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  • 268862c5e4 Address denicija's comments for AppRTCMobile video codec setting. sakal 2017-04-11 05:36:43 -07:00
  • 24da37b0bf ObjC: RTCVideoSource cleanup magjed 2017-04-11 04:50:15 -07:00
  • 29dbb1992a Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ ) ilnik 2017-04-11 04:49:07 -07:00
  • dee5eb14e1 Android Logging.java: Load native library only when needed magjed 2017-04-11 04:21:50 -07:00
  • 382f2b2c45 Fix swarming tests not running in parallel kjellander 2017-04-11 04:07:01 -07:00
  • 4fa0c4f97f Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) ilnik 2017-04-11 04:01:43 -07:00
  • 5721866808 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ ) ilnik 2017-04-11 03:59:43 -07:00
  • cabc25c7e7 Roll chromium_revision 860f7b94d4..419b6b1a41 (463558:463589) buildbot 2017-04-11 03:15:32 -07:00
  • cde2528d28 Enabling 'gn check' on //webrtc/ortc. mbonadei 2017-04-11 02:52:49 -07:00
  • 10fc0e6385 Delay based logging. philipel 2017-04-11 01:50:23 -07:00
  • 64e739aeae Add content type information to Encoded Images and add corresponding RTP extension header. Use it to separate UMA e2e delay metric between screenshare from video. Content type extension is set based on encoder settings and processed and decoders. ilnik 2017-04-11 01:46:04 -07:00
  • 93cda2ebde APM-QA tool, renaming noise generators into input-reference generators. alessiob 2017-04-11 01:06:28 -07:00
  • 9765370416 Resolve dependency between rtc_event_log_api and remote_bitrate_estimator michaelt 2017-04-11 00:49:44 -07:00
  • 810eecf2f4 Roll chromium_revision c57654688e..860f7b94d4 (463520:463558) buildbot 2017-04-11 00:48:22 -07:00
  • 7fb7bbd179 Revert of Add first part of the network_tester functionality. (patchset #13 id:260001 of https://codereview.webrtc.org/2779233002/ ) michaelt 2017-04-11 00:16:51 -07:00
  • 333d0ff631 Add first part of the network_tester functionality. michaelt 2017-04-10 23:26:35 -07:00
  • e0ab0ad85d Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT kjellander 2017-04-10 23:21:43 -07:00
  • 0d4e068d0a Make safe_cmp::* constexpr kwiberg 2017-04-10 22:44:07 -07:00
  • d491109d2d Roll chromium_revision 1af3c1a4a8..c57654688e (463476:463520) buildbot 2017-04-10 22:13:44 -07:00
  • 4a9d08fbc0 Roll chromium_revision d3a2a83fbf..1af3c1a4a8 (463418:463476) buildbot 2017-04-10 19:18:48 -07:00
  • 8c459f9eee Restore old (deprecated) signature of initializeAndroidGlobals. deadbeef 2017-04-10 18:07:55 -07:00
  • 20c84ccd48 Making FakeNetworkPipe demux audio and video packets. minyue 2017-04-10 16:57:57 -07:00
  • d9ce76444f Make RtpTransport actually implement RtpTransportInterface zstein 2017-04-10 16:17:57 -07:00
  • 0f92c796b4 Roll chromium_revision 5d7042a87c..d3a2a83fbf (463209:463418) buildbot 2017-04-10 15:59:16 -07:00
  • b4fc73a3ab Removing unnecessary parameters from initializeAndroidGlobals. deadbeef 2017-04-10 15:08:02 -07:00
  • 6799553a2c Add information about microphone gain changes to AEC3 peah 2017-04-10 14:12:41 -07:00
  • 6d822adac4 Added forced zero AEC output after call startup and echo path changes peah 2017-04-10 13:52:14 -07:00
  • ca31f175e1 Remove deprecated RTPPayloadStrategy Remove deprecated set_use_rtx_payload_mapping_on_restore() Remove unused headers danilchap 2017-04-10 08:45:29 -07:00
  • a1ef71f622 Add parser to visualise the ana dump michaelt 2017-04-10 08:31:26 -07:00
  • 8d609f6b6d Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) hbos 2017-04-10 07:39:05 -07:00
  • b0f7e39fd4 Move IsIntlike to type_traits.h kwiberg 2017-04-10 06:56:58 -07:00
  • 37e99fd3fa Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ kwiberg 2017-04-10 05:15:48 -07:00
  • 2fa97fdbe5 Roll chromium_revision 0a53e4a670..5d7042a87c (463181:463209) buildbot 2017-04-10 04:56:59 -07:00
  • 0642b3297d Remove duplicate entries from AUTHORS file henrik.lundin 2017-04-10 04:54:00 -07:00
  • fbcc5cb386 Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) olka 2017-04-10 04:38:13 -07:00
  • 925e9d762c Removed workaround for the WARN_UNUSED_RESULT issue. peah 2017-04-10 04:18:38 -07:00
  • 4fb651dd22 Event log cleanup in tests. philipel 2017-04-10 03:54:05 -07:00
  • fca900aa37 Fix two invalid DCHECKs related to audio BWE. stefan 2017-04-10 03:53:00 -07:00
  • 49cad02cf3 Ignore some UBSan errors kwiberg 2017-04-10 02:29:33 -07:00
  • 9f2c18e237 Changed OLA window for neteq. Old code didnt work well with 48khz soren 2017-04-10 02:22:46 -07:00
  • c547e84ec5 Allow rtp::Packet::*RawExtension to take 0 as an extension id danilchap 2017-04-10 01:31:49 -07:00
  • 02465b8a11 Add some unit tests to vie_encoder. asapersson 2017-04-10 01:12:52 -07:00
  • 36e6a8f1bd WavReaderAdaptor is a simple adaptor of the existing class WavReader from webrtc/common_audio/wav_file.h. The adaptor was mainly needed to use dependency injection and easily test the MultiEndCall class (see https://codereview.webrtc.org/2761853002/). alessiob 2017-04-10 00:53:53 -07:00
  • 2042c16be0 Revert of Delete class ScopedPtrCollection. Replaced with vector of unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2808463002/ ) nisse 2017-04-10 00:31:33 -07:00
  • 64c93c380f Roll chromium_revision 1ab7c6059c..0a53e4a670 (463170:463181) buildbot 2017-04-10 00:17:32 -07:00
  • 188596f20f Delete class ScopedPtrCollection. Replaced with vector of unique_ptr. nisse 2017-04-10 00:02:52 -07:00
  • bd1a6814ea Roll chromium_revision c8a0f6b4c5..1ab7c6059c (463161:463170) buildbot 2017-04-09 21:13:06 -07:00
  • 1b0882d44c Roll chromium_revision c30f6366d7..c8a0f6b4c5 (463148:463161) buildbot 2017-04-09 18:13:24 -07:00
  • 1d9ef47bbf Roll chromium_revision b0bf8e8ed3..c30f6366d7 (463140:463148) buildbot 2017-04-09 15:13:25 -07:00
  • a3d4d944f1 Roll chromium_revision 5e27d4b8d1..b0bf8e8ed3 (463138:463140) buildbot 2017-04-09 12:12:55 -07:00
  • 4b37127414 Fix compilation issues of std::unique_ptr steweg 2017-04-09 09:09:06 -07:00
  • 8cc642b555 Roll chromium_revision d5df028c94..5e27d4b8d1 (463137:463138) buildbot 2017-04-09 06:14:22 -07:00
  • 89292d1ad3 Roll chromium_revision 270af5af87..d5df028c94 (463130:463137) buildbot 2017-04-09 00:48:31 -07:00
  • 5253411e74 Roll chromium_revision 9d8fbbd04c..270af5af87 (463126:463130) buildbot 2017-04-08 03:13:51 -07:00
  • 01f279337d Roll chromium_revision a35a1e2ce2..9d8fbbd04c (463121:463126) buildbot 2017-04-08 00:14:54 -07:00
  • 50ddc634d8 Roll chromium_revision 98c8321fe9..a35a1e2ce2 (463081:463121) buildbot 2017-04-07 21:14:17 -07:00
  • 9d8052d9db Roll chromium_revision 762665735a..98c8321fe9 (462871:463081) buildbot 2017-04-07 18:12:00 -07:00
  • 66e9f7630f Adjust parameter in vp9 videoprocessor_integration test. jianj 2017-04-07 15:07:18 -07:00
  • 8d23c050f2 MultiEndCall::CheckTiming() verifies that a set of audio tracks and timing information is valid to simulate conversational speech. Unordered turns are rejected. Self cross-talk and cross-talk with 3 or more speakers are not permitted since it would require mixing at the simulation step. alessiob 2017-04-07 12:05:08 -07:00
  • 292084c376 Added the GetSources() to the RtpReceiverInterface and implemented it for the AudioRtpReceiver. zhihuang 2017-04-07 10:57:22 -07:00
  • bb16a483f2 Roll chromium_revision 61577bba5a..762665735a (462822:462871) buildbot 2017-04-07 09:14:50 -07:00
  • 89420459d1 Adding support for handling highly reverberant echoes in AEC3. peah 2017-04-07 06:13:39 -07:00
  • 38415b2d8e Reland of Adding PRESUBMIT check on google::protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2791583002/ ) mbonadei 2017-04-07 05:38:01 -07:00
  • 423f106e8c Add support for 64-bit architectures in build_aar.py. sakal 2017-04-07 05:10:15 -07:00
  • 2ce640fada Fixing sample-rate dependent band-split filter issues in AEC3 peah 2017-04-07 03:57:48 -07:00
  • ea44ad4efc Roll chromium_revision d65c1d7370..61577bba5a (462801:462822) buildbot 2017-04-07 03:14:10 -07:00
  • 7c2c8438f1 Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ ) mbonadei 2017-04-07 00:59:12 -07:00
  • 0944a80f3e Update stats for cpu/quality adaptation changes to excluded time when video is suspended. asapersson 2017-04-07 00:57:58 -07:00
  • 533b7ac8e1 Roll chromium_revision c405301b70..d65c1d7370 (462743:462801) buildbot 2017-04-07 00:50:37 -07:00
  • 77492860a9 Make AudioProcessing::GetConfig() pure virtual henrik.lundin 2017-04-06 23:28:09 -07:00
  • abd101b91f Support multiple connected Android devices in low bandwidth audio test oprypin 2017-04-06 23:21:30 -07:00
  • 225bfc0971 Make PacketTransportInternal inherit from PacketTransportInterface. deadbeef 2017-04-06 21:47:33 -07:00
  • 81c899de49 Roll chromium_revision fbddb8a080..c405301b70 (462718:462743) buildbot 2017-04-06 21:12:10 -07:00
  • f49011b1a8 Roll chromium_revision 300bd53cd8..fbddb8a080 (462634:462718) buildbot 2017-04-06 18:13:25 -07:00
  • 4b572d048a Correction of the AEC3 underrun behavior and minor other corrections peah 2017-04-06 16:33:06 -07:00
  • 86afe9d661 Major updates to the echo removal functionality in AEC3 peah 2017-04-06 15:45:32 -07:00
  • f51517a64f Roll chromium_revision bb330d1b8f..300bd53cd8 (462561:462634) buildbot 2017-04-06 15:21:06 -07:00
  • 8cd1cce52a Roll chromium_revision 86718dcdf2..bb330d1b8f (462494:462561) buildbot 2017-04-06 12:21:02 -07:00
  • 1ffbd6c93c Injectable audio encoders: voice_engine/channel changes. ossu 2017-04-06 12:05:04 -07:00
  • 5f4aaeb0a4 Re-enable FullStackTest.ScreenshareSlidesVP9_2SL test. marpan 2017-04-06 10:52:23 -07:00
  • 4b62001e4f Adding AudioDeviceDataObserver interface Lu Liu 2017-04-06 10:17:01 -07:00
  • a1a040a4a4 Injectable audio encoders: BuiltinAudioEncoderFactory ossu 2017-04-06 10:03:21 -07:00
  • ac4bbdf9d6 Roll chromium_revision 3014f8b41e..86718dcdf2 (462374:462494) buildbot 2017-04-06 09:27:50 -07:00
  • cab8d888c5 Revert of Enable rtc_unittests on iOS simulator (patchset #2 id:20001 of https://codereview.webrtc.org/2799033004/ ) kjellander 2017-04-06 07:32:51 -07:00
  • 6167b2621f Make RtpTransportControllerSend::send_side_cc_ a direct member. nisse 2017-04-06 06:34:25 -07:00
  • cde46b7278 Resolve cyclic dependency between audio network adaptor and event log api michaelt 2017-04-06 05:59:10 -07:00
  • 28dc285f22 Adding cbr support for Opus soren 2017-04-06 05:48:36 -07:00
  • 388fe425c7 Make WARN_UNUSED_RESULT a no-op on gcc kwiberg 2017-04-06 04:32:27 -07:00
  • 177b17e050 Move AndroidVideoTrackSourceObserver from API to src magjed 2017-04-06 04:31:26 -07:00
  • 639d46a7ab Delete system_wrappers logging facility. nisse 2017-04-06 03:53:06 -07:00
  • be77920021 Revert of CQ: Remove Linux ARM64 Debug trybot from default set. (patchset #1 id:1 of https://codereview.webrtc.org/2790263003/ ) kjellander 2017-04-06 03:07:02 -07:00
  • 24180017d5 ACM: Change test output files from PCM to WAV henrik.lundin 2017-04-06 02:40:37 -07:00
  • 4fcfdd8ff0 Enable rtc_unittests on iOS simulator kjellander 2017-04-06 02:21:45 -07:00
  • a280f7c026 Added integer parsing functions in base/string_to_number.h ossu 2017-04-06 02:02:15 -07:00
  • b1e3fc4515 Enable tools_unittests and rtc_stats_unittests on iOS Simulator kjellander 2017-04-06 01:48:37 -07:00
  • e24991d7d6 Adds AudioDeviceTest.RunPlayoutAndRecordingInFullDuplex unittest. henrika 2017-04-06 01:14:23 -07:00
  • 978504e25c Move rtp header extension length check from Packet::FindExtension to ExtensionT::Parse to allow to read variable-length extensions. danilchap 2017-04-06 01:03:53 -07:00
  • ed6343d90a Roll chromium_revision 875e8893e9..3014f8b41e (462360:462374) buildbot 2017-04-06 00:13:25 -07:00
  • 251eb2715a Roll chromium_revision 75820eb165..875e8893e9 (460410:462360) kjellander 2017-04-05 22:49:24 -07:00
  • f6a4f37685 Reland of Fixed error for missing explict class initialization error on iOS WebRTC buildbots (patchset #1 id:1 of https://codereview.webrtc.org/2803933002/ ) guidou 2017-04-05 22:06:36 -07:00