Implements a two-pass approach to packetization which creates
packets of an even size similar to RtpPacketizer::SplitAboutEqually.
This improves the bandwidth estimation.
The algorithm does a first pass with the existing packetizer, then
iterates through the resulting packet sizes and sums up the bytes left unused in each packet.
It then calculates a new maximum packet length as
configured_max_packet_len - ((unused_bytes - packets + 1) / packets)
adjusts for the overhead and re-runs the packetization algorithm.
For example, a list of OBUs with sizes
{1206, 1476, 1431}
currently gets packetized greedily as payload sizes
{1200, 1200, 1200, 523}
With this change, it gets packetized as
{1032, 1032, 1032, 1028}
This change is guarded by the field trial
WebRTC-Video-AV1EvenPayloadSizes
which is acting as a rollout flag.
BUG=webrtc:15927
Co-authored-by: Shyam Sadhwani <shyamsadhwani@meta.com>
Change-Id: I4f0b3c27de6f06104908dd769c4dd1f34115712c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348100
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42203}
Start introducing ArrayView to AudioFrame and code that flows down
from there. In this first step:
* Add `data_view()` that returns a read-only ArrayView for the
audio buffer. When AudioFrame is not initialized however, data_view()
will return a nullptr whereas the current data() method never returns
nullptr.
* Add `mutable_data()` that requires two arguments for properly setting
the samples per channel and number of channels that's required for
accurately reserving the returned mutable ArrayView.
A notable behavior change is that if the requested number of channels
is larger than supported or the calculated buffer size is too large,
the function will trigger a check.
* Add TODOs for following work.
Bug: chromium:335805780
Change-Id: I2937de800422589ebe6a3840b3caadf3d9ff8b00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347982
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42202}
Instead of from the global field trial string.
Bug: webrtc:42220378
Change-Id: Iddb41429e388792de02f702b4caa35689c57d9e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347720
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42201}
This prevents clangd from complaining about unused includes from
test/gmock.h and test/gtest.h
Bug: b/42226242
Change-Id: I2bd0f61f63981dff697d60f353d198fd81ab1457
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#42200}
This is the first new style trace event so this CL adds and updates
WebRTCs Perfetto configuration.
* Changes all #includes to target "third_party/perfetto". Added this
to DEPS.
* Expose the Perfetto public config in the "tracing" group using
an all_dependent_configs statement. This means the config is included
in all parts that include the "//:tracing" group. However, direct
perfetto includes are banned per DEPS.
* In order to expose Perfetto types (ie Flow/TerminatingFlow) the
perfetto headers are a dependancy on all targets. This should not
affect binary size as these are not used when perfetto is not enabled
and will not be linked.
Bug: b/42226290
Change-Id: I5711d81dae95ee907cbcd41bf1ee9b55d2ec595c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349161
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#42197}
This CL also creates a docs/monorail-bog-tracker-migration where
we can add more information (if needed) about the migration.
No-Try: True
Bug: None
Change-Id: Ieec36a793daf8e30c6181f7cd595fce922948838
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349323
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42195}
These are aliases for cricket::Codec.
Also remove internal usage
Bug: b/42225532
Change-Id: I220b95260dc942368cb6280432a058159eec8700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349321
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42194}
Calculate the RMS audio level of audio packets being sent before
invoking an encoded frame transform, and pass them with the encode frame
object.
Before this, the audio level was calculated at send time by having rms_levels_ look at all audio samples encoded since the last send. This
is fine without a transform, as this is done synchronously after
encoding, but with an async transform which might take arbitrarily long,
we could end up marking older audio packets with newer audio levels, or
not at all.
This also makes things work correctly if external encoded frames are
injected from elsewhere to be sent, and exposes the AudioLevel on the
TransformableFrame interface.
Bug: chromium:337193823, webrtc:42226202
Change-Id: If55d2c1d30dc03408ca9fb0193d791db44428316
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349263
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#42193}
Old target and call/simulated.h exist but refer to new target in test/network.
Bug: webrtc:14525
Change-Id: Ida04cef17913f2f829d7e925ae454dc40d5e8240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349264
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42191}
This migration was done semi-automatically. I didn't manage to find any
corresponding bug ID for chromium:413437 nor chromium:949536 in the new
issue tracker. Since these are policy-exempt anyway I opted for setting
the ID to NO_BUG and leaving a comment with the old ID.
Bug: None
Change-Id: If2d212ba554e40c42193b51f62a7da8a7f783d41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349267
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42190}
In its stead, there's now a SetEventCallback() method.
Bug: webrtc:11943
Change-Id: If936d6e1e23e8a584f06feb123ecf2d450ea4145
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319040
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42187}
This is the first step in implementing custom codecs in SDP.
Bug: none
Change-Id: I7789478208a769eaefd58b410ae6f488c604594d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348662
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#42171}
Since the code measuring the RTT has been converted to using TimeDelta
which internally stores the duration in microseconds, from DurationMs
which uses milliseconds, the RTO calculation can use the higher
precision to calculate lower non-zero durations on really fast networks
such within a data center.
Before this CL, which is from the initial drop of dcSCTP, the RTO
calculation was done using the algorithm from the paper "V. Jacobson:
Congestion avoidance and control", but now we're using the original
algorith from https://tools.ietf.org/html/rfc4960#section-6.3.1, which
comes from https://datatracker.ietf.org/doc/html/rfc6298#section-2.
Two issues were found and corrected:
1. The min RTT variance that is specified in the config file was
previously incorrectly divided by 8. That was not its intention,
but we're keeping that behaviour as other clients have actually
measured a good value to put there. This represents "G" in
the "basic algorithm" above, and since that is multiplied with K,
which is four, the default value of 220 wouldn't make sense if it
wasn't scaled down, as that would make the RTO easily saturate to
the RTO_max (800ms).
2. The previous algorithm had large round-off errors (probably because
the code used milliseconds), which makes fairly big changes to the
calculated RTO in some situations.
Bug: webrtc:15593
Change-Id: I95a3e137c2bbbe7bf8b99c016381e9e63fd01d87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349000
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42170}
The integration_test_helpers.h file was too long and had too many
big functions inline.
This CL takes some of the largest and puts them in the .cc file.
Bug: None
Change-Id: Ibaaf9675ca8b5efa29878b4883b21f14104451a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349020
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42169}
This adds Perfetto support to WebRTC with a GN flag rtc_use_perfetto.
The configuration of perfetto depends on whether or not webrtc is
build within Chrome or not. When in Chrome, WebRTC will depend on
//third_party/perfetto:libperfetto. When building standalone, specific includes required for Perfetto are exposed with the library webrtc_libperfetto.
The perfetto trace API is exposed with a header export in
trace_event.h which is used instead of the legacy API.
The addition of Perfetto means there are 4 compilation modes for
tracing in WebRTC,
1. No tracing implementation.
2. Legacy tracing (AddTraceEvent/GetCategoryEnabled).
3.a. Perfetto statically linked (webrtc_libperfetto).
3.b. Perfetto in Chrome (Chrome's libperfetto).
This CL removes the tracing expectations from
rtc_stats_integrationtest.cc because those directly used the old API.
Integration into Chrome is a follow up CL which depends on
https://chromium-review.googlesource.com/c/chromium/src/+/5471691.
Tested: Ran Chrome with Perfetto and traces appear. WebRTC Unit test tracing working: https://ui.perfetto.dev/#!?s=04ea2613ea36b814394639a1ec4b60be5b5097527f1a485995ecc13469885468
Bug: webrtc:15917
Change-Id: I537d79dc247c2b759689910c621087286a4d8fdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347880
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mikhail Khokhlov <khokhlov@google.com>
Cr-Commit-Position: refs/heads/main@{#42166}
the rollout has happened a while ago with no issues requiring the use
of the killswitch
BUG=chromium:40066610
Change-Id: I2c8148976a1da219ebbfbe6908224b6384348194
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348823
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42164}
Environment provides non-null interface for FieldTrialsView and thus VideoCodingModule no longer need to rely on FieldTrialBasedConfig class to provide field_trials when not passed at construction.
Bug: webrtc:10335
Change-Id: Iedfb29e8b29056618a85f2e7a1528da29e3be5c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347701
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42163}
This changes Flag to not inherit from a virtual interface.
Also fixing iwyu and build dependencies.
Bug: none
Change-Id: Iba6e095ec771d8975a32059041185270d32e51be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348940
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42162}
Landing when last remaining usage in Chrome has been removed.
Bug: webrtc:11943
Change-Id: I62817e2cc0b67113126b82424b6f843c77e66f31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341001
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42157}
