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 reverts commit 9a0662ac7e4a3bc6b3a316397a7fdf25f0025d35.
Reason for revert: breaks some av sync perf tests
Original change's description:
> Use only first payload timestamp for RTCP SR generation for audio
>
> Since now RTP rate is set correctly for audio, there's no need to
> use the very last data packet rtp/capture timestamps for generating
> RTCP SR packets.
>
> Using only one (first) packet timestamp eliminates the jitter between
> rtp and capture timestamps for audio. This jitter comes from the fact
> that capture timestamp for audio is unknown and we generate bogus
> timestamp at arbitrary, non-constant offset from the real capture time.
>
> Bug: webrtc:9905
> Change-Id: I855556184cfe994be39ab7780836a050f5a38c35
> Reviewed-on: https://webrtc-review.googlesource.com/c/108580
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25430}
TBR=danilchap@webrtc.org,ilnik@webrtc.org,ossu@webrtc.org
Change-Id: I208a659379b1075258ee94613e42afd9aebe4754
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9905
Reviewed-on: https://webrtc-review.googlesource.com/c/108623
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25435}
Since now RTP rate is set correctly for audio, there's no need to
use the very last data packet rtp/capture timestamps for generating
RTCP SR packets.
Using only one (first) packet timestamp eliminates the jitter between
rtp and capture timestamps for audio. This jitter comes from the fact
that capture timestamp for audio is unknown and we generate bogus
timestamp at arbitrary, non-constant offset from the real capture time.
Bug: webrtc:9905
Change-Id: I855556184cfe994be39ab7780836a050f5a38c35
Reviewed-on: https://webrtc-review.googlesource.com/c/108580
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25430}
Use helper TimeMicrosToNtp() on clock TimeInMicroseconds()
instead of CurrentNtpTime() and CurrentNtpTimeMillis()
Also update TimeMicrosToNtp() to not introduce fractional in
milliseconds offset. Expose that offset in time_utils.h
Add test showing indended behavior.
Bug: webrtc:9919
Change-Id: I8b019e11ae5b79d0b8ba113a84066b0369cd2575
Reviewed-on: https://webrtc-review.googlesource.com/c/107889
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25391}
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.
bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
deprecated version guarantees (using PostTaskAndReply) callback task will run on the task queue,
and thus doesn't guarantee to run it if task queue is destroyed,
new callback versions instead guarantee callback will always run,
but may run off the task queue if task queue is destroyed.
Both keep guarantee observer callbacks will not run after on_destroyed/on_removed is called.
Bug: None
Change-Id: I61bf52127f3084c0186aa8bc89037bf9296801d8
Reviewed-on: https://webrtc-review.googlesource.com/c/107305
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25301}
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}
Assumption extra needed bytes for single packet needs is sum
of extra bytes for first and last packet
moved up to RTPSenderVideo from individual packetizers.
There it can be fixed.
Bug: webrtc:9868
Change-Id: I24c80ffa5c174afd3fe3e92fa86ef75560bb961e
Reviewed-on: https://webrtc-review.googlesource.com/c/105662
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25160}
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}
Automatic detection if one-byte header or two-byte header should be used based
on extension ID and extension length.
Bug: webrtc:7990
Change-Id: I9fc848ecc59458d1ca97bace0e57ea04d3d0ced6
Reviewed-on: https://webrtc-review.googlesource.com/c/103422
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25004}
If layers have been enabled or disabled, send immediate instead of on
next available report.
Bug: webrtc:9823
Change-Id: Ifd774641d4b8c03a9efa8ad48ff5e88328ed2ba9
Reviewed-on: https://webrtc-review.googlesource.com/c/103802
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24997}
The RtpReceiver class is no longer used. Together with it, delete
RTPPayloadRegistry, RtpReceiverStrategy, and the tests under
modules/rtp_rtcp/test/testAPI/.
Bug: webrtc:8995
Change-Id: Ia9924d2f0f4315914a0dce6b7375ebb3601a6f96
Reviewed-on: https://webrtc-review.googlesource.com/c/103503
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24968}
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 CL makes the RtpGenericFrameDescriptor available in
RTPSenderVideo::SendVideo for encryption and in
RtpVideoStreamReceiver::OnReceivedFrame for decryption.
Bug: webrtc:9361
Change-Id: I5b6d10138c0874657862f103c8c9a2328e6d4a66
Reviewed-on: https://webrtc-review.googlesource.com/102720
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24929}
to extract byte representation of a built extension without rebuilding it.
Bug: webrtc:9361
Change-Id: I5e2a5caeb8ff28dcb58dc25d53407c449c86df44
Reviewed-on: https://webrtc-review.googlesource.com/102940
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24925}
Add new method OnRtpPacket, but leave
ReceiveStatisticsImpl::IncomingPacket and most of the implementation
unchanged. Deleting the old method and converting implementation from
RTPHeader to RtpPacketreceived is planned for a followup, after
downstream code is updated.
Bug: webrtc:7135, webrtc:8016
Change-Id: I697ec12804618859f8d69415622d1b957e1d0847
Reviewed-on: https://webrtc-review.googlesource.com/100104
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24889}
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}
A potential bug was introduced in "Refactor to remove direct memory
dependency on kMaxId" due to a memory restructuring,
commit c5744b8b21b627213286f1b6f2c65da5df9ce8d0
Bug: webrtc:7990
Change-Id: I0dcaf47e1c1e361d65220c278a2326d6f2686af7
Reviewed-on: https://webrtc-review.googlesource.com/101642
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24814}
