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}
Before this change, there was no way for a client to indicate to the
dcSCTP library if a packet that was supposed to be sent, was actually
sent. It was assumed that it always was.
To handle temporary failures better, such as retrying to send packets
that failed to be sent when the send buffer was full, this information
is propagated to the library.
Note that this change only covers the API and adaptations to clients.
The actual implementation to make use of this information is done as a
follow-up change.
Bug: webrtc:12943
Change-Id: I8f9c62e17f1de1566fa6b0f13a57a3db9f4e7684
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228563
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34767}
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}
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}
jitterBufferDelay and jitterBufferEmittedCount are defined
in RTCMediaStreamStats for both audio and video.
But for video, they were not populated in RTCInboundRtpStreamStats.
Bug: webrtc:12910
Change-Id: I135d473f055ecfb2c39b078ccf18c1bb9bc4f210
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224280
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34398}
When the transport is terminated, if an error has occured, it will
be propagated to the channels.
When such errors can happen at the SCTP level (e.g. out of resources),
RTCError may contain an error code matching the definition at
https://www.iana.org/assignments/sctp-parameters/sctp-parameters.xhtml#sctp-parameters-24
If the m= line is rejected or removed from SDP, an error will again be sent
to the data channels, signaling their unexpected transition to closed.
Bug: webrtc:12904
Change-Id: Iea3d8aba0a57bbedb5d03f0fb6f7aba292e92fe8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223541
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34386}
* 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}
The factory allows us to isolate the implementation from users who only
need to depend directly on the public folder now.
Bug: webrtc:12614
Change-Id: Ied09cf772ed427eaf17a7b5705f587da57405640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220939
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34330}
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 usrsctp_dumppacket which prefixes its output with a newline.
This makes the packets easier to grep and process with text2pcap.
BUG=webrtc:12614
Change-Id: I67bc2e0026250b21b030daf967ebc697640f2d7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220102
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/master@{#34114}
This reverts commit 096ad02c02b4bc6c046282b8793ef84d041dd0d8.
Reason for revert: Including a fix for the test issue.
Original change's description:
> Revert "Fix race between enabled() and set_enabled() in VideoTrack."
>
> This reverts commit 5ffefe9d2d743c66f8a8bcbc5ad9662a3138840a.
>
> Reason for revert: Breaks Chromium Android browser tests on fyi bots.
>
> Original change's description:
> > Fix race between enabled() and set_enabled() in VideoTrack.
> >
> > Along the way I introduced VideoSourceBaseGuarded, which is equivalent
> > to VideoSourceBase except that it applies thread checks. I found that
> > it's easy to use VideoSourceBase incorrectly and in fact there appear
> > to be tests that do this.
> >
> > I made the source object const in VideoTrack, as it already was in
> > AudioTrack, and that allowed for making the GetSource() accessors
> > bypass the proxy thread hop and give the caller direct access.
> >
> > Bug: webrtc:12773, b/188139639, webrtc:12780
> > Change-Id: I022175c4239a1306ef54059c131d81411d5124fe
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34096}
>
> TBR=mbonadei@webrtc.org,tommi@webrtc.org,landrey@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I16323d459c76eb6a87cc602a0048f6ee01c81626
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12773
> Bug: b/188139639
> Bug: webrtc:12780
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219637
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#34101}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12773
Bug: b/188139639
Bug: webrtc:12780
Change-Id: Ib35fe15a6c43de8f286d60aff02b19df1ab76925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219639
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34104}
This reverts commit 5ffefe9d2d743c66f8a8bcbc5ad9662a3138840a.
Reason for revert: Breaks Chromium Android browser tests on fyi bots.
Original change's description:
> Fix race between enabled() and set_enabled() in VideoTrack.
>
> Along the way I introduced VideoSourceBaseGuarded, which is equivalent
> to VideoSourceBase except that it applies thread checks. I found that
> it's easy to use VideoSourceBase incorrectly and in fact there appear
> to be tests that do this.
>
> I made the source object const in VideoTrack, as it already was in
> AudioTrack, and that allowed for making the GetSource() accessors
> bypass the proxy thread hop and give the caller direct access.
>
> Bug: webrtc:12773, b/188139639, webrtc:12780
> Change-Id: I022175c4239a1306ef54059c131d81411d5124fe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34096}
TBR=mbonadei@webrtc.org,tommi@webrtc.org,landrey@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I16323d459c76eb6a87cc602a0048f6ee01c81626
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12773
Bug: b/188139639
Bug: webrtc:12780
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219637
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34101}
Along the way I introduced VideoSourceBaseGuarded, which is equivalent
to VideoSourceBase except that it applies thread checks. I found that
it's easy to use VideoSourceBase incorrectly and in fact there appear
to be tests that do this.
I made the source object const in VideoTrack, as it already was in
AudioTrack, and that allowed for making the GetSource() accessors
bypass the proxy thread hop and give the caller direct access.
Bug: webrtc:12773, b/188139639, webrtc:12780
Change-Id: I022175c4239a1306ef54059c131d81411d5124fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34096}
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}
Reduced the level so that the library can be run with INFO level without
a lot of spam. VERBOSE is still reserved for frequent logs.
Also, using WARNING for logs that are not fatal and which can easily
be triggered by the user.
Bug: webrtc:12614
Change-Id: If09c302b2b5bfc002471f86a8aeb74ba1172c705
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219465
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34054}
This is to address flakiness of "DoubleThread" tests for the media
channel class. More investigation is in order though, so I'm adding
a TODO. The bug appears to be in test code only though, so this is
just to deflake the bots.
Bug: webrtc:12783
Change-Id: Ib6cf78927f2a9be9d2c6aa7f6915b1131a206e7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219460
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34049}
When verbose logs are enabled, SCTP packets will be dumped to debug
logs, allowing text2pcap to be used to generate PCAP files.
First start Chrome with verbose logs, and write those to file:
/path/to/chrome --enable-logging=stderr --v=4 2> out.log
Then extract the SCTP_PACKET traces and run text2pcap:
grep SCTP_PACKET out.log > sctp.log
text2pcap -n -i 132 -D -t '%H:%M:%S.' sctp.log sctp.pcapng
You may have to cut away more from the beginning if the debug logs
contain additional timestamps and more, e.g. like:
grep SCTP_PACKET out.log | cut -d ' ' -f 2- > sctp.log
Note that if there are multiple RTCPeerConnection objects created, each
will print out their packets to log, so to filter for a specific one:
grep "SCTP_PACKET DcSctpTransport0" out.log > sctp.log
Bug: webrtc:12614
Change-Id: Ibbceaf33719d09e7606247cb0496ddd827ea58bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218200
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33999}
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}
Xcode 12.5 triggers some warnings for -Wdeprecated-copy, and I believe
it is better to fix this problem than to suppress this warning.
Bug: webrtc:12749
Change-Id: I5ca5fd8fdcae18fe7d3941f78b3366b5f03b8c00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33990}
