EglThread can be shared by multiple clients each using their own
EglBase instance, but sharing thread and EglConnection.
go/meet-android-eglcontext-reduction
Bug: b/225229697
Change-Id: I2d18b92bdef51362a9dbd9c0af56cb868e29869d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305462
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Linus Nilsson <lnilsson@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40121}
Recreating the VirtualDisplay will require new user permission dialog,
so resize instead when possible.
Bug: b/281978124
Change-Id: I3b6939720897c038c9e598433372342cf72e001e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305560
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Linus Nilsson <lnilsson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40084}
This enables clients of EglBase to keep using it but
share underlying EGLContext with other clients.
go/meet-android-eglcontext-reduction
Bug: b/225229697
Change-Id: I42719f25be7db169c39878b57a5f1487e3c1894e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301941
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Linus Nilsson <lnilsson@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39961}
https://webrtc-review.googlesource.com/c/src/+/240680 made encoder aware of stride and slice height of input buffer but calculation of buffer size passed to queueInputBuffer() was not updated.
Bug: webrtc:13427
Change-Id: Iba8687f56eda148ac67b331d35c45317a4ec5c59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301321
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39895}
Before this change we first released output frame buffer in the code path which prepends config buffer to a keyframe and then called getOutputFormat(index). getOutputFormat(index) throws an exception if index points to a released buffer. This change rearranges calls such that getOutputFormat(index) always executed before releaseOutputBuffer(index).
Bug: webrtc:15015
Change-Id: Ia64f5d8e7483aeeb316d1a71c0cb79233f4bbee9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301364
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39874}
There was no check for null in the code that prepends config buffer to key frame buffer. It is not a requirement for codec to deliver config buffer. Some codecs probably do not do that.
Bug: none
Change-Id: Id9c57efc5d1de5f569fa773313da4db3cd8b074c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299900
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39807}
It is a part of "encoding statistics" feature [1] available in Android SDK 33. Local testing revealed that for HW VP8/9 encoders we get QP in range [0,64] which is not what WebRTC quality scaler expects. Exclude VP8/9 encoders for now.
[1] https://developer.android.com/reference/android/media/MediaFormat#VIDEO_ENCODING_STATISTICS_LEVEL_1
Bug: webrtc:15015
Change-Id: I8af2fd96afb34e18cb3e2cc3562b10149324c16e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298306
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39722}
Replace locally-defined keys and values with constants from MediaCodec API (MediaFormat.KEY_..., etc). Value of a constant field is resolved at compile time according to 13.1.1 [1]. I.e., it is safe to reference a constant field not available in older API (MediaCodec API ignores unrecognized MediaFormat.KEY_ values).
[1] https://docs.oracle.com/javase/specs/jls/se20/html/jls-13.html#jls-13.1
Bug: none
Change-Id: I3c63cfd67cc22db1b957f908508b434d36d88806
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298940
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Linus Nilsson <lnilsson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39668}
This enables testing HW H265 codecs on devices where the support is available.
Bug: b/261160916, webrtc:14852
Change-Id: I32d102fcf483ea4ba46d6f5161342dbb584e7cc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298040
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39591}
Keeping the headers to allow compatibility with current users
that expect the headers to be in that target before they are
also updated.
Bug: webrtc:9838
Change-Id: I8b1e88850958e92c043686587a37791f01860220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39031}
making it clear what unit is being used.
BUG=webrtc:13756
Change-Id: I6354d35a8e02bb93a905ccf32cb0b294b4813e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289460
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39008}
This helps in figuring out which dependencies exist, and gets closer
to obeying the "one target per .cc file" rule.
Test failures seem unrelated, so using No-Try.
No-Try: true
Bug: webrtc:14775
Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38910}
Notifier is thread-hostile, and we have added a SequenceChecker
on https://webrtc-review.googlesource.com/c/src/+/252520 ,
so this comment is no longer needed.
Bug: None
Change-Id: I39f7f75a786dd27d2f6299d85676e7182d9032eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38899}
This cl/ attempts to fix (rather) rare crashes in
OnNetworkConnected_n by loosening the assumptions
that a network handle will keep it's network name.
With this cl/ it is possible that a NetworkHandle
can call OnNetworkConnected_n with one interface name
and then directly afterwards call it with another (
w/o an OnNetworkDisconnected_n inbetween).
This is the only scenario in which I could see the previous
crash occurring.
i.e
OnNetworkConnected(handle, "some-if-name")
OnNetworkConnected(handle, "some-other-name")
- previously this caused crash,
- now this is treated as if there was an OnNetworkDisconnected(handle) in between.
---
Also 1: shamelessly copy TYPE_MOBILE_DUN & TYPE_MOBILE_HIPRI from chromium: 87987f0e76
Also 2: Modify testcase not to use real interface names, so I can ran them on personal test phone w/o the real networks interfering.
Bug: webrtc:13741
Change-Id: I5480d5ce7031c2b5c09b958064076d02b3db1248
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285980
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38808}
AndroidNetworkMonitor::SetNetworkInfos assumes this method is called
only once, but unittests calls it twice.
One is called by the startMonitoring Java method, and the other is
called by each test.
Because of this, these tests will not succeed if dcheck_always_on is true.
To solve this problem, use OnNetworkConnected_n
instead of SetNetworkInfos in each test.
Bug: None
Change-Id: I027706ad5ccd597a91e3a66f15e181ee22d4aaa9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285861
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#38798}
On Android bindings, do not build a DtmfSender instance in a
RtpSender if its video kind is Video.
This will prevent showing an error when trying to access
that DtmfSender instance that has no native reference
Bug: webrtc:14680
Change-Id: Iba67a12cae8604c032915156b581af269f6ed265
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283742
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38724}
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.
Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
This is a reland of commit 937a59268e
Check if codec requested in createEncoder/Decoder is supported and return null if not.
Original change's description:
> Call native codec factories from Android ones.
>
> Android video codec factories are expected to be synchronised with the native ones in terms on supported codecs. But before this change there were differences:
>
> 1. Native decoder factory keeps AV1 support behind RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY while Android decoder factory advertises AV1 unconditionally;
>
> 2. Native encoder factory advertises AV1 if RTC_USE_LIBAOM_AV1_ENCODER is enabled while Android encoder factory never advertises AV1.
>
> This CL synchronises the codecs set in Android factories with that of native factories by calling native factories from Android ones.
>
> Bug: webrtc:13573, b/257272020
> Change-Id: I99d801eda0c5f3400bac222b9b08d719f1a6ed72
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282240
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38583}
Bug: webrtc:13573, b/257272020
Change-Id: Ida7bb9a2954b836a07ad560de29c1f8088264b77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282802
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38607}
Android video codec factories are expected to be synchronised with the native ones in terms on supported codecs. But before this change there were differences:
1. Native decoder factory keeps AV1 support behind RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY while Android decoder factory advertises AV1 unconditionally;
2. Native encoder factory advertises AV1 if RTC_USE_LIBAOM_AV1_ENCODER is enabled while Android encoder factory never advertises AV1.
This CL synchronises the codecs set in Android factories with that of native factories by calling native factories from Android ones.
Bug: webrtc:13573, b/257272020
Change-Id: I99d801eda0c5f3400bac222b9b08d719f1a6ed72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282240
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38583}
It's deprecated and has been removed from Chrome. Let's follow suite.
// Passing all but unrelated bots
NOTRY=True
Bug: webrtc:14608
Change-Id: I6f2601af5b1dc08164230ebf15db2d2f1754f9e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280740
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38488}
Currently if you want to obtain the stats for a specific sender/receiver
in Android, you need to call peerConnection.getStats() and filter
manually the result by sender.
pc.getStats(receiver/sender) exists in c++ and ios but was not exposed
in Android
Bug: webrtc:14547
Change-Id: I9954434880f0f93821fcd2e2de24a875e8d136ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275880
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38428}
This is needed in order to use jint and make the header self contained.
Bug: b/251890128
Change-Id: Ie6c323113370a1d49f68c783137292e1c0be07d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278780
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38351}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
