Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
fp2-dev
/
platform
/
external
/
chromium_org
/
third_party
/
webrtc
/
b7e5b2741f9cbd68e7be54e3445d28d266477c92
/
video_engine
/
vie_encoder.cc
c1696da
Small refactor on ViE to remove redudant conditions and long ifdefs.
by andresp@webrtc.org
· 10 years ago
6111d79
Print an info log instead of return an error if an external encoder is de-registered, but no corresponding internal encoder can be registered automatically.
by stefan@webrtc.org
· 10 years ago
fedbe8b
Thread annotations for vie_encoder.cc/.h
by stefan@webrtc.org
· 10 years ago
c9995bc
Introduces PacedVideoSender to test framework and moves the Pacer to use Clock.
by stefan@webrtc.org
· 10 years ago
4ee6348
Add tests of texture frames in video_send_stream_test.
by wuchengli@chromium.org
· 10 years ago
1bdf186
Add support of texture frames for video capturer.
by wuchengli@chromium.org
· 10 years ago
5424828
Revert "Add support of texture frames for video capturer."
by wuchengli@chromium.org
· 10 years ago
a3b8c85
Add support of texture frames for video capturer.
by wuchengli@chromium.org
· 10 years ago
4d61a36
Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels.
by stefan@webrtc.org
· 10 years ago
ba47616
Replace scoped_array<T> with scoped_ptr<T[]>.
by andrew@webrtc.org
· 10 years ago
9d10769
Propagate capture ntp timestamp from rtp to renderer.
by wu@webrtc.org
· 10 years ago
8edccce
Cleaned up logging in video_coding.
by stefan@webrtc.org
· 10 years ago
022615b
Log Fixit for parts of video_engine folder.
by mflodman@webrtc.org
· 10 years ago
3b6c0e5
Refactor in BitrateController module.
by andresp@webrtc.org
· 10 years ago
1d61e3a
Simplify pacer interface.
by pbos@webrtc.org
· 10 years ago
76cd2f7
Fix a deadlock in ViEEncoder::DeliverFrame.
by wuchengli@chromium.org
· 10 years ago
3f83f9c
Implement minimum transmit bitrate.
by pbos@webrtc.org
· 10 years ago
bbbe9b8
Avoid crash in ViEEncoder::DeRegisterExternalEncoder().
by fischman@webrtc.org
· 10 years ago
c766098
Adding a new ramp-up-down-up test
by henrik.lundin@webrtc.org
· 10 years ago
55a2a27
Adds APIs for reporting pacer queuing delay.
by jiayl@webrtc.org
· 10 years ago
5a6e691
Adding a critical section missing in r5543.
by stefan@webrtc.org
· 10 years ago
5fcd7df
Fixes a race when writing to send_padding_.
by stefan@webrtc.org
· 10 years ago
8e1a876
Set pacing bitrates in SetEncoder.
by pbos@webrtc.org
· 10 years ago
1fc02eb
Removing DropDeltaAfterKey functionality which is unused.
by andresp@webrtc.org
· 11 years ago
04d6593
Ensure that no packet stays in the pacer queue for longer than 2 seconds.
by stefan@webrtc.org
· 11 years ago
2e98d45
Implement and test EncodedImageCallback in new ViE API.
by sprang@webrtc.org
· 11 years ago
82b883c
Renaming ViEEncoderObserver::VideoSuspended
by henrik.lundin@webrtc.org
· 11 years ago
7673871
Protect reads of ViEEncoder::video_suspended_.
by pbos@webrtc.org
· 11 years ago
b9f1eb8
Connect pacer/padding to SuspendBelowMinBitrate
by henrik.lundin@webrtc.org
· 11 years ago
4590177
Rename AutoMute to SuspendBelowMinBitrate
by henrik.lundin@webrtc.org
· 11 years ago
b748c9d
Fix for RTX in combination with pacing.
by stefan@webrtc.org
· 11 years ago
24e2089
Separate Call API/build files from video_engine/.
by pbos@webrtc.org
· 11 years ago
4ce7590
Removing the threshold from the auto-mute APIs
by henrik.lundin@webrtc.org
· 11 years ago
221798a
Upgrade scoped_ptr to Chromium's latest version.
by andrew@webrtc.org
· 11 years ago
7c46e95
AutoMute: Adding channel_id parameter to callback.
by henrik.lundin@webrtc.org
· 11 years ago
63301bd
Implement I420FrameCallbacks in Call.
by pbos@webrtc.org
· 11 years ago
5e74d96
Have padding decay to zero if no frames are being captured.
by stefan@webrtc.org
· 11 years ago
93cd397
Don't pad if only one stream is sent, except if auto muted.
by stefan@webrtc.org
· 11 years ago
39079d1
Piping AutoMuter interface through to ViE API
by henrik.lundin@webrtc.org
· 11 years ago
7dc1790
Improving padding rules and breaking out bw allocation to ViEEncoder.
by stefan@webrtc.org
· 11 years ago
7fc75bb
Update talk to 50918584.
by wu@webrtc.org
· 11 years ago
3f45c2e
Switch C++-style C headers with their C equivalents.
by pbos@webrtc.org
· 11 years ago
bf76ae2
Hooking up first simple CPU adaptation version.
by mflodman@webrtc.org
· 11 years ago
0ba496b
Revert r4301
by tnakamura@webrtc.org
· 11 years ago
9c0f14d
Cleanup WebRTC tracing
by hclam@chromium.org
· 11 years ago
a32d18f
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
by stefan@webrtc.org
· 11 years ago
0291c80
Removed ViE file API.
by mflodman@webrtc.org
· 11 years ago
0f6f7cb
Enqueue packet in pacer if sending fails
by hclam@chromium.org
· 11 years ago
69f7605
Wire up pacer-based padding.
by stefan@webrtc.org
· 11 years ago
695ff2a
Add support for padding in pacer.
by stefan@webrtc.org
· 11 years ago
453f9c0
Add functions to ViE API to enable/disable the absolute send time header extension.
by solenberg@webrtc.org
· 11 years ago
281cff8
Include files from webrtc/.. paths in video_engine/
by pbos@webrtc.org
· 11 years ago
5a22c40
Cleanup traces in WebRTC
by hclam@chromium.org
· 11 years ago
d474c13
Add more tracing for key frames.
by justinlin@chromium.org
· 11 years ago
ac6d919
Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation.
by andresp@webrtc.org
· 11 years ago
36bdba4
Adding trace and changing pacing constants
by pwestin@webrtc.org
· 11 years ago
3816c52
Fix the encoder pause logic. BUG=1691
by pwestin@webrtc.org
· 11 years ago
74472fe
More trace events
by hclam@chromium.org
· 11 years ago
73ebe67
Revert "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
67879bc
WebRtc_Word32 -> int32_t in video_engine/
by pbos@webrtc.org
· 11 years ago
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
dca71b2
Add interface to signal a network down event.
by stefan@webrtc.org
· 11 years ago
6313692
Follow-up fix for r3681.
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
c6242c9
Destroy VCM and VPM instead of delete.
by mflodman@webrtc.org
· 11 years ago
9d6fcb3
Adding a receive side API for buffering mode.
by mikhal@webrtc.org
· 11 years ago
0c66de6
Updates to send side streaming mode:
by mikhal@webrtc.org
· 11 years ago
71f3f68
Enable external encoders with internal picture source.
by stefan@webrtc.org
· 12 years ago
ac09423
Changed assert to log.
by mflodman@webrtc.org
· 12 years ago
c0539d9
Properly remove the bitrate observer when ViEEncoder is destructed.
by stefan@webrtc.org
· 12 years ago
1d50745
Remove ViE lint warnings that should have been caught at upload time.
by mflodman@webrtc.org
· 12 years ago
5e87b5f
Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016
by pwestin@webrtc.org
· 12 years ago
28f76e5
Fix uninitialzed memory and cleanup.
by pwestin@webrtc.org
· 12 years ago
b6d9cfc
Revert the revert in r2988 since that wasn't the issue.
by mflodman@webrtc.org
· 12 years ago
9f269d2
Reverse Merged r2884 & r2888 from trunk.
by vikasmarwaha@webrtc.org
· 12 years ago
3bbed74
Switching to I420VideoFrame
by mikhal@webrtc.org
· 12 years ago
b015cbe
Move src/ -> webrtc/
by andrew@webrtc.org
· 12 years ago