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  • 9b9910dda7 Roll chromium_revision 3d5a0fb164..c8b7ee41e0 (425604:425645) buildbot 2016-10-17 03:58:35 -07:00
  • 7e76560c35 Enable logging to console in DirectRTCClientTest. sakal 2016-10-17 03:17:09 -07:00
  • 2f255d8d67 Replace const -> constexpr for rtcp Packet Type for consistency with other rtcp packet classes. danilchap 2016-10-17 02:07:54 -07:00
  • c1f40b7bae Remove RtcpPacket dependency on rtcp_utility and thus IP_PACKET_SIZE constant: Build() use BlockLength() instead of constant IP_PACKET_SIZE for packet capacity, adding extra checks about packet generation in tests. Build(callback) removed as unused. definitions reordered to follow style. danilchap 2016-10-17 01:44:44 -07:00
  • 27c3d5b652 Restore thread name consistency for webrtc/p2p/ . johan 2016-10-17 00:54:57 -07:00
  • 883ad662f4 Removed the deprecated audioproc executable peah 2016-10-17 00:08:51 -07:00
  • e40a7ee007 GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. kjellander 2016-10-16 23:56:12 -07:00
  • 671534e76d Roll chromium_revision 90b4cf429c..3d5a0fb164 (425384:425604) buildbot 2016-10-16 19:46:56 -07:00
  • 9c4b4b47f4 Add path for recovered packets from internal::Call to RtpStreamReceiver. brandtr 2016-10-16 14:10:51 -07:00
  • e5ddf52b97 Delete unused file webrtcvideochannelfactory.h. nisse 2016-10-15 11:29:55 -07:00
  • 285e55816a Removed suppressions for the data race inside the APM that is now fixed. peah 2016-10-14 14:44:45 -07:00
  • 2c4c42217a Roll chromium_revision a3c4a78675..90b4cf429c (425286:425384) buildbot 2016-10-14 11:49:00 -07:00
  • 0d4b129ab2 Roll chromium_revision 00384b2217..a3c4a78675 (425234:425286) buildbot 2016-10-14 03:38:27 -07:00
  • 8f7cc7e77d This CL corrects the emptying of the render queues for the AEC and AECM when these become full to also work when not in debug mode. peah 2016-10-14 03:23:32 -07:00
  • 5d2e58c8fb Roll chromium_revision 61fb879aaf..00384b2217 (425083:425234) buildbot 2016-10-13 19:37:12 -07:00
  • 91902cb6c0 Remove DesktopRegion parameter from DesktopCapturer::Capture. zijiehe 2016-10-13 16:47:49 -07:00
  • 794d535e51 Roll chromium_revision 50c7b3ce18..61fb879aaf (424992:425083) buildbot 2016-10-13 11:37:53 -07:00
  • 9ae585de8d Cleanup of voice_engine includes. aleloi 2016-10-13 06:57:16 -07:00
  • 3283cf917b Add asyncstuntcpsocket_unittest.cc to rtc_unittests kjellander 2016-10-13 06:35:45 -07:00
  • 982bf89444 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) sprang 2016-10-13 06:23:11 -07:00
  • b593bc06ad Suggest myself as owner of api/ solenberg 2016-10-13 04:09:45 -07:00
  • 81b8a0749f Roll chromium_revision 2cabef4e7d..50c7b3ce18 (424936:424992) buildbot 2016-10-13 03:56:49 -07:00
  • 0d8385770b NetEq: Convert AverageIAT from int to float calculations henrik.lundin 2016-10-13 03:35:55 -07:00
  • c9ec8758db NetEq: Remove special case for Merge without Expand henrik.lundin 2016-10-13 02:43:34 -07:00
  • 722b0dc108 Revert of Android audio playout now supports non-call media streams (patchset #3 id:10004 of https://codereview.webrtc.org/2411263003/ ) henrika 2016-10-13 01:12:33 -07:00
  • dd7a1cff68 Landmine due to corrupt .pdb files on Windows. Henrik Kjellander 2016-10-12 21:07:02 -07:00
  • da3303fda0 Revert of Remove tools dir from root webrtc target (patchset #1 id:1 of https://codereview.webrtc.org/2412353004/ ) kjellander 2016-10-12 20:50:17 -07:00
  • 91a5759733 Roll chromium_revision 316b880c55..2cabef4e7d (421519:424936) Henrik Kjellander 2016-10-12 20:25:34 -07:00
  • 163b1a2d0a Remove tools dir from root webrtc target charujain 2016-10-12 20:22:47 -07:00
  • 614f68f112 Remove duplicate entry in webrtc .gn file exec_script_whitelist fbarchard 2016-10-12 18:11:47 -07:00
  • db158f9b9e Fix experiment name in BitrateControllerTest. Stefan Holmer 2016-10-13 02:11:51 +02:00
  • 77c663d0f5 Give FeedbackTimeout experiment the correct name. stefan 2016-10-12 15:57:45 -07:00
  • 12a39f4100 Don't crash on unexpected stap-a or fu-a. stefan 2016-10-12 15:30:14 -07:00
  • 75c8fb4b2c DataChannelInterface default impl of [messages/bytes]_[sent/received]. hbos 2016-10-12 14:48:15 -07:00
  • 84ffdee879 DataChannel[Interface]::[message/bytes]_[sent/received]() added. hbos 2016-10-12 14:14:39 -07:00
  • 73fdc317b2 Several fixes to screen_capturer_mac. erikchen 2016-10-12 12:24:09 -07:00
  • e280cdeb74 Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz. ossu 2016-10-12 11:04:10 -07:00
  • 872f614111 Android audio playout now supports non-call media streams. henrika 2016-10-12 08:11:42 -07:00
  • 116ec6da50 Implemented further mixer interface change suggestions from https://codereview.webrtc.org/2386383003/ aleloi 2016-10-12 06:07:07 -07:00
  • 7e30432b36 Hooking up audio network adaptor to VoE. minyue 2016-10-12 05:00:55 -07:00
  • 917d4e1e71 Removed the legacy behavior of stopping playout when setting new receive codecs. solenberg 2016-10-12 03:20:29 -07:00
  • e97974d203 Cleanup of the mixer interface. aleloi 2016-10-12 03:06:09 -07:00
  • 73a28ee066 The AudioProcessing class is used as an interface to the functionality in the audio processing module. Therefore, it should be a pure interface. This CL ensures that is the case. peah 2016-10-12 03:01:49 -07:00
  • 4b8bfb8ed3 Changed ramping functionality of the AudioMixer. aleloi 2016-10-12 02:14:59 -07:00
  • 3ec3da6fc2 Set screenshare.enabled parameter to false when running video_loopback test. asapersson 2016-10-11 16:20:33 -07:00
  • c6ca544295 Using relative path for GN for iOS. nicholss 2016-10-11 16:12:44 -07:00
  • c47a0c3ac4 RTCIceCandidatePairStats[1] added. hbos 2016-10-11 14:54:49 -07:00
  • dd0e1e0070 GN: Build iOS framework in build_ios_libs.sh Zeke Chin 2016-10-11 13:27:26 -07:00
  • e7c338fed4 Reland of "Remove the obsolete enum webrtc::PeerConnectionInterface::IceState." (patchset #1 id:1 of https://codereview.webrtc.org/2402993002/ ) sprang 2016-10-11 09:04:43 -07:00
  • 716978d075 Revert of Prune connections based on network name. (patchset #3 id:130001 of https://codereview.webrtc.org/2395243005/ ) sprang 2016-10-11 06:43:28 -07:00
  • e89141500a Moved MixerAudioSource and removed audio_mixer_defines.h. aleloi 2016-10-11 06:18:31 -07:00
  • 14acf658ad AudioTransport::NeedMorePlayData is no longer called from different threads using OpenSL ES on Android henrika 2016-10-11 06:15:41 -07:00
  • 99df6c03c3 Fix bug in DTMF generation where events with level > 36 would be ignored. solenberg 2016-10-11 04:35:34 -07:00
  • 113bdcadf3 Make sure VideoReceiveStream can be restarted sprang 2016-10-11 03:10:10 -07:00
  • defc21e0aa Removes usage of hardware AGC and any related APIs on Android. henrika 2016-10-11 01:29:09 -07:00
  • e0729c56d3 Add RtcpRttStats to AudioStream michaelt 2016-10-11 00:29:27 -07:00
  • a8b38559a5 Add a FlexfecReceiver class. brandtr 2016-10-10 16:44:57 -07:00
  • aae2784c1f Prune connections based on network name. Previously we prune connections on the same network pointer. So if an IPv6 and an IPv4 network are on the same network interface, IPv4 connection won't be pruned even if an IPv6 connection with higher priority becomes writable. honghaiz 2016-10-10 16:00:41 -07:00
  • 8ff52cc7bc Remove useless debugging code mattdr 2016-10-10 15:56:56 -07:00
  • 8cab52db48 Fix externalhmac.h/.cc to compile with libsrtp 1 and 2 mattdr 2016-10-10 15:33:37 -07:00
  • 440b4be4b7 Use non-deprecated screen update callbacks. erikchen 2016-10-10 15:18:30 -07:00
  • e606a172d4 Remove deprecated Gestalt methods. erikchen 2016-10-10 11:19:15 -07:00
  • 41aab327ad Fix delay plot crash in event_log_visualizer. stefan 2016-10-10 08:16:30 -07:00
  • ae0b3338e3 Prep to remove APM-related #defines from voice_engine_configurations.h henrik.lundin 2016-10-10 07:24:53 -07:00
  • 57cb873707 Revert of "Remove the obsolete enum webrtc::PeerConnectionInterface::IceState." (patchset #1 id:1 of https://codereview.webrtc.org/2361053003/ ) sprang 2016-10-10 05:59:10 -07:00
  • e0928d8002 Added logging for audio send/receive stream configs. ivoc 2016-10-10 05:12:51 -07:00
  • fc9414ab51 Reland of "Remove the obsolete enum webrtc::PeerConnectionInterface::IceState." johan 2016-10-10 03:25:57 -07:00
  • d36dd499c8 Revert of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #9 id:160001 of https://codereview.webrtc.org/2315663002/ ) kjellander 2016-10-08 22:21:35 -07:00
  • cbfb033f1d MB: Update Android bots after moving to Swarming and client.webrtc.perf Henrik Kjellander 2016-10-09 06:28:58 +02:00
  • 9fa49759e5 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped. - Rename the data codec payload types to end with "PlType" instead of "Id", for consistency. solenberg 2016-10-08 13:02:44 -07:00
  • 9e31cca8f4 iOS: Fix rotation bug for back camera magjed 2016-10-08 02:57:50 -07:00
  • c19f312f54 This CL adds functionality in the level controller to receive a signal level to use initially, instead of the default initial signal level. peah 2016-10-07 14:54:10 -07:00
  • 11a9cbfa50 Refactoring: move ownership of RtcEventLog from Call to PeerConnection skvlad 2016-10-07 11:53:05 -07:00
  • 0d348d69e6 Avoid race in VideoReceiveStream shutdown sprang 2016-10-07 08:28:39 -07:00
  • dda6ec008a Make cricket::VideoFrame inherit webrtc::VideoFrame. Delete all methods but a few constructors. And similarly for the subclass cricket::WebRtcVideoFrame. nisse 2016-10-07 08:16:54 -07:00
  • 0d382efbdc Cleaning build file for audio network adaptor. minyue 2016-10-07 07:59:28 -07:00
  • bf369fe3dd Replace rtcp parser in rtc event log handlers. RTCPUtility is going away. danilchap 2016-10-07 07:39:54 -07:00
  • 05f3ec1356 Fix "left shift of negative value" bug kwiberg 2016-10-07 07:38:48 -07:00
  • d2cf1ce8aa Remove legacy Camera2Enumerator.isSupported method. sakal 2016-10-07 07:06:03 -07:00
  • c7901c6491 Delete macsocketserver.h and related files. nisse 2016-10-07 05:57:21 -07:00
  • df494b0908 Android: Split out EGL rendering from SurfaceViewRenderer to separate class magjed 2016-10-07 05:32:35 -07:00
  • 36542514f6 Reland of https://codereview.webrtc.org/2396483002/ aleloi 2016-10-07 05:28:32 -07:00
  • a485dabc78 Revert of Made MixerAudioSource a pure interface. (patchset #7 id:350001 of https://codereview.webrtc.org/2396483002/ ) aleloi 2016-10-07 05:04:51 -07:00
  • 2ae5fdff86 Made MixerAudioSource a pure interface. aleloi 2016-10-07 04:30:10 -07:00
  • 2334b70851 Add support for component builds on Windows Swarming. Attempt 2 ehmaldonado 2016-10-07 02:47:38 -07:00
  • ab9f6e4dea RTCIceCandidateStats[1] added. hbos 2016-10-07 02:18:47 -07:00
  • 3dcfd64c26 Make bwe_simulator a separate test target that doesnt run on the bots but remove the #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE so that it always builds. terelius 2016-10-07 00:39:56 -07:00
  • 23345594e0 Revert of Add support for component builds on Windows Swarming. (patchset #2 id:100001 of https://codereview.webrtc.org/2389133002/ ) ehmaldonado 2016-10-07 00:31:58 -07:00
  • 66d4ce9324 Add support for component builds on Windows Swarming. ehmaldonado 2016-10-07 00:27:09 -07:00
  • ae04e36f56 Cleanup unused dependency on video_capture_module. perkj 2016-10-07 00:13:21 -07:00
  • e402a14668 Make process interval configurable for MaxCounter class. asapersson 2016-10-06 23:39:15 -07:00
  • 9cc012daf5 GN: Add missing config //webrtc:common_objc for rtc_sdk_common_objc kjellander 2016-10-06 22:18:13 -07:00
  • 7056be937f Delete old video defines in engine config. mflodman 2016-10-07 07:07:28 +02:00
  • 0b4b72797e Use NtpTime in RtcpReceiver instead of pair of uints danilchap 2016-10-06 09:24:45 -07:00
  • 15d8357bab Remove OnLocalSsrcChanged and rename EncoderStateFeedback. mflodman 2016-10-06 08:35:11 -07:00
  • 41b9c801c2 Adding audio network adaptor to AudioEncoderOpus. minyue 2016-10-06 07:13:54 -07:00
  • 81b9291dfd The FFT functionality in aec_rdft* is based on legacy C code which is not thread-safe in the sense that the rdft_init method can only be run in a single-threaded. peah 2016-10-06 06:46:20 -07:00
  • 21a18ee267 Revert of Delete webrtc::VideoFrame::CopyFrame. (patchset #2 id:20001 of https://codereview.webrtc.org/2371363003/ ) ivoc 2016-10-06 06:29:30 -07:00
  • 327e9d0821 Make MediaCodecEncoder fallback to a software encoder on failure. sakal 2016-10-06 05:55:11 -07:00
  • a669a3a0dc Revert "Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )" stefan 2016-10-06 05:04:52 -07:00