Tested with a Web Platform Test; the test added here is useful, but
does not exercise the bug.
Bug: chromium:959128
Change-Id: Ia2e7f9e015b2345dd02d341b0fe27f58b64aa81e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135575
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27889}
The main purpose right now of this CL is to avoid the situation
where multiple retransmissions are queued for sending (normally after
network glitch with increased pacer queue length), and some of those
fail sending because the can't be retrieved from the packet history
due to too short time since last sent.
Bug: webrtc:8975, webrtc:10607
Change-Id: I9f6369d83f0b8208e5f57b2dc2fd3f2db7c6fea1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135164
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27884}
The state this set tracks (whether this is new feedback for a packet
belonging to a media ssrc) can already be inferred from data provided
by the SendTimeHistory: if packet send time is not populated in the
feedback it's either because:
1. The feedback has already been processed
2. The receiver is sending feedback for bogus non-existent packets
If the first case, this maps to |feedback_packet_seq_num_set_|
containing the packet, if the ssrc (present in the feedback) is part
of the media ssrcs.
In the second case, this data should be ignored anyway.
Bug: webrtc:10604
Change-Id: If4828091142d68baa8dbb62be9d0b24ccaaa9546
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135163
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27882}
This CL makes it possible to configure the priority of audio streams in
bitrate allocations using field trials.
It also adds the option to forcibly ignore any injected audio allocation
strategy, so that experimentation with allocation won't be blocked on
the work to remove the strategy injection.
Bug: webrtc:10603
Change-Id: Ic36ceee6c15eb0fad275866f77e2a121066e516c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135467
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27881}
This CL removes the usage of absl::flat_hash_map because it transitively
depends on CCTZ which fails to link with lld-link after the switch to
libc++.
Since std::map doesn't support heterogeneous lookup until C++14, this
CL also stops using absl::string_view and switches to
`const std::string&`.
Bug: webrtc:10605
Change-Id: I4fc93969c6fc0cc7e7e62b4d2f801bdd27cff0f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135566
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27877}
It's more convenient to have it next to other freeze-related metrics
when sorted alphabetically.
Bug: None
Change-Id: I8f73e32a9cdb8166e139c193ffcd1dcc1fa18533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135563
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27874}
This will allow call audio to function when audio session attributes like `preferredInputNumberOfChannels` cannot be set due to intermittent OS errors.
Bug: webrtc:10602
Change-Id: Ie9f3e58a6ab54a26a9bd795575d16c3a9fe5c65f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135440
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27871}
If the receiver has indicated that a packet has been received, via a
TransportFeedback RTCP message, it is safe to remove it from the
RtpPacketHistory as we can be sure it won't be needed anymore.
This will reduce memory usage, reduce the risk of overflow in the
history at very high bitrates, and hopefully make payload based padding
a little more useful.
This is code stems partly from
https://webrtc-review.googlesource.com/c/src/+/134208
but without the RtpPacketHistory changes which were landed in
https://webrtc-review.googlesource.com/c/src/+/134307
Bug: webrtc:8975
Change-Id: Iea9d3d32bee5512473744e9ef3a18018567fc272
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135160
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27868}
This was added in CL 135122, but the bitratePriority parameter is not
standard and not implemented in a way users would expect. So it should
actually not be exposed in the Obj-C SDK.
Bug: webrtc:10438
Change-Id: I801ce940a32701d2703e951ef2b601c606aa2111
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135287
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27861}
optional<int> min_frames: The minimum number frames to observe to make a
scaling decision.
Default: kMinFramesNeededToScale in quality_scaler.cc
optional<double> initial_scale_factor: The sample period scale factor.
Default: kSamplePeriodScaleFactor in quality_scaler.cc
optional<double> scale_factor: Option to use a reduced sampling interval when
last check did not result in an adaptation (if
unset the initial_scale_factor is used).
Bug: none
Change-Id: I3bb955d1f8d7d7d49bc118361614b5aa59605231
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135125
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27860}
This reverts commit 14b2758726879d21671a21291dfed8fb4fd5c21c.
Reason for revert: Internal import failed.
Original change's description:
> Version 2 "Refactoring DataContentDescription class"
>
> (substantial changes since version 1)
>
> This CL splits the cricket::DataContentDescription class into
> two classes: cricket::RtpDataContentDescription (used for RTP data)
> and cricket::SctpDataContentDescription (used for SCTP only).
>
> SctpDataContentDescription no longer inherits from
> MediaContentDescriptionImpl, and no longer contains "codecs".
>
> Due to usage of internal interfaces by consumers, shimming the old
> DataContentDescription API is needed.
>
> A new cricket::DataContentDescription class is defined, which is
> a shim over RtpDataContentDescription and SctpDataContentDescription.
> It exposes as little functionality as possible, but supports the
> concerned consumer's usage
>
> Design document:
> https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
>
> Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
>
> Bug: webrtc:10358
> Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27853}
TBR=danilchap@webrtc.org,steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,shampson@webrtc.org
Change-Id: Ibc16ba14c1cbf50345a9b79151b79df140482539
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135280
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27855}
(substantial changes since version 1)
This CL splits the cricket::DataContentDescription class into
two classes: cricket::RtpDataContentDescription (used for RTP data)
and cricket::SctpDataContentDescription (used for SCTP only).
SctpDataContentDescription no longer inherits from
MediaContentDescriptionImpl, and no longer contains "codecs".
Due to usage of internal interfaces by consumers, shimming the old
DataContentDescription API is needed.
A new cricket::DataContentDescription class is defined, which is
a shim over RtpDataContentDescription and SctpDataContentDescription.
It exposes as little functionality as possible, but supports the
concerned consumer's usage
Design document:
https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
Bug: webrtc:10358
Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27853}
Add support for potentially out-of-order removals of packets, using a
vector of sequence numbers that have been acknowledges as received.
Additionally, make kStoreAndCull storage method by default with a
field-trial kill-switch if things go wrong unexpectedly.
Bug: webrtc:8975
Change-Id: I6da8b92d85fc362c12db82976f115626cb1d32d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134307
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27850}
This CL removes the redundant enable flags from AEC2 and AECM
Bug: webrtc:5298
Change-Id: Icc575abf1c368dda02ca77f057d166f1c921f662
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135100
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27848}
(Re-land reverted cr).
Use (in order from lowest to highest precedence):
-- fixed 32000bps
-- fixed target bitrate from codec
-- explicit values from the rtp encoding parameters
-- Final precedence is given to field trial values from
WebRTC-Audio-Allocation
Bug: webrtc:10487
Change-Id: I573e996fa1f243e673785cdbe687e029fd5cbf4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133483
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Daniel Lee <dklee@google.com>
Cr-Commit-Position: refs/heads/master@{#27847}
