Use it to separate UMA e2e delay metric between screenshare from video.
Content type extension is set based on encoder settings and processed and decoders.
Also,
Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
BUG=webrtc:7420
Review-Url: https://codereview.webrtc.org/2772033002
Cr-Commit-Position: refs/heads/master@{#17640}
The "initialize audio/video" parameters are no longer needed, but
at the same time were required to be true, causing a lot of confusion.
This CL removes them, but leaves the old method signature around,
marked "deprecated".
BUG=webrtc:3416
TBR=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2800353002
Cr-Commit-Position: refs/heads/master@{#17626}
Reason for revert:
Reland with temporary deprecated API to not break chromium and google3.
Original issue's description:
> Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ )
>
> Reason for revert:
> Suspect of breaking Chrome FYI bots.
>
> See
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder
>
> Example logs:
> ../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory
> #include "third_party/webrtc/video_encoder.h"
> ^
>
> Original issue's description:
> > Move video_encoder.h and video_decoder.h to /api and create GN targets for them
> >
> > BUG=webrtc:5881
> > # Because PRESUBMIT ignores LINT blacklist for moved files and these
> > # headers have some not easy to resolve issues.
> > NOPRESUBMIT=True
> >
> > Review-Url: https://codereview.webrtc.org/2780943003
> > Cr-Commit-Position: refs/heads/master@{#17511}
> > Committed: c42f540570
>
> TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5881
>
> Review-Url: https://codereview.webrtc.org/2794033002
> Cr-Commit-Position: refs/heads/master@{#17514}
> Committed: 716d7ac5c1TBR=solenberg@webrtc.org,sprang@webrtc.org,guidou@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5881
Review-Url: https://codereview.webrtc.org/2795163002
Cr-Commit-Position: refs/heads/master@{#17537}
Reason for revert:
Breaks Chrome FYI Android bots.
See:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus9%29/builds/20418https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus6%29/builds/14724https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/builds/20133https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/15111
Original issue's description:
> Deliver video frames on Android, on the decode thread.
>
> VideoCoding
> * Adding a method for polling for frames on Android only until the capture implementation takes care of this (longer term plan).
>
> CodecDatabase
> * Add an accessor for the current decoder
> * Use std::unique_ptr<> for ownership.
> * Remove "Release()" and "ReleaseDecoder()". Instead just delete.
> * Remove |friend| relationship between CodecDatabase and VCMGenericDecoder.
>
> VCMDecodedFrameCallback
> * DCHECKs for thread correctness.
> * Remove |lock_| now that a threading model has been established and verified.
>
> VCMGenericDecoder
> * All methods now have thread checks.
> * Variable access associated with thread checkers.
>
> VideoReceiver
> * Added two notification methods, DecoderThreadStarting() and DecoderThreadStopped()
> * Allows us to establish a period when the decoder thread is not running and it is safe to modify variables such as callbacks, that are only read when the decoder thread is running.
> * Allows us to DCHECK thread guarantees.
> * Allows synchronizing callbacks from the module process thread and have them only active while the decoder thread is running.
> * The above, allows us to establish two modes for the thread, single-threaded-mutable and multi-threaded-const.
> * Using that knowledge, we can remove |receive_crit_| as well as locking for a number of member variables.
>
> MediaCodecVideoDecoder
> * Removed frame polling code from this class, since this is now done from the root thread function in VideoReceiveStream.
>
> VideoReceiveStream
> * On Android: Polls for decoded frames every 10ms (same interval as previously in MediaCodecVideoDecoder)
> * [Un]Registers the |video_receiver_| with the module thread only around the time the decoder thread is started/stopped.
> * Notifies the receiver of start/stop events of the decoder thread.
> * Changed the decoder thread to use the new PlatformThread callback type.
>
> BUG=webrtc:7361, 695438
>
> Review-Url: https://codereview.webrtc.org/2764573002
> Cr-Commit-Position: refs/heads/master@{#17527}
> Committed: e3aa88bbd5TBR=sakal@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7361, 695438
Review-Url: https://codereview.webrtc.org/2792033003
Cr-Commit-Position: refs/heads/master@{#17530}
VideoCoding
* Adding a method for polling for frames on Android only until the capture implementation takes care of this (longer term plan).
CodecDatabase
* Add an accessor for the current decoder
* Use std::unique_ptr<> for ownership.
* Remove "Release()" and "ReleaseDecoder()". Instead just delete.
* Remove |friend| relationship between CodecDatabase and VCMGenericDecoder.
VCMDecodedFrameCallback
* DCHECKs for thread correctness.
* Remove |lock_| now that a threading model has been established and verified.
VCMGenericDecoder
* All methods now have thread checks.
* Variable access associated with thread checkers.
VideoReceiver
* Added two notification methods, DecoderThreadStarting() and DecoderThreadStopped()
* Allows us to establish a period when the decoder thread is not running and it is safe to modify variables such as callbacks, that are only read when the decoder thread is running.
* Allows us to DCHECK thread guarantees.
* Allows synchronizing callbacks from the module process thread and have them only active while the decoder thread is running.
* The above, allows us to establish two modes for the thread, single-threaded-mutable and multi-threaded-const.
* Using that knowledge, we can remove |receive_crit_| as well as locking for a number of member variables.
MediaCodecVideoDecoder
* Removed frame polling code from this class, since this is now done from the root thread function in VideoReceiveStream.
VideoReceiveStream
* On Android: Polls for decoded frames every 10ms (same interval as previously in MediaCodecVideoDecoder)
* [Un]Registers the |video_receiver_| with the module thread only around the time the decoder thread is started/stopped.
* Notifies the receiver of start/stop events of the decoder thread.
* Changed the decoder thread to use the new PlatformThread callback type.
BUG=webrtc:7361, 695438
Review-Url: https://codereview.webrtc.org/2764573002
Cr-Commit-Position: refs/heads/master@{#17527}
Reason for revert:
Suspect of breaking Chrome FYI bots.
See
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder
Example logs:
../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory
#include "third_party/webrtc/video_encoder.h"
^
Original issue's description:
> Move video_encoder.h and video_decoder.h to /api and create GN targets for them
>
> BUG=webrtc:5881
> # Because PRESUBMIT ignores LINT blacklist for moved files and these
> # headers have some not easy to resolve issues.
> NOPRESUBMIT=True
>
> Review-Url: https://codereview.webrtc.org/2780943003
> Cr-Commit-Position: refs/heads/master@{#17511}
> Committed: c42f540570TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5881
Review-Url: https://codereview.webrtc.org/2794033002
Cr-Commit-Position: refs/heads/master@{#17514}
BUG=webrtc:5881
# Because PRESUBMIT ignores LINT blacklist for moved files and these
# headers have some not easy to resolve issues.
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2780943003
Cr-Commit-Position: refs/heads/master@{#17511}
We currently leak one local reference to MediaCodecVideoDecoder in
every call to MediaCodecVideoDecoderFactory::CreateVideoDecoder. After
the decoder has been re-initialized 512 times, JNI will crash due to
local reference table overflow (max=512).
The actual leak is in the member initializer list of
MediaCodecVideoDecoder. This CL fixes the leak by adding a
ScopedLocalRefFrame outside of the ctor. All JNI code that originate
from a C++ thread (i.e. the entry point is not a Java thread) must use
a ScopedLocalRefFrame in order to avoid leaking local references.
BUG=webrtc:6969,b/36713034
Review-Url: https://codereview.webrtc.org/2780273002
Cr-Commit-Position: refs/heads/master@{#17464}
Add an attribute to the RTCConfiguration which can be used by specific
mobile devices so that the IPv6 ICE candidates on WiFi will not be collected.
BUG=b/35725283
Review-Url: https://codereview.webrtc.org/2731813002
Cr-Commit-Position: refs/heads/master@{#17100}
The check: 'RTC_CHECK_GE(slice_height, height);' has been observed to
fail after a reconfig. It looks like |slice_height| is still using the
previous resolution. |slice_height| isn't used for texture output and
hopefully this issue is texture specific. This CL only extracts and
checks |slice_height| when it's actually used.
BUG=b/35932686
Review-Url: https://codereview.webrtc.org/2736603003
Cr-Commit-Position: refs/heads/master@{#17065}
When textures are not enabled and we are using byte buffer outputs, the
decoder is currently crashing for odd heights because of an RTC_CHECK.
This CL removes the check and handles the pointer offset to the chroma
planes for the odd height case instead.
This has been verified to work correctly on a Pixel device.
BUG=webrtc:6651
Review-Url: https://codereview.webrtc.org/2709923005
Cr-Commit-Position: refs/heads/master@{#16805}
After this change, all calls to MediaCodecVideoEncoder must be made on
the same task queue. Removes OnCodecThread suffix from methods since it
is no longer meaningful.
BUG=webrtc:6290
Review-Url: https://codereview.webrtc.org/2669093004
Cr-Commit-Position: refs/heads/master@{#16792}
Moves CameraCapturer, CameraSession, Camera1Session and Camera2Session
away from the public API.
BUG=webrtc:7172
Review-Url: https://codereview.webrtc.org/2699713004
Cr-Commit-Position: refs/heads/master@{#16723}
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16675}
Previously, was only checking the Android SDK version. But it also needs
to check for the presence of the connectivity manager service.
BUG=webrtc:7026
Review-Url: https://codereview.webrtc.org/2697943002
Cr-Commit-Position: refs/heads/master@{#16631}
Reason for revert:
Breaks AppRTCMobile interoperability. The ICE candidate URL shouldn't be signaled between endpoints, it's only there for informational purposes.
Original issue's description:
> Add the url attribute to the IceCandidate (Java Wrapper)
>
> The url of the ICE server is added to the IceCandiate class.
> This can be used to tell which server this candidate was gathered from.
>
> BUG=webrtc:7128
>
> Review-Url: https://codereview.webrtc.org/2690593002
> Cr-Original-Commit-Position: refs/heads/master@{#16593}
> Committed: 8586c8ee88
> Review-Url: https://codereview.webrtc.org/2690593002
> Cr-Commit-Position: refs/heads/master@{#16615}
> Committed: 45efce01c7TBR=magjed@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2699533002
Cr-Commit-Position: refs/heads/master@{#16616}
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Original-Commit-Position: refs/heads/master@{#16593}
Committed: 8586c8ee88
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16615}
If android_setsocknetwork() is available, and it fails, then bind()
should *not* be called, and an error should be returned.
If it succeeds, then bind should be called, but with an "any" address.
This is to prevent cases where sockets are sent with a source address
that doesn't match the network interface they're sent on. See bug below.
This CL also changes "NetworkBinderResults" to an enum class, and
renames it to "NetworkBinderResult".
BUG=webrtc:7026
Review-Url: https://codereview.webrtc.org/2646863005
Cr-Commit-Position: refs/heads/master@{#16597}
To ensure compliance with older version high profile should appear in local SDP
before baseline profile.
BUG=b/34816463
Review-Url: https://codereview.webrtc.org/2696733002
Cr-Commit-Position: refs/heads/master@{#16596}
Reason for revert:
Breaks downstream application's build
Original issue's description:
> Add the url attribute to the IceCandidate (Java Wrapper)
>
> The url of the ICE server is added to the IceCandiate class.
> This can be used to tell which server this candidate was gathered from.
>
> BUG=webrtc:7128
>
> Review-Url: https://codereview.webrtc.org/2690593002
> Cr-Commit-Position: refs/heads/master@{#16593}
> Committed: 8586c8ee88TBR=magjed@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2692993002
Cr-Commit-Position: refs/heads/master@{#16595}
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16593}
Currently filed trial value which controls H.264 high profile support is
read once only when factory is created. If field trial value is changed for
the next WebRTC call supported codec list need to be updated as well.
BUG=b/34816463
Review-Url: https://codereview.webrtc.org/2685183004
Cr-Commit-Position: refs/heads/master@{#16543}
These structs will be used for ORTC objects (and their WebRTC
equivalents).
This CL also introduces some minor changes to the existing implemented
structs:
- max_bitrate_bps uses rtc::Optional instead of "-1 means unset"
- "mime_type" turned into "name"/"kind" (which can be used to form the
MIME type string, if needed).
- clock_rate and channels changed to rtc::Optional, since they will
need to be for RtpSender.send().
- Renamed "channels" to "num_channels" (the ORTC name, which I prefer).
BUG=webrtc:7013, webrtc:7112
Review-Url: https://codereview.webrtc.org/2651883010
Cr-Commit-Position: refs/heads/master@{#16437}
If an application sets a non-null value in RTCConfiguration.iceCheckMinInterval, we do not sent STUN pings more often than that. This is useful for bandwidth constrained scenarios.
This CL also increases the maximum STUN ping timeout to 60 seconds up from its previous value of 5 (which meant that a ping response received 5 seconds later would not be counted), and allows the RTT estimate to go up to 60 seconds from its previous limit of 3. RTTs above 3 seconds are possible on mobile links. (webrtc:7109)
This CL was originally written by pthatcher@, I am just submitting it after a minor cleanup.
BUG=webrtc:7082, webrtc:7109
Review-Url: https://codereview.webrtc.org/2670053002
Cr-Commit-Position: refs/heads/master@{#16421}
Bulk of the changes were done using
git grep -l '#include "webrtc/base/common.h"' | \
xargs sed -i '\,^#include.*webrtc/base/common\.h,d'
followed by adding back the include in the few places where it is
still needed, and in one case (pseudotcp.cc) instead deleting its use
of RTC_UNUSED.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2644103002
Cr-Commit-Position: refs/heads/master@{#16263}
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.
Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.
Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
Reason for revert:
The new method doesn't work as intended.
It can't pass ownership, because the StatsReports is a vector of raw pointers to StatReport objects owned by the StatsCollector.
Original issue's description:
> New method StatsObserver::OnCompleteReports, passing ownership.
>
> The new name, OnCompleteReports rather than OnComplete, is needed
> because in C++ method lookup, overriding a method hides all otherwise
> inherited methods with the same name, even if they have a different
> signature. And here, the intention is that each subclass should
> override one or the other of the two methods, and inherit the method it
> doesn't override.
>
> This cl is a prerequisite for
> https://codereview.webrtc.org/2567143003/, because the Chrome glue
> code needs to retain the stats report after the OnComplete method has
> returned.
>
> Currently, Chrome makes a copy of the stats mapping (which breaks when
> changing ValuePtr from an rtc::linked_ptr to an std::unique_ptr). After
> this cl, Chrome can be fixed to take ownership and no longer needs to
> copy anything, unblocking cl 2567143003.
>
> BUG=webrtc:6424
>
> Review-Url: https://codereview.webrtc.org/2584553002
> Cr-Commit-Position: refs/heads/master@{#15708}
> Committed: b36ee8d498TBR=solenberg@webrtc.org,magjed@webrtc.org,tkchin@webrtc.org,hbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2641783002
Cr-Commit-Position: refs/heads/master@{#16144}
Also did some slight refactoring of the code that turns constraints
into rtc::Optionals. Used a template method to avoid code duplication,
and used the same pattern for "CopyConstraintsIntoAudioOptions" as was
being used for "CopyConstraintsIntoRtcConfiguration".
BUG=webrtc:6752
Review-Url: https://codereview.webrtc.org/2628523003
Cr-Commit-Position: refs/heads/master@{#16063}
We currently leak one local reference to MediaCodecVideoEncoder in
every call to MediaCodecVideoEncoderFactory::CreateVideoEncoder. After
the encoder has been re-initialized 512 times, JNI will crash due to
local reference table overflow (max=512).
The actual leak is in the member initializer list of
MediaCodecVideoEncoder. This CL fixes the leak by adding a
ScopedLocalRefFrame outside of the ctor. All JNI code that originate
from a C++ thread (i.e. the entry point is not a Java thread) must use
a ScopedLocalRefFrame in order to avoid leaking local references.
BUG=webrtc:6969,b/34056152
Review-Url: https://codereview.webrtc.org/2627973004
Cr-Commit-Position: refs/heads/master@{#16034}
Bulk of the changes were produced using
git grep -l ' ASSERT(' | grep -v test | grep -v 'common\.h' |\
xargs -n1 sed -i 's/ ASSERT(/ RTC_DCHECK(/'
followed by additional includes of base/checks.h in affected files,
and git cl format.
Also had to do some tweaks to #if !defined(NDEBUG) logic in the
taskrunner code (webrtc/base/task.cc, webrtc/base/taskparent.cc,
webrtc/base/taskparent.h, webrtc/base/taskrunner.cc), replaced to
consistently use RTC_DCHECK_IS_ON, and some of the checks needed
additional #if protection.
Test code was excluded, because it should probably use RTC_CHECK
rather than RTC_DCHECK.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2620303003
Cr-Commit-Position: refs/heads/master@{#16030}
Bulk of changes done using
git grep -l 'RTC_DCHECK(false)' | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/'
peerconnection.cc also used RTC_DCHECK(false && "msg") in two places,
which were updated manually.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2623313004
Cr-Commit-Position: refs/heads/master@{#16026}
This cl was produced by
git grep -l 'ASSERT(false)' |\
xargs -n1 sed -i 's/ASSERT(false)/RTC_NOTREACHED()/'
followed by additional includes of base/checks.h in affected files,
git cl format to adjust spacing in webrtc/base/transformadapter.cc.
Finally, to make presubmit happy, one unnamed TODO marker was deleted
in that file.
This is a step towards deletion of base/common.h.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2625003003
Cr-Commit-Position: refs/heads/master@{#16009}
The purpose is to be able to add field trials in Java code.
BUG=webrtc:6683
Review-Url: https://codereview.webrtc.org/2621003002
Cr-Commit-Position: refs/heads/master@{#15994}
Moves webrtc/common_video/rotation.h and parts of
webrtc/common_video/include/video_frame_buffer.h and
webrtc/video_frame.h, and adds to a new GN target api:video_frame_api.
BUG=webrtc:5880
Review-Url: https://codereview.webrtc.org/2517173004
Cr-Commit-Position: refs/heads/master@{#15993}
The intention of SetConfiguration is that it modifies the configuration,
while keeping the constraints passed into CreatePeerConnection. Right
now that's now happening. See bug for more explanation.
BUG=webrtc:6942
Review-Url: https://codereview.webrtc.org/2603653002
Cr-Commit-Position: refs/heads/master@{#15974}
Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS. Maintaining deprecated
backwards compatible support for "OPT_SSLTCP".
Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.
PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.
TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.
Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.
For security reasons we ensure that:
1) The policy is always explicitly initialized to secure.
2) API users have to explicitly integrate with the feature to
use it, and will otherwise get no change in behavior.
3) The feature is not immediately exposed in non-native
contexts. For example, disabling of certificate validation
is not implemented via URI parsing since this would
immediately allow it to be used from a web page.
This is a second attempt of https://codereview.webrtc.org/2557803002/
which was rolled back in https://codereview.webrtc.org/2590153002/
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2594623002
Cr-Commit-Position: refs/heads/master@{#15967}
