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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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ef45669acf331e536414d4b08cc683c7116f8f77
ef45669
Adds GetSentPacket to PacketResult.
by Sebastian Jansson
· 6 years ago
449c1c0
Adds unit tests for safe reset trial.
by Sebastian Jansson
· 6 years ago
7286496
Download aap2 and bundletool as part of required dependencies.
by Yves Gerey
· 6 years ago
6fcf6ca
Modified PressEnterToContinue() to actualy check if Enter is pressed
by Danail Kirov
· 6 years ago
2c16cc6
Replace some usage of EventWrapper with rtc::Event.
by Niels Möller
· 6 years ago
88d8d7d
Add missing assignment in RTCConfiguration.mm
by Piotr (Peter) Slatala
· 6 years ago
f3ff14c
Properly setup MockPeerConnectionObserver in tests.
by Yves Gerey
· 6 years ago
22a8f98
Formatted sslidenty.cc and moved non referenced functions into an
by Benjamin Wright
· 6 years ago
428320c
Formatting OpenSSLCertificate and doing some minor code cleanup.
by Benjamin Wright
· 6 years ago
5d35554
Rename private member functions to use CamelCase.
by Benjamin Wright
· 6 years ago
61c5cc8
Makes OpenSSL concrete implementations final.
by Benjamin Wright
· 6 years ago
2616594
Refactor: Make SSLCertChain a final class.
by Benjamin Wright
· 6 years ago
150a907
FrameEncryption Video End To End Testcase.
by Benjamin Wright
· 6 years ago
c462a6e
Prevent the frame decryptor being set if the channel is stopped.
by Benjamin Wright
· 6 years ago
625771d
Roll chromium_revision a539a24569..62e33bd2f0 (603045:603177)
by chromium-webrtc-autoroll
· 6 years ago
59ebf23
Refactor structs in rtc_event_log_parser_new.h
by Elad Alon
· 6 years ago
ff43541
Delta compression efficiency improvement for non-existent base
by Elad Alon
· 6 years ago
436ebca
Fix extra setdscp call introduced by bad merge.
by Tim Haloun
· 6 years ago
0f08d22
Add a function for enabling the congestion window and pushback controller in the webrtc::SendSideCongestionController.
by erikvarga@webrtc.org
· 6 years ago
99b71df
Use function_video_(en|de)coder_factory from api
by Danil Chapovalov
· 6 years ago
88c2c50
Use monotonic clock to derive NTP timestamps in RTCP module
by Ilya Nikolaevskiy
· 6 years ago
fdee701
Add parser and unittests for new RTC event log format.
by Bjorn Terelius
· 6 years ago
916ae08
Makes critsect_.Leave() more visible in PacedSender.
by Sebastian Jansson
· 6 years ago
6dd7f91
Remove deprecated deregistration functions in RtcpTransceiver
by Danil Chapovalov
· 6 years ago
06aa209
Add support to adapt video without preserving aspect ratio
by Magnus Jedvert
· 6 years ago
9049037
Simplify api/DEPS presubmit check.
by Mirko Bonadei
· 6 years ago
7d76a31
Use MediaTransportInterface, for audio streams.
by Niels Möller
· 6 years ago
2769cd5
Roll chromium_revision f54583b6a0..a539a24569 (602763:603045)
by chromium-webrtc-autoroll
· 6 years ago
0d24772
Allocate CMBlockBuffers using a memory pool.
by Kári Tristan Helgason
· 6 years ago
c35096d
Reland "Encode RTC event logs in new format."
by Bjorn Terelius
· 6 years ago
a5d543c
Set minSdkVersion to 16 for AppRTCMobile_stubbed_video_io_test_apk.
by Mirko Bonadei
· 6 years ago
95dfa52
Clarify the desired semantics of AsyncResolverInterface::GetResolvedAddress.
by Zach Stein
· 6 years ago
327b753
Split out a separate target for VP8EncoderSimulcastProxy
by Jonathan Yu
· 6 years ago
5abd541
Stop exporting simulcast_encoder_adapter.h in :rtc_internal_video_codecs
by Jonathan Yu
· 6 years ago
b19b497
Refactor: Removing IgnoreBadCert from SSLStreamAdapter. Make test methods more explicit.
by Benjamin Wright
· 6 years ago
dcd40ca
Roll chromium_revision d68fb50e14..f54583b6a0 (602627:602763)
by chromium-webrtc-autoroll
· 6 years ago
8c27cca
Promotoing webrtc::CryptoOptions to RTCConfiguration.
by Benjamin Wright
· 6 years ago
78410ad
Fixes use after free error when setting a new FrameEncryptor on ChannelSend.
by Benjamin Wright
· 6 years ago
f26e290
fuchsia: Stub out timing and memory functions
by Scott Graham
· 6 years ago
9c8ae4b
Disable probe delay warning in release builds.
by Jamie Walch
· 6 years ago
6c6c9df
Refactor: Renaming ssl_cert_chain to GetSSLCertificateChain()
by Benjamin Wright
· 6 years ago
359d60a
Adds target rate to audio send stream stats.
by Sebastian Jansson
· 6 years ago
57ba7e1
Normalize baseline in network delay plot to RTT/2.
by Bjorn Terelius
· 6 years ago
039743e
Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
by Niels Möller
· 6 years ago
e2754c9
Fixes bug in AudioPriorityBitrateAllocationStrategy field trial.
by Sebastian Jansson
· 6 years ago
c0e4d45
Adds BitrateAllocation struct to OnBitrateUpdated.
by Sebastian Jansson
· 6 years ago
4ba6c26
Delete MessageData when a message is posted to a quitting MessageQueue
by Niels Möller
· 6 years ago
9516c38
[Fuzzer] Check FieldTrial bitmask size at compile time.
by Yves Gerey
· 6 years ago
1803bb2
Fix for clock read race in FakeNetworkPipe.
by Christoffer Rodbro
· 6 years ago
3284b61
Fix for packet loss tracking in network emulation.
by Christoffer Rodbro
· 6 years ago
2620470
Update fuzzer max input length handling
by Sam Zackrisson
· 6 years ago
ddc84e9
Publish function_video_(en|de)coder_factory into api
by Danil Chapovalov
· 6 years ago
23524ce
Add HDR metadata struct
by Johannes Kron
· 6 years ago
977b46a
Export symbols needed by the Chromium component build (part 7).
by Mirko Bonadei
· 6 years ago
3eb1c72
Removes deprecated BitrateAllocation alias.
by Sebastian Jansson
· 6 years ago
2506839
Add DCHECK for wrap around in RtpVideoSender::OnBitrateUpdated.
by Bjorn Terelius
· 6 years ago
370bae4
APM: Adding more explicit handling of failures in the json config data
by Per Åhgren
· 6 years ago
487e694
Use default value if field trial switch is set to an invalid number
by Johannes Kron
· 6 years ago
273c851
Remove obsolete android ndk copy from //third_party/android_tools/ndk
by Yongje Lee
· 6 years ago
7a95e0f
APM: Add ability to turn on/off dumping of internal data
by Per Åhgren
· 6 years ago
e2fd86a
Move encoder metadata into EncoderInfo struct.
by Erik Språng
· 6 years ago
2d3a1fb
Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit
by Oleh Prypin
· 6 years ago
4191a81
Revert "Move relay server code to a test-only target p2p_server_utils."
by Oleh Prypin
· 6 years ago
8fb20ce
Roll chromium_revision f57bd4785e..d68fb50e14 (602511:602627)
by chromium-webrtc-autoroll
· 6 years ago
e284c52
Move relay server code to a test-only target p2p_server_utils.
by Niels Möller
· 6 years ago
01c68b8
Roll chromium_revision b9a687f112..f57bd4785e (602396:602511)
by chromium-webrtc-autoroll
· 6 years ago
09a49fa
Roll chromium_revision 869181c2dc..b9a687f112 (602275:602396)
by chromium-webrtc-autoroll
· 6 years ago
4cb4786
Add expected default values to video configuration tests.
by Benjamin Wright
· 6 years ago
44a262a
Declares BitrateAllocator methods const.
by Sebastian Jansson
· 6 years ago
583d6d9
Add missing directory to api/DEPS and PRESUBMIT.py.
by Mirko Bonadei
· 6 years ago
62ae178
Remove deprecated pipe field from VideoQualityTestFixtureInterface::Params
by Artem Titov
· 6 years ago
825f83b
Revert "Encode RTC event logs in new format."
by Mirko Bonadei
· 6 years ago
e943d43
Remove deprecated DefaultNetworkSimulationConfig
by Artem Titov
· 6 years ago
a418e67
Use checkdeps to ensure API headers don't include internal headers.
by Mirko Bonadei
· 6 years ago
ec9b77b
Remove deprecated API: NetwrokSimulationInterface.
by Artem Titov
· 6 years ago
257ed43
Add support for optional fields in FixedLengthDeltaEncoder
by Elad Alon
· 6 years ago
c6ec4b1
Fix w3c URL for RTCIceTransport
by Niels Möller
· 6 years ago
5e58bcb
Forward audio rtp frequency to Rtcp sender and use it for SR packets
by Ilya Nikolaevskiy
· 6 years ago
ece3c22
Encode RTC event logs in new format.
by Bjorn Terelius
· 6 years ago
fb5c1ec
AEC3: Included missing parsing of config parameter
by Per Åhgren
· 6 years ago
8e6749e
Improve fileutils_override implementation internal API.
by Artem Titov
· 6 years ago
e068ad6
Use a sufficiently large bitmask.
by Jonas Olsson
· 6 years ago
511fe0b
Roll chromium_revision 5e5003737d..869181c2dc (602066:602275)
by chromium-webrtc-autoroll
· 6 years ago
d38a2b8
Increase the UDP receive buffer for video
by Johannes Kron
· 6 years ago
f0c449e
APM: Correct includes required for the data dumping functionality
by Per Åhgren
· 6 years ago
700b4a4
AEC3: Allow limiting dominant nearend to the non-initial phase
by Per Åhgren
· 6 years ago
41ed3e0
Roll chromium_revision b1cb85713b..5e5003737d (601125:602066)
by chromium-webrtc-autoroll
· 6 years ago
73f3917
Add support for signed deltas in FixedLengthDeltaEncoder
by Elad Alon
· 6 years ago
4b31cf5
Disable CertificateTest.CertificateIsUsedInConfig
by Elad Alon
· 6 years ago
087e9be
AGC2 Limiter class renamed.
by Alessio Bazzica
· 6 years ago
4842c78
Increasing APM fuzzer coverage.
by Alex Loiko
· 6 years ago
c6de47e
Added supported H264 profiles for new iPhones
by Yura Yaroshevich
· 6 years ago
8f726be
Add ability to override detection of resource location and source root
by Artem Titov
· 6 years ago
877dc89
Fix errors in AEC3 JSON parsing
by Sam Zackrisson
· 6 years ago
6e8e299
Revert "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
by Oleh Prypin
· 6 years ago
7e6b528
Removes FakeBaseEngine.
by Sebastian Jansson
· 6 years ago
362cb50
Remove redundant RTC_DCHECK of max/min RTP header extension id
by Johannes Kron
· 6 years ago
93922dc
Fix flaky unit test in rtc_unittests
by Johannes Kron
· 6 years ago
848273a
Revert "Increase coverage of AEC3 JSON config unit tests, fix bugs"
by Sam Zackrisson
· 6 years ago
80cd25b
Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
by Niels Möller
· 6 years ago
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