admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Commit Graph

  • 52b4e8871a Adding trace and changing pacing constants pwestin@webrtc.org 2013-05-02 19:02:17 +00:00
  • a5961b855e Update third party license file niklas.enbom@webrtc.org 2013-05-02 18:49:28 +00:00
  • 0d95e06a2f Bugfix custom call stop. pwestin@webrtc.org 2013-05-02 18:25:03 +00:00
  • ea83c6ac9d Allow voe_cmd_test to select Opus mono (now the default). andrew@webrtc.org 2013-05-02 15:57:36 +00:00
  • 8c845cb623 Relax VoE's max packet length threshold. andrew@webrtc.org 2013-05-02 15:28:02 +00:00
  • 258f55efc0 Disabled flaky test. phoglund@webrtc.org 2013-05-02 12:35:00 +00:00
  • 68e5a68f07 Revert 3933 "Remove traces of deprecated WebRtc_Word types." pbos@webrtc.org 2013-05-02 09:30:12 +00:00
  • 265a5d298a Remove traces of deprecated WebRtc_Word types. pbos@webrtc.org 2013-05-02 09:11:20 +00:00
  • 3c48f31e5b WebRTCDemo Android app to route audio to headphone when it's plugged in. braveyao@webrtc.org 2013-05-02 03:18:00 +00:00
  • 03d0c66376 Make libyuv fat on linux instead of thin. BUG=none TEST=none Review URL: https://webrtc-codereview.appspot.com/1382004 fbarchard@google.com 2013-05-02 01:01:24 +00:00
  • 28e82bfec6 Replace Resampler with PushResampler in transmit_mixer. andrew@webrtc.org 2013-05-02 00:30:36 +00:00
  • 342353780d Consolidate common_audio into a single target. andrew@webrtc.org 2013-04-30 23:43:26 +00:00
  • dff69c56b0 Add AEC suppression level option to audioproc. andrew@webrtc.org 2013-04-30 23:01:09 +00:00
  • 23516638fa Move WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME to system_wrappers.gypi . sergeyu@chromium.org 2013-04-30 22:53:51 +00:00
  • 72d0b0cf1f Add self to video_engine watchlist. andresp@webrtc.org 2013-04-30 22:20:53 +00:00
  • 4980679d35 Fixes two bugs in receive statistics. stefan@webrtc.org 2013-04-30 22:05:07 +00:00
  • d35964a1ce Fixing AV sync. Increased 2 const to allow for a bigger difference in AV sync. pwestin@webrtc.org 2013-04-30 16:06:10 +00:00
  • 6faba6edc9 VCM: Setting buffering delay in timing mikhal@webrtc.org 2013-04-30 15:39:34 +00:00
  • dd807ac474 Adding buffered mode to loopback test mikhal@webrtc.org 2013-04-30 15:19:47 +00:00
  • 8efc623fc2 Apply Chromium C++ style to RemoteRateControl. solenberg@webrtc.org 2013-04-30 08:33:46 +00:00
  • 15e32ccd30 Add DesktopCapturer interface for desktop capturers. sergeyu@chromium.org 2013-04-29 20:10:57 +00:00
  • 865ada3a52 Don't reset the last je value and mode mikhal@webrtc.org 2013-04-29 19:09:41 +00:00
  • 50b2efef6e Add a wrapper around PushSincResampler and the old Resampler. andrew@webrtc.org 2013-04-29 17:27:29 +00:00
  • 5b7120c81b Fix two issues where we might end up busy looping in decoder_render mode. stefan@webrtc.org 2013-04-29 16:41:30 +00:00
  • b0061f94b2 Enable Nack pacing. pwestin@webrtc.org 2013-04-27 00:41:08 +00:00
  • 47128ab5ab Removing vie file related code from vie_custom_call Follow up on https://code.google.com/p/webrtc/source/detail?r=3900 mikhal@webrtc.org 2013-04-26 20:09:54 +00:00
  • 4e545b33b3 Fixed remaining nits from Stefan pwestin@webrtc.org 2013-04-26 15:23:34 +00:00
  • 8fc05feed4 Add a push-based wrapper around SincResampler. andrew@webrtc.org 2013-04-26 14:56:51 +00:00
  • 42b0b84367 libyuv r680 fixes arm version of I444ToARGB and some lint changes BUG=none TEST=libyuv unittests pass on arm with Neon disabled. Review URL: https://webrtc-codereview.appspot.com/1356005 fbarchard@google.com 2013-04-26 02:23:32 +00:00
  • 1acb3b33bc Add comfort noise disabling and routing mode selection to audioproc. andrew@webrtc.org 2013-04-26 00:39:27 +00:00
  • 4cea79b830 Removing another instance of file api mikhal@webrtc.org 2013-04-25 23:31:27 +00:00
  • 77ac84814d Added new demo states.html & updated existing demos to work on firefox. vikasmarwaha@webrtc.org 2013-04-25 23:22:03 +00:00
  • 91563e42da Fix the encoder pause logic. BUG=1691 pwestin@webrtc.org 2013-04-25 22:20:08 +00:00
  • 381da4be9c VCM: Adding API for the size(duration) of the jitter buffer. Refers to the duration in time of the frames which are ready to be sent to the decoder. mikhal@webrtc.org 2013-04-25 21:45:29 +00:00
  • 8392cd9edd VCM/JB: Using last decoded state for waiting for key mikhal@webrtc.org 2013-04-25 21:30:50 +00:00
  • dc3cd217b2 VCM/JB: FrameForDecoding->IncompleteFrameForDecoding - Update complete frame for decoding - Remove FrameForDecodingNack mikhal@webrtc.org 2013-04-25 20:27:04 +00:00
  • b84f13f185 Disabling avi file interface mikhal@webrtc.org 2013-04-25 18:07:32 +00:00
  • 52aa019e98 Avoid adding duplicates in pacer lists. pwestin@webrtc.org 2013-04-25 17:35:56 +00:00
  • cb60fb2e6c Make sure timestamps are monotonically increasing. stefan@webrtc.org 2013-04-24 23:54:31 +00:00
  • c63772eb39 libyuv license file updates for Android WebView license check. BUG=none TEST=try bots Review URL: https://webrtc-codereview.appspot.com/1313013 fbarchard@google.com 2013-04-24 02:12:49 +00:00
  • df9c0e5ec9 Revert 3892 "VCM/JB: Using last decoded state for waiting for key" andrew@webrtc.org 2013-04-24 02:13:18 +00:00
  • 2e65346e98 Add a root codereview.settings file. andrew@webrtc.org 2013-04-24 02:08:50 +00:00
  • 6fec15941a Update template to follow chromium copyright style BUG=none TEST=none Review URL: https://webrtc-codereview.appspot.com/1322008 fbarchard@google.com 2013-04-24 01:01:28 +00:00
  • b5eeaa92ba Adding extra options to interact with external encoder/decoder. andresp@webrtc.org 2013-04-23 22:50:53 +00:00
  • 1248d4effc VCM/JB: Using last decoded state for waiting for key mikhal@webrtc.org 2013-04-23 20:57:06 +00:00
  • 3a9a3cdaa2 Roll libyuv to r676 for improved llvm compatibility BUG=none TESTED=try bots Review URL: https://webrtc-codereview.appspot.com/1313010 fbarchard@google.com 2013-04-23 18:05:46 +00:00
  • 8ca8a71de2 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." stefan@webrtc.org 2013-04-23 16:48:32 +00:00
  • a942692725 Buf fix for r3883. turaj@webrtc.org 2013-04-23 16:08:29 +00:00
  • ccd4b2aec8 Add a default RTT to CallStats and use different values for buffered/real-time mode. stefan@webrtc.org 2013-04-23 15:58:23 +00:00
  • d25b602dc0 VP8: Avoid copying the codec struct on Reset(). pbos@webrtc.org 2013-04-23 13:08:04 +00:00
  • efdf778d3f BUG=1351 mflodman@webrtc.org 2013-04-23 11:57:56 +00:00
  • c1f243f8e7 VCM/JB: Skip to the next complete key frame mikhal@webrtc.org 2013-04-22 22:24:38 +00:00
  • 63117339dc Updated the sync module with a slow moving filter pwestin@webrtc.org 2013-04-22 18:57:14 +00:00
  • 28d54ab18f Improve AV-sync when initial delay is set and NetEq has long buffer. turaj@webrtc.org 2013-04-22 18:53:35 +00:00
  • 1b427719dc emove desktop_capture.gypi from modules.gyp kjellander@webrtc.org 2013-04-22 13:13:49 +00:00
  • 7c9e992d05 Removed unused variable. mflodman@webrtc.org 2013-04-22 13:05:41 +00:00
  • aeff4f3003 Fixing Coverity issues. mflodman@webrtc.org 2013-04-22 12:41:57 +00:00
  • 8aa4a90bd9 Set mime type on device-switch.html tommi@webrtc.org 2013-04-22 05:43:13 +00:00
  • 006acc45c2 Update iOS build script to run on bots. kjellander@webrtc.org 2013-04-20 13:04:05 +00:00
  • e0e029e8cb Revert 3876 mikhal@webrtc.org 2013-04-19 21:11:41 +00:00
  • ee184b9520 VCM/Receiver: Only update render time when decoding mikhal@webrtc.org 2013-04-19 19:15:47 +00:00
  • c41478f7eb Ensure build_demo.py run subprocesses with bash shell. kjellander@webrtc.org 2013-04-19 11:50:42 +00:00
  • a39a8fec16 Add owner to Apprtc braveyao@webrtc.org 2013-04-19 02:34:45 +00:00
  • 1fea17dae4 Add the build script of the voice engine for iOS. sjlee@webrtc.org 2013-04-18 23:22:02 +00:00
  • a73d52ca52 revert r3871 mikhal@webrtc.org 2013-04-18 20:26:28 +00:00
  • 9756017717 - Replace the BWE_MIN and BWE_MAX macros with std::min and std::max - Add 'virtual' to a bunch of overridden methods of RemoteBitrateEstimatorMultiStream and RemoteBitrateEstimatorSingleStream. solenberg@webrtc.org 2013-04-18 19:12:42 +00:00
  • d26457fdeb Apply Chromium C++ style to BitRateStats. solenberg@webrtc.org 2013-04-18 12:25:32 +00:00
  • 65f995a3df New ViE interface. mflodman@webrtc.org 2013-04-18 12:02:52 +00:00
  • c14b728b71 Add lock to prevent possible rare race condition in Win coreAudio capture implementation. braveyao@webrtc.org 2013-04-18 09:32:07 +00:00
  • ceaedc0014 Remove executable bit from dc1.html. andrew@webrtc.org 2013-04-18 01:56:07 +00:00
  • a0cd9182aa Add desktop_capture directory for screen and window capturers. sergeyu@chromium.org 2013-04-17 18:21:42 +00:00
  • dbd6a6d653 Updating delay for first value mikhal@webrtc.org 2013-04-17 16:23:22 +00:00
  • 48c5882f2a Remove libvpx pre-processor conditions and conditional compile of default temporal layers files. andresp@webrtc.org 2013-04-17 15:31:40 +00:00
  • f09016744d Revert "Updating test file contents to emmastjernloef" kjellander@webrtc.org 2013-04-17 14:59:28 +00:00
  • f1bf3a00b2 A device switcher code example, with fake. hta@webrtc.org 2013-04-17 14:24:21 +00:00
  • 11959d3476 Updating test file contents to emmastjernloef kjellander@webrtc.org 2013-04-17 13:58:54 +00:00
  • db11fab49e Adding Opus unit test tina.legrand@webrtc.org 2013-04-17 10:39:41 +00:00
  • 4392d5f9f8 Fix for "RTP dynamic payload type 100 is reserved" henrika@webrtc.org 2013-04-17 07:34:25 +00:00
  • f1a3b4bc0c Issue 1647. Avoid unsequenced modification. turaj@webrtc.org 2013-04-16 17:01:35 +00:00
  • 6e788df19e Remove vim/emacs modelines from .gypi files pbos@webrtc.org 2013-04-16 12:40:34 +00:00
  • 56b5f77a2b Add support for multiple streams to RtpPlayer: - Tests video_rtp_play.cc, video_rtp_play_mt.cc, decode_from_storage.cc rewritten - rtp_player.cc/.h rewritten; added interfaces for externally setting up sinks - Support for reading .rtp files pulled out into rtp_file_reader namespace - Added support for reading .pcap (libpcap/wireshark/tcpdump) files, see pcap_file_reader solenberg@webrtc.org 2013-04-16 10:31:56 +00:00
  • 885cd13356 Start NACKing as soon as we have the first packet of a key frame. stefan@webrtc.org 2013-04-16 09:38:26 +00:00
  • bdb9b971be Change receive statistics bitrate to be provided in bps instead of kbps. stefan@webrtc.org 2013-04-16 09:02:03 +00:00
  • e44a064915 Make win_support_condition_variables_primitive global to aligned with |library| so that once we set it to true it will remain. Review URL: https://webrtc-codereview.appspot.com/1319006 wu@webrtc.org 2013-04-15 18:06:42 +00:00
  • 92d1f07551 Elevate NetEq short-term activity statistics to ACM level for logging. Review URL: https://webrtc-codereview.appspot.com/1313004 turaj@webrtc.org 2013-04-15 16:52:04 +00:00
  • 4b8de90dce Disable -Wunsequenced warning in audio_coding_module kjellander@webrtc.org 2013-04-15 06:38:56 +00:00
  • f806ad23db Roll chromium_revision 182149:193311 kjellander@webrtc.org 2013-04-15 05:52:57 +00:00
  • c83b35661d Roll libvpx to 192165. -pick up libvpx roll to 3db60c8. marpan@webrtc.org 2013-04-12 21:43:48 +00:00
  • c2a3aa7926 Partial revert of r3844 mikhal@webrtc.org 2013-04-12 19:53:30 +00:00
  • d6bd7cd2b1 removing redundant calls to cleanframes mikhal@webrtc.org 2013-04-12 17:09:51 +00:00
  • 9f5ebb5251 Adding a payload type for RTX. mflodman@webrtc.org 2013-04-12 14:55:46 +00:00
  • b8e7f4cc97 Change capture interface to use NTP capture time. stefan@webrtc.org 2013-04-12 11:56:23 +00:00
  • d35dff7664 Move to Chrome infra try server. kjellander@webrtc.org 2013-04-12 07:40:33 +00:00
  • 1de01354e6 Adding playout buffer status to the voe video sync Review URL: https://webrtc-codereview.appspot.com/1311004 pwestin@webrtc.org 2013-04-11 20:23:35 +00:00
  • 9da751715f VCM/JB:Removing hybrid and setting a decodable state. mikhal@webrtc.org 2013-04-11 18:49:13 +00:00
  • 7bc465bd21 Fix issues with incorrect wrap checks when having big buffers and high bitrate. stefan@webrtc.org 2013-04-11 17:48:02 +00:00
  • 122d209e67 Fixes an issue where the start bitrate is stored in kbps instead of bps. stefan@webrtc.org 2013-04-11 17:21:40 +00:00
  • eac36b8561 Fix -Wstring-conversion warnings. Review URL: https://webrtc-codereview.appspot.com/1299007 wu@webrtc.org 2013-04-11 15:37:46 +00:00
  • 523f93729b Re-write the build of the nacklist. Review URL: https://webrtc-codereview.appspot.com/1304008 andresp@webrtc.org 2013-04-11 11:30:39 +00:00
  • f2a97fc2b4 WebRTCDemo: handle stride!=width from first frame. Previously only mid-stream frames handled stride!=width correctly. fischman@webrtc.org 2013-04-10 23:21:10 +00:00