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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
6117068af4a18e16755ea4908d86f652e293b57e
/
media
/
base
493a650
Propagate base minimum delay from video jitter buffer to webrtc/api.
by Ruslan Burakov
· 6 years ago
22f9925
webrtc: Remove semicolons.
by Nico Weber
· 6 years ago
ea7ef2a
Refactoring RtpSenderInternal to share implementation for Audio & Video.
by Amit Hilbuch
· 6 years ago
2297d33
Rejected simulcast layers will no longer appear in GetParameters().
by Amit Hilbuch
· 6 years ago
71aee3a
Reland "Propagate VideoFrame::UpdateRect to encoder"
by Ilya Nikolaevskiy
· 6 years ago
7ea4605
Add latency to remote source api.
by Ruslan Burakov
· 6 years ago
c664314
Clean up implementation in stream_params
by Steve Anton
· 6 years ago
429b67d
Revert "Propagate VideoFrame::UpdateRect to encoder"
by Mirko Bonadei
· 6 years ago
efa72a1
Propagate VideoFrame::UpdateRect to encoder
by Ilya Nikolaevskiy
· 6 years ago
fd5d473
Revert "Partial frame capture API part 6"
by Ilya Nikolaevskiy
· 6 years ago
9f3aabb
Delete obsolete class cricket::VideoCapturer
by Niels Möller
· 6 years ago
ef288dd
Reland: Remove dead code from stream_params.h
by Steve Anton
· 6 years ago
cf416e4
Revert "Remove dead code from stream_params.h"
by Oleh Prypin
· 6 years ago
38c83b9
Remove unused file.
by Fredrik Solenberg
· 6 years ago
3f408d0
Remove dead code from stream_params.h
by Steve Anton
· 6 years ago
7752ad6
Partial frame capture API part 6
by Ilya Nikolaevskiy
· 6 years ago
167316b
Remove proxy layer from AndroidVideoTrackSource
by Magnus Jedvert
· 6 years ago
0237106
Expose video freeze metrics in GetStats.
by Sergey Silkin
· 6 years ago
c84f661
Stop using Googletest legacy APIs.
by Mirko Bonadei
· 6 years ago
042bb00
Fix RTP transport accepting invalid RTCP headers.
by Piotr (Peter) Slatala
· 6 years ago
c1a0bcb
Implement the encoding RtpParameter scaleResolutionDownBy
by Florent Castelli
· 6 years ago
2c9ebef
Use Abseil container algorithms in media/
by Steve Anton
· 6 years ago
bcd39d4
Creating Simulcast offer and answer in Peer Connection.
by Amit Hilbuch
· 6 years ago
f380284
(7) Rename files to snake_case: remove forwarding headers
by Steve Anton
· 6 years ago
d970807
Remove rtc_base/scoped_ref_ptr.h.
by Mirko Bonadei
· 6 years ago
1e27fec
Negate flag name for prerender smoothing and update comments.
by Rasmus Brandt
· 6 years ago
805a27e
Reland "Refactor WebRtcVideoEngine tests to not use cricket::VideoCapturer, part 2."
by Niels Möller
· 6 years ago
0acffb5
Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`.
by Chen Xing
· 6 years ago
aec15aa
(5) Rename files to snake_case: install forwarding headers
by Steve Anton
· 6 years ago
10542f2
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
by Steve Anton
· 6 years ago
1c05765
(3) Rename files to snake_case: move the files
by Steve Anton
· 6 years ago
1ebfb6a
Introduce VideoFrame::id to keep track of frames inside application.
by Artem Titov
· 6 years ago
fd87da7
Delete WebRtcVideoCapturer and related classes.
by Niels Möller
· 6 years ago
3793bb4
Refactor TestVideoCapturer to support multiple sinks.
by Niels Möller
· 6 years ago
41f3a43
Remove CodecInst pt.3
by Fredrik Solenberg
· 6 years ago
cca13f6
Remove unused cryptoparams.h header
by Steve Anton
· 6 years ago
e1301a8
Revert "Implement read-only codecPayloadType in RtpParameters"
by Henrik Grunell
· 6 years ago
806e06d
Implement read-only codecPayloadType in RtpParameters
by Florent Castelli
· 6 years ago
c57d573
RID parsing for Simulcast support.
by Amit Hilbuch
· 6 years ago
514f084
New statistic added to VideoReceiveStream to determine latency to first decode.
by Benjamin Wright
· 6 years ago
3e70781
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
by Yves Gerey
· 6 years ago
352ce5c
Expose delayed packet outage as a cumulative metric of samples in the new getStats API.
by Jakob Ivarsson
· 6 years ago
8af8896
Expose jitter buffer flushes metric in new getStats api.
by Ruslan Burakov
· 6 years ago
5f2ffee
Clean up deprecated APM stats
by Sam Zackrisson
· 6 years ago
2222a80
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
by Niels Möller
· 6 years ago
38332cd
Add RTCP and simulcast support for RTCRtpReceiver::getParameters()
by Florent Castelli
· 6 years ago
6eb8a16
Exposing audio and video engines directly.
by Sebastian Jansson
· 6 years ago
fa0aa39
Removes templating from CompositeMediaEngine.
by Sebastian Jansson
· 6 years ago
84848f2
Adds interfaces for audio and video engines.
by Sebastian Jansson
· 6 years ago
dd9390c
Prevent channels being set on stopped transceiver.
by Amit Hilbuch
· 6 years ago
5571812
Adding rtcp report interval into RTCConfiguration.
by Jiawei Ou
· 6 years ago
175aa2e
Implement data channels over media transport.
by Bjorn Mellem
· 6 years ago
e693381
Delete struct rtc::PacketTime.
by Niels Möller
· 6 years ago
15ca5a9
Add implicit conversion between rtc:PacketTime and int64_t.
by Niels Möller
· 6 years ago
9190b82
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
by Johannes Kron
· 6 years ago
06aa209
Add support to adapt video without preserving aspect ratio
by Magnus Jedvert
· 6 years ago
039743e
Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
by Niels Möller
· 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
93922dc
Fix flaky unit test in rtc_unittests
by Johannes Kron
· 6 years ago
80cd25b
Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
by Niels Möller
· 6 years ago
648d28a
Media engine and channel support for per-channel dscp values, specified by RtpParameter
by Tim Haloun
· 6 years ago
3c7d599
Replace _stricmp with absl::EqualsIgnoreCase
by Niels Möller
· 6 years ago
1ddc5b6
Export symbols needed by the Chromium component build (part 5).
by Mirko Bonadei
· 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
d65d179
Export symbols needed by the Chromium component build (part 4).
by Mirko Bonadei
· 6 years ago
98a462c
Reland "Reland "Propagate media transport to media channel.""
by Anton Sukhanov
· 6 years ago
bfb444c
Adds new CryptoOption crypto_options.frame.require_frame_encryption.
by Benjamin Wright
· 6 years ago
9accc9f
Revert "Reland "Propagate media transport to media channel.""
by Oleh Prypin
· 6 years ago
da65ed2
Reland "Propagate media transport to media channel."
by Anton Sukhanov
· 6 years ago
276827c
Export symbols needed by the Chromium component build (part 3).
by Mirko Bonadei
· 6 years ago
37cf245
Revert "Propagate media transport to media channel."
by Oleh Prypin
· 6 years ago
8c16f74
Propagate media transport to media channel.
by Anton Sukhanov
· 6 years ago
3d25530
Reland "Export symbols needed by the Chromium component build (part 1)."
by Mirko Bonadei
· 6 years ago
5526e45
vp9: change x-google-profile-id to profile-id
by Philipp Hancke
· 6 years ago
16fe3f2
Revert "Export symbols needed by the Chromium component build (part 1)."
by Mirko Bonadei
· 6 years ago
99eea42
Reland "Reland "Export symbols needed by the Chromium component build (part 1).""
by Mirko Bonadei
· 6 years ago
84583f6
Enable End-to-End Encrypted Audio Payloads.
by Benjamin Wright
· 6 years ago
b49520b
Revert "Reland "Export symbols needed by the Chromium component build (part 1).""
by Mirko Bonadei
· 6 years ago
588f464
Reland "Export symbols needed by the Chromium component build (part 1)."
by Mirko Bonadei
· 6 years ago
2ea9af2
Revert "Export symbols needed by the Chromium component build (part 1)."
by Mirko Bonadei
· 6 years ago
9e24dcf
Export symbols needed by the Chromium component build (part 1).
by Mirko Bonadei
· 6 years ago
6ca9836
Prepare for per-media DSCP values. Push dscp for stun packets to the port layer where they are created.
by Tim Haloun
· 6 years ago
23eba22
Add support for RtpEncodingParameters num_temporal_layers.
by Åsa Persson
· 6 years ago
892acf0
Add support for send_encodings parameters in addTransceiver
by Florent Castelli
· 6 years ago
49ac595
Add GetSources to VideoRtpReceiver
by Jonas Oreland
· 6 years ago
965e794
Add sanity checks to UpdateDelayStatistics and patch unit tests.
by Johannes Kron
· 6 years ago
84df1c7
Make fewer copies when using StringBuilder.
by Jonas Olsson
· 6 years ago
bfd412e
Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel.
by Benjamin Wright
· 6 years ago
2e4419e
Add option to only request a frame interval change via OnOutputFormatRequest.
by Åsa Persson
· 6 years ago
366a50c
Remove simple stringstream usages.
by Jonas Olsson
· 6 years ago
3288168
Enable video adaptation for all screenshare content
by Ilya Nikolaevskiy
· 6 years ago
3df1d5d
Revert removal of simulcast screenshare experimental code (killswitch checks)
by Ilya Nikolaevskiy
· 6 years ago
f5f5373
Delete unused member MediaSenderInfo::packets_cached.
by Niels Möller
· 6 years ago
17aff35
Enable clang::find_bad_constructs for sdk/ (part 1).
by Mirko Bonadei
· 6 years ago
a3df0f2
Remove simulcast screenshare experimental code
by Ilya Nikolaevskiy
· 6 years ago
e41c433
Move sigslot to proper third_party directory
by Artem Titov
· 6 years ago
79eb4dd
Enabling clang::find_bad_constructs for libjingle_peerconnection_api.
by Mirko Bonadei
· 6 years ago
89b2963
Reland "Enable simulcast screenshare by default"
by Ilya Nikolaevskiy
· 6 years ago
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