The RTC event log analyzer would previously only plot network latency
for incoming video streams. (The latency is computed from the capture
time in the RTP header, and the packet receive time.) This CL adds
support for audio packets, which requires estimating the RTP clock
frequency for the incoming packets.
Bug: None
Change-Id: Idf1ff9febfdd4097976b22a61f1c5679deb6068c
Reviewed-on: https://webrtc-review.googlesource.com/c/108784
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25580}
In a handful of places we wait for the old IceConnectionState to reach some value and then we assume that the new connection states have also been updated. However those are updated in response to different events that might not have fired yet, so sometimes these tests will fail.
This change makes us wait explicitly for those states to update.
Bug: webrtc:9983
Change-Id: I5cb6652ee29c0b86c0834174442140a3863e08e4
Reviewed-on: https://webrtc-review.googlesource.com/c/110441
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25579}
Some targets used to be included transitively via ortc.
Since ortc module has been removed (Bug: webrtc:9824),
this CL explicitly add them in main //:webrtc target.
As a result, the following functions are exposed again:
CreateBuiltinVideoDecoderFactory()
CreateBuiltinVideoEncoderFactory()
CreatePeerConnectionFactory()
[...]
Bug: webrtc:9824
Bug: webrtc:9973
Change-Id: Iebfae582f8887bf76338c73fc85c4608e96c3f0d
Reviewed-on: https://webrtc-review.googlesource.com/c/110248
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25576}
This changes PeerConnection to allow sending and receiving data channel
messages over the media transport. If |use_media_transport_for_data_channels|
is set, PeerConnection will use a DCT_MEDIA_TRANSPORT mode for data
channels.
DCT_MEDIA_TRANSPORT acts exactly like DCT_SCTP within the data channel
and peer connection layers. On the transport layer, it uses the media
transport instead of SCTP. It appears as an RTP data channel in SDP
(just as media over media-transport appears as RTP in SDP).
Bug: webrtc:9719
Change-Id: I6a90142bd3f43668479c825ed02689dcd0d58b78
Reviewed-on: https://webrtc-review.googlesource.com/c/109740
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25575}
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782
This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.
Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.
One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.
Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
GetEncoderInfo() is now called every frame, so we should not do
expensive parsing or logging in there. Instead, prepare an EncoderInfo
instance in InitEncode() and just return that in GetEncoderInfo().
Bug: webrtc:9890
Change-Id: Idc9e79e681c6f7ff4f9b446aa298c156f25bc6f6
Reviewed-on: https://webrtc-review.googlesource.com/c/110161
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25569}
This will allow clients to include only the software codecs they need
rather than being forced to bundle them all.
- libjingle_peerconnection_jni keeps its allow_poison for now, until
dependent targets bundle their own codecs explicitly.
- native_api_codecs and native_api_video lose their allow_poison
because dependent targets are already bundling codecs explicitly.
- libjingle_peerconnection_metrics_default_jni and
native_api_peerconnection lose their allow_poison because they
were not actually poisoned.
legacy_hwcodecs_jni and default_video_codec_factory_jni exist for
clients that want to continue bundling the same codecs they get by
default today.
Bug: webrtc:7925
Change-Id: Idf853a6bc77f43decd35ad2a0f467937fec8f8b5
Reviewed-on: https://webrtc-review.googlesource.com/c/108221
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#25564}
The .cc source files listed below #ifdef for WEBRTC_CODEC_OPUS and
WEBRTC_CODEC_ILBC but the build files don't include the defines.
modules/audio_coding/neteq/tools/neteq_test.cc
modules/audio_coding/neteq/tools/neteq_test_factory.cc
Bug: None
Change-Id: I6065021f68e58d0e5663acd006a9865bf265adc0
Reviewed-on: https://webrtc-review.googlesource.com/c/109925
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25555}
Determining the EncoderInfo meta data is now done during InitEncode().
This implementation assums that no dynamic wrappers are wrapped in this
simulcast encoder adapter. Ie, if supports_native_handle changes,
InitEncode() must be called again for it to be reported properly.
Bug: webrtc:9722
Change-Id: I7901effe11e89ac011659a4ea862ab2a42577eb5
Reviewed-on: https://webrtc-review.googlesource.com/c/109620
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25549}
This is a workaround for the case when there are no video frames in a
call for a very long time, such that RTP timestamps wraparound and
FrameBuffer can't figure out if the frame is older or newer.
Bug: webrtc:9974
Change-Id: Ie1eaa4938813dbbd637ddcbe7ff118ead2bfa4a9
Reviewed-on: https://webrtc-review.googlesource.com/c/109882
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25548}
This reverts commit 61c6e5643e7ea058e653956980a90e033249c055.
Reason for revert: downstream projects prepared for this change
Original change's description:
> Revert "Isolating APM API build target: making :api an actual target."
>
> This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.
>
> Reason for revert: breaking downstream
>
> Original change's description:
> > Isolating APM API build target: making :api an actual target.
> >
> > This CL is part of a refactoring work to unblock other CLs
> > that would generate a circular dependency when including
> > modules/audio_processing. It will also allow to easily move
> > the APM interface part under //api.
> >
> > More in detail, this change moves the APM interface files from
> > the build target modules/audio_processing to
> > modules/audio_processing:api. It also adds :api as dependency
> > where needed.
> >
> > Bug: webrtc:9535
> > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25539}
>
> TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
>
> Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9535
> Reviewed-on: https://webrtc-review.googlesource.com/c/109820
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25540}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109884
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25547}
A preparation for deleting EventFactory and EventWrapper, to instead
use rtc::Event directly.
Bug: webrtc:3380
Change-Id: I4c40daca9268e57b06d506d91e09365091c42ad6
Reviewed-on: https://webrtc-review.googlesource.com/c/109880
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25545}
We want clients to be able to build their own factories around these
codecs.
Bug: webrtc:7925
Change-Id: Ia8f62d5d85e63ac6e3eb402c5996d8b986625615
Reviewed-on: https://webrtc-review.googlesource.com/c/109529
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#25543}
This removes the constructor that takes all members as parameters.
A main goal with this struct is to make it easy to add or remove fields
without having to refactor implementations and usage all over the palce,
this breaks down if we need to update the arguments in this constructor
every time.
Also cleans up a few commented-out lines that were forgotten in an
earliers cl.
Bug: webrtc:9722
Change-Id: I22db35d399469bdb3bfd8bdc531b24ead9fe1c3e
Reviewed-on: https://webrtc-review.googlesource.com/c/109570
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25542}
This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.
Reason for revert: breaking downstream
Original change's description:
> Isolating APM API build target: making :api an actual target.
>
> This CL is part of a refactoring work to unblock other CLs
> that would generate a circular dependency when including
> modules/audio_processing. It will also allow to easily move
> the APM interface part under //api.
>
> More in detail, this change moves the APM interface files from
> the build target modules/audio_processing to
> modules/audio_processing:api. It also adds :api as dependency
> where needed.
>
> Bug: webrtc:9535
> Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25539}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109820
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25540}
This CL is part of a refactoring work to unblock other CLs
that would generate a circular dependency when including
modules/audio_processing. It will also allow to easily move
the APM interface part under //api.
More in detail, this change moves the APM interface files from
the build target modules/audio_processing to
modules/audio_processing:api. It also adds :api as dependency
where needed.
Bug: webrtc:9535
Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
Reviewed-on: https://webrtc-review.googlesource.com/c/109501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25539}
cricket::Port::SignalPortComplete.
The mDNS name registration is asynchronously executed by the mDNS
responder, and a host candidate with an mDNS name is only gathered after
this completes. SignalPortComplete however is currently done
synchronously by UDPPort, and any candidate gathered by a UDPPort after
this signal is fired would be discarded.
Bug: webrtc:9964, webrtc:9605
Change-Id: If8aaf193ef26c06bd118e6418b62ba0de5e87e3c
Reviewed-on: https://webrtc-review.googlesource.com/c/109541
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Zach Stein <zstein@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25534}
