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  • f5bebd40f3 (Auto)update libjingle 64247466-> 64326665 henrike@webrtc.org 2014-04-04 18:39:07 +00:00
  • 653c325af2 Fix the library path for android 64-bit build michaelbai@google.com 2014-04-04 04:44:19 +00:00
  • 40ee3d07ed Consolidate audio conversion from Channel and TransmitMixer. andrew@webrtc.org 2014-04-03 21:56:01 +00:00
  • cca888a5bf Removed rehydrate.html dutton@google.com 2014-04-03 21:25:54 +00:00
  • be8e8ee6f6 Remove bad *s from filename. andrew@webrtc.org 2014-04-03 20:51:41 +00:00
  • c7b8b2f2a7 PRESUBMIT.py: use new way to specify default try builders kjellander@webrtc.org 2014-04-03 20:19:36 +00:00
  • fe165ded46 Added warning for Github move ****THESE_FILES_ARE_MOVING**** dutton@google.com 2014-04-03 19:57:06 +00:00
  • 240eec3cd4 Delay Estimator: Minor refactoring and added a setter function. bjornv@webrtc.org 2014-04-03 08:11:47 +00:00
  • 148149138d (Auto)update libjingle 64147530-> 64247466 wu@webrtc.org 2014-04-02 23:25:15 +00:00
  • 5e760e7b94 Check the return value of the FromString call and return failure when then value is invalid. I.e. uses bool FromString(const std::string& s, T* t) instead of T FromString(const std::string& str) wu@webrtc.org 2014-04-02 23:19:09 +00:00
  • e387771b98 Remove webrtc_unittest.cc from talk presubmit script. wu@webrtc.org 2014-04-02 22:23:16 +00:00
  • 184b913eb5 Rename RTPanalyze to rtp_analyze and remove old version henrik.lundin@webrtc.org 2014-04-02 20:56:17 +00:00
  • c7c432aa9b Remove AudioDevice::{Microphone,Speaker}IsAvailable. andrew@webrtc.org 2014-04-02 16:49:26 +00:00
  • 7549ff4257 This is to get rid of a bug relating to the return of NULL in calling GetDecoder when there are DTMF packets. minyue@webrtc.org 2014-04-02 15:03:01 +00:00
  • 1092ea0192 Add format specification to output file names henrik.lundin@webrtc.org 2014-04-02 07:46:49 +00:00
  • 620d444c0b Extends max sample rate from 96kHz to 192kHz on the input side. henrika@webrtc.org 2014-04-02 07:22:34 +00:00
  • 790385fee4 sink_filter_ds.cc: add lock to Receive procedure to Pause(). braveyao@webrtc.org 2014-04-02 02:14:55 +00:00
  • 19018ddb17 Make ACM2 the default in voe_cmd_test. andrew@webrtc.org 2014-04-01 20:58:05 +00:00
  • 05e7b44b83 (Auto)update libjingle 63948945-> 64147530 wu@webrtc.org 2014-04-01 17:44:24 +00:00
  • f8f7c8b618 Added simulations of capacity variations and wifi recordings. stefan@webrtc.org 2014-04-01 14:00:05 +00:00
  • 7e889b7126 Add /third_party/syzygy/binaries to .gitignore kjellander@webrtc.org 2014-04-01 13:46:48 +00:00
  • d10bdd3f78 Roll chromium_revision 255773:260462 kjellander@webrtc.org 2014-04-01 10:40:03 +00:00
  • ca9d038ac8 Fix ARM64 detection. andrew@webrtc.org 2014-04-01 01:19:08 +00:00
  • a789f3720a VoiceEngine(iOS & Android): removed NOT_SUPPORTED fischman@webrtc.org 2014-04-01 00:16:35 +00:00
  • 8f8119409d Roll libyuv to 994 for arm64 initial support using C versions of code. BUG=chromium:354539 TESTED=GYP_DEFINES="OS=ios target_arch=armv7 target_subarch=64" GYP_CROSSCOMPILE=1 GYP_GENERATOR_FLAGS="output_dir=out_ios" ./build/gyp_chromium -f ninja --depth=. libyuv_test.gyp && ninja -j7 -C out_ios/Debug-iphoneos R=andrew@webrtc.org, thorcarpenter@google.com fbarchard@google.com 2014-03-31 21:35:01 +00:00
  • 49c5ba32bb AppRTCDemo(iOS): now works in the iOS Simulator! fischman@webrtc.org 2014-03-31 20:22:19 +00:00
  • 61e78fca6c AppRTCDemo(iOS): remote-video reliability fixes fischman@webrtc.org 2014-03-31 20:16:49 +00:00
  • 30cd5b5278 libyuv roll to r986 for c89 fix to cpu_id. BUG=none TESTED=cl cpu_id.cc R=andrew@webrtc.org fbarchard@google.com 2014-03-31 17:28:46 +00:00
  • caeae4680c Add tests for the RBE RemoveStream() API. solenberg@webrtc.org 2014-03-31 13:33:39 +00:00
  • d0a81d91ff VoE Channel: Don't register codecs when stopping receiver henrik.lundin@webrtc.org 2014-03-31 07:31:53 +00:00
  • fe16488184 AppRTCDemo(android): specify DtlsSrtpKeyAgreement:true in CreatePeerConnection's constraints. fischman@webrtc.org 2014-03-28 19:58:03 +00:00
  • 4f2bd68744 Silence pointless LS_WARNING about port 0 for active-only candidates. fischman@webrtc.org 2014-03-28 18:13:34 +00:00
  • 987f2c9aae (Auto)update libjingle 63913264-> 63948945 wu@webrtc.org 2014-03-28 16:22:19 +00:00
  • 0aa04f9f24 Restore support for code coverage in WebRTC kjellander@webrtc.org 2014-03-28 13:14:00 +00:00
  • f7d501d48a (Auto)update libjingle 63884381-> 63913264 wu@webrtc.org 2014-03-27 23:48:25 +00:00
  • a5586b50e5 Protect ENABLE_PROFILING to fix profiling=1. andrew@webrtc.org 2014-03-27 22:44:13 +00:00
  • dd0b99debb Roll libvpx 258445:259973. fischman@webrtc.org 2014-03-27 22:27:54 +00:00
  • fff3fd35a6 Add arm64 to typedefs.h andrew@webrtc.org 2014-03-27 19:48:53 +00:00
  • 5a0218c794 Allow loopback tests to do TURN when served from webrtc.googlecode.com. BUG=3037 R=fischman@webrtc.org andresp@webrtc.org 2014-03-27 19:24:45 +00:00
  • cfe5e9c894 (Auto)update libjingle 63837929-> 63884381 wu@webrtc.org 2014-03-27 17:03:58 +00:00
  • 6b17be0bf8 Add svn mime-type properties to loopback_test files so they can be served from: https://webrtc.googlecode.com/svn/trunk/webrtc/tools/loopback_test/loopback_test.html andresp@webrtc.org 2014-03-27 10:52:09 +00:00
  • b13a7d5b1c Don't disable experimental AGC in audioproc. andrew@webrtc.org 2014-03-27 00:11:11 +00:00
  • b0ecc1c6fb (Auto)update libjingle 63777286-> 63837929 henrike@webrtc.org 2014-03-26 22:44:28 +00:00
  • b6dfbed1dc Exclude TwoStreamsSendAndFailUnsignalledRecvInOneToOne from TSAN. andrew@webrtc.org 2014-03-26 22:22:46 +00:00
  • b25576a75b talk/: enable _DEBUG in Debug for all posix fischman@webrtc.org 2014-03-26 21:53:47 +00:00
  • 44caf01c34 Re-submit: rev5775 andresp@webrtc.org 2014-03-26 21:00:21 +00:00
  • 1ca08f65e3 Fix after auto update in r5787. APPRTCVideoView.h/m was removed incorrectly. henrike@webrtc.org 2014-03-26 16:42:14 +00:00
  • 7ee0c16edd Makes ScreenCapturerMac exclude the window specified in DesktopCapturer::SetExcludedWindow. No behavior change for now since Chromium has not been updated to call SetExcludedWindow. jiayl@webrtc.org 2014-03-26 15:57:43 +00:00
  • 4e65602886 Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams. solenberg@webrtc.org 2014-03-26 14:32:47 +00:00
  • d09d074827 Protect write of send_target_bitrate. andresp@webrtc.org 2014-03-26 14:27:34 +00:00
  • 5fb7428496 (Auto)update libjingle 63775799-> 63776369 henrike@webrtc.org 2014-03-26 02:00:10 +00:00
  • a92fd74f40 (Auto)update libjingle 63773382-> 63775799 henrike@webrtc.org 2014-03-26 01:46:18 +00:00
  • dce3feb0b0 (Auto)update libjingle 63738002-> 63773382 henrike@webrtc.org 2014-03-26 01:17:30 +00:00
  • 440fa23553 Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out. solenberg@webrtc.org 2014-03-25 19:57:07 +00:00
  • 6cd201cf31 Revert 5775 "Modify bitrate controller to update bitrate based o..." andrew@webrtc.org 2014-03-25 19:42:39 +00:00
  • 681d448d88 Removing VideoCodecDerived and moving methods inside VideoCodec. mallinath@webrtc.org 2014-03-25 18:44:58 +00:00
  • 39f8ddae70 Updated WebRTC version to 3.51 elham@webrtc.org 2014-03-25 18:41:14 +00:00
  • ae3347a546 Fix after auto update: removed files were brought back. henrike@webrtc.org 2014-03-25 18:17:02 +00:00
  • e52b3b9c95 iOS video_capture: move @private vars to impl. fischman@webrtc.org 2014-03-25 18:15:07 +00:00
  • 76d4f389bb AppRTCDemo(iOS): allow rooms with no incoming audio. fischman@webrtc.org 2014-03-25 17:40:38 +00:00
  • 6e3dbc2a77 (Auto)update libjingle 63648983-> 63738002 henrike@webrtc.org 2014-03-25 17:09:47 +00:00
  • efcad39f77 Fix race condition in RTPSEnder. sprang@webrtc.org 2014-03-25 16:51:35 +00:00
  • ff7908abfd Roll Opus with ARM optimizations enabled to WebRTC tina.legrand@webrtc.org 2014-03-25 16:12:43 +00:00
  • 02e749f848 Change sprintf format string from %zu to %i henrik.lundin@webrtc.org 2014-03-25 13:39:11 +00:00
  • da07737e68 Modify bitrate controller to update bitrate based on process call and not only whenever a RTCP receiver block is received. andresp@webrtc.org 2014-03-25 12:48:42 +00:00
  • 0f0c992336 Temporarily use older protobuf library. sprang@webrtc.org 2014-03-25 12:43:58 +00:00
  • a16147c037 Adding API for setting bandwidth estimation configurations. stefan@webrtc.org 2014-03-25 10:37:31 +00:00
  • b64d52c292 iOS video_capture: start camera in the background. fischman@webrtc.org 2014-03-25 05:23:32 +00:00
  • 385a722646 PeerConnection(iOS): make ARC-clean talk/.../objc* and talk/examples/ios fischman@webrtc.org 2014-03-25 05:16:29 +00:00
  • e68102e046 iOS VideoEngine: move video_{capture,render} to ARC. fischman@webrtc.org 2014-03-25 05:15:44 +00:00
  • e42b8ab129 Cleanups in libjingle to make it compile with chromium_code=1 sergeyu@chromium.org 2014-03-25 00:31:35 +00:00
  • 7fa1fcb72c AppRTCDemo(ios): style/cleanup fixes following cr/62871616-p10 fischman@webrtc.org 2014-03-25 00:11:56 +00:00
  • ce12f1fd32 Add configuration for ability to use the encode usage measure for triggering overuse/underuse. asapersson@webrtc.org 2014-03-24 21:59:16 +00:00
  • b70c8e9dfd Disable flaky WebRtcVideoMediaChannelTests on memcheck and tsan. andrew@webrtc.org 2014-03-24 20:57:42 +00:00
  • 3fb8f7bbb0 Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. solenberg@webrtc.org 2014-03-24 20:28:11 +00:00
  • c693a2a624 PeerConnection(iOS): fix case in #import statements. fischman@webrtc.org 2014-03-24 18:56:37 +00:00
  • 9d4762e8b6 Have changes to REMB trigger RTCP to be sent immediately. stefan@webrtc.org 2014-03-24 17:13:00 +00:00
  • 1e6cb2c5d2 (Auto)update libjingle 63560528-> 63648983 wu@webrtc.org 2014-03-24 17:01:50 +00:00
  • 28e83d1a56 DelayEstimator: Updates delay_quality and adds soft reset. bjornv@webrtc.org 2014-03-24 15:26:52 +00:00
  • 92c0e29963 Run Opus with lower complexity setting on Android, iOS and/or ARM tina.legrand@webrtc.org 2014-03-24 14:38:36 +00:00
  • 3c412b24d9 Add targetBitrate to VideoCodec struct. pbos@webrtc.org 2014-03-24 12:36:52 +00:00
  • 7e3ee8362b Disabled some of the remote bitrate estimator baseline tests. stefan@webrtc.org 2014-03-24 10:45:13 +00:00
  • b1f5010075 VoE changes to allow forwarding of packets from VoE to ViE BWE. solenberg@webrtc.org 2014-03-24 10:38:25 +00:00
  • 37ca765650 Add fir_filter to common_audio aluebs@webrtc.org 2014-03-24 10:16:11 +00:00
  • af839b28b0 Add AIMD option to BWE API. stefan@webrtc.org 2014-03-24 09:42:08 +00:00
  • ba5a6c3d89 ACM2/NetEq4 did not decode Opus in stereo tina.legrand@webrtc.org 2014-03-23 09:58:48 +00:00
  • 152208adeb (Auto)update libjingle 63547048-> 63560528 henrike@webrtc.org 2014-03-21 21:43:26 +00:00
  • 07bc734459 Refactor in BitrateController module. - Move condition of 0 bps as max meaning 1gbps from SendSideBandwidthEstimation to BitrateController. - Remove condition on bitrate=0 meaning bandwidth estimation off as that could only happen when no observers existed and in which case the estimation would be ignored. - Add MaybeTriggerOnNetworkChanged which only runs rate allocation if any of the dependent variables has changed thus allowing to remove many of the bool returns that try to indicate if the estimation has changed which would not be aware if the observers have changed. - SendSideBandwidthEstimation now has a UpdateBitrate and has clear code paths to which calls update bitrate. - Changes in enforce_min_bitrate so the 10kbps min is set from the BitrateController and not from the outside this keep valid as observers are changed. andresp@webrtc.org 2014-03-21 16:51:01 +00:00
  • be7e26d229 (Auto)update libjingle 63503990-> 63547048 henrike@webrtc.org 2014-03-21 16:40:18 +00:00
  • 6f9c48348b Fixing crash in video_render_tests in release mode. henrikg@webrtc.org 2014-03-21 14:41:27 +00:00
  • 16b75c2c7a Remove locks in SendSideBandwidthEstimation since those are only accessed while owning locks in BitrateControllerImpl (excluding AvailableBandwidth). andresp@webrtc.org 2014-03-21 14:00:51 +00:00
  • b28bfa7efc Adding FEC support in NetEq 4. minyue@webrtc.org 2014-03-21 12:07:40 +00:00
  • 0e65fdaa3b Fix "unreachable code" warnings (MSVC warning 4702) in webrtc. pbos@webrtc.org 2014-03-21 10:26:42 +00:00
  • 0209e565de Adding operator== and != methods for CodecInst and VideoCodec structures. mallinath@webrtc.org 2014-03-21 00:41:28 +00:00
  • 9c2bd2b288 Roll libvpx 254609:258445. fischman@webrtc.org 2014-03-21 00:20:01 +00:00
  • 0c6f0f94f1 Revert 5737 "Add system wrapper dependency to libjingle targets." mallinath@webrtc.org 2014-03-20 23:07:03 +00:00
  • 5e83c65aee (Auto)update libjingle 63493960-> 63503990 henrike@webrtc.org 2014-03-20 22:33:30 +00:00
  • 062e6e539f ARGBScale fix for bilinear down sampling overread when source size is odd. BUG=chromium:352592 TESTED=drmemory on libyuv new unittest scaling unittest. R=andrew@webrtc.org fbarchard@google.com 2014-03-20 21:44:15 +00:00
  • a8ebdb71e3 Revert "(Auto)update libjingle 63363208-> 63493960" (r5740) henrike@webrtc.org 2014-03-20 20:27:25 +00:00
  • 5f768adc27 (Auto)update libjingle 63363208-> 63493960 henrike@webrtc.org 2014-03-20 20:18:06 +00:00