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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
3c3aef44deb4cca21ece8532e8dfae1b92c7989a
/
webrtc
/
video
/
BUILD.gn
3c3aef4
Revert of Reland "Move smoothing filter to common audio". (patchset #5 id:100001 of https://codereview.webrtc.org/2520003005/ )
by minyue
· 8 years ago
223641f
Reland "Move smoothing filter to common audio".
by minyue
· 8 years ago
f6acc2a
Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/
by magjed
· 8 years ago
509e4fe
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ )
by magjed
· 8 years ago
eacbaea
Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ )
by magjed
· 8 years ago
42043b9
Stop using hardcoded payload types for video codecs
by Magnus Jedvert
· 8 years ago
614d5b7
Move VideoEncoderSoftwareFallbackWrapper from webrtc/video_encoder.h to webrtc/media/engine/
by magjed
· 8 years ago
445fb8f
Use correct define in H264 end-to-end tests.
by brandtr
· 8 years ago
13ceeea
Revert of H.264 packetization mode 0 (try 2) (patchset #27 id:520001 of https://codereview.webrtc.org/2337453002/ )
by magjed
· 8 years ago
3bba101
Implement H.264 packetization mode 0.
by hta
· 8 years ago
bf6a45b
Moved transport_adapter.h/.cc from call/ to video/ dir to remove circular dependency
by charujain
· 8 years ago
e5ba44e
Implement framesDecoded stat in video receive ssrc stats.
by sakal
· 8 years ago
a3cac05
GN: move webrtc/video/ targets from webrtc/BUILD.gn into webrtc/video/BUILD.gn
by kjellander
· 8 years ago
e40a7ee
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
by kjellander
· 8 years ago
ae04e36
Cleanup unused dependency on video_capture_module.
by perkj
· 8 years ago
15d8357
Remove OnLocalSsrcChanged and rename EncoderStateFeedback.
by mflodman
· 8 years ago
cc91d28
Moved RtcEventLog files from call/ to logging/
by skvlad
· 8 years ago
89a3a1a
Moved Gn target rtc_event_log to one directory above.
by charujain
· 8 years ago
b62dbbe
GN: Change rtc_source_set targets --> rtc_static_library
by kjellander
· 8 years ago
e9cc686
GN Templates: Move common_inherited_config to the template.
by ehmaldonado
· 8 years ago
7a2ce0b
GN Templates: Move common_config to the template.
by ehmaldonado
· 8 years ago
38a2132
GN: Introduce templates.
by ehmaldonado
· 8 years ago
26091b1
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
by perkj
· 8 years ago
8eb37a3
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
by perkj
· 8 years ago
cc16836
- Add task queue to Call with the intent of replacing the use of one of the process threads.
by perkj
· 8 years ago
4cd2790
Move RTP for synchroninzation and rename classes, files and variables.
by mflodman
· 8 years ago
4e417b2
Reland of Switch to use SequencedTaskChecker instead of ThreadChecker where needed.
by perkj
· 8 years ago
efd902c
Revert of Switch to use SequencedTaskChecker instead of ThreadChecker where needed. (patchset #1 id:1 of https://codereview.webrtc.org/2149553002/ )
by perkj
· 8 years ago
ec7cef8
Switch to use SequencedTaskChecker instead of ThreadChecker where needed.
by perkj
· 8 years ago
37ad337
Remove EncodedFrameCallbackAdapter.
by sergeyu
· 8 years ago
0208322
GN: Add video_engine_tests
by Peter Boström
· 8 years ago
0e9d6d9
Add class which periodically computes statistics.
by asapersson
· 8 years ago
cfc8e3b
Removed all RTP dependencies from ViEChannel and renamed class.
by mflodman
· 8 years ago
35151f3
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
by asapersson
· 8 years ago
b99395a
Reland of Delete video_render module. (patchset #1 id:1 of https://codereview.webrtc.org/1923613003/ )
by nisse
· 8 years ago
fa66659
Rename ViEReceiver and move ownership to VideoReceiveStream.
by mflodman
· 8 years ago
0190367
Revert of Delete video_render module. (patchset #12 id:220001 of https://codereview.webrtc.org/1912143002/ )
by kjellander
· 8 years ago
97cfd1e
Delete video_render module.
by nisse
· 8 years ago
7ffeab5
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
by kjellander@webrtc.org
· 9 years ago
7324eb9
Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
by kjellander
· 9 years ago
99b345c
Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
by kjellander@webrtc.org
· 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
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
03ef053
Merge webrtc/video_engine/ into webrtc/video/
by Peter Boström
· 9 years ago
c4a1c37
Removed vie_defines.h
by mflodman
· 9 years ago
0c478b3
Rename ChannelGroup to CongestionController and move to webrtc/call/.
by mflodman
· 9 years ago
5c389d3
Split webrtc/video into webrtc/{audio,call,video}.
by Peter Boström
· 9 years ago
ebbf8a8
Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it.
by sprang
· 9 years ago
3641185
Includes webrtc/build/protoc.gypi instead of build/protoc.gypi
by Bjorn Terelius
· 9 years ago
b933667
Revert "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly."
by Bjorn Terelius
· 9 years ago
c159b04
Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly.
by Bjorn Terelius
· 9 years ago
4b91bd0
Move frame input (ViECapturer) to webrtc/video/.
by Peter Boström
· 9 years ago
2ee2439
Merge video_engine_core into webrtc target.
by Peter Boström
· 9 years ago
57e5fd2
PRESUBMIT: Improve PyLint check and add GN format check.
by Henrik Kjellander
· 9 years ago
4d71ede
Add HW fallback option to software encoding.
by Peter Boström
· 9 years ago
7252a2b
Add HW fallback option to software decoding.
by Peter Boström
· 9 years ago
23fba1f
Add AudioReceiveStream to Call API.
by Fredrik Solenberg
· 9 years ago
ac2d27d
Fix style violations in common_types.h and config.h
by kwiberg@webrtc.org
· 10 years ago
f21ea91
GN: Add common configs to all targets.
by kjellander@webrtc.org
· 10 years ago
1b9a188
GN: Fix webrtc/video/BUILD.gn for Chromium build.
by kjellander@webrtc.org
· 10 years ago
788f058
GN: Implement video_engine, video_capture and video_render.
by kjellander@webrtc.org
· 10 years ago
1227ab8
GN: Add BUILD.gn files + kjellander to OWNERS
by kjellander@webrtc.org
· 10 years ago