to clearly signal passed ownership.
Drop support for accepting nullptr clock to avoid copying the Configuration structure.
Update all calls in webrtc to the new factory function
Bug: None
Change-Id: Ic5a78da8e59ba3988a757a9d9634fa31499ce0db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125901
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26994}
The initial implementation forced the sender to use different sizes
of the RTP header extension depending on if a feedback request is
included or not. This can be a problem if the RTP header is pre-
allocated.
This CL changes this so that a static size of 4 bytes can be used
for the TransportSequenceNumberV2 RTP header extension. The change
in the protocol to get this to work is that
FeedbackRequest::sequence_count == 0 means that no feedback is
requested, and FeedbackRequest::sequence_count == 1 means that
feedback is requested for the current packet only.
Bug: webrtc:10262
Change-Id: Ia5134b3daf49f8a5b89f6c717894f6e055f39c8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125420
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26985}
The idea is to let the RtpRtcp and RTPSender classes be responsible for
media-agnostic RTP transport, and move out the media-specific processing,
such as packetization and media-specific headers.
Bug: webrtc:7135
Change-Id: Ib0ce45bf06713b3eb6c06acd91c5168856874e4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123187
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26954}
This CL applies clang-tidy's bugprone-argument-comment [1] to the
WebRTC codebase.
All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.
[1] - https://clang.llvm.org/extra/clang-tidy/checks/bugprone-argument-comment.html
Bug: webrtc:10252
Change-Id: I77fec17509311275f18e730e482fb9f3fb2998ad
Reviewed-on: https://webrtc-review.googlesource.com/c/124989
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26900}
* LossNotificationController is the class that decides when to issue
LossNotification RTCP messages.
* RtpRtcp handles the technicalities of producing RTCP messages.
Bug: webrtc:10336
Change-Id: I292536257a984ca85d21d9cfa38e7ff2569cbb39
Reviewed-on: https://webrtc-review.googlesource.com/c/124123
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26840}
The FMT 15 is not specific only to REMB or loss notification messages.
Rather, it is the Application Layer FB (AFB) of Psfb (Payload Specific
Feedback Messages).
See https://tools.ietf.org/html/rfc4585#section-6.3TBR=terelius@webrtc.org
Bug: webrtc:10336
Change-Id: I8cd27ef9ee044bf7b7e7c1bd1a53c1dae2d95006
Reviewed-on: https://webrtc-review.googlesource.com/c/123886
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26827}
Replaces use of field trials in RtpSender and RtpVideoSender with injectable WebRtcKeyValueConfig.
Implementation still defaults to field trials.
BUG: webrtc:10335
Change-Id: I5fc6d327ee5d011ccc41385734b38344df172627
Reviewed-on: https://webrtc-review.googlesource.com/c/123447
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26795}
This cl change RtpSender to feed the PacedSender with RTP packet size rather than payload size in experiment WebRTC-SendSideBwe-WithOverhead. Before this cl, the congestion controller was feed with packet size but not the pacer. That means that the pacer budget was updated with an estimate that includes the RTP headers, but the media budget only use the payloads.
BUG: webrtc:10325 webrtc:6762
Change-Id: I35c8350603a7881ea162debcd89ed901cbb50950
Reviewed-on: https://webrtc-review.googlesource.com/c/123444
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26788}
To reflect what this value actually contain.
BUG: webrtc:10325
Change-Id: Ic3c5efbd16157bfae1a2749df17051f105720997
Reviewed-on: https://webrtc-review.googlesource.com/c/123500
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26787}
The discardability flag denotes whether the frame may be dropped by
the decoder with no effect on the decodability of subsequent frames.
Bug: webrtc:10214
Change-Id: I3654951d8863b50effe9670b8d1d7eb051240039
Reviewed-on: https://webrtc-review.googlesource.com/c/122241
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26763}
This reverts commit 4f36b7a478.
Reason for revert: Failing tests fixed.
Original change's description:
> Revert "Delete test/constants.h"
>
> This reverts commit 389b1672a3.
>
> Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate
>
> Original change's description:
> > Delete test/constants.h
> >
> > It's not possible to use constants.h for all RTP extensions
> > after the number of extensions exceeds 14, which is the maximum
> > number of one-byte RTP extensions. This is because some extensions
> > would have to be assigned a number greater than 14, even if the
> > test only involves 14 extensions or less.
> >
> > For uniformity's sake, this CL also edits some files to use an
> > enum as the files involved in this CL, rather than free-floating
> > const-ints.
> >
> > Bug: webrtc:10288
> > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> > Commit-Queue: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26728}
>
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
>
> Bug: webrtc:10288, chromium:933127
> Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/123381
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26744}
TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954
Bug: webrtc:10288, chromium:933127
Reviewed-on: https://webrtc-review.googlesource.com/c/123384
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26750}
This reverts commit 389b1672a3.
Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate
Original change's description:
> Delete test/constants.h
>
> It's not possible to use constants.h for all RTP extensions
> after the number of extensions exceeds 14, which is the maximum
> number of one-byte RTP extensions. This is because some extensions
> would have to be assigned a number greater than 14, even if the
> test only involves 14 extensions or less.
>
> For uniformity's sake, this CL also edits some files to use an
> enum as the files involved in this CL, rather than free-floating
> const-ints.
>
> Bug: webrtc:10288
> Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26728}
TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
No-Presubmit: True
Bug: webrtc:10288, chromium:933127
Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
Reviewed-on: https://webrtc-review.googlesource.com/c/123381
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26744}
It's not possible to use constants.h for all RTP extensions
after the number of extensions exceeds 14, which is the maximum
number of one-byte RTP extensions. This is because some extensions
would have to be assigned a number greater than 14, even if the
test only involves 14 extensions or less.
For uniformity's sake, this CL also edits some files to use an
enum as the files involved in this CL, rather than free-floating
const-ints.
Bug: webrtc:10288
Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
Reviewed-on: https://webrtc-review.googlesource.com/c/123048
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26728}
Set padding bit if the TransportFeedback packet contains zero padding.
Also write number of padding elements at the last position of the packet.
Bug: webrtc:10263
Change-Id: I8d17bc0e889f658ac383dec64ddcb95d438c9702
Reviewed-on: https://webrtc-review.googlesource.com/c/122240
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26646}
rtx packet may have addition extension (mid) and may use different
header size for extension (e.g. if repaired rtp stream id registered
to larger id than rtp stream id)
As a result rtx packet size calculation as orginial size + 2 bytes in
some scenarious may be incorrect. This chenage avoids crash in that cases.
Bug: None
Change-Id: I620d95e0592d6bdac0d3623b2675a49fc2177580
Reviewed-on: https://webrtc-review.googlesource.com/c/122180
Reviewed-by: Erik Varga <erikvarga@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26635}
We are about to split RtpGenericFrameDescriptorExtension
into v00 and v01. Allow downstream projects to refer to
RtpGenericFrameDescriptorExtension00 now, so that we may later
delete references to RtpGenericFrameDescriptorExtension
without breaking their build.
Bug: webrtc:10214
Change-Id: I45528699bf7d8cc6c22c22a601f248cca2ba6c90
Reviewed-on: https://webrtc-review.googlesource.com/c/121769
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26588}
Also refactor most of the RtpSender tests to not use the frame-level
method RTPSender::SendOutgoingData.
Bug: webrtc:7135
Change-Id: I9b0af6aa45e9b908d8197e48b143fede7e2804c7
Reviewed-on: https://webrtc-review.googlesource.com/c/121461
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26577}
Configurability via SetSelectiveRetransmissions was deleted in
https://webrtc-review.googlesource.com/c/119920.
Delete constants kRetransmitFECPackets and kRetransmitAllPackets,
which are never enabled in production code. Also move the declaration
of RetransmissionMode from rtp_rtcp_defines.h to rtp_sender_video.h,
to reduce visibility to applications.
Bug: None
Change-Id: I70dcf7532aa3415a2449d8d807c500c1f149bf6d
Reviewed-on: https://webrtc-review.googlesource.com/c/120053
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26570}
Update RtpSenderAudioTest to call methods on RTPSenderAudio rather
than RTPSender, when possible. In particular, avoid
RTPSender::SendOutgoingData. Drop parameterization on the
WebRTC-SendSideBwe-WithOverhead field trial, since that appears
unrelated to these tests.
Also delete some unused parts of the RtpSender test.
Bug: webrtc:7135
Change-Id: I535bf48bb1720e2727f4a62fa3e49b2bb84394a0
Reviewed-on: https://webrtc-review.googlesource.com/c/120920
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26516}
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
Moved from RtpSender to RtpSenderVideo, since currently the
PlayoutDelay extension is used for video only, and configured via
RTPVideoHeader.
Bug: webrtc:7135
Change-Id: Idfcc90cefea83e40a4e79164d7914cdcd50e41fe
Reviewed-on: https://webrtc-review.googlesource.com/c/120357
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26484}
There's now one const method PlayoutDelayToSend to produce the delay
values to insert into outgoing packets, and two update methods,
OnSentPacket, and OnReceivedAck, to observe outgoing packets and acks,
respectively.
Bug: webrtc:7135
Change-Id: I07498c30f0de87ae0113f7e2eb6357a091a1f0af
Reviewed-on: https://webrtc-review.googlesource.com/c/120603
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26474}
This prepares for making the Clock interface fully mutable.
Calls to the time functions in Clock can have side effects in some
circumstances. It's also questionable if it's a good idea to allow
repeated calls to a const method return different values without
any changed to the class instance.
Bug: webrtc:9883
Change-Id: I96fb9230705f7c80a4c0702132fd9dc73899fc5e
Reviewed-on: https://webrtc-review.googlesource.com/c/120347
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26467}
This CL applies clang-tidy's performance-for-range-copy [1] on the
WebRTC codebase.
All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.
[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-for-range-copy.html
Bug: webrtc:10215
Change-Id: I7c83290b8866d76129bbec4e24e6701f5014102e
Reviewed-on: https://webrtc-review.googlesource.com/c/120043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26420}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
This is a reland of 171df93262
Original change's description:
> Delete RtpUtility::Payload, and refactor RTPSender to not use it
>
> Replaced by a payload type --> video codec map in RTPSenderVideo,
> where it is used to select the right packetizer.
>
> Bug: webrtc:6883
> Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
> Reviewed-on: https://webrtc-review.googlesource.com/c/119263
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26380}
Tbr: danilchap@webrtc.org
Bug: webrtc:6883
Change-Id: I30771b86bbe50de609353e23e80dc532dc884ad4
Reviewed-on: https://webrtc-review.googlesource.com/c/119661
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26394}
Create a new RTCP feedback message for reporting the loss and/or non-decodability of video frames, to be used by the upcoming injectable VideoFrameBufferController. The new feedback message should report:
1. The sequence number of the last decoded non-discardable video frame. (TBD: If a multi-packet frame, should it be the sequence number of the first, last, or any of the packets?)
2. The sequence number of the last received RTP packet in the stream.
3. A decodability flag, whose specific meaning depends on the last-received
RTP sequence number. The decodability flag is true if and only if all of
the frame's dependencies are known to be decodable, and the frame itself
is not yet known to be unassemblable.
* Clarification #1: In a multi-packet frame, the first packet's
dependencies are known, but it is not yet known whether all parts
of the current frame will be received.
* Clarification #2: In a multi-packet frame, the dependencies would be
unknown if the first packet was not received. Then, the packet will
be known-unassemblable.
Bug: webrtc:10226
Change-Id: I1563c944477e3ed40235e82ab99a439414632aff
Reviewed-on: https://webrtc-review.googlesource.com/c/118931
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26387}
This reverts commit 171df93262.
Reason for revert: Breaks downstream project
Original change's description:
> Delete RtpUtility::Payload, and refactor RTPSender to not use it
>
> Replaced by a payload type --> video codec map in RTPSenderVideo,
> where it is used to select the right packetizer.
>
> Bug: webrtc:6883
> Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
> Reviewed-on: https://webrtc-review.googlesource.com/c/119263
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26380}
TBR=danilchap@webrtc.org,brandtr@webrtc.org,nisse@webrtc.org
Change-Id: I76489c29541827aaba72515a76db54bdb7495e28
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6883
Reviewed-on: https://webrtc-review.googlesource.com/c/119640
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26385}
Replaced by a payload type --> video codec map in RTPSenderVideo,
where it is used to select the right packetizer.
Bug: webrtc:6883
Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
Reviewed-on: https://webrtc-review.googlesource.com/c/119263
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26380}
These methods should only be used when parsing frames produced
by an older client; newer clients should not attempt to set
these values.
(When talking to older clients, TRUE is hard-coded. When talking
to newer clients, these flags are deprecated.)
Bug: webrtc:10214
Change-Id: I8537869ef3112f4ce9531c6becc33951715685a1
Reviewed-on: https://webrtc-review.googlesource.com/c/118421
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26360}
Several PSFB messages might be supported, distinguished using
the unique identifier. If the unique identifier is not REMB, it's
not an error, and so a warning should not be issued.
Bug: webrtc:10226
Change-Id: I5e79b473bd54cf0964f19329efb33354f63f5d5e
Reviewed-on: https://webrtc-review.googlesource.com/c/118686
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26359}
In preparation for adding a discardability flag in
RtpGenericFrameDescriptor, deprecate two bits which are always
in practice set to TRUE.
This is conceptual deprecation. RTC_DEPRECATED cannot actually be
applied, because we still want to be able to parse those bits
and make sure they are truly set to TRUE when TRUE is expected.
Bug: webrtc:10214
Change-Id: I7d6cb640fe27f142578883389cc67d326c90f7bb
Reviewed-on: https://webrtc-review.googlesource.com/c/118381
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26340}
