To unblock replacing RTCStatsMember<T> with absl::optional<T>.
RTCStats::GetAttribute() is added to allow obtaining the name of a
metric even after the replacement, the only usage of this AFAIK is this
test file.
Bug: webrtc:15164
Change-Id: I21e8b782ba86d9f864cb90885134c6724ba3398b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334163
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41515}
To unblock code cleanup, uses of Members() are replaced by
Attributes(). This will allow replacing RTCStatsMember<T> with
absl::optional<T> in a future CL.
Bug: webrtc:15164
Change-Id: Ic4ca0d333a9cdaa78739bd8dd4515d2ecfacffbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334162
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41512}
To unblock code cleanup, uses of Members() are replaced by
Attributes(). This will allow replacing RTCStatsMember<T> with
absl::optional<T> in a future CL.
Bug: webrtc:15164
Change-Id: I4ed32ef8ea3d8fd7506415848e044e34377fce8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334161
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41511}
One in series of CLs to migrate from Members() to Attributes() to
unblock code deletion.
Adds and makes use of has_value(), holds_alternative<T>() and get<T>()
convenience methods.
Bug: webrtc:15164
Change-Id: I2ae9869a5d1de3f3875c70494c9781d69da7f936
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334160
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41510}
It only worked with VP8 before.
Tested: out/debug/video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --encoder=openh264 --decoder=ffmpeg-h264 --num_frames=30 --scalability_mode=S2T2 --dump_encoder_output -> 360p and 720p streams with two temporal layers each were produced. Bitrate allocation across temporal layers is done by OpenH264 encoder (no API to control this).
Bug: webrtc:14852
Change-Id: I58e2e1f595bdd6653701a97874766752bd2e3d58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332342
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41508}
The plan is to replace Members() with Attributes() instead.
For backwards-compatability during the transition, Attribute implements
RTCStatsMemberInterface but the two classes serve the same purpose
which is to allow iterating all metrics of a stats object.
The reason for moving away from "members" is that we already have a way
to express a variable that maybe has a value: absl::optional<T>. The
only information the member adds is the const char* name(), which we'll
move to Attribute in a future CL.
We don't need to maintain an RTCStatsMemberInterface::Type enum in the
future because absl::variant<T> has absl::holds_alternative<T>.
Step 1: Add Attributes().
Step 2: Migrate to Attributes() and delete Members().
Step 3: Replaces all uses of RTCStatsMember<T> with absl::optional<T>
and delete RTCStatsMember + RTCStatsMemberInterface.
Bug: webrtc:15164
Change-Id: I3fdd5b24214bb5cc340a54a0171df73b516e1803
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333840
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41507}
Remove member variables that point to objects owned externally (in practice by SdpOfferAnswerHandler). The objects also live on the
signaling thread whereas JsepTransportController performs
operations on the network thread. Removing the raw pointers avoids
the risk of referencing the description objects after they've been
deleted or if the state is inconsistent across threads.
Bug: webrtc:1515832
Change-Id: I852b2a3993964be817f93c46b5bc4b03121cde86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334061
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41505}
As of https://webrtc-review.googlesource.com/c/src/+/332200, socket is
no longer pure virtual and needs to be exported for Chromium component
build tests.
Bug: webrtc:15368
Change-Id: I3d2fcd329bc859d07106ae740eb2791eccaea1f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334060
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41504}
Bug: webrtc:15719
Change-Id: I7daf8ee5b90fbe9f1246f1d99211ffa0d8a19f73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330780
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41503}
rtc::TaskQueue is a wrapper of TaskQueueBase providing no extra functionality in this case
Bug: webrtc:14169
Change-Id: I5eb27a5dbb16f6097a9c71c2633c807808e50c05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333800
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41501}
Ensure top_active_prio_level_ is set to -1 in MaybeUpdateTopPrioLevel if
last packet is purged.
Bug: webrtc:15740
Change-Id: I81df9ee084de89f79b8ab79db8ce52fe1e20738a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333883
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41498}
Intention is to gradually stop using raw pointers and make it easier to introduce
new meta data types.
A default implementation is added that use existing int RecvFrom(void*
pv,..)
In this cl, async_udp_socket.cc use the new method. There should be no
behaviour change.
Bug: webrtc:15368
Change-Id: I8f9773a65d24ab5bbac3534dcc37ee1ed874a2c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41497}
This field trial was added 5 years ago in
https://webrtc-review.googlesource.com/c/src/+/111883
probably as a safe guard, but looks never used.
Bug: webrtc:11503
Change-Id: Ia9544b652b25fad4c614d66fe020f3d994c96505
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333380
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41490}
Measures the time consumed by OnFrame (e.g. the encoding time) and
sets an overload flag during N subsequent frames if the time is
longer than the current frame time. N is set to the number of
received frames on the network thread while being blocked by
encoding.
The queue overload mechanism for zero hertz can be disabled using the
WebRTC-ZeroHertzQueueOverload kill switch.
Also adds a UMA called WebRTC.Screenshare.ZeroHz.QueueOverload.
Bug: webrtc:15539
Change-Id: If81481c265d3e845485f79a2a1ac03dcbcc3ffc3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332381
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41489}
Add ulpfec and flexfec to list of resiliency mechanisms taken
into account and in general exclude Comfort Noise (CN) from media
codecs.
Also introduce RtpCodecCapability::IsMediaCodec & ::IsResiliencyCodec
behaving like the MediaCodec methods.
BUG=webrtc:15396
Change-Id: I79041898928190bfdd33a06d8f6975d7556c46b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330424
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41485}
The DcSctpSocket is thread compatible. As long as you serialize accesses
to it - either by calling it from the same thread, or using some kind of
concurrency primitive (e.g. mutex) to avoid calling the API methods from
different threads concurrently, it's fine.
Using the sequence checker, we can verify that the socket is called from
the thread it was created on, or from the same task queue. This provided
a more strict verification, as it didn't allow e.g. creating a socket on
one thread, and then handing it to a different thread where it was used.
Nor did it allow having multiple threads use it, protecting any calls to
it using an external mutex.
One can avoid these checks using webrtc::CurrentTaskQueueSetter to allow
the sequence checker to believe it's running where it's not running, but
this is a hack.
This CL removes the sequence checker in the socket, to simplify using it
in environments that don't use task queues for synchronization. Since it
is still kept in dcsctp::TaskQueueTimeoutFactory, it's still used in all
environments where the task queue is used (e.g. Chrome).
This makes it easier to use dcSCTP without WebRTC.
Bug: None
Change-Id: I2674d7cd902bad45ed3d0816c908ecf3ee971727
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333801
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41482}
* changing the defaults to unified-plan,
* expect unified-plan by default
* only include the msid-semantic line in answers when the
offer contained a sign of msid being used
BUG=webrtc:10421
Change-Id: I79364c2dd801d37adea119375e4dc01edf55b3be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331800
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41480}
This is a reland of commit b39c2a8464c48306a495f14beccf431b91e51efd
Original change's description:
> FrameCadenceAdapter: align video encoding to metronome
>
> This CL aligns the video encoding tasks to metronome tick which
> similar with the metronome decoding.
>
> Design doc: https://docs.google.com/document/d/18PvEgS-DehClK6twCSCATOlX-j9acmXd-3vjb0tR9-Y
>
> Bug: b/304158952
> Change-Id: I262bd4a5097fdaeed559b9d7391a059ae86e2d63
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327460
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
> Cr-Commit-Position: refs/heads/main@{#41469}
Bug: b/304158952
Change-Id: Icf4e1ad91f5c98f3c32a88ffe4d6277e907353e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333464
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41479}
To allow custom FecController use propagated rather than global field trials
note that there is one FecControllerFactory per peer connection factory,
but FecController is created per peer connection and may use per peer connection field trials.
Bug: webrtc:10335
Change-Id: Id25bfaf4b49d4f6d551730c8fd55596ddc49ab47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41478}
same functionality can be achieved by using TaskQueueBase directly
Bug: webrtc:14169
Change-Id: Id7c27afdbf61b4cab3d006d9a7b690b915759cdb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333440
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41476}
The source code checked out on a Linux desktop
has CRLF line endings.
$ git grep -Il $'\r'
Bug: None
Change-Id: I9dfb388346561821f33b21dc6dbed9abb4445aa6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333580
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41475}
The intent is to catch compile errors when tracing is turned off.
Bug: None
Change-Id: I48a42e53e7bd8053984b7d0687b1f6abe538cba1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333760
Reviewed-by: Christoffer Dewerin <jansson@google.com>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41474}
With intention to propagte it futher into RtcEventLogEncoderNewFormat
where it can replace usage of the global field trials
Same environment can be saved in RtcEventLogImpl itself wthere it can
replace usage of the global clock
Bug: webrtc:10335
Change-Id: Ia147d7073af5aab54190fdf192cd5c046c3d40a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330423
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41472}
This CL aligns the video encoding tasks to metronome tick which
similar with the metronome decoding.
Design doc: https://docs.google.com/document/d/18PvEgS-DehClK6twCSCATOlX-j9acmXd-3vjb0tR9-Y
Bug: b/304158952
Change-Id: I262bd4a5097fdaeed559b9d7391a059ae86e2d63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327460
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Cr-Commit-Position: refs/heads/main@{#41469}
as this includes the deprecated timeKillEvent symbol which leads
to runtime errors on platforms where this is already removed.
See discussion in
https://webrtc-review.googlesource.com/c/src/+/328860
for why this causes problems.
BUG=webrtc:15656
Change-Id: I95d07ceed105d35ac76fe97dbd1c454de398f52e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333260
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41468}
This is a reland of commit 63d03f586bb668f72113b61030ec0930aa192010
Original change's description:
> Unify access to SDP codec parameters
>
> which come from the a=fmtp:<pt> lines in the SDP and were used as either
> std::map<std::string, std:string>
> with three aliases,
> cricket::CodecParameterMap
> SdpAudioFormat::Parameters
> SdpVideoFormat::Parameters
>
> Use webrtc::CodecParameterMap in all places.
>
> BUG=None
>
> Change-Id: If47692bde7347834c349c6539b43309d8770e67b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41375}
Bug: None
Change-Id: I5f8f45688df232eb37b12fa3e56a893a1c754e17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331402
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41467}
