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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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08126349f5cc5d682a3398014a45088b4bb310f5
0812634
Pass a real audio codec pair ID to decoders that we create
by Karl Wiberg
· 7 years ago
92be1ca
Revert "Move rtp-specific config out of EncoderSettings."
by Niels Moller
· 7 years ago
247e0b4
Disabling periodic tasks on SSCC in unit tests.
by Sebastian Jansson
· 7 years ago
81e8a43
Roll chromium_revision 3ad75d0441..89bed79700 (544573:544673)
by Autoroller
· 7 years ago
29b204e
Tracking packet feedback availability in BitrateAllocator.
by Sebastian Jansson
· 7 years ago
fe617a3
Adding has_packet_feedback to LimitObserver callback.
by Sebastian Jansson
· 7 years ago
bc900cb
Move rtp-specific config out of EncoderSettings.
by Niels Möller
· 7 years ago
c3d1e09
Make sure RTCMTLVideoView.h ends up in framework headers.
by Anders Carlsson
· 7 years ago
08006d4
Android AppRTCMobile: Use new audio device code
by Magnus Jedvert
· 7 years ago
82fad3d
Remove TemporalLayersFactory and associated classes
by Erik Språng
· 7 years ago
8fc7948
Android: Generate audio JNI code
by Magnus Jedvert
· 7 years ago
37e3602
Android: Add henrika@ as owner of audio code
by Magnus Jedvert
· 7 years ago
e7fac68
Introduce Nullable annotation.
by Sami Kalliomäki
· 7 years ago
7531a76
Delete unused header media/base/test/mock_mediachannel.h.
by Niels Möller
· 7 years ago
9611442
Remove unneeded migration helper.
by Sami Kalliomäki
· 7 years ago
e61631d
Roll chromium_revision 2f0e6b63b5..3ad75d0441 (544446:544573)
by Autoroller
· 7 years ago
b88bfc9
Roll chromium_revision 8c0344a12e..2f0e6b63b5 (544337:544446)
by Autoroller
· 7 years ago
db67ba1
Report SRTP error codes to UMA
by Steve Anton
· 7 years ago
e9d2e4d
Provide the option of injecting rtc::TaskQueue when creating RtcEventLogImpl via factory methods.
by Dino Radaković
· 7 years ago
fe48ee9
Fixing zlib license generation.
by Mirko Bonadei
· 7 years ago
b38b05b
Adding srte as owner in modules/congestion_controller.
by Sebastian Jansson
· 7 years ago
89dd7bf
Move android audio device code into sdk/android
by Paulina Hensman
· 7 years ago
4d22a6d
Delete unneeded includes of wav_file.h and file_wrapper.h.
by Niels Möller
· 7 years ago
8ef59a4
Added data member access methods to FakeNetworkPipe.
by Christoffer Rodbro
· 7 years ago
3dc0125
Moving ConfigureEncoderTask to the calling scope.
by Sebastian Jansson
· 7 years ago
68a7168
Roll chromium_revision ee966518c2..8c0344a12e (544233:544337)
by Autoroller
· 7 years ago
2e0da5a
Remove EncodedFrame picture_id/spatial_layer references.
by philipel
· 7 years ago
f18072e
Enable SVC based on number of SSRCs.
by Sergey Silkin
· 7 years ago
eb98c72
Minor improvements in ADM unittest for Windows.
by henrika
· 7 years ago
faed538
Delete obsolete alias RateLimiter and rtc_base/ratelimiter.h
by Niels Möller
· 7 years ago
9047dac
Disable flaky test SendSideCongestionControllerTest/PacerQueueEncodeRatePushback.
by Rasmus Brandt
· 7 years ago
d328ec3
Change valueOf -> parseBoolean to avoid unneeded boxing
by Oleh Prypin
· 7 years ago
a98bb2d
Roll chromium_revision 02de71e34e..ee966518c2 (544129:544233)
by Autoroller
· 7 years ago
114155b
Roll chromium_revision 163641576c..02de71e34e (544029:544129)
by Autoroller
· 7 years ago
01cb5f2
Fix issue where sockets bound to temporary IPv6 addresses are discarded.
by Taylor Brandstetter
· 7 years ago
3d976f6
Discard link to media channel when audio sender stopped.
by Harald Alvestrand
· 7 years ago
bb60a3a
Refactor VP8 TemporalLayers
by Erik Språng
· 7 years ago
d757356
Fixing -Wstrict-prototypes warnings.
by Mirko Bonadei
· 7 years ago
def1ef5
New equality operators, for structs related to webrtc::VideoCodec.
by Niels Möller
· 7 years ago
5b3541f
RTCStatsCollector::GetStatsReport() with optional selector argument.
by Henrik Boström
· 7 years ago
4a73cd4
Adding tests of TaskQueueCongestionControl field trial.
by Sebastian Jansson
· 7 years ago
e62f600
Extend WavReader and WavWriter API.
by Artem Titov
· 7 years ago
451dfdf
Roll chromium_revision 7a8a322ad7..163641576c (543921:544029)
by Autoroller
· 7 years ago
0fa82a6
Moved FrameKey to api/video/encoded_frame.h and renamed it to VideoLayerFrameId.
by philipel
· 7 years ago
9c1ee36
Fix low_bandwidth_audio_perf_test resource dependency on Android
by Oleh Prypin
· 7 years ago
b9a02e5
Change place of UMA logging in AudioMixer.
by Alex Loiko
· 7 years ago
5370124
Replacing unique pointer with raw pointer in SSCC checks.
by Sebastian Jansson
· 7 years ago
04d4950
Revert "Using safe casts of allocation limits in Call."
by Oleh Prypin
· 7 years ago
4a9b4d6
Using safe casts of allocation limits in Call.
by Sebastian Jansson
· 7 years ago
8d8cb56
Delete obsolete methods from MockRtpTransportControllerSend
by Sebastian Jansson
· 7 years ago
b708e93
Bring mb up to date with Chromium's changes
by Oleh Prypin
· 7 years ago
8d2c5a8
Detangling target dependencies in rtc_base_approved.
by Tommi
· 7 years ago
7b2676f
Fix low_bandwidth_audio_perf_test binary dependency on Windows
by Oleh Prypin
· 7 years ago
d2c8332
Revert "Relaxing no-streams presubmit check (streams are allowed in tests)."
by Mirko Bonadei
· 7 years ago
7311918
Add an example app for iOS native API.
by Anders Carlsson
· 7 years ago
8cf0a87
Reland "Split perf-test-specific resources in low_bandwidth_audio_test"
by Oleh Prypin
· 7 years ago
73ac908
Relaxing no-streams presubmit check (streams are allowed in tests).
by Mirko Bonadei
· 7 years ago
9fa35e5
Fix path to proto in py_event_log_analyzer/pb_parse.py
by Oleh Prypin
· 7 years ago
8a1b20a
Roll chromium_revision 33aa22e76e..7a8a322ad7 (543816:543921)
by Autoroller
· 7 years ago
cdd2a97
Roll chromium_revision ce851e47bd..33aa22e76e (543685:543816)
by Autoroller
· 7 years ago
317a522
Fixes to posting delayed process tasks in SSCC.
by Sebastian Jansson
· 7 years ago
4ccc1c4
Don't destroy a receive stream's sink before reassigning it.
by Oskar Sundbom
· 7 years ago
3bb1194
Revert "Add 'is_chrome_branded' guard to the default of 'rtc_use_h264'"
by Patrik Höglund
· 7 years ago
3546835
Roll chromium_revision e0e02de5a7..ce851e47bd (543578:543685)
by Autoroller
· 7 years ago
bf3dbb4
Delete payload_type from VCMEncoderDatabase and vcm::VideoSender.
by Niels Möller
· 7 years ago
5bf8ccd
Delete encoder caching in WebRtcVideoSendStream.
by Niels Möller
· 7 years ago
677f42c
Enable ContinuousAfterStreamCountChangeSimulcastEncoderAdapter picture id tests.
by Åsa Persson
· 7 years ago
d132ce1
Remove unnecessary copies from AsyncInvoke
by Cameron Pickett
· 7 years ago
465a5d9
Refactor payload types constants in CallTest
by Ilya Nikolaevskiy
· 7 years ago
af9e87b
Delete unused methods from vcm::VideoCodingModule.
by Niels Möller
· 7 years ago
eef09fc
Fix race in DegradedCall::DestroyVideoSendStream
by Erik Språng
· 7 years ago
883d00f
Add support of AAudio in native WebRTC on Android O and above
by henrika
· 7 years ago
815f3b6
Fix podspec iOS version.
by Kári Tristan Helgason
· 7 years ago
7696bef
Remove the public_deps to fileutils from test_support.
by Patrik Höglund
· 7 years ago
4de9eb2
Roll chromium_revision bc2c5b551b..e0e02de5a7 (543473:543578)
by Autoroller
· 7 years ago
8870f55
Roll chromium_revision d94f7320ab..bc2c5b551b (543368:543473)
by Autoroller
· 7 years ago
24c220c
Changed target_angle_degrees in audioproc_float to float to avoid integer division when converting to radians
by Alex Luebs
· 7 years ago
895ae9a
Improving the speed of the delay estimator in AEC3
by Per Åhgren
· 7 years ago
1d037ae
Don't crash in SingleNalu packetization for h264 if no space in packet
by Ilya Nikolaevskiy
· 7 years ago
4425b05
Add video send stream test to check switch to and from screenshare
by Ilya Nikolaevskiy
· 7 years ago
aca5a7d
Improvements to network control types.
by Sebastian Jansson
· 7 years ago
dfe6bcd
Roll chromium_revision 670c468885..d94f7320ab (543262:543368)
by Autoroller
· 7 years ago
56f9f0e
Make task_queue_ injectable by wrapping it into a std::unique_ptr and adding an optional arg to the constructor of RtcEventLogImpl.
by Dino Radaković
· 7 years ago
3ab1d26
Exposing WebRTC-Audio-SendSideBwe-For-Video field trial
by Alex Narest
· 7 years ago
650a826
Revert "Reland "Split perf-test-specific resources in low_bandwidth_audio_test""
by Oleh Prypin
· 7 years ago
63b7574
Add check for negative max bitrate in VideoSendStream.
by Sebastian Jansson
· 7 years ago
7bd79a0
Split up audio_device build target
by Paulina Hensman
· 7 years ago
5f1a31c
Adding a smooth transition from the startup phase parameter set in AEC3
by Per Åhgren
· 7 years ago
b3808dc
Reland "Split perf-test-specific resources in low_bandwidth_audio_test"
by Oleh Prypin
· 7 years ago
2cb4105
Moved audioproc_f interface into api directory.
by Ivo Creusen
· 7 years ago
bb894ff
Make PayloadRouter own the picture id and tl0 pic idx sequences.
by Niels Möller
· 7 years ago
9f64b9c
Reland "Remove unnecessary dependency on base."
by Patrik Höglund
· 7 years ago
853715c
Min BWE default is 10kbps but for audio send side BWE it was overridden to 5kbps. Now audio send side BWE is used for video calls too and should set min to 10kbps in case of video call.
by Alex Narest
· 7 years ago
cc681cc
Split vp8_impl into webm_vp8_encoder and webm_vp8_decoder
by Erik Språng
· 7 years ago
b3bac5e
Revert "Remove unnecessary dependency on base."
by Patrik Höglund
· 7 years ago
6f2fcb4
Add more Audio Mixer and Fixed Gain Controller metrics.
by Alex Loiko
· 7 years ago
aaa882c
Revert "Split perf-test-specific resources in low_bandwidth_audio_test"
by Oleh Prypin
· 7 years ago
e0eb13c
Remove unnecessary dependency on base.
by Patrik Höglund
· 7 years ago
180d992
Style guide: State what version of C++ we should use
by Karl Wiberg
· 7 years ago
4bbc150
Split perf-test-specific resources in low_bandwidth_audio_test
by Oleh Prypin
· 7 years ago
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