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  • 1379f1f1e6 Extract the parameters for the encoder stack from the CodecManager kwiberg 2015-11-23 04:30:52 -08:00
  • 30a5e563ea Roll chromium_revision 3796a7a..e038f1d (361065:361088) kjellander 2015-11-23 04:22:43 -08:00
  • f0a431a283 Exclude EndToEndTest.ReceivesTransportFeedback and TransportFeedbackNotConfigured from DrMemory. Stefan Holmer 2015-11-23 12:59:08 +01:00
  • db81ffd6f5 Request keyframe if too many packets are missing and NACK is disabled. jbauch 2015-11-23 03:59:02 -08:00
  • fa8ae9a535 Remove <iostream> include from file_audio_device.cc kjellander@webrtc.org 2015-11-23 12:44:01 +01:00
  • 8becec3b49 talk: remove deprecated *processor.h files tfarina 2015-11-23 02:19:55 -08:00
  • 87d584597c Fix androidmediadecoder_jni TS logging. And fix pragma warning about deprecated "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h include. perkj 2015-11-23 01:46:27 -08:00
  • c3c4cdb88c Add Android x86 and x64 trybots to CQ. kjellander@webrtc.org 2015-11-23 09:59:48 +01:00
  • d5674c30aa Roll chromium_revision c29f20c..3796a7a (361043:361065) kjellander 2015-11-22 19:52:21 -08:00
  • 50c5136cb2 RTCP Bye packet moved to own file Bye class got support for Parsing Reason field implemented danilchap 2015-11-22 09:03:11 -08:00
  • c9829138c0 Roll chromium_revision 6018759..c29f20c (361030:361043) kjellander 2015-11-22 04:39:58 -08:00
  • 485b5b01ec Roll chromium_revision 4df2d47..6018759 (361029:361030) kjellander 2015-11-21 19:53:59 -08:00
  • 82581a03a8 Roll chromium_revision 3966d2c..4df2d47 (361020:361029) kjellander 2015-11-21 11:59:39 -08:00
  • b4a29d9760 Roll chromium_revision b854092..3966d2c (360794:361020) kjellander 2015-11-21 04:01:39 -08:00
  • 13f6b8f7f4 Increase transport feedback frequency to 20 Hz. stefan 2015-11-20 18:14:14 -08:00
  • 43edf0ffb9 Require negotiation to send transport cc feedback over RTCP. stefan 2015-11-20 18:05:48 -08:00
  • bd13838ccc Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack. solenberg 2015-11-20 16:08:07 -08:00
  • 672304a654 NetEq: Remove overly verbose logging henrik.lundin 2015-11-20 11:57:05 -08:00
  • 5def7b9fde Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ ) deadbeef 2015-11-20 11:43:22 -08:00
  • 7add058439 Move some receive stream configuration into webrtc::AudioReceiveStream. solenberg 2015-11-20 09:59:34 -08:00
  • 6834fa10f1 Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) deadbeef 2015-11-20 09:49:59 -08:00
  • 0a43fef6dc Allow pacer to boost bitrate in order to meet time constraints. sprang 2015-11-20 09:00:37 -08:00
  • 34911ad55c Improved error handling in iOS ADM to avoid race during init henrika 2015-11-20 15:47:09 +01:00
  • 76a31ca3d4 Avoids hitting DCHECK in OpenSL ES player henrika 2015-11-20 13:40:44 +01:00
  • 1afae74246 Roll chromium_revision 5c83f4e..b854092 (360728:360794) kjellander 2015-11-20 04:36:16 -08:00
  • 30e918278c This cl add support to encode from textures to MediaCodecVideoEncoder. perkj 2015-11-20 01:31:25 -08:00
  • 5663b4fa9a iOS: Set enable_protobuf=1 by default. kjellander@webrtc.org 2015-11-20 09:37:41 +01:00
  • 7e63ef0e8f Allow default audio receive channel to receive on any unsignalled SSRC. solenberg 2015-11-20 00:19:43 -08:00
  • b0ad43baa0 Add aecdump support to audioproc_f aluebs 2015-11-20 00:11:53 -08:00
  • ceb450b51d Roll chromium_revision c8eec9a..5c83f4e (360565:360728) kjellander 2015-11-20 00:05:52 -08:00
  • 17c0aff9ea Enable VP9 HW decoder on Exynos chips. Alex Glaznev 2015-11-19 15:56:11 -08:00
  • 7593aad163 Re-enable mistakenly disabled PEM tests. Misc cleanup and alignment fixes. torbjorng 2015-11-19 12:20:51 -08:00
  • 7755e2064b Chrome has now been updated. perkj 2015-11-19 12:02:21 -08:00
  • 726b1f7a14 Removed dummy "mediastreamsignaling.h" perkj 2015-11-19 12:00:55 -08:00
  • 191c1f9d5b Disable all JsepPeerConnectionP2PTestClient tests on Mac due to flakiness on Mac Debug bots. ivoc 2015-11-19 11:12:06 -08:00
  • 12e21a0d6c Remove dead code (we no longer support SILK) kwiberg 2015-11-19 11:08:29 -08:00
  • ef453238aa Android: Make classes non-final Magnus Jedvert 2015-11-19 17:54:04 +01:00
  • 062e14e8e6 Roll chromium_revision bb7899a..c8eec9a (360504:360565) kjellander 2015-11-19 07:03:46 -08:00
  • f399f2174c Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 on linux due to flakiness on the Linux64 Debug bot. ivoc 2015-11-19 06:44:32 -08:00
  • f22695c3d8 Remove build_with_libjingle and exclude failing iOS tests from 'All' target. kjellander@webrtc.org 2015-11-19 15:39:29 +01:00
  • 1503867850 Disabled several JsepPeerConnectionP2PTestClient tests on Mac, due to flakiness on Debug Mac trybots. ivoc 2015-11-19 05:28:07 -08:00
  • e488a0dbe4 Fix DTLS packet boundary handling in SSLStreamAdapterTests. jbauch 2015-11-19 05:17:58 -08:00
  • 87097a8be5 Roll chromium_revision ed2e3fb..bb7899a (360379:360504) kjellander 2015-11-19 02:59:38 -08:00
  • b6755ab6df Revert of Adding thread timeout for audio recorer thread in Java (patchset #2 id:20001 of https://codereview.webrtc.org/1444313002/ ) henrika 2015-11-19 02:43:11 -08:00
  • 488e75f11b Patchset 1 yet again relands without modification https://codereview.webrtc.org/1422963003/ It do the following: Per 2015-11-19 10:43:36 +01:00
  • 0969398277 Revert of Remove android_rel from CQ since all of its machines are offline. (patchset #1 id:1 of https://codereview.webrtc.org/1459083002/ ) kjellander 2015-11-18 23:57:44 -08:00
  • 392d0c2701 Remove android_rel from CQ since all of its machines are offline. Henrik Kjellander 2015-11-19 08:04:46 +01:00
  • 521ed7bf02 Reland Convert internal representation of Srtp cryptos from string to int Guo-wei Shieh 2015-11-18 19:41:53 -08:00
  • 318166bed7 Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ ) guoweis 2015-11-18 19:03:38 -08:00
  • 2764e1027a Convert internal representation of Srtp cryptos from string to int. guoweis 2015-11-18 18:02:32 -08:00
  • 3c652b6746 modules/audio_coding: Remove some codec include dirs kjellander@webrtc.org 2015-11-18 23:07:57 +01:00
  • b7ce96470b modules/video_coding/utility: Remove include kjellander@webrtc.org 2015-11-18 23:04:10 +01:00
  • 1b20d81d35 Roll chromium_revision 64f2817..ed2e3fb (360275:360379) kjellander 2015-11-18 13:31:58 -08:00
  • 0f59a88b32 modules/video_processing: refactor interface->include + more. Henrik Kjellander 2015-11-18 22:31:24 +01:00
  • ed7d6ec63e WebRTC: Add compability header for video_coding refactoring. Henrik Kjellander 2015-11-18 22:26:25 +01:00
  • ad948c42a1 Preliminary support of VP9 HW encoder on Android. Alex Glaznev 2015-11-18 13:06:42 -08:00
  • 2557b86e76 modules/video_coding refactorings Henrik Kjellander 2015-11-18 22:00:21 +01:00
  • 4dd7a653b5 Temporarily disable VERIFY while bug is investigated. phoglund 2015-11-18 09:54:55 -08:00
  • 223692aa85 Remove dead code kwiberg 2015-11-18 08:27:51 -08:00
  • e1a27d48ad Move CNG/RED payload type extraction to Rent-A-Codec kwiberg 2015-11-18 07:32:49 -08:00
  • 49a6c99d1f Disables BitrateEstimatorTest.SwitchesToASTThenBackToTOFForVideo on win_drmemory_full due to flakiness. ivoc 2015-11-18 07:04:37 -08:00
  • 2446e5a2de Fixed the render queue item size preallocation and verification, moved constant declarations, removed redundant queue allocation peah 2015-11-18 06:11:13 -08:00
  • 0219c9b4bf rtcp::App moved into own file and got Parse function danilchap 2015-11-18 05:56:53 -08:00
  • 2aff615bd7 Remove spammy logging of RTCP delivery failures. Peter Boström 2015-11-18 13:47:16 +01:00
  • f70568c04b So long and thanks for all the code reviews! andrew 2015-11-18 03:07:39 -08:00
  • cb50c96be2 Set temporal up switch bit to false for flexible mode (one temporal layer is configured currently). asapersson 2015-11-18 01:58:55 -08:00
  • aa45843723 Roll chromium_revision a6d9f7f..64f2817 (360123:360275) kjellander 2015-11-17 20:24:27 -08:00
  • 310b093aa4 Fix active tcp port to 9 Guo-wei Shieh 2015-11-17 19:15:50 -08:00
  • 2935e01419 Several Tick counter improvements try #2." thaloun 2015-11-17 15:02:44 -08:00
  • c073615d56 Update references to TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305, etc. davidben 2015-11-17 12:58:11 -08:00
  • 0a75749da4 Roll chromium_revision 04756fa..a6d9f7f (360053:360123) kjellander 2015-11-17 11:58:04 -08:00
  • 32f39968ce Re-apply change https://codereview.webrtc.org/1426673007/ honghaiz 2015-11-17 11:36:31 -08:00
  • 5c489c9d3e Add OpenSL ES enable setting to AppRTCDemo (part 2). henrika 2015-11-17 10:12:39 -08:00
  • 2be7c54b13 Remove ViEEncoder::ScaleInputImage. Peter Boström 2015-11-17 15:52:51 +01:00
  • bd05f0ba52 Unconditionally build VP9 support. Peter Boström 2015-11-17 15:27:32 +01:00
  • 18adf0a79d Add UMA for send bwe and pacer bitrate. stefan 2015-11-17 06:24:56 -08:00
  • d9eec762ce Trace encoding/decoding time in a generic way. pbos 2015-11-17 06:03:43 -08:00
  • 5a71f03f8b Deactivate the audio session after a call ends using the AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation constant since it is recommended for VoIP apps. henrika 2015-11-17 14:54:48 +01:00
  • 45e998d958 Roll chromium_revision a2e8a40..04756fa (359987:360053) kjellander 2015-11-17 05:09:08 -08:00
  • fd614c2149 Adding thread timeout for audio recorer thread in Java henrika 2015-11-17 04:28:26 -08:00
  • e66339296b Add OpenSL ES enable setting to AppRTCDemo. glaznev 2015-11-17 04:05:29 -08:00
  • 3c12f4dadb Revert of Create rtc::AtomicInt POD struct. (patchset #12 id:220001 of https://codereview.webrtc.org/1420043008/ ) pbos 2015-11-17 03:21:02 -08:00
  • 192164eebc Preparational work before introducing the locks in order to harmonize the code: -Moved the initialize function -Moved api_format into the shared state peah 2015-11-17 02:16:45 -08:00
  • 4d291f7d5e Applied the render queueing to the agc. peah 2015-11-16 23:52:25 -08:00
  • 03179cd850 Roll chromium_revision 6fd4bdd..a2e8a40 (359891:359987) kjellander 2015-11-16 19:57:28 -08:00
  • 740c4f11e0 Remove packet initializer in RtpRtcpRtxNackTest. pbos 2015-11-16 17:19:34 -08:00
  • 854e84c7fb Use webrtc/base/logging.h for video coding/processing. pbos 2015-11-16 16:39:06 -08:00
  • c91d173870 Revert of Several Tick counter improvements. (patchset #8 id:140001 of https://codereview.webrtc.org/1415923010/ ) thaloun 2015-11-16 16:28:49 -08:00
  • fa6228e221 Introduced the render sample queue for the aec and aecm. peah 2015-11-16 16:27:42 -08:00
  • 4c27e4b62d Several Tick counter improvements. Tim Haloun 2015-11-16 14:37:41 -08:00
  • eb8b388273 Fix VP9 support in AppRTCDemo. Alex Glaznev 2015-11-16 14:11:53 -08:00
  • 6f8ce060a2 common_video: rename interface -> include kjellander 2015-11-16 13:52:24 -08:00
  • 591cb1fea4 Roll chromium_revision c958aa7..6fd4bdd (359816:359891) kjellander 2015-11-16 12:34:01 -08:00
  • b27f590ece Create rtc::AtomicInt POD struct. pbos 2015-11-16 11:03:04 -08:00
  • 3528a27b1b Flesh out webrtc/.gitignore brucedawson 2015-11-16 11:01:54 -08:00
  • 482b12e2c3 Remove BundleFilter filtering of RTCP. pbos 2015-11-16 10:19:58 -08:00
  • 8b85de2ba1 Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail. solenberg 2015-11-16 09:48:04 -08:00
  • 9a7c838ec4 Adding stddef.h to opus_inst.h. minyue 2015-11-16 08:07:01 -08:00
  • 3a94154035 Move some send stream configuration into webrtc::AudioSendStream. solenberg 2015-11-16 07:34:50 -08:00
  • 633a3aa26f ThreadUtils: Add joinUninterruptibly() with timeout magjed 2015-11-16 05:12:30 -08:00