This reverts commit 487f9a17e426fd14bb06b13e861071b3f15d119b.
Reason for revert: speculative revert
Original change's description:
> Reland "Refactor SCTP data channels to use DataChannelTransportInterface."
>
> Also clears SctpTransport before deleting JsepTransport.
>
> SctpTransport is ref-counted, but the underlying transport is deleted when
> JsepTransport clears the rtp_dtls_transport. This results in crashes when
> usrsctp attempts to send outgoing packets through a dangling pointer to the
> underlying transport.
>
> Clearing SctpTransport before DtlsTransport removes the pointer to the
> underlying transport before it becomes invalid.
>
> This fixes a crash in chromium's web platform tests (see
> https://chromium-review.googlesource.com/c/chromium/src/+/1776711).
>
> Original change's description:
> > Refactor SCTP data channels to use DataChannelTransportInterface.
> >
> > This change moves SctpTransport to be owned by JsepTransport, which now
> > holds a DataChannelTransport implementation for SCTP when it is used for
> > data channels.
> >
> > This simplifies negotiation and fallback to SCTP. Negotiation can now
> > use a composite DataChannelTransport, just as negotiation for RTP uses a
> > composite RTP transport.
> >
> > PeerConnection also has one fewer way it needs to manage data channels.
> > It now handles SCTP and datagram- or media-transport-based data channels
> > the same way.
> >
> > There are a few leaky abstractions left. For example, PeerConnection
> > calls Start() on the SctpTransport at a particular point in negotiation,
> > but does not need to call this for other transports. Similarly, PC
> > exposes an interface to the SCTP transport directly to the user; there
> > is no equivalent for other transports.
>
> Bug: webrtc:9719
> Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29120}
TBR=steveanton@webrtc.org,mellem@webrtc.org,benwright@webrtc.org
Change-Id: Ibd1a7f30931c114212c90824fec414d276d3f915
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152421
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29141}
Also clears SctpTransport before deleting JsepTransport.
SctpTransport is ref-counted, but the underlying transport is deleted when
JsepTransport clears the rtp_dtls_transport. This results in crashes when
usrsctp attempts to send outgoing packets through a dangling pointer to the
underlying transport.
Clearing SctpTransport before DtlsTransport removes the pointer to the
underlying transport before it becomes invalid.
This fixes a crash in chromium's web platform tests (see
https://chromium-review.googlesource.com/c/chromium/src/+/1776711).
Original change's description:
> Refactor SCTP data channels to use DataChannelTransportInterface.
>
> This change moves SctpTransport to be owned by JsepTransport, which now
> holds a DataChannelTransport implementation for SCTP when it is used for
> data channels.
>
> This simplifies negotiation and fallback to SCTP. Negotiation can now
> use a composite DataChannelTransport, just as negotiation for RTP uses a
> composite RTP transport.
>
> PeerConnection also has one fewer way it needs to manage data channels.
> It now handles SCTP and datagram- or media-transport-based data channels
> the same way.
>
> There are a few leaky abstractions left. For example, PeerConnection
> calls Start() on the SctpTransport at a particular point in negotiation,
> but does not need to call this for other transports. Similarly, PC
> exposes an interface to the SCTP transport directly to the user; there
> is no equivalent for other transports.
Bug: webrtc:9719
Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29120}
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}
This is a reland of 5b4fcb5bf69218c2f42ca2b0cada6c15f2f638e9
Original change's description:
> New build target p2p:stun_types
>
> The media:rtc_media_base target needs definitions of various
> stun-related types and constant. With this new smaller target, it no
> longer needs to depend on all of p2p.
>
> Bug: webrtc:8733
> Change-Id: I05910b6915f6d2c96e8f52a017adbc7eb693dca8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150945
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29036}
Tbr: steveanton@webrtc.org
Bug: webrtc:8733
Change-Id: I1847007ecf29e0e6a27f559b92df632a1cd69280
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151880
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29092}
This reverts commit 5b4fcb5bf69218c2f42ca2b0cada6c15f2f638e9.
Reason for revert: Breaks build
Original change's description:
> New build target p2p:stun_types
>
> The media:rtc_media_base target needs definitions of various
> stun-related types and constant. With this new smaller target, it no
> longer needs to depend on all of p2p.
>
> Bug: webrtc:8733
> Change-Id: I05910b6915f6d2c96e8f52a017adbc7eb693dca8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150945
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29036}
TBR=steveanton@webrtc.org,mbonadei@webrtc.org,nisse@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8733
Change-Id: I6e00657a6137ff773325f37ec02ee1014b6fe96b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151740
Reviewed-by: Hannes Landeholm <hnsl@webrtc.org>
Commit-Queue: Hannes Landeholm <hnsl@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29085}
This reverts commit eec5fff4df92b2330e5fec67ff08c7cbb4c4ab8d.
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}
The media:rtc_media_base target needs definitions of various
stun-related types and constant. With this new smaller target, it no
longer needs to depend on all of p2p.
Bug: webrtc:8733
Change-Id: I05910b6915f6d2c96e8f52a017adbc7eb693dca8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150945
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29036}
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/+/144942https://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}
This reverts commit 4c85828ab272d9bd58789bad7b135b6287395f97.
Reason for revert:
Speculatively reverting this because it makes several web platform tests relating to RTCDataChannel flaky, see first failing roll:
https://chromium-review.googlesource.com/c/chromium/src/+/1776711
Original change's description:
> Refactor SCTP data channels to use DataChannelTransportInterface.
>
> This change moves SctpTransport to be owned by JsepTransport, which now
> holds a DataChannelTransport implementation for SCTP when it is used for
> data channels.
>
> This simplifies negotiation and fallback to SCTP. Negotiation can now
> use a composite DataChannelTransport, just as negotiation for RTP uses a
> composite RTP transport.
>
> PeerConnection also has one fewer way it needs to manage data channels.
> It now handles SCTP and datagram- or media-transport-based data channels
> the same way.
>
> There are a few leaky abstractions left. For example, PeerConnection
> calls Start() on the SctpTransport at a particular point in negotiation,
> but does not need to call this for other transports. Similarly, PC
> exposes an interface to the SCTP transport directly to the user; there
> is no equivalent for other transports.
>
> Bug: webrtc:9719
> Change-Id: I0d3151c48c1a511368277981fc4cf818a9f8ebb4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150341
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29012}
TBR=steveanton@webrtc.org,mellem@webrtc.org,benwright@webrtc.org
Change-Id: I074b9e68f298d20d0cabb4239084b4843e76e910
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150944
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29025}
This change moves SctpTransport to be owned by JsepTransport, which now
holds a DataChannelTransport implementation for SCTP when it is used for
data channels.
This simplifies negotiation and fallback to SCTP. Negotiation can now
use a composite DataChannelTransport, just as negotiation for RTP uses a
composite RTP transport.
PeerConnection also has one fewer way it needs to manage data channels.
It now handles SCTP and datagram- or media-transport-based data channels
the same way.
There are a few leaky abstractions left. For example, PeerConnection
calls Start() on the SctpTransport at a particular point in negotiation,
but does not need to call this for other transports. Similarly, PC
exposes an interface to the SCTP transport directly to the user; there
is no equivalent for other transports.
Bug: webrtc:9719
Change-Id: I0d3151c48c1a511368277981fc4cf818a9f8ebb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150341
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29012}
There are a few reasons for making this test only:
* The code is only used by tests and utilities.
* The pure interface has only a single implementation so an interface isn't really needed.
(a followup change could remove it altogether)
* The implementation always incorporates locking regardless of how the class gets used.
See e.g. previous use in the Packet class.
* The implementation is a layer on top of RtpUtility::RtpHeaderParser which is
sufficient for most production cases.
Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29010}
The packets belonging to a frame were kept in PacketBuffer
until the frame was decoded. This CL clears the dependencies
of an existing RtpFrameObject to PacketBuffer so that we can
free up PacketBuffer as soon as the RtpFrameObject is created.
Bug: none
Change-Id: Ic939be91815519ae1d1c67ada82006417b2d26a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149818
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28977}
This CL will make AudioDecoderIsacT symmetrical to AudioEncoderIsacT.
Bug: webrtc:10826
Change-Id: I78d1cf7bc2245bf4a282aabd81c8ece6ca23f285
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146683
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28847}
Make sure that the packets in the packet buffer belonging to the
first and last sequence numbers are marked as first and last,
respectively.
Bug: chromium:989856
Change-Id: I57bdd7d62d585be2d2083a6b5ce67fce89ab4389
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147875
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28769}
This reverts commit 7325bc3917e6dd4c92e7a18fd879ba91f0b2851f.
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}
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}
This change adds the writing and parsing of the `abs-capture-time` RTP header extension defined at:
http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
We are still missing the code to:
- Negotiate the header extension.
- Collect capture time for audio and video and have the info sent with the header extension.
- Receive the header extension and use its info.
Bug: webrtc:10739
Change-Id: I75af492e994367f45a5bdc110af199900327b126
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144221
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28468}
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}
This change adds the plumbing of RtpPacketInfo from RtpVideoStreamReceiver::OnRtpPacket() to VideoReceiveStream::OnFrame() for video. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
Bug: webrtc:10668
Change-Id: Ib97d430530c5a8487d3b129936c7c51e118889bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139891
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28332}
When the LossNotifications field trial is in effect, LNTF should
be offered/accepted in the SDP message, not assumed to be configured
on both sides equally.
Bug: webrtc:10662
Change-Id: Ibd827779bd301821cbb4196857f6baebfc9e7dc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138079
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28056}
Follows https://webrtc-review.googlesource.com/c/src/+/129768 closely.
Adds an ENCODER and sets it up to parse SDP config for multistream
opus.
E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"
Bug: webrtc:8649
Change-Id: I3fc341e76f5c41dab0243cf65f6461e4c3d9d67d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132001
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27775}
Fuzzer test was configured in a wrong way in
https://webrtc-review.googlesource.com/c/src/+/129768
This fixes it (verified locally on libfuzzer MSAN and ASAN).
Bug: webrtc:8649, chromium:950813
Change-Id: I52647bb12c4c412252cdcd931c9e210606bdb12d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132009
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27516}
See https://webrtc-review.googlesource.com/c/src/+/121764 for the
overall vision.
This CL adds a multistream Opus decoder. It's a new code-path to not
interfere with the standard Opus decoder. We introduce new SDP syntax,
which uses terminology of RFC 7845. We also set up the decoder side to
parse it. The encoder part will come in a later CL.
E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"
Bug: webrtc:8649
Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27493}
rtcp_receiver_fuzzer was running over inputs of unreasonable
length, leading to timeouts. RTCP typically runs over UDP.
This CL limits the inputs to a bit over the max UDP payload length.
Bug: chromium:948469
Change-Id: I669a5b24c265bb3b6da2503da109efed32c25182
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131393
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27482}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
It is possible for the fuzzer to just never deliver packets if the packet delay
is set long enough in the RtpReplayer. This is simply fixed by setting an upper
bound. This change is in the test code setup.
Bug: webrtc:10493,chromium:943420
Change-Id: I54f56e1aa7700f1151e0b58a5a53bc789d032c18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130365
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27369}
This is to avoid time-outs in the fuzzer bots.
Notry: true
Bug: chromium:942886, webrtc:10415
Change-Id: If5e0bcda4e56bb4916bc4479e5b4c822c654c734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129925
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27335}
The pointer-to-submodule interfaces are being removed.
This CL:
1) introduces AudioProcessing::Config::GainController1 with most config,
2) adds functions to APM for setting and getting analog gain,
3) creates a temporary GainControlConfigProxy to support the transition
to the new config.
4) Moves the lock references in GainControlForExperimentalAgc and
GainControlImpl into the GainControlConfigProxy, as it becomes the
sole AGC object with functionality exposed to the client.
Bug: webrtc:9947, webrtc:9878
Change-Id: Ic31e15e9bb26d6497a92b77874e0b6cab21ff2b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126485
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27316}
