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• a6c39d9902 Remove unimplemented VideoChannel code. Peter Boström 2016-02-01 19:30:33 +01:00
• 6f7557e9e2 Disable useless BWE tests. stefan 2016-02-01 10:24:19 -08:00
• e37a2d1802 Reland "Removing webrtc::AudioFrame::energy_." minyue 2016-02-01 10:02:40 -08:00
• d8de1154c9 Remove mutable from rtc::CriticalSections. pbos 2016-02-01 09:00:51 -08:00
• 34877eeec9 Revert of Added validation between RTP and RTCP timestamps (patchset #7 id:120001 of https://codereview.webrtc.org/1633843003/ ) danilchap 2016-02-01 08:25:04 -08:00
• 74451a5ea9 Prevent zero division in VCMJitterEstimator. Peter Boström 2016-02-01 16:31:10 +01:00
• b46c333139 Roll chromium_revision 126e210..65f9b34 (372588:372637) kjellander 2016-02-01 04:53:50 -08:00
• bba9dec4d5 Use separate rtp module lists for send and receive in PacketRouter. stefan 2016-02-01 04:39:55 -08:00
• 1f611fa58b Fixed minor issue: added missing semicolons to metric_recorder.cc cesar.ilharco 2016-02-01 03:09:15 -08:00
• d06d4d8dee Add linux_ubsan_vptr to default trybots. kjellander 2016-02-01 02:54:48 -08:00
• e1f2f1fbb8 Unwrap timestamps in VideoAnalyzer sprang 2016-02-01 02:04:52 -08:00
• 44efbece68 Converting picture_id to bitstring pushed from WithPictureId to Create function. Added Parse and accessor functions. danilchap 2016-02-01 01:36:37 -08:00
• 6b231e0719 Roll chromium_revision d785e7c..126e210 (372580:372588) kjellander 2016-01-31 20:35:57 -08:00
• 25f17d762a Roll chromium_revision 2084e1d..d785e7c (372575:372580) kjellander 2016-01-31 13:07:56 -08:00
• ac53c883a5 Roll chromium_revision 750447f..2084e1d (372566:372575) kjellander 2016-01-31 04:12:45 -08:00
• 430e40084d Roll chromium_revision f5d1a9c..750447f (372546:372566) kjellander 2016-01-30 20:36:08 -08:00
• 3f70562bbb Fix WebRtc ninja x86 build using Visual Studio 2015 (set GYP_MSVS_VERSION=2015). conceptgenesis 2016-01-30 14:40:44 -08:00
• c97c886e3a Roll chromium_revision f07b6b8..f5d1a9c (372524:372546) kjellander 2016-01-30 04:06:26 -08:00
• f0269a6a15 Roll chromium_revision e9e4e90..f07b6b8 (372389:372524) kjellander 2016-01-29 20:19:12 -08:00
• eee86a6aa3 Add option to disable particular HW video codec from app. Alex Glaznev 2016-01-29 14:17:07 -08:00
• 9dfed79f3f Stop processing any incoming packets when turn port is disconnected. honghaiz 2016-01-29 13:22:31 -08:00
• 083b8e91a9 Roll chromium_revision 3784ca9..e9e4e90 (372326:372389) kjellander 2016-01-29 12:35:12 -08:00
• de13882d94 rtcp::ExtenededReports packet class got Parse function danilchap 2016-01-29 11:26:14 -08:00
• ff63ed2888 Format changes achieved by running clang-format -i -style=Chromium peah 2016-01-29 07:46:13 -08:00
• f5b804bb9c Fix implicit bool casts in producer_fec_fuzzer.cc. Peter Boström 2016-01-29 16:26:43 +01:00
• 3a8cac8410 Roll chromium_revision 105cb5f..3784ca9 (372268:372326) kjellander 2016-01-29 04:06:48 -08:00
• b163c3f1ba Delete unused members from VideoOptions nisse 2016-01-29 01:14:38 -08:00
• a37babebfa Roll chromium_revision ffa6c99..105cb5f (372122:372268) kjellander 2016-01-28 20:06:58 -08:00
• 378dc770a6 Consolidate setters into SetRecvParameters. pbos 2016-01-28 15:58:41 -08:00
• 5e8351b325 Prevent division-by-zero in VCMFecMethod. Peter Boström 2016-01-28 23:55:29 +01:00
• 46eed76207 Removing "candidates" attribute from TransportDescription. deadbeef 2016-01-28 13:24:37 -08:00
• e8f0836c58 Roll chromium_revision ea1b30c..ffa6c99 (371978:372122) kjellander 2016-01-28 12:07:04 -08:00
• fb152707ed Replace const-reference with pointer in SendData. Peter Boström 2016-01-28 19:11:58 +01:00
• f4b9c77512 Changed test to validate rtp timstamps not just in RTP packets but also in RTCP Sender Reports. Altered it to accept negative value since it is normal for RTCP packet coming before RTP packet to have slightly later time. danilchap 2016-01-28 06:14:24 -08:00
• 55b97fe388 clang-format -i -style=file webrtc/voice_engine/channel.* kwiberg 2016-01-28 05:22:45 -08:00
• 6043f2e5d6 Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #5 id:80001 of https://codereview.webrtc.org/1581693006/ ) terelius 2016-01-28 05:06:07 -08:00
• e73afbaf17 New rtc::VideoSinkInterface. nisse 2016-01-28 04:47:08 -08:00
• 533a4e4882 Switch critical section locks out for atomic operations tommi 2016-01-28 02:44:12 -08:00
• bec70ab0fd https://github.com/w3c/webrtc-stats/pull/10/files added mediaType to the tracks. The closest in the current stats is the ssrc type. fippo 2016-01-28 01:27:15 -08:00
• 6a062bd7af Deleted method AudioTrackInterface::GetRenderer. nisse 2016-01-28 00:38:10 -08:00
• edc978dcd9 Roll chromium_revision da1acd5..ea1b30c (371832:371978) kjellander 2016-01-27 20:10:43 -08:00
• ab8f82ffe0 Make ECDSA default for RTCPeerConnection tkchin 2016-01-27 17:50:11 -08:00
• 691b8369ff Using buffered signal to calculate the level of echo cancellation. minyue 2016-01-27 15:44:52 -08:00
• d162a5e379 Add shouldDisableBuffering to RTCFileLogger. tkchin 2016-01-27 15:11:47 -08:00
• 919ff75376 Use high QP threshold for HW VP8 encoder frame downscaling. glaznev 2016-01-27 15:01:03 -08:00
• da2183c86f Update API for Objective-C RTCDataChannelConfiguration. hjon 2016-01-27 13:42:28 -08:00
• 08a6eab75e Adding "first packet received" notification to PeerConnectionObserver. Taylor Brandstetter 2016-01-27 13:38:49 -08:00
• d7a75d798b Roll chromium_revision c6ec25c..da1acd5 (371549:371832) kjellander 2016-01-27 13:33:24 -08:00
• 7b3c72ffa9 Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #4 id:60001 of https://codereview.webrtc.org/1581693006/ ) deadbeef 2016-01-27 13:03:42 -08:00
• 42265a8cc3 Adding "first packet received" notification to PeerConnectionObserver. Taylor Brandstetter 2016-01-27 12:10:36 -08:00
• 80f1db971d Include relay protocol type when computing the turn candidate foundation. Honghai Zhang 2016-01-27 11:54:45 -08:00
• 3afc8c40be Consolidate SetSendParameters into one setter. Peter Boström 2016-01-27 16:45:21 +01:00
• ec2922f864 Change PeerConnectionFactory.setVideoHwAccelerationOptions to create shared Egl context for harware encoders and decoders. Before this fix, it was required that the EGL context used as an argument was kept open until all PeerConnections using the context had been closed. With this fix, that is no longer required. Also, if a released EGLContext (EGL_NO_CONTEXT) is used, harware codecs will fallback to use byte buffers for encoding and decoding. BUG=b/26583522 R=glaznev@webrtc.org Per 2016-01-27 15:25:46 +01:00
• 2098fca39a Revert of New rtc::VideoSinkInterface. (patchset #7 id:120001 of https://codereview.webrtc.org/1594973006/ ) nisse 2016-01-27 06:12:49 -08:00
• a862d4563f New rtc::VideoSinkInterface. Niels Möller 2016-01-27 14:40:49 +01:00
• f5dca48dc0 Add transport sequence number on the non-pacer path of the rtp sender. Stefan Holmer 2016-01-27 12:58:51 +01:00
• 1c3909899d Use rtc::time for all your timing needs! Erik Språng 2016-01-27 12:55:33 +01:00
• d673b0fa5d [rtp_rtcp] Fix potentional time difference between rtp and rtcp packets. Danil Chapovalov 2016-01-27 12:54:37 +01:00
• b11e97a552 Move talk/media/webrtc/OWNERS to talk/media. Peter Boström 2016-01-27 12:39:30 +01:00
• 0b518bf6fc Remove incorrect cast to AsyncSocketAdapter. Peter Boström 2016-01-27 12:35:40 +01:00
• bab934bffe H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding. hbos 2016-01-27 01:36:03 -08:00
• fab0a2886d Fix BasicNetworkManager not to spam logs when internet is unreacheable. Sergey Ulanov 2016-01-26 22:12:50 -08:00
• 3ea1852f9e Sync build_ios_libs.sh script with http://webrtc.org/native-code/ios/ hjon 2016-01-26 13:43:21 -08:00
• 4cb3e3997b Fix compilation if HAVE_WEBRTC_VIDEO is not defined. jbauch 2016-01-26 13:07:54 -08:00
• 6d49a8ed17 Update API for Objective-C RTCConfiguration. hjon 2016-01-26 13:06:42 -08:00
• 7b582a2e4f Roll chromium_revision 2ca77c1..c6ec25c (371488:371549) kjellander 2016-01-26 12:09:32 -08:00
• a2c55235ca Allow packets to be reordered in the fake network pipe. philipel 2016-01-26 08:41:53 -08:00
• 7fd881743c Fix type of local encoded length variable from uint32_t to size_t. asapersson 2016-01-26 07:26:09 -08:00
• 59b2d3ebe7 Remove zero-divide in VCMContentMetricsProcessing. Peter Boström 2016-01-26 16:18:48 +01:00
• 83277131b1 AudioCodingModuleImpl: Put CodecManager and Rent-A-Codec in a separate struct kwiberg 2016-01-26 06:06:13 -08:00
• d0c7bba4f4 [rtp_rtcp] Dlrr::SubBlock struct renamed to ReceiveTimeInfo Danil Chapovalov 2016-01-26 14:12:46 +01:00
• 5c7f110ec6 Roll chromium_revision fb2e77c..2ca77c1 (371273:371488) kjellander 2016-01-26 04:12:47 -08:00
• 6a07f12e95 AudioCodingModuleImpl: Initialize encoder_stack_ to nullptr kwiberg 2016-01-26 03:28:51 -08:00
• 2bdcfadc8a Revert of Removing webrtc::AudioFrame::energy_. (patchset #2 id:20001 of https://codereview.webrtc.org/1589953002/ ) terelius 2016-01-26 02:29:50 -08:00
• ffa3fdc8d6 Reallocate encoded buffer size if needed for VP8. Initially set to the input image size. Issue may occur for very small input images (e.g. 4x4) when encoded image length > input image size. asapersson 2016-01-26 01:56:32 -08:00
• e791ffd638 Remove non-monotonic clock support sprang 2016-01-26 01:53:20 -08:00
• 4fd6cda067 Add tracing to VCMGenericEncoder::Release. Peter Boström 2016-01-26 10:19:53 +01:00
• 86956ded42 Small cleanup in VP9EncoderImpl::GetEncodedLayerFrame. asapersson 2016-01-26 01:05:20 -08:00
• bacae81a1d Remove webrtc::AudioFrame::energy_. minyue 2016-01-25 14:10:36 -08:00
• 58a80b5b12 Roll chromium_revision 717238e..fb2e77c (370438:371273) kjellander 2016-01-25 12:21:43 -08:00
• 85b22e2306 Remove vp8_factory.{cc,h}. Peter Boström 2016-01-25 17:58:00 +01:00
• b332e5d3bd Roll chromium_revision 6a04368..717238e (370362:370438) + tcmalloc primiano 2016-01-25 08:13:05 -08:00
• 28ba92731d Switch to use new implementation in metrics.h. Sparse macro replaced for all video histograms that have a constant name. asapersson 2016-01-25 05:58:23 -08:00
• 5ad935cb56 Remove mutable from rtc::CriticalSection members. pbos 2016-01-25 03:52:44 -08:00
• 7d0d0e0763 Remove dead code from webrtc/base/timing.* tommi 2016-01-25 03:09:29 -08:00
• 9de632a100 Deleted unused enums MediaChannelOptions and VoiceMediaChannelOptions, with constants OPT_CONFERENCE and OPT_AGC_MINUS_10DB. nisse 2016-01-25 01:56:50 -08:00
• 7a83951b27 Fix a bug in webrtc::ByteReader henrik.lundin 2016-01-24 23:47:51 -08:00
• f91e6d0438 Enable cpplint for webrtc/modules/bitrate_controller and fix all uncovered cpplint errors. jbauch 2016-01-24 23:05:21 -08:00
• e373dc20c4 Update API for Objective-C RTCDataChannel. hjon 2016-01-22 14:04:27 -08:00
• 38b39d59d1 Temporary hack to avoid assert errors when time moves backwards. sprang 2016-01-22 13:36:32 -08:00
• cc71c4107f Revert "Disable P2PTransport...TestFailoverControlledSide on Memcheck" tnakamura 2016-01-22 12:45:02 -08:00
• 0a37497842 Deleted unused method SetDumpPath and unneeded includes. nisse 2016-01-22 11:56:53 -08:00
• c8930ba8ac Disable WebRtcSessionTest.TestStunError on Win. minyue 2016-01-22 06:17:46 -08:00
• 9846845da6 Calculate audio levels in AEC in time domain. minyue 2016-01-22 05:46:43 -08:00
• 5447934728 Remove implementation of CriticalSectionWrapper and use rtc::CriticalSection While transitioning over to rtc::CriticalSection completely, this gives perf benefits that rtc::CriticalSection has on Mac to current users of CriticalSectionWrapper. tommi 2016-01-22 05:26:14 -08:00
• 7406b96abc CriticalSection: Use types+methods from base/platform_thread*.*. Use PlatformThreadRef, CurrentThreadRef and IsThreadRefEqual instead of pthread_t, pthread_self and operator== (or !=). tommi 2016-01-22 05:13:33 -08:00
• 32e590ec13 class doesn't rely on structures in RTCPUtility to store data. supports several fci items in same packet. got accessors to read data Danil Chapovalov 2016-01-22 11:04:56 +01:00
• 3fe2c6a161 VideoProcessorImpl using EncodedImage::GetBufferPaddingBytes. hbos 2016-01-22 00:07:12 -08:00
• ed281e9c9b New lock implementation for mac. According to my measurements, it's about 100x faster than the native mutex implementation in OSX. Google "OSX mutex performance" for more info. tommi 2016-01-21 23:47:25 -08:00
• 2bf9a5f11b Update API for Objective-C RTCMediaStream. Jon Hjelle 2016-01-21 16:14:11 -08:00