`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.
Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.
CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn
Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
Since this macro can be considered public, it makes sense to prefix it
with WEBRTC_ (also to avoid potential conflicts with client code).
This CL also removes some definitions of this macro in order to define
it only where it is strictly needed (it is only used in a .cc file).
Bug: webrtc:11142
Change-Id: Idce7389301e71d8434e238b3cf4ceaa9cf97cd87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161008
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29957}
in support of converging on single way to run asynchronous tasks in webrtc
Bug: b/144982320
Change-Id: I200ad298136d11764a3f5c0547ebcba51aceafa0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158782
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29896}
Otherwise we require that the destructor must run on the same thread as
previous calls. This is not necessary since we can assume there's no
other references to the instance when we enter the destructor.
Bug: webrtc:9883
Change-Id: Ia254bce9265979da0e25ba33598edd8f807d7e00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159704
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29793}
This CL adds ParseStatus/ParseStatusOr classes and returns those instead
of CHECKing that the log is well formed. Some refactoring was required.
We also add a allow_incomplete_logs parameter to the parser which by
default is false. Setting it to true will make the parser log a warning
but return success for errors that typically indicate that the log has
been truncated. "Deeper" errors indicating log corruption still return
an error.
Bug: webrtc:11064
Change-Id: Id5bd6e321de07e250662ae3aaa5ef15f48db6d55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158746
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29679}
Due to changes in how the transport feedback is processed, the late
feedback results plot doesn't get any entries anymore.
Bug: webrtc:9883
Change-Id: I9df8e86a35bedddf78407128f0ab0b6b321a6f28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158668
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29643}
Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to
move it to its own target and allow its dependant to avoid to take a
dependency rtc_base:on rtc_base_approved.
Bug: webrtc:9419
Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29548}
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
Add missing includes to files that were transactivly depending on removed includes.
Bug: None
Change-Id: Id5923bb8dc3e1d8fbb664e460278ad3e5993be7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155963
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29396}
Also annotate a few of the remaining uses, to guide further splits of
that large build target.
Bug: webrtc:8733
Change-Id: I16ac33ab48e6d39a1a8dbc2a3fc671d8db6dbfe9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150789
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29001}
The new target does not depend on libjingle_peerconnection_api, and to
do this, the named "audio" and "video" string literals had to be moved from
media_stream_interface.cc to media_types.cc.
In this cl, the dependency on libjingle_peerconnection_api can be
dropped from a few targets.
No-Presubmit: True
Bug: webrtc:8733
Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28998}
Scenario tests runs all its activities on task queues. This is not
allowed by the default event log writer, causing a DCHECK failure.
This CL makes it possible to stop the event asynchronously,
thereby avoiding the need for the DCHECK.
Bug: webrtc:10365
Change-Id: I1206982b29fd609ac85b4ce30ae9291cbec52041
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136685
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28027}
This prepares for adding a logged event for route changes.
Bug: webrtc:10614
Change-Id: Iee989e029530c991da38697a30e0a585316d9a05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135951
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27915}
This sequence checker was necessary back when a concern existed
over calling StopLogging() twice. That is no longer a concern.
Bug: webrtc:10613
Change-Id: Ib28d876a8c1940e76d4914287043cce2a1d974b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135949
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27914}
This is a reland of 0870c70b04
Original change's description:
> Reland "Improving robustness of feedback matching code in event log parser."
>
> This is a reland of a1e4fbb253
>
> Original change's description:
> > Improving robustness of feedback matching code in event log parser.
> >
> > Removes the dependency on TransportFeedbackAdapter thereby removing
> > some of the complexity that came with it, in particular, we don't fill
> > in missing packets. This makes the code easier to debug and avoids some
> > confusing logging that's not relevant for the parser.
> >
> > Bug: webrtc:9883
> > Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27739}
>
> Bug: webrtc:9883
> Change-Id: I460d0c576626614fb4ce2c3d5e3ddbb5d1c122cf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134106
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27763}
Bug: webrtc:9883
Change-Id: I1f80ed1f63ad75fbb97f5f401fe486d19c057f75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134462
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27829}
This reverts commit 0870c70b04.
Reason for revert: Failed to handle lost packets.
Original change's description:
> Reland "Improving robustness of feedback matching code in event log parser."
>
> This is a reland of a1e4fbb253
>
> Original change's description:
> > Improving robustness of feedback matching code in event log parser.
> >
> > Removes the dependency on TransportFeedbackAdapter thereby removing
> > some of the complexity that came with it, in particular, we don't fill
> > in missing packets. This makes the code easier to debug and avoids some
> > confusing logging that's not relevant for the parser.
> >
> > Bug: webrtc:9883
> > Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27739}
>
> Bug: webrtc:9883
> Change-Id: I460d0c576626614fb4ce2c3d5e3ddbb5d1c122cf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134106
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27763}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9883
Change-Id: Ibcfc4f7425fe202d86f0c3a33de51e605dc17c04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134312
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27790}
This is a reland of a1e4fbb253
Original change's description:
> Improving robustness of feedback matching code in event log parser.
>
> Removes the dependency on TransportFeedbackAdapter thereby removing
> some of the complexity that came with it, in particular, we don't fill
> in missing packets. This makes the code easier to debug and avoids some
> confusing logging that's not relevant for the parser.
>
> Bug: webrtc:9883
> Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27739}
Bug: webrtc:9883
Change-Id: I460d0c576626614fb4ce2c3d5e3ddbb5d1c122cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134106
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27763}
This reverts commit a1e4fbb253.
Reason for revert: Breaks downstream.
Original change's description:
> Improving robustness of feedback matching code in event log parser.
>
> Removes the dependency on TransportFeedbackAdapter thereby removing
> some of the complexity that came with it, in particular, we don't fill
> in missing packets. This makes the code easier to debug and avoids some
> confusing logging that's not relevant for the parser.
>
> Bug: webrtc:9883
> Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27739}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: Icdf3231f5a32b6f63a903c7dffc8ca505680a72a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134105
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27742}
Removes the dependency on TransportFeedbackAdapter thereby removing
some of the complexity that came with it, in particular, we don't fill
in missing packets. This makes the code easier to debug and avoids some
confusing logging that's not relevant for the parser.
Bug: webrtc:9883
Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27739}
With this change, both the normal RTP and the transport-wide sequence
numbers are propagated with with AddPacket() call via a new
RtpPacketSendInfo struct, replacing the previous set of parameters.
The intent with this is that SendTimeHistory can hold a mapping from
transport-wide to rtp sequence numbers, and then via callbacks let the
RTP modules know when packets have been received by the remote end.
Bug: webrtc:8975
Change-Id: I6a24fc6282cbb041393752d39593c2867b242192
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133021
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27708}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: I2ea59dc66230182bee6ae7a0925aed0fe9ef823c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127643
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27133}
This CL adds a mode to simulate roughly what GoogCC could have been
doing during the recording of an rtc event log by using the logged
events as input to GoogCC and visualizing the resulting target rate.
This is similar to the existing simulated_sendside_bwe mode, but uses
the new NetworkControllerInterface to ensure more reliable GoogCC
simulation.
Bug: None
Change-Id: I57894aa666151efc8405407d928b5257fb9b7d61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123924
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27095}
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}
This fixes a build error on C++17 mode due to missing #include, plus
adds a presubmit check to prevent further breakage.
Bug: chromium:752720
Change-Id: I5c7d1dca0079dfe7a042650402e6f7ae28a797ba
Reviewed-on: https://webrtc-review.googlesource.com/c/124940
Commit-Queue: Taiju Tsuiki <tzik@chromium.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26898}
Import proto_library.gni when rtc_enable_protobuf is true instead of when
build_with_mozilla is false.
Makes it maybe easier to reason about the intention (e.g. intention is to not
compile any protobuf in, hence flag rtc_enable_protobuf)
The build file could not work if build_with_mozilla = true but
rtc_enable_protobuf = true.
Bug: webrtc:10338
Change-Id: I26e5983bd1519aa46c308b11796d518de5ef7597
Reviewed-on: https://webrtc-review.googlesource.com/c/123763
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26868}
The FMT 15 is not specific only to REMB or loss notification messages.
Rather, it is the Application Layer FB (AFB) of Psfb (Payload Specific
Feedback Messages).
See https://tools.ietf.org/html/rfc4585#section-6.3TBR=terelius@webrtc.org
Bug: webrtc:10336
Change-Id: I8cd27ef9ee044bf7b7e7c1bd1a53c1dae2d95006
Reviewed-on: https://webrtc-review.googlesource.com/c/123886
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26827}
absl/memory/memory.h needs to be included for absl::WrapUnique on the
C++17 mode build, as absl/types/optional.h has a conditional include of
memory.h.
Tbr: terelius@webrtc.org
Bug: chromium:752720
Change-Id: I7bdfb200f29ac8ec11f0b102304b7788595e7eb9
Reviewed-on: https://webrtc-review.googlesource.com/c/123360
Commit-Queue: Taiju Tsuiki <tzik@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26815}
This change adds incoming & outgoing packet rates to the
event_log_visualizer.
The outgoing packet rate is drawn on the graph with outgoing RTP rate,
because we want to see it together with bandwidth estimate and probe
clusters.
The incoming packet rate is drawn separately.
Bug: webrtc:9719
Change-Id: I32648d016359af110837440ed1a5f9c31c841ea7
Reviewed-on: https://webrtc-review.googlesource.com/c/122941
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26696}
The old format is still used.
Bug: None
Change-Id: I2c8c9a8c7b09282ba8c375403cbee7fd7ab2d056
Reviewed-on: https://webrtc-review.googlesource.com/c/122960
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26679}
This change does not include receive_timestamps for ACKs, because there is 1 problem.
That problem will be resolved in a separate change.
I am getting receive_timestamp errors that have to do with delta compression with optional fields.
Two failure modes that I noticed:
1) the base event does not have the timestamp: it crashes with length validation
# Check failed: base <= MaxUnsignedValueOfBitWidth(params_.value_width_bits()) (1820716 vs. 131071)
2) all events are null, it crashes with assert that X events were expected, but no events were deserialized.
Bug: webrtc:9719
Change-Id: I5d1bbb95dfd15ca7321667aad5e4d89c085e9c06
Reviewed-on: https://webrtc-review.googlesource.com/c/122360
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26668}
Hard-coding default values forces IDs over 14 to be used even
when we offer less than 15 different extensions.
Note that the code relies on MergeRtpHdrExts for making sure
that extension IDs are kept consistent and non-colliding between
different streams (audio/video).
Bug: webrtc:10288
Change-Id: I3e59f7ddc8ca43cea91084a6b7f36df70fb6be4a
Reviewed-on: https://webrtc-review.googlesource.com/c/121646
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26622}
This CL is a manual spin-off of [1], which tried to apply clang-tidy's
performance-move-const-arg [1] to the WebRTC codebase.
Since there were some wrong fixes to correct, this CL lands a few
different fixes, like adding a constructor overload to take an rvalue
reference or remove 'const' to make std::move effective.
[1] - https://webrtc-review.googlesource.com/c/src/+/120350
[2] - https://clang.llvm.org/extra/clang-tidy/checks/performance-move-const-arg.html
Bug: webrtc:10252
Change-Id: I42a777247fee2cb788efcd7c2035148330056b7a
Reviewed-on: https://webrtc-review.googlesource.com/c/120928
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26553}
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
This CL applies clang-tidy's performance-for-range-copy [1] on the
WebRTC codebase.
All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.
[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-for-range-copy.html
Bug: webrtc:10215
Change-Id: I7c83290b8866d76129bbec4e24e6701f5014102e
Reviewed-on: https://webrtc-review.googlesource.com/c/120043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26420}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
This fixes a bug where the event_log_visualizer hits a DCHECK when the RTP timestamp jumps.
TBR = kwiberg
Bug: webrtc:10170
Change-Id: I127a8e6165265d0726892a912f5bcdc33d98ced5
Reviewed-on: https://webrtc-review.googlesource.com/c/119664
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26410}
Create a new RTCP feedback message for reporting the loss and/or non-decodability of video frames, to be used by the upcoming injectable VideoFrameBufferController. The new feedback message should report:
1. The sequence number of the last decoded non-discardable video frame. (TBD: If a multi-packet frame, should it be the sequence number of the first, last, or any of the packets?)
2. The sequence number of the last received RTP packet in the stream.
3. A decodability flag, whose specific meaning depends on the last-received
RTP sequence number. The decodability flag is true if and only if all of
the frame's dependencies are known to be decodable, and the frame itself
is not yet known to be unassemblable.
* Clarification #1: In a multi-packet frame, the first packet's
dependencies are known, but it is not yet known whether all parts
of the current frame will be received.
* Clarification #2: In a multi-packet frame, the dependencies would be
unknown if the first packet was not received. Then, the packet will
be known-unassemblable.
Bug: webrtc:10226
Change-Id: I1563c944477e3ed40235e82ab99a439414632aff
Reviewed-on: https://webrtc-review.googlesource.com/c/118931
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26387}
Instead timestamps required for processing are provided explicitly.
This makes it easier to ensure correct usage in log processing
and simulation.
Bug: webrtc:10170
Change-Id: I724a6b9b94e83caa22b8e43b63ef4e6b46138e6a
Reviewed-on: https://webrtc-review.googlesource.com/c/118702
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26339}
In particular, time_utils.h is currently pulled in via rtc_event.h
This CL is in preparation of moving parts of the RTC event log to api/.
Bug: webrtc:10206
Change-Id: Idd35aa9404afded4d29b1296344996c45b8c2e91
Reviewed-on: https://webrtc-review.googlesource.com/c/117921
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26326}
This is in preparation for letting Chrome extract DTLSTransport
information after SLD/SRD instead of doing it on-demand.
Bug: chromium:907849
Change-Id: Iac6b174c98d3d14136e1fd25bce4a9292f6c8b41
Reviewed-on: https://webrtc-review.googlesource.com/c/116984
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26289}
- Remove visibility of encoder target.
- Remove unnecessary dependency on task_queue.
- Remove CreateRtcEventLogFactory() declaration from the rtc_event_log_api target
since the function is not defined in that target.
Bug: None
Change-Id: Id9edee86f358d08ea063d62bd96e9653c5b06d55
Reviewed-on: https://webrtc-review.googlesource.com/c/116060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26215}
This CL moves the code that stores SSRCs used by different media types
so that it will be used by the new format too. This is sufficient to
get the correct media types printed in e.g. event_log_visualizer.
Bug: webrtc:8111
Change-Id: Ife11bc49b2af7577c7b5326c0b0fadd2e5b48b94
Reviewed-on: https://webrtc-review.googlesource.com/c/113942
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25965}
This has been deprecated for a long time. Simulcast streams are now logged as
one RtcEventVideoSendStreamConfig per SSRC instead of one RtcEventVideoSendStreamConfig
containing a group of SSRCs
Bug: webrtc:8111
Change-Id: I4da62a4b2151a841413cde222a5154638dbb2e47
Reviewed-on: https://webrtc-review.googlesource.com/c/113811
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25957}
The header extensions for a receive stream should also be used
for the associated RTX stream, but not for the (RTCP) send stream.
Bug: webrtc:10113
Change-Id: Ibeb25a4490d7f628f1b360bf4d6f7edf444ba22a
Reviewed-on: https://webrtc-review.googlesource.com/c/113807
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25955}
This handles an unlikely corner case where you receive a RTCP feedback for a packet the same millisecond that you send it.
Bug: None
Change-Id: I77f460bef4073d4d9c5633c88f4d2dd8470f8577
Reviewed-on: https://webrtc-review.googlesource.com/c/113305
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25911}
This was previously only done for the legacy wire format.
Bug: webrtc:8111
Change-Id: I82767c8eafb35a50967c6ff3d0131c3981957c1f
Reviewed-on: https://webrtc-review.googlesource.com/c/112590
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25909}
The event log implementation will be simpler if it creates its own TaskQueue.
If we really need the "injectable" functionality, it could be achieved via a
TaskQueueFactory that returns a move-constructible TaskQueue.
Bug: webrtc:10085
Change-Id: I538be3dd77c09be2f5bae015227067acd6af8355
Reviewed-on: https://webrtc-review.googlesource.com/c/113140
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25908}
1. Make |output_period_ms_| optional, so as to clarify where
it gets assigned a value. (I.e. the value set by the ctor
is not retained.)
2. Some extra const modifiers.
Bug: webrtc:8111
Change-Id: I9f3ad7ff763cfbc9c9385f7fd4325ba696772765
Reviewed-on: https://webrtc-review.googlesource.com/c/112588
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25877}
This is a reland of a4dcb749fb
Original change's description:
> Fix output period in RtcEventLogImpl
>
> RtcEventLogImpl::StartLogging() was ignoring one of its parameters.
> This CL fixes the issue.
>
> Bug: webrtc:10082
> Change-Id: Ie1790c1a7299748dabe99909d967384ad9895635
> Reviewed-on: https://webrtc-review.googlesource.com/c/112586
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25858}
Bug: webrtc:10082
Change-Id: I783fba84aa35e489f6235538c624b19f2f98a962
Reviewed-on: https://webrtc-review.googlesource.com/c/112860
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25876}
RtcEventLogImpl::task_queue_ is a std::unique_ptr<rtc::TaskQueue>.
When a unique_ptr is destroyed, it first sets its internal pointer
to point to null, and only then invokes the destructor of that
object. However, the code in RtcEventLogImpl relies on
rtc::TaskQueue's property, that its destructor blocks on executing
tasks.
We solve by manually invoking the destructor, and only resetting
the internal pointer thereafter. In theory, we could have changed
the unique_ptr to a raw pointer at this point. We avoid that, so
as to keep the ownership clearer to readers of the code.
Bug: webrtc:10085
Change-Id: I54bbf5d6bae019757ca2e31ee960d558058ccc42
Reviewed-on: https://webrtc-review.googlesource.com/c/112598
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25875}
This reverts commit a4dcb749fb.
Reason for revert: Speculative revert. Tsan failure has been consistently generated after this CL.
Original change's description:
> Fix output period in RtcEventLogImpl
>
> RtcEventLogImpl::StartLogging() was ignoring one of its parameters.
> This CL fixes the issue.
>
> Bug: webrtc:10082
> Change-Id: Ie1790c1a7299748dabe99909d967384ad9895635
> Reviewed-on: https://webrtc-review.googlesource.com/c/112586
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25858}
TBR=eladalon@webrtc.org,terelius@webrtc.org
Change-Id: I6b79c207d537ab6ca44bb418958854acebc886ac
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10082
Reviewed-on: https://webrtc-review.googlesource.com/c/112740
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25864}
This is a reland of c308bdfa45
Original change's description:
> Add transaction id to CandidatePairEvents.
>
> The transaction id is a randomly generated number used to link stun
> requests and responses (https://tools.ietf.org/html/rfc5389#section-6).
> Logging this will help us debug ICE network issues.
>
> Bug: webrtc:9972
> Change-Id: I93167cb119aad99156e8727b6e4eeeff5198f924
> Reviewed-on: https://webrtc-review.googlesource.com/c/109720
> Commit-Queue: Zach Stein <zstein@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25848}
TBR=terelius@webrtc.org,qingsi@webrtc.org
Bug: webrtc:9972
Change-Id: I32b55befddfcb8dc98babd0b64e756eaeb9fab09
Reviewed-on: https://webrtc-review.googlesource.com/c/112661
Reviewed-by: Zach Stein <zstein@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25862}
RtcEventLogImpl::StartLogging() was ignoring one of its parameters.
This CL fixes the issue.
Bug: webrtc:10082
Change-Id: Ie1790c1a7299748dabe99909d967384ad9895635
Reviewed-on: https://webrtc-review.googlesource.com/c/112586
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25858}
This reverts commit c308bdfa45.
Reason for revert: The msan bot has been consistently failing since this commit. See eg https://ci.chromium.org/p/webrtc/builders/luci.webrtc.ci/Linux%20MSan/16989
Original change's description:
> Add transaction id to CandidatePairEvents.
>
> The transaction id is a randomly generated number used to link stun
> requests and responses (https://tools.ietf.org/html/rfc5389#section-6).
> Logging this will help us debug ICE network issues.
>
> Bug: webrtc:9972
> Change-Id: I93167cb119aad99156e8727b6e4eeeff5198f924
> Reviewed-on: https://webrtc-review.googlesource.com/c/109720
> Commit-Queue: Zach Stein <zstein@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25848}
TBR=eladalon@webrtc.org,terelius@webrtc.org,zstein@webrtc.org,qingsi@webrtc.org,jeroendb@webrtc.org
Change-Id: Ib3b0a845f2300f4fcba2061650e17522735f08b3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9972
Reviewed-on: https://webrtc-review.googlesource.com/c/112581
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25852}
The transaction id is a randomly generated number used to link stun
requests and responses (https://tools.ietf.org/html/rfc5389#section-6).
Logging this will help us debug ICE network issues.
Bug: webrtc:9972
Change-Id: I93167cb119aad99156e8727b6e4eeeff5198f924
Reviewed-on: https://webrtc-review.googlesource.com/c/109720
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25848}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
RSID is only useful if we store the RSID header extension.
Since we don't do that at the moment, there is no need to
store RSID in the stream configs.
Bug: webrtc:8111
Change-Id: I978f335d05984346f225c4781a8bfaa228f3f4c8
Reviewed-on: https://webrtc-review.googlesource.com/c/111759
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25763}
This allows the fall back list to be used instead.
Bug: webrtc:9718
Change-Id: Ie17a4b740fef60385c6019ea167c73eff07e8ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/111246
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25676}
The new parser provides the same functionality (with a slightly
different API) and is backwards compatible with the legacy wire format.
Downstream projects seem to have transitioned to the new parser API.
Bug: webrtc:8111
Change-Id: Icb458f0d55e0a4566c4b7b4a53cab48f0b9d6fd8
Reviewed-on: https://webrtc-review.googlesource.com/c/110782
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25635}
This tool does not seem useful enough to justfy the maintenance cost.
If we want something like this in the future, then the core logic
should be added to the parser.
Bug: webrtc:8111
Change-Id: Ifc3dc9b91e85246d35d7775c68d0f2dc687516aa
Reviewed-on: https://webrtc-review.googlesource.com/c/110724
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25623}
When delta encoding, use ToUnsigned() and ToSigned() on
timestamp_ms, since it's a signed type. This is only relevant
for delta-encoding/decoding.
Bug: webrtc:8111
Change-Id: I1fabfcb2be64793c281f5bc0d38a2f8035dd0d18
Reviewed-on: https://webrtc-review.googlesource.com/c/110504
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25602}
Since the number of channels is always greater than 0, smaller
deltas can be accomplished by encoding a sequence of (1, 2, 1)
as if the sequence were (0, 1, 0). This way, wrap around to the
first value is a delta of 1, rahter than a delta of 3.
For simplicity's sake, though at the cost of consistency, we still
encode the base event's number of channels unshifted. We do so
because there are no bits to be gained by doing it otherwise, and
the value there is more likely to be manually inspected, than are
the deltas, so a simpler scheme has merit.
Bug: webrtc:8111
Change-Id: I2d4def67da85c42802fe13cd0494fdd9f2b38f7a
Reviewed-on: https://webrtc-review.googlesource.com/c/110242
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25601}
The new event log format makes use of delta encoding to compress
parts of the log.
Bug: webrtc:8111
Change-Id: I7bec839555323a7537dcec831d4ac1d5eb109932
Reviewed-on: https://webrtc-review.googlesource.com/c/109161
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25584}
Until this CL, RtcEvent::Copy() was a virtual function returning
a std::unique_ptr<RtcEvent>, which was used only in unit tests.
Whenever it was used, the exact sub-class was known, so returning
as a std::unique_ptr<RtcEvent> lost information, but the
additional flexibility of calling Copy on an unknown RtcEvent
was never used. This CL changes the function to a non-virtual one.
Bug: webrtc:8111
Change-Id: I30dbea5f4ec84c41282550aa77a855bfae9ffbd1
Reviewed-on: https://webrtc-review.googlesource.com/c/109140
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25487}
Add some constructors to the structs in rtc_event_log_parser_new.h,
so that they may be emplaced into containers.
Bug: webrtc:8111
Change-Id: I2ccc3026673eef1237c7de2405e500fe9d7a33d0
Reviewed-on: https://webrtc-review.googlesource.com/c/108121
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25396}
Before this CL, when we encoded a sequence with a non-existent
base, we pretended that the delta was 0, and the first delta was
based on that. However, in a sequence where the deltas are small,
but where the first element is big, that would produce
unnecessarily wide deltas. Therefore, we change the behavior in
cases where the base is non-existent, to encode the first existent
value (if any) as a varint; the delta width may then be smaller.
This CL include two piggy-backed changes:
1. Varint encoding/decoding moved to its own file (and an
additional flavor added).
2. The unit tests for delta encoding are further parameterized
with a random seed.
Bug: webrtc:8111
Change-Id: I76fff577c86d019c8334bf74b76bd35db06ff68d
Reviewed-on: https://webrtc-review.googlesource.com/c/107860
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25395}
This is a reland of ece3c228a2
Original change's description:
> Encode RTC event logs in new format.
>
> This CL adds the encoder and wires it up to the event log.
> Parser and unit tests are uploaded in a separate CL.
>
> Bug: webrtc:8111
> Change-Id: I6470003e55c2c4006cd8349a2c4bdc3f9491d869
> Reviewed-on: https://webrtc-review.googlesource.com/c/106708
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25333}
Bug: webrtc:8111
Change-Id: I22eeca36d6b1f7cfa1ac65347571ebe33cecc1fc
Reviewed-on: https://webrtc-review.googlesource.com/c/108082
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25382}
Optional fields are those which only occur sometimes. For example,
the sequence number field in an RTP packet always occurs, but
fields in optional RTP extensions only occur sometimes.
Bug: webrtc:8111
Change-Id: Iff2c35b73530c0a1db68e547b4caf34434aa4ace
Reviewed-on: https://webrtc-review.googlesource.com/c/103362
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25336}
This CL adds the encoder and wires it up to the event log.
Parser and unit tests are uploaded in a separate CL.
Bug: webrtc:8111
Change-Id: I6470003e55c2c4006cd8349a2c4bdc3f9491d869
Reviewed-on: https://webrtc-review.googlesource.com/c/106708
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25333}
Signed deltas can yield a more efficient encoding when the encoded
sequence sometimes moves backwards.
Bug: webrtc:8111
Change-Id: Ib1a50192851214ccc3f2bd7eaf88f4be97e4beb0
Reviewed-on: https://webrtc-review.googlesource.com/c/100423
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25324}
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}
A blob is a string of binary information, whose length may not
necessarily be determined by looking into the string, so that
concatenating all blobs without explicitly including their lengths
as part of their encoding is not a viable option.
Bug: webrtc:8111
Change-Id: I89fdca660e89a6a71eff3ecb7b86416312b81f23
Reviewed-on: https://webrtc-review.googlesource.com/c/104201
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25278}
This is a reland of 5ccdc1331f
Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
> for y in bool int float string FLAG; do
> git grep -l "\b$x\_$y\b" | \
> xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
> done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}
TBR=kwiberg@webrtc.org
Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
This reverts commit 5ccdc1331f.
Reason for revert: Breaks downstream project.
Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
> for y in bool int float string FLAG; do
> git grep -l "\b$x\_$y\b" | \
> xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
> done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: Ia79cd6066ecfd1511c34f1b30fd423e560ed6854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9884
Reviewed-on: https://webrtc-review.googlesource.com/c/107160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25276}
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).
They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.
This CL adds the 'WEBRTC_' prefix to them.
Generated with:
for x in DECLARE DEFINE; do
for y in bool int float string FLAG; do
git grep -l "\b$x\_$y\b" | \
xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
done
done
git cl format
Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}
Add code for delta-encoding and decoding, to be used when producing
WebRTC event logs of the new format.
This CL supports fixed-size encoding only. Also, no support for
signed deltas or optional values yet. These will be added in
subsequent CLs.
Bug: webrtc:8111
Change-Id: I531abd99fd924f4c9e692abe565bc6f66c875ad5
Reviewed-on: https://webrtc-review.googlesource.com/c/100304
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25256}
Padding size and header size are not part of the header, but we still
want to log them. Add the values as separate fields to the log events.
Bug: webrtc:8111
Change-Id: I8dfa2ccafe679f96b8911b538a8512b0170bc642
Reviewed-on: https://webrtc-review.googlesource.com/c/106321
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25237}
We're no longer verifying CSRCs or configurations for remb, rtcp mode
and codec since we're planning to drop those fields from the log in an upcoming CL.
Bug: webrtc:8111
Change-Id: I38a7d87b21f8e6d8a791d8e27a0f54c293f3d340
Reviewed-on: https://webrtc-review.googlesource.com/c/106380
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25229}
Microsecond resolution is system dependent anyway, so it wasn't reliable.
This CL verifies millisecond timestamps instead of microsecond in tests.
Bug: webrtc:8111
Change-Id: I14aab9a807f747a88b2b84f51becf54f4097931e
Reviewed-on: https://webrtc-review.googlesource.com/c/105561
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25138}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
Add helper class to process RtcEventLog events in order.
Use helper class to migrate rtc_event_log2rtp_dump.cc
to new parser API.
Bug: webrtc:8111
Change-Id: I7cbc220dad1f50be3a985ed44de27b38e5f20476
Reviewed-on: https://webrtc-review.googlesource.com/98601
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24806}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I443c0c77cef30a3f5712f72ce88db80d1fb29874
Reviewed-on: https://webrtc-review.googlesource.com/96642
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24567}
Since the linux_memcheck trybot is no more, this CL removes all the
code needed to make it work.
Bug: webrtc:7737, webrtc:8356, webrtc:9570
Change-Id: I09a9467b8bf895146a3384c2c915b54662721af6
Reviewed-on: https://webrtc-review.googlesource.com/90863
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24201}
This doesn't cause problems because seems like GN just ignores these attempts and confines everything to the root.
Bug: None
Change-Id: Ief50b6e10f3950b319d0bde547f91dcc7f237cda
Reviewed-on: https://webrtc-review.googlesource.com/90041
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24063}
The Copy() function previously did not copy the logging timestamp.
To be able to use Copy() in this test, we add private copy
constructors for RtcEvents which the Copy() can use to copy
everything including the timestamp.
Also adds missing test for RtcEventAlrState,
RtcEventIceCandidatePairConfig and RtcEventIceCandidatePair.
Bug: webrtc:8111
Change-Id: I3901231735baa4e671173c921eada0a4be6de7c9
Reviewed-on: https://webrtc-review.googlesource.com/86042
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23774}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script passing top level directories except rtc_base and api
find $@ -type f \( -name \*.h -o -name \*.cc -o -name \*.mm \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -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: I9465c172e65ba6e6ed4e4fdc35b0b265038d6f71
Reviewed-on: https://webrtc-review.googlesource.com/84584
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23697}
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 CL fixes a bug that the RtcEventLog owned by PeerConnection was not
passed to P2PTransportChannel after JsepTransportController was
introduced to deprecate the legacy TransportController.
Bug: webrtc:9337
Change-Id: I406cd9c0761dfe67f969aa99c6141e1ab38249d5
Reviewed-on: https://webrtc-review.googlesource.com/79964
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23572}
