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  • ea40bd0cc8 Presubmit script for preventing changes to protected files and add the full list of those files. henrike@webrtc.org 2013-07-29 18:20:07 +00:00
  • c0aa29c98c Updated WebRTC version to 3.37 TBR=tnakamura@webrtc.org elham@webrtc.org 2013-07-29 16:57:21 +00:00
  • 8400246fce Improved error messages when binaries are missing. Also stderr = stdout now. phoglund@webrtc.org 2013-07-29 11:01:03 +00:00
  • f87177a757 To fix a bug in InverseFFTAndWindow() function in AECM. It's a bufer overwritting issue, and thus Android AppRTCDemo app was broken (reported by Ami). Tested with audioproc offline test. Bit-exact. kma@webrtc.org 2013-07-26 23:43:33 +00:00
  • 1e09a71126 Update talk folder to revision=49952949 henrike@webrtc.org 2013-07-26 19:17:59 +00:00
  • 367f640bea webrtc/.gitignore: add parts of talk/examples/android and third_party/llvm to the list. fischman@webrtc.org 2013-07-26 17:50:10 +00:00
  • b6a6a24fda Updated WebRtcNsx_PrepareSpectrumNeon() in accordance with the new real FFT interface in APM. For reference, you can check https://webrtc-codereview.appspot.com/1830004/diff/92001/webrtc/modules/audio_processing/ns/nsx_core.c, line 594 "static void PrepareSpectrumC()". Tested with audioproc. Bit exact. kma@webrtc.org 2013-07-26 16:24:34 +00:00
  • b6433b7a1e Access receiving_ under receive_cs critical section Note: InsertRTPPacket/InsertRTCPPacket could be merged into ReceivedRTPPacket, as there are no other callers. braveyao@webrtc.org 2013-07-26 09:02:46 +00:00
  • abab1d8456 Don't set clang_use_chrome_plugins in common.gypi sergeyu@chromium.org 2013-07-26 00:55:46 +00:00
  • 14c966c706 Fixes resources and data path in modules_unittests.isolate. henrike@webrtc.org 2013-07-25 22:44:04 +00:00
  • b86fbaf1d4 Downstream latest Chromium SincResampler changes. andrew@webrtc.org 2013-07-25 22:04:30 +00:00
  • e691b4f952 Roll libvpx 211873:212975 to pick up build fixes for ninja iOS device build. fischman@webrtc.org 2013-07-25 21:22:41 +00:00
  • 6439afce5b Revert 4403 "Roll chromium_revision 205140:212975 to pick up bui..." Broke bot: http://chromegw.corp.google.com/i/client.webrtc/builders/Mac32%20Release/builds/312 fischman@webrtc.org 2013-07-25 21:02:21 +00:00
  • 60e4b0e472 Roll chromium_revision 205140:212975 to pick up build fixes for ninja iOS device build. fischman@webrtc.org 2013-07-25 20:49:59 +00:00
  • 099b8c9e8e Update include paths in device_info_external.cc sergeyu@chromium.org 2013-07-25 18:41:43 +00:00
  • 61e596fc49 Add a Config class interface to AudioProcessing for passing options. andrew@webrtc.org 2013-07-25 18:28:29 +00:00
  • c46967dc53 Revert 4391 "Roll chromium 205140:212975 to support ninja iOS ar..." turaj@webrtc.org 2013-07-25 17:48:24 +00:00
  • 8e3bbedacd Fix include path in video_capture_external.cc Fix build error introduced in r4337 niklas.enbom@webrtc.org 2013-07-25 16:55:58 +00:00
  • 33584f942c Introduce a bit of sanity to talk/PRESUBMIT.py's license checking. fischman@webrtc.org 2013-07-25 16:43:30 +00:00
  • 9fbc558dd4 talk/OWNERS: add libjingle team members from internal webrtc/files/OWNERS fischman@webrtc.org 2013-07-25 16:22:37 +00:00
  • 880c842627 AppRTCDemo: don't render frames that are already outdated. fischman@webrtc.org 2013-07-24 20:40:03 +00:00
  • 87f8a7eb67 Roll chromium 205140:212975 to support ninja iOS armv7 build. fischman@webrtc.org 2013-07-24 20:15:43 +00:00
  • fc8aaf02e1 Formalized Real 16-bit FFT for APM. It also prepares for introducing Real 16-bit FFT Neon code from Openmax to SPL. CL https://webrtc-codereview.appspot.com/1819004/ takes care of that, but this CL is a prerequisite of that one. Tested audioproc with an offline file. Bit exact. kma@webrtc.org 2013-07-24 17:38:23 +00:00
  • b63c29f48c Minor bug fix in r4388, had to change pc_config variable to pcConfig for apprtc demo. vikasmarwaha@webrtc.org 2013-07-23 23:13:35 +00:00
  • 59fb7a60f2 Use Mozilla STUN server in apprtc demo for FF. Currently FF cannot work with Google STUN server as it expects XOR-MAPPED address while Google STUN server provides MAPPED address. vikasmarwaha@webrtc.org 2013-07-23 22:06:51 +00:00
  • d102e66ef9 Fix ScreenCapturerLinux not to use XDamage when requested. sergeyu@chromium.org 2013-07-23 20:05:42 +00:00
  • 678cf29d8b webrtc/common_types.h: Document bitrate fields' units. fischman@webrtc.org 2013-07-23 18:32:10 +00:00
  • 8d27a1c723 Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle henrike@webrtc.org 2013-07-23 18:15:11 +00:00
  • 6879c8adad Hooking up first simple CPU adaptation version. mflodman@webrtc.org 2013-07-23 11:35:00 +00:00
  • 5c280ecd57 Revert 4382 "Makes webrtc and libjingle build from the same gyp-..." henrike@webrtc.org 2013-07-23 03:30:32 +00:00
  • 5fcddf2334 Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle henrike@webrtc.org 2013-07-23 00:27:43 +00:00
  • 390fcb7a20 Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder. henrike@webrtc.org 2013-07-22 22:32:50 +00:00
  • 28654cbc22 Update talk folder to revision=49713299. henrike@webrtc.org 2013-07-22 21:07:49 +00:00
  • 5bb8e7e220 Adjusted net delay perf expectations slightly. phoglund@webrtc.org 2013-07-22 13:15:59 +00:00
  • 129afc29fb Correctly rebuild WebRTCDemo after jni/ source file changes yujie.mao@webrtc.org 2013-07-20 04:43:08 +00:00
  • dd02935d5d Roll libvpx to 211873. -pickup public roll to: 33149cbb marpan@webrtc.org 2013-07-19 21:05:45 +00:00
  • 5e44b8fac1 Add libjingle's valgrind suppressions henrike@webrtc.org 2013-07-19 16:00:41 +00:00
  • 0df5b8dfa6 Revert 4372 "Makes webrtc and libjingle build from the same gyp-..." henrike@webrtc.org 2013-07-18 18:58:29 +00:00
  • 4e4bf4db8b Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. henrike@webrtc.org 2013-07-18 18:33:55 +00:00
  • c6d5b50b41 AppRTCDemo: build fixes for iOS build in webrtc fischman@webrtc.org 2013-07-18 02:02:07 +00:00
  • 9f07ea4c8a Roll tools/android 4235:4258, to pick up an x86 md5sum_bin binary yujie.mao@webrtc.org 2013-07-18 01:57:04 +00:00
  • d2102afa2a Undo libvpx include changes in r4348 to fix build. tnakamura@webrtc.org 2013-07-17 18:48:24 +00:00
  • 8c7347124c talk: DataChannel.java repeated contents. This removes the duplicate. henrike@webrtc.org 2013-07-17 16:28:37 +00:00
  • 9de257d00f Update talk folder to revision=49470012. Same as 375 in libjingle's google code repository. henrike@webrtc.org 2013-07-17 14:42:53 +00:00
  • a3f30143b7 Default constructor for RtcpAppHandler. pbos@webrtc.org 2013-07-17 14:25:45 +00:00
  • 64e2cbf184 clean up incomplete revert in r4357 Also revert r4319, will follow up with pbos tnakamura@webrtc.org 2013-07-16 21:52:59 +00:00
  • aa4d96a134 Revert r4301 tnakamura@webrtc.org 2013-07-16 19:25:04 +00:00
  • 7b2f955e56 Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README. henrike@webrtc.org 2013-07-16 18:04:56 +00:00
  • 42581545eb Fixes: Resolves conflict that will happen when merging libjingle's and WebRTC's supplemental.gyp. By separating build_with_chromium and build_with_libjingle one can now just define build_with_libjingle in libjingle's supplemental.gyp. Once that is done it will be possible to merge the two supplemental.gyp-files. I.e. in WebRTC the supplemental.gyp would only set build_with_chromium to 0 since there is no longer any reason to disable logging and tests as they will be accessible in the same repository as libjingle. henrike@webrtc.org 2013-07-16 16:37:22 +00:00
  • 3d8647f17d Include files from webrtc/.. paths in signal_processing/. pbos@webrtc.org 2013-07-16 13:32:03 +00:00
  • 0c4e05afbb Include files from webrtc/.. paths in media_file/. pbos@webrtc.org 2013-07-16 13:05:40 +00:00
  • 9b82dced8d Make sure first RTP packet counts as in-order. pbos@webrtc.org 2013-07-16 13:03:35 +00:00
  • 2e10b8e4a0 Include files from webrtc/.. paths in bitrate_controller/. pbos@webrtc.org 2013-07-16 12:54:53 +00:00
  • a4407329d4 Include files from webrtc/.. paths in video_coding/. pbos@webrtc.org 2013-07-16 12:32:05 +00:00
  • 4a44ea21d7 Revert r4320 "Fix three uninitialized members in rtp_receiver_impl.cc" TBR=pwestin@webrtc.org elham@webrtc.org 2013-07-15 21:46:06 +00:00
  • 4888fd4827 Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered" elham@webrtc.org 2013-07-15 21:21:48 +00:00
  • b7eda43810 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on several SSRCs" elham@webrtc.org 2013-07-15 21:08:27 +00:00
  • 6f5707e184 Revert r4328 elham@webrtc.org 2013-07-15 20:59:52 +00:00
  • 8543c1c77c Updated WebRTC version to 3.36 TBR=tnakamura@webrtc.org elham@webrtc.org 2013-07-15 17:19:45 +00:00
  • ca35c19e5a Roll libvpx to 208227. -pick up libvpx roll to 93f88ab. marpan@webrtc.org 2013-07-12 21:08:26 +00:00
  • df119c9a45 Remove dead video_capture for QuickTime. pbos@webrtc.org 2013-07-12 18:08:13 +00:00
  • 723d683ecb Update talk folder to revision=49260075. Same as 369 in libjingle's google code repository. henrike@webrtc.org 2013-07-12 16:04:50 +00:00
  • a9b74ad716 Include files from webrtc/.. paths in video_capture/. pbos@webrtc.org 2013-07-12 10:03:52 +00:00
  • 8b06200802 Include files from webrtc/.. paths in utility/. pbos@webrtc.org 2013-07-12 08:28:10 +00:00
  • 0ed57c51a3 Remove dead code testAPI.cc. pbos@webrtc.org 2013-07-12 08:23:05 +00:00
  • 5aa3f1b4c0 Include files from webrtc/.. paths in video_render/. pbos@webrtc.org 2013-07-12 08:12:08 +00:00
  • 5b10d8fb18 Fix some voe_auto_test uninitialised-value errors. pbos@webrtc.org 2013-07-11 15:50:07 +00:00
  • ffe16bdae9 trunk/talk: removes empty folders. henrike@webrtc.org 2013-07-11 15:42:10 +00:00
  • 811269df40 Include files from webrtc/.. paths in audio_device/. pbos@webrtc.org 2013-07-11 13:24:38 +00:00
  • db6e3f8bc5 Fix root-relative includes for pacing/. pbos@webrtc.org 2013-07-11 09:50:05 +00:00
  • e4736eee20 Fixes a crash when sending SR reports from a sender only module. stefan@webrtc.org 2013-07-11 08:28:35 +00:00
  • aeba6e8740 ModuleRTPRTCP call rtcp_sender_.TMMBR() directly instead of calling its own API. braveyao@webrtc.org 2013-07-11 08:06:37 +00:00
  • 96edd56170 Sorted headers under rtp_rtcp/. pbos@webrtc.org 2013-07-10 15:40:42 +00:00
  • 69215d8432 Include files from webrtc/.. paths in video_engine/. pbos@webrtc.org 2013-07-10 15:02:02 +00:00
  • adf23a55f8 Direct3D renderer for new VideoEngine API tests. pbos@webrtc.org 2013-07-10 14:07:56 +00:00
  • 717d147ebb Support sending multiple report blocks and keeping track of statistics on several SSRCs. stefan@webrtc.org 2013-07-10 13:39:27 +00:00
  • 9de89a6f6b Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered. stefan@webrtc.org 2013-07-10 12:42:15 +00:00
  • 452d853c43 Fix three uninitialized members in rtp_receiver_impl.cc. stefan@webrtc.org 2013-07-10 10:54:56 +00:00
  • 08933a5dfb Initialize payload-type frequency in channel.cc. pbos@webrtc.org 2013-07-10 10:06:29 +00:00
  • 28e2075280 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk henrike@webrtc.org 2013-07-10 00:45:36 +00:00
  • 6aa6229953 Update version number to 3.35 tnakamura@webrtc.org 2013-07-09 18:43:02 +00:00
  • c79b9295cd Update version number to 3.34 tnakamura@webrtc.org 2013-07-09 18:40:52 +00:00
  • fc496d95df Add root_path_android.cc to webrtc/test/Android.mk. pbos@webrtc.org 2013-07-09 15:24:16 +00:00
  • f3f1358360 Fixed implicit-int-conversion bugs. pbos@webrtc.org 2013-07-09 14:04:46 +00:00
  • cab716cc7d Fix a circular dependency by removing an unnecessary dependency, add a missing include_tests check and missing lib references for android. stefan@webrtc.org 2013-07-09 13:43:24 +00:00
  • f56d612c70 Create gyp target for bwe components. stefan@webrtc.org 2013-07-09 12:32:35 +00:00
  • af8d5afec9 Initial port of FullStackTest to new VideoEngine API. pbos@webrtc.org 2013-07-09 08:02:33 +00:00
  • 5fc4d34f54 Arguments need to be separated when implementing gyp-actions. henrike@webrtc.org 2013-07-09 02:08:25 +00:00
  • 1a7b9b94be Cleanup WebRTC tracing hclam@chromium.org 2013-07-08 21:31:18 +00:00
  • e80a934b36 Added modules_unittests.isolate for ndk-apk builds. henrike@webrtc.org 2013-07-08 21:19:57 +00:00
  • a950300b0e Disables unit tests that don't work on Android for Android. henrike@webrtc.org 2013-07-08 18:53:54 +00:00
  • a2073af728 Fixes build breakage when building WebRTC in Chromium and having include_tests=1. henrike@webrtc.org 2013-07-08 18:14:58 +00:00
  • bd3eee3e24 Fixes broken gyp-condition. henrike@webrtc.org 2013-07-08 17:34:20 +00:00
  • 34773d9b6b Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests. henrike@webrtc.org 2013-07-08 14:55:23 +00:00
  • 1932fe1865 Use scoped_ptr<> for loopback.cc pbos@webrtc.org 2013-07-05 17:02:37 +00:00
  • 66b2e5c05a Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the rtp_rtcp implementation. stefan@webrtc.org 2013-07-05 14:30:48 +00:00
  • d4d9480c05 Added gum4.html, a multiple camera opening demo, each opening with a different resolution and/or frame rate. mcasas@webrtc.org 2013-07-05 09:12:04 +00:00
  • db7d82f26f Revert 4298 "Makes it possible to find files used by some unit t..." pbos@webrtc.org 2013-07-05 08:49:09 +00:00
  • caf2fcca6a Makes it possible to find files used by some unit tests when running them as Chrome native tests. henrike@webrtc.org 2013-07-05 04:15:38 +00:00
  • 21beaf97e7 Adding Stefan as VideoEngine owner, removing Per. mflodman@webrtc.org 2013-07-04 12:29:08 +00:00