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}
after dependencies adopted the RtpMediaContentDescription which
this is currently aliased to.
Also move definition of AudioCodecs and VideoCodecs to the place
where codecs are defined.
BUG=webrtc:15214
Change-Id: I9b0456e1c69c8b23e0cc7665a59baae268872d9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325021
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41020}
If we have been sending padding for 1s and estimate still is unchanged, then stop padding by transitioning to decrease state.
Bug: webrtc:12707
Change-Id: I0dca2e5cd98263fc7fae9882c23c21634413c7a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324740
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41018}
Reasons:
- the code is no longer used in Chrome
- it is conceptually weird for WebRTC to have JSON parsing in its API
- there are concerns around the reliability of the underlying JSON library
Additionally, this CL removes the rtc_json "poisonous" attribute: the scheme is incompatible and redundant with testonly.
Bug: webrtc:1493351
Change-Id: I0b621b0e3f183df7315919d9c89242fbe387928f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325062
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41014}
2 main reasons:
1. Packet sizes are much different thus a lost audio packet should not be treated similar to a lost video packet. In low bandwidth/traffic policing scenario, the number of send packet is few, thus the computed loss can be imprecise.
2. Given a candidate bandwidth estimate, the objective function (how good the candidate is) is computed by recomputing loss rate = send rate/estimate bandwith + inherent loss. It means the objective function is byte based rather than packet based.
Potential risk: the current algorithm params are tuned based on packet count, thus it might not work with byte count, which is much higher than packet count.
The change is under field trial and disabled by default.
Bug: webrtc:12707
Change-Id: I8b832e7920d2b4cadcd4a072b3a4d4f26a213a20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325065
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41013}
This gives the option to synchronize rendering updates with
the display refresh cycle and limit effective updates to a certain frame
rate.
go/meet-android-synchronized-rendering
Bug: b/217863437
Change-Id: I4938a10f4e80d98a17e28f2e397fbb95117a3e4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325061
Reviewed-by: Ranveer Aggarwal <ranvr@webrtc.org>
Commit-Queue: Linus Nilsson <lnilsson@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41012}
Using RTC_DCHECK for test validation is wrong to begin with (gets
compiled out in non-debug builds, which measn we may miss validations),
but becomes extra problematic when we include code with side-effects
inside the DCHECK, which results in release-build tests having a
different flow than intended
Bug: webrtc:15572
Change-Id: I89d5b55f903b9d93fe4a929548d1b9fcde8941be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323182
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41005}
RenderSynchronizer is used to coordinate video rendering updates
to a specific frame rate target and aligned to display refresh cycles.
go/meet-android-synchronized-rendering
Bug: b/217863437
Change-Id: Ie329c4c2eccfb0c9aee9b90f7ddbc370919d5e86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324840
Reviewed-by: Ranveer Aggarwal <ranvr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Linus Nilsson <lnilsson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41004}
Calls the AudioOutput implementation of GetStats, which is currently
not implemented.
Bug: webrtc:14653
Change-Id: Ieaf0e0c030a95d23c8950ff9038a64426142a789
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324800
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41003}
