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  • aff1c8489f Roll chromium_revision ccef3cb..7779e7d (331232:332119) Henrik Kjellander 2015-06-01 11:49:20 +02:00
  • 5263b3c1dd Add options for NetEq fast accelerate mode through libjingle Henrik Lundin 2015-06-01 10:29:41 +02:00
  • 0908d0dcf2 Fix issue with RTT computations in simulator. Stefan Holmer 2015-06-01 10:20:26 +02:00
  • 9b07368bfb Revert "Roll chromium_revision ccef3cb..7779e7d (331232:332119)" Henrik Kjellander 2015-05-31 21:57:30 +02:00
  • a8d686d174 Roll chromium_revision ccef3cb..7779e7d (331232:332119) Henrik Kjellander 2015-05-31 20:49:54 +02:00
  • f69f1fbc98 Testing and improving NADA algorithm. Cesar Magalhaes 2015-05-30 17:49:18 +02:00
  • 4765070b8d Rename I420VideoFrame to VideoFrame. Miguel Casas-Sanchez 2015-05-29 17:21:40 -07:00
  • c2cb266c93 Match video orientation with device orientation for portrait and portrait upside down Jon Hjelle 2015-05-29 16:38:26 -07:00
  • 7be99bdea1 Revert "Match video orientation with device orientation for portrait and portrait upside down" Zeke Chin 2015-05-29 16:34:38 -07:00
  • 14c2695f29 Match video orientation with device orientation for portrait and portrait upside down Jon Hjelle 2015-05-29 15:24:52 -07:00
  • bc7dd7e023 Add RTCConfiguration constructor to RTCPeerConnection wrapper. Zeke Chin 2015-05-29 14:59:14 -07:00
  • d935f912b1 Don't try to parse empty Ice urls. Joachim Bauch 2015-05-29 22:14:21 +02:00
  • a8202aadd5 Roll chromium_revision 1b9c098..ccef3cb (330302:331232) Henrik Kjellander 2015-05-29 20:13:24 +02:00
  • 5a8bad60ca Update a comment that mentions the nonexistent Reset() method. Wan-Teh Chang 2015-05-29 09:41:38 -07:00
  • 5c6c6e026b Implements TODOs for webrtc::datachannel state management when the SCTP association is congested. Adds missing state variables for each step in the transitions between DataChannelInterface::DataStates (kConnecting, kOpen, etc.), and uses them. Lally Singh 2015-05-29 11:52:39 -04:00
  • c28a896a7b VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation Jelena Marusic 2015-05-29 15:05:44 +02:00
  • bf738d7130 Temporarily disabling OpenSL ES for playout. henrika 2015-05-29 11:42:43 +02:00
  • 04e5b49827 Make maximum SSL version configurable through PeerConnectionFactory::Options Joachim Bauch 2015-05-29 09:40:39 +02:00
  • cc84649389 Add LappedTransform accessors. Michael Graczyk 2015-05-28 18:01:33 -07:00
  • e70028e43f Protect access to shared list of SRTP sessions. Joachim Bauch 2015-05-29 01:20:52 +02:00
  • 9e3cb336d4 AppRTCDemo: check for necessary permissions before starting the call. Alex Glaznev 2015-05-28 15:51:52 -07:00
  • 770cc380eb Don't call CRYPTO_add in BoringSSL. Jiayang Liu 2015-05-28 15:36:29 -07:00
  • 35448372be Disable reusing of ECDHE keys with NSS. Joachim Bauch 2015-05-29 00:29:09 +02:00
  • 5ee9f679a5 Remove webrtcvideoengine.cc. Peter Boström 2015-05-29 00:13:05 +02:00
  • 603175a395 Improve comments. Wan-Teh Chang 2015-05-28 14:10:14 -07:00
  • 7c4e7458b5 Support multiple URLs in PeerConnectionInterface::IceServer Joachim Bauch 2015-05-28 23:06:30 +02:00
  • 45b229cc89 Remove an unnecessary webrtc:: namespace prefix. Wan-Teh Chang 2015-05-28 13:45:28 -07:00
  • 92d9489881 Miscellaneous cleanups in VCMReceiver and its unit tests. Wan-Teh Chang 2015-05-28 13:36:06 -07:00
  • 645299d4e0 Add frequency smoothing to postfilter. Andrew MacDonald 2015-05-28 13:10:18 -07:00
  • d4f769d8fc Stop video candidates getting down to audio. Donald Curtis 2015-05-28 09:48:21 -07:00
  • a743794d06 audio_processing/aecm: Create() now returns a pointer to the object Bjorn Volcker 2015-05-28 15:58:42 +02:00
  • 71861a0e20 Remove GetSendSideDelay from RtpRtcp. Peter Boström 2015-05-28 14:45:36 +02:00
  • 7cd16b03b2 video_processing_unittest: Only create files for visual inspection if the boolean flag 'gen_files' is set. Henrik Boström 2015-05-28 14:44:28 +02:00
  • c3deaa30d5 common_audio/vad: Removes head allocation failure check Bjorn Volcker 2015-05-28 14:30:22 +02:00
  • 796e17237b Fixes crash in WebRtcAudioManager.setCommunicationMode henrika 2015-05-28 14:18:33 +02:00
  • c41fe5d5d0 Force 8 kHz sampling rate on Android emulator. Patrik Höglund 2015-05-28 14:16:36 +02:00
  • 2251d6e174 Remove ViESender. Peter Boström 2015-05-28 14:10:39 +02:00
  • 259bd2034c Report ssrc_groups in GetStats(). Peter Boström 2015-05-28 13:39:50 +02:00
  • 8bb6ea3da9 Reset speech encoder before hooking it up to RED or CNG Karl Wiberg 2015-05-28 13:37:19 +02:00
  • 8051832a9d Adding a new Matlab tool rtpAnalyze Henrik Lundin 2015-05-28 12:37:46 +02:00
  • 3b187b9c0c Removed unnecessary includes of webrtcvideocapturer.h Henrik Boström 2015-05-28 11:43:38 +02:00
  • 11beccd712 Remove external report blocks from RtcpSender and rtp_rtcp interface. Erik Språng 2015-05-28 11:10:25 +02:00
  • 23c2e55479 Remove remaining .mk files. Peter Boström 2015-05-28 11:05:01 +02:00
  • b444b3f0ff Redirect logs to stderr in audioproc_f. Andrew MacDonald 2015-05-27 17:26:03 -07:00
  • 9b720f7016 Add GetChunkLength to LappedTransform. Michael Graczyk 2015-05-27 17:09:47 -07:00
  • fec2c6d7eb Prevent potential double-free if srtp_create fails. Joachim Bauch 2015-05-27 23:41:43 +02:00
  • 10602605f8 Added buildbucket bucket definitions Henrik Kjellander 2015-05-27 23:01:39 +02:00
  • 92fbbb21f8 Switch acm_receiver over to using base/logging.h Tommi 2015-05-27 22:07:35 +02:00
  • 9303eaf512 Don't unnecessarily set mode/category on AVAudioSession. Noah Richards 2015-05-27 10:23:50 -07:00
  • def39883f0 Configure default render delay as 10 ms. Peter Boström 2015-05-27 17:59:11 +02:00
  • cf808d2366 Add new fast mode for NetEq's Accelerate operation Henrik Lundin 2015-05-27 14:33:29 +02:00
  • cbe408aa11 WebRtcVideoCapturer: Getting rid of the |critical_section_stopping_| lock and all of its critical sections. Henrik Boström 2015-05-27 10:11:34 +02:00
  • c065cc797d Clarify boolean flags in neteq_opus_quality_test. Minyue Li 2015-05-27 10:01:10 +02:00
  • c13cacbb39 Remove an unused method in NetEq::Expand Henrik Lundin 2015-05-27 09:23:48 +02:00
  • de4703c5d1 Refactor common_audio/vad: Create now returns the handle directly instead of an error code Bjorn Volcker 2015-05-27 07:22:58 +02:00
  • afef4bfd1c Reland "Adding a test framework for conference mode application in VoE." Minyue 2015-05-27 00:21:18 +02:00
  • a4b7e5e35a Revert "Adding a test framework for conference mode application in VoE." Minyue 2015-05-26 23:21:50 +02:00
  • 6a1ba8c17f Fix coding style nits. Wan-Teh Chang 2015-05-26 14:11:41 -07:00
  • e87d48719f Fix ARM64 detection for VP8 and VP9 wrappers. Stefan Holmer 2015-05-26 22:10:28 +02:00
  • fc052055e9 Adding a test framework for conference mode application in VoE. Minyue 2015-05-26 21:00:40 +02:00
  • 5d55c98cd2 WebRTC 4521: Remove usage of deprecated timezone global variable Guo-wei Shieh 2015-05-26 11:53:29 -07:00
  • 8d3ad82d30 Script for auto-rolling chromium_revision in DEPS. Henrik Kjellander 2015-05-26 19:52:05 +02:00
  • 5a3ebd761c Revert "Remove default encoder/decoders." Peter Boström 2015-05-26 11:44:05 +02:00
  • e14e5f4468 Solve TSan warning about unlocking an unlocked mutex. Brave Yao 2015-05-26 16:29:16 +08:00
  • f09e09c7ee VoE: Remove unused interfaces Jelena Marusic 2015-05-26 10:24:55 +02:00
  • 32c2023fb6 Attempt at fixing error on the Chrome Windows FYI bots. It looks like our basictypes.h file in the overrides folder is including the file it is overriding due to include path precedence (Chrome's is lower than WebRTCs). Tommi 2015-05-25 21:22:13 +02:00
  • 905495cfaa Introduce NetEq::Config::ToString and use it in NetEq's constructor Henrik Lundin 2015-05-25 16:58:41 +02:00
  • e982a70ae3 PRESUBMIT: Fix typo. Henrik Kjellander 2015-05-25 16:33:54 +02:00
  • 54be3e0049 Remove some WebRtcVideoEngine2 unittest stubs. Peter Boström 2015-05-25 15:04:24 +02:00
  • d8399e630f Also provide sample rate when registering decoders Karl Wiberg 2015-05-25 14:39:56 +02:00
  • 323b132f5e Protect ACM decoder buffer in stereo. Minyue 2015-05-25 13:49:37 +02:00
  • 57e5fd2e60 PRESUBMIT: Improve PyLint check and add GN format check. Henrik Kjellander 2015-05-25 12:55:39 +02:00
  • 00aac5aacf Some cleanup for base/logging and base/stream.h Tommi 2015-05-25 11:25:59 +02:00
  • 23edcff7a9 Move base/logging.* to rtc_base_approved. Tommi 2015-05-25 10:45:43 +02:00
  • ee369e4277 Refactoring of AudioTrackJni and AudioRecordJni using new JVM/JNI classes henrika 2015-05-25 10:11:27 +02:00
  • a26c4e5df6 Script to generate CL descriptions when rolling chromium_revision. Henrik Kjellander 2015-05-23 11:55:37 +02:00
  • 0eefb4d5c3 Detach base/logging.* from base/stream.*. This is being done in preparation of moving base/logging.* to rtc_base_approved. base/stream.* has libjingle dependencies that webrtc can't use, so logging.* can't depend on streams. It does look like stream.* isn't used much, so cleaning that up as well as cleaning up usage of the actual stream support (now LogStream) in the logging code, is in order, but I'll leave that to another cl. Tommi 2015-05-23 09:54:07 +02:00
  • 469c2c04aa Make Config::default_value leak instead of having an exit-time destructor. Andrew MacDonald 2015-05-22 17:50:26 -07:00
  • 4bf12eafba Revert "Fix sending wrong candidates down to transportchannel." Alejandro Luebs 2015-05-22 15:32:47 -07:00
  • f65de8483e Fix sending wrong candidates down to transportchannel. Donald Curtis 2015-05-22 14:55:19 -07:00
  • 67b635a47e Fix simulcast_encoder_adapter giving full target_bitrate to the 2nd layer of any simulcast setup during InitEncode. Noah Richards 2015-05-22 14:12:10 -07:00
  • e4cb4e9aae Fix jitter buffer bug around out-of-order packets and non-RTX padding. Noah Richards 2015-05-22 14:03:00 -07:00
  • 477487410a Enable AudioProcessing48kHzSupport by default Alejandro Luebs 2015-05-22 12:00:21 -07:00
  • 3548dd2154 Set local SSRCs on receivers added before senders. Peter Boström 2015-05-22 18:48:36 +02:00
  • 367c868c99 AudioEncoderCng: Handle case where speech encoder is reset Henrik Lundin 2015-05-22 15:13:41 +02:00
  • f761d10393 Update NetEq Quality Test. Minyue Li 2015-05-22 11:22:11 +02:00
  • 915df4fc30 CaptureManager: Don't stop a capturer at UnregisterVideoCapturer if it did not start in the first place. Henrik Boström 2015-05-22 09:43:26 +02:00
  • 9a416bd14e Get rid of unnecessary Terminate() method and worker_thread_ from WebRtcVideoEngine2 Fredrik Solenberg 2015-05-22 09:04:09 +02:00
  • 5af6d47d26 Code style change for quality_scaler. jackychen 2015-05-21 14:11:36 -07:00
  • 98d8cf58ee Hardware VP8 encoding: Use QP as metric for resize. jackychen 2015-05-21 11:12:02 -07:00
  • 5fdcdf66d0 Enable ciphers to get ECDHE with NSS. Joachim Bauch 2015-05-21 18:06:19 +02:00
  • 6f2ef74b42 Keep track of DTLS packet sizes to prevent partial reads. Joachim Bauch 2015-05-21 17:52:01 +02:00
  • a3ba0c7f5a RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits Magnus Jedvert 2015-05-21 17:39:21 +02:00
  • 36a1438a66 Remove ViEFrameProviderBase. Peter Boström 2015-05-21 17:00:24 +02:00
  • af55ccc054 Add RtcpMuxPolicy support to PeerConnection. Peter Thatcher 2015-05-21 07:48:41 -07:00
  • 02ff9117b5 Feature merge request: Add support for iOS http proxy detection Yuriy Shevchuk 2015-05-21 13:50:59 +02:00
  • 523183b4aa Disables AudioDeviceTest.StartStopPlayout for Nexus 9 only henrika 2015-05-21 13:43:08 +02:00
  • 280ed11493 Roll gtest-parallel. Peter Boström 2015-05-21 13:25:28 +02:00
  • 848d524879 Revert "RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits" https://webrtc-codereview.appspot.com/47229004/ Magnus Jedvert 2015-05-21 13:25:24 +02:00
  • 10022cdeae RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits Magnus Jedvert 2015-05-21 11:40:52 +02:00