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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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78a71386005ceac9bca63329a1c247f91a53e12c
78a7138
Remove MediaTransport from Call.
by Tommi
· 5 years ago
44327c3
Update test::CreateVideoStreams to use configured scale_resolution_down_by if set.
by Åsa Persson
· 5 years ago
383adc0
Delete shim of PRId64 et al. on Windows
by Oleh Prypin
· 5 years ago
0d210ee
Change return type of of ReceiveStatistics::Create to unique_ptr.
by Niels Möller
· 5 years ago
c2fe547
Remove unused fallbacks in PacedSender
by Erik Språng
· 5 years ago
eac47f7
Removing unused fallback variant for the reverb computation
by Per Åhgren
· 5 years ago
891d393
Call Call::GetStats() from the correct thread in ProbingEndToEndTest.
by Tommi
· 5 years ago
aaaf804
Call Call::GetStats() from the correct thread in VideoSendStreamTest.
by Tommi
· 5 years ago
efffd0a
Roll chromium_revision 3d0c04364f..5744654b26 (684897:685023)
by chromium-webrtc-autoroll
· 5 years ago
307448f
Roll chromium_revision 006302cd2e..3d0c04364f (684781:684897)
by chromium-webrtc-autoroll
· 5 years ago
5b5d97c
Reland of "Reporting of decoding_codec_plc events""
by Alex Narest
· 5 years ago
2d2bbb1
Filter out duplicate receive codecs in the media engine
by Steve Anton
· 5 years ago
3cc2f70
Roll chromium_revision 192da69226..006302cd2e (684664:684781)
by chromium-webrtc-autoroll
· 5 years ago
b168678
Add RTC_ prefix to non-standard format specifier macro "PRIdNS"
by Oleh Prypin
· 5 years ago
12ebfa6
Delete RtcpStatisticsCallback from ReceiveStatistics
by Niels Möller
· 5 years ago
b668542
Delete unused format specifier macros for NSInteger and NSUInteger
by Oleh Prypin
· 5 years ago
83bbe91
Delete deprecated rtc_event_log header
by Danil Chapovalov
· 5 years ago
e08648d
Add `AbsoluteCaptureTime` to `RtpPacketInfo`.
by Chen Xing
· 5 years ago
f40a340
Remove deprecated code related to AEC2
by Per Åhgren
· 5 years ago
75caef7
Delete unused ACM members isac_decoder_16k_ and isac_decoder_32k_
by Niels Möller
· 5 years ago
d2845f8
Removes unused AudioAllocationSettings from voice engine.
by Sebastian Jansson
· 5 years ago
d23f67e
Call Call::GetStats() from the correct thread in StatsEndToEndTest.
by Tommi
· 5 years ago
c24a5b1
Fix CallPerfTests to call Call::GetStats() from the right thread.
by Tommi
· 5 years ago
c653172
Delete obsolete method AudioCodingModule::SetBitRate
by Niels Möller
· 5 years ago
e71edc5
Roll chromium_revision 838e9d2793..192da69226 (684401:684664)
by chromium-webrtc-autoroll
· 5 years ago
1e49ab2
Migrate part of Vp9 SVC tests on PC framework. Add temporal layers support.
by Artem Titov
· 5 years ago
8dcaed9
Split VideoFrameWriter into yuv and y4m writers
by Artem Titov
· 5 years ago
9d62a56
Roll chromium_revision 9d357a520c..838e9d2793 (684300:684401)
by chromium-webrtc-autoroll
· 5 years ago
00c7ecf
Surface CandidatePairChange event
by Alex Drake
· 5 years ago
63c38e2
Fix for incorrect transport sequence number config for audio in scenario tests.
by Sebastian Jansson
· 5 years ago
7cbee84
Reland "Adds PeerConnection scenario test framework."
by Sebastian Jansson
· 5 years ago
c648819
DegradedCall: fake network using TaskQueue instead of ProcessThread
by Erik Språng
· 5 years ago
bb1f245
Disable RunPythonTests on rtc_tools.
by Mirko Bonadei
· 5 years ago
61b1590
Roll chromium_revision 8776a3887d..9d357a520c (684182:684300)
by chromium-webrtc-autoroll
· 5 years ago
2e6c294
Refactor test_peer.cc to reduce amount of arguments passing around
by Artem Titov
· 5 years ago
e6b7b66
Fix CallClient so that it calls Call::GetStats() on the right thread.
by Tommi
· 5 years ago
a22cab8
Calling DebugBreak() on Windows during fatal checks instead of relying on abort().
by Tommi
· 5 years ago
7ba3b81
Delete class PlatformFile.
by Niels Möller
· 5 years ago
10da4a0
Fix RtpFrameReferenceFinderFuzzer to not generate invalid input
by Johannes Kron
· 5 years ago
c89468a
Fix CallStatsUnittests to update the RTT on the process thread (as in production).
by Tommi
· 5 years ago
4d7c405
Split out RtcpCnameCallback from RtcpStatisticsCallback
by Niels Möller
· 5 years ago
ed44f54
In ChannelReceive, use AcmReceiver directly, not AudioCodingModule
by Niels Möller
· 5 years ago
e80885a
Call Call::GetStats() from the correct thread in our bandwidth tests.
by Tommi
· 5 years ago
5e005f4
Fix RampUp tests to call Call::GetStats() from the right thread - and remove the need for a dedicated polling thread.
by Tommi
· 5 years ago
bdc9096
Roll chromium_revision 2c4c2e2ea6..8776a3887d (684065:684182)
by chromium-webrtc-autoroll
· 5 years ago
074f0d2
Roll chromium_revision 7c6275bdfa..2c4c2e2ea6 (683711:684065)
by chromium-webrtc-autoroll
· 5 years ago
9b1700c
Enable field trial LegacySimulcastLayerLimit by default
by Florent Castelli
· 5 years ago
45231be
AEC3: Removing unused code in the echo subtractor
by Per Åhgren
· 5 years ago
cdbaeeb
Aec3:Remove unused legacy code
by Per Åhgren
· 5 years ago
d7ee76c
Wire up field trials for some experimental screenshare settings
by Erik Språng
· 5 years ago
8d41058
Remove unused rtc_tools/video_analysis.py.
by Mirko Bonadei
· 5 years ago
b56cca3
Remove the old `ContributingSources` class.
by Chen Xing
· 5 years ago
3d351c6
Revert "Adds PeerConnection scenario test framework."
by Sebastian Jansson
· 5 years ago
ad5c4ac
Adds PeerConnection scenario test framework.
by Sebastian Jansson
· 5 years ago
139f4dc
QualityScaler: Add option to try fast adapt down at start up based on initial bw estimates.
by Åsa Persson
· 5 years ago
fedd625
Change 2g network pc audio test to more realistic network
by Artem Titov
· 5 years ago
d75b3c4
Roll chromium_revision 2ab7c1917b..7c6275bdfa (683574:683711)
by chromium-webrtc-autoroll
· 5 years ago
a285909
Revert "Adding new top-level directory crypto/"
by Mirko Bonadei
· 5 years ago
df7c5f1
Roll chromium_revision 01452febf2..2ab7c1917b (683465:683574)
by chromium-webrtc-autoroll
· 5 years ago
8bbdb5b
Update VideoBitrateAllocator allocate to take a struct with more fields
by Florent Castelli
· 5 years ago
9a9f18a
Get WebRTC.Video.ReceivedPacketsLostInPercent from ReceiveStatistics
by Niels Möller
· 5 years ago
054e3bb
Reland "Replace the implementation of `GetContributingSources()` on the audio side."
by Chen Xing
· 5 years ago
59bbd65
Add ToString method for AudioProcessing::Config
by Artem Titov
· 5 years ago
6563934
Revert "Sanitize the codec list before sending it to the media engine"
by Artem Titov
· 5 years ago
5e155a6
ReportBlockStatsTest: Remove usage of RTCPReportBlock (no longer used).
by Åsa Persson
· 5 years ago
916fda5
Sync download_tools.py with changes in gclient_utils module.
by Sergey Silkin
· 5 years ago
9160b62
Improve thread safety of AndroidVideoTrackSource::SetState.
by Sami Kalliomäki
· 5 years ago
b3f78de
Reland "Don't use all_dependent_configs for sdk frameworks"
by Oleh Prypin
· 5 years ago
28ee3da
Roll chromium_revision 2d438bebcd..01452febf2 (683346:683465)
by chromium-webrtc-autoroll
· 5 years ago
402f625
Roll chromium_revision 9c759119a4..2d438bebcd (683187:683346)
by chromium-webrtc-autoroll
· 5 years ago
0c67c80
Guard video analyzer against race conditions.
by Yves Gerey
· 5 years ago
59a041d
Roll chromium_revision 4b9d901264..9c759119a4 (683063:683187)
by chromium-webrtc-autoroll
· 5 years ago
bd3f305
Request a new key frame if packet buffer is cleared
by Johannes Kron
· 5 years ago
77d3efc
Simplify ReportBlockStats
by Niels Möller
· 5 years ago
84de3d9
Factor framework dependencies out of audio_device_impl
by Oleh Prypin
· 5 years ago
32eaa7b
Roll chromium_revision d488661c95..4b9d901264 (682945:683063)
by chromium-webrtc-autoroll
· 5 years ago
da4f093
Reland "Only include payload in bytes sent/received."
by Bjorn A Mellem
· 5 years ago
fd643a4
Build core audio for older windows versions
by Paul Roberts
· 5 years ago
076bdba
Roll chromium_revision 1bdd79185b..d488661c95 (682806:682945)
by chromium-webrtc-autoroll
· 5 years ago
ea9798c
Update video quality tests units reporting.
by Artem Titov
· 5 years ago
032ea9c
Remove "Missing track ID" StatsCollector log message
by Steve Anton
· 5 years ago
add7ef9
Sanitize the codec list before sending it to the media engine
by Steve Anton
· 5 years ago
f6f3ecf
Roll chromium_revision 6ba1e90223..1bdd79185b (682701:682806)
by chromium-webrtc-autoroll
· 5 years ago
8d3e4bd
Adds simulated network node builder.
by Sebastian Jansson
· 5 years ago
e05ae5b
Adds non-forwarding frame tap to video frame matcher.
by Sebastian Jansson
· 5 years ago
53571c7
Adds config based creation of frame generator capturer.
by Sebastian Jansson
· 5 years ago
d0b67c2
Roll chromium_revision 1443fbac59..6ba1e90223 (682581:682701)
by chromium-webrtc-autoroll
· 5 years ago
928e7a3
Make ID of datachannel stats not depend on dc.id
by Harald Alvestrand
· 5 years ago
97e015f
Update metrics units for better fit in DefaultVideoQualityAnalyzer
by Artem Titov
· 5 years ago
60ffc31
Fix potential crash if nack is being processed when media gets disabled
by Erik Språng
· 5 years ago
f034b86
Remove dead code from DefaultVideoQualityAnalyzer
by Artem Titov
· 5 years ago
a1b99b3
Use total_decode_time_ms in VideoAnalyzer
by Johannes Kron
· 5 years ago
6acb069
Adds default for PeerConnectionObserver::OnIceConnectionChange
by Sebastian Jansson
· 5 years ago
7702c8a
Fix SendSideBweWithOverhead using new pacer code path.
by Erik Språng
· 5 years ago
e529ecb
Roll chromium_revision bb33f8cf70..1443fbac59 (682476:682581)
by chromium-webrtc-autoroll
· 5 years ago
8e967df
Use unique_ptr in JsepCandidateCollection
by Steve Anton
· 5 years ago
e08ca23
Include overlapping dialogs when capturing a window via screen cropping
by Bryan Ferguson
· 5 years ago
5ae853a
Roll chromium_revision 394c077f2b..bb33f8cf70 (682353:682476)
by chromium-webrtc-autoroll
· 5 years ago
4ed5b08
Skip empty strings in ToUtf(8|16).
by Noah Richards
· 5 years ago
fe431e4
Roll chromium_revision 0959b866c2..394c077f2b (682230:682353)
by chromium-webrtc-autoroll
· 5 years ago
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