Check for the case when PostThreadMessage returns false but GetLastError returns ERROR_SUCCESS.
TBR=olka
NOTRY=true
Review-Url: https://codereview.webrtc.org/2063313003
Cr-Commit-Position: refs/heads/master@{#13143}
Current number of threads selection code does not work well
for Android builds - middle and low end devices are having hard time
encoding VGA and QVGA with just one thread.
Increase the amount of vp8 encoder threads for 180p and above resolution.
Also limit maximum number of thread to 3, since for 8 core devices
most of time 4 cores are idle when thermal throttling kicks in.
BUG=b/27946721
R=marpan@webrtc.org
Review URL: https://codereview.webrtc.org/2058753003 .
Cr-Commit-Position: refs/heads/master@{#13142}
The RtpReceiverObserverInterface is created.
The SignalFirstPacketReceived will be forwarded from BaseChannel to WebRtcSession.
WebRtcSession will forward SignalFirstAudioPacketReceived and SignalFirstVideoPacketReceived to the RtpReceiverInterface.
The application can listen to the Signal by implementing and registering a RtpReceiverObserver.
Review-Url: https://codereview.webrtc.org/1999853002
Cr-Commit-Position: refs/heads/master@{#13139}
It was already disabled for browsers by design, and for everyone else
because of a bug.
BUG=webrtc:5922
Review-Url: https://codereview.webrtc.org/2055493003
Cr-Commit-Position: refs/heads/master@{#13138}
It appears there the encode and send operation can happen over multiple
threads. Also, padding data itself may be sent on a different thread.
Remove thread checker and protect all data with crit_sect.
BUG=webrtc:5998
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2066863002
Cr-Commit-Position: refs/heads/master@{#13137}
Added the plumbing necessary to get two different lists of codecs from
WebRtcVoiceEngine up to MediaSessionDescriptionFactory.
This should be the last step in this set of CLs. Once
https://codereview.webrtc.org/1991233004/ has landed, it's possible to
implement the ReceiveCodecs getter with the info from the
AudioDecoderFactory. The factory needs to be updated to actually
produce the correct list, as well.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2013053002
Cr-Commit-Position: refs/heads/master@{#13131}
Camera1 tests are now separated from general CameraVideoCapturer tests.
Main motivation behind these changes is that Camera2 implementation can
be tested using the same tests.
CL also reduces code duplication on tests using textures.
BUG=webrtc:5519
Review-Url: https://codereview.webrtc.org/2024843002
Cr-Commit-Position: refs/heads/master@{#13130}
This CL fixes an issue with the non-smoothing implementation where frames were delivered on the decoder thread. No-smoothing is now done in a separate class that uses a TaskQueue. The implementation may drop frames if the renderer doesn't keep up and it doesn't block the decoder thread.
Further work done:
* I added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 5 locks.
* I removed the Start/Stop methods from the IncomingVideoStream implementations. Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running". This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
* I changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface. This means that any implementation of that interface can be used and the decoder can be made to just use the 'renderer' from the config. Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing. The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
* The non-smoothing IncomingVideoStream implementation currently allows only 1 outstanding pending frame. If we exceed that, the current frame won't be delivered to the renderer and instead we deliver the next one (since when this happens, the renderer is falling behind).
* The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
* Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
* Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes. The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
* Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction. This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
* Made the render delay value in VideoRenderFrames, const.
BUG=
Review-Url: https://codereview.webrtc.org/2035173002
Cr-Commit-Position: refs/heads/master@{#13129}
Added initial support for MediaSessionDescriptionFactory to pick different codecs based on communications direction (sendrecv, sendonly, recvonly, inactive) specifically for audio.
This adds some more degradation options for the answer: depending on answer options, it's now possible to degrade to INACTIVE from any offer, as well as to either RECVONLY or SENDONLY from a SENDRECV offer.
The set of "codecs" used for testing the answer was compiled using this spreadsheet:
https://docs.google.com/a/google.com/spreadsheets/d/1nVIfZLsFo5YK10_e80BCAADZnnRQ1devwwwAGmqJPow/edit?usp=sharing
I should probably condense it into a smaller table and put in the source.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/1956343002
Cr-Commit-Position: refs/heads/master@{#13126}
Move all its configuration to all.gyp instead, which is
not processed by Chromium builds (webrtc.gyp is).
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2060873002
Cr-Commit-Position: refs/heads/master@{#13124}
Earlier, no statistics were reported if no frames were being delivered
for encoding. This makes statics always be reported regardless of if
there are frames being delivered to the encoder.
Review-Url: https://codereview.webrtc.org/2051403002
Cr-Commit-Position: refs/heads/master@{#13122}
not to be created in apprtc on Android.
The path separator was missing when the path for the aecdump
file was created. This CL adds that path separator.
Note that the change of the formatting of the rest of the
line was caused by "git cl format" (the clang automatic
formatting).
BUG=webrtc:5991
Review-Url: https://codereview.webrtc.org/2053263002
Cr-Commit-Position: refs/heads/master@{#13121}
Right now if an exception is thrown, it doesn't seem to be logged
anywhere. This CL makes it show a pop-up with the error message.
This should save time debugging issues.
Review-Url: https://codereview.webrtc.org/2049933004
Cr-Commit-Position: refs/heads/master@{#13120}
We were passing the pointer to the sockaddr to usrsctp_dumppacket,
instead of the pointer to the data. So we were just logging random
bytes. The dangers of void*...
NOTRY=True
TBR=pthatcher@webrtc.org
BUG=619372
Review-Url: https://codereview.webrtc.org/2061093003
Cr-Commit-Position: refs/heads/master@{#13119}
BUG=webrtc:5949
TESTED=Built and ran the tests on Mac.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2054413002
Cr-Commit-Position: refs/heads/master@{#13116}
This CL eliminates repeated calls to AudioEffect.queryEffects() on Android when configuring the audio device. Each of these calls was taking 5-10 milliseconds on the devices I was testing (Nexus 4, Nexus 5), and setting up the audio device involved around 10 of these calls.
This change adds a method that checks the cached list of effects before calling the underlying operating system API; this eliminated about half of these calls. The other half happened inside static methods such as NoiseSuppressor.isAvailable(), which are just convenience wrappers for searching through the list of effects. These calls have been replaced with searching through the cached list of effects, reducing the time to configure audio processing effects from 60-80 ms to 5-10. This results in a similar improvement in call setup time.
BUG=
Review-Url: https://codereview.webrtc.org/2051323002
Cr-Commit-Position: refs/heads/master@{#13115}
CreatePeerConnectionFactory does not yet expose the ability to set the
factory from the outside.
Added notry due to android_dbg being broken.
NOTRY=True
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/1991233004
Cr-Commit-Position: refs/heads/master@{#13112}
The error hasn't been noticed since we don't really do
(or support) Mac 32-bit builds.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2065583002
Cr-Commit-Position: refs/heads/master@{#13111}
Introduce a new method I420Buffer::CropAndScale, and a static
convenience helper I420Buffer::CenterCropAndScale. Use them for almost
all scaling needs.
Delete the Scaler class and the cricket::VideoFrame::Stretch* methods.
BUG=webrtc:5682
R=pbos@webrtc.org, perkj@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2020593002 .
Cr-Commit-Position: refs/heads/master@{#13110}
Changes:
* Enabled protobuf for iOS globally.
* Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global
scope similar to GYP since tests depend on it.
* Added missing rtc_libvpx_build_vp9 variable.
* Moved out audio_coding defines into .gni file to avoid code duplication
* Renamed files to avoid object naming conflicts that GN disallows:
* webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc}
* webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc}
BUG=webrtc:5949
TESTED=Built and ran the tests on Mac. Also ran:
gn gen out/Default --args="rtc_enable_bwe_test_logging=true"
and verified that more objects are being built (1885 vs 1883)
when compiling modules_unittests.
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2041233006
Cr-Commit-Position: refs/heads/master@{#13108}
Changes:
* Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope
to match GYP.
* Enable sctpdataengine_unittest.cc for iOS, which should have
been done in https://codereview.webrtc.org/1587193006
* Renamed GN target rtc_base_test_utils -> rtc_base_tests_utils
to match GYP.
* Added dependencies on call, modules/video_coding and video for
rtc_media.
* Added dependency on audio for rtc_media_unitttests (couldn't be
added to rtc_media due to circular dependency problem).
BUG=webrtc:5949
TESTED=Built and ran the tests on Mac.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2050313002
Cr-Commit-Position: refs/heads/master@{#13106}
Due to a bug, the NetworkManager was reconnecting to the NetworkMonitor's NetworkChanged signal every time the network manager is stopped and restarted. After each calls, one more listener was added to the signal and never removed - which caused OnNetworksChanged to be called multiple times on each actual network change.
Not sure if this had any negative effect other than the extraneous "Network changed" lines in WebRTC logs, but it wasn't working correctly either way.
The fix is to only subscribe to the signal once, when the NetworkMonitor is created.
TBR=pthatcher
BUG=
NOTRY=True
Review-Url: https://codereview.webrtc.org/2054583002
Cr-Commit-Position: refs/heads/master@{#13105}
Every message will now be traced with the location from which it was
posted, including function name, file and line number.
This CL also writes a normal LOG message when the dispatch took more
than a certain amount of time (currently 50ms).
This logging should help us identify messages that are taking
longer than expected to be dispatched.
R=pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2019423006 .
Cr-Commit-Position: refs/heads/master@{#13104}
To avoid the case where a single data point or too short window is used,
causing bad behavior due to bad stats, update RateStatistics to return
an Optional rather than a plain rate.
There was also a strange off by one bug where the rate was slightly
overestimated (N + 1 buckets, N ms time window).
These changes requires updates to a number of places, and may very well
cause seeming perf regressions (but the stats were probablty more wrong
previously).
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2029593002 .
Cr-Commit-Position: refs/heads/master@{#13103}
Instead of the default copy constructor, the Copy() method has to be used. In this CL, the number of copies has been reduced significantly in production code. One case in the video engine remains, where we need to restart a video stream. Even in that case, I'm sure we could avoid it, but for this particular CL, I decided against it to keep things simple (and it's also an edge case). Most importantly, creating copies is made harder and the interface encourages ownership transfers.
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/2042603002 .
Cr-Commit-Position: refs/heads/master@{#13102}
NOTRY=True # two of the androids bots are not cooperating
BUG=
Review-Url: https://codereview.webrtc.org/2057533002
Cr-Commit-Position: refs/heads/master@{#13099}
Chromium uses gn gen --check, which doesn't like some of the includes
used in the new gn targets the fuzzers use. This breaks Chromium
libfuzzer compiles, for which there isn't yet a webrtc FYI bot.
I'm working on fixing the includes, at which point these can come back.
BUG=chromium:618901
NOTRY=true
Review-Url: https://codereview.webrtc.org/2053293002
Cr-Commit-Position: refs/heads/master@{#13098}
These parts were commented out to avoid breaking the Chromium
WebRTC FYI bots. Include them in the WebRTC build to make our bots
build as many as possible of our GN targets.
BUG=webrtc:5949
NOTRY=True
TBR=phoglund@webrtc.org
Review-Url: https://codereview.webrtc.org/2054903002
Cr-Commit-Position: refs/heads/master@{#13097}
