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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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80cd25bcfb2264fa0f1192de942a6f063879dd42
80cd25b
Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
by Niels Möller
· 6 years ago
988cc08
[Cleanup] Add missing #include. Remove useless ones.
by Yves Gerey
· 6 years ago
8ee06a7
Increase coverage of AEC3 JSON config unit tests, fix bugs
by Sam Zackrisson
· 6 years ago
f7a7c8a
Stop adding RTT delay if there was not packet loss for enough time
by Gustavo Garcia
· 7 years ago
0627e21
Removes unused DeliverPacket from CallClient.
by Sebastian Jansson
· 6 years ago
9581bc4
Rename too long variable name to extmap_allow_mixed
by Johannes Kron
· 6 years ago
2edab4c
Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase.
by Niels Möller
· 6 years ago
01cf44d
AEC3: Adding missing elements to the json parser
by Per Åhgren
· 6 years ago
3583693
3 TLs: add full stack test for short pattern + base heavy alloc.
by Rasmus Brandt
· 6 years ago
6ed37ba
AEC3: Enable fuzzer testing of old render buffering code.
by Gustaf Ullberg
· 6 years ago
d34597c
Update test::CreateVideoStreams to use num_temporal_layers.
by Åsa Persson
· 6 years ago
98f5f6c
In RtcpTransceiver functions with callback avoid relying on PostTaskAndReply
by Danil Chapovalov
· 6 years ago
b0ab2ce
Reland "Remove the HighPassFilter interface"
by Sam Zackrisson
· 6 years ago
c9f9b87
AEC3: Improve dominant nearend detection
by Gustaf Ullberg
· 6 years ago
ac19414
Export symbols needed by the Chromium component build (part 6).
by Mirko Bonadei
· 6 years ago
9a5da49
Split out a separate target for SimulcastEncoderAdapter
by Jonathan Yu
· 6 years ago
165148d
Reland "Remove deprecated barcode scanning functionality"
by Magnus Jedvert
· 6 years ago
39feabe
Enables FrameDecryptor to do an initial key request on frame decryption.
by Benjamin Wright
· 6 years ago
201596f
Make packet max buffer size configurable via field trial flag
by Johannes Kron
· 6 years ago
68b2df7
Make protection_overhead_rate configurable through field trial.
by Ying Wang
· 6 years ago
6d21650
Don't decode frames with an older timestamp than the last decoded timestamp.
by philipel
· 6 years ago
38a3419
Revert "Remove deprecated barcode scanning functionality"
by Alessio Bazzica
· 6 years ago
67b011d
Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream
by Niels Möller
· 6 years ago
ff292f3
Remove deprecated barcode scanning functionality
by Magnus Jedvert
· 6 years ago
635474e
Compute RTCConnectionState and RTCIceConnectionState.
by Jonas Olsson
· 6 years ago
800e121
Adds support to change transport routes in Scenario tests.
by Sebastian Jansson
· 6 years ago
8d33c0c
Adds field trial to do safer reset on route change.
by Sebastian Jansson
· 6 years ago
c98849c
AEC3: changes the signal used for deciding when to update the erle so the reverb render signal is now used
by Jesús de Vicente Peña
· 6 years ago
ecdd432
Routing unacknowledged data in TransportFeedbackAdapter.
by Sebastian Jansson
· 6 years ago
e482ff8
Audio codecs API: Remove some weasel words in the docs
by Karl Wiberg
· 6 years ago
57dd881
Delete dead code in webrtc_libyuv.cc
by Niels Möller
· 6 years ago
9acf1c1
Reland "Make sure Chromium will pick the correct field_trial/metric impl."
by Mirko Bonadei
· 6 years ago
648d28a
Media engine and channel support for per-channel dscp values, specified by RtpParameter
by Tim Haloun
· 6 years ago
51cc30c
Fix a null reference bug in NetworkMonitorAutoDetect.getNetworkState.
by Qingsi Wang
· 6 years ago
cb21ffe
Add blob-encoding support for RTC event logs
by Elad Alon
· 6 years ago
2dfa998
Reland "Prefix flag macros with WEBRTC_."
by Mirko Bonadei
· 6 years ago
c538fc7
Revert "Prefix flag macros with WEBRTC_."
by Mirko Bonadei
· 6 years ago
1cb20de
Revert "Make sure Chromium will pick the correct field_trial/metric impl."
by Niklas Enbom
· 6 years ago
3c7d599
Replace _stricmp with absl::EqualsIgnoreCase
by Niels Möller
· 6 years ago
53347b7
Mute failed tests when no sanitizer defects.
by Yves Gerey
· 6 years ago
2baa3c4
Roll chromium_revision c66210d3ab..b1cb85713b (601019:601125)
by chromium-webrtc-autoroll
· 6 years ago
0d26c99
Set renderThreadHandler to null on uncaught exception in EglRenderer.
by Sami Kalliomäki
· 6 years ago
5ccdc13
Prefix flag macros with WEBRTC_.
by Mirko Bonadei
· 6 years ago
8dc280d
Make sure Chromium will pick the correct field_trial/metric impl.
by Mirko Bonadei
· 6 years ago
1ddc5b6
Export symbols needed by the Chromium component build (part 5).
by Mirko Bonadei
· 6 years ago
cf58bf7
Move the SocketStream class to test target
by Niels Möller
· 6 years ago
bc6a06c
Adding missing #include on absl/memory/memory.h.
by Mirko Bonadei
· 6 years ago
82d4329
Delete unused test class StreamSource
by Niels Möller
· 6 years ago
2461c31
Roll chromium_revision 343f58e4df..c66210d3ab (600903:601019)
by chromium-webrtc-autoroll
· 6 years ago
97fc11f
Fix the 'SetConfiguration(RTCConfiguration::use_media_transport)' setting.
by Piotr (Peter) Slatala
· 6 years ago
28c437c
Roll chromium_revision 834490b775..343f58e4df (600802:600903)
by chromium-webrtc-autoroll
· 6 years ago
aad5d36
Roll chromium_revision fc405b495a..834490b775 (600654:600802)
by chromium-webrtc-autoroll
· 6 years ago
cb06cac
Moves fake media engine implementation to cc file.
by Sebastian Jansson
· 6 years ago
7dc9774
Delete unused code from media/base/testutils.{cc,h}
by Niels Möller
· 6 years ago
192eeec
Enable End-to-End Encrypted Video Frames.
by Benjamin Wright
· 6 years ago
6714bf9
Fix up OpenSSL/BoringSSL forward declarations.
by David Benjamin
· 6 years ago
50b1e6b
Add fixed-size delta-encoding/decoding code for WebRTC event logs
by Elad Alon
· 6 years ago
608298b
Move RtcEventLog::CreateNull implementation near declaration.
by Danil Chapovalov
· 6 years ago
78416b6
Adds time to initial config in analyzer code.
by Sebastian Jansson
· 6 years ago
f203d73
Correctly slice MediaBitrateRecieved on content type in ReceiveStatisticsProxy
by Ilya Nikolaevskiy
· 6 years ago
d28efe5
Adds field trial to AudioPriorityBitrateAllocationStrategy.
by Sebastian Jansson
· 6 years ago
65faede
AEC3: Introduce partial adaptive filter resets at echo path changes
by Per Åhgren
· 6 years ago
1ffee36
AEC3: Remove ERLE uncertainty code that has no effect
by Per Åhgren
· 6 years ago
4b7a412
Relieve perkj@ of some OWNER duties
by Karl Wiberg
· 6 years ago
6347bda
Remove expat from generate_licenses.py.
by Mirko Bonadei
· 6 years ago
d0be002
Add missing #include to absl/memory/memory.h
by tzik
· 6 years ago
d65d179
Export symbols needed by the Chromium component build (part 4).
by Mirko Bonadei
· 6 years ago
9d24795
rtc::ZeroOnFreeBuffer: Don't forget to zero memory we free in operator=
by Karl Wiberg
· 6 years ago
b5541a0
Fix: Argv may be corrupted after InitGoogleMock found any related flags
by Artem Titov
· 6 years ago
576a333
Roll chromium_revision c926d3bb2f..fc405b495a (600547:600654)
by chromium-webrtc-autoroll
· 6 years ago
f05cae3
Roll chromium_revision 8bef2e268b..c926d3bb2f (600433:600547)
by chromium-webrtc-autoroll
· 6 years ago
7fa6ee6
Adds support for "-" to a=ssrc msid lines.
by Seth Hampson
· 6 years ago
98a462c
Reland "Reland "Propagate media transport to media channel.""
by Anton Sukhanov
· 6 years ago
55fab32
Roll chromium_revision c5242283d9..8bef2e268b (600305:600433)
by chromium-webrtc-autoroll
· 6 years ago
bfb444c
Adds new CryptoOption crypto_options.frame.require_frame_encryption.
by Benjamin Wright
· 6 years ago
d932fba
Track padding and header size in log event.
by Bjorn Terelius
· 6 years ago
b9972fa
Adds AudioNetworkAdaptation support to Scenario tests.
by Sebastian Jansson
· 6 years ago
09beff2
Add UseMediaTransport RTCConfiguration support in Java class
by Piotr (Peter) Slatala
· 6 years ago
2bff543
Removes undefined declarations in channel.h.
by Sebastian Jansson
· 6 years ago
4f3ce27
rtc::Buffer: Handle move self-assignment
by Karl Wiberg
· 6 years ago
d189252
Delete more rtc_base/stringutils.*
by Niels Möller
· 6 years ago
fab9129
Get frame type, width and height from the generic descriptor.
by philipel
· 6 years ago
34d990f
Adding NetEq buffer full metric to UMA.
by Minyue Li
· 6 years ago
a240daa
Change verification of stream configs in RTC event log unittest.
by Bjorn Terelius
· 6 years ago
5a464d3
Add resolution to generic frame descriptor extension
by Danil Chapovalov
· 6 years ago
4744e5b
Reland "Remove old video_bitrate_allocator.h"
by Rasmus Brandt
· 6 years ago
dbb47b8
Roll chromium_revision d06a979d44..c5242283d9 (600199:600305)
by chromium-webrtc-autoroll
· 6 years ago
f25303e
Reland: Modernize rtc::SSLCertificate
by Steve Anton
· 6 years ago
28b6d1d
Roll chromium_revision 2419220cab..d06a979d44 (600044:600199)
by chromium-webrtc-autoroll
· 6 years ago
9accc9f
Revert "Reland "Propagate media transport to media channel.""
by Oleh Prypin
· 6 years ago
c76b8ff
Roll chromium_revision f8cad916e6..2419220cab (599923:600044)
by chromium-webrtc-autoroll
· 6 years ago
aa1e7c2
Allow 'use_media_transport' to be modified on PeerConnection before local/remote description are set.
by Piotr (Peter) Slatala
· 6 years ago
da65ed2
Reland "Propagate media transport to media channel."
by Anton Sukhanov
· 6 years ago
4905edb
Reland: Use unique_ptr and ArrayView in SSLFingerprint
by Steve Anton
· 6 years ago
243cabe
Formatting openssladapter to be more consistent.
by Benjamin Wright
· 6 years ago
4e5074e
Add MediaTransportInterface factory to the Jni bindings
by Piotr (Peter) Slatala
· 6 years ago
9b1d679
Remove 'iOS32 Sim Debug (iOS 9.0)' from client.webrtc.
by Mirko Bonadei
· 6 years ago
d895f42
Revert "Remove the HighPassFilter interface"
by Niklas Enbom
· 6 years ago
c1bfe1a
Avoids creating empty call_order file when no call order data is written
by Per Åhgren
· 6 years ago
6026f05
Calculate max payload size for an rtp packet to fit full video frame
by Danil Chapovalov
· 6 years ago
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