This was an oversight when integrating datagram-based data channels into
JsepTransportController. If a DatagramTransport exists, but only to be
used for data channels, JsepTransportController will still create an RTP
transport for it and use it for RTP.
Bug: webrtc:9719
Change-Id: I93cdb8bfc03159882a83a9f5097d3ef99fed215f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150241
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28993}
This reverts commit 87bed4793ff8f463202f442381339626d0b27f0d.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Reland "Preserve min and max playout delay from RTP header extension"
>
> This reverts commit f31cc08ba01ed403e89255b5f3f38d5dbdde855e.
>
> Reason for revert: Reland with fixes
>
> Original change's description:
> > Revert "Preserve min and max playout delay from RTP header extension"
> >
> > This reverts commit 85ba9972c42544c4771e394c9aa1d20bf5d09a91.
> >
> > Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream.
> >
> > Original change's description:
> > > Preserve min and max playout delay from RTP header extension
> > >
> > > Audio and video synchronization can sometimes override the minimum
> > > and maximum playout delay that is set through the RTP header
> > > extension. This CL makes sure that the playout delay always is
> > > within the limits set by the RTP header extension.
> > >
> > > Bug: webrtc:10886
> > > Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#28980}
> >
> > TBR=stefan@webrtc.org,kron@webrtc.org
> >
> > Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:10886
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28984}
>
> TBR=stefan@webrtc.org,kron@webrtc.org
>
> Change-Id: I5a3908a8c45f7faedab6f009b22df81d674e13a0
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10886
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150653
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28985}
TBR=stefan@webrtc.org,kron@webrtc.org
Change-Id: Id2e5d1ff804881e956a07fa4ae0f8301895dcc95
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10886
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150654
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28986}
This reverts commit f31cc08ba01ed403e89255b5f3f38d5dbdde855e.
Reason for revert: Reland with fixes
Original change's description:
> Revert "Preserve min and max playout delay from RTP header extension"
>
> This reverts commit 85ba9972c42544c4771e394c9aa1d20bf5d09a91.
>
> Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream.
>
> Original change's description:
> > Preserve min and max playout delay from RTP header extension
> >
> > Audio and video synchronization can sometimes override the minimum
> > and maximum playout delay that is set through the RTP header
> > extension. This CL makes sure that the playout delay always is
> > within the limits set by the RTP header extension.
> >
> > Bug: webrtc:10886
> > Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28980}
>
> TBR=stefan@webrtc.org,kron@webrtc.org
>
> Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10886
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28984}
TBR=stefan@webrtc.org,kron@webrtc.org
Change-Id: I5a3908a8c45f7faedab6f009b22df81d674e13a0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10886
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150653
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28985}
This reverts commit 85ba9972c42544c4771e394c9aa1d20bf5d09a91.
Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream.
Original change's description:
> Preserve min and max playout delay from RTP header extension
>
> Audio and video synchronization can sometimes override the minimum
> and maximum playout delay that is set through the RTP header
> extension. This CL makes sure that the playout delay always is
> within the limits set by the RTP header extension.
>
> Bug: webrtc:10886
> Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28980}
TBR=stefan@webrtc.org,kron@webrtc.org
Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10886
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28984}
Audio and video synchronization can sometimes override the minimum
and maximum playout delay that is set through the RTP header
extension. This CL makes sure that the playout delay always is
within the limits set by the RTP header extension.
Bug: webrtc:10886
Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28980}
The packets belonging to a frame were kept in PacketBuffer
until the frame was decoded. This CL clears the dependencies
of an existing RtpFrameObject to PacketBuffer so that we can
free up PacketBuffer as soon as the RtpFrameObject is created.
Bug: none
Change-Id: Ic939be91815519ae1d1c67ada82006417b2d26a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149818
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28977}
Number of received FEC bytes is used for the
WebRTC.Video.FecBitrateReceivedInKbps UMA histogram. Before this cl,
that value is based on a FEC packet counter updated by
ReceiveStatistics::FecPacketReceived. This cl deletes that method, and
instead adds a byte count to the FecPacketCounter struct, which is
maintained by the UlpFecReceiver and used for other FEC-related stats.
Bug: webrtc:10917
Change-Id: I24bd494b6909a2fe109d28e2b71ca8f413d05911
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150533
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28976}
Previously the kDontRetransmit value was used to indicate packets that
should not be retransmitted but were put in the RtpPacketHistory anyway
as a temporary storage while waiting for a callback from PacedSender.
Since PacedSender now always owns the delayed packets directly, we can
remove all usage of StorageTye in RtpPacketHistory, and only put
packets there after pacing if RtpPacketToSend::allow_retransmission()
returns true.
Bug: webrtc:10633
Change-Id: I003b76ba43bd87658cc2a39e908fd28ebcd403f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150521
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28974}
The performance cost is not trivial but according to my profiling,
it is acceptable.
Bug: b/139745386
Change-Id: I0e63221ccf22e9f6fb32c630ff63a279e765994a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150539
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28973}
Add separate setting for configuring min bitrate that only applies when
adapting up in resolution.
Bug: none
Change-Id: I83d33ac3110a22602065b8d83130e3f619cb1eba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150329
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28970}
The |frames_dropped| statistics contain not only frames that are dropped
but also frames that are in internal queues. This CL changes that so
that |frames_dropped| only contains frames that are dropped.
Bug: chromium:990317
Change-Id: If222568501b277a75bc514661c4f8f861b56aaed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150111
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28968}
This patch adds a new (optional) attribute to TURN_ALLOCATE_REQUEST,
TURN_LOGGING_ID (0xFF05).
The attribute is put into the comprehension-optional range
so that a TURN server should ignore it if it doesn't know if.
https://tools.ietf.org/html/rfc5389#section-18.2
The intended usage of this attribute is to correlate client and
backend logs.
Bug: webrtc:10897
Change-Id: I51fdbe15f9025e817cd91ee8e2c3355133212daa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149829
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28966}
This CL removes the old non-paced code path and instead uses a helper
class to just immediately pass the packet through the same code path as
when an actual pacer is used.
Bug: webrtc:10633
Change-Id: Id9a3ee4719829ad07710f5468e5452aa4bc8570b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150530
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28963}
The methods are no longer in use, this CL cleans away references and
updates any tests using them.
Bug: webrtc:10633
Change-Id: I2db301e0a021a2f85a8b9a74e409303baba407da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28956}
This CL corrects the way the audio processing module handles sample rates that
don't allow partitioning the data into evenly sized 10 ms chunks, examples
being 22050 Hz and 11025 Hz.
Bug: webrtc:10882
Change-Id: I35d738f8a0e1debc443fe5d473c0d666a7ba8d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150526
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28953}
This CL removes and replaces the legacy fixed-point high-pass filter in
APM with the floating point high-pass filter in AEC3.
Bug: webrtc:10907
Change-Id: I88cf8f622ab139e4ffa97f89a72425aa3becfc58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150103
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28950}
This is a reland of b7b8e30cb44c41f51dbbefb9a9160e6dfe869c5b
Original change's description:
> Reland Process 8 kHz audio as 16 kHz internally of the audio processing module
>
> This CL relands the code from the CL "Process 8 kHz audio as 16 kHz internally
> of the audio processing module" which by mistake was reverted via a rebase in
> another CL.
>
> The CL changes the behavior of APM for 8 kHz so that it is internally
> processed as 16 kHz.
>
> Bug: webrtc:10863
> Change-Id: I32a57b2d279c2134125667c19b09cfda381a33c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150221
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28944}
Bug: webrtc:10863
Change-Id: Ic626b99b099248f0d8a677dc4cfe1505e14ae7cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150330
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28949}
This reverts commit b7b8e30cb44c41f51dbbefb9a9160e6dfe869c5b.
Reason for revert: Broke ApmTest.Process test in internal iOS waterfall
Original change's description:
> Reland Process 8 kHz audio as 16 kHz internally of the audio processing module
>
> This CL relands the code from the CL "Process 8 kHz audio as 16 kHz internally
> of the audio processing module" which by mistake was reverted via a rebase in
> another CL.
>
> The CL changes the behavior of APM for 8 kHz so that it is internally
> processed as 16 kHz.
>
> Bug: webrtc:10863
> Change-Id: I32a57b2d279c2134125667c19b09cfda381a33c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150221
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28944}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: Ia49e07b0c25c49da646917516e317f1d57cc4e84
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10863
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150326
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28948}
Some devices have issues decoding the resolutions that result when using 4
as a factor.
Bug: webrtc:9381
Change-Id: I5055923ca318a1bde62bcefb452cae8f33165e43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150102
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28945}
