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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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e2405c1a823f3baf90a9c72f2e058f91eb659c20
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
26968ba
Delete unused utf8 conversion utilities
by Niels Möller
· 6 years ago
e8038e9
Adds IP overhead info to PacketInfo.
by Sebastian Jansson
· 6 years ago
74cd1ef
AEC3: Enabling by default the use of the stationarity properties at render at init
by Jesús de Vicente Peña
· 6 years ago
5350d1c
RtcEventLogSource no longer uses deprecated parsing functions.
by Bjorn Terelius
· 6 years ago
499bc6c
Fix race conditions for ReofferDoesNotCallOnTrack test.
by Yves Gerey
· 6 years ago
53e2211
AEC3: Kill kill-switches
by Gustaf Ullberg
· 6 years ago
8b3cc49
Adds default values for feedback/allocation indicators.
by Sebastian Jansson
· 6 years ago
fb226af
Remove some old logging in goog_cc for congestion window.
by Ying Wang
· 6 years ago
a1d9ca4
Revert "Add ability to specify if rate controller of video encoder is trusted."
by Oleh Prypin
· 6 years ago
cdc959f
Compute video freeze metrics on rendered frames instead of on decoded
by Ilya Nikolaevskiy
· 6 years ago
3bdbc84
Moves pushback controller to GoogCC
by Sebastian Jansson
· 6 years ago
f81170b
Add error logs to RtpPacketHistory::GetBestFittingPacket when no packet is found.
by Per Kjellander
· 6 years ago
ade98c9
Adds srte to WATCHLISTS.
by Sebastian Jansson
· 6 years ago
2b15626
Revert "Use unique_ptr and ArrayView in SSLFingerprint"
by Henrik Grunell
· 6 years ago
703259c
Don't CHECK when parsing AEC3 parameters from json
by Sam Zackrisson
· 6 years ago
80bf775
Roll chromium_revision 2499289737..f34485ffde (598606:598711)
by chromium-webrtc-autoroll
· 6 years ago
f7fcaf0
Use zero octets for rtp packet padding
by Danil Chapovalov
· 6 years ago
3d25530
Reland "Export symbols needed by the Chromium component build (part 1)."
by Mirko Bonadei
· 6 years ago
3e335d1
Add ability to specify if rate controller of video encoder is trusted.
by Erik Språng
· 6 years ago
88be972
Delete post_encode_callback
by Niels Möller
· 6 years ago
74f6c7e
AEC3: Cleanup test code for platforms with clock-drift
by Per Åhgren
· 6 years ago
d6b0796
AEC3: Ensure that the usage of stationary signal properties is not unset
by Per Åhgren
· 6 years ago
23b2a25
Remove unlimited retransmission for screenshare experiment code
by Ilya Nikolaevskiy
· 6 years ago
cc21e61
Use unique_ptr and ArrayView in SSLFingerprint
by Steve Anton
· 6 years ago
e8d2b1b
Roll chromium_revision 8afdf16764..2499289737 (598496:598606)
by chromium-webrtc-autoroll
· 6 years ago
f7dd9df
Change TurnPort::Create to return a unique_ptr
by Steve Anton
· 6 years ago
9cfce17
Roll chromium_revision 0d09089dd5..8afdf16764 (598349:598496)
by chromium-webrtc-autoroll
· 6 years ago
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