This CL uses |width| and |height| in RTPVideoHeaderVP9 to pass information
about enabled layers from encoder to packetizer.
Bug: webrtc:11319
Change-Id: Idc1c337f8dfb3f7631506acb784d2a634b41b955
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167724
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30428}
To free up RtpVideoHeader::generic field for codec agnostic details
from an rtp header extension.
Bug: webrtc:10342
Change-Id: I7b9d869b2ecfedb96dfd860be47ed8dffa058749
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166175
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30396}
The extension (and thus structures to carry it) are designed
in particular for client<->SFU link. Putting the structure into api
acknowledges it can be reused by SFU projects
Bug: webrtc:10342
Change-Id: I8ca1f5046abadf6aa16200443c4892e9a2a928b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166467
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30324}
to allow writing DependencyDescriptor value that is not copiable.
and avoid copying RtpGenericFrameDescriptor
Bug: webrtc:10342
Change-Id: I6eefa9d06b90d7e858f224443ba6769975b556fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166171
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30322}
These constants describes how value should be put on the wire and thus
belong to the extension builder/writer class rather than extension value class
Bug: None
Change-Id: I65ca3923eddcc2e48563ad69b98356c159ad86be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166461
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30305}
The idea to communicate spatial dependencies with spatial layers bitmask
wasn't fully implemented and was dropped in later version of the descriptor.
Bug: webrtc:10342
Change-Id: I1ed191c3a2a9d2e1e9ddf313f781ca8257c34dfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166165
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30278}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
If for some reason capture timestamp is unset, the default value of 0 would be
passed to RtcpSender. This will cause rtp timestamps to grow at double the rate
in Sender Reports because it has time since the last frame capture as a term.
Bug: none
Change-Id: I2fe09dabef6b0957fb504deaa06393dedc4a9e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162481
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30105}
Some clients will not count audio packets into the bandwidth estimate
despite negotiating e.g. abs-send-time for that SSRC.
If padding is sent on such an RTP module, we might get stuck in a low
resolution.
This CL works around that by preferring to send padding on video SSRCs.
Bug: webrtc:11196
Change-Id: I1ff503a31a85bc32315006a4f15f8b08e5d4e883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161941
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30066}
As of https://webrtc-review.googlesource.com/c/src/+/158899, FEC may be
used on packets with VideoTimingExtension. This may result in creation
of FEC packets that exceed the maximum configured RTP packet size.
This problem occurs most frequently with datagram transports that define a
smaller maximum packet size.
Bug: webrtc:9719
Change-Id: I842216a6696a695f0a3f01a221e538605fc5b9bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161557
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30045}
while suboptimal, these implementions are complete and allow to
swap code from using RtpDepacketizer interface to VideoRtpDepacketizer
Bug: webrtc:11152
Change-Id: Ie7823feeb5b0563b74754255aaedfada9d446ac5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161380
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30031}
This allows one to request the same sequence number again
in the case of resending an FIR to the a sender before the
sender has time to send a key-frame.
Bug: webrtc:11171
Change-Id: Idd8e8120ccbcc194cefb8d0cf3f7cc64e7f76aa5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161236
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30006}
instead of guessing based on presence of the sequence header OBU.
Bug: webrtc:11042
Change-Id: I9a0674249feceebb07299ea965c62e87499f6baf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161013
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29958}
This part of the effort to implement A/V sync metric.
Bug: webrtc:10739
Change-Id: I4adba1b99b37b31868168e37d9aa8e03f8ea6d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159886
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ruslan Burakov <kuddai@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29849}
* Removes legacy defines from rtp_rtcp_defines.
* Simplifies the feedback adaptation logic, this is achieved
by using the ability to preserve lost packets information
from the RTCP message.
* Extracts in flight data tracking to a separate helper class.
* Removes legacy fields and constructors from the PacketFeedback
structure.
* Removes the legacy GetTransportFeedbackVector method.
Apart from reducing total LOC, this prepares for moving the adaptation
to run on a TaskQueue.
Bug: webrtc:9883
Change-Id: I5ef4eace0948f119f283cd71dc2b8d0954a1449b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158781
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29674}
