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webrtc
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src
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9b7b75324f2deb9a17c1d25292123ae1f4021d9a
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webrtc
/
video
/
rtp_streams_synchronizer.cc
4cd2790
Move RTP for synchroninzation and rename classes, files and variables.
by mflodman
· 9 years ago
[Renamed (65%) from webrtc/video/vie_sync_module.cc]
d28db7f
Delete all use of tick_util.h.
by Niels Möller
· 9 years ago
0b25072
Use vcm::VideoReceiver on the receive side.
by Peter Boström
· 9 years ago
74f6e9e
Replace NULL with nullptr in webrtc/video.
by Peter Boström
· 9 years ago
f8cdd18
Add histogram stats for AV sync stream offset: "WebRTC.Video.AVSyncOffsetInMs"
by asapersson
· 9 years ago
a26ac92
Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ )
by pbos
· 9 years ago
da33a8a
Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
by torbjorng
· 9 years ago
f14c47a
Remove ignored return code from modules.
by Peter Boström
· 9 years ago
1794b26
Extract ViESyncModule outside ViEChannel.
by Peter Boström
· 9 years ago
97888bd
Swap use of CriticalSectionWrapper for rtc::CriticalSection in webrtc/video.
by Tommi
· 9 years ago
7623ce4
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
[Renamed (97%) from webrtc/video_engine/vie_sync_module.cc]
8237abf
Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
by kjellander
· 9 years ago
[Renamed (97%) from webrtc/video/vie_sync_module.cc]
03ef053
Merge webrtc/video_engine/ into webrtc/video/
by Peter Boström
· 9 years ago
[Renamed (97%) from webrtc/video_engine/vie_sync_module.cc]
2557b86
modules/video_coding refactorings
by Henrik Kjellander
· 9 years ago
ff761fb
modules: more interface -> include renames
by Henrik Kjellander
· 9 years ago
74f0f35
Delete a chain of methods in ViE, VoE and ACM
by henrik.lundin
· 9 years ago
98f5351
system_wrappers: rename interface -> include
by Henrik Kjellander
· 9 years ago
415d2cd
Use webrtc/base/logging.h for video.
by Peter Boström
· 9 years ago
e4f9650
Remove system_wrappers/interface/trace_event.h
by tommi
· 9 years ago
8fc7fa7
Base A/V synchronization on sync_labels.
by pbos
· 10 years ago
36a1438
Remove ViEFrameProviderBase.
by Peter Boström
· 10 years ago
0b1534c
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
by pkasting@chromium.org
· 10 years ago
4e2806d
Remove WEBRTC_TRACE uses in video_engine/
by pbos@webrtc.org
· 11 years ago
66773a0
Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.
by wu@webrtc.org
· 11 years ago
cd70119
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
by wu@webrtc.org
· 11 years ago
48df381
Fix for making sure that the packet in order checks are done prior to updating the last received packet state.
by stefan@webrtc.org
· 11 years ago
822fbd8
Update talk to 50918584.
by wu@webrtc.org
· 12 years ago
aa4d96a
Revert r4301
by tnakamura@webrtc.org
· 12 years ago
66b2e5c
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
by stefan@webrtc.org
· 12 years ago
7262ad1
Fix AV sync issue
by hclam@chromium.org
· 12 years ago
9b23ecb
Log current and target AV delay in ViESyncModule
by hclam@chromium.org
· 12 years ago
e46c8d3
API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay.
by turaj@webrtc.org
· 12 years ago
f5d4cb1
Include files from webrtc/.. paths in video_engine/
by pbos@webrtc.org
· 12 years ago
1de0135
Adding playout buffer status to the voe video sync
by pwestin@webrtc.org
· 12 years ago
806dc3b
More trace events
by hclam@chromium.org
· 12 years ago
b238d12
WebRtc_Word32 -> int32_t in video_engine/
by pbos@webrtc.org
· 12 years ago
79b0289
Adds event traces and counters for WebRTC receive side.
by edjee@google.com
· 12 years ago
efe4edb
Enabling bufffering mode with no sync module or VoE
by mikhal@webrtc.org
· 12 years ago
ef9f76a
Adding a receive side API for buffering mode.
by mikhal@webrtc.org
· 12 years ago
8d18526
Fixes an incorrect if statement in vie_sync_module.cc.
by stefan@webrtc.org
· 12 years ago
14b43be
Move src/ -> webrtc/
by andrew@webrtc.org
· 12 years ago
[Renamed from src/video_engine/vie_sync_module.cc]
7c3523c
Change audio/video sync to be based on mapping RTP timestamps to NTP.
by stefan@webrtc.org
· 13 years ago
6f8db36
Reorganize voice_engine/.
by andrew@webrtc.org
· 13 years ago
ab2610f
Removed the last lint warnings in video_engine.
by mflodman@webrtc.org
· 13 years ago
5f28498
First step in refactoring audio/video synchronization. Adds unittests.
by stefan@webrtc.org
· 13 years ago
139c467
Fixed a/v sync issue.
by mflodman@webrtc.org
· 13 years ago
2853dde
Refactor the internal API to the rtp/rtcp module.
by pwestin@webrtc.org
· 13 years ago
3c383ab
Revert 2211 - Refactor the internal API to the rtp/rtcp module.
by turaj@webrtc.org
· 13 years ago
0774838
Refactor the internal API to the rtp/rtcp module.
by pwestin@webrtc.org
· 13 years ago
39e9659
Correct wrong usage of WebRtc_Word8 in video enigne
by leozwang@webrtc.org
· 13 years ago
d32c447
Changed constructor used for CriticalSectionScoped in ViE.
by mflodman@webrtc.org
· 13 years ago
d2ee5d9
Changed sync bug introduced in refactoring.
by mflodman@webrtc.org
· 13 years ago
511f82e
Refactored ViESyncModule.
by mflodman@webrtc.org
· 13 years ago
94ea32e
Move video_engine/source* to video_engine/. No code changes except paths in gyp-files.
by mflodman@webrtc.org
· 13 years ago
[Renamed from src/video_engine/main/source/vie_sync_module.cc]
470e71d
by niklase@google.com
· 14 years ago