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gerrit-public.fairphone.software
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platform
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external
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webrtc
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ab1aee0be4d6ce2ca38cccd5763d6ab9ef7b1d71
ab1aee0
Reland "Deprecate the adaptive level controller"
by Sam Zackrisson
· 6 years ago
e6826d2
Add configurable connectivity check intervals.
by Qingsi Wang
· 6 years ago
6d72c32
Revert "Rework rtp packet history"
by Taylor Brandstetter
· 6 years ago
8493594
Cleanup of TransportFeedbackObserver interface
by Erik Språng
· 6 years ago
25e022f
Deliver cached stats reports asynchronously.
by Taylor Brandstetter
· 6 years ago
6328d7c
Rework rtp packet history
by Erik Språng
· 6 years ago
d062a3c
Prepare VideoProcessor for async simulcast support.
by Rasmus Brandt
· 6 years ago
2222921
Roll chromium_revision 90346c9657..6f54ca247b (541700:541802)
by Autoroller
· 6 years ago
ad09d74
Extend the audioproc_f input parameters to match what is supported by AEC3
by Per Åhgren
· 6 years ago
a51bbd8
Adding PRESUBMIT check to stop using istream, ostream and sstream.
by Mirko Bonadei
· 6 years ago
564299a
Fix applicationContext passed to NetworkMonitor from NDK being unused.
by Sami Kalliomäki
· 6 years ago
bbf1465
Delete dead code for video quality calculation.
by Rasmus Brandt
· 6 years ago
9823ee4
Fix native api in preparation for native_api example.
by Anders Carlsson
· 6 years ago
a2d89fc
Remove ObjC static library target.
by Kári Tristan Helgason
· 6 years ago
a9bb99e
Roll chromium_revision 344076e53b..90346c9657 (541591:541700)
by Autoroller
· 6 years ago
081136f
Revert "Reland "Add multiplex case to webrtc_perf_tests""
by Taylor Brandstetter
· 6 years ago
7c5bc1c
Reland "Add multiplex case to webrtc_perf_tests"
by Emircan Uysaler
· 6 years ago
8861da8
Roll chromium_revision f5178a2db6..344076e53b (541490:541591)
by Autoroller
· 6 years ago
5b32f23
Securely clear memory containing key information / passwords before freeing.
by Joachim Bauch
· 6 years ago
fdd5eae
Fix incorrect explicit marks
by Emircan Uysaler
· 6 years ago
5aac372
Revert "Add multiplex case to webrtc_perf_tests"
by Emircan Uysaler
· 6 years ago
a06e919
Removing interface to access pacer via SSCC.
by Sebastian Jansson
· 6 years ago
d90a7e8
Add multiplex case to webrtc_perf_tests
by Emircan Uysaler
· 6 years ago
98bb968
Roll chromium_revision ec5ab775db..f5178a2db6 (541385:541490)
by Autoroller
· 6 years ago
12edf4c
Separate build target for rtc_base/numerics/safe_minmax.h
by Karl Wiberg
· 6 years ago
1b2e90b
Replaced DestructAndGetRtpStateTask with lambda.
by Sebastian Jansson
· 6 years ago
e4d79c7
Replaced EncoderReconfiguredTask with lambda.
by Sebastian Jansson
· 6 years ago
0f03973
Separate test/fake_audio_device on API and implementation. Step 1.
by Artem Titov
· 6 years ago
2e18061
Count key protocol for all media sections
by Harald Alvestrand
· 6 years ago
db4fa4b
Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 3.
by philipel
· 6 years ago
7dbb701
Fix crash when setting duplicate receive codecs.
by Anders Carlsson
· 6 years ago
69d2dc9
Roll chromium_revision 92107f0efb..ec5ab775db (541279:541385)
by Autoroller
· 6 years ago
6deb65c
Move hardwarevideoencoderfactory jni for building w/o SW codecs.
by Anders Carlsson
· 6 years ago
b46c552
Roll chromium_revision c005394adc..92107f0efb (541159:541279)
by Autoroller
· 6 years ago
9a58cc0
Update the android AppRTC to use PeerConnection Unified Plan API.
by Seth Hampson
· 6 years ago
c384e14
Add new PeerConnection APIs to the Java SDK.
by Seth Hampson
· 6 years ago
513449e
Changes name of RtpTransceiverInit's stream_labels to stream_ids.
by Seth Hampson
· 6 years ago
8e20f17
Report UMA metrics for received SDP format
by Steve Anton
· 6 years ago
cd7d86b
Roll chromium_revision a76818d7bf..c005394adc (541053:541159)
by Autoroller
· 6 years ago
87d5a74
Fix crash that occurs if GetStats is called from within OnStatsDelivered
by Taylor Brandstetter
· 6 years ago
12eb858
Separating the AEC3 suppressor gain rampup behavior for call startup and in-call resets
by Per Åhgren
· 6 years ago
e3927c5
Allow to turn RtcpTransciever on and off at runtime.
by Danil Chapovalov
· 6 years ago
70473fc
Reland "Add hugeFramesSent GetStats metric"
by Ilya Nikolaevskiy
· 6 years ago
1e0c804
Make DTMF sender wait before sending the first tone.
by Harald Alvestrand
· 6 years ago
52f8188
Revert "Deprecate the adaptive level controller"
by Sam Zackrisson
· 6 years ago
2e1d784
Delete the VideoCodec::plName string.
by Niels Möller
· 6 years ago
f9fbce9
Add back OWNERS entris that went missing
by Karl Wiberg
· 6 years ago
6f37ed7
Deprecate the adaptive level controller
by Sam Zackrisson
· 6 years ago
9771c50
Clear the FrameBuffer if it's full and a keyframe is being inserted.
by philipel
· 6 years ago
8ddc2e6
Revert "Add hugeFramesSent GetStats metric"
by Max Morin
· 6 years ago
87f827d
Roll chromium_revision 4f34e798b7..a76818d7bf (540949:541053)
by Autoroller
· 6 years ago
6b54228
Adapt audio codec factory templates to the new codec pair ID arguments
by Karl Wiberg
· 6 years ago
94ffe90
Roll chromium_revision 6562397ff8..4f34e798b7 (540826:540949)
by Autoroller
· 6 years ago
b2bfba6
Declare the RtpHeaderExtensionMap* as const in RtpHeaderParser::Parse.
by Bjorn Terelius
· 6 years ago
0807d15
Remove more dead code from BaseChannel
by Steve Anton
· 6 years ago
0867260
Fixing data race on vptr of Thread subclasses.
by Taylor Brandstetter
· 6 years ago
98cd810
Production code: Pass codec ID argument to audio codecs
by Karl Wiberg
· 6 years ago
2e7a459
Reland "Fix race conditions in NetworkMonitor. This change makes the class thread-safe."
by Honghai Zhang
· 6 years ago
9406569
Use round robin packet queue in the pacer by default
by Ilya Nikolaevskiy
· 6 years ago
012b7e7
Add a couple of logs.
by Ying Wang
· 6 years ago
dcb4cd1
Roll chromium_revision bb5e87b52c..6562397ff8 (540725:540826)
by Autoroller
· 6 years ago
6723cdc
Revert "Separate test/fake_audio_device on API and implementation."
by Artem Titov
· 6 years ago
f9f71b9
Add hugeFramesSent GetStats metric
by Ilya Nikolaevskiy
· 6 years ago
8ea5f9a
Separate test/fake_audio_device on API and implementation.
by Artem Titov
· 6 years ago
a5797c2
Clean up Android API audio settings
by Sam Zackrisson
· 6 years ago
f9d0f1d
UMA counters for SDES x media type
by Harald Alvestrand
· 6 years ago
0133d46
Use a smaller size of sequence number set, to improve performance
by Ying Wang
· 6 years ago
8d9dcb1
Adding bps_or method to DataRate class.
by Sebastian Jansson
· 6 years ago
e90636a
Codec instantiation tests for testing device capabilities in batch.
by Rasmus Brandt
· 6 years ago
6375ef7
Enable VideoProcessorIntegrationTest on devices.
by Kári Tristan Helgason
· 6 years ago
4d36449
Add stub draft of audio generator to APM
by Sam Zackrisson
· 6 years ago
4c09d7a
In webrtc_perf_tests always create screenshare stream after realtime video
by Ilya Nikolaevskiy
· 6 years ago
3f693b9
Delete unused method SetPeriodicKeyFrames.
by Niels Möller
· 6 years ago
9e981f0
Clean up iOS API audio settings
by Sam Zackrisson
· 6 years ago
4ff5443
Fix bugs in collecting STUN candidate stats and configuring STUN
by Qingsi Wang
· 6 years ago
afe9eb3
Roll chromium_revision b36a6c8d2c..bb5e87b52c (540587:540725)
by Autoroller
· 6 years ago
845e878
Name change from stream label to stream id for spec compliance.
by Seth Hampson
· 6 years ago
1d287b5
Roll chromium_revision 6373ee9653..b36a6c8d2c (540478:540587)
by Autoroller
· 6 years ago
3ba7a57
Fixing typo in log messages.
by Taylor Brandstetter
· 6 years ago
a646d30
Enables configuration of transmission max bitrate multiplier and fec protection level.
by Ying Wang
· 6 years ago
13e7434
Merge SimulcastStream and SpatialLayer structures.
by Sergey Silkin
· 6 years ago
9039969
Call vpx_codec_destroy() only if vpx_codec_init() call preceded.
by Sergey Silkin
· 6 years ago
ede748f
Adding waits to SendSideCongestionController tests.
by Sebastian Jansson
· 6 years ago
38c15d3
Template argument and corpora for Audio Processing Fuzzer.
by Alex Loiko
· 6 years ago
3e871ea
Single exit point in VPx wrapper Release().
by Sergey Silkin
· 6 years ago
a944732
Removing Android32 MIPS from mb_config.pyl.
by Mirko Bonadei
· 6 years ago
048805e
Make it possible to clear the VideoFormat set on VideoAdapter by making VideoAdapter::OnOutputFormatRequest take an rtc::Optional
by Chris Dziemborowicz
· 6 years ago
12196b4
Update prebuilt AppRTC
by Oleh Prypin
· 6 years ago
459f4e3
VP9 GoF/temporal index check.
by philipel
· 6 years ago
36482cd
Roll chromium_revision 2eea119a99..6373ee9653 (540206:540478)
by Autoroller
· 6 years ago
fc85371
Crash if PeerConnection methods are called with the wrong SdpSemantics.
by Steve Anton
· 6 years ago
6d30631
Bug Fix: Multiplex Codec Crash
by Qiang Chen
· 6 years ago
243cd32
Updating android/README.
by Seth Hampson
· 6 years ago
17668ec
Audio codec implementations: Take optional codec ID argument
by Karl Wiberg
· 6 years ago
cb415b2
Roll chromium_revision a410274678..2eea119a99 (539759:540206)
by Oleh Prypin
· 6 years ago
5a26a3a
Remove public sync_label from StreamParams
by Steve Anton
· 6 years ago
a796a7e
Reland "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers."
by Sergey Silkin
· 6 years ago
af23b75
Update iOS AppRTC to use PeerConnection Unified Plan
by Steve Anton
· 6 years ago
45087cd
Moved retransmission rate limiter to Call class.
by Sebastian Jansson
· 6 years ago
546d7f9
Added OnAddTrack to Objective C SDK.
by Yura Yaroshevich
· 6 years ago
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