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fp2-dev
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platform
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chromium_org
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webrtc
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65deb268159caec998295346b75877b97f7d2551
65deb26
With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps.
by stefan@webrtc.org
· 11 years ago
a97bf1c
WebRtc_Word32 -> int32_t in common_video.
by pbos@webrtc.org
· 11 years ago
f85a509
WebRtc_Word32 -> int32_t in utility/
by pbos@webrtc.org
· 11 years ago
283c29a
WebRtc_Word32 -> int32_t in media_file/
by pbos@webrtc.org
· 11 years ago
5d9a1bc
Fixing the flakiness of ThreadWakesTwice.
by hta@webrtc.org
· 11 years ago
91cab71
WebRtc_Word32 -> int32_t in test/
by pbos@webrtc.org
· 11 years ago
64a144f
WebRtc_Word32 -> int32_t in audio_device/
by pbos@webrtc.org
· 11 years ago
54f03bc
WebRtc_Word32 -> int32_t in voice_engine/
by pbos@webrtc.org
· 11 years ago
c0231af
WebRtc_Word32 -> int32_t in system_wrappers
by pbos@webrtc.org
· 11 years ago
208a648
Always set render delay in ViEChannel::RegisterExternalDecoder.
by pbos@webrtc.org
· 11 years ago
fbda0fc
WebRtc_Word32 => int32_t etc. in audio_coding/
by pbos@webrtc.org
· 11 years ago
8ec8955
Remove the old unused udp_transport
by pwestin@webrtc.org
· 11 years ago
7deebae
Reduce execution time of rate control test.
by marpan@webrtc.org
· 11 years ago
715275c
Fixed a bug in isac-fix's entropy coding function: out of bounds acces to array.
by kma@webrtc.org
· 11 years ago
48c4b75
WebRtc_Word32 => int32_t in video_coding/
by pbos@webrtc.org
· 11 years ago
b57da65
WebRtc_Word32 => int32_t for rtp_rtcp/
by pbos@webrtc.org
· 11 years ago
a9f28d5
Clean packets on the network when closing + made loopback test actually run again.
by mflodman@webrtc.org
· 11 years ago
63a1ebd
WebRtc_Word32 => int32_t remote_bitrate_estimator/
by pbos@webrtc.org
· 11 years ago
14e22dd
Fix a crash issue on WinXP where LoadLibrary(TEXT("Kernel32.dll")) may fail.
by wu@webrtc.org
· 11 years ago
a2576cf
In streaming mode it is preferable to fade to silence when sender stops sending, or long period of packet loss.
by turaj@webrtc.org
· 11 years ago
1d25eac
Resolves TSan v2 reports data races in voe_auto_test.
by henrika@webrtc.org
· 11 years ago
004f462
Add GYP target for WebRTC Video demo for Android.
by kjellander@webrtc.org
· 11 years ago
2ed1cd9
Permit arbitrary payload names for kVideoCodecGeneric.
by pbos@webrtc.org
· 11 years ago
ef91cbf
Remove WEBRTC_*_ENGINE_NETWORK_API use
by pwestin@webrtc.org
· 11 years ago
dded206
Adds event traces and counters for WebRTC receive side.
by edjee@google.com
· 11 years ago
c39749a
Fix no received audio in tests.
by pwestin@webrtc.org
· 11 years ago
84423e9
Disabling MixingTests due to race conditions.
by henrika@webrtc.org
· 11 years ago
c9f8871
Two more sleep calls converted to use SleepMs().
by hta@webrtc.org
· 11 years ago
45ce6a8
TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC
by henrika@webrtc.org
· 11 years ago
e493218
Remove UDP transport API from VoE
by pwestin@webrtc.org
· 11 years ago
9e8a401
Fixes memory leak in AudioLevel class reported by memory try bots.
by henrika@webrtc.org
· 11 years ago
c4efe71
Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer
by henrika@webrtc.org
· 11 years ago
065b64d
Remove UDP transport API from ViE
by pwestin@webrtc.org
· 11 years ago
dba5f45
Webrtc_Word32 => int32_t in video_coding/main/
by pbos@webrtc.org
· 11 years ago
7873061
Revert of r3747.
by henrike@webrtc.org
· 11 years ago
f535877
Two more sleep calls converted to use SleepMs().
by hta@webrtc.org
· 11 years ago
f46e1fa
Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer
by henrika@webrtc.org
· 11 years ago
b514117
Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher
by justinlin@chromium.org
· 11 years ago
a4b78ff
For VGA (640x360), currently 1 thread is used. This change increases it to 2 threads. For HD, 4 threads are enabled.
by fbarchard@google.com
· 11 years ago
f3ac3ba
Updated Webrtc version to 3.28
by elham@webrtc.org
· 11 years ago
55742e5
Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549
by marpan@webrtc.org
· 11 years ago
46144bb
Fixes build break in previous cl (https://code.google.com/p/webrtc/source/detail?r=3739) found by Android bots.
by henrike@webrtc.org
· 11 years ago
3b6f728
Removed CPU APIs from VoEHardware. Code is now only used by test applications.
by henrike@webrtc.org
· 11 years ago
fa2dd22
Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.
by solenberg@webrtc.org
· 11 years ago
2ffc8bf
Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..."
by wu@webrtc.org
· 11 years ago
365ca40
Removed CPU APIs from VoEHardware. Code is now only used by test applications.
by henrike@webrtc.org
· 11 years ago
0c0795e
Fix broken audio.
by leozwang@webrtc.org
· 11 years ago
a0bba27
G722-stereo has been missing when creating AudioDecoder.
by turaj@webrtc.org
· 11 years ago
88a7940
NetEq4 fails if the first packets inserted in are out-of-band DTMFs.
by turaj@webrtc.org
· 11 years ago
88f12ab
Fix flakiness in network up/down event tests when running under memcheck.
by stefan@webrtc.org
· 11 years ago
6fc5215
WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14.
by fischman@webrtc.org
· 11 years ago
dca71b2
Add interface to signal a network down event.
by stefan@webrtc.org
· 11 years ago
6e34ceb
Split condition_variable_win.cc into native (for Vista and newer OS versions) and generic implementation (based on events).
by henrike@webrtc.org
· 11 years ago
f386e2b
Remove VoE's default call in Trace::SetLevelFilter.
by andrew@webrtc.org
· 11 years ago
aa0fcd7
Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust.
by solenberg@webrtc.org
· 11 years ago
31b4448
Alphabetize include order in fake_voe_external_media.h.
by andrew@webrtc.org
· 11 years ago
a7a643e
Restart Android capture after orientation change. Also prevent an NPE on exit.
by fischman@webrtc.org
· 11 years ago
13f66d1
Add some VoE and AudioProcessing mocks.
by andrew@webrtc.org
· 11 years ago
8826e34
Refactor unittest trace printouts to a separate class.
by andrew@webrtc.org
· 11 years ago
0c1f10b
Enable the below APIs for iOS.
by sjlee@webrtc.org
· 11 years ago
9522792
Introduced pause and resume to the pacer
by pwestin@webrtc.org
· 11 years ago
f49577f
Updated WebRTC version to 3.27
by elham@webrtc.org
· 11 years ago
7fd368f
Bugfix for extended RTP/RTCP test
by pwestin@webrtc.org
· 11 years ago
c075e25
Move the VIE tests to use external transport instead of the built in udp transport
by pwestin@webrtc.org
· 11 years ago
4c27c03
Add trace printouts to all unit tests.
by andrew@webrtc.org
· 11 years ago
e1198e6
Add min and target bitrate to VideoCodec.
by marpan@webrtc.org
· 11 years ago
aa922de
Move the VoE tests to use external transport instead of the built in udp transport
by pwestin@webrtc.org
· 11 years ago
af6aa7b
Creating a copy of Udp transport under webrtc/test
by pwestin@webrtc.org
· 11 years ago
495c563
Cleanup nanosleep -> SleepMs Remove some leftover stuff
by hta@webrtc.org
· 11 years ago
4a48fd6
WebRtc_Word -> stdint in audio_coding/g711/
by pbos@webrtc.org
· 11 years ago
a2df078
Remove incorrect asserts.
by stefan@webrtc.org
· 11 years ago
e49f252
WebRtc_Word -> stdint in audio_coding/cng/
by pbos@webrtc.org
· 11 years ago
0f2782f
Fix -Wstring-conversion warnings.
by wu@webrtc.org
· 11 years ago
5815b7c
Thread safety issue fix in incoming_video_stream.cc. See issue 1465.
by vikasmarwaha@webrtc.org
· 11 years ago
757bf0f
Account for header inside I420Encoder::InitEncode.
by pbos@webrtc.org
· 11 years ago
6313692
Follow-up fix for r3681.
by stefan@webrtc.org
· 11 years ago
46618db
Fixed initialization of SPL in echo_control_mobile.
by kma@webrtc.org
· 11 years ago
9cd73ed
Android: rename android_build_type gyp variable.
by wjia@webrtc.org
· 11 years ago
ffe2ec6
Updated WebRTC version number to 3.26
by elham@webrtc.org
· 11 years ago
692c69d
Fix framerate sent to account for actually sent frames.
by stefan@webrtc.org
· 11 years ago
72e204a
Change VCM interface to take target bitrate in bits per second.
by stefan@webrtc.org
· 11 years ago
e3339fc
Generic video-codec support.
by pbos@webrtc.org
· 11 years ago
0b5c7f1
Revert the deletion of test_api_nack.cc in r3674.
by stefan@webrtc.org
· 11 years ago
87ef38e
Truncated delay quality to avoid negative return values
by bjornv@webrtc.org
· 11 years ago
946d240
Adding RTX on source
by mikhal@webrtc.org
· 11 years ago
cf9ab12
Adding Opus frame length test
by tina.legrand@webrtc.org
· 11 years ago
aecc559
Fixed a crash issue in NSX module.
by kma@webrtc.org
· 11 years ago
912b7f7
Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004
by pwestin@webrtc.org
· 11 years ago
40749c1
Added destructors for tests to control destruct order
by pwestin@webrtc.org
· 11 years ago
b793abe
Increasing size of nack list in buffered mode.
by mikhal@webrtc.org
· 11 years ago
2daec4c
Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work.
by pwestin@webrtc.org
· 11 years ago
2637d61
Refactor webrtc specific Event implementation to an EventFactory.
by stefan@webrtc.org
· 11 years ago
15a03fd
Remove DTMF detection. Talk team has been in the loop and there is no need for
by turaj@webrtc.org
· 11 years ago
df00856
Tool found: pass by value when pass by reference is better in system wrapper unit test.
by henrike@webrtc.org
· 11 years ago
0c46af6
Change intrinsic code in isac fix to let it pass chrome clang compiler.
by kma@webrtc.org
· 11 years ago
160b327
Fixes issue detected by tool.
by henrike@webrtc.org
· 11 years ago
f0f1dc2
Removed redundant VP8 width/height and made sure the generic width/height is set.
by stefan@webrtc.org
· 11 years ago
23a7047
Revert "Internal clean up: removing unused include line."
by dwkang@webrtc.org
· 11 years ago
177ec87
Internal clean up: removing unused include line.
by dwkang@webrtc.org
· 11 years ago
d451969
Fixed issue 1497 in iSAC fixed point.
by kma@webrtc.org
· 11 years ago
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