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53c1d3c561c92948cada27409bab1c5b9747ae8a
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36208c2
Properly account for retransmitted packets when not using the pacer.
by stefan@webrtc.org
· 10 years ago
fbd6f47
Fixes RTX related bugs.
by stefan@webrtc.org
· 10 years ago
8abfc7f
Disabling SendsSetSimulcastSsrcs.
by pbos@webrtc.org
· 10 years ago
14e5fbb
Revert "Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets"
by henrik.lundin@webrtc.org
· 10 years ago
256de0f
Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets
by henrik.lundin@webrtc.org
· 10 years ago
45ae9e4
Disable flaky CanSwitchToUseAllSsrcs.
by pbos@webrtc.org
· 10 years ago
1d61e3a
Simplify pacer interface.
by pbos@webrtc.org
· 10 years ago
e2a7a77
Remove internal codecs from VideoSendStream.
by pbos@webrtc.org
· 10 years ago
76cd2f7
Fix a deadlock in ViEEncoder::DeliverFrame.
by wuchengli@chromium.org
· 10 years ago
dd6cf04
Adds a method to WindowCapturer to bring a window to the front.
by jiayl@webrtc.org
· 10 years ago
c4c3021
Adding thread annotations to NetEq4
by henrik.lundin@webrtc.org
· 10 years ago
fee2074
Add #include <cstdlib> for std::abs.
by pbos@webrtc.org
· 10 years ago
df08c5d
Resolves TSan v2 warnings in voe_auto_test.
by henrika@webrtc.org
· 10 years ago
0c0c604
Re-comitting r5711: "Fixing a flaky test in video_engine_tests"
by henrik.lundin@webrtc.org
· 10 years ago
a50f107
Revert 5711 "Fixing a flaky test in video_engine_tests"
by turaj@webrtc.org
· 10 years ago
323af23
Fixing a flaky test in video_engine_tests
by henrik.lundin@webrtc.org
· 10 years ago
608b331
Small refactor on send_side_bandwidth_estimation.
by andresp@webrtc.org
· 10 years ago
c6f6696
Refactor rampup tests:
by andresp@webrtc.org
· 10 years ago
d073362
Tool to establish a loopback call via apprtc turn server.
by andresp@webrtc.org
· 10 years ago
dc7e89c
References to includes in third_party should be relative, not absolute.
by sprang@webrtc.org
· 10 years ago
8b04780
Add support for YUV4MPEG file reading to tools files. (Minor fix).
by mcasas@webrtc.org
· 10 years ago
280ab2a
Add support for YUV4MPEG file reading to tools files.
by mcasas@webrtc.org
· 10 years ago
fcc76aa
Fix a bug where network freeze during CNG causes delay
by henrik.lundin@webrtc.org
· 10 years ago
56bf2cd
Remove legacy weirdness in Merge::Downsample
by henrik.lundin@webrtc.org
· 10 years ago
96616cb
Stopping network threads before tearing down test
by henrik.lundin@webrtc.org
· 11 years ago
66cff19
Race condition in RTPSender
by sprang@webrtc.org
· 11 years ago
b67e9b7
Add max delay to trace based filters and enhances drop tail queues with delay statistics.
by stefan@webrtc.org
· 11 years ago
9376c69
Re-landing "Routing SuspendChange to VideoSendStream::Stats"
by henrik.lundin@webrtc.org
· 11 years ago
3f83f9c
Implement minimum transmit bitrate.
by pbos@webrtc.org
· 11 years ago
691c5b2
Enable all RampUpTest.UpDownUp* tests
by henrik.lundin@webrtc.org
· 11 years ago
2a25b6c
Replace labs with std::abs.
by pbos@webrtc.org
· 11 years ago
2b43f8b
Disable all protobuf dependent targets when enable_protobuf=0.
by andrew@webrtc.org
· 11 years ago
18e5838
Enable VS2013 for Windows compilation by default.
by kjellander@webrtc.org
· 11 years ago
ee86b90
Remove platform-specific code from new-API tests.
by pbos@webrtc.org
· 11 years ago
758d0b8
Implement a test for an old corner-case in NetEq
by henrik.lundin@webrtc.org
· 11 years ago
174152b
Developing NetEqImpl unit tests
by henrik.lundin@webrtc.org
· 11 years ago
e462911
Disable TestOpusNewACM on Android.
by andrew@webrtc.org
· 11 years ago
9759355
Revert "Routing SuspendChange to VideoSendStream::Stats"
by henrik.lundin@webrtc.org
· 11 years ago
dce00af
Reorder includes in audio_processing_impl_unittest.
by andrew@webrtc.org
· 11 years ago
3f7753c
Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now.
by braveyao@webrtc.org
· 11 years ago
2ae3c62
Routing SuspendChange to VideoSendStream::Stats
by henrik.lundin@webrtc.org
· 11 years ago
f83bc05
Classes and tests for audio an classifier. The class can be used to classify whether a frame of audio contains speech or music. The classifier uses the music/speech classifier in Opus.
by jan.skoglund@webrtc.org
· 11 years ago
b5b49c0
Add tests and modify tools for new float deinterleaved interface.
by andrew@webrtc.org
· 11 years ago
6a5ccc5
Add a float interface to PushSincResampler.
by andrew@webrtc.org
· 11 years ago
23531a3
iOS video_render: omit no-op setNeedsDisplay
by fischman@webrtc.org
· 11 years ago
b0de73f
AppRTCDemo(iOS): video support; part 1 of 2: webrtc/.
by fischman@webrtc.org
· 11 years ago
5b67882
Adding a link to issue
by henrik.lundin@webrtc.org
· 11 years ago
da666ce
Roll chromium_revision 249215:255773
by kjellander@webrtc.org
· 11 years ago
4e9a04b
Fix build breakage introduce with r5665.
by stefan@webrtc.org
· 11 years ago
dc07590
Add option to bwe_rtp_to_text to output arrival times only in nanoseconds.
by stefan@webrtc.org
· 11 years ago
bbbe9b8
Avoid crash in ViEEncoder::DeRegisterExternalEncoder().
by fischman@webrtc.org
· 11 years ago
9201f0d
Moves WEBRTC_POSIX define from header file to gyp-settings.
by henrike@webrtc.org
· 11 years ago
bcf0a10
Remove std:: prefixes from C functions in webrtc/.
by pbos@webrtc.org
· 11 years ago
52db76a
adding FEC support to WebRTC Opus wrapper and tests.
by minyue@webrtc.org
· 11 years ago
1f2820b
This CL is to add Opus complexity knob and to test it.
by minyue@webrtc.org
· 11 years ago
9a82322
Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.
by wu@webrtc.org
· 11 years ago
7a56079
Fixes a bug in the simulation framework where the time offset is accumulating as the packet trace is repeated, causing increasingly large gaps with no packets being transmitted.
by stefan@webrtc.org
· 11 years ago
70e2ce9
NetEq4: Changing the behavior of playout_timestamp_ update
by henrik.lundin@webrtc.org
· 11 years ago
379c349
Potential deadlock in VideoSendStreamTest::ProducesStats
by sprang@webrtc.org
· 11 years ago
0435a83
Use DISABLE_ instead of commenting out tests
by henrik.lundin@webrtc.org
· 11 years ago
c766098
Adding a new ramp-up-down-up test
by henrik.lundin@webrtc.org
· 11 years ago
bc55f91
Remove upper check for number of cores in VCM, I didn't find any good reasons for checking this.
by mflodman@webrtc.org
· 11 years ago
80eeab5
Switch to correct interpretation of int and float input data in audio_processing_unittest
by bjornv@webrtc.org
· 11 years ago
5350e31
Add a deinterleaved float interface to AudioProcessing.
by andrew@webrtc.org
· 11 years ago
78f1f4a
VideoCaptureAndroid: don't deliver frames after stopCapture().
by fischman@webrtc.org
· 11 years ago
dbe9516
Including algorithm header to avoid VS2013 breakage
by henrik.lundin@webrtc.org
· 11 years ago
6a9c344
Fix compilation errors under clang 3.5.
by pbos@webrtc.org
· 11 years ago
f9747a8
Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed.
by fischman@webrtc.org
· 11 years ago
55a2a27
Adds APIs for reporting pacer queuing delay.
by jiayl@webrtc.org
· 11 years ago
a1a6001
Remove ProcessingComponent's dependence on AudioProcessingImpl.
by andrew@webrtc.org
· 11 years ago
84641bf
Call PrintWindow for the first time of capturing to capture the window frames correctly.
by jiayl@webrtc.org
· 11 years ago
e9d42e6
Clean up CPU detection defines in SincResampler a little.
by andrew@webrtc.org
· 11 years ago
935694d
Invalidate the whole screen when the frame size is changed.
by jiayl@webrtc.org
· 11 years ago
b7306ea
Use scoped_ptr<T[]> in SincResampler to avoid .get()[] weirdness.
by andrew@webrtc.org
· 11 years ago
9900e37
Add SetConfig method to FakeNetworkPipe and to DirectTransport
by henrik.lundin@webrtc.org
· 11 years ago
fa35a17
Add experimental noise suppression flag to audioproc test
by aluebs@webrtc.org
· 11 years ago
1f9bc4f
Missing include in experiments.h
by sprang@webrtc.org
· 11 years ago
dea824b
Split the implementation of VP8Encoder|Decoder::Create into a seperated file
by wu@webrtc.org
· 11 years ago
9aef225
Fix to get total number of sent and received rtcp packets.
by asapersson@webrtc.org
· 11 years ago
7efba48
AviRecorder is missing a critical section.
by braveyao@webrtc.org
· 11 years ago
b1a7102
Disable libjingle_peerconnection_java_unittest
by kjellander@webrtc.org
· 11 years ago
5cf6d75
Removed unused mock methods in audio_processing
by bjornv@webrtc.org
· 11 years ago
663ba07
Add RTCP packet class. Adds packet types: sr, rr, bye, fir.
by asapersson@webrtc.org
· 11 years ago
af83c70
MIPS optimizations for AEC audio processing module
by andrew@webrtc.org
· 11 years ago
fd1b455
Updated WebRTC version to 3.50 TBR= wu@webrtc.org
by elham@webrtc.org
· 11 years ago
1d2f5dc
Add an AlignedFreeDeleter and remove scoped_ptr_malloc.
by andrew@webrtc.org
· 11 years ago
3528ad9
Minor improvement in RoundToInt16 implementation.
by turaj@webrtc.org
· 11 years ago
0b95677
Modified overuse detection thresholds.
by asapersson@webrtc.org
· 11 years ago
64007c6
Removing a variable that was never read
by henrik.lundin@webrtc.org
· 11 years ago
46b817d
ifdef the alsa code based on macro USE_X11
by fbarchard@google.com
· 11 years ago
6ad768a
Fix the break caused by r5579.
by turaj@webrtc.org
· 11 years ago
bc43ec2
Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable.
by turaj@webrtc.org
· 11 years ago
ab23a2d
Make WindowCapturerLinux handling window resize events.
by jiayl@webrtc.org
· 11 years ago
1af2c14
Added architecture for compiling under chrome NaCl.
by andresp@webrtc.org
· 11 years ago
891f05e
This CL separate all ACM tests with new and old implementation of ACM and NetEq. The reason is to debug an issue with failure on Android try bots. We need to see if the error only occurs with the new ACM/NetEq, or if it is a flakiness that affects both.
by tina.legrand@webrtc.org
· 11 years ago
4a15560
Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).
by asapersson@webrtc.org
· 11 years ago
4845ee0
Removes VoERTP_RTCP::InsertExtraRTPPacket.
by henrika@webrtc.org
· 11 years ago
d4ce7ff
Fix DesktopAndCursorComposer not to crash
by sergeyu@chromium.org
· 11 years ago
ad065d0
Move the volume quantization workaround from VoE to AGC.
by andrew@webrtc.org
· 11 years ago
f6c4fc3
Remove obsolete voe_unit_test.
by solenberg@webrtc.org
· 11 years ago
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