Log - 88a38e32e78f49510845490e8529241281d5f366 - platform/external/webrtc

57daeb7 Reland "Moved congestion controller to task queue." by Sebastian Jansson · 7 years ago
cc7125f Sets sending status for active RtpRtcp modules. by Seth Hampson · 7 years ago
5a503b0 Revert "Moved congestion controller to task queue." by Sebastian Jansson · 7 years ago
0cbcba7 Moved congestion controller to task queue. by Sebastian Jansson · 7 years ago
970b088 Reland "Break up rtc_event_log_api to solve circular dependencies." by Qingsi Wang · 7 years ago
06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
3587b83 Make RTCP report interval configurable by Jiawei Ou · 7 years ago
75df728 Revert "Break up rtc_event_log_api to solve circular dependencies." by Mirko Bonadei · 7 years ago
001546d Break up rtc_event_log_api to solve circular dependencies. by Qingsi Wang · 7 years ago
f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
dee9191 Use rtc::ToString instead of std::to_string by Jiawei Ou · 7 years ago
65ce311 Removing useless dependencies on //testing/gmock. by Mirko Bonadei · 7 years ago
3b790f3 Make fec controller plug-able. by Ying Wang · 7 years ago
2ffe3e8 Reland Use runtime enabled features API to enable dual stream mode by Ilya Nikolaevskiy · 7 years ago
6539f69 Add VideoSendStream::Config::EncoderSettings::experiment_cpu_load_estimator. by Niels Möller · 7 years ago
c1094eb Revert "Use runtime enabled features API to enable dual stream mode" by Lu Liu · 7 years ago
6f011df Use runtime enabled features API to enable dual stream mode by Ilya Nikolaevskiy · 7 years ago
a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
9c68613 Update gn files to support Mozilla build by Dan Minor · 7 years ago
8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
fedc00c Optional: Use nullopt and implicit construction in /call by Oskar Sundbom · 7 years ago
3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
a7f2d84 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" by Per Kjellander · 7 years ago
c73e1f4 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" by Per Kjellander · 7 years ago
588c548 GN rtc_* templates: Set default visibility to webrtc_root + "/*" by Karl Wiberg · 7 years ago
dfe9ffc Added active field to constructor and ToString() of VideoStream. by Seth Hampson · 7 years ago
62337e5 Use AudioProcessingBuilder everywhere AudioProcessing is created. by Ivo Creusen · 7 years ago
7f331fa Add metric name for MinVideoAndAudioBitRate. by Edward Lemur · 7 years ago
24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
e66572b Reland "iOS: Save perf results under Documents/perf_result.json" by Edward Lemur · 7 years ago
947f3fe Fix reporting of perf results on PlaysOutAudioAndVideoInSync* tests by Edward Lemur · 7 years ago
dd3987f Add _[no]red suffix to RampUpTests. by Edward Lemur · 7 years ago
9e19403 Move videosourceinterface to api. by Patrik Höglund · 7 years ago
be214a2 Move videosinkinterface.h to common_video to solve a circular dep. by Patrik Höglund · 7 years ago
6213929 Add missing files to audio_processing. by Patrik Höglund · 7 years ago
8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
36193c3 Adds active field to VideoStream struct. by Seth Hampson · 7 years ago
f32795e Updates to video config to allow changes in google3 tests, in order to not break anything. by Seth Hampson · 7 years ago
2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
3e11343 Fix circular dependencies in webrtc_common. by Patrik Höglund · 7 years ago
712989d Revert "Reland "iOS: Save perf results under Documents/perf_result.json"" by Mirko Bonadei · 7 years ago
a8005cf Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
8b886bb Reland "iOS: Save perf results under Documents/perf_result.json" by Edward Lemur · 7 years ago
d37709b Revert "Fix circular dependencies between optional, array_view, and rtc_base." by Patrik Höglund · 7 years ago
081c651 Revert "iOS: Save perf results under Documents/perf_result.json" by Rasmus Brandt · 7 years ago
a9e0924 Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
10a8e7a iOS: Save perf results under Documents/perf_result.json by Edward Lemur · 7 years ago
cbf5b73 Explicitly convert size_t to int in Call::DeliverPacket by Danil Chapovalov · 7 years ago
292a73e Deliver packet to Call as rtc::CopyOnWriteBuffer by Danil Chapovalov · 7 years ago
a498ae8 Stop using public_deps in system_wrappers. by Mirko Bonadei · 7 years ago
b5728d9 Stop using public_deps in modules/rtp_rtcp. by Mirko Bonadei · 7 years ago
03d6f2f Stop using public_deps in modules/audio_mixer. by Mirko Bonadei · 7 years ago
a0e1a55 Stop using public_deps in the call module. by Mirko Bonadei · 7 years ago
cf73c96 Add AudioDeviceModule to AudioState::Config. by Fredrik Solenberg · 7 years ago
ad62792 Fixing hidden dependencies. by Mirko Bonadei · 7 years ago
1eb051c Made functions on BitrateAllocator::ObserverConfig member functions by srte · 7 years ago
2f06168 Make PrintResultList receive a vector of doubles instead of a string. by Edward Lemur · 7 years ago
d0e196b Adding two tests: by Alex Narest · 7 years ago
56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
a6092a9 Deprecated the Get BitrateController method by srte · 7 years ago
e40468b Move some numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 7 years ago
d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
63e6072 Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine. by Fredrik Solenberg · 7 years ago
c3ed630 Add stats googHasEnteredLowResolution. by Åsa Persson · 7 years ago
e0b2ff5 Add kTransmissionMaxBitrateMultiplier logic to audio priority bitrate allocation strategy similarly to default bitrate allocation logic. by Alex Narest · 7 years ago
fe73d6a Extended the bitrate allocator to allow allocation to tracks based upon priorities which are planned to be defined as a relative bitrate in the RTCRtpEncodingParameters. by Seth Hampson · 7 years ago
c0e6804 Fix deps of audio:audio_tests. by Patrik Höglund · 7 years ago
61a7b14 Removing conditional visibility. by Mirko Bonadei · 7 years ago
675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
fd6c091 Delete deprecated constructor of SendSideCongestionController. by Niels Möller · 7 years ago
f3850f6 Voice Engine: Require caller to supply an AudioDecoderFactory by Karl Wiberg · 7 years ago
5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
de69145 Remove pbos@webrtc.org from all OWNERS. by Peter Boström · 7 years ago
990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
d79314f Reland "Add fine grained dropped video frames counters on sending side" by Ilya Nikolaevskiy · 7 years ago
1c1a681 Revert "Add fine grained dropped video frames counters on sending side" by Ilya Nikolaevskiy · 7 years ago
78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
4b1a363 Add fine grained dropped video frames counters on sending side by Ilya Nikolaevskiy · 7 years ago
a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago
54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 7 years ago
b3944f0 Media track ID visibility at BWE level by Alex Narest · 7 years ago
05d9822 Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss. by Taylor Brandstetter · 7 years ago
b709cf8 Remove Call::ParseRtpPacket by Danil Chapovalov · 7 years ago
245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 7 years ago
4bece9a Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled. by Åsa Persson · 7 years ago
4332d09 Reland "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
a32dd01 Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
39cefdb Revert "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
68007e9 Reland "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
4a87e1c Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead by Elad Alon · 7 years ago
729b910 Revert "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
2209b90 Remove WEBRTC_TRACE. by Fredrik Solenberg · 7 years ago
d4404c2 Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
34cdd2d Remove AudioDeviceObserver and make ADM not inherit from the Module interface. by Fredrik Solenberg · 7 years ago
b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
440216f Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets. by Bjorn Terelius · 7 years ago