Just like the C++ API, add a method in Java VideoFrame.Buffer that
describes the underlying implementation.
Use this method to properly select AndroidVideoBuffer
or AndroidVideoI420Buffer in Java -> C++ Video Frame Conversion.
Also, add a test case for WrappedNativeI420Buffer
in VideoFrameBufferTest for consistency.
Bug: webrtc:12602
Change-Id: I4c0444e8af6f6a1109bc514e7ab6c2214f1f6d60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223080
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34545}
AV1X->AV1 mapping added to SdpVideoFormatToVideoCodecInfo in
https://webrtc-review.googlesource.com/c/src/+/215586 results in
discrepancy of codec name between SDP and VideoCodecInfo. That violates
VideoCodecInfo design and breaks downstream projects.
This CL moves the mapping from VideoCodecInfoToSdpVideoFormat and
SdpVideoFormatToVideoCodecInfo to VideoCodecTypeMime.
Bug: b/181690054
Change-Id: I2a76524c29b082241f2ec72a60a209ce9b0c7c5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221205
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34230}
In the old jitter buffer the two VCMVideoProtection modes |kProtectionNone| and |kProtectionFEC| could be set on the jitter buffer for it to not wait for NACK and instead generate incomplete frames. This has not been possible for a long time.
Bug: webrtc:9378, webrtc:7408
Change-Id: I0a2d3ec34d721126c1128306d5fad88314f8d59f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32513}
This patch adds an interface for os/firmware to set a network
preference NOT_PREFERRED / NEUTRAL that can be picked up by
an IceController and used when selection ice candidate pair.
The patch exposes this using an Android Intent based interface.
BUG: webrtc:11825
Change-Id: Ic12b6bf704fde7f9c912020dd7bc79ccae4613ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180883
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31877}
The current camera switch API sequentially cycles through each
camera name for each method invocation. This policy provides
reasonable behavior for devices with 2 or 3 cameras, but
presents challenges with devices that contain several cameras.
For example in a scenario where the current camera is oriented
on the same side as the next camera name, a developer would need to
call switchCamera multiple times to capture from a camera oriented on
a different side of the device.
This commit allows a developer to specify a camera name when switching
cameras. This flexibility allows developers to have more control over
which device they switch to in cases where a device contains several cameras.
Bug: webrtc:11261
Change-Id: I93d46d70b2c7cf735a411a4ef4f33e926bf3a5ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165040
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30199}
This wires the current degradation preference in the SDK, it will later
be nullable in a follow up change once the native API supports it.
Bug: webrtc:11164
Change-Id: I8324e6e0af996dfddfa07e3aff4ba242d9533388
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161321
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30170}
This is a reland of 11dfff0878c949f2e19d95a0ddc209cdad94b3b4
Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org
Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
This reverts commit 1fa06041bcd8a0119e557d16e7b54a9110c5ad03.
Reason for revert: Likely cause for breaking downstream projects
Original change's description:
> Make negotiationneeded processing in PeerConnection spec compliant.
>
> This CL fixes the problem of misfired negotiationneeded notifications due
> to the lack of a NegotiationNeeded slot and the proper procedure to
> update it.
>
>
> Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
> Bug: chromium:740501
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27594}
TBR=steveanton@webrtc.org,magjed@webrtc.org,sakal@webrtc.org,hbos@webrtc.org,guidou@webrtc.org
Change-Id: Iad7b7d4e37227fa6a76ff830160ca3da9dbe4719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:740501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132761
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27599}
This CL fixes the problem of misfired negotiationneeded notifications due
to the lack of a NegotiationNeeded slot and the proper procedure to
update it.
Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
Bug: chromium:740501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27594}
This is a reland of 177670afd6d4aa414e4aa75983da538b7f350ee8
Fixing failing tests.
TBR=magjed@webrtc.org
Original change's description:
> Add bindings for simulcast and RIDs in Android SDK.
>
> This adds the bindings for rid in RtpParameters.Encoding and bindings
> for send_encodings in RtpTransceiverInit to allow creating a transceiver
> with multiple send encodings.
>
> Bug: webrtc:10464
> Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27323}
Bug: webrtc:10464
Change-Id: I95fac3967217c20a9fdddb490aea30eca2061ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130362
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27402}
This adds the bindings for rid in RtpParameters.Encoding and bindings
for send_encodings in RtpTransceiverInit to allow creating a transceiver
with multiple send encodings.
Bug: webrtc:10464
Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27323}
This CL adds a way to extract the underlying android.opengl.EGLContext
and javax.microedition.khronos.egl.EGLContext for EglBase14 and
EglBase10 respectively. The reason is that clients can't be expected to
use only WebRTC's OpenGL code and might need to integrate with their
own GL code.
Bug: None
Change-Id: Ie00a564de45a090683542a52005da7e43c586ced
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127888
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27205}
This is a propagation of upstream chromium change needed to
resume DEPS autorolls into WebRTC.
Original comment from upstream change:
> This change is made in preparation for an ErrorProne
> check to catch this at compile time. See bug for details.
Bug: chromium:771683
Change-Id: I56aed15f73a633dcadae7ece6c645cd3596f9257
Reviewed-on: https://webrtc-review.googlesource.com/c/113505
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25951}
These changes simplify the code, and also fix the issue where the peerconnectionstate would sometimes return to "new" during connection setup.
Bug: webrtc:9308
Change-Id: I895cd2f94a2b9688c821cca64d1a077317b99d44
Reviewed-on: https://webrtc-review.googlesource.com/c/111964
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25942}
This reverts commit 1e87b4f32b73526f9caaae2a7bccfbd0cd84dcb9.
Reason for revert: Breaks internal project
Original change's description:
> Replace the IceConnectionState implementation.
>
> PeerConnection::ice_connection_state() used to return a value based on both DTLS and ICE transports.
> Now that we have PeerConnection::peer_connection_state() to fill that role we can change the implementation of ice_connection_state over to match the spec.
>
> Bug: webrtc:6145
> Change-Id: Ia4f348f728f24faf4b976c63dea2187bb1f01ef0
> Reviewed-on: https://webrtc-review.googlesource.com/c/108780
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25773}
TBR=kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,jonasolsson@webrtc.org
Change-Id: Icc4368d120a4167286fa6ba2e884a3650b453eff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6145
Reviewed-on: https://webrtc-review.googlesource.com/c/111925
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25775}
PeerConnection::ice_connection_state() used to return a value based on both DTLS and ICE transports.
Now that we have PeerConnection::peer_connection_state() to fill that role we can change the implementation of ice_connection_state over to match the spec.
Bug: webrtc:6145
Change-Id: Ia4f348f728f24faf4b976c63dea2187bb1f01ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/108780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25773}
Some imports of classes in the same package are a bit silly.
Removing = false for booleans is safe because Java guarantees that
an uninitialized bool will always be false.
Tbr: sakal@chromium.org
Bug: None
Change-Id: I04baa78a6e21b1c4fc74c5e46665e66481da2495
Reviewed-on: https://webrtc-review.googlesource.com/c/111243
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25678}
This support is needed if there is a big delay between the creation of
frames and the time they are delivered to the WebRTC C++ layer in
AndroidVideoTrackSource. This is the case if e.g. some heavy video
processing is applied to the frames that takes a couple of hundred
milliseconds. Currently, timestamps coming from Android video sources
are aligned to rtc::TimeMicros() once they reach the WebRTC C++ layer in
AndroidVideoTrackSource. At this point, we "forget" any latency that
might occur before this point, and audio/video sync consequently
suffers.
Bug: webrtc:9991
Change-Id: I7b1aaca9a60a978b9195dd5e5eed4779a0055607
Reviewed-on: https://webrtc-review.googlesource.com/c/110783
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25654}
This change makes it possible for android apps to use the new standards-compliant PeerConnectionState.
Bug: webrtc:9977
Change-Id: Iad19c38e664a59e86879715ec7a04a59a9894bee
Reviewed-on: https://webrtc-review.googlesource.com/c/109883
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25652}
Support Injectable Audio Codecs from the Java SDK.
The PeerConnectionFactory.Builder defaults to
BuiltinAudio(Encoder|Decoder)Factory, but other implementations are
permitted via the Audio(Encoder|Decoder)FactoryFactory interface.
Bug: webrtc:9916
Change-Id: I61ad4a6e57666bc1be79daf5f40b129e0eacad84
Reviewed-on: https://webrtc-review.googlesource.com/c/107711
Commit-Queue: Lennart Kolmodin <kolmodin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25478}
This is implemented by allowing users to set two different aspect
ratios, one for landscape input and one for portrait input. This extra
control might be useful in other scenarios as well.
Bug: webrtc:9903
Change-Id: I91676737f4aa1f5d94cfe79ac51d5f866779945b
Reviewed-on: https://webrtc-review.googlesource.com/c/108086
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25387}
With the expanding use cases for webrtc::CryptoOptions it makes more sense for
it to be be available per peer connection instead of only as a factory option.
To support backwards compatability for now this code will support the factory
method of setting crypto options by default. However it will completely
overwrite these settings if an RTCConfiguration.crypto_options is provided.
Got LGTM offline from Sami, adding him to TBR if he has any further comments.
TBR=sakal@webrtc.org
Bug: webrtc:9891
Change-Id: I86914cab69284ad82afd7285fd84ec5f4f2c4986
Reviewed-on: https://webrtc-review.googlesource.com/c/107029
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25375}
Also enables support for all hardware implementations. Renames
HardwareVideoDecoderFactory to MediaCodecVideoDecoderFactory. Renames
HardwareVideoDecoder to AndroidVideoDecoder.
Bug: webrtc:8538
Change-Id: I9b351f387526af4da61fb07c07fb4285bd833e19
Reviewed-on: https://webrtc-review.googlesource.com/97680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24586}
