This reverts commit 9d0730942677a520ce7e184d081b4c5a2469fc48.
Reason for revert: Speculative revert due to failing downstream test. If the test recovers, I'll assign the issue to the tests owners.
Original change's description:
> red: generate and parse the red fmtp format
>
> generates a fmtp line like
> a=fmtp:<red payloadtype> <opus payloadtype>/<opus payloadtype>
> and matches the incoming redundant payload types against the
> send codec one. Offers without an FMTP line will not use RED.
> Redundancy levels of 1 (plus main packet ) to 32 are accepted but
> this is not wired up to the encoder since the O/A semantic of
> RFC 2198 is not clear.
>
> This decreases the chance of a collision with the SATIN codec
> which also runs on 48khz (but so far does not specify a channelCount of 2)
>
> BUG=webrtc:11640
>
> Change-Id: I8755e5b1e944d105212f1bbe4f330cf4e0753e67
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229583
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34848}
TBR=henrik.lundin@webrtc.org,hta@webrtc.org,philipp.hancke@googlemail.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I5a0816a22a2a213679ab047c61e3b1dda40c4f59
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230140
Reviewed-by: Björn Terelius <terelius@google.com>
Commit-Queue: Björn Terelius <terelius@google.com>
Cr-Commit-Position: refs/heads/main@{#34850}
generates a fmtp line like
a=fmtp:<red payloadtype> <opus payloadtype>/<opus payloadtype>
and matches the incoming redundant payload types against the
send codec one. Offers without an FMTP line will not use RED.
Redundancy levels of 1 (plus main packet ) to 32 are accepted but
this is not wired up to the encoder since the O/A semantic of
RFC 2198 is not clear.
This decreases the chance of a collision with the SATIN codec
which also runs on 48khz (but so far does not specify a channelCount of 2)
BUG=webrtc:11640
Change-Id: I8755e5b1e944d105212f1bbe4f330cf4e0753e67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229583
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34848}
Setting different number of temporal layers is supported by SimulcastEncodeAdapter and LibvpxVp8Encoder will fallback to SimulcastEncoderAdapter if InitEncode fails.
Bug: none
Change-Id: I8a09ee1e6c70a0006317957c0802d019a0d28ca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228642
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34785}
turning the current field trial into a killswitch.
Note that RED is not used by default since it is listed after opus in the SDP.
To enable RED for opus the setCodecPreferences can be used to change
the order of codecs.
BUG=webrtc:11640
Change-Id: I248f4340ca0a3f7c4ea6d6a41b566bc92ab6f19d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228426
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34781}
When layers are activated/deactivated via UpdateActiveSimulcastLayers,
the flag wasn't being updated. This resulted in calls to Stop() getting
ignored after an implicit start via activating layers.
Bug: chromium:1234779
Change-Id: I4a72e624874526d27d3e97d6903112367c5e77fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227700
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34654}
Some hardware H.264 encoders does not place average QP delta in
slice_qp_delta field. Adding an optional flag in EncoderInfo to notify
quality scaler about this.
Bug: webrtc:12942
Change-Id: I3ee29c5ae9bd7bb34d26eba7e6bede3798ca44b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226921
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34627}
stats.rtp_stats.jitter is a RTP timestamp so we needed to convert it back to regular timestamps
See https://bugs.chromium.org/p/webrtc/issues/detail?id=12980#c7
Bug: webrtc:12980
Change-Id: I0d94a22e043ac6ecec4926d950abbdcf787b7168
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227100
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Di Wu <meetwudi@gmail.com>
Cr-Commit-Position: refs/heads/master@{#34590}
rtp_rtcp_format is lighter build target than rtc_media_base and
a more natural place to keep rtp parsing functions.
Bug: None
Change-Id: Ibcb5661cc65edbdc89a63f3e411d7ad1218353cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226330
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34504}
Software fallback wrapper now reports least common multiple of requirements
for two encoders.
SimulcastEncoderAdapter queries actual encoder before InitEncode call
and requests alignment for all layers if simulcast is not supported by
any of the encoders.
Bug: chromium:1084702
Change-Id: Iaed8190737125d447036b6c664b863be72556a5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225881
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34466}
Remove unused functions GetRtpHeader/GetRtpHeaderLength
Replace usage of SetRtpHeader with webrtc::RtpPacket
Move SetRtpSsrc next to the only place it is used.
Bug: None
Change-Id: I3ecc244b1a2bdb2d68e0dbdb34dd60160a3101f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225547
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34447}
This is a step in the direction of being able to make configuration
changes without having to tear down and reconstruct the object
during renegotiation.
Bug: none
Change-Id: If594fd41f3a561060f64212c479a25d19adf8598
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223740
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34402}
* Remove unnecessary decoder factory pointer.
* Set video decoder factory in the ctor of the config class.
* Prepare SetRecvParameters for not needing RecreateWebRtcVideoStream.
Bug: none
Change-Id: I48fbf2920c9fe50f3995ceab5667eb2f70618f25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223067
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34351}
This builds on a few other CLs that avoid recreating the audio receive
streams on config changes and removes redundant config state in
WebRtcAudioReceiveStream, constructs the embedded receive stream in the
initializer list and keeps it const.
Bug: webrtc:11993
Change-Id: Iad28e0170bee6bf1e08713a89af7c81435b4265e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222100
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34317}
This is a consistent way to get to common config parameters for
all receive streams and avoids storing a copy of the extension
headers inside of Call. This is needed to get rid of the need of
keeping config and copies in sync, which currently is part of why
we repeatedly delete and recreate audio receive streams on config
changes.
Bug: webrtc:11993
Change-Id: Ia356b6cac1425c8c6766abd2e52fdeb73c4a4b4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222040
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34285}
Also including common Rtp config members.
Follow up changes will remove the ReceiveRtpConfig class in Call
and copy of extension headers, instead use the config directly
from the receive streams and not require stream recreation for changing
the headers.
Bug: webrtc:11993
Change-Id: I29ff3400d45d5bffddb3ad0a078403eb102afb65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221983
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34283}
This is to be consistent with how things work on the video side but
also much less drastic than the current implementation. Aim is to
remove RecreateAudioReceiveStream(), which would improve efficiency
as well as allow for specific handling of the cases that currently
trigger recreation.
Bug: webrtc:11993
Change-Id: Ia81a5e66d44e41ea4eb2bff800e0b1583821c96a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34282}
This CL adds a cooldown of 0.5 seconds where if the WebRtcVideoChannel
created an unsignalled receive stream within that amount of time, if we
receive even more unknown ssrcs we simply drop those RTP packets.
This prevents getting into a state of spawning new decoders on every
single packet which could happen e.g. if PT based demuxing is enabled
and MIDs are missing from the packets.
Bug: webrtc:12815
Change-Id: Id7675fb0cbfbc72281dcfe030d1a35629df3eb9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221520
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34263}
This reverts commit 8a18e5b3c954a3f9cc006c90356a3d850bcc352f.
Reason for revert: Removing the problematic DCHECK.
Original change's description:
> Revert "Remove AudioReceiveStream::Reconfigure() method."
>
> This reverts commit e2561e17e29e62c02731f1d214d7ee5ffdaeb941.
>
> Reason for revert: Speculative revert: breaks an downstream project
>
> Original change's description:
> > Remove AudioReceiveStream::Reconfigure() method.
> >
> > Instead, adding specific setters that are needed at runtime:
> > * SetDepacketizerToDecoderFrameTransformer
> > * SetDecoderMap
> > * SetUseTransportCcAndNackHistory
> >
> > The whole config struct is big and much of the state it holds, needs to
> > be considered const. For that reason the Reconfigure() method is too
> > broad of an interface since it overwrites the whole config struct
> > and doesn't actually handle all the potential config changes that might
> > occur when the config changes.
> >
> > Bug: webrtc:11993
> > Change-Id: Ia5311978f56b2e136781467e44f0d18039f0bb2d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221363
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34252}
>
> TBR=saza@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I15ca2d8ee5fd612e13dc1f4b3bfb9c885c21dc66
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11993
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221746
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34253}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:11993
Change-Id: I0d3bf9abdcdc8d3f9259d014e6074a5e6b6cc73c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221747
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34255}
This reverts commit e2561e17e29e62c02731f1d214d7ee5ffdaeb941.
Reason for revert: Speculative revert: breaks an downstream project
Original change's description:
> Remove AudioReceiveStream::Reconfigure() method.
>
> Instead, adding specific setters that are needed at runtime:
> * SetDepacketizerToDecoderFrameTransformer
> * SetDecoderMap
> * SetUseTransportCcAndNackHistory
>
> The whole config struct is big and much of the state it holds, needs to
> be considered const. For that reason the Reconfigure() method is too
> broad of an interface since it overwrites the whole config struct
> and doesn't actually handle all the potential config changes that might
> occur when the config changes.
>
> Bug: webrtc:11993
> Change-Id: Ia5311978f56b2e136781467e44f0d18039f0bb2d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221363
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34252}
TBR=saza@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I15ca2d8ee5fd612e13dc1f4b3bfb9c885c21dc66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11993
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221746
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34253}
Instead, adding specific setters that are needed at runtime:
* SetDepacketizerToDecoderFrameTransformer
* SetDecoderMap
* SetUseTransportCcAndNackHistory
The whole config struct is big and much of the state it holds, needs to
be considered const. For that reason the Reconfigure() method is too
broad of an interface since it overwrites the whole config struct
and doesn't actually handle all the potential config changes that might
occur when the config changes.
Bug: webrtc:11993
Change-Id: Ia5311978f56b2e136781467e44f0d18039f0bb2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221363
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34252}
UpdateActiveSimulcastLayers has been blocking
WebRtcVideoChannel::SetSend which may be called quite frequently during
negotiations. This CL changes UpdateActiveSimulcastLayers to not
synchronize with the transport's task queue to wait for the changes to
get applied.
This synchronization is quite costly, but so too are other remaining
things in VideoSendStream, so we should aim to get rid of the
`thread_sync_event_` in VideoSendStream.
Bug: webrtc:12840, webrtc:12854
Change-Id: Idb48d29b6b8382881c7c1e6f1d0f5e708dbca30f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221203
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34228}
Both flexfec and AV1 seem not to have created interop issues and falling
back to the lower range is better than skipping the codecs.
BUG=webrtc:12295
Change-Id: I58459133beae4f17b767af92a4e2c9028ab8cbe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217888
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/master@{#34182}
Starting new audio codecs from the top of the lower range
reduces collisions with video codecs which are assigned from
the bottom of the lower range
BUG=webrtc:11640
Change-Id: If6d2b849b8e1de777a1d4352df533e4f1845fde9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220022
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34166}
for consistency with the definition in IsRtcpPacket which takes
into account a collision of H261 feedback for payload types 64 + 65:
https://datatracker.ietf.org/doc/html/rfc5761#section-4
BUG=webrtc:12194
Change-Id: I2ebb0456ae2aff1b1735f26221c7c4ae79698ac9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220021
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34131}
With this CL, SimulcastEncoderAdapter no longer configures its encoder
as multi-layered if we only have a single active layer. Instead we
create a single single-layered encoder for that one and only active
layer. When using VP8 SW encoder this means that LibvpxVp8Encoder is
configured to only prepare a single video frame which avoids the cost of
scaling down to layers that we do not send. (A multi-layered
LibvpxVp8Encoder is required to scale even layers we don't encode.)
When profiling this CL I found very small but measurable gains for
representative downscale factors of 20.1 ms of 60 s profile. This is
just 0.0335% CPU so it's not much, but skipping a downscale might be
worth a lot more if we have to map/unmap buffers or do GPU round-trips
in the future (which I have not measured).
When downscaling to factors 4 and 2 due to libyuv having a
"fast-path" for these (i.e. no adaptation active), zero difference was
found for NV12. For I420 there was small regression of 16.1 ms
(0.026% CPU) for this one edge-case. It's possible to work around this,
but considering the tiny changes we're talking about, I really don't
think it's worth the additional complexity. I'll file a bug on libyuv
about scaling factors 2+2 vs 4 and leave it at that.
Bug: webrtc:12603
Change-Id: Id462140c6a829cf6b460baae868e94243f477db3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219683
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34092}
its content is duplicated in the report_block_data member
Bug: webrtc:10678
Change-Id: I89421ae4ab5f727a233161924372105e222ed404
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219080
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34039}
This is one step in getting rid of cricket::MediaType.
Bug: webrtc:12754
Fixes: webrtc:12764
Change-Id: Idee832572bdc4c0e3bfdec6fb31ec0ba9db3e995
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218346
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33994}
Pending messages on network thread for MediaChannel, will be dropped
when the MediaChannel object is deleted (without blocking).
Remove MessageHandler inheritance from Channel since Post-ing to the
network thread has been removed from there.
Copy/pasted code for SendRtp/SendRtcp in WebRtcVideoChannel and
WebRtcVoiceMediaChannel consolidated in MediaChannel.
Bug: webrtc:11993
Change-Id: I05320eb7f86b98adba50ca5eb8b76b78f4111263
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217720
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33955}
This CL mostly adds plumbing to get awareness of the network thread
to the media channel classes. Currently this pointer is only used
to DCHECK that `SetInterface` for the `NetworkInterface` pointer, is
called on the network thread. Follow up changes will establish that
most of the methods are called on the network thread and the mutex
in the MediaChannel base class, can be removed.
Most of the changes in the CL are in channel_unittest.cc. They're mostly
around updating the tests to incorporate the network thread in ways
that reflect how the classes are used in production. Another change is
to use accessor methods for the media channel instances instead of
caching potentially dangling pointers.
Bug: webrtc:11993
Change-Id: I8e2ed1bc23724e238554dbce386789d69660f7e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217682
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33951}
This is related to upcoming changes whereby it will be enforced that
calls to SetInterface(<valid ptr>) and SetInterface(nullptr) be matched
up correctly.
Bug: webrtc:11993
Change-Id: Ic022f9487a7ab297adaced8e620e2384e055673b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217241
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33903}
There's a bit of copy/pasted code in the channel code, which is
making moving network traffic consistently over to the network thread
a bit trickier than it needs to be, so I'm also updating variable
names used in Set[Local|Remote]Content_w to be more explicitly the same
and make it clear that the code is copy/pasted (and future updates can
consolidate more of it).
Also removing some code from the video/voice media channels that's
effectively dead code (vector + registration methods that aren't needed)
Bug: webrtc:12705
Change-Id: I2e14e69fbc489a64fc1e8899aaf1cfc979fe840b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215978
Reviewed-by: Sam Zackrisson <saza@google.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33847}
