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src
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webrtc
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b8896d2584f63f26c00bb6c1f3be19ac2b6add62
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test
671d805
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1983583002/ )
by nisse
· 9 years ago
bcf3191
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 9 years ago
e5d98bc
Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #2 id:290001 of https://codereview.webrtc.org/1963413004/ )
by tommi
· 9 years ago
fe1461a
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
by nisse
· 9 years ago
97aa5c2
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
by kwiberg
· 9 years ago
8b348aa
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 9 years ago
b90dcbd
Corrected bug in checking the third number and added extra checks
by dkirovbroadsoft
· 9 years ago
b670f85
Replace scoped_ptr with unique_ptr everywhere
by kwiberg
· 9 years ago
44dc40f
Enable -Winconsistent-missing-override flag.
by nisse
· 9 years ago
5d9726a
Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
by nisse
· 9 years ago
dd62c69
Delete webrtc::VideoFrame methods buffer and stride.
by nisse
· 9 years ago
0d62df9
Reland of Delete video_render module. (patchset #1 id:1 of https://codereview.webrtc.org/1923613003/ )
by nisse
· 9 years ago
823f908
Switch voice transport to use Call and Stream instead of VoENetwork.
by mflodman
· 9 years ago
8083b4d
Revert of Delete video_render module. (patchset #12 id:220001 of https://codereview.webrtc.org/1912143002/ )
by kjellander
· 9 years ago
dabe8ad
Delete video_render module.
by nisse
· 9 years ago
2945e42
Fix producer_fec_fuzzer.
by Peter Boström
· 9 years ago
5fb5bd2
#include "webrtc/base/constructormagic.h" where appropriate
by kwiberg
· 9 years ago
413dd56
Fix test.gyp dependency.
by nisse
· 9 years ago
6ad2ded
RtpPacket class introduced.
by danilchap
· 9 years ago
eb1bef4
Add rotation to EncodedImage and make sure it is passed through encoders.
by Per
· 9 years ago
1e4f606
Added new VideoFrameBuffer methods Data[YUV]() etc.
by nisse
· 9 years ago
60cf421
Add isolate files for Android tests
by kjellander
· 9 years ago
a186ef6
Delete method webrtc::VideoFrame::native_handle.
by nisse
· 9 years ago
e8b98bd
Fixed rtcp rpsi parsing of invalid packets.
by danilchap
· 9 years ago
5422696
GN: Fix some build errors for iOS.
by kjellander
· 9 years ago
6bf669c
Remove webrtc::ScopedVector
by kwiberg
· 9 years ago
5ed7972
New method I420Buffer::Copy.
by nisse
· 9 years ago
a2761a5
Delete class webrtc::VideoRenderer and its header file.
by nisse
· 9 years ago
aec92fa
Replace RefCountImpl with rtc::RefCountedObject.
by Peter Boström
· 9 years ago
f8e52ce
This is an initial cleanup step, aiming to delete the
by nisse
· 9 years ago
5e79cc2
Added function for parsing single rtcp packet in tests.
by Danil Chapovalov
· 9 years ago
0b54e5a
Reland "Add check_deps rules in DEPS files."
by kjellander@webrtc.org
· 9 years ago
de82d23
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
by kjellander
· 9 years ago
181e867
Add check_deps rules in DEPS files.
by kjellander@webrtc.org
· 9 years ago
f2e3315
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
by solenberg
· 9 years ago
b9a65af
Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ )
by solenberg
· 9 years ago
9ef75db
- Clean up unused voice engine DTMF code following removal of VoEDtmf APIs.
by solenberg
· 9 years ago
39673bb
Remove the VoEDtmf interface.
by solenberg
· 9 years ago
4bd2aae
Remove webrtc/test/webrtc_test_common.gyp
by kjellander
· 9 years ago
b60d949
Replace scoped_ptr with unique_ptr in webrtc/common_video/
by kwiberg
· 9 years ago
c961c78
Cleanup of webrtc::VideoFrame.
by Niels Möller
· 9 years ago
12c9c87
Revert of Cleanup of webrtc::VideoFrame. (patchset #6 id:100001 of https://codereview.webrtc.org/1679323002/ )
by kjellander
· 9 years ago
c937f30
Cleanup of webrtc::VideoFrame.
by nisse
· 9 years ago
3637ad2
Simple RTCP receiver fuzzer.
by Peter Boström
· 9 years ago
c2dc777
Roll chromium_revision 1d144ca..fa5d546 (375480:376142)
by kjellander@webrtc.org
· 9 years ago
484e0cd
Replace scoped_ptr with unique_ptr in webrtc/voice_engine/
by kwiberg
· 9 years ago
c9c53c4
iOS: Add resource files for tests and implement OutputPath
by kjellander
· 9 years ago
a4777ed
Added A/V sync tests with drifting clocks.
by danilchap
· 9 years ago
872d5a3
Fuzzer tests for AudioDecoder's DecodeRedundant and IncomingPacket
by henrik.lundin
· 9 years ago
db68549
Fix GYP and GN references that are invalid in Chromium builds.
by kjellander
· 9 years ago
b3f6beb
Move gtest_prod_util.h out of webrtc/test tree.
by kjellander
· 9 years ago
abec1e4
Remove implicit downcast in producer_fec_fuzzer.cc.
by Peter Boström
· 9 years ago
3913313
Deprecate VideoDecoder::Reset() and remove calls.
by Peter Boström
· 9 years ago
3dad57b
Use separate rtp module lists for send and receive in PacketRouter.
by stefan
· 9 years ago
10d7c19
Fix implicit bool casts in producer_fec_fuzzer.cc.
by Peter Boström
· 9 years ago
65451ba
H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding.
by hbos
· 9 years ago
781be9e
Allow packets to be reordered in the fake network pipe.
by philipel
· 9 years ago
1e5b805
Remove mutable from rtc::CriticalSection members.
by pbos
· 9 years ago
441719f
Add stefan@webrtc.org to webrtc/test/OWNERS.
by Peter Boström
· 9 years ago
9b91023
Enable transport seq num extension on receive channel to suppress log warning.
by stefan
· 9 years ago
094d31f
Reenables several NetEq unittests on android.
by ivoc
· 9 years ago
4cb6ce4
Remove use of ConditionVariableWrapper and CriticalSectionWrapper from UdpSocket2Windows.
by Tommi
· 9 years ago
47d6a3d
Remove dependency on ConditionVariableWrapper and CriticalSectionWrapper in UdpSocketPosix.
by Tommi
· 9 years ago
bd2fe3c
Add tests for verifying transport feedback for audio and video.
by Stefan Holmer
· 9 years ago
bf4689d
Re-land: "Use an explicit identifier in Config"
by aluebs
· 9 years ago
5bfff64
Add ramp-up tests for transport sequence number with and w/o audio.
by Stefan Holmer
· 9 years ago
da2fece
Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
by tommi
· 9 years ago
770ba12
Use an explicit identifier in Config
by aluebs
· 9 years ago
26f9c18
Add CreateSend/ReceiveTransport() methods to CallTest.
by stefan
· 9 years ago
a2932fd
Reland "Add APK targets to build libjingle tests for Android."
by phoglund
· 9 years ago
8c0d4cb
Add audio streams to CallTest and a first A/V call test.
by Stefan Holmer
· 9 years ago
4685e62
Add implementation in metrics.h that uses atomic pointer.
by asapersson
· 9 years ago
98836c0
Remove DISABLED_ON_ macros.
by Peter Boström
· 9 years ago
840364d
Move fake-handle frame creation into test target.
by Peter Boström
· 9 years ago
4d003e9
[rtp_rtcp] Lint errors cleaned from rtp_utility
by danilchap
· 9 years ago
19cc283
Step 1 to prepare call_test.* for combined audio/video tests.
by stefan
· 9 years ago
8468abc
Expose codec implementation names in stats.
by Peter Boström
· 9 years ago
1626ef5
Add VP8 and H264 depacketizer fuzzers.
by Peter Boström
· 9 years ago
cfab413
Add DrFuzz support to webrtc fuzzers.
by pbos
· 9 years ago
1841c4e
Disable warnings failing when using Clang on Windows.
by kjellander
· 9 years ago
e8878af
Reland of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1528043002/ )
by Peter Boström
· 9 years ago
71ae2ea
Revert of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1524993002/ )
by tommi
· 9 years ago
89fb85d
Base webrtc fuzzers on a template.
by Peter Boström
· 9 years ago
aff0ce7
Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ )
by stefan
· 9 years ago
dbf6eb3
Add APK targets to build libjingle_peerconnection_unittests for Android.
by perkj
· 9 years ago
4325517
Add FEC producer fuzzing and a unittest for one of the issues found.
by Stefan Holmer
· 9 years ago
1cddafe
Replace EventWrapper in video/, test/ and call/.
by Peter Boström
· 9 years ago
0739714
Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/.
by terelius
· 9 years ago
e8f0735
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
by Peter Boström
· 9 years ago
a24951b
Nuke TickTime::UseFakeClock.
by Peter Boström
· 9 years ago
32949e5
Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
by kjellander
· 9 years ago
d838d8e
Create fuzzer tests for audio decoders
by Henrik Lundin
· 9 years ago
5be013d
Merge webrtc/video_engine/ into webrtc/video/
by Peter Boström
· 9 years ago
f95302f
Prepare the AudioSendStream to be hooked up to send-side BWE.
by Stefan Holmer
· 9 years ago
ae4b1f0
- Remove calls to VoEDtmf from WVoE/MC.
by Fredrik Solenberg
· 9 years ago
21ca0a4
Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream.
by solenberg
· 9 years ago
29a929d
Default to LS_INFO logging for release builds.
by Peter Boström
· 9 years ago
daf1aa4
Clean up PlatformThread.
by Peter Boström
· 9 years ago
87fc684
Fix bug in calculation of averge queue time in paced sender.
by Erik Språng
· 9 years ago
ed2a0fc
Add fuzzing of VP8 QP parsing.
by Peter Boström
· 9 years ago
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