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}
It was used only in examples/peerconnection/server/peer_channel.cc,
for questionable utility.
Bug: webrtc:6663
Change-Id: I4047eb12f35615621dd0b34a694dead51c5fd20d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128869
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27279}
This change integrates fuzzing support for RtpDumps in WebRTC. This allows
LibFuzzer to directly fuzz the RTP code path from packet arrival all the way
to actual decoding and rendering. It does this by replaying each RTP packet
in the RTPDump which can be mutated directly by the fuzzer.
For fuzzing support the RtpFileReader needs to support reading from a
buffer instead of an file. The test class requires FILE* for all its
parsing operations and is deeply coupled this way. I chose to solve this
problem at an OS level by using the tmpfile() option and copying the buffer
to the tmpfile(). fmemopen() is no available on most platforms so couldn't
be used as a generic solution. The additional copy isn't ideal but won't
be a bottleneck for the fuzzing.
In the future I plan for the fuzzers to read from a configuration file. But
given the current packaging strategy for fuzzers in WebRTC this isn't easy.
Bug: webrtc:9860
Change-Id: I2560120e82663f9e9fb5b9640e6a6d16f9c1a360
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126682
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27151}
This change reduces the risk of echo due to noise in the headroom
of the linear filter.
Changes:
- Use shorter delay headroom
- Delay headroom is specified in samples (not blocks)
- No hysteresis limit when delay is reduced
Bug: chromium:119942,webrtc:10341
Change-Id: I708e80e26d541dff8ca04b6da2d346a1d59cbfcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126420
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27126}
This change simply calls through all code paths in the SSLCertificate interface
after passing in an untrusted PEM string. Corpus will follow in another CL.
Bug: webrtc:10395
Change-Id: I001642fa89a84ce01505780f5e76f01a0e46a785
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127640
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27118}
Adding an example of a request to send simulcast (from the PC).
Adding an example of a request to receive simulcast (from the SFU).
Bug: webrtc:10409
Change-Id: I13b689621e2f89f8e00b7ee8bc542157ccebb873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127621
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27116}
rtp_header_parser currently has 0% fuzzing coverage. To improve this I have
added a basic fuzzer which fuzzes all of the available paths.
Bug: webrtc:10395
Change-Id: I30324b2bfa7629b0110527258b33b7e048e89fcf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127040
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27115}
This change adds a basic fuzzer to exercise parsing of SCTP messages.
Bug: webrtc:10395
Change-Id: I1fd7a8560add3463c1978ebcad30082ae31f2073
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127042
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27113}
This function is called on each incoming RTCP payload.
Bug: webrtc:10395
Change-Id: I164746fe45912cc503565e77046b5d884e0204e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127122
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27110}
StringToNumber is directly used in parsing the SDP so it should be fuzzed.
Bug: webrtc:10395
Change-Id: I85b520fbefd34d3dba49950c5ff297b482c572b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127123
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27089}
The generic video depacketizer was missed in the initial fuzzing pass.
Bug: webrtc:10395
Change-Id: I166f27fc5897a2eafe38dad8e074834fefcc330e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127041
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27088}
This simple fuzzer is intended to detect potential issues in the field trial
parsing code. Since these can be set by the browser it is better to have some
fuzzing coverage around this area.
Bug: webrtc:10395
Change-Id: I1b8b859d2107a0bc99cb7520cf0ef96f3d110547
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127121
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27087}
The fuzzer times out on too long inputs.
This CL limits tests to 400 000 bytes, ~ 12 seconds of 8 kHz float audio.
Bug: chromium:940209
Change-Id: I86b772f9d8989a8b129d933d25ece3631a6a365f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126780
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27059}
Ignore rtc_link_task_queue_impl flag,
instead use build_with_chromium for custom chromium implementation injection
This changes TaskQueue implementation used in webrtc fuzzers in chromium:
from own webrtc implementation to chromium's.
Bug: webrtc:10191
Change-Id: I63be28b680ae8ea8ee1dbf0c699263c392ce29d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125196
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26977}
This avoids timeouts on the server.
Bug: chromium:935089
Change-Id: I8b46664a7cf4d5f14a76b5d034a67453e730eb9e
Reviewed-on: https://webrtc-review.googlesource.com/c/124484
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26940}
Since there is no way to enable/disable these diagnostics at runtime,
this CL moves the suppression into the rtc_* templates in order to
remove the need to explicitly add the snippet of code needed to
suppress it (currently copy/pasted in 144 locations).
The diagnostic that causes the most problems is the one about "complex
class/struct explicit ctor/dtor" [1] because WebRTC doesn't find
it useful enough.
Other diagnostics are good (for example the one that warns about
using "virtual" instead of "override", but that will be covered by
this clang-tidy check [2]) while others are Chromium related so
they have never triggered.
[1] - https://cs.chromium.org/chromium/src/tools/clang/plugins/FindBadConstructsConsumer.cpp?l=147-167&rcl=b4bebe1aa15dba7ca5fcc6456a81a55665327c3a
[2] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html
Bug: webrtc:163
Change-Id: Icbf27efa5b369100a31e6a32df1a0913729b3b34
Reviewed-on: https://webrtc-review.googlesource.com/c/125088
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26918}
The new name fits better.
Bug: None
Change-Id: I1f201ff07915ed6c18efeefb7380e2b286742bb9
Reviewed-on: https://webrtc-review.googlesource.com/c/123800
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26814}
This change disables the ERLE estimation of onsets and instead assumes
minimum ERLE. This reduces the risk of echo leaks during onsets. The
estimated ERLE was sometimes incorrect due to:
- Not enough data to train on.
- Platform noise suppression can change the echo-path.
Bug: chromium:119942,webrtc:10341
Change-Id: I1dd1c0f160489e76eb784f07e99af02f44f387ec
Reviewed-on: https://webrtc-review.googlesource.com/c/123782
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26794}
