through streams related to a call object.
The Call object does not know directly when packets pass through it, only which
AudioSendStreams are used. Each AudioSendStream has a pointer to the Transport
object through which its packets are send.
This CL:
By registering an internal wrapper class, TimedTransport, the AudioSendStream
can stay up-to-date on when packets have passed through its Transport. This
lifetime (as an interval) is then queried by the Call when the AudioSendStream
is destroyed. When Call is destroyed, all streams are guaranteed to have been
destroyed and hence Call is up-to-date on packet activity.
The class TimeInterval keeps the code in Call and AudioSendStream smaller, with
fewer get methods in their APIs and less code for updating values.
Also modifies the unit test for AudioSendStream: it previously enforced that
the stream registers (with its channel proxy) the same transport that it was
constructed with.
BUG=webrtc:7882
Review-Url: https://codereview.webrtc.org/2979833002
Cr-Commit-Position: refs/heads/master@{#19087}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
This functionality already exists for video streams, so not having it
for audio is unexpected and has lead to problems.
BUG=webrtc:7631
Review-Url: https://codereview.webrtc.org/2887733002
Cr-Commit-Position: refs/heads/master@{#18231}
probing_interval as a name is used for the period that BWE attempt to increase its estimate. The name is confusing since it is not related to "probing" which is a special mechanism for estimating BWE.
BUG=None
Review-Url: https://codereview.webrtc.org/2888893002
Cr-Commit-Position: refs/heads/master@{#18203}
These are the changes made to WebRtcVoiceEngine and surrounding
code. It still contains some things that are inelegant, like how
AudioCodecSpec and AudioFormatInfo is ferried around in
SendCodecSpec. This should probably be resolved before landing.
There are also a few test still that are disabled. They should be
removed or fixed, as the case may be.
I've put this CL up to get a better overview of the changes made and
how reviewable they are.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2705093002
Cr-Commit-Position: refs/heads/master@{#17904}
Also move RtpTransportControllerSendInterface to its own header file.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2808043002
Cr-Commit-Position: refs/heads/master@{#17840}
These are invalid since:
- An allocated bitrate of 0 means that the stream should be disabled. Changing the behavior to send audio at min bitrate even though the allocator asks for the stream to be disabled.
- It should be OK to set a min bitrate equal to the start bitrate.
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2806163003
Cr-Commit-Position: refs/heads/master@{#17613}
This is in preparation for merging the ViERemb logic in packet_router,
to send REMB feedback as sender reports if possible, otherwise as
receiver reports.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2774623006
Cr-Commit-Position: refs/heads/master@{#17489}
Implementation owned by call, and passed to VideoSendStream and
AudioSendStream.
BUG=webrtc:6847, webrtc:7135
Review-Url: https://codereview.webrtc.org/2685673003
Cr-Commit-Position: refs/heads/master@{#17389}
This is part of a series of CLs. Next CLs:
1. CL for RPLR-based FecController
2. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2661043003
Cr-Commit-Position: refs/heads/master@{#17368}
This CL is one in a series. To finish the work, the following CLs will be added:
1. CL for connecting RPLR as well
2. CL for RPLR-based FecController
3. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2638083002
Cr-Commit-Position: refs/heads/master@{#17365}
New class ReceiveSideCongestionController, extracted from CongestionController, and responsible for the
OnReceivedPacket processing.
Rest of the CongestionController moved to a new class
SendSideCongestionController.
To avoid breaking applications, CongestionController is redefined
as a union of these two classes, with no intended change in behavior.
With one exception: CongestionController::SetBweBitrates used to call
remote_bitrate_estimator_.SetMinBitrate, but after remote_bitrate_estimator_ was moved to ReceiveSideCongestionController,
it no longer does this.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2752233002
Cr-Commit-Position: refs/heads/master@{#17321}
And delete the method CongestionController::packet_router.
BUG=None
Review-Url: https://codereview.webrtc.org/2516983004
Cr-Commit-Position: refs/heads/master@{#15323}
Reason for revert:
Unfortunately, this change breaks internal projects. Specifically the change to the CongestionController interface means anything implementing it will be forced to change in lock-step.
Original issue's description:
> Pass time constanct to bwe smoothing filter.
>
> BUG=webrtc:6443, webrtc:6303
>
> Committed: https://crrev.com/9abbf5ae4ec7d688a9b4aa03a405f3faadb74b90
> Cr-Commit-Position: refs/heads/master@{#15266}
TBR=minyue@webrtc.org,stefan@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6443, webrtc:6303
Review-Url: https://codereview.webrtc.org/2532993002
Cr-Commit-Position: refs/heads/master@{#15272}
Payload type -> RtpCodecParameters maps added for sender and receiver.
This is a follow-up to https://codereview.webrtc.org/2484193002/ which
did the same thing for VideoMediaInfo. This information will be used to
produce RTCCodecStats[1].
Voice[Sender/Receiver]Info is updated with current codec payload type
for every stream which can be used to look up the codec in
VoiceMediaInfo.
[1] https://w3c.github.io/webrtc-stats/#codec-dict*
BUG=chromium:659117
Review-Url: https://codereview.webrtc.org/2503383002
Cr-Commit-Position: refs/heads/master@{#15144}
We support multiple payload types, and one which matches the audio codec the closest, is picked (or the one with lowest clock rate, if no perfect match is found).
The exact clock rate is then ignored and DTMF packets are time stamped with the rate of the current audio codec. This is exactly the way the code has worked up to this point, but until now we have been under the impression that we were in fact sending 8k DTMF.
In other words, this is an improvement over the current situation, since we will most likely find a payload type which matches the codec clock rate.
This CL also does a little cleaning of the DTMFQueue and RTPSenderAudio classes.
BUG=webrtc:2795
Review-Url: https://codereview.webrtc.org/2392883002
Cr-Commit-Position: refs/heads/master@{#15129}
Updates GetStats() function in AudioSendStream to use the new GetStatistics function in APM instead of the corresponding VoE functions.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2463813002
Cr-Commit-Position: refs/heads/master@{#15065}
The unit was kbps but the one default use of it is in bps. The inconsistency should be fixed.
BUG=webrtc:6670
Review-Url: https://codereview.webrtc.org/2247213005
Cr-Commit-Position: refs/heads/master@{#14955}
As a side effect:
- Moved the AudioSendStream::Config::SendCodecSpec methods into the .cc.
- Which exposed an issue with event_visualizer_utils not having a dependency on api:call_api set up.
- Which further exposed clang warnings about large inlined default methods in webrtc/config.h.
BUG=webrtc:4690
Committed: https://crrev.com/1836fd6257a692959b3b49ba99ef587ad9995871
Review-Url: https://codereview.webrtc.org/2446963003
Cr-Original-Commit-Position: refs/heads/master@{#14771}
Cr-Commit-Position: refs/heads/master@{#14780}
Reason for revert:
Breaks downstream project
Original issue's description:
> Clean up logging in AudioSendStream::SetupSendCodec().
>
> As a side effect:
> - Moved the AudioSendStream::Config::SendCodecSpec methods into the .cc.
> - Which exposed an issue with event_visualizer_utils not having a dependency on api:call_api set up.
> - Which further exposed clang warnings about large inlined default methods in webrtc/config.h.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/1836fd6257a692959b3b49ba99ef587ad9995871
> Cr-Commit-Position: refs/heads/master@{#14771}
TBR=kwiberg@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2452643002
Cr-Commit-Position: refs/heads/master@{#14774}
