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path: root/webrtc/modules/audio_conference_mixer
AgeCommit message (Expand)Author
2016-01-13Convert channel counts to size_t.Peter Kasting
2015-12-16OWNERS: Add * to .gyp{i,} everywhere.kjellander@webrtc.org
2015-11-18So long and thanks for all the code reviews!andrew
2015-11-06Remove interface directories kept to avoid breaking downstream.kjellander
2015-11-04modules: more interface -> include renamesHenrik Kjellander
2015-10-28system_wrappers: rename interface -> includeHenrik Kjellander
2015-09-23Fix the maximum native sample rate in AudioProcessingAlejandro Luebs
2015-09-03Reland "Adding unittests to AudioConferenceMixer."minyue
2015-09-01Revert "Adding unittests to AudioConferenceMixer."minyuel
2015-09-01Adding unittests to AudioConferenceMixer.minyuel
2015-08-31Stylizing AudioConferenceMixer.minyuel
2015-08-24Update a ton of audio code to use size_t more correctly and in general reducePeter Kasting
2015-07-28Fix to "Removing AudioMixerStatusReceiver and ParticipantStatistics"Minyue Li
2015-07-28Removing AudioMixerStatusReceiver and ParticipantStatistics.Minyue
2015-04-01Remove henrike@ from OWNERSHenrik Kjellander
2015-03-04Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macrokjellander@webrtc.org
2015-02-26Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going awaykwiberg@webrtc.org
2015-02-25Moved gypi to avoid presubmit warning about '..' when touching the files.andresp@webrtc.org
2015-01-29Remove ChangeUniqueID.tommi@webrtc.org
2015-01-14Move system_wrappers.gyp files to the proper directory.andresp@webrtc.org
2014-12-15Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.pkasting@chromium.org
2014-09-28GN: Add common configs to all targets.kjellander@webrtc.org
2014-09-11Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.henrik.lundin@webrtc.org
2014-09-08Convert GN visibility to be a list.brettw@chromium.org
2014-08-26Remove Android.mk build files.pbos@webrtc.org
2014-08-26Remove former team members from OWNERS and WATCHLISTSkjellander@webrtc.org
2014-08-25GN: Disable Chromium clang plugins for standalone build.kjellander@webrtc.org
2014-07-18This is related to an earlier CL of enabling Opus 48 kHz.minyue@webrtc.org
2014-07-17Use _numMixedParticipants instead of audioFrameList->size() to determine if t...wu@webrtc.org
2014-06-23GN: Add BUILD.gn files + kjellander to OWNERSkjellander@webrtc.org
2014-06-05Fix the chain that propagates the audio frame's rtp and ntp timestamp including:wu@webrtc.org
2014-05-13The webrtc::AudioFrame struct contains a variable energy_. Since the energy i...henrik.lundin@webrtc.org
2014-04-14Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.fischman@webrtc.org
2014-01-29Add Config struct for experimental AGC.andrew@webrtc.org
2014-01-27Fix bug introduced during replace of list wrapper with std equivalents in r5378.andresp@webrtc.org
2014-01-13Removes the remaining uses of the list wrapper class and the list wrapper class.henrike@webrtc.org
2014-01-13Removes usage of ListWrapper from several files.henrike@webrtc.org
2014-01-07Remove the requirement to call set_sample_rate_hz and friends.andrew@webrtc.org
2013-10-22Upgrade scoped_ptr to Chromium's latest version.andrew@webrtc.org
2013-08-12Replace MapWrapper with std::map<>.pbos@webrtc.org
2013-08-09Remove unused unreferenced code in webrtc/pbos@webrtc.org
2013-07-03Proper spacing for end-of-namespace comments.pbos@webrtc.org
2013-05-28Include files from webrtc/.. paths in audio_conference_mixer/pbos@webrtc.org
2013-04-16Remove vim/emacs modelines from .gypi filespbos@webrtc.org
2013-04-09WebRtc_Word32 -> int32_t in audio_conference_mixer/pbos@webrtc.org
2013-01-22Replace AudioFrame's operator= with CopyFrom().andrew@webrtc.org
2013-01-18Remove '<(library)' in gyp files.wjia@webrtc.org
2012-12-07Properly zero out unmixed frames.andrew@webrtc.org
2012-12-03Removing redundant codec unittest targets.kjellander@webrtc.org
2012-10-22Move src/ -> webrtc/andrew@webrtc.org
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