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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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99f8e0841a9d5d49bb3bcccc1e19df56449b4a2d
99f8e08
Add a chart for packet loss on incoming streams.
by Stefan Holmer
· 8 years ago
073378e
Avoids crash at device switch on iOS by ensuring that audio parameters can be updated on the fly driven by e.g. switching audio device.
by henrika
· 8 years ago
2b11fd2
rtc::Optional: Tell sanitizers that unset values aren't OK to access
by kwiberg
· 8 years ago
463d301
Added ClearTo(seq_num) to RtpFrameReferenceFinder.
by philipel
· 8 years ago
d547224
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
by kthelgason
· 8 years ago
27c7b8f
VadCore: Allow signed multiplication overflow that we don't know how to fix
by kwiberg
· 8 years ago
3fa3517
Filter objc headers in cpplint presubmit check
by Kári Tristan Helgason
· 8 years ago
9c8c586
MB: Disable more parts of the GYP build.
by kjellander
· 8 years ago
499dcb1
Remove references to .isolate files that are no longer needed.
by kjellander
· 8 years ago
bd3dda6
Renamed RTCStatsReport to RTCLegacyStatsReport in objc files.
by hbos
· 8 years ago
b0afd97
Revert of Only expose gflags target in non-Chromium and non-fuzzer builds. (patchset #1 id:40001 of https://codereview.webrtc.org/2321963002/ )
by kjellander
· 8 years ago
961168a
Add sakal as an OWNER to some Android files.
by sakal
· 8 years ago
ce2e136
Update AvgCounter to have the ability to include last period metric for subsequent intervals without samples (e.g. for non-periodic updated stats).
by asapersson
· 8 years ago
2a5f371
Make UMA stats creation available in the Java interface.
by sakal
· 8 years ago
9365338
Only expose gflags target in non-Chromium and non-fuzzer builds.
by kjellander
· 8 years ago
aa85cac
Add magjed@ as owner of webrtc/common_video
by magjed
· 8 years ago
432950c
Revert of Add a DEPS gclient hook to prune corrupt mockito remote. (patchset #1 id:1 of https://codereview.webrtc.org/2326523002/ )
by kjellander
· 8 years ago
5865f48
Revert of Separating video settings in VideoQualityTest. (patchset #2 id:20001 of https://codereview.webrtc.org/2312613003/ )
by kjellander
· 8 years ago
906f403
This CL refactors the buffering of the incoming near-end signal inside
by peah
· 8 years ago
0e62f2b
Change owner of webrtc/test/channel_transport to solenberg@.
by henrikg
· 8 years ago
f07fb00
Separating video settings in VideoQualityTest.
by minyue
· 8 years ago
3115b06
Add a DEPS gclient hook to prune corrupt mockito remote.
by ehmaldonado
· 8 years ago
13eef785
Revert of Don't use VoE legacy APIs in force_mic_volume_max tool. (patchset #5 id:80001 of https://codereview.webrtc.org/2268183007/ )
by solenberg
· 8 years ago
0f8ea0d
Avoids crash in WebRtcAudioTrack.initPlayout (part II)
by henrika
· 8 years ago
ae9f2bd
Don't use VoE legacy APIs in force_mic_volume_max tool.
by solenberg
· 8 years ago
49fbbe0
Force a Chromium sync on all bots.
by Henrik Kjellander
· 8 years ago
4e0581f
Revert of move all reference to carbon api (patchset #2 id:300001 of https://codereview.webrtc.org/2321493002/ )
by magjed
· 8 years ago
7e4b604
Android ThreadUtils: Propagate exceptions in invoke functions
by magjed
· 8 years ago
22c8d5a
Revert of Setting up an RTP input fuzzer for NetEq (patchset #2 id:20001 of https://codereview.webrtc.org/2315633002/ )
by henrik.lundin
· 8 years ago
17e3fa1
Removed sync packet support from NetEq.
by ossu
· 8 years ago
2c993ce
Avoids crash in WebRtcAudioTrack.initPlayout
by henrika
· 8 years ago
5b356f4
FilePlayer: Remove backwards compatibility stuff that we no longer need
by kwiberg
· 8 years ago
acf9f47
GN Templates: Introduce rtc_shared_library
by ehmaldonado
· 8 years ago
76cd281
MB: Move Linux 32 bots from the WebRTC FYI to the main waterfall.
by ehmaldonado
· 8 years ago
a90879b
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2316563002/ )
by kthelgason
· 8 years ago
71eb61c
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
by magjed
· 8 years ago
4e869e9
A more useful gyp_flag_compare script
by ehmaldonado
· 8 years ago
243c0e8
Fixing NetEqReplacementInput for reordered and missing packets
by henrik.lundin
· 8 years ago
ac398f2
Python event log analyzer tool: fix of indexing issue.
by aleloi
· 8 years ago
a4c2106
This CL contains the following small changes:
by aleloi
· 8 years ago
250fd97
Use RateCounter for received bitrate stats:
by asapersson
· 8 years ago
14f1250
Do not report bucket delay for stats when pacer is paused (zero returned).
by asapersson
· 8 years ago
a264ecc
Copy RTCAudioSource.h and RTCMediaSource.h with other public header files when building the WebRTC framework for iOS / Mac
by VladimirTechMan
· 8 years ago
14b9d79
If encoding is inactive, don't start sending when stream is reconfigured.
by Taylor Brandstetter
· 8 years ago
7610f85
Adding AudioNetworkAdaptor interfaces.
by minyue
· 8 years ago
656ad48
Revert of CQ: Remove linux_baremetal until it's back (patchset #1 id:1 of https://codereview.webrtc.org/2322463002/ )
by kjellander
· 8 years ago
0f49dac
Reland of [WebRTC] A real ScreenCapturer test (patchset #1 id:1 of https://codereview.webrtc.org/2310953002/ )
by zijiehe
· 8 years ago
f581eb7
Renamed and restructured the protobuf definitions for the rtc_event_log graphs.
by skvlad
· 8 years ago
a4d40cb
Fixing stack buffer overflow (read) in SctpDataEngine.
by Taylor Brandstetter
· 8 years ago
652ac89
Simplifications of the mixing algorithm.
by aleloi
· 8 years ago
88499ec
Moving/renaming webrtc/common.h.
by solenberg
· 8 years ago
4016a0b
GN: Move variables from //build_overrides/webrtc.gni to //webrtc/build/webrtc.gni
by ehmaldonado
· 8 years ago
d52063f
Change OverUseFrameDetector to use a task queue instead of ProcessThread to periodically check for overuse. It is made to only operate on a single task queue.
by perkj
· 8 years ago
311525e
Several lock acquisitions and one of the two lock members are removed. ENSURE_LOCKS_REQUIRED and CalledOnValidThread annotations are added.
by aleloi
· 8 years ago
08b0351
Implement PlayoutDelay extension as a trait to be used with rtp::Packet class
by Danil Chapovalov
· 8 years ago
10e8f8e
Relax expectation in EndToEndTest.CallReportsRttForSender test
by Danil Chapovalov
· 8 years ago
2d273f1
Setting up an RTP input fuzzer for NetEq
by henrik.lundin
· 8 years ago
947c66c
CQ: Remove linux_baremetal until it's back
by Henrik Kjellander
· 8 years ago
9881cb2
Merge min_ms and max_ms accessors in PlayoutDelayOracle
by Danil Chapovalov
· 8 years ago
126ee72
Only parse PPS up to PPS and SPS ids in the depacketizater.
by Stefan Holmer
· 8 years ago
d4626e5
Disable -Wsentinel warning for Linux 32-bit builds.
by kjellander
· 8 years ago
60e4346
Add time line for acked bitrate.
by Stefan Holmer
· 8 years ago
b9d8d10
Fixed flaky StunRequestTests which depended on the wall clock
by skvlad
· 8 years ago
b460fd8
Increase timeout for flaky tests for ProcessThreadImpl
by skvlad
· 8 years ago
cfaca03
Add dynamic bitrate tracker and adjustment for Exynos VP8 HW encoder.
by Alex Glaznev
· 8 years ago
37062ed
Fix chromium-style errors in IntelligibilityEnhancer
by Alejandro Luebs
· 8 years ago
306d52b
Reland of Delete cricket::VideoFrame::GetTimeStamp. (patchset #1 id:1 of https://codereview.webrtc.org/2315703002/ )
by nisse
· 8 years ago
eb24dd0
Improvements to UI to python event log analyzer tool.
by aleloi
· 8 years ago
d52bef7
iSAC float: Handle errors in upper band decoding
by kwiberg
· 8 years ago
92b2e08
Revert of Delete cricket::VideoFrame::GetTimeStamp. (patchset #2 id:150001 of https://codereview.webrtc.org/2310043002/ )
by nisse
· 8 years ago
fa1ba19
Reland of Delete cricket::VideoFrame::GetTimeStamp. (patchset #1 id:1 of https://codereview.webrtc.org/2306953002/ )
by nisse
· 8 years ago
857a8fb
Remove dedicated unittest file for remb format
by Danil Chapovalov
· 8 years ago
dedc822
MB: Cleanup no-longer-used GN configurations
by kjellander
· 8 years ago
edaa4a4
GN: Set WEBRTC_RESTRICT_LOGGING as is set in GYP.
by ehmaldonado
· 8 years ago
859e861
Remove stop method from VideoTrackSourceInterface.
by sakal
· 8 years ago
c928453
Significantly increased max_num_buffers_ of Vp9FrameBufferPool.
by hbos
· 8 years ago
d4ec970
neteq_rtpplay: Add an error message for unmatched SSRC
by henrik.lundin
· 8 years ago
891eaac
MB: Add Linux 32-bit Debug and Release trybots
by Henrik Kjellander
· 8 years ago
b096aa7
Revert of Remove all reference to carbon api (patchset #2 id:20001 of https://codereview.webrtc.org/2299633002/ )
by kthelgason
· 8 years ago
dbd8b6b
Remove all reference to carbon api
by kthelgason
· 8 years ago
0b1b472
Enable gyp_compare_flag_script to compare all targets.
by ehmaldonado
· 8 years ago
b27c091
Exclude flaky PhysicalSocketTest tests on Dr Memory bots.
by Henrik Kjellander
· 8 years ago
aa90b31
[WebRTC] Two DirectX capturers cannot work concurrently
by zijiehe
· 8 years ago
fef8653
An early analysis shows in DirectX based capturer, Windows API returns larger
by zijiehe
· 8 years ago
f71d298
GN: Apply optimize_max only on windows
by ehmaldonado
· 8 years ago
1e714ae
RtcpReceiverTest rewritten using public available interface
by danilchap
· 8 years ago
d313403
rtc::Buffer: Let SetData and AppendData accept anything with .data() and .size()
by kwiberg
· 8 years ago
84c8528
Use RtpPacketToSend in RtpSenderAudio.
by danilchap
· 8 years ago
4c44202
Revert of [WebRTC] A real ScreenCapturer test (patchset #8 id:240001 of https://codereview.webrtc.org/2268093002/ )
by asapersson
· 8 years ago
d6e65b3
MB: Add WebRTC FYI bots to mb_config.pyl.
by ehmaldonado
· 8 years ago
e9cc686
GN Templates: Move common_inherited_config to the template.
by ehmaldonado
· 8 years ago
bd43172
Restrict the 1-argument ArrayView constructor to types with .size() and .data()
by kwiberg
· 8 years ago
59af8b7
GN Templates: Use the optimize_max compiler config.
by ehmaldonado
· 8 years ago
ccd7489
Plot accumelated packets over time.
by philipel
· 8 years ago
7556282
FrameBuffer::NextFrame now return a ReturnReason and take an additional std::unique_ptr<FrameObject>* as output parameter.
by philipel
· 8 years ago
c82f2e1
RTCStatsCollector collecting stats on multiple threads.
by hbos
· 8 years ago
7a2ce0b
GN Templates: Move common_config to the template.
by ehmaldonado
· 8 years ago
74c10b5
Introduce webrtc::VideoFrame::timestamp_us, and corresponding constructor.
by nisse
· 8 years ago
3821ff8
Only log BWE updates if the actual estimate changed or if we have non-zero loss reports.
by stefan
· 8 years ago
a973f95
Remove restart method from VideoTrackSourceInterface.
by sakal
· 8 years ago
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