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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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e59647b991f61cf1cf61b020356705e6c0f81257
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13f1a0a
Wire up x-google-{min,start,max}-bitrate to WebRtcVoiceMediaChannel.
by stefan
· 8 years ago
127387e
Delete nalu parser in mediaencoder
by kthelgason
· 8 years ago
2305b70
Remove unused build override
by kthelgason
· 8 years ago
cb861e0
Templatize percentile_filter.h and move it to base/analytics.
by terelius
· 8 years ago
4b9a2cb
Reland "Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate."
by minyue
· 8 years ago
26bddb9
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac
by ehmaldonado
· 8 years ago
18f7c8d
Remove warning suppression from VideoToolboxEncoder
by kthelgason
· 8 years ago
076c011
Change unit of logged bitrate stats in bytes/s to bits/s.
by asapersson
· 8 years ago
aff9636
Greatly reduce number of level controller tests.
by phoglund
· 8 years ago
6a2e20a
Make sure GetLastError on a PlatformThread return an error that is relevant to the thread.
by perkj
· 8 years ago
b49fc14
RtpDataEngine, FindCodecByName: Don't reassign codecs
by magjed
· 8 years ago
998df1d
Create webrtc/sdk/android folder
by magjed
· 8 years ago
78b4d56
Relanding "Pass time constant to bwe smoothing filter."
by minyue
· 8 years ago
8d66a5a
Disabled flaky P2PTestConductor tests on ASAN and MSAN.
by ossu
· 8 years ago
0245da0
Move ownership of PacketRouter from CongestionController to Call.
by nisse
· 8 years ago
06251f1
Reduce ProbeController::kDefaultMaxProbingBitrateBps to 10 mbps.
by philipel
· 8 years ago
f4a5942
Disabled all ScreenCapturerIntegrationTests on Windows
by ossu
· 8 years ago
3a864d2
MB: Add swarming bots in the FYI waterfall and remove memcheck swarming.
by ehmaldonado
· 8 years ago
b7e1dd7
Revert of Adding memcheck suppression. (patchset #1 id:1 of https://codereview.webrtc.org/2537563003/ )
by philipel
· 8 years ago
706a45e
Added missing include to fix waterfall compile error.
by hbos
· 8 years ago
f15a2c5
Delete deprecated versions of Copy, ScaleFrom and CropAndScaleFrom.
by nisse
· 8 years ago
0583b28
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
by hbos
· 8 years ago
0c43f77
Update video histograms that do not have a minimum lifetime limit before being recorded.
by asapersson
· 8 years ago
759e0b7
Fix memory leak in video_coding::PacketBuffer::InsertPacket.
by philipel
· 8 years ago
be74270
Calculate JitterBufferDelayInMs in the new jitter buffer.
by philipel
· 8 years ago
e69b468
Revert of Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (patchset #5 id:240001 of https://codereview.webrtc.org/2411613002/ )
by minyue
· 8 years ago
1731c9c
Use swap instead of copy in RtcHistogram::GetAndReset.
by asapersson
· 8 years ago
84e56d5
Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate.
by minyue
· 8 years ago
097529f
Remove 3 defines in voice_engine_configurations.h
by henrik.lundin
· 8 years ago
e61fbff
Use RotateDesktopFrame in DirectX capturer
by zijiehe
· 8 years ago
166e59a
Enable ScreenCapturerIntegrationTests
by zijiehe
· 8 years ago
6a46cf7
Roll chromium_revision db14e1adbd..b66d8ae9dc (435041:435081)
by buildbot
· 8 years ago
c9e80ee
Adding packet overhead to audio network adaptor.
by minyue
· 8 years ago
821dc7a
Roll chromium_revision 683745f53c..db14e1adbd (434997:435041)
by buildbot
· 8 years ago
a332877
Remove overhead from video bitrate.
by michaelt
· 8 years ago
c4dc4a5
Adding RTCStatsIntegrationTest to memcheck supressions.
by deadbeef
· 8 years ago
75f9d8c
Roll chromium_revision ffe8e7b51d..683745f53c (434954:434997)
by buildbot
· 8 years ago
290d43a
Add a new UMA metric in APM to track incoming capture-side audio level
by henrik.lundin
· 8 years ago
939e08f
Added webrtc/audio/utility directory and empty GN target.
by aleloi
· 8 years ago
ee414d9
Added sanity check to VCMDecodingState::UsingFlexibleMode to prevent OOB error.
by philipel
· 8 years ago
ad6f646
Use //build/dotfile_settings.gni to reduce blocked auto-rolls
by kjellander
· 8 years ago
768d625
Fix spelling mistake in RTP module declaration.
by brandtr
· 8 years ago
b890c95c
Replace some asserts with DCHECKs
by kwiberg
· 8 years ago
5049942
Refactor RMSLevel and give it new functionality
by henrik.lundin
· 8 years ago
1308c69
Roll chromium_revision 0496be2799..ffe8e7b51d (434847:434954)
by buildbot
· 8 years ago
f17cae2
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
by kjellander
· 8 years ago
668eb3b
Add overhead to transport feedback observer.
by michaelt
· 8 years ago
19223ac
Ignore newly added resource files.
by charujain
· 8 years ago
455b512
Landmine to clobber Windows builders
by Henrik Kjellander
· 8 years ago
1b5b22d
Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly.
by charujain
· 8 years ago
db346a7
RTCStatsIntegrationTest added.
by hbos
· 8 years ago
876222f
Move usage of QualityScaler to ViEEncoder.
by kthelgason
· 8 years ago
320e45a
Use RateCounter for input/sent fps stats. Reports average of periodically computed stats over a call.
by asapersson
· 8 years ago
65e5f5a
Roll chromium_revision d74a300097..0496be2799 (434704:434847)
by buildbot
· 8 years ago
bdd6f4c
Adding memcheck suppression.
by deadbeef
· 8 years ago
6cf94a0
Only use BoringSSL time callback in unit tests.
by deadbeef
· 8 years ago
352444f
RTC_[D]CHECK_op: Remove superfluous casts
by kwiberg
· 8 years ago
af476c7
RTC_[D]CHECK_op: Remove "u" suffix on integer constants
by kwiberg
· 8 years ago
80ed35e
Implement periodic bandwidth probing in application-limited region.
by sergeyu
· 8 years ago
bf22be9
Roll chromium_revision 2b5aa49038..d74a300097 (434640:434704)
by buildbot
· 8 years ago
fd87f4a
Opus: Move complexity variable out of conditional build flag
by henrik.lundin
· 8 years ago
1bc3146
Disable more VideoProcessorIntegrationTest tests on Linux 32-bit
by Henrik Kjellander
· 8 years ago
bb58435
Fix potential synchronization issues with framelisteners in EglRenderer.
by sakal
· 8 years ago
266f0a4
Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.
by philipel
· 8 years ago
d1aaaaa
Remove surface size mismatch logic from EglRenderer.
by sakal
· 8 years ago
6287e82
Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ )
by ossu
· 8 years ago
7703b27
Disable PeerConnectionEndToEndTest.CallWithLegacySdp on Asan bots.
by philipel
· 8 years ago
ceecea4
Pass selected cricket::VideoCodec down to internal H264 encoder
by magjed
· 8 years ago
20dce34
Fixed bug in PacketBuffer to correctly detect new complete frames after ClearTo has been called.
by philipel
· 8 years ago
e1a13f8
MB: Remove a --target-devices-file flag for JUnit tests on android.
by ehmaldonado
· 8 years ago
a8eb756
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
by aleloi
· 8 years ago
9abbf5a
Pass time constanct to bwe smoothing filter.
by michaelt
· 8 years ago
ffc6118
Don't cache video codec list in VideoEngine2.
by brandtr
· 8 years ago
ec1a670
Only create |remote_rate| when needed in RemoteBitrateEstimatorSingleStream.
by Rasmus Brandt
· 8 years ago
fb4a37a
Add memcheck suppressions.
by ehmaldonado
· 8 years ago
26fa6b2
Revert of Bug in ExtractFrame API (extracts frames incorrectly) (patchset #9 id:130001 of https://codereview.webrtc.org/2529923002/ )
by charujain
· 8 years ago
566cba1
Roll chromium_revision 5c22c2afac..2b5aa49038 (434448:434640)
by buildbot
· 8 years ago
b7636b4
Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly.
by charujain
· 8 years ago
2f58ec8
Add I420Buffer::Copy method taking plane pointers as input.
by nisse
· 8 years ago
e441bdb
Cleanup RtpSender hiding RtpHeaderExtensionLength function.
by danilchap
· 8 years ago
2fedf9c
Smooth BWE and pass it to Audio Network Adaptor.
by michaelt
· 8 years ago
847f689
Roll chromium_revision 5e821a778b..5c22c2afac (432715:434448)
by kjellander
· 8 years ago
deb95f3
Change rtc::TimeNanos and rtc::TimeMicros return value from uint64_t to int64_t.
by nisse
· 8 years ago
71b9b58
Revert of Move ADM specific Android files into modules/audio_device/android/ (patchset #2 id:20001 of https://codereview.webrtc.org/2533573002/ )
by solenberg
· 8 years ago
e8d8a2b
Move ADM specific Android files into modules/audio_device/android/
by solenberg
· 8 years ago
e69a1a9
Reland of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:1 of https://codereview.webrtc.org/2529143002/ )
by magjed
· 8 years ago
d7e6ccb
Revert of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:40001 of https://codereview.webrtc.org/2525693003/ )
by magjed
· 8 years ago
c7805db
Fix perf regression in screenshare temporal layer bitrate allocation
by sprang
· 8 years ago
fd34d30
iOS HW encoder: Enable H264 High profile support
by magjed
· 8 years ago
bdbc4b7
Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload
by magjed
· 8 years ago
1da1a09
Android HW encoder: Set constrained baseline as the profile
by magjed
· 8 years ago
03d6b08
Get rid of webrtc/base/latebindingsymboltable*
by ehmaldonado
· 8 years ago
f3feeff
Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ )
by magjed
· 8 years ago
0fa164a
Make Valgrind memcheck work in swarming.
by ehmaldonado
· 8 years ago
5732910
Revert of CQ: Disable android_more_configs trybot (patchset #1 id:1 of https://codereview.webrtc.org/2522953003/ )
by ehmaldonado
· 8 years ago
76622ce
Adding a unit test for RMSLevel
by henrik.lundin
· 8 years ago
293bc2a
Add 'Update LASTCHANGE' hook to DEPS
by ehmaldonado
· 8 years ago
5f7226f
Turn off error resilience for vp8 for no temporal layers if nack is enabled.
by asapersson
· 8 years ago
5dfac56
Keep all codec parameters in VideoReceiveStream::Decoder
by magjed
· 8 years ago
a6a699a
Sent bitrate stats are incorrect if FlexFEC is configured:
by asapersson
· 8 years ago
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