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}
This should make it more clear what's supported and what's
only checked due to legacy code. No longer uses the word deprecated
since it may be confusing.
BUG=webrtc:5095
NOTRY=True
Review-Url: https://codereview.webrtc.org/1513483006
Cr-Commit-Position: refs/heads/master@{#13094}
Otherwise, we'll read out of bounds if the packet is too small.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2040953003
Cr-Commit-Position: refs/heads/master@{#13093}
This CL implements auto pausing video streams per stream with logic to
avoid toggling state too often.
Also re-enabling tests disabled for Mac, with the assumption the new
logic removes flakiness.
BUG=webrtc:5868,webrtc:5407
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2035383002 .
Cr-Commit-Position: refs/heads/master@{#13092}
to be compatible with projects that has own base/thread_annotation.h
NOTRY=TRUE
Review-Url: https://codereview.webrtc.org/2055473002
Cr-Commit-Position: refs/heads/master@{#13090}
This CL turns nativeConfiguration into createNativeConfiguration returning a
pointer or nil on failure. This method's certificate generation is updated to
use the new API and reports failure (nil) if unsuccessful instead of relying on
the default certificate. We also remove the implicit assumption (now incorrect)
that RSA is the default. This is the same type of changes as was done in
https://codereview.webrtc.org/1965313002 but this file
(RTCPeerConnectionInterface.mm) was forgotten.
With no more usages of kIdentityName it and dtlsidentitystore.cc is removed.
Also removes unnecessary #include in peerconnectioninterface.h that was still
remnant due to an indirect include of kIdentityName.
RTCConfiguration+Private.h now lists method nativeEncryptionKeyTypeForKeyType
which was added in the above mentioned prior CL.
BUG=webrtc:5707, webrtc:5708
Review-Url: https://codereview.webrtc.org/2035473004
Cr-Commit-Position: refs/heads/master@{#13089}
Sync the GYP and GN targets and update the name of the GN one
to 'remote_bitrate_estimator'.
Move the GYP variable 'enable_bwe_test_logging' into the local scope.
Remove redundant entries in modules.gyp.
These are preparations related to the GN migration.
BUG=webrtc:5949
TESTED=Ran GYP with the default variables and with
-Denable_bwe_test_logging=1. Compiled remote_bitrate_estimator
and verified that bwe_test_logging.cc is compiled only when
set.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2040313004
Cr-Commit-Position: refs/heads/master@{#13087}
In org.webrtc.VideoCapturerAndroidTest#startWhileCameraIsAlreadyOpenAndCloseCamera,
use a video renderer instead of a capture observer. The video renderer
automatically returns the texture buffers, which resolves the bug.
There shouldn't be any changes to the effectiveness of the test.
BUG=webrtc:5982
Review-Url: https://codereview.webrtc.org/2042283004
Cr-Commit-Position: refs/heads/master@{#13085}
Reason for revert:
Plan to reland with InitToBlack kept, to be able to update Chrome to use the new I420Buffer::SetToBlack method.
Original issue's description:
> Revert of New static method I420Buffer::SetToBlack. (patchset #4 id:60001 of https://codereview.webrtc.org/2029273004/ )
>
> Reason for revert:
> Breaks chrome, in particular, the tests in
>
> media_stream_remote_video_source_unittest.cc
>
> use the InitToBlack method which is being deleted.
>
> Original issue's description:
> > New static method I420Buffer::SetToBlack.
> >
> > Replaces cricket::VideoFrame::SetToBlack and
> > cricket::WebRtcVideoFrame::InitToBlack, which are deleted.
> >
> > Refactors the black frame logic in VideoBroadcaster, and a few of the
> > tests.
> >
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/663f9e2ddc86e813f6db04ba2cf5ac1ed9e7ef67
> > Cr-Commit-Position: refs/heads/master@{#13063}
>
> TBR=perkj@webrtc.org,pbos@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/271d74078894bb24f454eb31b77e4ce38097a2fa
> Cr-Commit-Position: refs/heads/master@{#13065}
TBR=perkj@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2049513005
Cr-Commit-Position: refs/heads/master@{#13083}
The AppRTCDemo app on Mac OSX does not show or send local video streams,
as ACFoundation capture session is not compiled in or implemented in
the appropriate places. This is the first part of a two-part patch
that implements local capture on the Mac for AppRTCDemo
P.S. This is my first patch to WebRTC. I didn't see any relevant tests, but I could write some if you can point me at a location. Also, I don't have access to the automated tests (I don't think)
BUG=webrtc:3417
Review-Url: https://codereview.webrtc.org/2046863004
Cr-Commit-Position: refs/heads/master@{#13080}
Previously RefCountedObject was passing all parameters by value.
This meant that it was hard to use it with movable types, such
as unique_ptr<>. Now there is a constructor that takes r-value,
which means that RefCountedObject<std::unique_ptr<foo>> can be
initialized by passing std::unique_ptr<foo> to the constructor.
Review-Url: https://codereview.webrtc.org/2036123002
Cr-Commit-Position: refs/heads/master@{#13079}
