generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Commit Graph

  • 3ea35fdb1b common_audio: Removed macro WEBRTC_SPL_LSHIFT_W16 bjornv@webrtc.org 2014-10-09 08:47:02 +00:00
  • 127ca3f8e5 Disable TestDTLSConnectWithSmallMtu on all platforms. pbos@webrtc.org 2014-10-09 07:52:03 +00:00
  • 0001adcfef Use openmax_dl on all ARM (v7 or higher) platforms. andrew@webrtc.org 2014-10-09 04:13:02 +00:00
  • 95bacfed08 Remove bad waiting code from video decoder release function. glaznev@webrtc.org 2014-10-09 00:00:11 +00:00
  • 97abeee282 (Auto)update libjingle 77263371-> 77296420 buildbot@webrtc.org 2014-10-08 22:24:30 +00:00
  • 536eb98408 Re-enables a bunch of base unittests. henrike@webrtc.org 2014-10-08 22:17:02 +00:00
  • 9ea539605e Roll chromium_revision fc668e2..2d714fa (298195:298667) andrew@webrtc.org 2014-10-08 19:16:10 +00:00
  • 4165f7aa22 Add a variable for deciding when to use openmax_dl. andrew@webrtc.org 2014-10-08 18:01:27 +00:00
  • f71785cd3b audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >> bjornv@webrtc.org 2014-10-08 15:36:30 +00:00
  • 575d126a3d Protect send_/recv_streams_ in WebRtcVideoEngine2. pbos@webrtc.org 2014-10-08 14:48:08 +00:00
  • 9c6dc46c6d CHECK/DCHECK: Explicitly state whether the condition can have side effects kwiberg@webrtc.org 2014-10-08 12:19:56 +00:00
  • 5e3d7c78de Change name of a NetEq internal member variable henrik.lundin@webrtc.org 2014-10-08 12:10:53 +00:00
  • 742922b313 Make the media content send only if offerToReceive is false while local streams exist. We previously do not add the media content if offerToReceive is false. jiayl@webrtc.org 2014-10-07 21:32:43 +00:00
  • d6bda09503 Initialize sctp_paddrparams in OpenSctpSocket(). pbos@webrtc.org 2014-10-07 19:23:43 +00:00
  • 27e5898f45 Explicitly unpoison FDs for MSan. pbos@webrtc.org 2014-10-07 17:56:53 +00:00
  • 46ffc70878 Temporary fix to allow Invoke() calls for VP8 HW encoder and decoder. glaznev@webrtc.org 2014-10-07 17:11:36 +00:00
  • 963b979510 Remove potential deadlock in WebRtcVideoEngine2. pbos@webrtc.org 2014-10-07 14:27:27 +00:00
  • a9e363e721 Roll chromium_revision c264a05..fc668e2 (297113:298195) kjellander@webrtc.org 2014-10-07 12:49:34 +00:00
  • 77d5a57e5c Revert "Only configure the SSL library in one place." pbos@webrtc.org 2014-10-07 11:43:03 +00:00
  • 6ed1cf49f0 Isolate: Remove use of --ignore_broken_items kjellander@webrtc.org 2014-10-07 09:17:35 +00:00
  • 9103953b58 Fix neteq_rtpplay so that empty SSRC is valid henrik.lundin@webrtc.org 2014-10-07 07:18:36 +00:00
  • 7cbc4f969a Set NetEq playout mode through the Config struct henrik.lundin@webrtc.org 2014-10-07 06:37:39 +00:00
  • 8b65d511a0 Add an SSRC filter to neteq_rtpplay henrik.lundin@webrtc.org 2014-10-07 05:30:04 +00:00
  • 532ed43e85 Prevent reading outside iSAC bitstream, if the stream is corrupted. turaj@webrtc.org 2014-10-07 00:21:02 +00:00
  • 8234fa6f0e Only configure the SSL library in one place. henrike@webrtc.org 2014-10-06 22:30:46 +00:00
  • 2fe5893748 Mac: adds missing _DEBUG flag to mac debug builds. henrike@webrtc.org 2014-10-06 22:04:11 +00:00
  • 528fc650d8 Fixing build issue with L-sdk henrike@webrtc.org 2014-10-06 17:56:43 +00:00
  • 9a742b4840 talk: removes empty directories base and sound. henrike@webrtc.org 2014-10-06 17:52:59 +00:00
  • 5d3e7ac1a3 Check on the existence of report directory houssainy@google.com 2014-10-06 17:21:27 +00:00
  • 42684be21b Wire up CPU adaptation in WebRtcVideoEngine2. pbos@webrtc.org 2014-10-03 11:25:45 +00:00
  • 31b75eae05 Moves xmllite's unittests to rtc_unittest. henrike@webrtc.org 2014-10-02 18:43:47 +00:00
  • 25cc745d6b Switch to SW video decoder on Android after getting 2 or more critical errors from HW decoder. glaznev@webrtc.org 2014-10-02 16:58:05 +00:00
  • 4b133da5fd Let RtpFileSource use RtpFileReader henrik.lundin@webrtc.org 2014-10-02 08:19:38 +00:00
  • 348eac641e audio_processing: Replaced WEBRTC_SPL_RSHIFT_U32 with >> bjornv@webrtc.org 2014-10-02 08:07:05 +00:00
  • 5fa8c458d8 Remove mouse cursor capturer from the ScreenCapturer interface sergeyu@chromium.org 2014-10-02 01:47:10 +00:00
  • 6138f0f89d Revert "Remove mouse cursor capturer from the ScreenCapturer interface" sergeyu@chromium.org 2014-10-02 01:36:43 +00:00
  • 1fced0f2aa Remove mouse cursor capturer from the ScreenCapturer interface sergeyu@chromium.org 2014-10-02 00:18:10 +00:00
  • 76819d315d Add error trap for XFixesGetCursorImage() sergeyu@chromium.org 2014-10-01 23:07:12 +00:00
  • 325cff01b4 Import LappedTransform and friends. andrew@webrtc.org 2014-10-01 17:42:18 +00:00
  • 593c3a0868 rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation. henrike@webrtc.org 2014-10-01 16:33:03 +00:00
  • 4530b2ca48 Revert 7355 "Fix parallelization in libjingle_p2p_unittest." henrike@webrtc.org 2014-10-01 15:43:55 +00:00
  • 36b0c1afae Adds PRESUBMIT.py dispensation for depending on rtc_base. henrike@webrtc.org 2014-10-01 14:40:58 +00:00
  • fd29205e6e Fix parallelization in libjingle_p2p_unittest. pbos@webrtc.org 2014-10-01 12:31:31 +00:00
  • c86e45d7c4 Fix parallelizability in modules_tests. pbos@webrtc.org 2014-10-01 10:05:40 +00:00
  • 4cebd84c79 Reland "Remove DTMF status methods from Voice Engine" r7276 henrik.lundin@webrtc.org 2014-10-01 08:23:21 +00:00
  • 4e4fe4f9ae Add support for MSan kjellander@webrtc.org 2014-10-01 08:03:19 +00:00
  • afefed5c93 Update checkdeps.py rules in DEPS kjellander@webrtc.org 2014-10-01 06:03:47 +00:00
  • 83fe69da95 Added presubmit protecting against inclusion of rtc_base, while allowing rtc_base_approved. henrike@webrtc.org 2014-09-30 21:54:26 +00:00
  • 3037bc3447 GN: Add common configs to tools and test. kjellander@webrtc.org 2014-09-30 19:07:58 +00:00
  • b8caf6a504 GN: Enable libvpx, add link target and convert some test targets kjellander@webrtc.org 2014-09-30 18:05:02 +00:00
  • d05756f0a2 Changed mips_arch_variant variable value corresponding to Chromium code changes. andrew@webrtc.org 2014-09-30 15:53:24 +00:00
  • 79a7148108 Revert 7337 "Reland 28629004: adding new AEC dump start interfac..." xians@webrtc.org 2014-09-30 15:29:13 +00:00
  • 7aad5e5cce Revert 7338 "Fixed the android build by making the interface pur..." xians@webrtc.org 2014-09-30 15:26:15 +00:00
  • d0bb5862f5 Collecting stats every fixed time in webrtc_video_streaming.js test and prepare the format these collected stats to be plotted using one of external dev-tools. houssainy@google.com 2014-09-30 15:20:15 +00:00
  • db75a66b0f Minor code change to fix some warnings in MIPS build. andrew@webrtc.org 2014-09-30 15:17:50 +00:00
  • 90d1979d77 Fixed the android build by making the interface pure virtual. xians@webrtc.org 2014-09-30 15:15:22 +00:00
  • 14092e00f1 Reland 28629004: adding new AEC dump start interface for chrome xians@webrtc.org 2014-09-30 14:35:15 +00:00
  • 792d1a0541 Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest. henrike@webrtc.org 2014-09-30 14:21:10 +00:00
  • 875206196c Revert 7334 "adding new AEC dump start interface for chrome." xians@webrtc.org 2014-09-30 13:30:05 +00:00
  • 2e417d6428 adding new AEC dump start interface for chrome. xians@webrtc.org 2014-09-30 13:11:27 +00:00
  • 38c121c484 Minor modifications to test::RtpFileReader henrik.lundin@webrtc.org 2014-09-30 11:08:44 +00:00
  • 1795c358fc Add default implementation of Add/RemoveObserver. pbos@webrtc.org 2014-09-30 09:45:25 +00:00
  • 65e56dba53 audio_processing/aecm: Added help function for calculating log of energy bjornv@webrtc.org 2014-09-30 09:31:28 +00:00
  • 23ec8372a6 audio_processing: Removed usage of macro WEBRTC_SPL_MUL bjornv@webrtc.org 2014-09-30 09:29:28 +00:00
  • 750423c722 audio_processing: Replaced trivial macro WEBRTC_SPL_LSHIFT_W32 with << bjornv@webrtc.org 2014-09-30 09:26:36 +00:00
  • 8cad9432d5 Revert 7327 "Update isolate.gypi files + link to isolate_driver.py" kjellander@webrtc.org 2014-09-30 08:44:00 +00:00
  • 02cd3067d2 Update isolate.gypi files + link to isolate_driver.py kjellander@webrtc.org 2014-09-30 08:34:57 +00:00
  • 359d720004 Allow Android apps to set video renderer scaling type. Also add type check for EGL context object received from apps and switch to byte buffer video decoding if EGL context is incorrect glaznev@webrtc.org 2014-09-29 23:07:08 +00:00
  • 7dfb7fa189 Reland disallowing blocking calls on the worker thread. This fixed the issue that invoking the call when the thread is not started. jiayl@webrtc.org 2014-09-29 22:45:55 +00:00
  • ea6c12e59f Set thread scheduling parameters inside the new thread. henrike@webrtc.org 2014-09-29 18:25:27 +00:00
  • 626624061e Disable flaky tests: JsepPeerConnectionP2PTestClient.ReceivedBweStatsCombined JsepPeerConnectionP2PTestClient.ReceivedBweStatsNotCombined asapersson@webrtc.org 2014-09-29 14:30:07 +00:00
  • e794c36637 Fix parallel test execution for tools, testsupport and metrics tests. kjellander@webrtc.org 2014-09-29 11:47:28 +00:00
  • d71118194f audio_processing: Replaced macro WEBRTC_SPL_LSHIFT_W16 with << bjornv@webrtc.org 2014-09-29 10:56:27 +00:00
  • 7c15510f38 common_audio refactoring: Removed macro WEBRTC_SPL_LSHIFT_U32 bjornv@webrtc.org 2014-09-29 09:40:38 +00:00
  • 24f62e1a28 Adding getStats function to the exposed PeerConnection in RtcBot houssainy@google.com 2014-09-29 09:36:28 +00:00
  • 730d270771 Remove callback from RtpDepacketizer::Parse(). pbos@webrtc.org 2014-09-29 08:00:22 +00:00
  • f21ea918ad GN: Add common configs to all targets. kjellander@webrtc.org 2014-09-28 17:37:22 +00:00
  • 34f2a9ea72 Initialize SSL in unittest_main.cc. pbos@webrtc.org 2014-09-28 11:36:45 +00:00
  • 3a10d2f64f Roll chromium_revision deaf2f7e..c264a056 (295079:297113) kjellander@webrtc.org 2014-09-28 10:33:45 +00:00
  • 6c6680a9d4 Cleanup .gclient.bot_entries to avoid sync problems on bots. kjellander@webrtc.org 2014-09-27 18:41:03 +00:00
  • 3902054b58 Roll chromium_revision 6455c69..deaf2f7 (293954:295079) kjellander@webrtc.org 2014-09-27 18:10:30 +00:00
  • bebc75e8bd Fix the duplicated candidate problem when using multiple STUN servers. jiayl@webrtc.org 2014-09-26 23:01:11 +00:00
  • 0a256acb67 Getting orientation is not working properly. VideoCaptureImpl::RotationFromDegrees returns -1 in case fails not 0. So we need to change the if statement. braveyao@webrtc.org 2014-09-26 22:50:06 +00:00
  • 5d0071fb1f Build one of NSS or BoringSSL but not both. pthatcher@webrtc.org 2014-09-26 18:53:40 +00:00
  • a21d071607 Reverting part of https://webrtc-codereview.appspot.com/15089004/diff/140001/talk/session/media/channelmanager.cc?context=10&column_width=80 because of a major regression hanging the executable on start. thorcarpenter@google.com 2014-09-26 17:19:14 +00:00
  • 1fd362c31e Do not assert for blocking call allowed in Thread::Join. We do not allow blocking call from the worker thread, but on Android the worker thread may stop/join a SignalThread, which hits the assert. AssertBlockingIsAllowedOnCurrentThread is used to make sure a thread does not do Invoke, so check that in Thread::Join does not seem to add much value. jiayl@webrtc.org 2014-09-26 16:57:07 +00:00
  • 384d05f362 Remove the different block lengths in ns_core aluebs@webrtc.org 2014-09-26 14:41:19 +00:00
  • 5088377d70 Revert 7297 "Remove the different block lengths in ns_core" aluebs@webrtc.org 2014-09-26 14:33:08 +00:00
  • ca110b808f Mark virtual overrides of ViENetwork and VoENetwork as such. henrikg@webrtc.org 2014-09-26 11:09:08 +00:00
  • 8b2e50cf83 Revert 7302 "Roll chromium revision: 6455c69:2687a76" marpan@webrtc.org 2014-09-25 22:20:11 +00:00
  • bfacaabfce Add accessors for array of channel pointers in AudioBuffer. They are needed as arguments to any multichannel audio processing unit. claguna@google.com 2014-09-25 20:52:08 +00:00
  • b38959ee0c Roll chromium revision: 6455c69:2687a76 marpan@webrtc.org 2014-09-25 20:28:08 +00:00
  • f1d751c7de Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup. jiayl@webrtc.org 2014-09-25 16:38:46 +00:00
  • 05305116d6 Explicitly initialize SSL for tests. pbos@webrtc.org 2014-09-25 15:50:26 +00:00
  • 61e811faa0 Bump to version 39 tnakamura@webrtc.org 2014-09-25 15:28:20 +00:00
  • 60fbd65482 Removing error triggered for disabling FEC on non-opus minyue@webrtc.org 2014-09-25 14:36:30 +00:00
  • 5f3965783b Remove the different block lengths in ns_core aluebs@webrtc.org 2014-09-25 13:53:43 +00:00
  • 741711a861 Revert r7049/r7123, which added unnecessary "u"s to "return 0"s. henrik.lundin@webrtc.org 2014-09-25 07:38:14 +00:00
  • 315669939a Fix typo from RtpPacketizerH264. pbos@webrtc.org 2014-09-25 07:31:42 +00:00
  • 37e1846d73 Revert "Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup." (rev 7293). andresp@webrtc.org 2014-09-25 07:30:14 +00:00