- Functionality now implemented in AudioReceiveStream and Call.
- Added some missing function to MockChannelProxy.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2461523002
Cr-Commit-Position: refs/heads/master@{#15072}
- The legacy API is not used in WVoE/MC.
- Removed use of the API (along with StartReceive()) from unit tests.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2453243003
Cr-Commit-Position: refs/heads/master@{#14858}
Simplify the AudioMixer::Source interface and update the mixer
implementation to the new interface.
Instead of asking a mixer source to provide a pointer to an AudioFrame
during each mixing iteration, a mixer should supply a pointer to its
own AudioFrame.
This simplifies lifetime issues as sources do not give away an
internal pointer.
Implementation: when an audio source is added, the mixer allocates a
new AudioFrame. The audio frame is kept together in the internal class
SourceStatus together with the audio source pointer until the source
is removed.
NOTRY=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2420913002
Cr-Commit-Position: refs/heads/master@{#14713}
Methods to facilitate this are added to ChannelProxy and voe::Channel.
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2378143004
Cr-Commit-Position: refs/heads/master@{#14707}
Reason for revert:
Speculative revert.
Intermittent memory access errors suspected to be caused by this cl.
See for instance https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/8018
UNADDRESSABLE ACCESS of freed memory: reading 0x0331d330-0x0331d334 4 byte(s)
# 0 webrtc::voe::RtcpRttStatsProxy::LastProcessedRtt
# 1 webrtc::ModuleRtpRtcpImpl::Process
Original issue's description:
> Add RtcpRttStats to AudioStream
>
> BUG=webrtc:6508
>
> Committed: https://crrev.com/e0729c56d35acfaf9738fdb32c6508cd78eaf089
> Cr-Commit-Position: refs/heads/master@{#14595}
TBR=stefan@webrtc.org,minyue@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6508
Review-Url: https://codereview.webrtc.org/2415943002
Cr-Commit-Position: refs/heads/master@{#14631}
This gets rid of a bit of codec-specific code in VoE.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2355483003
Cr-Commit-Position: refs/heads/master@{#14614}
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.
After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).
The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).
This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.
BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2380683005 .
Cr-Commit-Position: refs/heads/master@{#14485}
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.
- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.
BUG=webrtc:5879
Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
The method is no longer used, since the jitter buffer delay is
obtained directly from AudioCodingModule instead of being calculated
and smoothed in VoiceEngine. Deleting a few obsolete member variables
as well.
BUG=webrtc:6237
Review-Url: https://codereview.webrtc.org/2290253002
Cr-Commit-Position: refs/heads/master@{#14007}
The new method returns the current total delay (packet buffer and sync
buffer) in ms, with smoothing applied to even out short-time
fluctuations due to jitter. The packet buffer part of the delay is not
updated during DTX/CNG periods.
This CL also pipes the new metric through ACM and uses it in
VoiceEngine. It replaces the previous method of estimating the buffer
delay (where an inserted packet's RTP timestamp was compared with the
last played timestamp from NetEq). The new method works better under
periods of DTX/CNG.
Review-Url: https://codereview.webrtc.org/2262203002
Cr-Commit-Position: refs/heads/master@{#13855}
Mostly, it's about replacing mutable reference arguments with pointer
arguments, and replacing C style casts with C++ style casts.
Review-Url: https://codereview.webrtc.org/2056653002
Cr-Commit-Position: refs/heads/master@{#13849}
Because passing ownership in raw pointers makes kittens cry.
This also means we can ditch the Destroy functions and the protected
destructors. (Well, almost. We need to keep the old CreateFilePlayer
and DestroyFilePlayer around for a little while longer because of an
external caller.)
Review-Url: https://codereview.webrtc.org/2049683003
Cr-Commit-Position: refs/heads/master@{#13797}
Reason for revert:
broke browser_tests
Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Original-Commit-Position: refs/heads/master@{#13613}
> Cr-Commit-Position: refs/heads/master@{#13615}
TBR=pbos@webrtc.org,mflodman@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2203233002
Cr-Commit-Position: refs/heads/master@{#13616}
Reason for revert:
broke internal tests
Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Commit-Position: refs/heads/master@{#13613}
TBR=pbos@webrtc.org,mflodman@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2206743002
Cr-Commit-Position: refs/heads/master@{#13614}
This message was printed to the receiver's log on every single video
frame if the remote side is sending video but not audio. This could
happen if, for example, one participant creates RtpSenders for both
audio and video but doesn't set the audio track, or if the track is
set but RtpParameters.encodings[0].active == false.
This CL changes the trace level to StateInfo as it is the expected
behavior in this case.
R=deadbeef@webrtc.org, henrika@webrtc.org
Review URL: https://codereview.webrtc.org/2125183002 .
Cr-Commit-Position: refs/heads/master@{#13408}
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.
Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}
TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749
Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
BUG=webrtc:4741,webrtc:5603,chromium:609749
R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1748403002 .
Cr-Commit-Position: refs/heads/master@{#13321}
Allows detecting large-enough audio packets as part of a probe,
speculative fix for a rampup-time regression in M50. These packets are
accounted on the send side when probing.
BUG=webrtc:5985
R=mflodman@webrtc.org, philipel@webrtc.org
Review URL: https://codereview.webrtc.org/2061193002 .
Cr-Commit-Position: refs/heads/master@{#13210}
It was already disabled for browsers by design, and for everyone else
because of a bug.
BUG=webrtc:5922
Review-Url: https://codereview.webrtc.org/2055493003
Cr-Commit-Position: refs/heads/master@{#13138}
CreatePeerConnectionFactory does not yet expose the ability to set the
factory from the outside.
Added notry due to android_dbg being broken.
NOTRY=True
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/1991233004
Cr-Commit-Position: refs/heads/master@{#13112}
This gets rid of the complex & icky state where the sample rate is not
yet determined.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2020353003
Cr-Commit-Position: refs/heads/master@{#13011}
VoEBase is plumbed to optionally take an AudioDecoderFactory, or create
a builtin factory if none is provided.
Retained the CreateChannel interfaces in Channel and ChannelManager
and added variants for injecting an AudioDecoderFactory. The
"old-style" variants call CreateBuiltinAudioDecoderFactory to get a
factory to use.
(Just realized this means each channel uses a separate factory with the
old-style calls. Probably ok.)
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/1993783002
Cr-Commit-Position: refs/heads/master@{#12961}
Adding a some checks and switching out a few assert for RTC_[D]CHECK.
These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled. I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.
TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/2014973002 .
Cr-Commit-Position: refs/heads/master@{#12925}
