Log - f7795df4261877af9eaf982cccfafc3fc52aeb4a - fp2-dev/platform/external/chromium_org/third_party/webrtc

b16a722 Reduced kMaxSampleDiffMs (limit to 22fps). by asapersson@webrtc.org · 10 years ago
a229768 Fix failing test introduced with r6111. by stefan@webrtc.org · 10 years ago
305fd94 Fixes log spam introduced with r6041. by stefan@webrtc.org · 10 years ago
0d47fe1 Raise kViEMaxNumberOfChannels from 32 to 64 by wu@webrtc.org · 10 years ago
569487d Updated WebRTC version to 3.53 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
60f1422 Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. by wu@webrtc.org · 10 years ago
73c2412 Remove timestamp_extrapolator's dependency to Clock and vcm defines. by wu@webrtc.org · 10 years ago
1a9e6ac Pointers were not dereferenced in GetRtpStatistics. by asapersson@webrtc.org · 10 years ago
42fe6b3 Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. by stefan@webrtc.org · 10 years ago
c6cfc5c Upping start bitrate to min, if set to a lower value i SetSendCodec. by mflodman@webrtc.org · 10 years ago
73e1a8b Disable flaky RunsRtpRtcpTestWIthoutErrors. by pbos@webrtc.org · 10 years ago
b0079ed Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
0a5fd54 Casting char to int in logs. by asapersson@webrtc.org · 10 years ago
b991cd0 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
a1626fe Propagate capture ntp timestamp from rtp to renderer. by wu@webrtc.org · 10 years ago
6b1114a Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
b18bff5 Cleaned up logging in video_coding. by stefan@webrtc.org · 10 years ago
4b0cd7f Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
ff46b81 Updated WebRTC version to 3.52 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
44c9b9a Log Fixit for parts of video_engine folder. by mflodman@webrtc.org · 10 years ago
6c57efd Re-submit: rev5775 by andresp@webrtc.org · 10 years ago
37f807f 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. by solenberg@webrtc.org · 10 years ago
765ea72 Revert 5775 "Modify bitrate controller to update bitrate based o..." by andrew@webrtc.org · 10 years ago
f50914a Removing VideoCodecDerived and moving methods inside VideoCodec. by mallinath@webrtc.org · 10 years ago
0ac0bca Updated WebRTC version to 3.51 by elham@webrtc.org · 10 years ago
539bbde Modify bitrate controller to update bitrate based on process call and not by andresp@webrtc.org · 10 years ago
1f49208 Adding API for setting bandwidth estimation configurations. by stefan@webrtc.org · 10 years ago
1a19092 Add configuration for ability to use the encode usage measure for triggering overuse/underuse. by asapersson@webrtc.org · 10 years ago
50ac4d6 Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
85101db Have changes to REMB trigger RTCP to be sent immediately. by stefan@webrtc.org · 10 years ago
5f804f8 VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
9b2b8ec Add AIMD option to BWE API. by stefan@webrtc.org · 10 years ago
209791d Refactor in BitrateController module. by andresp@webrtc.org · 10 years ago
40fee00 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 10 years ago
27bd3be Add ability to configure cpu overuse options via an API. by asapersson@webrtc.org · 10 years ago
f9d5709 Fixes RTX related bugs. by stefan@webrtc.org · 10 years ago
bef6e62 Simplify pacer interface. by pbos@webrtc.org · 10 years ago
3a70e6e Fix a deadlock in ViEEncoder::DeliverFrame. by wuchengli@chromium.org · 10 years ago
9420a1f Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
f35f098 Avoid crash in ViEEncoder::DeRegisterExternalEncoder(). by fischman@webrtc.org · 10 years ago
0bf5a2f Adding a new ramp-up-down-up test by henrik.lundin@webrtc.org · 10 years ago
ecee063 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 10 years ago
c0d56c0 Fix to get total number of sent and received rtcp packets. by asapersson@webrtc.org · 10 years ago
23c8d6b Updated WebRTC version to 3.50 TBR= wu@webrtc.org by elham@webrtc.org · 10 years ago
072bab2 Modified overuse detection thresholds. by asapersson@webrtc.org · 10 years ago
2fa9f7e Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
ae50521 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
15e3511 Reset estimate if no frame has been seen for a certain time (to avoid large jitter if stop sending). by asapersson@webrtc.org · 10 years ago
46b22d8 Adding a critical section missing in r5543. by stefan@webrtc.org · 10 years ago
8e98655 Increase overuse and normal use thresholds for Mac. by asapersson@webrtc.org · 10 years ago
8cb4c8d Fixes a race when writing to send_padding_. by stefan@webrtc.org · 10 years ago
6cfc58d Set pacing bitrates in SetEncoder. by pbos@webrtc.org · 10 years ago
ddbd31e Remove ViE external encryption API. by solenberg@webrtc.org · 10 years ago
a68379b Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
49e9e15 Connect webrtc::Config to WrappingBitrateEstimator by henrik.lundin@webrtc.org · 10 years ago
a1e140d Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." by mallinath@webrtc.org · 10 years ago
c091c50 Revert 5421 "Fix deadlock on register/unregister observer while ..." by mallinath@webrtc.org · 11 years ago
aa2c3ae Fix deadlock on register/unregister observer while there is a an going callback. by andresp@webrtc.org · 11 years ago
224933c Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
64339f0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 11 years ago
c1792c5 Roll Chromium 238260 -> 243863 by wjia@webrtc.org · 11 years ago
b95f445 Updated Webrtc version to 3.49 by elham@webrtc.org · 11 years ago
9c8f391 Removes usage of ListWrapper from several files. by henrike@webrtc.org · 11 years ago
ca72300 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
7e4053c Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
d3f0617 If the configured start bitrate is higher than the configures max by mflodman@webrtc.org · 11 years ago
4a185e9 Race condition in ViECapturer::RegisterObserver by sprang@webrtc.org · 11 years ago
e83367b Update WebRTC to version 3.48 by tnakamura@webrtc.org · 11 years ago
acc2e43 Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
cd117d2 Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
ef1f6c3 Remove media_file from VideoEngine dependencies. by pbos@webrtc.org · 11 years ago
39139dc Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
0af1d21 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
ee867fa Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
0e4512b Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
e4d538a Make sure channels in the same call are in the same channel group. by mflodman@webrtc.org · 11 years ago
e6dc4ff Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
3a4fc4b Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
9b3d2bf Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
cde78d6 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
a4670a1 Complete rewrite of demo application. by henrike@webrtc.org · 11 years ago
0ceb51f Remove overloaded CpuOveruseMeasure function. by asapersson@webrtc.org · 11 years ago
7123a80 Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
66e84b0 Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
894dab9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
f1d22d4 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
ed8c496 Fraction lost statistics not being reported by sprang@webrtc.org · 11 years ago
cf5c552 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
8db148e Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
adc238a Fixing the android build by henrik.lundin@webrtc.org · 11 years ago
b669e60 Remove default implementations for SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
3bcea52 Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics. by stefan@webrtc.org · 11 years ago
8911937 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
5459e0b Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
382cfdd Removing DropDeltaAfterKey functionality which is unused. by andresp@webrtc.org · 11 years ago
9435a17 Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
f2c136b Added a delay measurement, measures the time between an incoming captured frame until the frame is being processed. Measures the delay per second. by asapersson@webrtc.org · 11 years ago
da3ae7c Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
0e2571d Lock frame in ViECapturer::IncomingFrameI420. by pbos@webrtc.org · 11 years ago
801822c Ensure that no packet stays in the pacer queue for longer than 2 seconds. by stefan@webrtc.org · 11 years ago