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gerrit-public.fairphone.software
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fp2-dev
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platform
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external
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chromium_org
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third_party
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webrtc
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0eb8ec621648d9b274322b2a1b82e5a4d839ad0f
0eb8ec6
Delay Estimator: Minor refactoring and added a setter function.
by bjornv@webrtc.org
· 10 years ago
3aa1ac2
Rename RTPanalyze to rtp_analyze and remove old version
by henrik.lundin@webrtc.org
· 10 years ago
c55faad
Remove AudioDevice::{Microphone,Speaker}IsAvailable.
by andrew@webrtc.org
· 10 years ago
acb49e5
This is to get rid of a bug relating to the return of NULL in calling GetDecoder when there are DTMF packets.
by minyue@webrtc.org
· 10 years ago
71c9ebd
Add format specification to output file names
by henrik.lundin@webrtc.org
· 10 years ago
0725df6
Extends max sample rate from 96kHz to 192kHz on the input side.
by henrika@webrtc.org
· 10 years ago
a0acb1f
sink_filter_ds.cc: add lock to Receive procedure to Pause().
by braveyao@webrtc.org
· 10 years ago
5ae01bf
Make ACM2 the default in voe_cmd_test.
by andrew@webrtc.org
· 10 years ago
15f109e
Added simulations of capacity variations and wifi recordings.
by stefan@webrtc.org
· 10 years ago
53b062b
Roll chromium_revision 255773:260462
by kjellander@webrtc.org
· 10 years ago
7a8dee4
Fix ARM64 detection.
by andrew@webrtc.org
· 10 years ago
8f5ab19
VoiceEngine(iOS & Android): removed NOT_SUPPORTED
by fischman@webrtc.org
· 10 years ago
24532e0
Add tests for the RBE RemoveStream() API.
by solenberg@webrtc.org
· 10 years ago
7c3f468
VoE Channel: Don't register codecs when stopping receiver
by henrik.lundin@webrtc.org
· 10 years ago
9136607
Restore support for code coverage in WebRTC
by kjellander@webrtc.org
· 10 years ago
ad239fe
Add arm64 to typedefs.h
by andrew@webrtc.org
· 10 years ago
4c6d59a
Allow loopback tests to do TURN when served from webrtc.googlecode.com.
by andresp@webrtc.org
· 10 years ago
66f5371
Add svn mime-type properties to loopback_test files so they can be served from:
by andresp@webrtc.org
· 10 years ago
bae92ab
Don't disable experimental AGC in audioproc.
by andrew@webrtc.org
· 10 years ago
6c57efd
Re-submit: rev5775
by andresp@webrtc.org
· 10 years ago
0ab635c
Makes ScreenCapturerMac exclude the window specified in DesktopCapturer::SetExcludedWindow.
by jiayl@webrtc.org
· 10 years ago
37f807f
Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams.
by solenberg@webrtc.org
· 10 years ago
1e05528
Protect write of send_target_bitrate.
by andresp@webrtc.org
· 10 years ago
0027f0a
Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out.
by solenberg@webrtc.org
· 10 years ago
765ea72
Revert 5775 "Modify bitrate controller to update bitrate based o..."
by andrew@webrtc.org
· 10 years ago
f50914a
Removing VideoCodecDerived and moving methods inside VideoCodec.
by mallinath@webrtc.org
· 10 years ago
0ac0bca
Updated WebRTC version to 3.51
by elham@webrtc.org
· 10 years ago
a090cc7
iOS video_capture: move @private vars to impl.
by fischman@webrtc.org
· 10 years ago
09fb237
Fix race condition in RTPSEnder.
by sprang@webrtc.org
· 10 years ago
0b11715
Change sprintf format string from %zu to %i
by henrik.lundin@webrtc.org
· 10 years ago
539bbde
Modify bitrate controller to update bitrate based on process call and not
by andresp@webrtc.org
· 10 years ago
1f49208
Adding API for setting bandwidth estimation configurations.
by stefan@webrtc.org
· 10 years ago
892cd1f
iOS video_capture: start camera in the background.
by fischman@webrtc.org
· 10 years ago
1dd9fb5
iOS VideoEngine: move video_{capture,render} to ARC.
by fischman@webrtc.org
· 10 years ago
1a19092
Add configuration for ability to use the encode usage measure for triggering overuse/underuse.
by asapersson@webrtc.org
· 10 years ago
50ac4d6
Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API.
by solenberg@webrtc.org
· 10 years ago
85101db
Have changes to REMB trigger RTCP to be sent immediately.
by stefan@webrtc.org
· 10 years ago
bb1d4c7
DelayEstimator: Updates delay_quality and adds soft reset.
by bjornv@webrtc.org
· 10 years ago
b45cf1e
Run Opus with lower complexity setting on Android, iOS and/or ARM
by tina.legrand@webrtc.org
· 10 years ago
5ca38d1
Add targetBitrate to VideoCodec struct.
by pbos@webrtc.org
· 10 years ago
825acb1
Disabled some of the remote bitrate estimator baseline tests.
by stefan@webrtc.org
· 10 years ago
5f804f8
VoE changes to allow forwarding of packets from VoE to ViE BWE.
by solenberg@webrtc.org
· 10 years ago
61e9201
Add fir_filter to common_audio
by aluebs@webrtc.org
· 10 years ago
9b2b8ec
Add AIMD option to BWE API.
by stefan@webrtc.org
· 10 years ago
95b5fde
ACM2/NetEq4 did not decode Opus in stereo
by tina.legrand@webrtc.org
· 10 years ago
209791d
Refactor in BitrateController module.
by andresp@webrtc.org
· 10 years ago
61e72f0
Fixing crash in video_render_tests in release mode.
by henrikg@webrtc.org
· 10 years ago
23e07d8
Remove locks in SendSideBandwidthEstimation since those are only accessed while owning locks in
by andresp@webrtc.org
· 10 years ago
97d92ed
Adding FEC support in NetEq 4.
by minyue@webrtc.org
· 10 years ago
d327be4
Fix "unreachable code" warnings (MSVC warning 4702) in webrtc.
by pbos@webrtc.org
· 10 years ago
40fee00
Adding operator== and != methods for CodecInst and VideoCodec structures.
by mallinath@webrtc.org
· 10 years ago
bcee6b7
Use codec width/height as the encoded_image width/height.
by wu@webrtc.org
· 10 years ago
88fa18b
Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets
by henrik.lundin@webrtc.org
· 10 years ago
27bd3be
Add ability to configure cpu overuse options via an API.
by asapersson@webrtc.org
· 10 years ago
55f4fe8
Prevent playout delay wrap-around in VoiceEngine
by henrik.lundin@webrtc.org
· 10 years ago
4d9df07
Removes error printout in voe_cmd_test which was caused by attempts to transmit RTCP packets even if a transport object was not registered.
by henrika@webrtc.org
· 10 years ago
a183edc
Extend perf tests to perform rampup on single stream.
by andresp@webrtc.org
· 10 years ago
b903292
Adjust the captured window rect when the window is maximized.
by jiayl@webrtc.org
· 10 years ago
5e18933
Properly account for retransmitted packets when not using the pacer.
by stefan@webrtc.org
· 10 years ago
f9d5709
Fixes RTX related bugs.
by stefan@webrtc.org
· 10 years ago
292e7f6
Disabling SendsSetSimulcastSsrcs.
by pbos@webrtc.org
· 10 years ago
ce06c77
Revert "Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets"
by henrik.lundin@webrtc.org
· 10 years ago
e8db41f
Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets
by henrik.lundin@webrtc.org
· 10 years ago
16c3dcc
Disable flaky CanSwitchToUseAllSsrcs.
by pbos@webrtc.org
· 10 years ago
bef6e62
Simplify pacer interface.
by pbos@webrtc.org
· 10 years ago
f39df52
Remove internal codecs from VideoSendStream.
by pbos@webrtc.org
· 10 years ago
3a70e6e
Fix a deadlock in ViEEncoder::DeliverFrame.
by wuchengli@chromium.org
· 10 years ago
3c85e1e
Adds a method to WindowCapturer to bring a window to the front.
by jiayl@webrtc.org
· 10 years ago
d8e33dc
Adding thread annotations to NetEq4
by henrik.lundin@webrtc.org
· 10 years ago
e85416a
Add #include <cstdlib> for std::abs.
by pbos@webrtc.org
· 10 years ago
48bbc5a
Resolves TSan v2 warnings in voe_auto_test.
by henrika@webrtc.org
· 10 years ago
ebae8bb
Re-comitting r5711: "Fixing a flaky test in video_engine_tests"
by henrik.lundin@webrtc.org
· 10 years ago
8d3c410
Revert 5711 "Fixing a flaky test in video_engine_tests"
by turaj@webrtc.org
· 10 years ago
f9a6ab0
Fixing a flaky test in video_engine_tests
by henrik.lundin@webrtc.org
· 10 years ago
48191a6
Small refactor on send_side_bandwidth_estimation.
by andresp@webrtc.org
· 10 years ago
ca626eb
Refactor rampup tests:
by andresp@webrtc.org
· 10 years ago
3bf1f38
Tool to establish a loopback call via apprtc turn server.
by andresp@webrtc.org
· 10 years ago
81fd3e7
References to includes in third_party should be relative, not absolute.
by sprang@webrtc.org
· 10 years ago
340b16e
Add support for YUV4MPEG file reading to tools files. (Minor fix).
by mcasas@webrtc.org
· 10 years ago
bc5c7bc
Add support for YUV4MPEG file reading to tools files.
by mcasas@webrtc.org
· 10 years ago
b6fb76a
Fix a bug where network freeze during CNG causes delay
by henrik.lundin@webrtc.org
· 10 years ago
78c101a
Remove legacy weirdness in Merge::Downsample
by henrik.lundin@webrtc.org
· 10 years ago
3c00b1c
Stopping network threads before tearing down test
by henrik.lundin@webrtc.org
· 10 years ago
64e7141
Race condition in RTPSender
by sprang@webrtc.org
· 10 years ago
7f78ae5
Add max delay to trace based filters and enhances drop tail queues with delay statistics.
by stefan@webrtc.org
· 10 years ago
15cf717
Re-landing "Routing SuspendChange to VideoSendStream::Stats"
by henrik.lundin@webrtc.org
· 10 years ago
9420a1f
Implement minimum transmit bitrate.
by pbos@webrtc.org
· 10 years ago
41da329
Enable all RampUpTest.UpDownUp* tests
by henrik.lundin@webrtc.org
· 10 years ago
c53e587
Replace labs with std::abs.
by pbos@webrtc.org
· 10 years ago
34bf4d6
Disable all protobuf dependent targets when enable_protobuf=0.
by andrew@webrtc.org
· 10 years ago
1507b50
Enable VS2013 for Windows compilation by default.
by kjellander@webrtc.org
· 11 years ago
af634a2
Remove platform-specific code from new-API tests.
by pbos@webrtc.org
· 11 years ago
10c488f
Implement a test for an old corner-case in NetEq
by henrik.lundin@webrtc.org
· 11 years ago
fb9e586
Developing NetEqImpl unit tests
by henrik.lundin@webrtc.org
· 11 years ago
0a1a92b
Disable TestOpusNewACM on Android.
by andrew@webrtc.org
· 11 years ago
f951dfc
Revert "Routing SuspendChange to VideoSendStream::Stats"
by henrik.lundin@webrtc.org
· 11 years ago
17757d1
Reorder includes in audio_processing_impl_unittest.
by andrew@webrtc.org
· 11 years ago
5ddb6fe
Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now.
by braveyao@webrtc.org
· 11 years ago
697cd78
Routing SuspendChange to VideoSendStream::Stats
by henrik.lundin@webrtc.org
· 11 years ago
253e312
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
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