The specializations for 4-byte reading did not return correct
values. This has to do with the order of casting and shifting. Also
adding a test to expose the bug (and verify the other byte sizes).
Review URL: https://codereview.webrtc.org/1615653011
Cr-Commit-Position: refs/heads/master@{#11364}
Once we have eliminated all non-monotonic clocks, revert this change.
BUG=webrtc:5452
Review URL: https://codereview.webrtc.org/1618333002
Cr-Commit-Position: refs/heads/master@{#11361}
This reverts commit 1a8240c32a14a31b1417b6e06f511f2a16d81b19.
Per comments in bug 5136, the affected test should no longer be flaky.
BUG=webrtc:5136
Review URL: https://codereview.webrtc.org/1616273004
Cr-Commit-Position: refs/heads/master@{#11360}
In AEC, audio levels are calculated in frequency domain. This makes the calculation dependent on FFT. We now make the calculation performed in time domain. The complexity is the same, but the dependence on FFT is removed.
BUG=
Review URL: https://codereview.webrtc.org/1542573002
Cr-Commit-Position: refs/heads/master@{#11357}
While transitioning over to rtc::CriticalSection completely, this gives perf benefits that rtc::CriticalSection has on Mac to current users of CriticalSectionWrapper.
BUG=
Review URL: https://codereview.webrtc.org/1614373002
Cr-Commit-Position: refs/heads/master@{#11356}
Use PlatformThreadRef, CurrentThreadRef and IsThreadRefEqual instead of pthread_t, pthread_self and operator== (or !=).
BUG=
Review URL: https://codereview.webrtc.org/1619153003
Cr-Commit-Position: refs/heads/master@{#11355}
According to my measurements, it's about 100x faster than the native mutex implementation in OSX. Google "OSX mutex performance" for more info.
BUG=
Review URL: https://codereview.webrtc.org/1594723003
Cr-Commit-Position: refs/heads/master@{#11352}
While doing this, I made a couple of minor changes:
* Removed unused variables (one lock and one video frame variable)
* Switched over to a scoped lock in remb.cc and removed an if() in a function where we can just return the expression being checked.
BUG=
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1613053003 .
Cr-Commit-Position: refs/heads/master@{#11349}
Also remove mischievous tab character!
This is a part of getting rid of CriticalSectionWrapper and makes the code slightly simpler.
BUG=
Review URL: https://codereview.webrtc.org/1607353002
Cr-Commit-Position: refs/heads/master@{#11346}
When the file was rewound, the remaining audio read was inserted at
the start of the destination array, not where the first reading
attempt ended.
R=ivoc@webrtc.org
Review URL: https://codereview.webrtc.org/1612053002
Cr-Commit-Position: refs/heads/master@{#11343}
We can't assume that the audio source implementation will be our own internal one and we shouldn't apply local audio options to a remote audio track this way either.
BUG=5423
R=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1576913002 .
Cr-Commit-Position: refs/heads/master@{#11341}
Callers can just remember the return value of
RentACodec::RentEncoderStack instead.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1612713002
Cr-Commit-Position: refs/heads/master@{#11340}
The FFmpeg video decoder requires up to 8 additional bytes to be allocated for its encoded image buffer input, due to optimized byte readers over-reading on some platforms.
We plan to use FFmpeg for a soon-to-land H.264 enc/dec.
This CL adds support for padding encoded image buffers based on codec type, and makes sure calls to VCMEncodedFrame::VerifyAndAllocate use the padding.
All padding constants are 0 but making H.264 pad with 8 bytes will be a one-line change.
Also, added -framework CoreFoundation to webrtc_h264_video_toolbox which was missing.
BUG=chromium:468365
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424
NOTRY=True
Review URL: https://codereview.webrtc.org/1602523004
Cr-Commit-Position: refs/heads/master@{#11337}
Adds logging to RTPSender and RTCPSender, pushing an event log pointer from Channel through ModuleRtpRtcpImpl to the Sender objects.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1571283002
Cr-Commit-Position: refs/heads/master@{#11336}
Delete methods MaybeSetRenderingSize and IsRendererRegistered, the latter replaced by std::find.
Delete return values from AddRenderer and RemoveRenderer.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1603423002
Cr-Commit-Position: refs/heads/master@{#11332}
This type is used for GL contexts in android hardware encode/decode, so
it seems correct/useful to make it publicly constructable.
Review URL: https://codereview.webrtc.org/1604403002
Cr-Commit-Position: refs/heads/master@{#11331}
Issue may occur for very small input images (e.g. 4x4) when encoded image length > input image size.
BUG=chromium:578193
Review URL: https://codereview.webrtc.org/1603643006
Cr-Commit-Position: refs/heads/master@{#11329}
This ensures the DtmfSender won't try to access it after it's
destroyed.
BUG=webrtc:5403
Review URL: https://codereview.webrtc.org/1590333004
Cr-Commit-Position: refs/heads/master@{#11327}
Several unittests were disabled on android, this CL will reenable them. One of
the tests was accidentally disabled on all platforms, and now no longer gives a
bitexact result.
BUG=webrtc:3343,webrtc:5349
Review URL: https://codereview.webrtc.org/1532903002
Cr-Commit-Position: refs/heads/master@{#11323}
Provides a better string (provides names of all implementations), but
also fixes a crash when accessing the ImplementationName() of
SimulcastEncoderAdapter where InitEncode has failed.
BUG=chromium:577932, webrtc:4897
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1599353003 .
Cr-Commit-Position: refs/heads/master@{#11321}
