28Rename neteq4 folder to neteqhenrik.lundin@webrtc.org
2014-05-13Deleting all NetEq3 fileshenrik.lundin@webrtc.org
2014-04-14Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.fischman@webrtc.org
2014-04-02Rename RTPanalyze to rtp_analyze and remove old versionhenrik.lundin@webrtc.org
2014-03-04Including algorithm header to avoid VS2013 breakagehenrik.lundin@webrtc.org
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