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  • e9f36d55da Make sure min and max bitrate is always set for audio. stefan 2017-01-24 08:18:45 -08:00
  • e256bc582a Delete left-over using declaration. nisse 2017-01-24 07:43:16 -08:00
  • a388310af2 Added api/webrtcsdp.h forwarding header to work around upstream projects. ossu 2017-01-24 07:13:59 -08:00
  • 9aa3f0a200 Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) mbonadei 2017-01-24 06:58:22 -08:00
  • b54c63ffb0 Moving no_op_function.cc out of webrtc/build mbonadei 2017-01-24 05:25:12 -08:00
  • dabbea670c Moving whitespace file up by one folder mbonadei 2017-01-24 05:19:42 -08:00
  • 69dc7dbe24 Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) mbonadei 2017-01-24 05:14:35 -08:00
  • e6b4723790 Statically linked zxing. Without dependency on libMagick mandermo 2017-01-24 05:02:35 -08:00
  • 4b7c952376 Reland of "Log audio network adapter decisions in event log." minyue 2017-01-24 04:54:59 -08:00
  • 35a32700fc Moving webrtc.gni up one level from build/ mbonadei 2017-01-24 04:49:35 -08:00
  • 62d02c328d Unit test out of band H264 SPS,PPS within RtpStreamReceiver. johan 2017-01-24 04:38:27 -08:00
  • 822d2586fb Move webrtc/build/android -> tools-webrtc/android mbonadei 2017-01-24 04:22:11 -08:00
  • 81eab61172 Count FlexFEC packets in |fec_bitrate_| in RTPSenderVideo. brandtr 2017-01-24 04:06:09 -08:00
  • 0608ffdcb4 Roll chromium_revision 59592eaa98..319b885718 (445345:445689) buildbot 2017-01-24 03:08:53 -08:00
  • 365aebdade Make CongestionController::remote_bitrate_estimator_ a non-pointer. nisse 2017-01-24 02:53:38 -08:00
  • d2b092f38a Reland of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. johan 2017-01-24 02:38:17 -08:00
  • 15389c034d Drop pacer and retransmission_rate_limiter from RtpStreamReceiver constructor. nisse 2017-01-24 02:36:58 -08:00
  • 568c9e72d1 New simulators to test BWE at low bitrates (15-50kbps range). terelius 2017-01-24 02:03:31 -08:00
  • a4a753857e Android: Script for building libwebrtc.aar. sakal 2017-01-24 01:25:50 -08:00
  • e04064deb2 Revert of Delete unused class/template ScopedMessageData. (patchset #1 id:1 of https://codereview.webrtc.org/2652663002/ ) aleloi 2017-01-24 01:20:39 -08:00
  • dc2b3f3b9f Delete unused class CompositeMediaEngineWithFakeVoiceEngine. nisse 2017-01-24 00:54:59 -08:00
  • d83fb921a8 Delete unused class/template ScopedMessageData. nisse 2017-01-24 00:36:38 -08:00
  • c23b0b26df Delete unused classes DesktopId and ScreencastEventCatcher. nisse 2017-01-24 00:03:32 -08:00
  • ad452287b0 Moving get_landmines.py (build/ -> tools-webrtc/) mbonadei 2017-01-24 00:01:49 -08:00
  • 2b7552663e Add linux_memcheck as default trybot. Henrik Kjellander 2017-01-24 05:22:40 +01:00
  • 914d49d0fd Revert of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. (patchset #3 id:40001 of https://codereview.webrtc.org/2638933002/ ) kjellander 2017-01-23 20:16:58 -08:00
  • 1b54a5f018 Relanding: Removing #defines previously used for building without BoringSSL/OpenSSL. deadbeef 2017-01-23 19:39:57 -08:00
  • 4c78702d12 iOS: Add MedianSlopeFilter field trial. tkchin 2017-01-23 11:24:57 -08:00
  • 5c4f24a141 Move implmentation specific constants out of rtp_header_extension.h danilchap 2017-01-23 11:10:20 -08:00
  • f53d7374cf H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. johan 2017-01-23 09:29:33 -08:00
  • e1405ad0d1 Removed double-special-casing of ISAC in libjingle and WebRtcVoE. ossu 2017-01-23 08:55:48 -08:00
  • cb893ee634 Removing unused code from webrtc/build mbonadei 2017-01-23 08:49:12 -08:00
  • 1bed2e486e video_loopback: fall back to fake capturer if we can't open camera. sprang 2017-01-23 08:46:51 -08:00
  • 435ddf978d Add TransportFeedbackPacketLossTracker. minyue 2017-01-23 08:07:05 -08:00
  • ed582f7e36 Script to start stubbed loopback video test with Espresso mandermo 2017-01-23 07:55:42 -08:00
  • 0ebdf2757c Delete or update left-over ASSERT use and comments. nisse 2017-01-23 07:43:05 -08:00
  • da25006431 Fixed public_deps for libjingle_peerconnection{,_api} ossu 2017-01-23 07:37:43 -08:00
  • 50cfe1fda7 RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector. hbos 2017-01-23 07:21:55 -08:00
  • 9c3d4c4d88 Stop leaking FlexfecReceiveStream objects after call shutdown. brandtr 2017-01-23 06:59:13 -08:00
  • a067013e90 Minor style change suggested by internal static analysis tool. aleloi 2017-01-23 06:09:35 -08:00
  • 7bb87ee4e8 Create //webrtc/api:libjingle_peerconnection_api + refactorings. ossu 2017-01-23 04:56:25 -08:00
  • f49ff260d1 GN: Make audio_processing_unittests compile with rtc_enable_protobuf=false ehmaldonado 2017-01-23 04:26:02 -08:00
  • fd870db0b2 Add metric for decode time and max decode time in video quality tests. philipel 2017-01-23 03:22:15 -08:00
  • 011240333e Minor style change suggested by internal static analysis tool. aleloi 2017-01-23 03:10:39 -08:00
  • a31cdbce13 Roll chromium_revision dcc5978539..59592eaa98 (445328:445345) buildbot 2017-01-23 03:09:34 -08:00
  • 0b56279da1 Catch failure to load native dependencies. sakal 2017-01-23 03:02:56 -08:00
  • de8ca92755 New script to count usage of C++ classes. nisse 2017-01-23 02:41:08 -08:00
  • b55bd97972 Reland of Creating libwebrtc bundle jar (patchset #1 id:1 of https://codereview.webrtc.org/2640023010/ ) mbonadei 2017-01-23 01:25:53 -08:00
  • 5d0f2e8753 Roll chromium_revision 269b6bc66e..dcc5978539 (445317:445328) buildbot 2017-01-22 23:26:44 -08:00
  • c1524347e9 Roll chromium_revision 7649e76842..269b6bc66e (445027:445317) buildbot 2017-01-22 21:03:34 -08:00
  • 3e4faae0ed Fixing memory leak in FakeTransportController. deadbeef 2017-01-20 22:43:34 -08:00
  • 8662f94023 Only set certificate on DTLS transport if fingerprint is found in SDP. deadbeef 2017-01-20 21:20:51 -08:00
  • 2197e91860 Remove dead code for GtkVideoRenderer. pbos 2017-01-20 18:13:07 -08:00
  • f33491ebaf Revert of Removing #defines previously used for building without BoringSSL/OpenSSL. (patchset #2 id:20001 of https://codereview.webrtc.org/2640513002/ ) deadbeef 2017-01-20 17:01:45 -08:00
  • eaa826c2ee Removing #defines previously used for building without BoringSSL/OpenSSL. deadbeef 2017-01-20 15:15:58 -08:00
  • cd3180c0c6 PATENTS: fix reference philipp.hancke 2017-01-20 12:45:07 -08:00
  • 7bcdb69957 Ignore ufrag/password in "a=candidate" lines in SDP. deadbeef 2017-01-20 12:43:58 -08:00
  • 0fc04b74c5 Finalize the support for building WebRTC library for iOS with bitcode VladimirTechMan 2017-01-20 08:01:36 -08:00
  • f64941f1a5 RTCMediaStreamTrackStats.framesDecoded collected. hbos 2017-01-20 07:39:09 -08:00
  • aea1a017ed Move webrtc/sdk/DEPS to webrtc/sdk/objc/DEPS magjed 2017-01-20 06:56:44 -08:00
  • 3c9151b953 Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ ) mbonadei 2017-01-20 06:48:03 -08:00
  • a62a82b7e7 Creating libwebrtc bundle jar mbonadei 2017-01-20 06:15:34 -08:00
  • fefe076789 RTCMediaStreamTrackStats.framesSent collected by RTCStatsCollector. hbos 2017-01-20 06:14:25 -08:00
  • 2d4d653e1f Fix msan flake in rtcstats_integrationtest.cc. hbos 2017-01-20 04:16:41 -08:00
  • c854ac3755 Stop camera onStop instead of onPause. sakal 2017-01-20 04:09:11 -08:00
  • 42f6d2fb6c RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector. hbos 2017-01-20 03:56:50 -08:00
  • 7319f26632 Roll chromium_revision 780d18a4ff..7649e76842 (445004:445027) buildbot 2017-01-20 03:28:46 -08:00
  • 30fe5e0d10 Prevent downstream linter warnings. sakal 2017-01-20 03:10:08 -08:00
  • 35564065ca Camera1Session: Fix camera sometimes getting stopped twice. sakal 2017-01-20 03:09:03 -08:00
  • 9e30274c03 RTCMediaStreamTrackStats collected on a per-attachment basis. hbos 2017-01-20 02:47:10 -08:00
  • fd6c94d311 Allow more config changes for CallActivity. sakal 2017-01-20 02:36:49 -08:00
  • 3e9229045e Load library dependencies in AppRTCMobile. sakal 2017-01-20 01:46:50 -08:00
  • be850e1b1d Clear out cached codecs when calculating new codec lists. noahric 2017-01-20 01:07:26 -08:00
  • 204030a256 Roll chromium_revision bdeae63b37..780d18a4ff (444971:445004) buildbot 2017-01-19 23:59:52 -08:00
  • a01d2f5cf7 Roll chromium_revision 34215edf2e..bdeae63b37 (444898:444971) buildbot 2017-01-19 21:00:21 -08:00
  • 888874f761 Allow GCC 4.9 to compile Chromium floppymaster 2017-01-19 20:20:45 -08:00
  • 8944ab3fec Roll chromium_revision 1a7fcf6220..34215edf2e (444851:444898) buildbot 2017-01-19 17:44:09 -08:00
  • b2cdd93fd6 Remove the dependency of TransportChannel and TransportChannelImpl. zhihuang 2017-01-19 16:54:25 -08:00
  • 9d643e87af Roll chromium_revision 113278e435..1a7fcf6220 (444801:444851) buildbot 2017-01-19 14:36:54 -08:00
  • 537798b7e9 Roll chromium_revision d50ce8a895..113278e435 (444743:444801) buildbot 2017-01-19 12:08:34 -08:00
  • 9410b51037 GN: Add audio_conference_mixer_unittests to modules_unittests. ehmaldonado 2017-01-19 11:57:56 -08:00
  • d74886350e Fix PseudoTcp to handle incoming packets with invalid SEQ field sergeyu 2017-01-19 10:53:35 -08:00
  • 3cd896c962 Roll chromium_revision 6d8c754784..d50ce8a895 (444712:444743) buildbot 2017-01-19 09:05:45 -08:00
  • eef94d9995 Video collected by VideoFileRenderer is first saved on the native heap, then saved to disk during release. mandermo 2017-01-19 09:02:29 -08:00
  • 3626865be2 GN: Refactor modules_unittests to eliminate package boundary violations. ehmaldonado 2017-01-19 08:27:11 -08:00
  • d32bf75721 Pass SdpAudioFormat through Channel, without converting to CodecInst kwiberg 2017-01-19 07:03:59 -08:00
  • 093dac142b Roll chromium_revision 0f65d3f753..6d8c754784 (444698:444712) buildbot 2017-01-19 05:58:07 -08:00
  • b93598465d Revert of Move congestion controller processing to the pacer thread. (patchset #5 id:80001 of https://codereview.webrtc.org/2637783003/ ) nisse 2017-01-19 05:41:25 -08:00
  • 6ce9259cb0 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) zhihuang 2017-01-19 04:49:47 -08:00
  • daeffb299b Roll chromium_revision be0566f991..0f65d3f753 (444668:444698) buildbot 2017-01-19 03:31:17 -08:00
  • 5aed06c8d3 make the DtlsTransportWrapper inherit form DtlsTransportInternal zhihuang 2017-01-19 01:48:02 -08:00
  • 04926b8264 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ ) kjellander 2017-01-19 00:06:17 -08:00
  • f847178285 Roll chromium_revision adc103db18..be0566f991 (444630:444668) buildbot 2017-01-18 23:37:22 -08:00
  • 4b9646623e Roll chromium_revision e9762587b9..adc103db18 (444575:444630) buildbot 2017-01-18 21:02:11 -08:00
  • f15825fbbe Roll chromium_revision 0bc260f9e8..e9762587b9 (444497:444575) buildbot 2017-01-18 18:05:41 -08:00
  • 3078b551ad Reduce the log verbosity in sslstreamadapter_unittest skvlad 2017-01-18 15:29:35 -08:00
  • d1c0998730 Adding OrtcFactory, and changing UdpTransport to match current plan. deadbeef 2017-01-18 15:16:37 -08:00
  • 27edfbcec4 Roll chromium_revision 10fecf4ab1..0bc260f9e8 (444374:444497) buildbot 2017-01-18 15:00:26 -08:00
  • a3c8c90f1a Roll chromium_revision d9e076c478..10fecf4ab1 (444338:444374) buildbot 2017-01-18 09:00:59 -08:00
  • d99a200fad Adding some features to proxy.h, and restructuring the macros. deadbeef 2017-01-18 08:55:23 -08:00