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}
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
Review-Url: https://codereview.webrtc.org/2446963003
Cr-Commit-Position: refs/heads/master@{#14771}
The stat is currently always set to zero until the residual echo detector has landed.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2431443003
Cr-Commit-Position: refs/heads/master@{#14721}
This class is logically parallel with the {Audio,Video}ReceiveStream
classes. Its purpose is to describe a receive stream of FlexFEC packets,
through the corresponding config.
Functionally, this class simply forwards the received RTP packets
to its FlexfecReceiver, which returns recovered packets to the
Call level, for appropriate demultiplexing based on SSRC.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2397843005
Cr-Commit-Position: refs/heads/master@{#14704}
This change adds a new statistic for logging how many calls to
NetEq::GetAudio resulted in a "muted output". A muted output happens
if the packet stream has been dead for some time (and the last decoded
packet was not comfort noise).
BUG=webrtc:5606
BUG=b/31256483
Review-Url: https://codereview.webrtc.org/2341293002
Cr-Commit-Position: refs/heads/master@{#14302}
