This is not used and adds a lot of maintenance overhead to
the code since it requires that the transport feedback adapter
communicates directly with audio send stream.
This also means that the packet loss tracker used as input for
this can be removed and a lot of wiring up code overall.
Bug: webrtc:9883
Change-Id: I25689fb622ed89cbb378c27212a159485f5f53be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156502
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29667}
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.
Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
The SetSSRC() method is slated for removal, make sure we set the local
SSRC at construction time.
Bug: webrtc:10774
Change-Id: I431e828caf60c5e0134adbe82d1d3345745cc6ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149827
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28926}
Changing the ssrc for a module is intended to be removed, and will in
the future require creating a new instance.
Bug: webrtc:10774
Change-Id: Ie96daa4a8cf00223ea040509037582f6b1c8eb19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145205
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28571}
This is part of implementing RTCRemoteInboundRtpStreamStats. The CL was
split up into smaller pieces for reviewability. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats
In [1], ReportBlockData was implemented and tested.
This CL adds the plumbing to make it available in MediaSenderInfo for
audio streams, but the code is not wired up to make use of these stats.
In follow-up CL [2], ReportBlockData will be used to implement
RTCRemoteInboundRtpStreamStats. The follow-up CL will add integration
tests exercising the full code path.
[1] https://webrtc-review.googlesource.com/c/src/+/136584
[2] https://webrtc-review.googlesource.com/c/src/+/138067
Bug: webrtc:10455
Change-Id: Id8940090cc9c121e8f06ccdb068a22ce24c07092
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138066
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28072}
Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.
TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.
Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}
Before this change the encoder tasks runs on a shared worker queue.
That makes the destruction require synchronization to avoid races.
By keeping a separate encode queue to ChannelSend, we can safely
destruct the object without worrying for left over tasks, as they
will be stopped when the task queue is destroyed.
For TaskQueue implementations using one thread per TaskQueue this
will increase the thread count by the number of AudioSendStreams,
which typically is just one.
This is partly a reland of 9b9344742b
Original change's description:
> Removes lock from ChannelSend.
>
> The lock isn't really needed as encoder_queue_is_active_ can be checked
> on the task queue to provide synchronization.
>
> There is one behavioral change due to this: We will not cancel any currently
> pending encoding tasks when we stop sending, they will be allowed to finish.
>
> Bug: webrtc:10365
> Change-Id: I2b4897dde8d49bc7ee5d2d69694616aee8aaea38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125096
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26963}
Bug: webrtc:10365
Change-Id: Iafb84e25d90ec8639359be80fad65763d08e5719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125740
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27038}
This change takes out responsibility for packetization from the
RtpRtcp class, and deletes the method RtpRtcp::SendOutgoingData.
Video packetization was similarly moved in cl
https://webrtc-review.googlesource.com/c/src/+/123187
Bug: webrtc:7135
Change-Id: I0953125a5ca22a2ce51761b83693e0bb8ea74cd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27000}
Since it's a common pattern it makes sense to explicitly provide the
interface rather than reimplementing it every time it's used.
Bug: webrtc:9883
Change-Id: I4dca84bd7c8616fcbcbaba511718671a3668e743
Reviewed-on: https://webrtc-review.googlesource.com/c/122300
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26664}
In order to have correct overhead adjustments for ANA in AudioSendStream we need to know both transport and audio packetization overheads in AudioSendStream. This change makes AudioSendStream to be OverheadObserver instead of ChannelSend. This change is required for https://webrtc-review.googlesource.com/c/src/+/115082.
This change was also suggested earlier in TODO:
// TODO(solenberg): Make AudioSendStream an OverheadObserver instead.
https://cs.chromium.org/chromium/src/third_party/webrtc/audio/channel_send.cc?rcl=71b5a7df7794bbc4103296fcd8bd740acebdc901&l=1181
I think we should also consider moving TargetTransferRate observer to AudioSendStream. Since AudioSendStream owns encoder and configures ANA, it makes sense to consolidate all rate (and overhead) calculation there. Added TODO to clean it up in next chanelists.
Bug: webrtc:10150
Change-Id: I48791b998ea00ffde9ec75c6bca8b6dc83001b42
Reviewed-on: https://webrtc-review.googlesource.com/c/119121
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26540}
Rids can now be sent using rtp_sender.
Hooking up the rid values in the voice and video engine is still WIP.
Bug: webrtc:10074
Change-Id: I245c7ecb23b67fc0ba65caaa5dbb4fcfd60c81bb
Reviewed-on: https://webrtc-review.googlesource.com/c/114505
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26092}
This will be used in a later CL to use the link capacity field in the
update to control the Opus encoder.
Bug: webrtc:9718
Change-Id: If2ad16a8f4656e8cdf10c33f5fb060ef7ca5caba
Reviewed-on: https://webrtc-review.googlesource.com/c/111510
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25761}
Replaced by interface ChannelSendInterface, implemented by ChannelSend
and mock class.
Thread checkers are moved to ChannelSend, which is also moved into
the anonymous namespace and exposed only via a function CreateChannelSend.
Bug: webrtc:9801
Change-Id: I73b2e2bfb67c1a5077709f2379533bf315babad9
Reviewed-on: https://webrtc-review.googlesource.com/c/111240
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25684}
So far ANA was not available for media transport interface. With recent changes to media transport, we can now account for packet overhead, network route (ip/tcp/udp/turn overheads) and we can also use bandwidth estimate from the media transport.
Bug: webrtc:9719
Change-Id: I98c9a09dd418b763c339ee2ee05592e164cf9199
Reviewed-on: https://webrtc-review.googlesource.com/c/110367
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25677}
'configured_bitrate_bps_' is accessed from different threads in
SetBitrate and GetBitrate (one comes back from OnNetworkRouteChange
callback, the other one is used in GetStats()) and so it should be
protected by a critical section.
Bug: webrtc:10010
Change-Id: I029baa729e0203b9f2d180d8835d61add26e6cef
Reviewed-on: https://webrtc-review.googlesource.com/c/111281
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25675}
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}
This is a follow up of https://webrtc-review.googlesource.com/c/src/+/43201.
Issue 43201 didn't do the job properly.
1. The audio rtcp report interval is not properly hooked up.
2. We don't need to propagate audio rtcp interval into video send stream or vice versa.
3. We don't need to propagate rtcp report interval to any receiving streams.
Bug: webrtc:8789
Change-Id: I1f637d6e5173608564ef0702d7eda6fc93b3200f
Reviewed-on: https://webrtc-review.googlesource.com/c/110105
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25610}
This change corrects a potential race condition when updating a FrameEncryptor
for the audio send channel. If a FrameEncryptor is set on an active audio
stream it is possible for the current FrameEncryptor attached to the audio channel to be deallocated due to
the FrameEncryptors reference count reaching zero before the new FrameEncryptor is set on the
channel.
To address this issue the ChannelSend is now holds a scoped_reftptr<FrameEncryptor>
to only allow deallocation when it is actually set on the encoder queue.
ChannelSend is unique in this respect as the Audio Receiver a long with the
Video Sender and Video Receiver streams all recreate themselves when they have
a configuration change. ChannelSend instead reconfigures itself using the
existing channel object.
Added Seth as TBR as this only introduces mocks.
TBR=shampson@webrtc.org
Bug: webrtc:9907
Change-Id: Ibf391dc9cecdbed1874e0252ff5c2cb92a5c64f4
Reviewed-on: https://webrtc-review.googlesource.com/c/107664
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25374}
This change adds a new subcategory to the public native webrtc::CryptoOptions
structure: webrtc::CryptoOptions::Frame.
This new structure has a single off by default property:
crypto_options.frame.require_frame_encryption.
This new flag if set prevents RtpSenders from sending outgoing payloads unless
a frame_encryptor_ is attached and prevents RtpReceivers from receiving
incoming payloads unless a frame_decryptor_ is attached.
This option is important to enforce no unencrypted data can ever leave the
device or be received.
I have also attached bindings for Java and Objective-C.
I have implemented this functionality for E2EE audio but not E2EE video
since the changes are still in review.
Bug: webrtc:9681
Change-Id: Ie184711190e0cdf5ac781f69e9489ceec904736f
Reviewed-on: https://webrtc-review.googlesource.com/c/105540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25238}
This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into
the audio media path. If a FrameEncryptorInterface is set on an outgoing audio RTPSender
then each outgoing audio payload will first pass through the provided FrameEncryptor which
will have a chance to modify the payload contents for the purposes of encryption.
If a FrameDecryptorInterface is set on an incoming audio RtpReceiver then each incoming
audio payload will first pass through the provided FrameDecryptor which have a chance to
modify the payload contents for the purpose of decryption.
While AEAD is supported by the FrameDecryptor/FrameEncryptor interfaces this CL does not
use it and so it is left as null.
Bug: webrtc:9681
Change-Id: Ic383a9dce280528739f9d271357c2220e0a0dccf
Reviewed-on: https://webrtc-review.googlesource.com/c/101702
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25001}
