And a corresponding struct RtpReceiveStats. This is intended
to hold the information exposed via GetStats, which is quite
different from the stats reported to the peer via RTCP.
This is a preparation for moving ReceiveStatistics out of the
individual receive stream objects, and instead have a shared instance
owned by RtpStreamReceiverController or maybe Call.
Bug: webrtc:10679,chromium:677543
Change-Id: Ibb52ee769516ddc51da109b7f2319405693be5d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148982
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28943}
Storing a fixed amount of packets does not work well with DTX since the history could include up to 20 seconds of packets which can potentially be negative in the event of clock drift or delay shifts.
Bug: webrtc:10333
Change-Id: Ifb8543b7e999e17845cb0e4171066862941f370e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149832
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28942}
Deprecated the field BitrateAllocationUpdate::link_capacity since it is only
used by the Opus codec in order to smooth the target bitrate, which is
equivalent to the stable_target_bitrate field.
The unused field trial WebRTC-Bwe-StableBandwidthEstimate is also removed.
Bug: webrtc:10126
Change-Id: Ic4a8a9ca4202136d011b91dc23c3a27cfd00d975
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149839
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28941}
The received frames statistics currently include also frames
that are dropped because they are duplicated, incomplete, or
the buffer being full. After this CL only frames that are
added to the decode queue are counted.
This CL is part of fixing the dropped frames statistics that
are currently also counting frames that are in the decode
queue.
Bug: chromium:990317
Change-Id: I7df31939ecb7b9e222086e1141a15420fa2819dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150108
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28939}
This is a reland of 81c0cf287c8514cb1cd6f3baca484d668c6eb128
Original change's description:
> Simplification and refactoring of the AudioBuffer code
>
> This CL performs a major refactoring and simplification
> of the AudioBuffer code that.
> -Removes 7 of the 9 internal buffers of the AudioBuffer.
> -Avoids the implicit copying required to keep the
> internal buffers in sync.
> -Removes all code relating to handling of fixed-point
> sample data in the AudioBuffer.
> -Changes the naming of the class methods to reflect
> that only floating point is handled.
> -Corrects some bugs in the code.
> -Extends the handling of internal downmixing to be
> more generic.
>
> Bug: webrtc:10882
> Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}
Bug: webrtc:10882
Change-Id: I2ddf327e80a03468c41662ae63c619ff34f2363a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28938}
This change adds helper classes to manipulate Absolute Capture Time header extensions. Both classes support the "timestamp interpolation" optimization.
Bug: webrtc:10739
Change-Id: I08eff46eb8910842a6dbaa3288b976004fabe1c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149801
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28936}
It's not removed from VideoBitrateAllocationParameters as that struct
is part of the API.
Bug: webrtc:9883
Change-Id: I69f683e3c1dc3a0edc1711f6289514b86b05ad77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149815
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28935}
It's propagated from ReceiveStatistics up to VoiceReceiverInfo,
and then not used. It's not part of the standard stats.
Bug: None
Change-Id: I90ce6a72e3ca846adbbba5d3023fef18a2169018
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149164
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28933}
PeerConnection now has a new setting in RTCConfiguration to enable use of
datagram transport for data channels. There is also a corresponding field
trial, which has both a kill-switch and a way to change the default value.
PeerConnection's interaction with MediaTransport for data channels has been
refactored to work with DataChannelTransportInterface instead.
Adds a DataChannelState and OnStateChanged() to the DataChannelSink
callbacks. This allows PeerConnection to listen to the data channel's
state directly, instead of indirectly by monitoring media transport
state. This is necessary to enable use of non-media-transport (eg.
datagram transport) data channel transports.
For now, PeerConnection watches the state through MediaTransport as well.
This will persist until MediaTransport implements the new callback.
Datagram transport use is negotiated. As such, an offer that requests to use
datagram transport for data channels may be rejected by the answerer. If the
offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data
channels with an extra x-opaque parameter for datagram transport. If the
opaque parameter is rejected (by an answerer without datagram support), the
offerer may fall back to SCTP.
If DTLS is not enabled, there is no viable fallback. In this case, the data
channels are negotiated as media transport data channels. If the receiver does
not understand the x-opaque line, it will reject these data channels, and the
offerer's data channels will be closed.
Bug: webrtc:9719
Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28932}
This reverts commit 81c0cf287c8514cb1cd6f3baca484d668c6eb128.
Reason for revert: internal test failures
Original change's description:
> Simplification and refactoring of the AudioBuffer code
>
> This CL performs a major refactoring and simplification
> of the AudioBuffer code that.
> -Removes 7 of the 9 internal buffers of the AudioBuffer.
> -Avoids the implicit copying required to keep the
> internal buffers in sync.
> -Removes all code relating to handling of fixed-point
> sample data in the AudioBuffer.
> -Changes the naming of the class methods to reflect
> that only floating point is handled.
> -Corrects some bugs in the code.
> -Extends the handling of internal downmixing to be
> more generic.
>
> Bug: webrtc:10882
> Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: I2729e3ad24b3a9b40b368b84cb565c859e79b51e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10882
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150084
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28931}
This flag has been default-off since Jul 24th (m77 branch) and apart
from a bug fixed on Aug 5th, there have been no reports of issues, so
let's remove it and start cleaning away the old code path.
Most of the usage within RtpSender/PacingController and their
respective unit tests are removed with this CL, but there will be
several more to follow.
Bug: webrtc:10633
Change-Id: I1986ccf093434ac8fbd8d6db82a0bb44f50b514e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149838
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28930}
This CL changes the behavior of APM for 8 kHz so that it is internally
processed as 16 kHz.
Bug: webrtc:10863
Change-Id: Ie17de6551c6e984b60534820374a49ca298f06ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148800
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28929}
This CL performs a major refactoring and simplification
of the AudioBuffer code that.
-Removes 7 of the 9 internal buffers of the AudioBuffer.
-Avoids the implicit copying required to keep the
internal buffers in sync.
-Removes all code relating to handling of fixed-point
sample data in the AudioBuffer.
-Changes the naming of the class methods to reflect
that only floating point is handled.
-Corrects some bugs in the code.
-Extends the handling of internal downmixing to be
more generic.
Bug: webrtc:10882
Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28928}
The SetSSRC() method is slated for removal, make sure we set the local
SSRC at construction time.
Bug: webrtc:10774
Change-Id: I431e828caf60c5e0134adbe82d1d3345745cc6ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149827
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28926}
silentConcealedSamples, insertedSamplesForDeceleration and
removedSamplesForAcceleration were implemented in M76, but we forgot to
add them to the WEBRTC_RTCSTATS_IMPL list, meaning the "iterate all
members" method, RTCStats::Members(), did not contain these metrics.
As a consequence, Chrome did not pick up these members for exposure to
JavaScript.
Also fix the test coverage in rtc_stats_integrationtest.cc where code
paths that did not apply to audio track stats were not explicitly
asserting that they must be undefined in those cases.
Bug: chromium:996146, webrtc:10903
Change-Id: I00e7ddee600818ee4d561b88e005391830adcf3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149816
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28925}
This CL changes the VideoFrameDumpingDecoder API to only expose a
factory function creating the wrapper instead of the full class.
Bug: webrtc:10902
Change-Id: I1e7e3a60accea1a7c48207d4262ed4bacacab4a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150040
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28924}
The name media_send_ssrc makes less sense when used mostly for the
RtcpReceiver functionality.
The old member is still there and used as a fallback. That will be
cleaned away after downstream code is fixed.
Bug: webrtc:10774
Change-Id: I4ec18db76910f31dfe76bc9b137ffe89220d3fa8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149836
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28923}
This change adds logic to WindowCapturerWin to capture overlapping
owned/pop-up windows (e.g. menus, dialogs, tooltips). This makes window
capture behavior more consistent regardless of whether
CroppingWindowCapturerWin is used & its conditions for using crop-from-
screen capture are met (in ShouldUseScreenCapturer). (I.e. regardless
of OS version, window shape / translucency, occlusion by another
potentially top-most window, or whether the capturing app has opted in
to using the cropping capturer).
Owned/pop-up windows associated with the selected window are enumerated
then captured individually, with their contents composited into the
final frame.
This change also:
- Crops out the top window border (which exposed a bit of the background
when using the cropping capturer, and resulted in an inconsistent
appearance compared to the side & bottom borders being cropped out).
Bug: chromium:980864
Change-Id: I81c504848a0c0e6bf122aeff437b400e44944718
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148302
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#28922}
DatagramDtlsAdaptor wraps a DatagramTransport in a DtlsTransport. This
is only used by wrapping it again, in an RtpTransport. It is simpler to
just wrap DatagramTransport directly into an RtpTransport.
DatagramTransport is never used as a DtlsTransport, and doesn't support
most of the functionality exposed by the DtlsTransport interface.
However, it supports *all* the functionality of the RtpTransport, making
this a much cleaner fit.
Bug: webrtc:9719
Change-Id: I699e8124ee4cb6c8c187162f9b444ff0431a4902
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149400
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28921}
This CL is a no-op since rtc_use_lto is always false and in general
such change should probably be implemented in
//build/config/compiler/BUILD.gn.
Bug: chromium:408997
Change-Id: Id37d3181e66e699f8cd535aee1af7609352a7259
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149833
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28919}
Before I had assumed that it wasn't critical for these tests
to have the stats polled at a very regular interval but the perf
waterfall disagrees, so I'm accounting for drift when scheduling
the callbacks.
Bug: chromium:993688
Change-Id: If7f1d3919093f97508774c0c635fff6fe5081c10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149809
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28914}
This CL removes all external access to the integer sample data in the
AudioBuffer class. It also removes the API in AudioBuffer that provides this.
The purpose of this is to pave the way for removing the sample
duplicating and implicit conversions between integer and floating point
sample formats which is done inside the AudioBuffer.
Bug: webrtc:10882
Change-Id: I1438b691bcef98278aef8e3c63624c367c2d12e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149162
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28912}
This silences a warning that appeared with iOS 13, and is more efficient
in general.
Bug: webrtc:10866
Change-Id: I23db6b78af36e59b1d825d3f0cccc6008f9b626a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149808
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28911}
This is a reland of 7c6f74ab0344e9c6201de711d54026e9990b8e6c
Compared to the previous commit, new bits are added to log calls of
AddIceCandidate, and the gathering and reception of IPv6 candidates.
Original change's description:
> Set the usage pattern bits for adding remote ICE candidates from SDP.
>
> Currently these bits are only set when a remote ICE candidate is
> successfully added via addIceCandidate. For non-trickled sessions in
> which the remote candidates are added via the remote description, these
> bits are lost. This also happens for trickled sessions, though a rare
> case, when addIceCandidate does not succeed because the peer connection
> is not ready to add any remote candidate.
>
> Bug: webrtc:10868
> Change-Id: Ib2f199f9ffc936060473934d25ba397ef31131a3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148880
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28844}
Bug: webrtc:10868
Change-Id: Ifac0593dcfb64d88619fd24b4ab61c14a0810beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149024
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28904}
That method is going away in favor in construction time setting.
Bug: webrtc:10774
Change-Id: I2aba5a2537e5846a3c9438a5b376b230e84c5f32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149826
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28901}
These tests are now run as part of PacingControllerUnittest instead.
Bug: webrtc:10809
Change-Id: If59e622e8a66565be678106d9341aa6eee78c299
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149803
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28900}
This CL changes the way that values are converted
between fixed and floating point to
-Avoid the former asymmetric conversion causing
nonlinear distortions.
-Reduce the complexity.
In contrast to the initial CL, the DCHECKS on the incoming sample
range was changed to limiting.
Bug: webrtc:6594
Change-Id: I8218dfd5c45388ad5aac61be453d2f28725a2475
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132783
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#28867}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149169
Cr-Commit-Position: refs/heads/master@{#28897}
This change fixes a bug where the initial delay could be set incorrectly.
Bug: webrtc:10896
Change-Id: I66b2234b69c46639488f4561e973384001230861
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149820
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28894}
