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  • 721ef633d0 Remove the codec_type_ member from AudioDecoder kwiberg@webrtc.org 2014-11-04 11:51:46 +00:00
  • c2dd5ee2c0 Prepare for removal of PeerConnectionObserver::OnError. Prepare for removal of constraints to PeerConnection::AddStream. perkj@webrtc.org 2014-11-04 11:31:29 +00:00
  • f37145f685 Enables AIMD control by default. stefan@webrtc.org 2014-11-04 09:08:21 +00:00
  • b0f4b3da05 Improving error message from neteq_rtpplay henrik.lundin@webrtc.org 2014-11-04 08:53:10 +00:00
  • a663d90ae3 (Auto)update libjingle 79104430-> 79104922 buildbot@webrtc.org 2014-11-03 22:29:18 +00:00
  • 5f38c8d1b8 Android AppRTCDemo improvements: - Add a room list to ConnectActivity with buttons to add/remove rooms. - Add loopback call button. - Add option to toggle full screen / letterbox video. - Add camera fps settings. - Fix device to landscape orientation for HD video until issue 3936 will be fixed. - Fix a few crashes by avoiding calling peer connection and GAE signaling function while connection is closing. - Better handling GAE channel error - catch channel exceptions and display dialog with error messages. glaznev@webrtc.org 2014-11-03 22:18:52 +00:00
  • 5804936052 Add format members to AudioConverter for DCHECKing. andrew@webrtc.org 2014-11-03 21:32:14 +00:00
  • e451b756a8 Update rate control parameter in vp9 test. marpan@webrtc.org 2014-11-03 21:26:08 +00:00
  • 4765ca55f9 Roll chromium_revision: 28d1981..d3db2ff marpan@webrtc.org 2014-11-03 20:10:26 +00:00
  • f866b2d9f9 Restore the void return type on WriteWavHeader. andrew@webrtc.org 2014-11-03 18:20:06 +00:00
  • b81e304ac0 replace inline assembly WebRtcNsx_AnalysisUpdate by intrinsics. andrew@webrtc.org 2014-11-03 17:17:51 +00:00
  • f9471807a2 Add Opus support to neteq_rtpplay henrik.lundin@webrtc.org 2014-11-03 15:19:58 +00:00
  • 96a93259b3 Implement external decoder support in WebRtcVideoEngine2. pbos@webrtc.org 2014-11-03 14:46:44 +00:00
  • 548b228c91 Add UMA metrics for the initial (after two seconds) packet loss, round-trip time and bandwidth estimate of a WebRTC call. stefan@webrtc.org 2014-11-03 14:42:43 +00:00
  • 96dc685143 Add stats for video: - number of sent/received RTCP NACK/FIR/PLI per minute - percentage of unique sent/received NACK requests - percentage of discarded/duplicated packets by the jitter buffer - permille of sent/received key frames asapersson@webrtc.org 2014-11-03 14:40:38 +00:00
  • 2236267b5e Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSan henrik.lundin@webrtc.org 2014-11-03 13:38:50 +00:00
  • bf09976e86 Add more sanity checks to workaround the unidentified problem that CaptureThread is still running while related resouces are destroyed already. braveyao@webrtc.org 2014-11-03 09:58:40 +00:00
  • ed45896759 Adjust/increase rate control thresold for a vp9 test. marpan@webrtc.org 2014-11-01 07:08:52 +00:00
  • 5b88317820 Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. marpan@webrtc.org 2014-11-01 06:10:48 +00:00
  • 5072e0f6cd Update Android projects to API level 21. kjellander@webrtc.org 2014-10-31 23:26:10 +00:00
  • 818c9f9e14 replace inline assembly WebRtcNsx_SynthesisUpdateNeon by intrinsics. andrew@webrtc.org 2014-10-31 22:07:35 +00:00
  • a3ed713dad Add a WavReader counterpart to WavWriter. andrew@webrtc.org 2014-10-31 21:51:03 +00:00
  • c2c94a9a9f Change default JVM location to /usr/lib/jvm/java-7-openjdk-amd64 kjellander@webrtc.org 2014-10-31 19:01:41 +00:00
  • 78c222bfae Update all .isolate files for the new format. kjellander@webrtc.org 2014-10-31 18:08:09 +00:00
  • 8a130c1084 Update Android projects to API level 20. kjellander@webrtc.org 2014-10-31 17:13:37 +00:00
  • 053c6abf8d Fix N7 camera aspect ratio. glaznev@webrtc.org 2014-10-31 16:58:58 +00:00
  • 508c91683c Build fix for MIPS32R6. andrew@webrtc.org 2014-10-31 16:26:17 +00:00
  • cc476aa038 Fix a name collision with Android libc++ andrew@webrtc.org 2014-10-31 16:01:25 +00:00
  • b7ed7799e7 Implement conference-mode temporal-layer screencast. pbos@webrtc.org 2014-10-31 13:08:10 +00:00
  • 3bf3d238c8 Configure A/V sync in WebRtcVideoEngine2. pbos@webrtc.org 2014-10-31 12:59:34 +00:00
  • 4abadab708 Simplify bwe tests. stefan@webrtc.org 2014-10-31 10:47:12 +00:00
  • 2dc6f3154d Adapting bitrate according to maxplaybackrate for Opus. minyue@webrtc.org 2014-10-31 05:33:10 +00:00
  • 8328e7c44d Revert "Revert part of r7561, "Refactor audio conversion functions."" andrew@webrtc.org 2014-10-31 04:58:14 +00:00
  • 14146e40aa arm64 iOS build. tkchin@webrtc.org 2014-10-31 00:14:39 +00:00
  • 50ca986bc1 Improve the logging when a TCP connection is deleted. jiayl@webrtc.org 2014-10-30 23:50:54 +00:00
  • d0cf68ee37 Add 15 fps support for Android devices with missing 15 fps camera mode. glaznev@webrtc.org 2014-10-30 18:38:26 +00:00
  • 8aa4d2d2cd Creating a C++ wrapper class for VAD henrik.lundin@webrtc.org 2014-10-30 13:23:25 +00:00
  • bcfb4d0403 Revert part of r7561, "Refactor audio conversion functions." kwiberg@webrtc.org 2014-10-30 11:16:06 +00:00
  • 8219529b98 Cleaning up r7562-7567. minyue@webrtc.org 2014-10-30 08:23:54 +00:00
  • 879fac81d1 (Auto)update libjingle 78822708-> 78823675 buildbot@webrtc.org 2014-10-30 07:50:13 +00:00
  • 5f73a37597 Revert 7563 "before rebase" due to wrong submission minyue@webrtc.org 2014-10-30 07:49:58 +00:00
  • c11cc8d947 Revert 7564 "to submit" due to wrong submission minyue@webrtc.org 2014-10-30 07:46:47 +00:00
  • de386bf67b to submit minyue@webrtc.org 2014-10-30 07:20:09 +00:00
  • c673bb9f29 before rebase minyue@webrtc.org 2014-10-30 07:19:57 +00:00
  • 0b62672576 adding default rates minyue@webrtc.org 2014-10-30 07:19:49 +00:00
  • 4fc4addc81 Refactor audio conversion functions. andrew@webrtc.org 2014-10-30 03:40:10 +00:00
  • 776e6f289c Use external VideoDecoders in VideoReceiveStream. pbos@webrtc.org 2014-10-29 15:28:39 +00:00
  • 2dd3134e50 Add stats for duplicate sent and received NACK requests. asapersson@webrtc.org 2014-10-29 12:42:30 +00:00
  • f567095f62 common_audio: Removed macro WEBRTC_SPL_RSHIFT_W32 bjornv@webrtc.org 2014-10-29 10:29:16 +00:00
  • 7f10513efc Remove unused code in overuse detector. asapersson@webrtc.org 2014-10-29 10:05:21 +00:00
  • decd9306ae AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket kwiberg@webrtc.org 2014-10-29 08:38:50 +00:00
  • cfe3845b66 Enable G.722 for Chromium builds henrik.lundin@webrtc.org 2014-10-29 08:32:44 +00:00
  • 1abc146aa5 (Auto)update libjingle 78738075-> 78738103 buildbot@webrtc.org 2014-10-29 08:14:14 +00:00
  • 7998089789 ApprtDemo Android: Switch between front and back camera. This adds a UI icon for switching between the front and back camera. This cl adds the possibility to change between the front and back camera while in a call or before the other end have connected. perkj@webrtc.org 2014-10-29 08:10:03 +00:00
  • 663fdd02fd Make an AudioEncoder subclass for Opus kwiberg@webrtc.org 2014-10-29 07:28:36 +00:00
  • 2623695dfb Renaming bandwidth to bitrate in webrtcvoiceengine. minyue@webrtc.org 2014-10-29 02:27:08 +00:00
  • ffeaeed8c1 Make NSinst_t* const and rename to self in ns_core aluebs@webrtc.org 2014-10-28 22:52:09 +00:00
  • 269fb4bc90 move xmpp and p2p to webrtc Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder. henrike@webrtc.org 2014-10-28 22:20:11 +00:00
  • 8b1b23f8f8 Make local functions static and dropWebRtcNs_ in ns_core aluebs@webrtc.org 2014-10-28 21:06:57 +00:00
  • 28b54671cb Make all comments whole sentences in ns_core aluebs@webrtc.org 2014-10-28 20:56:53 +00:00
  • bd6bdca57f scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots. henrike@webrtc.org 2014-10-28 18:06:42 +00:00
  • ae694effd8 (Auto)update libjingle 78642371-> 78680406 buildbot@webrtc.org 2014-10-28 17:37:17 +00:00
  • a296725d0e audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>" bjornv@webrtc.org 2014-10-28 13:05:43 +00:00
  • 67ca26e087 common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16 bjornv@webrtc.org 2014-10-28 13:03:10 +00:00
  • ff8a98e352 Use neteq_unittest_tools in audio_decoder_unittests henrik.lundin@webrtc.org 2014-10-28 09:47:13 +00:00
  • 820efd5b55 Fix double backslashes in incoming_video_stream.cc perkj@webrtc.org 2014-10-28 08:47:16 +00:00
  • fbd55cb27d (Auto)update libjingle 78616359-> 78642371 buildbot@webrtc.org 2014-10-28 05:35:35 +00:00
  • f15dee6980 Check if a datachannel in the current local description is an sctp channel before assuming rtp. When generating an offer from a local description when 'sctp' is not explicitly set in the media session options, we were generating an offer with an RTP datachannel even though the channel in the local description was already sctp. tommi@webrtc.org 2014-10-27 22:15:04 +00:00
  • aada86b261 Add a simple AudioConverter class. andrew@webrtc.org 2014-10-27 18:18:17 +00:00
  • 33a0e2d7ef Only configure the SSL library in one place. henrike@webrtc.org 2014-10-27 18:13:40 +00:00
  • aca5803b19 Move (test) RtpFileReader to a lightweight target. pbos@webrtc.org 2014-10-27 18:01:03 +00:00
  • b787f4c593 Move scoped_ptr "free" functions into the webrtc namespace. andrew@webrtc.org 2014-10-27 17:42:22 +00:00
  • 243eb8e9af Adding setting screen to AppRTCDemo. glaznev@webrtc.org 2014-10-27 17:22:15 +00:00
  • 068b529f46 (Auto)update libjingle 78583324-> 78583691 buildbot@webrtc.org 2014-10-27 16:20:42 +00:00
  • df429882af Upgrade our scoped_ptr copy to match Chromium's latest. andrew@webrtc.org 2014-10-27 16:12:38 +00:00
  • 2e7ee4b28b Fix the SrtpFilter crash caused by two local offers. pthatcher@webrtc.org 2014-10-27 16:10:29 +00:00
  • efc82c2c73 Implement screencast settings for WebRtcVideoEngine2. pbos@webrtc.org 2014-10-27 13:58:00 +00:00
  • a37f1dd6b8 Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile henrik.lundin@webrtc.org 2014-10-27 12:58:18 +00:00
  • 0552356fda isacfix: Refactor big-endian reading and writing kwiberg@webrtc.org 2014-10-27 11:25:37 +00:00
  • 9fed099208 Increase max trace message size to 1024 characters. pbos@webrtc.org 2014-10-27 09:31:05 +00:00
  • c86ec3e3bc Fix ::~LogMessage to print as a string. pbos@webrtc.org 2014-10-27 09:22:03 +00:00
  • 1732df6129 Use flags set by the port allocator. braveyao@webrtc.org 2014-10-27 03:01:37 +00:00
  • 3b839d008f PRESUBMIT: Add linux_msan to default trybots. kjellander@webrtc.org 2014-10-24 21:41:24 +00:00
  • 3f7bcc126d (Auto)update libjingle 78430441-> 78445452 buildbot@webrtc.org 2014-10-24 17:26:28 +00:00
  • c7ed8db7fd (Auto)update libjingle 78427027-> 78430441 buildbot@webrtc.org 2014-10-24 12:59:08 +00:00
  • 470988742a Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected. perkj@webrtc.org 2014-10-24 11:38:19 +00:00
  • 39b1743116 Adding the subtool rtcBot report visualizer houssainy@google.com 2014-10-24 09:26:16 +00:00
  • ad3b5a5c16 Move min transmit bitrate to VideoEncoderConfig. pbos@webrtc.org 2014-10-24 09:23:21 +00:00
  • c9d6d14020 patch from issue 25469004 pthatcher@webrtc.org 2014-10-23 23:37:22 +00:00
  • 8fe75ee234 (Auto)update libjingle 78381351-> 78389679 buildbot@webrtc.org 2014-10-23 23:07:23 +00:00
  • fb5e9fc44e (Auto)update libjingle 78344087-> 78381351 buildbot@webrtc.org 2014-10-23 21:36:17 +00:00
  • 7e19a11a71 Break out WebRtcNs_ComputeDdUpdate function in ns_core aluebs@webrtc.org 2014-10-23 19:54:33 +00:00
  • f8ea0d5518 Break out WebRtcNs_UpdateNoise function in ns_core aluebs@webrtc.org 2014-10-23 19:49:42 +00:00
  • 799e88ae19 Break out FFT function in ns_core aluebs@webrtc.org 2014-10-23 19:36:42 +00:00
  • 8454ad88ed Break out ComputeSnr function in ns_core aluebs@webrtc.org 2014-10-23 19:34:14 +00:00
  • 0d3e254c89 Adding three video conference bots test houssainy@google.com 2014-10-23 16:45:07 +00:00
  • 0e19d0c2aa Adding file from test.webrtc.org domain to be downloaded houssainy@google.com 2014-10-23 15:41:30 +00:00
  • 580d367b14 Add macros and APIs for webrtc histograms. asapersson@webrtc.org 2014-10-23 12:57:56 +00:00
  • 9d446f2e16 (Auto)update libjingle 78296920-> 78342456 buildbot@webrtc.org 2014-10-23 12:22:06 +00:00
  • 8539bd0184 Download full Chromium checkouts by default kjellander@webrtc.org 2014-10-23 12:17:58 +00:00