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

Commit Graph

  • 5964fe0f86 audio_processing: DestroyHandle() now returns void bjornv@webrtc.org 2014-04-22 06:52:28 +00:00
  • 2a796720f8 common_audio: VADFree() now returns void bjornv@webrtc.org 2014-04-22 04:45:35 +00:00
  • 3dfabf928c libyuv r1000 roll for DEPS update to new chromium moving location of gold linker on linux. BUG=libyuv:323 TESTED=try bots R=andrew@webrtc.org fbarchard@google.com 2014-04-21 22:06:06 +00:00
  • ec3d8ecdcc Fix typo by renaming RTCSessionDescriptonDelegate -> RTCSessionsDescriptionDelegate tkchin@webrtc.org 2014-04-21 18:47:24 +00:00
  • 229e16e254 Add resource audio for audio processing tests. andrew@webrtc.org 2014-04-20 03:54:46 +00:00
  • 54fd70046d Remove ASSERT in TransportChannelProxy::SetImplementation, when proxy already set to same transport channel impl. mallinath@webrtc.org 2014-04-19 01:03:33 +00:00
  • f5a33f145b Resampler modifications in preparation for arbitrary audioproc rates. andrew@webrtc.org 2014-04-19 00:32:07 +00:00
  • 3d9ec1fed4 Fix multi-monitor support in the screen capturer for Mac. sergeyu@chromium.org 2014-04-19 00:25:35 +00:00
  • 8e5ec52e76 (Auto)update libjingle 65152644-> 65219629 buildbot@webrtc.org 2014-04-19 00:00:31 +00:00
  • 7d055a6e63 Revert r5937 "Fix multi-monitor support in the screen capturer for Mac." sergeyu@chromium.org 2014-04-18 23:45:38 +00:00
  • 0daa8be9d6 Add Chromium's ScopedVector. andrew@webrtc.org 2014-04-18 21:20:54 +00:00
  • be7585b150 Fix multi-monitor support in the screen capturer for Mac. sergeyu@chromium.org 2014-04-18 18:22:41 +00:00
  • a596a389ea Fix iSAC/48000 issue with ACM2. turaj@webrtc.org 2014-04-17 23:30:49 +00:00
  • 29540b1879 Revert "PeerConnectionFactory: delay deletion of owned threads." fischman@webrtc.org 2014-04-17 22:54:30 +00:00
  • 1a87f529a2 (Auto)update libjingle 65151416-> 65151642 buildbot@webrtc.org 2014-04-17 22:41:30 +00:00
  • cea024d672 PeerConnectionFactory: delay deletion of owned threads. fischman@webrtc.org 2014-04-17 22:36:00 +00:00
  • b476d36120 Roll libvpx 259973:264320 marpan@webrtc.org 2014-04-17 20:35:03 +00:00
  • aeb0c28193 Update PRESUBMIT.py's list of "DO_NOT_SUBMIT_FILES". henrike@webrtc.org 2014-04-17 14:15:43 +00:00
  • e57ae02327 WebRtcAecm_Process: Reduce code duplication kwiberg@webrtc.org 2014-04-17 12:28:33 +00:00
  • d2f366f28c StereoToMono: Remove useless call to WebRtcSpl_SatW32ToW16 kwiberg@webrtc.org 2014-04-17 12:17:39 +00:00
  • 66803489f9 Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. henrika@webrtc.org 2014-04-17 10:45:01 +00:00
  • 0f7375504a Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. henrika@webrtc.org 2014-04-17 10:38:08 +00:00
  • e2e9abb3bc Revert "Make VoiceEngine choose ACM2 by default" henrik.lundin@webrtc.org 2014-04-17 10:12:27 +00:00
  • 0b3c6c3191 (Auto)update libjingle 65086785-> 65104022 buildbot@webrtc.org 2014-04-17 10:03:57 +00:00
  • adaf809612 Removing AudioCoding duplicate tests henrik.lundin@webrtc.org 2014-04-17 08:29:10 +00:00
  • 6cec07f6a7 Make VoiceEngine choose ACM2 by default henrik.lundin@webrtc.org 2014-04-17 08:07:18 +00:00
  • c0a15b7ddc Fix crashes due to dangling external decoder pointer. fischman@webrtc.org 2014-04-17 01:22:48 +00:00
  • 39b868bad3 (Auto)update libjingle 65055925-> 65086785 buildbot@webrtc.org 2014-04-17 00:04:39 +00:00
  • 8f88f20af2 Expand the test max wait time from 1000ms to 2000ms. The createOffer/createAnswer methods sometimes times out due to slow identity generation under memcheck. jiayl@webrtc.org 2014-04-16 17:14:21 +00:00
  • c18729165a Set include_internal_video_capture=1 for video_capture_tests kjellander@webrtc.org 2014-04-16 12:59:49 +00:00
  • f927fd6481 Re-enable AGC tests: * AgcConfigTest.HasCorrectDefaultConfiguration * AgcConfigTest.DealsWithInvalidParameters * AgcConfigTest.CanGetAndSetAgcStatus * AgcConfigTest.HasCorrectDefaultRxConfiguration * AgcConfigTest.DealsWithInvalidRxParameters * AgcConfigTest.CanGetAndSetRxAgcStatus * AudioProcessingTest.AgcIsOnByDefault * AudioProcessingTest.CanEnableAgcWithAllModes * AudioProcessingTest.RxAgcShouldBeOffByDefault * AudioProcessingTest.CanTurnOnDigitalRxAcg * AudioProcessingTest.CannotTurnOnAdaptiveAnalogRxAgc aluebs@webrtc.org 2014-04-16 11:58:18 +00:00
  • 7de47bce12 Remove use of tmpnam. kjellander@webrtc.org 2014-04-16 08:04:26 +00:00
  • 2c3f1abb69 Replace flooding logs in rtp_sender.cc with a comment. andrew@webrtc.org 2014-04-15 21:26:34 +00:00
  • 36eda7cf0e Workaround for https://bugzilla.mozilla.org/show_bug.cgi?id=996329, where the m line from firefox have a space at the end. wu@webrtc.org 2014-04-15 20:37:30 +00:00
  • ca539bbed0 iOS: baby steps to being able to include_tests=1 fischman@webrtc.org 2014-04-15 20:26:41 +00:00
  • 7c6e3d188a Moved voe_neteq_stats_unittest to audio_coding_module_unittest henrik.lundin@webrtc.org 2014-04-15 17:59:25 +00:00
  • 6c75c98964 Propagate capture ntp timestamp from rtp to renderer. wu@webrtc.org 2014-04-15 17:46:33 +00:00
  • 1fd5b45a0e (Auto)update libjingle 64956819-> 64982143 buildbot@webrtc.org 2014-04-15 17:39:43 +00:00
  • 2f8d5f3302 Check if a header extension is registered before updating it and fail silently if it's not. stefan@webrtc.org 2014-04-15 12:28:46 +00:00
  • 190b72a350 Make libjingle Android example build without sourcing envsetup.sh kjellander@webrtc.org 2014-04-15 08:35:49 +00:00
  • 6e105ede38 Make WebRTC Android examples build without sourcing envsetup.sh kjellander@webrtc.org 2014-04-15 08:35:00 +00:00
  • ad4440a64e In shared socket mode, use udp port as default receiver even if stun server address is not set. mallinath@webrtc.org 2014-04-15 01:10:58 +00:00
  • 505f400f27 (Auto)update libjingle 64909599-> 64919255 buildbot@webrtc.org 2014-04-14 20:33:47 +00:00
  • e98598d3f0 Make everyone an OWNER for .gyp/.gypi add/delete purposes, talk/ edition. fischman@webrtc.org 2014-04-14 20:31:16 +00:00
  • 2c89b5cb27 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. fischman@webrtc.org 2014-04-14 20:08:03 +00:00
  • 35ead381f8 Adding a config struct to NetEq henrik.lundin@webrtc.org 2014-04-14 18:49:17 +00:00
  • 810acbc93e New Packet and PacketSource classes for NetEq tests henrik.lundin@webrtc.org 2014-04-14 18:42:23 +00:00
  • 1da6047132 (Auto)update libjingle 64813990-> 64909599 buildbot@webrtc.org 2014-04-14 18:15:15 +00:00
  • cf0b46c762 iosdeviceinfo.cc: remove unnecessary file fischman@webrtc.org 2014-04-14 18:12:32 +00:00
  • 5cf73962e6 Fix gyp for video_capture/ensure_initialized.cc. primiano@chromium.org 2014-04-14 17:26:31 +00:00
  • f875f15afb (Auto)update libjingle 64709629-> 64813990 buildbot@webrtc.org 2014-04-14 16:06:21 +00:00
  • b9309beea4 Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. henrika@webrtc.org 2014-04-14 14:12:50 +00:00
  • 5692531f18 Added a new OnMoreData() interface which will not feed the playout data to APM. xians@webrtc.org 2014-04-14 10:50:37 +00:00
  • a956ec2019 Add win_drmemory_light trybot to default trybot list. kjellander@webrtc.org 2014-04-14 08:38:27 +00:00
  • 940894d207 DrMemory: Excluding failing tests for Dr Memory Full kjellander@webrtc.org 2014-04-13 11:45:43 +00:00
  • ef79fd7965 DrMemory: Excluding failing tests for Dr Memory Full BUG=3158 TEST=None TBR=kjellander,phoglund kjellander@webrtc.org 2014-04-13 08:31:42 +00:00
  • 41e8774113 DrMemory: Excluding failing tests for Dr Memory Full kjellander@webrtc.org 2014-04-12 19:53:06 +00:00
  • 8ce7c72456 Fix the captured screen rect conversion. device_mode.dmPosition is already relative to the primary display's top-left, while the expected value of GetScreenRect() is also relative to the primary display's top-left. jiayl@webrtc.org 2014-04-11 22:31:15 +00:00
  • 8d1cdaa84e NetEq changes. turaj@webrtc.org 2014-04-11 18:47:55 +00:00
  • ffd242432d DrMemory: Suppress and exclude more tests to green up the full build. kjellander@webrtc.org 2014-04-11 14:16:27 +00:00
  • 34c5da6b5e Cleaned up logging in video_coding. stefan@webrtc.org 2014-04-11 14:08:35 +00:00
  • 8b2ec15d1e Convert WEBRTC_TRACE to LOG in utility. asapersson@webrtc.org 2014-04-11 07:59:43 +00:00
  • b884eb6118 (Auto)update libjingle 64630087-> 64709629 wu@webrtc.org 2014-04-10 16:59:16 +00:00
  • 8dce41b3c6 Remove erronuous commit message from auto sync. henrike@webrtc.org 2014-04-10 14:39:38 +00:00
  • 22cf7472a0 Disable UsesTraceCallback pbos@webrtc.org 2014-04-10 14:39:22 +00:00
  • e6013bb7be Fix loopback test for case where no constraint is given. R=stefan@webrtc.org TBR=stefan@webrtc.org andresp@webrtc.org 2014-04-10 14:21:45 +00:00
  • 2a770828d8 Remove usage of webrtc trace in video processing modules. asapersson@webrtc.org 2014-04-10 11:30:49 +00:00
  • 0273fa98e0 Add ability to control peer connection constraints for the loopback test. andresp@webrtc.org 2014-04-10 09:40:16 +00:00
  • 15192f909e (Auto)update libjingle 64594651-> 64630087 buildbot@webrtc.org 2014-04-10 06:34:32 +00:00
  • f93021430d Remove self-assignment hacks that were added to avoid unused variable warnings. Instead, appear to use the variables. fischman@webrtc.org 2014-04-09 21:19:55 +00:00
  • 0569d93db7 Move a chatty creation log in neteq to LS_VERBOSE. andrew@webrtc.org 2014-04-09 17:48:48 +00:00
  • 8f89497949 Remove erronuous commit message. henrike@webrtc.org 2014-04-09 14:43:43 +00:00
  • f4357f3530 Make Android-APK compile in release again. solenberg@webrtc.org 2014-04-09 14:21:37 +00:00
  • 52fd65b16a Partial revert of "Removing samples directory following move to Github" kjellander@webrtc.org 2014-04-09 13:52:24 +00:00
  • 8883a0f47f (landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds henrika@webrtc.org 2014-04-09 13:04:12 +00:00
  • 7ecc142d6b Removing samples directory following move to Github dutton@google.com 2014-04-09 09:55:54 +00:00
  • 61c1b8ea32 (Auto)update libjingle 64585415-> 64594651 buildbot@webrtc.org 2014-04-09 06:06:38 +00:00
  • 2e9d89cf77 Unbreak android APK buildbots by emptying the video_capture_tests_apk target. fischman@webrtc.org 2014-04-09 02:34:50 +00:00
  • b0b135e4c2 VideoCaptureAndroid: support multiple frame-rates per resolution. fischman@webrtc.org 2014-04-09 01:18:32 +00:00
  • 74f6074ec1 Fix DesktopSize::is_empty() for the case when only width or only height is 0. sergeyu@chromium.org 2014-04-09 01:04:22 +00:00
  • a78a41f985 Move output_mixer_unittest.cc to utility_unittest.cc. andrew@webrtc.org 2014-04-08 23:09:28 +00:00
  • f4c9444c65 VideoCaptureAndroid: stop referencing ViERenderer fischman@webrtc.org 2014-04-08 22:55:07 +00:00
  • f824fde36f (Auto)update libjingle 64326665-> 64585415 henrike@webrtc.org 2014-04-08 22:13:01 +00:00
  • 984e4fbaaa video_capture(iOS): move stopCapture to background thread fischman@webrtc.org 2014-04-08 21:06:52 +00:00
  • 2a03498825 Implement FEC support in VideoReceiveStream. pbos@webrtc.org 2014-04-08 11:21:45 +00:00
  • dc80bae2a6 Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. Clean some logs and add asserts in the way. andresp@webrtc.org 2014-04-08 11:06:12 +00:00
  • b287d968d9 New NetEq test to verify correct timestamp propagation henrik.lundin@webrtc.org 2014-04-07 21:21:45 +00:00
  • 74a7c482b9 Removes unused thread causing compiler warnings. henrike@webrtc.org 2014-04-07 20:49:34 +00:00
  • 4e393070be Compare the answer's media type against offer to make sure they are match. Otherwise we should return failure. wu@webrtc.org 2014-04-07 17:04:35 +00:00
  • 413d001132 Removed the disabling of include_tests from r2729. henrike@webrtc.org 2014-04-07 15:52:31 +00:00
  • 9337c839da Updated WebRTC version to 3.52 TBR=wu@webrtc.org elham@webrtc.org 2014-04-07 15:49:00 +00:00
  • b08db28958 Clean up traces and logs in RemoteBitrateEstimator. stefan@webrtc.org 2014-04-07 12:53:28 +00:00
  • 5574dacd1f Log Fixit for parts of video_engine folder. mflodman@webrtc.org 2014-04-07 10:56:31 +00:00
  • e8d1865408 Disable more tests for DrMemory to speed up execution. kjellander@webrtc.org 2014-04-07 09:00:12 +00:00
  • 36947bb635 Fix logging calls in bitrate_controller module. andresp@webrtc.org 2014-04-07 08:45:16 +00:00
  • 9f57404334 Excluding and suppressing Dr Memory test failures. kjellander@webrtc.org 2014-04-07 08:01:06 +00:00
  • 0fefb1041c Remove WEBRTC_TRACE use in common_video/ pbos@webrtc.org 2014-04-07 07:29:18 +00:00
  • 09b0c10eed Talk: fixes warning: local variable is initialized but not referenced due to only using the variable in question for asserts. henrike@webrtc.org 2014-04-04 22:33:34 +00:00
  • d1fe6b728e AppRTCDemo(android): fix a couple of SDP-related regressions. fischman@webrtc.org 2014-04-04 21:40:46 +00:00
  • f040bd8fa3 Fix a crash in WindowCapturereMac when capture() fails. jiayl@webrtc.org 2014-04-04 20:26:41 +00:00