- Change how the transmission offset is calculated, to
incorporate the time since the frame was captured.
- Break out RtpRtcpClock and move it to system_wrappers.
- Use RtpRtcpClock to set the capture time in ms in the capture module.
We must use the same clock as in the RTP module to be able to measure
the time from capture until transmission.
- Enables the RTP header extension for packet transmission time offsets.
BUG=
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/666006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2489 4adac7df-926f-26a2-2b94-8c16560cd09d
Description:
On ChromeOS/ARM, compiler enforces to check return result of a function.
Currently, we don't check return result of fwrite, it causes building errors.
The following files need to patch. The patch should be similar, before I patch all
of them, I will start with 3 files, once we agree upon the solution, we will expand
it to all of them.
The question is should we do
1. if (error) { return -1;}
or
2. if (error) { /*ignor the error*/ }
I took "return -1" in this patch, but I'm OK with either. Please let me know your
thoughts and I will upload a new patch.
Review URL: https://webrtc-codereview.appspot.com/583010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2315 4adac7df-926f-26a2-2b94-8c16560cd09d
1) code cleanup and some updates to selection logic for qm_select.
2) added unit test for the QmResolution class.
3) update codec frame size and reset/update frame rate in media-opt:
4) removed unused motion vector metrics and some related code of content metrics processing.
Review URL: https://webrtc-codereview.appspot.com/405008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1791 4adac7df-926f-26a2-2b94-8c16560cd09d
also removed other related unused variables and code.
Reset frame rate estimate in mediaOpt when frame rate reduction is decided.
Update content_metrics with frame rate and qm_resolution with frame size.
Review URL: https://webrtc-codereview.appspot.com/395007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1718 4adac7df-926f-26a2-2b94-8c16560cd09d
When targeting 32-bit Linux, we need to pass -msse2 to gcc to compile
SSE2 intrinsics. However, -msse2 also gives gcc license to automatically
generate SSE2 instructions wherever it pleases. This will crash our code
on processors without SSE2 support.
This change breaks the files with SSE2 intrinsics into separate targets,
such that we can limit the scope of -msse2 to where it's needed.
We no longer need to employ the WEBRTC_USE_SSE2 define; the build system
decides when SSE2 is supported and compiles the appropriate files.
TBR=bjornv@webrtc.org
TEST=audioproc (performance testing), audioproc_unittest, video_processing_unittests, build on Linux (targeting ia32/x64, with disable_sse2==0/1), Mac, Windows
Review URL: http://webrtc-codereview.appspot.com/352008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1425 4adac7df-926f-26a2-2b94-8c16560cd09d
Removed some Valgrind warnings by closing output files. There are still some Valgrind warnings left, that needs to be fixed by a developer with more insight.
Updated all include paths to contain full paths to header files.
Tested in Debug+Release on Linux, Mac and Windows.
All tests ran successfully except the VideoProcessingModuleTest.ContentAnalysis test that fails on Windows with the following error:
unknown file: error: SEH exception with code 0xc0000005
thrown in the test body.
Fixing that is out of scope for this CL.
Review URL: http://webrtc-codereview.appspot.com/266011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1217 4adac7df-926f-26a2-2b94-8c16560cd09d
General Notes:
1. In general, API structure was not modified and is based on VPLIB.
2. Modification to API: Return values are based on libyuv, i.e. 0 if ok, a negative value in case of an error (instead of length).
3. All scaling (inteprolation) is now done via the scale interface. Crop/Pad is not being used.
4. VPLIB was completely removed. All tests are now part of the libyuv unit test (significantly more comprehensive and based on gtest).
5. JPEG is yet to be implemented in LibYuv and therefore existing implementation remains.
Review URL: http://webrtc-codereview.appspot.com/258001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1140 4adac7df-926f-26a2-2b94-8c16560cd09d
