Log - 18a6625221e4c4413c83a85e452a09d10716e18c - platform/external/webrtc

8f119ca Enable experiments with audio bitrate priority. by Jonas Olsson · 5 years ago
9356252 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
8d8ffdb Expose new audio stats on the API by Ivo Creusen · 6 years ago
44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 6 years ago
fc02a79 Revert "Piping audio interruption metrics to API layer" by Henrik Andreassson · 6 years ago
413ccc4 Stop DCHECK which occurs in ANA BitrateController when overhead is zero. by Bjorn A Mellem · 6 years ago
299c4e6 Piping audio interruption metrics to API layer by Henrik Lundin · 6 years ago
c35b6e6 Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData by Niels Möller · 6 years ago
30a276b Add RTP sequence number to TransportFeedbackObserver::AddPacket() by Erik Språng · 6 years ago
63658d0 Revert "Ensure that we always set values for min and max audio bitrate." by Daniel Lee · 6 years ago
e47aee3 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
cf96e0f Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. by Henrik Boström · 6 years ago
01738c6 Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. by Henrik Boström · 6 years ago
0810a7c Add base class NetworkPredictor and NetworkPredictorFactory and wire up. by Ying Wang · 6 years ago
2af5dcb Reland "Refactor FrameDecryptorInterface::Decrypt to use new API." by Benjamin Wright · 6 years ago
6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 6 years ago
7dd83e2 Revert "Refactor FrameDecryptorInterface::Decrypt to use new API." by Henrik Boström · 6 years ago
642aa81 Refactor FrameDecryptorInterface::Decrypt to use new API. by Benjamin Wright · 6 years ago
c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 6 years ago
31660fd Avoid using global task queue factory in audio/ unittests by Danil Chapovalov · 6 years ago
741daaf Move rtc::FunctionView to the public API by Artem Titov · 6 years ago
94b57c0 Cleanup BUILD.gn files from imports like foo:foo by Artem Titov · 6 years ago
53de725 Fix outdated android sdk path in tests. by Oleksandr Iakovenko · 6 years ago
ef1052a Reland "Move api/rtp_headers.h to its own build target." by Niels Möller · 6 years ago
2baef35 Revert "Move api/rtp_headers.h to its own build target." by Steve Anton · 6 years ago
a67050d Move api/rtp_headers.h to its own build target. by Niels Möller · 6 years ago
c936cb6 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h by Niels Möller · 6 years ago
f0b8dee Qualify cmath functions. by Mirko Bonadei · 6 years ago
17b050f Fixes ClangTidy errors in audio/ by Benjamin Wright · 6 years ago
471783f Remove rtc::QueuedTask alias, use webrtc::QueuedTask directly by Danil Chapovalov · 6 years ago
9ffb5df Removes unused mock_bitrate_controller. by Sebastian Jansson · 6 years ago
ad89528 Reland "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" by Danil Chapovalov · 6 years ago
42d8c93 Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" by Yves Gerey · 6 years ago
44dd9f2 Adds ChannelSend specific encoder task queue. by Sebastian Jansson · 6 years ago
304e9d2 Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current by Danil Chapovalov · 6 years ago
d5af402 Add overhead observers to MediaTransportInterface by Niels Möller · 6 years ago
87e2d78 Prepare for splitting FrameType into AudioFrameType and VideoFrameType by Niels Möller · 6 years ago
0b69826 Don't inject worker queue into send streams. by Sebastian Jansson · 6 years ago
8672cac Trigger audio bitrate allocation update on overhead change. by Sebastian Jansson · 6 years ago
ee5ccbc Move ownership of RTPSenderAudio to ChannelSend. by Niels Möller · 6 years ago
232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 6 years ago
c44f6cc Modernize RtpRtcp factory function: use unique_ptr as return type by Danil Chapovalov · 6 years ago
110c64b Delete unused key WebRTC-Audio-SendSideBwe-For-Video. by Christoffer Rodbro · 6 years ago
8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
7949f21 Revert "Removes lock from ChannelSend." by Sebastian Jansson · 6 years ago
977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
9b93447 Removes lock from ChannelSend. by Sebastian Jansson · 6 years ago
da6806c Injecting Clock into BitrateAllocator. by Sebastian Jansson · 6 years ago
fc52b91 Implicitly suppress //build/config/clang:find_bad_constructs. by Mirko Bonadei · 6 years ago
3cdd4d5 Fix: Ignore empty frames in Media Transport by Piotr (Peter) Slatala · 6 years ago
d8d3248 Reland "Delete test/constants.h" by Elad Alon · 6 years ago
4f36b7a Revert "Delete test/constants.h" by Oleh Prypin · 6 years ago
e049eba Revert "Add Sender and Receiver interfaces for MediaTransport audio" by Sergey Silkin · 6 years ago
0d8eed6 Add Sender and Receiver interfaces for MediaTransport audio by Niels Möller · 6 years ago
afb5dbb Update ACM to use RTPHeader instead of WebRtcRTPHeader by Niels Möller · 6 years ago
389b167 Delete test/constants.h by Elad Alon · 6 years ago
914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 6 years ago
397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 6 years ago
fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 6 years ago
14a7cf9 Adds CallEncoder to ChannelSend. by Sebastian Jansson · 6 years ago
464a557 Adds audio priority bitrate field trial parameter. by Sebastian Jansson · 6 years ago
3b50f9f Propagate base minimum delay to audio_receiver_stream by Ruslan Burakov · 6 years ago
1c54605 [clang-tidy] Apply performance-move-const-arg fixes (misc). by Mirko Bonadei · 6 years ago
626015d Make AudioSendStream to be OverheadObserver by Anton Sukhanov · 6 years ago
80a8687 [clang-tidy] Apply performance-move-const-arg fixes (mutable lambdas). by Mirko Bonadei · 6 years ago
432c833 Remove redundant check in channel_receive.cc. by Ruslan Burakov · 6 years ago
01dc691 Delete sequence number save and restore in ChannelSend. by Niels Möller · 6 years ago
c84f661 Stop using Googletest legacy APIs. by Mirko Bonadei · 6 years ago
fe055c1 [clang-tidy] Apply modernize-use-override fixes. by Mirko Bonadei · 6 years ago
b4977de Receive-side ready for multiple channels. by Alex Loiko · 6 years ago
d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 6 years ago
79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
5c2f1f0 Add some missing includes and dependencies. by Bjorn Terelius · 6 years ago
e7d08df Fix chromium roll into WebRTC. by Artem Titov · 6 years ago
0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
36faf0b Delete setting of unused variable nack_window_ms by Niels Möller · 6 years ago
10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
ba50223 Make voiceengine/audio transport stop using voice_detection() interface by Sam Zackrisson · 6 years ago
53eae87 Add PeerConnection option to enable RTX handling in the audio jitter buffer. by Jakob Ivarsson · 6 years ago
ac63ac7 Update refcounting of AudioState to use rtc::RefCountedObject by Niels Möller · 6 years ago
40d5533 Include absl/memory/memory.h if absl::make_unique is used by Steve Anton · 6 years ago
77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
31d8b52 Delete unneeded includes of rtc_base/stringutils.h. by Niels Möller · 6 years ago
3d2ed19 Remove Transport implementation from ChannelSend by Fredrik Solenberg · 6 years ago
f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
2a977cf For audio receive channel use default max reordering threshold instead of 0 by Danil Chapovalov · 6 years ago
10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
e977199 Delete ChannelSend::RegisterTransport, replacing by construction argument by Niels Möller · 6 years ago
ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
8ce0d2b In ReceiveStatistic require callbacks during construction by Danil Chapovalov · 6 years ago
e3abb81 Decouple //rtc_base:rtc_base_tests_utils from gunit. by Mirko Bonadei · 6 years ago
8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
254d869 Routing BitrateAllocationUpdate to audio codec. by Sebastian Jansson · 6 years ago
8b5d9d8 Remove the audio/video split for the RTCP report intervals. by Jiawei Ou · 6 years ago
13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
985a1f3 Add const or GUARDED_BY on a few ChannelSend members by Niels Möller · 6 years ago