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}
Reason for revert:
Doing a reland where systeminfo.cc includes basictypes.h so that CPU_X86 etc. are defined when they are checked/used.
Original issue's description:
> Revert of Replace basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2604043002/ )
>
> Reason for revert:
> Very likely cause of Chromium import bot breakage (unused function '__cpuid'), TBD why.
>
> Original issue's description:
> > Replace basictypes.h with stdint.h for int_t types.
> >
> > Removes basictypes.h for types that only makes use of it for fixed-size-int
> > typedefs and replaces it with stdint.h.
> >
> > BUG=webrtc:6853
> > R=tommi@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2604043002
> > Cr-Commit-Position: refs/heads/master@{#15867}
> > Committed: 7fd1a75300
>
> TBR=tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6853
>
> Review-Url: https://codereview.webrtc.org/2603203003
> Cr-Commit-Position: refs/heads/master@{#15869}
> Committed: 7eb0e23bcf
BUG=webrtc:6853
TBR=tommi@webrtc.org
Review-Url: https://codereview.webrtc.org/2609783002
Cr-Commit-Position: refs/heads/master@{#15873}
Reason for revert:
Very likely cause of Chromium import bot breakage (unused function '__cpuid'), TBD why.
Original issue's description:
> Replace basictypes.h with stdint.h for int_t types.
>
> Removes basictypes.h for types that only makes use of it for fixed-size-int
> typedefs and replaces it with stdint.h.
>
> BUG=webrtc:6853
> R=tommi@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2604043002
> Cr-Commit-Position: refs/heads/master@{#15867}
> Committed: 7fd1a75300TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6853
Review-Url: https://codereview.webrtc.org/2603203003
Cr-Commit-Position: refs/heads/master@{#15869}
Removes basictypes.h for types that only makes use of it for fixed-size-int
typedefs and replaces it with stdint.h.
BUG=webrtc:6853
R=tommi@webrtc.org
Review-Url: https://codereview.webrtc.org/2604043002
Cr-Commit-Position: refs/heads/master@{#15867}
Created a java wrapper for the callback OnAddTrack in this CL since it has been added to native C++ API
The callback function is called when a track is signaled by remote side and a new RtpReceiver is created.
The application can tell when tracks are added to the streams by listening to this callback.
BUG=webrtc:6112
Review-Url: https://codereview.webrtc.org/2513723002
Cr-Commit-Position: refs/heads/master@{#15745}
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}
Reason for revert:
This CL broke all Chromium WebRTC FYI bots. A roll+fix was attempted here: https://codereview.chromium.org/2590783003/, but failed to land. I'm reverting this CL now to make the tree green again. Make the API change gradual when you reland so that we can update Chromium between.
Original issue's description:
> Add disabled certificate check support to IceServer PeerConnection API.
>
> 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.
>
> 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.
>
> BUG=webrtc:6840
>
> Review-Url: https://codereview.webrtc.org/2557803002
> Cr-Commit-Position: refs/heads/master@{#15670}
> Committed: b0f04fdb9eTBR=pthatcher@webrtc.org,deadbeef@webrtc.org,hnsl@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2590153002
Cr-Commit-Position: refs/heads/master@{#15703}
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.
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.
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2557803002
Cr-Commit-Position: refs/heads/master@{#15670}
Current implementation of JavaToStdString applies additional encoding that modifies ISO-8859-1 encoded strings (e.g. byte array). This CL is to fix this.
A planned use of this is to pass a protobuf serialized string as a MediaConstraint to WebRTC to configure audio network adaptor.
BUG=webrtc:6815
Review-Url: https://codereview.webrtc.org/2549783002
Cr-Commit-Position: refs/heads/master@{#15509}
Reason for revert:
There was a bug in the implementation where the adapter could get stuck at really low resolutions. That has now been fixed.
Original issue's description:
> Revert of Add ability to scale to arbitrary factors (patchset #7 id:120001 of https://codereview.webrtc.org/2555483005/ )
>
> Reason for revert:
> Issue discovered with scaling back up.
>
> Original issue's description:
> > Add ability to scale to arbitrary factors
> >
> > This CL adds a fallback for the case when no optimized scale factor produces a low enough resolution for what was requested. It also ensures that all resolutions provided by the video adapter are divisible by four. This is required by some hardware implementations.
> >
> > BUG=webrtc:6837
> >
> > Committed: https://crrev.com/710c335d785b104bda4a912bd7909e4d27f9b04f
> > Cr-Commit-Position: refs/heads/master@{#15469}
>
> TBR=magjed@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6837
>
> Committed: https://crrev.com/7722a4cc8d31e5e924e9e6c5c97412ce8bbbe59d
> Cr-Commit-Position: refs/heads/master@{#15470}
R=magjed@webrtc.org
BUG=webrtc:6837,webrtc:6848
Review-Url: https://codereview.webrtc.org/2558243003
Cr-Commit-Position: refs/heads/master@{#15485}
Reason for revert:
Issue discovered with scaling back up.
Original issue's description:
> Add ability to scale to arbitrary factors
>
> This CL adds a fallback for the case when no optimized scale factor produces a low enough resolution for what was requested. It also ensures that all resolutions provided by the video adapter are divisible by four. This is required by some hardware implementations.
>
> BUG=webrtc:6837
>
> Committed: https://crrev.com/710c335d785b104bda4a912bd7909e4d27f9b04f
> Cr-Commit-Position: refs/heads/master@{#15469}
TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6837
Review-Url: https://codereview.webrtc.org/2557323002
Cr-Commit-Position: refs/heads/master@{#15470}
This CL adds a fallback for the case when no optimized scale factor produces a low enough resolution for what was requested. It also ensures that all resolutions provided by the video adapter are divisible by four. This is required by some hardware implementations.
BUG=webrtc:6837
Review-Url: https://codereview.webrtc.org/2555483005
Cr-Commit-Position: refs/heads/master@{#15469}
It should be 'JNIEXPORT rettype JNICALL' not 'rettype JNIEXPORT JNICALL'.
BUG=webrtc:6660
Review-Url: https://codereview.webrtc.org/2557793003
Cr-Commit-Position: refs/heads/master@{#15467}
Create the RtpReceiver.Observer which is a Java wrapper over the webrtc::RtpReceiverObserverInterface.
The callback function onFirstPacketReceived will be called whenever the first audio or video packet it received.
BUG=webrtc:6742
Review-Url: https://codereview.webrtc.org/2531333003
Cr-Commit-Position: refs/heads/master@{#15464}
I decided to make one webrtc/sdk/android/BUILD.gn for both tests and Java/jni src.
External dependencies needs to be updated after this CL.
Future work is required to clean up the Android api and move
implementation details to /webrtc/sdk/android/src.
BUG=webrtc:5882,webrtc:6804
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2547483003
Cr-Commit-Position: refs/heads/master@{#15443}
