Only caller was the RtpFrameObject constructor, so it's
not needed in the interface.
To be able to delete downstream overrides, add a temporary
default implementation. Method will be completely deleted in part 2.
Bug: webrtc:9378
Change-Id: I9083b6284313b6ebce854c6f2cec4617953331d9
Reviewed-on: https://webrtc-review.googlesource.com/c/112128
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25822}
- Enable flexible mode in loopback tools and quality tests
- Ensure duplicate references are not set by the sender in video header
- Reset first active spatial layer on keyframe in encoder
- Make vp9 encoder to not generate spatial references for first active
layer with external reference control in svc flexible mode
Bug: webrtc:10049
Change-Id: If9ff576ea8a1a2fef6116b17b5b5adff08c5f8c6
Reviewed-on: https://webrtc-review.googlesource.com/c/112080
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25795}
Pass encoded frames to packetizer immediately if encoder is configured
to drop whole superframe.
Bug: webrtc:9950
Change-Id: Iedee9618bb146307efd5a86cb35bf14b5e64b341
Reviewed-on: https://webrtc-review.googlesource.com/c/109002
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25771}
Move HdrMetadata to ColorSpace as part of preparing for joint transmission
of these two objects.
Bug: webrtc:8651
Change-Id: Ie948011a2c0106d5967cb5ef3b9565217e798272
Reviewed-on: https://webrtc-review.googlesource.com/c/111481
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25730}
This increases expected value of maximum buffer level in VP8/9 tests
up to 1 second and thus alignes it with the value that WebRTC uses by
default for these codecs.
Bug: webrtc:10017
Change-Id: I8fd41e8006f11c230d844a053c04656408c2ec97
Reviewed-on: https://webrtc-review.googlesource.com/c/111503
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25716}
Libvpx has been recently updated and this test was failing because
of a slightly different value.
TBR=sprang@webrtc.org
Bug: webrtc:10017
Change-Id: I5fe9161eef5c3e1ff8e0dceb36a663648d8f4617
Reviewed-on: https://webrtc-review.googlesource.com/c/111461
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25709}
Instead of optionally compile VP9 source files based on the value of
the GN argument 'rtc_libvpx_build_vp9', this CL uses the preprocessor
macro RTC_ENABLE_VP9 to decide if VP9 related code needs to be compiled
or not.
Bug: None
Change-Id: I5c1b69d7ec35e8446181d98c912277d0ae8fdba2
Reviewed-on: https://webrtc-review.googlesource.com/c/111063
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25685}
RTC_ENABLE_VP9 is more natural to deal with then RTC_DISABLE_VP9.
In all the places this macro is used, WebRTC needs to do more things
so it is easier to "do more if RTC_ENABLE_VP9 is defined" than
"do more if RTC_DISABLE_VP9 is not defined".
Bug: None
Change-Id: If992e5c554173e6af3f030f6e0fd21bd82acf9eb
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/111242
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25679}
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782
This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.
Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.
One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.
Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
If rate controller is trusted, we disable the frame dropper in the
media optimization module.
This is a re-land of
https://webrtc-review.googlesource.com/c/src/+/105020
Bug: webrtc:9890
Change-Id: I418e47a43a1a98cb2fd5295c03360b954f2288f2
Reviewed-on: https://webrtc-review.googlesource.com/c/109141
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25570}
Will be deleted as soon as downstream calls of
VideoCodingModule::Create are updated.
Tbr: sprang@webrtc.org # Trivial change in video/
Bug: webrtc:3380
Change-Id: Iaeb6da2fb68991225fe9086ddddd4a553e1620b4
Reviewed-on: https://webrtc-review.googlesource.com/c/107890
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25554}
This is a workaround for the case when there are no video frames in a
call for a very long time, such that RTP timestamps wraparound and
FrameBuffer can't figure out if the frame is older or newer.
Bug: webrtc:9974
Change-Id: Ie1eaa4938813dbbd637ddcbe7ff118ead2bfa4a9
Reviewed-on: https://webrtc-review.googlesource.com/c/109882
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25548}
A preparation for deleting EventFactory and EventWrapper, to instead
use rtc::Event directly.
Bug: webrtc:3380
Change-Id: I4c40daca9268e57b06d506d91e09365091c42ad6
Reviewed-on: https://webrtc-review.googlesource.com/c/109880
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25545}
The SetChannelParameters function was used when WebRTC supported decoding
with errors, which we no longer do.
This cleanup CL is related to the work tracked by 9946.
Bug: webrtc:9946
Change-Id: Id2d5ed23031388f890c42651bfbe5f79eda701e5
Reviewed-on: https://webrtc-review.googlesource.com/c/108861
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25505}
This utility class is needed in rtcp_rtp. Instead of reimplementing it
again, the existing class is moved to rtc_base, cleaned from unused
features and extended as required for the new usage.
Bug: webrtc:9914
Change-Id: I3b0d83d08d8fa5e1384b4721a93c6a90781948fd
Reviewed-on: https://webrtc-review.googlesource.com/c/109081
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25498}
Currently we send Nack as soon as we see packets out of order(a skip in packet sequence number). Sometimes this is not necessary because these "missing" packets just late for a couple of millisecond, or these packets can be recovered by FEC. This CL add a field trial parameter to configure a delay before sending Nack.
Bug: webrtc:9953
Change-Id: Ia8f5995d874f7c55a74091bc90d8395b9b88e66b
Reviewed-on: https://webrtc-review.googlesource.com/c/109080
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25488}
The frame time deltas are now capped based on the current noise.
This has been tested in various conditions using both screen content
and typical mobile video settings, to produce delays that are not overly
high screen content, and simultaneously not negatively affect mobile
calls on really bad network that may have high natural jitter.
Bug: webrtc:9898
Change-Id: I51ad279af156aba1b5cc75ae203334a34bce9d48
Reviewed-on: https://webrtc-review.googlesource.com/c/107349
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25469}
This is a cleanup CL related to the work tracked by 9946.
Bug: webrtc:9946
Change-Id: I3d879196af83856ece1418fa786aab03a3dd3c8c
Reviewed-on: https://webrtc-review.googlesource.com/c/108820
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25466}
This is just a cleanup CL related to the work tracked by 9946.
Bug: webrtc:9946
Change-Id: I9a8347aa382bf44f3cd6c38d89bea0e9d68a50e0
Reviewed-on: https://webrtc-review.googlesource.com/c/108781
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25464}
When FlexFEC is enabled, sometimes media packet will be recovered by FEC before the actual media packet's arrival. In current implementation this will be considered as packet out of order and nack will be sent, thus cause large increase in retransmit bitrate.
This fix:
1. Avoid sending nack for packet out of order caused by "early" recovered media packets.
2. Save recovered media packet in a set, and do not send nack for these packets.
Bug: None
Change-Id: I008ef4e33668bce6d2cb9ff52b4b5c8e3f349965
Reviewed-on: https://webrtc-review.googlesource.com/c/108090
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25444}
Remove them from test.
It is completion of the move started with
https://webrtc-review.googlesource.com/c/src/+/107705
Bug: None
Change-Id: Ib0b26db04a1ee814322851280ba1e59b4b3f7ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/107891
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25392}
This deprecates the following methods in VideoEncoder:
virtual ScalingSettings GetScalingSettings() const;
virtual bool SupportsNativeHandle() const;
virtual const char* ImplementationName() const;
Though they are not marked RTC_DEPRECATED since we still want to call
them from within the default GetEncoderInfo() until downstream
projects have been updated.
Furthmore, implementation name is changed from const char* to
std:string, which prevents some lifetime issues with dynamic encoder
names, and CodecSpecificInfo.codec_name is removed in favor of getting
the implementation name via GetEncoderInfo().
This CL removes calls to these deprecated methods, follow-ups will also
remove implementations of the methods and replace them with new
GetEncoderInfo() substitutions.
Bug: webrtc:9890
Change-Id: I6fd6e531480c0b952f53dbd5105e0b0adc3e3b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/106905
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25351}
This will allow downstream projects to use it to construct their own
injected codecs without pulling in dependencies on the software codecs.
Bug: webrtc:7925
Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8
Reviewed-on: https://webrtc-review.googlesource.com/c/107027
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#25297}
This change prevents decoding corruption by not allowing keyframes with a
newer frame id but an older timestamp to be decoded. This does not handle
reordering well.
Bug: none
Change-Id: I4a67ca84ee86a782da74a10530c531d893d3bd3c
Reviewed-on: https://webrtc-review.googlesource.com/c/107304
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25292}
This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into
the video send and receive path. If a FrameEncryptorInterface is set on an outgoing video RTPSender
then each outgoing video frame will first pass through the provided FrameEncryptor which
will have a chance to modify the payload contents for the purposes of encryption. In addition to
this the new GenericFrameDescriptor will be added as additional data.
If a FrameDecryptorInterface is set on an incoming video RtpReceiver then each incoming
video payload will first pass through the provided FrameDecryptor which have a chance to
modify the payload contents for the purpose of decryption.
Bug: webrtc:9795
Change-Id: I9f743ce0cb63df0cf070f6144be7ada078b4e5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/103920
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25258}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
Bug: webrtc:9419
Change-Id: I4d4e2ae52ee01de68147fd0f2cfe4c92d600ad94
Reviewed-on: https://webrtc-review.googlesource.com/c/106343
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25207}
Only used for output filename nowadays. Previously, it was used for
selecting the codec implementation. That is now done by injecting
the appropriate codec factory.
Bug: webrtc:9317
Change-Id: Ia2bf28f7df165fb65410ecd1f5d646ee6604e1be
Reviewed-on: https://webrtc-review.googlesource.com/c/106023
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25204}
