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}
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}
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 is a reland of 171df9326200d1e01bce530e2ff01ac5890e6cb7
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}
This reverts commit 171df9326200d1e01bce530e2ff01ac5890e6cb7.
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}
That is, the payload packetization overhead (eg. vp8 payload header),
not the RTP headers, extensions, etc.
The encoder and pacer both look at payload rate, but are currently not
aware of the bytes that are added in between them.
Bug: webrtc:10155
Change-Id: I4cdb04849d762360374d47a496983c8c6df191d2
Reviewed-on: https://webrtc-review.googlesource.com/c/115410
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26163}
Rids can now be sent using rtp_sender.
Hooking up the rid values in the voice and video engine is still WIP.
Bug: webrtc:10074
Change-Id: I245c7ecb23b67fc0ba65caaa5dbb4fcfd60c81bb
Reviewed-on: https://webrtc-review.googlesource.com/c/114505
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26092}
before this CL it was only configured when pacer is used.
This CL sets it also when pacer is not used.
Move block for setting TransmissionOffset/AbsoluteTime extensions after pacer_ check
to stress in pacer case there are set(overwritten) in another function.
Bug: None
Change-Id: I06a6dd6ec689a25439a75b3baa71340535cd1ff8
Reviewed-on: https://webrtc-review.googlesource.com/c/112126
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25794}
Delete the deprecated IncomingPacket method, and convert implementation
to use RtpPacketReceived rather than RTPHeader.
Part 1 was https://webrtc-review.googlesource.com/c/src/+/100104
Bug: webrtc:7135, webrtc:8016
Change-Id: Ib4840d947870403deea2f9067f847e4b0f182479
Reviewed-on: https://webrtc-review.googlesource.com/c/6762
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25648}
The replacements are absl::EqualsIgnoreCase and
absl::StartsWithIgnoreCase. Also delete the alias
RtpUtility::StringCompare.
Bug: webrtc:6424
Change-Id: I4bed71540264450f85137ad0c2564125c5c6213f
Reviewed-on: https://webrtc-review.googlesource.com/c/109006
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25481}
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}
Pacer may accept same packet serveral time for resending,
packet may spend non-zero time in pacer queue.
As a result packet can be resend several time within one rtt
wasting bandwidth.
Bug: None
Change-Id: I753a5400b47d3804735e66e539a1b103916d0c94
Reviewed-on: https://webrtc-review.googlesource.com/c/106260
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25205}
Streams that are part of transport feedback are assumed to be part of
allocation. A SetAsPartOfAllocation method is introduced to be used by
media streams that are part of bitrate allocation but not included in
feedback.
This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.
Bug: webrtc:9796
Change-Id: If7ac1ad3e6f3c28b2ada2aef1607a42689d899b2
Reviewed-on: https://webrtc-review.googlesource.com/c/104881
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25079}
This CL moves the action of acquiring the lock outside
UpdateTransportSequenceNumber. This prepares for an upcoming CL where
the lock is used outside this call at the call sites and avoids the lock-unlock
overhead that would otherwise occur.
Also removing the const declaration as it modifies the state of
transport_sequence_number_allocator_.
Bug: webrtc:9796
Change-Id: I0bd4a0fd2fdbf6291867eb913690c61269eab8c5
Reviewed-on: https://webrtc-review.googlesource.com/c/102684
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25068}
Rename to better match what it does,
Adjust to support two-byte header extension
Bug: webrtc:7990
Change-Id: I2786d70e7cf9cd3d722f54fb1d07c9cfaafab947
Reviewed-on: https://webrtc-review.googlesource.com/103201
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24958}
This reverts commit 8b7bc5d7010c84ac57459518fe18309ef5fee1dd.
Reason for revert: Slow RTC_DCHECK has been removed.
Original change's description:
> Revert "Second reland of "Optimize execution time of RTPSender::UpdateDelayStatistics""
>
> This reverts commit 9def3b45ef06de9e068e8f4d1644e9d508baa913.
>
> Reason for revert: webrtc_perf_tests fails on Mac-10.12.
>
> Original change's description:
> > Second reland of "Optimize execution time of RTPSender::UpdateDelayStatistics"
> >
> > The reland has a lot of additional DCHECKS for easier debugging,
> > so in debug builds it will actually be a ~2x slowdown compared to the old code.
> > The excessive DCHECKS should be removed in a followup CL.
> >
> > Bug: webrtc:9439
> > Change-Id: I8389cd84f1ca12c29cc6993f0d2cf7e6d7dd8379
> > Reviewed-on: https://webrtc-review.googlesource.com/101761
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24821}
>
> TBR=terelius@webrtc.org,asapersson@webrtc.org,kron@webrtc.org
>
> Change-Id: I98c4c96d552858d0299d49993e9b9be6a6204dfe
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9439
> Reviewed-on: https://webrtc-review.googlesource.com/101860
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24825}
TBR=terelius@webrtc.org,asapersson@webrtc.org,kron@webrtc.org
Change-Id: I260c56932710d26f9d7201c07279fef8d2150bd9
Bug: webrtc:9439
Reviewed-on: https://webrtc-review.googlesource.com/102000
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24843}
This reverts commit 3f4a4fad8cd661309ff5d9a631e89518f32e7c5e.
Reason for revert: Breaking internal tests
Original change's description:
> Added field trial WebRTC-GenericDescriptor for the new generic descriptor.
>
> Also parameterized tests to test the new generic descriptor and
> added --generic_descriptor flag to loopback tests.
>
> Bug: webrtc:9361
> Change-Id: I2b76889606fe2e81249ecdebb0b7b61151682485
> Reviewed-on: https://webrtc-review.googlesource.com/101900
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24835}
TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org
Change-Id: I4d4714a9f4ab0e95adf0f4130bc1a932efc448fa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9361
Reviewed-on: https://webrtc-review.googlesource.com/101940
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24839}
Also parameterized tests to test the new generic descriptor and
added --generic_descriptor flag to loopback tests.
Bug: webrtc:9361
Change-Id: I2b76889606fe2e81249ecdebb0b7b61151682485
Reviewed-on: https://webrtc-review.googlesource.com/101900
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24835}
This reverts commit 9def3b45ef06de9e068e8f4d1644e9d508baa913.
Reason for revert: webrtc_perf_tests fails on Mac-10.12.
Original change's description:
> Second reland of "Optimize execution time of RTPSender::UpdateDelayStatistics"
>
> The reland has a lot of additional DCHECKS for easier debugging,
> so in debug builds it will actually be a ~2x slowdown compared to the old code.
> The excessive DCHECKS should be removed in a followup CL.
>
> Bug: webrtc:9439
> Change-Id: I8389cd84f1ca12c29cc6993f0d2cf7e6d7dd8379
> Reviewed-on: https://webrtc-review.googlesource.com/101761
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24821}
TBR=terelius@webrtc.org,asapersson@webrtc.org,kron@webrtc.org
Change-Id: I98c4c96d552858d0299d49993e9b9be6a6204dfe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9439
Reviewed-on: https://webrtc-review.googlesource.com/101860
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24825}
The reland has a lot of additional DCHECKS for easier debugging,
so in debug builds it will actually be a ~2x slowdown compared to the old code.
The excessive DCHECKS should be removed in a followup CL.
Bug: webrtc:9439
Change-Id: I8389cd84f1ca12c29cc6993f0d2cf7e6d7dd8379
Reviewed-on: https://webrtc-review.googlesource.com/101761
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24821}
RtpPacket::UpdateDelayStatistics was previously optimized with several
sanity checks added. These sanity checks caused many of the unit tests
in peerconnection_integration_unittests to fail and the CL was therefore
reverted. This CL contains the sanity checks along with patches so that
the unit tests pass.
Bug: webrtc:9439
Change-Id: Ia5f5e8b125e5f3f4b79d433e2282901143530a25
Reviewed-on: https://webrtc-review.googlesource.com/99802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24813}
Remove function for converting uri into ExtensionType
This removes one of the lists of all supported extensions
Bug: webrtc:7472
Change-Id: I0c27239d91ef14ca4a3aa0c00588fa2b9cf10e0c
Reviewed-on: https://webrtc-review.googlesource.com/100523
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24752}
as iniditcation retransmission shouldn't be limited because of rate.
Bug: None
Change-Id: I579261749515260b972631779dadc6349dfcab46
Reviewed-on: https://webrtc-review.googlesource.com/99541
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24690}
This reverts commit 7bcd2a98be3fa8c246866d6b343c7f94752977b3.
Reason for revert: peerconnection_unittests fails on downstream test runner.
Original change's description:
> Reland "Optimize execution time of RTPSender::UpdateDelayStatistics"
>
> The reland has a lot of additional DCHECKS for easier debugging,
> so in debug builds it will actually be a ~2x slowdown compared to the old code.
> The excessive DCHECKS should be removed in a followup CL.
>
> Bug: webrtc:9439
> Change-Id: I493de337bf20c998aa32c2532212cac85c5517fb
> Reviewed-on: https://webrtc-review.googlesource.com/96641
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24501}
TBR=terelius@webrtc.org,asapersson@webrtc.org,philipel@webrtc.org
Change-Id: Ia48444d2a7647cf826ef93b4720f6d7ff9a712c3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9439
Reviewed-on: https://webrtc-review.googlesource.com/96960
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24502}
The reland has a lot of additional DCHECKS for easier debugging,
so in debug builds it will actually be a ~2x slowdown compared to the old code.
The excessive DCHECKS should be removed in a followup CL.
Bug: webrtc:9439
Change-Id: I493de337bf20c998aa32c2532212cac85c5517fb
Reviewed-on: https://webrtc-review.googlesource.com/96641
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24501}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated using script:
#!/bin/bash
dir=modules/rtp_rtcp
find $dir -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $dir -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc
Reviewed-on: https://webrtc-review.googlesource.com/83584
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23624}
