There have been several bugs where the members of PlayoutDelay were
zero initialized when handling RTP packets without the corresponding
extensions. Initializing to {-1, -1} (meaning not provided) is less
brittle.
Bug: None
Change-Id: I196850377128d5e67a19bdaf9298403b2e9f5a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/111181
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25670}
The header modules/audio_device/include/audio_device_default.h was not
owned by any build target.
Bug: webrtc:8946
Change-Id: I3266a613c10963688c3bea701384e1d1bb68daac
Reviewed-on: https://webrtc-review.googlesource.com/c/111201
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25669}
Usage of templates makes it harder for tooling to help the user. This
can be experienced when trying to investigate compile failures and using
editor tools to browse the code.
This CL replaces usage of templates with injection of unique pointers to
interfaces that implements the behavior that previously was assumed by
the templated implementation.
Bug: webrtc:9883
Change-Id: Ica17af9646f68a9b063988f9e85d6acc8ca37c10
Reviewed-on: https://webrtc-review.googlesource.com/c/106703
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25668}
This makes the currently implicit interfaces explicit and
prepares for making CompositeMediaEngine non-templated.
Bug: webrtc:9883
Change-Id: I57452acc9ada60a801f6d624894440a942c12ded
Reviewed-on: https://webrtc-review.googlesource.com/c/106940
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25667}
This is a preparation for deleting ChannelSendProxy. Signature is
changed on a couple of methods. Unused methods
EnableAudioNetworkAdaptor, DisableAudioNetworkAdaptor,
SetReceiverFrameLengthRange and RtpRtcpModulePtr are deleted. Some
methods are demoted to private: SendData, SendRtp, SendRtcp,
PreferredSampleRate, Sending, and OnOverheadChanged.
Bug: webrtc:9801
Change-Id: I982e72418a32e66fb5de410350b1bfebd9a3219c
Reviewed-on: https://webrtc-review.googlesource.com/c/110605
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25666}
Replaced by an interface ChannelReceiveInterface, implemented
by ChannelReceive and the corresponding mock class.
Moved thread checkers to ChannelReceive. That class is moved to the
anonymous namespace in the .cc file, and exposed only via a function
CreateChannelReceive.
Bug: webrtc:9801
Change-Id: Iecacbb1858885bf86da9484f2422e53323dbe87a
Reviewed-on: https://webrtc-review.googlesource.com/c/110610
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25665}
Currently (and this has to change), media transport is created two times if audio&video is used (even if bundling is enabled).
The second time it's destroyed really quickly (but given lack of 'Connect' method, the connection has already started).
This change adds a TODO and modifies existing tests to prevent creation of 2 media transports.
Bug: webrtc:9719
Change-Id: I872e98dcd10685beb0326d501f0e0abf36c0fdfc
Reviewed-on: https://webrtc-review.googlesource.com/c/110887
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25660}
This reverts commit 586725dc9a508c7d3e82b5a625a5ee7e8b1a4e17.
Reason for revert: misses a check to see if the optional callback is implemented.
Original change's description:
> Add ios bindings for PeerConnectionState.
>
> This change makes it possible for ios apps to use the new standards-compliant PeerConnectionState.
>
> Bug: webrtc:9977
> Change-Id: Icf69bb1faa0383ae239cb7508f2a740a2d489697
> Reviewed-on: https://webrtc-review.googlesource.com/c/110502
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25651}
TBR=kthelgason@webrtc.org,jonasolsson@webrtc.org
Change-Id: Iff919e9876e6b8dddc6d8ab7df302081d0cfa917
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9977
Reviewed-on: https://webrtc-review.googlesource.com/c/111062
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25659}
This adds a presubmit check that warns missing memory.h inclusion
when a source file use absl::make_unique. That header tends to be
included transitively on pre-C++17 mode, but doesn't on C++17 mode.
Bug: chromium:752720
Change-Id: I235287f4f7407d48bfad35da86da47bc602f03ce
Reviewed-on: https://webrtc-review.googlesource.com/c/111040
Commit-Queue: Taiju Tsuiki <tzik@chromium.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25655}
This support is needed if there is a big delay between the creation of
frames and the time they are delivered to the WebRTC C++ layer in
AndroidVideoTrackSource. This is the case if e.g. some heavy video
processing is applied to the frames that takes a couple of hundred
milliseconds. Currently, timestamps coming from Android video sources
are aligned to rtc::TimeMicros() once they reach the WebRTC C++ layer in
AndroidVideoTrackSource. At this point, we "forget" any latency that
might occur before this point, and audio/video sync consequently
suffers.
Bug: webrtc:9991
Change-Id: I7b1aaca9a60a978b9195dd5e5eed4779a0055607
Reviewed-on: https://webrtc-review.googlesource.com/c/110783
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25654}
This change makes it possible for android apps to use the new standards-compliant PeerConnectionState.
Bug: webrtc:9977
Change-Id: Iad19c38e664a59e86879715ec7a04a59a9894bee
Reviewed-on: https://webrtc-review.googlesource.com/c/109883
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25652}
This change makes it possible for ios apps to use the new standards-compliant PeerConnectionState.
Bug: webrtc:9977
Change-Id: Icf69bb1faa0383ae239cb7508f2a740a2d489697
Reviewed-on: https://webrtc-review.googlesource.com/c/110502
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25651}
Fixing bug that allows a channel to be set on a stopped transceiver.
This CL contains the following refactoring:
1. Extracted ChannelInterface from BaseChannel
2. Unified SetXxxMediaChannel (Voice, Video) into SetMediaChannel
Bug: webrtc:9932
Change-Id: I2fbf00c823b7848ad4f2acb6e80b1b58ac45ee38
Reviewed-on: https://webrtc-review.googlesource.com/c/110564
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25641}
We are currently trying to print a nice "μs" to the log, but this often
ends up as a weird character. This CL replaces the unicode 'μ' to a
simple ascii 'u'.
TBR=sakal
Bug: None
Change-Id: Ibe90e0d2f12004676fc531aec0a2b33d59a8cb3f
Reviewed-on: https://webrtc-review.googlesource.com/c/110608
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25636}
The new parser provides the same functionality (with a slightly
different API) and is backwards compatible with the legacy wire format.
Downstream projects seem to have transitioned to the new parser API.
Bug: webrtc:8111
Change-Id: Icb458f0d55e0a4566c4b7b4a53cab48f0b9d6fd8
Reviewed-on: https://webrtc-review.googlesource.com/c/110782
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25635}
Expected checksums depend on whether libopus is built with SSE or not.
Since we have no robust way to know that and we cannot enforce all
clients to use SSE, we accept both results.
Bug: webrtc:9530
Bug: webrtc:9995
Change-Id: I9f0464ffec15df91eafe15d89c61e2140f341cb1
Reviewed-on: https://webrtc-review.googlesource.com/c/110789
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25633}
Downstream projects need to be able to configure ANA without hacking or redefining protos.
Bug: webrtc:9719
Change-Id: Idd80471066ff41a9265adbdb738cc98cc97b2e6e
Reviewed-on: https://webrtc-review.googlesource.com/c/110765
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25629}
Those alias do not save much typing, but may cause conflicts, specially the one in the header
Bug: None
Change-Id: Ifb17f639e528aaff72861ff55dcd7a96a229715d
Reviewed-on: https://webrtc-review.googlesource.com/c/110784
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25628}
So far the code assumed that there is only one implementation of AudioSourceInterface: LocalAudioSource.
That is not true. This change allows custom implementations to still set options (such as audio network adaptation) on the source.
Long term solution should include refactoring options so that they are passed to peer connection or call object, and not be defined on audio source.
Bug: webrtc:9719
Change-Id: Ic3b92219502bc73a964adbbb9c5cd7156aa382e1
Reviewed-on: https://webrtc-review.googlesource.com/c/110681
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25626}
Specifically, the tests that only use data channels shouldn't need any
audio codec support; by using an audio encoder factory that supports
no codecs, we ensure that this is the case.
(The tests were already using empty *de*coder factories; however, it
was only recently that it became possible to use empty *en*coder
factories as well.)
Bug: webrtc:7529
Change-Id: Ied84283fe88073704a66bc82007b0dfcd7bf377f
Reviewed-on: https://webrtc-review.googlesource.com/c/110726
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25625}
All implementations of ModifyReceiverCallConfig and
ModifySenderCallConfig configure the bitrate_config member only. So
replace these methods by ModifyReceiverBitrateConfig and
ModifySenderBitrateConfig.
This is a preparation for injecting RtpTransportControllerSend via
CallConfig. Then bitrates should be passed when constructing
RtpTransportControllerSend, and they can be deleted from CallConfig.
Bug: webrtc:7135
Change-Id: I6714158bd463dd485018713d0e26815919e5afcc
Reviewed-on: https://webrtc-review.googlesource.com/c/110780
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25624}
This tool does not seem useful enough to justfy the maintenance cost.
If we want something like this in the future, then the core logic
should be added to the parser.
Bug: webrtc:8111
Change-Id: Ifc3dc9b91e85246d35d7775c68d0f2dc687516aa
Reviewed-on: https://webrtc-review.googlesource.com/c/110724
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25623}
