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  • 0393de4b56 Roll chromium_revision b032878ebd..e438353b8b (480186:480311) kjellander 2017-06-18 13:21:21 -07:00
  • 1a610f15c3 Revert of Opus implementation of the AudioEncoderFactoryTemplate API (patchset #4 id:80001 of https://codereview.webrtc.org/2930243003/ ) charujain 2017-06-18 02:38:58 -07:00
  • eb2d2d31d1 Revert of Opus implementation of the AudioDecoderFactoryTemplate API (patchset #1 id:1 of https://codereview.webrtc.org/2942733003/ ) charujain 2017-06-18 02:37:17 -07:00
  • af6293517f Support building WebRTC without audio and video for Android zhihuang 2017-06-17 22:31:24 -07:00
  • d053fe4ab3 Opus implementation of the AudioDecoderFactoryTemplate API kwiberg 2017-06-17 18:40:52 -07:00
  • fe1aa82c63 Opus implementation of the AudioEncoderFactoryTemplate API kwiberg 2017-06-17 18:23:03 -07:00
  • b8727aebc1 G722 implementation of the AudioEncoderFactoryTemplate API kwiberg 2017-06-17 17:41:59 -07:00
  • b1ed7f09c0 G722 implementation of the AudioDecoderFactoryTemplate API kwiberg 2017-06-17 17:30:09 -07:00
  • 0eacd88568 Templated AudioDecoderFactory kwiberg 2017-06-17 15:45:28 -07:00
  • 19b3a554e8 Fixing incorrect use of erase/remove idiom. deadbeef 2017-06-16 20:19:08 -07:00
  • dab1d2d34e Enable SNI in ssl adapter. Emad Omara 2017-06-16 15:43:11 -07:00
  • 43b39de0ca Roll chromium_revision 175bf817db..b032878ebd (480112:480186) buildbot 2017-06-16 15:24:03 -07:00
  • 5369a98951 Roll chromium_revision 1d3617187c..175bf817db (480056:480112) buildbot 2017-06-16 12:21:13 -07:00
  • 653158338e Templated AudioEncoderFactory kwiberg 2017-06-16 10:42:05 -07:00
  • 9fbbdc2fef Create the VideoEncoderFactory and implement it. Bjorn Mellem 2017-06-16 09:23:50 -07:00
  • c1db79beb7 Roll chromium_revision 804cd4b03e..1d3617187c (480025:480056) buildbot 2017-06-16 09:19:02 -07:00
  • 5cb19827e7 Tune loss-based BWE to be more compatible with the low frequency loss reports of audio streams. stefan 2017-06-16 07:47:00 -07:00
  • 8fa21c49ef Style fixes in rtcp_packet/ eladalon 2017-06-16 07:07:47 -07:00
  • 6b826ef66d Add cropping to VIEEncoder to match simulcast streams resolution ilnik 2017-06-16 06:53:48 -07:00
  • f79dbadc09 Add has_value() and value() methods to rtc::Optional. terelius 2017-06-16 06:48:13 -07:00
  • 0ef8fb91b4 Roll chromium_revision ff467ab402..804cd4b03e (480004:480025) buildbot 2017-06-16 06:46:56 -07:00
  • af35f833b7 Reduces sensitivity in audio-glitch detector for iOS henrika 2017-06-16 13:22:13 +02:00
  • bf5a2fc11b Use RaceChecker instead of ThreadChecker in a few places. erikvarga 2017-06-16 05:02:05 -07:00
  • bd09ebc71e Remove unused #include "libyuv/compare.h" eladalon 2017-06-16 04:54:22 -07:00
  • e150058236 Move setting switches in AppRTCMobile to Settings screen Anders Carlsson 2017-06-15 16:05:13 +02:00
  • 7c303907a4 Roll chromium_revision 2696d5a95c..ff467ab402 (479974:480004) buildbot 2017-06-16 03:27:08 -07:00
  • bc061b4280 Create AndroidVideoBuffer and allow renderers to consume it. Sami Kalliomäki 2017-06-16 10:08:23 +02:00
  • af99b6d67a Delete SignalSrtpError. nisse 2017-06-16 00:57:21 -07:00
  • 92b607b033 Roll chromium_revision 0292e51a15..2696d5a95c (479940:479974) buildbot 2017-06-16 00:17:26 -07:00
  • 2b6e65820e Roll chromium_revision 3a8e71c423..0292e51a15 (479885:479940) buildbot 2017-06-15 21:26:35 -07:00
  • 7ca9e3621f Roll chromium_revision 14085ad54d..3a8e71c423 (479854:479885) buildbot 2017-06-15 18:16:04 -07:00
  • 3fc2350ef9 Support H.264 high profile encoding on Exynos devices. glaznev 2017-06-15 16:24:37 -07:00
  • 24d00ddd8c Roll chromium_revision 3a1b0a576e..14085ad54d (479771:479854) buildbot 2017-06-15 15:15:49 -07:00
  • 38ede13042 Support building WebRTC without audio and video. zhihuang 2017-06-15 12:52:32 -07:00
  • beae92c784 Roll chromium_revision 408bddbca7..3a1b0a576e (479728:479771) buildbot 2017-06-15 12:21:41 -07:00
  • 65f3150412 Roll chromium_revision 5cda18fb97..408bddbca7 (479679:479728) buildbot 2017-06-15 09:45:43 -07:00
  • 112adf9ca9 Validate references of frames inserted into FrameBuffer2. philipel 2017-06-15 09:06:21 -07:00
  • eb02c03a53 Allow WebRtcMediaEngine to be created from any thread. deadbeef 2017-06-15 08:29:25 -07:00
  • d03a578333 Roll chromium_revision 47f5ca5642..5cda18fb97 (479659:479679) buildbot 2017-06-15 06:53:01 -07:00
  • 67561a6411 Use the same QP max for tests as in production sprang 2017-06-15 06:34:42 -07:00
  • fda496a31e Set overuse detector max frame interval based on target frame rate. sprang 2017-06-15 04:21:07 -07:00
  • 19e087fc91 This CL finalizes the Conversational Speech tool. The following changes have been made: - command line args wired, - user output added, - final polishing. alessiob 2017-06-15 03:49:57 -07:00
  • b2152b7b61 Roll chromium_revision 5423b19357..47f5ca5642 (479628:479659) buildbot 2017-06-15 03:33:17 -07:00
  • 6af9399117 ACM: Make AcmReceiver's ownership of NetEq more obvious Henrik Lundin 2017-06-14 14:13:02 +02:00
  • f9784f23d7 Reland of Conversational speech tool, simualtor + unit tests (patchset #1 id:1 of https://codereview.webrtc.org/2925123003/ ) alessiob 2017-06-15 02:24:59 -07:00
  • f4dd191b28 Change existing aec dump tests to use webrtc::AecDump. aleloi 2017-06-15 01:55:38 -07:00
  • af66f2ca8a Roll chromium_revision c4920d627d..5423b19357 (479593:479628) buildbot 2017-06-15 00:19:12 -07:00
  • 08cfc84278 Roll chromium_revision e7ba6263e5..c4920d627d (479563:479593) buildbot 2017-06-14 21:12:20 -07:00
  • 24b60ef75c Roll chromium_revision 6dec5dff0b..e7ba6263e5 (479516:479563) buildbot 2017-06-14 18:17:20 -07:00
  • f7f6572d59 Roll chromium_revision 4062088042..6dec5dff0b (479467:479516) buildbot 2017-06-14 15:19:33 -07:00
  • 4d2e0a874b Roll chromium_revision 78764cfda4..4062088042 (479277:479467) buildbot 2017-06-14 12:57:21 -07:00
  • a5e0df6438 Move MinPositive to call.h as discussed here: https://codereview.chromium.org/2888303005/#msg19 zstein 2017-06-14 11:41:48 -07:00
  • 23bc51bd87 Reland of ll chromium_revision 05ba7bc226..78764cfda4 (479231:479277) (patchset #1 id:1 of https://codereview.webrtc.org/2939903002/ ) kjellander 2017-06-14 11:06:42 -07:00
  • 62faaabce9 Android: Add functionality for wrapping C++ I420 buffers to Java Magnus Jedvert 2017-06-14 19:03:31 +02:00
  • cca0f6cc68 Support H.264 high profile decoding on Exynos devices. glaznev 2017-06-14 10:20:54 -07:00
  • 4f1f458a14 Also scan stderr for audio files to test, due to change in Android test_runner oprypin 2017-06-14 09:35:11 -07:00
  • 386e49690a Revert "Revert "Update webrtc/sdk/objc to new VideoFrameBuffer interface"" Magnus Jedvert 2017-06-14 14:04:36 +00:00
  • 26ecfcc1c1 Remove timeStampMs from EncodedImage. Sami Kalliomäki 2017-06-14 13:46:58 +02:00
  • 4eccdaa314 Fix a numerical issue in NetEq delay plotting henrik.lundin 2017-06-14 07:02:17 -07:00
  • 7a721e84f8 Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface Magnus Jedvert 2017-06-14 11:28:08 +02:00
  • 3c938fc5ea Add NetEq delay plotting to event_log_visualizer henrik.lundin 2017-06-14 06:09:58 -07:00
  • 3c81a1afd8 Add field trial for balanced degradation preference. asapersson 2017-06-14 05:52:21 -07:00
  • c417d9e558 NetEq: Removing LastError and LastDecoderError Henrik Lundin 2017-06-14 12:29:03 +02:00
  • 2b3aa14ee2 Fix Chromium style checker warnings for MockAudioDecoder kwiberg 2017-06-14 03:31:17 -07:00
  • 96444aecfc Implement operator<< for AudioCodecInfo and AudioCodecSpec kwiberg 2017-06-14 03:27:40 -07:00
  • 6c4ba9f77d Plot acknowledged bitrate when compiled with rtc_enable_bwe_test_logging. terelius 2017-06-14 02:41:59 -07:00
  • 58c5a7d4ac Revert of Roll chromium_revision 05ba7bc226..78764cfda4 (479231:479277) (patchset #1 id:1 of https://codereview.webrtc.org/2936153002/ ) mbonadei 2017-06-13 23:32:45 -07:00
  • f7e294d568 Implement kBalanced degradation preference. asapersson 2017-06-13 23:25:22 -07:00
  • b749e5e1f5 Fix for broken test BweFeedbackTest. tschumim 2017-06-13 22:58:21 -07:00
  • b7d6015529 Roll chromium_revision 05ba7bc226..78764cfda4 (479231:479277) buildbot 2017-06-13 21:17:34 -07:00
  • 7dce727c88 Roll chromium_revision 53ecf9341f..05ba7bc226 (479165:479231) buildbot 2017-06-13 18:25:48 -07:00
  • 6eb03b81bb Remove dependency on gunit headers in virtualsocketserver. Bjorn Mellem 2017-06-13 15:07:41 -07:00
  • 1ee2125909 Adding PortAllocator option to support cases where sockets can't be bound. deadbeef 2017-06-13 15:49:45 -07:00
  • 1d560e1b9a Roll chromium_revision 4ddaa6f836..53ecf9341f (479034:479165) buildbot 2017-06-13 15:24:21 -07:00
  • 179f997307 Remove DCHECK from PeerConnectionFactory::worker_thread. zstein 2017-06-13 15:01:49 -07:00
  • da4eba1e0a Tune vp9 quality scaler parameters glaznev 2017-06-13 11:34:49 -07:00
  • 0c61a36f49 Roll chromium_revision 4f7c2dc196..4ddaa6f836 (478995:479034) buildbot 2017-06-13 09:24:00 -07:00
  • 5c4eebb62b Implement org.webrtc.VideoEncoder using the android MediaCodec. Bjorn Mellem 2017-06-12 09:21:03 -07:00
  • 7be7883a01 Adds detection of audio glitches for playout on iOS (reland) henrika 2017-06-13 17:34:16 +02:00
  • 6e286cba7e Revert "Adds detection of audio glitches for playout on iOS. " Henrik Andreasson 2017-06-13 15:20:52 +00:00
  • 33e4e65706 Adds detection of audio glitches for playout on iOS. henrika 2017-06-13 16:11:08 +02:00
  • dea075c7a6 Log an error in RtpDemuxer::FindSsrcAssociations() if kMaxProcessedSsrcs exceeded eladalon 2017-06-13 07:57:31 -07:00
  • 7ed35f4643 Replacing WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP with WEBRTC_ENABLE_PROTOBUF. minyue-webrtc 2017-06-13 11:49:29 +02:00
  • 10e1f7548a Roll chromium_revision 9061a92f5c..4f7c2dc196 (478958:478995) buildbot 2017-06-13 07:32:31 -07:00
  • 29860331f1 Remove webrtcvideoengine2.h eladalon 2017-06-13 07:28:31 -07:00
  • 659a0101f6 Delete old include file webrtc/video_frame.h. nisse 2017-06-13 06:05:05 -07:00
  • a65ad22939 Delete unused method FilesystemInterface::GetFileTime. nisse 2017-06-13 05:37:44 -07:00
  • 8c6afef954 Make sure UI methods get called on the main thread adam.fedor 2017-06-13 05:25:33 -07:00
  • fdfeb8361e Declaring rtc_base_approved dep on webrtc_common mbonadei 2017-06-13 04:53:27 -07:00
  • 7339712256 Removing backward compatible header mbonadei 2017-06-13 04:25:37 -07:00
  • a735d4ebb5 Roll chromium_revision 0ca6ede735..9061a92f5c (478917:478958) buildbot 2017-06-13 03:56:00 -07:00
  • 2c9f9f2bc9 Only create H264 frames if there are no gaps in the packet sequence number. philipel 2017-06-13 02:47:28 -07:00
  • fc309750a9 Access UIApplication on main thread Anders Carlsson 2017-06-12 14:16:20 +02:00
  • 5b383c0ebd Revert "Update webrtc/sdk/objc to new VideoFrameBuffer interface" Magnus Jedvert 2017-06-13 09:14:36 +00:00
  • 1edbda09d4 Don't hardcode gn target path for licence generation. Kári Tristan Helgason 2017-06-13 10:45:42 +02:00
  • f3ba6484e3 Change rtp header extension AbsoluteSendTime::Write to take time in 24bit format Danil Chapovalov 2017-06-12 15:43:55 +02:00
  • 29f0d453aa Delete ApplicationName and OrganizationName. nisse 2017-06-13 02:04:51 -07:00
  • b008b45f1e Update webrtc/sdk/objc to new VideoFrameBuffer interface Magnus Jedvert 2017-06-12 15:09:48 +02:00
  • 687bc3e27b Delete unused method Win32Filesystem::GetAppPathname. nisse 2017-06-13 01:06:07 -07:00
  • 418b7d34d1 Increase number of unsignaled audio streams we handle to 4. solenberg 2017-06-13 00:38:27 -07:00