This commit replaces the internal use of DurationMs, with millisecond
precision, to webrtc::TimeDelta, which uses microsecond precision.
This is just a refactoring. The only change to the public API is
convenience methods to convert between DurationMs and webrtc::TimeDelta.
Bug: webrtc:15593
Change-Id: Ida861bf585c716be5f898d0e7ef98da2c15268b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325402
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41062}
(NOTE: This and dependent CLs will be reverted in a few days after
data collection from the field is complete.)
This change introduces a new task queue concept, Voucher. They
are associated with a currently running task tree. Whenever
tasks are posted, the current voucher is inherited and set as
current in the new task.
The voucher exists for as long as there are direct and indirect
tasks running that descend from the task where the voucher was
created.
Vouchers aggregate application-specific attachments, which perform
logic unrelated to Voucher progression. This particular change adds
an attachment that measures time from capture to all encode operations
complete, and places it into the WebRTC.Video.CaptureToSendTimeMs UMA.
An accompanying Chrome change crrev.com/c/4992282 ensures survival of
vouchers across certain Mojo IPC.
Bug: chromium:1498378
Change-Id: I2a27800a4e5504f219d8b9d33c56a48904cf6dde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41061}
Update most of the webrtc tests to use EnableMediaWithDefaults instead of SetMediaEngineDefaults
Bug: webrtc:15574
Change-Id: I489a09e4ea3479dc26829ee0c1235e67bcbca7c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325485
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41059}
To ensure padding, we increase 1 bit instead of 1kbps to avoid that 1kbps adds up over time.
Not have unit test for this, but did manual/hamrit tests many times.
Bug: webrtc:12707
Change-Id: I9b3160ab1808cb3a21ff0609446359a4ec3a4949
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325520
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41056}
Build fix for H265 on Android so that https://webrtc-review.googlesource.com/c/src/+/325482 can land.
gn args:
target_os = "android"
proprietary_codecs = true
Bug: webrtc:15620
Change-Id: I8a134afbc50137ac17ce9a4a57d68dd3f3c6d52f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325483
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41053}
The OnFrameToEncode probe had no END in passthrough mode and it
resulted in infinitely long OnFrameToEncode TRACE events.
We now exclude the probes altogether in passthrough mode.
Bug: webrtc:15456
Change-Id: Ia96a5d2b1f5b5470527e904a3ab07de5aa712ca4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325401
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41051}
instead of requiring to pass in call_factory and media_engine
webrtc users should set media_factory member and media dependencies into PeerConnectionFactoryDependencies
Bug: webrtc:15574
Change-Id: I2dc584fe7afa41c9f170bdc51533396155cdcb06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325320
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41049}
This CL just moves the definition and adds a forward.
Actually using the new definition is left for later CLs.
Bug: webrtc:15622
Change-Id: I6d97ef45b98f9eb193c59dd7f8a89c99cfe0ba9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325381
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41048}
The change is under field trial use_in_start_phase.
Bug: webrtc:12707
Change-Id: I2ba8245c5d126b3c8a2e54b826853d98aad6e4f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325184
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41047}
Increasing BWE by 1kbps should be safe/no-op in practice, and it ensures that padding in kIncreasing state will be triggered.
Bug: webrtc:12707
Change-Id: I82493d07a80abd60c93d9cff74baf0a55e77f2b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325286
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41046}
When a timer expires, it can optionally return a new expiration value.
Clearly, that value can't be zero, as that would make it expire
immediately again.
To simplify the interface, and make it easier to migrate to
rtc::TimeDelta, change it from an optional value to an always-present
value that - if zero - means that the expiration time should be
unchanged.
This is just an internal refactoring, and not part of any external
interface.
Bug: webrtc:15593
Change-Id: I6e7010d2dbe774ccb260e14fd6b9861c319e2411
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325281
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41045}
...for signaling and worker thread members in BaseChannel classes.
Bug: webrtc:15099
Change-Id: I83611ed2564e143aca19d0f12ce060b77eb9d2a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325260
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41041}
Before this change, a timer could have an optional max duration. Either
that value was present, and that limited the max duration of the timer,
or it was absl::nullopt, which represented "no limit".
To simplify the interface, this CL makes that value "not optional" by
having it always present. The previous "no limit" is now represented by
DurationMs::InfiniteDuration.
This is just a refactoring of internal interfaces - public interfaces
are left untouched.
Bug: webrtc:15593
Change-Id: I80df1d9b2f4d208411ce6cb5045db0a57865e3b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325280
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41040}
No functionality that depends on this information has been identified; no tests break when it is taken out.
Bug: webrtc:15224
Change-Id: I37298479c6b8a4acb82f59d32130c105371936b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324760
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41037}
The new reference time contains a monotonically increasing clock time and represents the time when the frame was captured. Not all platforms provide the "true" sample capture time in |reference_time| but might instead use a somewhat delayed (by the time it took to capture the frame) version of it.
Bug: webrtc:15539
Change-Id: I95eff8b0f7bff8d3ae65798bf82046e1ac2b0cf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325261
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Cr-Commit-Position: refs/heads/main@{#41036}
Despite being in an "internal" header, IceTransportInternal is already
exported and used outside WebRTC. IceConfig is a counterpart to
IceTransportInternal, so they should be either exported or not exported
together.
See
https://chromium-review.googlesource.com/c/chromium/src/+/4980065/comment/a3a77a56_6d6c2c84/
Bug: chromium:1394755, webrtc:15609
Change-Id: I750d0de81da6ad50fade15d8f7cc57b1ca89e4be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: David Benjamin <davidben@webrtc.org>
Auto-Submit: David Benjamin <davidben@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41029}
Removed thread safety: for a low level helper it adds overhead that users may not need. In particular RtpSenderVideo doesn't need it because calls to SendVideo are already synchronized.
Added a feature to force producing extension as requested by downstream.
Cleanup and document api:
Changed rtp_frequency type to int as it has no reason to use uint32_t per style guide
Changed absolute_capture_time to NtpTime to clarify both units and offset of the time. NtpTime has trivial conversion to/from uint64_t
Documented all the parameters.
Cleanup tests.
Bug: b/307553606
Change-Id: I0922ca4d3c89f124eeb561742dca79ed5c2327fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325022
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/main@{#41023}
RegisterReceivedPacketCallback is used instead of
sigslot::SignalReadPacket. The callback use a new data class ReceivedPacket that combine meta
data and packet payload from a received packet.
This is the first step in an attempt to cleanup the data types used in
the packet receive pipeline.
Eventually, the ReceivedPacket class can contain more meta data such as
ECN information.
Bug: webrtc:11943,webrtc:15368
Change-Id: I984c561b9262fe4aa00176529bd8d901adf66640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325060
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41021}
