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webrtc
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src
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webrtc
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f427d57aaab7d5260b788a0192cb3ae01244f489
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modules
c5805c3
Remove deprecated RTPSender::SendPadData
by danilchap
· 8 years ago
28cdc2d
Remove static cast from H264SpropParameterSets.
by johan
· 8 years ago
87e0263
Improvements to the reliability of the echo detector perf test.
by ivoc
· 8 years ago
604324f
Corrected access of null pointer in audioproc_f:
by peah
· 8 years ago
9817ab9
Removes verification of audio parameters on Android
by henrika
· 8 years ago
d8cd9cc
Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
by nisse
· 8 years ago
0c0354e
Disable flaky test VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9
by skvlad
· 8 years ago
d7e5b79
This CL adds the basic framework for AEC3 in the audio processing module.
by peah
· 8 years ago
e7a1876
Delete stl_util.h. Unused since cl https://codereview.webrtc.org/2447103002
by nisse
· 8 years ago
3dc2796
Disable AudioDeviceTest.StartPlayoutOnTwoInstances on iOS
by Henrik Kjellander
· 8 years ago
6dab241
Delete VideoFrame default constructor, and IsZeroSize method.
by nisse
· 8 years ago
a0aa420
Disable flaky QualityScaler tests for now.
by kthelgason
· 8 years ago
f11c1ff
Fix for negative shift value in NetEq.
by ivoc
· 8 years ago
0fd94b0
Making audio network adaptor config proto a JAVA package.
by minyue
· 8 years ago
4899054
Destroy encoders that fail to InitEncode.
by noahric
· 8 years ago
c7eca0b
Add OWNERS to BWE modules.
by stefan
· 8 years ago
a329255
Remove sequenced task checker from FlexfecSender.
by brandtr
· 8 years ago
a1daebb
Delete voice_engine_configurations.h
by henrik.lundin
· 8 years ago
72e2c02
Disabling the potentially flaky test
by peah
· 8 years ago
469b3a4
Replace VideoCaptureDataCallback by VideoSinkInterface.
by nisse
· 8 years ago
b2da7b1
Change MANUAL -> DISABLED for ScreenCapturerIntegrationTest tests
by Henrik Kjellander
· 8 years ago
722b793
MANUAL tests of GDI capturers
by zijiehe
· 8 years ago
dcc26fa
Implement Theil-Sen's method for fitting a line to noisy data (used in bandwidth estimation).
by terelius
· 8 years ago
c47a9e1
During AEC development, it is handy to be able to simulate different
by peah
· 8 years ago
6c8d156
When recreating a call based on an aecdump recording the nearend used
by peah
· 8 years ago
7e1744b
Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
by asapersson
· 8 years ago
1940860
Fix issue with deprecated CongestionController interface not working.
by stefan
· 8 years ago
9d0f636
Enable screen capturer tests for Linux / DirectX capturer / magnifier capturer
by zijiehe
· 8 years ago
c4367eb
Refactor webrtc/modules/video_{capture,coding} for GN check
by ehmaldonado
· 8 years ago
5fbf11e
Do not update OnReceivedRtcpReceiverReport if report block list is empty (and rtt zero).
by asapersson
· 8 years ago
b45ca0b
Delete deprecated CongestionController constructor and packet_router method.
by nisse
· 8 years ago
7a45bfc
Re-enable disabled VideoProcessorIntegrationTest tests
by kjellander
· 8 years ago
d2ddca2
Simplify an always true condition.
by nisse
· 8 years ago
aab4e12
Refactor webrtc/modules/audio_processing for GN check
by ehmaldonado
· 8 years ago
a18a530
Decode h264 fmtp sprop-parameter-sets to binary.
by johan
· 8 years ago
c0d258d
Injectable output rate calculater for AudioMixer.
by aleloi
· 8 years ago
8fd4fc6
Remove unused arguments and variable in MediaOptimization.
by asapersson
· 8 years ago
d9d5bdf
Log BitBlt failure
by zijiehe
· 8 years ago
27e08d9
Refactor webrtc/{api,audio} and modules/audio_coding for GN check
by kjellander
· 8 years ago
31212ba
Moved call.h and most of api/call/* into call/
by ossu
· 8 years ago
c49a0f7
Adding OnReceivedOverhead to AudioEncoder.
by minyue
· 8 years ago
666ab2c
Make ostream<< for enum class H264PacketizationMode
by hta
· 8 years ago
8e73c64
Remove deprecated comments
by zijiehe
· 8 years ago
afc16ca
Reject XR TargetBitrate items with unsupported layer indices
by sprang
· 8 years ago
c939473
Reland of H.264 packetization mode 0 (try 3) (patchset #1 id:1 of https://codereview.webrtc.org/2558453002/ )
by hta
· 8 years ago
6fe85c0
Move /webrtc/api/android files to /webrtc/sdk/android
by magjed
· 8 years ago
dcff992
Reland of APM: Change 3 UMA metrics to fewer but linearly distributed buckets (patchset #1 id:1 of https://codereview.webrtc.org/2548333002/ )
by henrik.lundin
· 8 years ago
f8ffba3
Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ )
by hta
· 8 years ago
f302aac
The level controller complexity tests have lately been
by peah
· 8 years ago
abda7f8
RTPPayloadRegistry: Stop using the rate to keep track of receive codecs
by kwiberg
· 8 years ago
54e6b81
This approach passes packetization mode to the encoder as part of
by hta
· 8 years ago
d270945
Fix spelling mistake in rtp_rtcp.h.
by brandtr
· 8 years ago
cf255e4
Split targets mixing .c and .cc sources.
by kjellander
· 8 years ago
d5f9b9b
AGC: Route clipping parameter from webrtc::Config to AGC
by henrik.lundin
· 8 years ago
1ea0829
Revert "Revert of Use different restrictions of acked bitrate lag depending on operating point. (patchset #3 id:40001 of https://codereview.webrtc.org/2542083003/ )"
by stefan
· 8 years ago
8de8a49
Revert of APM: Change 3 UMA metrics to fewer but linearly distributed buckets (patchset #2 id:20001 of https://codereview.webrtc.org/2547593002/ )
by henrik.lundin
· 8 years ago
23a5296
APM: Change 3 UMA metrics to fewer but linearly distributed buckets
by henrik.lundin
· 8 years ago
73981fc
RtpPacketizer::NextPacket fills RtpPacket instead of just payload.
by danilchap
· 8 years ago
d7ef04b
Move functionality out from AudioFrame and into AudioFrameOperations.
by aleloi
· 8 years ago
d799351
Revert of Use different restrictions of acked bitrate lag depending on operating point. (patchset #3 id:40001 of https://codereview.webrtc.org/2542083003/ )
by deadbeef
· 8 years ago
643ce24
Fix exponential probing in ProbeController.
by sergeyu
· 8 years ago
e1d9bcd
Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
by asapersson
· 8 years ago
0ec4ee5
Use different restrictions of acked bitrate lag depending on operating point.
by stefan
· 8 years ago
4a86445
Wire up rtcp xr target bitrate on receive side.
by sprang
· 8 years ago
11cda43
fix coding and documentary ambiguity in AimdRateControl::TimeToReduceFurther.
by howtofly
· 8 years ago
b0e0142
VP8DecoderImpl: Fix uninitialized memory crash
by magjed
· 8 years ago
0d9fb71
Deprecated SetAudioPacketSize from RTPSender and removed calls to it.
by ossu
· 8 years ago
88a348f
Cleanup RtpHeaderExtensionMap removing use of two legacy functions
by danilchap
· 8 years ago
d26e05a
Remove bitrate cap for AdaptiveVideoSource and increase other caps to 25 Mbps.
by terelius
· 8 years ago
63c39fa
Wire up BitrateAllocation to be sent as RTCP TargetBitrate
by sprang
· 8 years ago
42ad2d4
Wire up RTCP XR target bitrate in rtp/rtcp module
by sprang
· 8 years ago
291986e
Increase test timeout to combat flakiness.
by kthelgason
· 8 years ago
cbd3eb5
Remove OnLocalSsrcChanged
by mflodman
· 8 years ago
2b52cd4
Change assert to RTC_DCHECK in bwe_test_logging.cc
by terelius
· 8 years ago
2bc91bf
Reland "Update rtt on audio only calls".
by michaelt
· 8 years ago
fef80b7
Delete all of the video_processing module but the denoiser code.
by nisse
· 8 years ago
82a8776
Templatize percentile_filter.h and move it to base/analytics.
by terelius
· 8 years ago
7937121
Reland "Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate."
by minyue
· 8 years ago
45a0bd3
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac
by ehmaldonado
· 8 years ago
562c020
Greatly reduce number of level controller tests.
by phoglund
· 8 years ago
c01dfe6
Relanding "Pass time constant to bwe smoothing filter."
by minyue
· 8 years ago
61e42da
Move ownership of PacketRouter from CongestionController to Call.
by nisse
· 8 years ago
3bd27b7
Reduce ProbeController::kDefaultMaxProbingBitrateBps to 10 mbps.
by philipel
· 8 years ago
b6dfe9d
Disabled all ScreenCapturerIntegrationTests on Windows
by ossu
· 8 years ago
fc3c29b
Update video histograms that do not have a minimum lifetime limit before being recorded.
by asapersson
· 8 years ago
c138a32
Fix memory leak in video_coding::PacketBuffer::InsertPacket.
by philipel
· 8 years ago
c3f6629
Calculate JitterBufferDelayInMs in the new jitter buffer.
by philipel
· 8 years ago
3d4d02b
Revert of Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (patchset #5 id:240001 of https://codereview.webrtc.org/2411613002/ )
by minyue
· 8 years ago
e334041
Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate.
by minyue
· 8 years ago
dc4b00b
Use RotateDesktopFrame in DirectX capturer
by zijiehe
· 8 years ago
69b2ab4
Enable ScreenCapturerIntegrationTests
by zijiehe
· 8 years ago
927b87d
Adding packet overhead to audio network adaptor.
by minyue
· 8 years ago
a6b65a8
Add a new UMA metric in APM to track incoming capture-side audio level
by henrik.lundin
· 8 years ago
d5a1889
Added sanity check to VCMDecodingState::UsingFlexibleMode to prevent OOB error.
by philipel
· 8 years ago
ccd06c1
Fix spelling mistake in RTP module declaration.
by brandtr
· 8 years ago
c1a5e1b
Replace some asserts with DCHECKs
by kwiberg
· 8 years ago
cf7d22e
Refactor RMSLevel and give it new functionality
by henrik.lundin
· 8 years ago
7b42932
Add overhead to transport feedback observer.
by michaelt
· 8 years ago
5f55ae6
Move usage of QualityScaler to ViEEncoder.
by kthelgason
· 8 years ago
13d9326
RTC_[D]CHECK_op: Remove superfluous casts
by kwiberg
· 8 years ago
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