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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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f25303efd173c440f8cb3f09b455fdbd211f8896
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
f5e767d
Don't send max allocation probe unless allocation changed.
by Sebastian Jansson
· 6 years ago
a1c9312
Update proto for new event log format.
by Bjorn Terelius
· 6 years ago
aba0633
Delete wrappers for snprintf and vsnprintf
by Niels Möller
· 6 years ago
3100fc1
Use color aligning in video quality analysis tool
by Magnus Jedvert
· 6 years ago
3e7b7b1
AEC3: Changes to initial behavior and handling of saturated echo
by Per Åhgren
· 6 years ago
276827c
Export symbols needed by the Chromium component build (part 3).
by Mirko Bonadei
· 6 years ago
0753675
Using more specific dependencies in rtc_base.
by Sebastian Jansson
· 6 years ago
6c78ff4
Always verify packet wasn't resend recently before resending it.
by Danil Chapovalov
· 6 years ago
2d0c687
Remove |hw_encoder| and |hw_decoder| from VideoCodecTestFixture::Config.
by Rasmus Brandt
· 6 years ago
f907c49
Delete unused code in rtc_base/stringutils.*
by Niels Möller
· 6 years ago
84d2827
Add generate_ios_coverage_command.py script
by Artem Titarenko
· 6 years ago
1298541
Removing unnecessary dependencies on socket.h.
by Sebastian Jansson
· 6 years ago
03d2801
Roll chromium_revision 0cecb6ce10..f8cad916e6 (599821:599923)
by chromium-webrtc-autoroll
· 6 years ago
be65d48
Remove AECM comfort noise setting from API
by Sam Zackrisson
· 6 years ago
e2405c1
Remove the HighPassFilter interface
by Sam Zackrisson
· 6 years ago
d419db9
Adding support for logging severity LS_NONE.
by Peter Hanspers
· 6 years ago
2e47f7c
Implement test class LoopbackMediaTransport
by Niels Möller
· 6 years ago
f06bacc
Add test that verifies that VideoEncoderConfig max_framerate is reported to source.
by Åsa Persson
· 6 years ago
2560e2e
Removes Clock instance from RoundRobinPacketQueue.
by Sebastian Jansson
· 6 years ago
1927dfa
Add tool for aligning color space of video files
by Magnus Jedvert
· 6 years ago
f0e926f
Add missing #include and deps to absl/memory
by tzik
· 6 years ago
1b26a0a
Roll chromium_revision 0e821c2fa2..0cecb6ce10 (599702:599821)
by chromium-webrtc-autoroll
· 6 years ago
a39a007
Reland "Deprecates legacy transport feedback adapter."
by Sebastian Jansson
· 6 years ago
acaed83
Roll chromium_revision 0df2607f98..0e821c2fa2 (599562:599702)
by chromium-webrtc-autoroll
· 6 years ago
c9e6b96
Add necessary frameworks to sdk objc audio targets.
by Jiawei Ou
· 6 years ago
3b56ee7
Export symbols needed by the Chromium component build (part 2).
by Mirko Bonadei
· 6 years ago
d4d5f8a
Formatting and style guide improvements for opensslstreamadapter.cc
by Benjamin Wright
· 6 years ago
f714ee1
Revert "Deprecates legacy transport feedback adapter."
by Mirko Bonadei
· 6 years ago
a5778e0
Deprecates legacy transport feedback adapter.
by Sebastian Jansson
· 6 years ago
5c94f55
Removes analyzer dependency on legacy congestion controller.
by Sebastian Jansson
· 6 years ago
82c71af
Revert "Modernize rtc::SSLCertificate"
by Niklas Enbom
· 6 years ago
1e3ed16
Fix force_fieldtrials documentation in video_loopback
by Elad Alon
· 6 years ago
0391446
Removing forward declarations in paced_sender.h.
by Sebastian Jansson
· 6 years ago
cd0ca2d
Adds unit test for RTT based backoff.
by Sebastian Jansson
· 6 years ago
74c066c
Merges ControlHandler and PacerController.
by Sebastian Jansson
· 6 years ago
7341ab6
Moves functionality to TransportFeedbackAdapter.
by Sebastian Jansson
· 6 years ago
ed04912
Stop simulations when a LOG_END event is reached.
by Ivo Creusen
· 6 years ago
961dbea
NetEq fuzzer: Restrict fuzzer input to 90000 bytes
by Henrik Lundin
· 6 years ago
d8a52b3
Make ivoc owner of audio_coding.
by Ivo Creusen
· 6 years ago
6932fb2
Revert "Reland: Use unique_ptr and ArrayView in SSLFingerprint"
by Mirko Bonadei
· 6 years ago
40a7a35
Extract functionality of test_main into separate library.
by Artem Titov
· 6 years ago
d2d2ecb
Add command-line flag for setting the max number of packets in the buffer.
by Ivo Creusen
· 6 years ago
c84cd95
Move MockVideoDecoder to api/test.
by Erik Språng
· 6 years ago
11539f0
AEC3: Simplify render buffering
by Gustaf Ullberg
· 6 years ago
e07864e
Moves rtc::SentPacket to separate target.
by Sebastian Jansson
· 6 years ago
76ad154
New method for precise packet reception time measurement.
by Christoffer Rodbro
· 6 years ago
2c7149b
Add field trial to disable unsignalled video.
by Åsa Persson
· 6 years ago
6003e7a
Fix FakeEncoder to produce correct bitrate for several temporal layers
by Ilya Nikolaevskiy
· 6 years ago
a85995a
Set frame duration per spatial layer.
by Sergey Silkin
· 6 years ago
9ac3c91
Refactor of extmap-allow-mixed in SessionDescription
by Johannes Kron
· 6 years ago
cae8802
Delete force_mic_volume_max.
by Patrik Höglund
· 6 years ago
83bd37c
Add field trials for configuring Opus encoder packet loss rate.
by Jakob Ivarsson
· 6 years ago
fcebe0e
in RtpPacketizers separate case 'frame fits into single packet'.
by Danil Chapovalov
· 6 years ago
1a35fbd
Add field trial for normalized simulcast size.
by Åsa Persson
· 6 years ago
09256c1
Remove ios32_sim_ios9_dbg from CQ.
by Mirko Bonadei
· 6 years ago
147038c
cq: explicitly mark presubmit tryjob as not re-usable in CQ.
by Oleh Prypin
· 6 years ago
9c18d21
Remove rtc_base/Dummy.java.
by Mirko Bonadei
· 6 years ago
28887a5
Roll chromium_revision 03013c95df..0df2607f98 (599460:599562)
by chromium-webrtc-autoroll
· 6 years ago
37cf245
Revert "Propagate media transport to media channel."
by Oleh Prypin
· 6 years ago
f409246
Roll chromium_revision 3b54b6aa8b..03013c95df (599343:599460)
by chromium-webrtc-autoroll
· 6 years ago
8c16f74
Propagate media transport to media channel.
by Anton Sukhanov
· 6 years ago
dbc2ea7
Roll chromium_revision c12ec9eedc..3b54b6aa8b (599188:599343)
by chromium-webrtc-autoroll
· 6 years ago
55cd3ac
Modernize rtc::SSLCertificate
by Steve Anton
· 6 years ago
47f3240
Reland: Use unique_ptr and ArrayView in SSLFingerprint
by Steve Anton
· 6 years ago
5e23a41
Removes backwards compatability CryptoOptions support.
by Benjamin Wright
· 6 years ago
23e48fb
Move expectations from eventlog unittests to helper functions.
by Bjorn Terelius
· 6 years ago
f7fee39
Remove rtc_base:rtc_base_generic.
by Mirko Bonadei
· 6 years ago
b354f74
Roll chromium_revision d47784f23e..c12ec9eedc (599082:599188)
by chromium-webrtc-autoroll
· 6 years ago
6af1c92
Add mock_video_encoder.h to api/test
by Erik Språng
· 6 years ago
3b4b4f5
Mitigate miscalculation of rtp packet size
by Danil Chapovalov
· 6 years ago
781b2bd
Restore "device type" for iOS internal.client.webrtc
by Artem Titarenko
· 6 years ago
62b1345
Get rid of thread_darwin file.
by Kári Tristan Helgason
· 6 years ago
c34cf71
Revert "Remove old video_bitrate_allocator.h"
by Oleh Prypin
· 6 years ago
93428bf
Move SdpType from/to string definition close to declaration.
by Mirko Bonadei
· 6 years ago
55d1af1
Remove support for microsecond resolution in RtcEventLogs.
by Bjorn Terelius
· 6 years ago
4529fbc
Move TemporalLayers to api/video_codecs.
by Erik Språng
· 6 years ago
28d200c
Roll chromium_revision 37b6d53f02..d47784f23e (598967:599082)
by chromium-webrtc-autoroll
· 6 years ago
a54daf1
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h"
by Benjamin Wright
· 6 years ago
edd204e
Roll chromium_revision 9d052f4b6f..37b6d53f02 (598839:598967)
by chromium-webrtc-autoroll
· 6 years ago
8f4bc41
Revert "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h"
by Oleh Prypin
· 6 years ago
1cd39fa
make CreateOffer/CreateAnswer use ice credentials of pooled sessions.
by Jonas Oreland
· 6 years ago
df1bf00
Headers shouldn't include themselves.
by Yves Gerey
· 6 years ago
ac2f3d1
Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h
by Benjamin Wright
· 6 years ago
8285841
Adds handling of untracked data to congestion controller.
by Sebastian Jansson
· 6 years ago
ca51189
Roll chromium_revision f34485ffde..9d052f4b6f (598711:598839)
by chromium-webrtc-autoroll
· 6 years ago
0d399a8
Removes socket addresses from PacketInfo struct.
by Sebastian Jansson
· 6 years ago
20ad254
Adds tracking of allocated but unacknowledged bitrate.
by Sebastian Jansson
· 6 years ago
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