Commit graph

38 commits

Author SHA1 Message Date
Danil Chapovalov
18d1d0f793 Fix perfect forwarding in RtpPacket::GetExtension
Thus allow to pass output parameter by reference.

Bug: None
Change-Id: I64821caf72875efee62d6cfc90691070dceba775
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334644
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41542}
2024-01-17 11:16:49 +00:00
Danil Chapovalov
7b42f35bcc Remove artifical extra RTP packet capacity
Instead allow RtpPacket to exceed configured capacity when setting payload

Bug: None
Change-Id: I02fc080ffa3127ffbe0dade1f200dd7456a6a128
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40471}
2023-07-25 06:33:09 +00:00
Danil Chapovalov
885ededbb8 Add move constructor and assign operator to RtpPacket
RtpPacket has CopyOnWriteBuffer and std::vector that can be moved more
efficiently than copied, thus move of the RtpPacket is also more efficient

Bug: None
Change-Id: I5509346e426cd32d0fb0649ef1a6883b7176df1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290726
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39053}
2023-01-10 11:12:45 +00:00
philipel
3bb6f6d4e8 Add RtpPacket::SetRawExtension function.
Bug: webrtc:14801
Change-Id: I1ce9361250a7ad2d932ee9ae5b8f93415d0ea7b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289980
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38974}
2023-01-02 16:18:16 +00:00
Per K
5e5d017c2b Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
Instead of getting header extension mapping from a receiver object, get the mapping from the received packet.

The purpose is to be able to remove extension information from webrtc/call/receive_stream.h.
Header extensions are negotiated per mid, not per receive stream.
The goal is to reduce the number of places where packets are parsed and demuxed.

Bug: webrtc:7135, webrtc:14795
Change-Id: I8944bc06a11dc572d9e14e7d7ee446a841096295
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288968
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38944}
2022-12-22 14:04:21 +00:00
Artem Titov
913cfa76ec Use backticks not vertical bars to denote variables in comments for /modules/rtp_rtcp
Bug: webrtc:12338
Change-Id: I52eb3b6675c4705e22f51b70799ed6139a3b46bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227164
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34686}
2021-08-09 15:51:03 +00:00
Danil Chapovalov
44450a073b Support header only parsing by RtpPacket
It is not uncommon to save rtp header of an rtp packet for later parsing
(e.g. rtc event log does that)
Such header is invalid as an rtp packet when padding bit is set.
This change suggest to treat header only packets with padding as valid.

Bug: webrtc:5261
Change-Id: If61d84fc37383d2e9cfaf9b618276983d334702e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225265
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34438}
2021-07-08 14:43:28 +00:00
Danil Chapovalov
7d5418233d Avoid assembling complicated but unused video rtp header extensions
Bug: chromium:1219407
Change-Id: I017de10813a1e80f4af0ba55d8d1aa73077dd131
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222615
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34326}
2021-06-17 16:09:13 +00:00
Tommi
d350006b70 Add rtp_config() accessor to ReceiveStream.
This is a consistent way to get to common config parameters for
all receive streams and avoids storing a copy of the extension
headers inside of Call. This is needed to get rid of the need of
keeping config and copies in sync, which currently is part of why
we repeatedly delete and recreate audio receive streams on config
changes.

Bug: webrtc:11993
Change-Id: Ia356b6cac1425c8c6766abd2e52fdeb73c4a4b4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222040
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34285}
2021-06-14 17:57:57 +00:00
Danil Chapovalov
e15dc58f32 Use rtc::CopyOnWriteBuffer::MutableData through webrtc
where mutable access is required.

Bug: webrtc:12334
Change-Id: I4b2b74f836aaf7f12278c3569d0d49936297716b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32936}
2021-01-11 11:31:33 +00:00
Henrik Grunell
c463a784c3 Clarification of RtpPacket constructor in comment.
See also b/175210069 for more context.

Bug: None
Change-Id: I06e9848028c0f11362db373af54b42cbc67aee77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198780
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32874}
2020-12-22 14:39:13 +00:00
Danil Chapovalov
cea929923b in RtpPacket packet pass rtp header extension value by const&
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}
2020-01-20 13:37:01 +00:00
Danil Chapovalov
629de6f7ed Merge RtpPacket HasExtension and IsExtensionReserved functions
RtpPacket doesn't keep difference between reserved and set extension.

Bug: None
Change-Id: I1c79f4ebd7ba20ae5da0194c3faa418050db7d8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30316}
2020-01-20 11:37:25 +00:00
Danil Chapovalov
a3ecb7a656 Migrate tests from RtpDepacketizer to VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: I1b1c5183d35b791c09c14c9d1f0ca240c1749d9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161455
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30055}
2019-12-10 17:37:46 +00:00
Ilya Nikolaevskiy
a5d952f4be Reland "Refactor FEC code to use COW buffers"
Reland with fixes for fuzzer found crashes.

This refactoring helps to reduce unnecessary memcpy calls on the receive side.

This CL replaces |uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| in Packet class, removes |length| field there, and does necessary changes.

Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145332

Bug: webrtc:10750
Change-Id: I6775a701bcb2ae25ec1666e1db90041cd49013b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151131
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29116}
2019-09-09 16:20:33 +00:00
Ilya Nikolaevskiy
082696efd9 Revert "Refactor FEC code to use COW buffers"
This reverts commit eec5fff4df.

Reason for revert: Some crashes found by the fuzzer

Original change's description:
> Refactor FEC code to use COW buffers
> 
> This refactoring helps to reduce unnecessary memcpy calls on the receive
> side.
> 
> This CL replaces
> |uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| in Packet class,
> removes |length| field there, and does necessary changes.
> 
> This is a reland of these two CLs with fixes:
> https://webrtc-review.googlesource.com/c/src/+/144942
> https://webrtc-review.googlesource.com/c/src/+/144881
> 
> Bug: webrtc:10750
> Change-Id: I76f6dee5a57ade59942ea2822ca4737edfe6438b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145332
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29035}

TBR=brandtr@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org

Change-Id: Id3d65fb1324b9f1b0446fe217012115ecacf2b40
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10750
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151130
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29043}
2019-09-03 07:53:05 +00:00
Ilya Nikolaevskiy
eec5fff4df Refactor FEC code to use COW buffers
This refactoring helps to reduce unnecessary memcpy calls on the receive
side.

This CL replaces
|uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| in Packet class,
removes |length| field there, and does necessary changes.

This is a reland of these two CLs with fixes:
https://webrtc-review.googlesource.com/c/src/+/144942
https://webrtc-review.googlesource.com/c/src/+/144881

Bug: webrtc:10750
Change-Id: I76f6dee5a57ade59942ea2822ca4737edfe6438b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145332
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29035}
2019-09-02 12:28:37 +00:00
Qingsi Wang
6ff9ebd070 Revert "Refactor FEC code to use COW buffers"
This reverts commit 7325bc3917.

Reason for revert: FecTest.UlpfecTest is consistently failing.

Original change's description:
> Refactor FEC code to use COW buffers
> 
> This refactoring helps to reduce unnecessary memcpy calls on the receive
> side.
> 
> This CL is the first stage of refactoring: it only replaces
> |uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| and does
> necessary changes.
> 
> A follow-up CL will remove length field of the Packet class.
> 
> 
> Bug: webrtc:10750
> Change-Id: Ie233da83ff33f6370f511955e4c65d59522389a7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144881
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28539}

TBR=brandtr@webrtc.org,ilnik@webrtc.org,asapersson@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org

Change-Id: I07c34256a76174f09a0d27eacbae6488e66f4b43
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10750
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145340
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28545}
2019-07-11 19:55:28 +00:00
Ilya Nikolaevskiy
7325bc3917 Refactor FEC code to use COW buffers
This refactoring helps to reduce unnecessary memcpy calls on the receive
side.

This CL is the first stage of refactoring: it only replaces
|uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| and does
necessary changes.

A follow-up CL will remove length field of the Packet class.


Bug: webrtc:10750
Change-Id: Ie233da83ff33f6370f511955e4c65d59522389a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144881
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28539}
2019-07-11 14:53:39 +00:00
Anton Sukhanov
ff25b873bf Implements method on RtpPacket to extract extension.
Removing extension will be used in DatagramDtlsAdaptor to remove transport sequence number to avoid having both datagram and RTP feedback loops. The sequence number will be stored in temporary map and used to re-create RTCP fdeedback packed when we receive datagram ACK. It would enable integration of Datagram transport without any changes in the upper layers of RTP stack. Note that Datagram adaptor changes will be implemented in a separate changelist.

In this change:
- Implement method to remove extension by rebuilding entire packet without given extension type.
- Fails if extension is not registered or not set.
- Unit test

Bug: webrtc:9719
Change-Id: I9d3811d5d97fadde5a294d5da643b2ebc6a8196e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145100
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28530}
2019-07-10 17:35:43 +00:00
Erik Språng
1d46f9c599 Add RtpPacket::IsExtensionReserved().
This is a small utility method to check whether an extension has been
reserved, so that can be checked before attempting to set an extension
without the need to actually try setting it and potentially failing
with warning loggins as a result.

Bug: webrtc:10633
Change-Id: Ie6f2c4f3f5e94a30dbf60aec6290ebee72681d9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144461
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28455}
2019-07-03 08:13:41 +00:00
Ilya Nikolaevskiy
2d821c3cbc Allow VideoTimingExtension to be used with FEC
This CL allows for FEC protection of packets with VideoTimingExtension by
zero-ing out data, which is changed after FEC protection is generated (i.e
in the pacer or by the SFU).

Actual FEC protection of these packets would be enabled later, when all
modern receivers have this change.

Bug: webrtc:10750
Change-Id: If4785392204d68cb8527629727b5c062f9fb6600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143760
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28396}
2019-06-27 07:38:49 +00:00
Danil Chapovalov
4284828887 Remove deprecated version of RtpPacket::SetPadding that used to randomize padding
was deprecated in
https://webrtc-review.googlesource.com/c/src/+/103983

Bug: None
Change-Id: I617b7b5112446deaa9be983978cabdb247638266
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141865
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28266}
2019-06-13 14:38:38 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Amit Hilbuch
77938e6409 Simulcast work to enable RID mux.
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}
2018-12-21 20:59:23 +00:00
Danil Chapovalov
c5dd3009b4 Introduce RtpPacket::GetExtension accessor that return result
instead of using output parameter.

Bug: None
Change-Id: I1d5c150b7cb6302aa29e040e8c9fe68bddfd8c0e
Reviewed-on: https://webrtc-review.googlesource.com/c/110240
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25565}
2018-11-08 15:34:40 +00:00
Danil Chapovalov
f7fcaf0885 Use zero octets for rtp packet padding
RFC3550 Section 4 mention
"Octets designated as padding have the value zero."

Bug: None
Change-Id: Ife4c6226143c79ad7d152bc6099ba1d81f5492dd
Reviewed-on: https://webrtc-review.googlesource.com/c/103983
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25109}
2018-10-11 10:22:36 +00:00
Johannes Kron
78cdde3df6 Add support for sending RTP two-byte header extensions.
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}
2018-10-05 08:45:52 +00:00
Danil Chapovalov
e19953bdcb Add RtpPacket::GetRawExtension function
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}
2018-10-02 09:53:23 +00:00
Johannes Kron
07ba2b9445 Parse two-byte header extensions.
Bug: webrtc:7990
Change-Id: I967d2065b85d6a2ca938ac0e83035cb92b45a907
Reviewed-on: https://webrtc-review.googlesource.com/98160
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
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@{#24881}
2018-09-28 08:32:17 +00:00
Johannes Kron
c5744b8b21 Refactor to remove direct memory dependency on kMaxId
When two-byte header extensions are enabled, kMaxId will change from 15
to 255. This CL is a refactor to remove the direct dependency between
memory allocation and kMaxId.

Bug: webrtc:7990
Change-Id: I38974a9c705eb6a0fdba9038a7d909861587d98d
Reviewed-on: https://webrtc-review.googlesource.com/101580
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24801}
2018-09-24 13:26:46 +00:00
Danil Chapovalov
7d2df3f848 Inline one-line RtpPacket getters
inlining these accessors both reduce binary size and, likely, slightly improve performance.

Bug: None
Change-Id: I4d1f3285afb044946b9611ad36d5d093299c19a9
Reviewed-on: https://webrtc-review.googlesource.com/94146
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24286}
2018-08-15 09:41:20 +00:00
Danil Chapovalov
527ff1eec2 Remove raw extensions accessors from rtp packet
These accessors were introduced in https://codereview.webrtc.org/2789773004
for dynamic size extensions.
They are now implemented without need of these raw functions

Bug: None
Change-Id: Id43f0bcbf951d60ebeece49556b093956c5ad2bf
Reviewed-on: https://webrtc-review.googlesource.com/92626
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24242}
2018-08-09 10:43:37 +00:00
Danil Chapovalov
9bf31584d1 Pass buffer with size when writing rtp header extension
Bug: chromium:826911
Change-Id: I617fecfee74745004067d892d6e31c94304f99ea
Reviewed-on: https://webrtc-review.googlesource.com/83945
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23641}
2018-06-18 13:04:33 +00:00
Niels Möller
65ec0fc81e Delete unneeded includes of basictypes.h.
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.

Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.

Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
2018-05-21 19:35:08 +00:00
Danil Chapovalov
61405bcb19 Fix infinite loop in rtp packet parsing
when rtp header extension is larger than 2^16 bytes

Bug: chromium:811613
Change-Id: I05b725d734dd628056d603b596d3523e827ddb54
Reviewed-on: https://webrtc-review.googlesource.com/52345
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22003}
2018-02-13 14:42:45 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/modules/rtp_rtcp/source/rtp_packet.h (Browse further)