This CL ensures that any previously set nondefault settings in the
audio processing module are not overwritten by the ApplyOptions
method in WebRtcVoiceEngine
BUG=webrtc:8018
Review-Url: https://codereview.webrtc.org/2985633002
Cr-Commit-Position: refs/heads/master@{#19144}
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 eliminates a thread hop in PeerConnectionFactory initialization,
and will allow some code to be simplified.
BUG=None
Review-Url: https://codereview.webrtc.org/2934103002
Cr-Commit-Position: refs/heads/master@{#18613}
Reason for revert:
Revert of revert of revert of revert of 'Activating..'. Or "reland of reland of 'Activate..'".
*Now* the internal projects are fixed and the fix is verified.
Original issue's description:
> Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2903153005/ )
>
> Reason for revert:
> Reverting again: internal project issues were apparently not completely fixed.
>
> Original issue's description:
> > Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2904893002/ )
> >
> > Reason for revert:
> > Revert the revert now that internal projects are updated.
> >
> > Original issue's description:
> > > Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ )
> > >
> > > Reason for revert:
> > > Breaks internal project.
> > >
> > > Original issue's description:
> > > > Activate 'offload debug dump recordings from audio thread to TaskQueue'.
> > > >
> > > > A low priority task queue is added to WebRTCVoiceEngine. The
> > > > start/stop debug calls make file logging happen on the task queue.
> > > >
> > > > In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> > > > so that it can be shared for low priority tasks between different
> > > > subcomponents. It will require some changes to MediaEngine,
> > > > CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> > > > projects.
> > > >
> > > > A task queue must be created and destroyed from the same thread. With
> > > > this CL that will be the worker thread, which creates and destroys
> > > > WebRTCVoiceEngine. With the dependent CL, it will probably change to
> > > > the signaling thread.
> > > >
> > > > NOTRY=True # tests just passed
> > > >
> > > > BUG=webrtc:7404
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2896813002
> > > > Cr-Commit-Position: refs/heads/master@{#18252}
> > > > Committed: c61bf947b4
> > >
> > > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:7404
> > >
> > > Review-Url: https://codereview.webrtc.org/2904893002
> > > Cr-Commit-Position: refs/heads/master@{#18255}
> > > Committed: be68b72cfa
> >
> > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:7404
> >
> > Review-Url: https://codereview.webrtc.org/2903153005
> > Cr-Commit-Position: refs/heads/master@{#18270}
> > Committed: d2303a2338
>
> TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2910633002
> Cr-Commit-Position: refs/heads/master@{#18272}
> Committed: fe9ecb07eaTBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2904423002
Cr-Commit-Position: refs/heads/master@{#18300}
Reason for revert:
Reverting again: internal project issues were apparently not completely fixed.
Original issue's description:
> Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2904893002/ )
>
> Reason for revert:
> Revert the revert now that internal projects are updated.
>
> Original issue's description:
> > Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ )
> >
> > Reason for revert:
> > Breaks internal project.
> >
> > Original issue's description:
> > > Activate 'offload debug dump recordings from audio thread to TaskQueue'.
> > >
> > > A low priority task queue is added to WebRTCVoiceEngine. The
> > > start/stop debug calls make file logging happen on the task queue.
> > >
> > > In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> > > so that it can be shared for low priority tasks between different
> > > subcomponents. It will require some changes to MediaEngine,
> > > CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> > > projects.
> > >
> > > A task queue must be created and destroyed from the same thread. With
> > > this CL that will be the worker thread, which creates and destroys
> > > WebRTCVoiceEngine. With the dependent CL, it will probably change to
> > > the signaling thread.
> > >
> > > NOTRY=True # tests just passed
> > >
> > > BUG=webrtc:7404
> > >
> > > Review-Url: https://codereview.webrtc.org/2896813002
> > > Cr-Commit-Position: refs/heads/master@{#18252}
> > > Committed: c61bf947b4
> >
> > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7404
> >
> > Review-Url: https://codereview.webrtc.org/2904893002
> > Cr-Commit-Position: refs/heads/master@{#18255}
> > Committed: be68b72cfa
>
> TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2903153005
> Cr-Commit-Position: refs/heads/master@{#18270}
> Committed: d2303a2338TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2910633002
Cr-Commit-Position: refs/heads/master@{#18272}
Reason for revert:
Revert the revert now that internal projects are updated.
Original issue's description:
> Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ )
>
> Reason for revert:
> Breaks internal project.
>
> Original issue's description:
> > Activate 'offload debug dump recordings from audio thread to TaskQueue'.
> >
> > A low priority task queue is added to WebRTCVoiceEngine. The
> > start/stop debug calls make file logging happen on the task queue.
> >
> > In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> > so that it can be shared for low priority tasks between different
> > subcomponents. It will require some changes to MediaEngine,
> > CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> > projects.
> >
> > A task queue must be created and destroyed from the same thread. With
> > this CL that will be the worker thread, which creates and destroys
> > WebRTCVoiceEngine. With the dependent CL, it will probably change to
> > the signaling thread.
> >
> > NOTRY=True # tests just passed
> >
> > BUG=webrtc:7404
> >
> > Review-Url: https://codereview.webrtc.org/2896813002
> > Cr-Commit-Position: refs/heads/master@{#18252}
> > Committed: c61bf947b4
>
> TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2904893002
> Cr-Commit-Position: refs/heads/master@{#18255}
> Committed: be68b72cfaTBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2903153005
Cr-Commit-Position: refs/heads/master@{#18270}
Reason for revert:
Breaks internal project.
Original issue's description:
> Activate 'offload debug dump recordings from audio thread to TaskQueue'.
>
> A low priority task queue is added to WebRTCVoiceEngine. The
> start/stop debug calls make file logging happen on the task queue.
>
> In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> so that it can be shared for low priority tasks between different
> subcomponents. It will require some changes to MediaEngine,
> CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> projects.
>
> A task queue must be created and destroyed from the same thread. With
> this CL that will be the worker thread, which creates and destroys
> WebRTCVoiceEngine. With the dependent CL, it will probably change to
> the signaling thread.
>
> NOTRY=True # tests just passed
>
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2896813002
> Cr-Commit-Position: refs/heads/master@{#18252}
> Committed: c61bf947b4TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2904893002
Cr-Commit-Position: refs/heads/master@{#18255}
A low priority task queue is added to WebRTCVoiceEngine. The
start/stop debug calls make file logging happen on the task queue.
In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
so that it can be shared for low priority tasks between different
subcomponents. It will require some changes to MediaEngine,
CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
projects.
A task queue must be created and destroyed from the same thread. With
this CL that will be the worker thread, which creates and destroys
WebRTCVoiceEngine. With the dependent CL, it will probably change to
the signaling thread.
NOTRY=True # tests just passed
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2896813002
Cr-Commit-Position: refs/heads/master@{#18252}
When operating on mobile devices, where hardware support is available
for the AEC and NS functionality, it is desirable to be able to
operate without hardcoded behaviors for the WebRTC AGC and HPF.
This CL adds support to allow a field trial to turn these off
whenever that is possible.
BUG=webrtc:6220, webrtc:6183, webrtc:6181
Review-Url: https://codereview.webrtc.org/2876133002
Cr-Commit-Position: refs/heads/master@{#18226}
Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2799033006
Cr-Commit-Position: refs/heads/master@{#17977}
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}
This will create another decoder instance, which isn't ideal, but
that's better than the current behavior where things don't work at all.
We still need to fix the root cause of the linked bug, which is that we
don't remember previous payload type mappings when generating an offer.
This CL also adds a check for what is a real error: when a payload type
that was mapped to one codec is changed to map to a different codec.
And lastly, this CL removes a DCHECK for an assumption that was not
valid: that subsequently applied codec lists will always be supersets of
previous lists.
BUG=webrtc:5847
Review-Url: https://codereview.webrtc.org/2831333002
Cr-Commit-Position: refs/heads/master@{#17897}
When SSRCs aren't signaled, an SSRC of 0 is used internally to mean
"the default receive stream". But this wasn't working with the
implementation of GetRtpReceiveParameters in the audio/video
engines. This was breaking RtpReceiver.GetParameters in this situation,
as well as the new getStats implementation (which relies on
GetParameters).
The new implementation will fail if *no* default receive stream is
configured (meaning no default sink is set), and otherwise will return
a default RtpEncodingParameters (later it will be filled with relevant
SDP parameters as they're implemented).
BUG=webrtc:6971
Review-Url: https://codereview.webrtc.org/2806173002
Cr-Commit-Position: refs/heads/master@{#17803}
Reason for revert:
Re-land, reverting did not fix bug.
https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
Original issue's description:
> Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
>
> Reason for revert:
> Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
>
> https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
>
> Original issue's description:
> > Added the GetSources() to the RtpReceiverInterface and implemented
> > it for the AudioRtpReceiver.
> >
> > This method returns a vector of RtpSource(both CSRC source and SSRC
> > source) which contains the ID of a source, the timestamp, the source
> > type (SSRC or CSRC) and the audio level.
> >
> > The RtpSource objects are buffered and maintained by the
> > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> > the info of the contributing source will be pulled along the object
> > chain:
> > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> > AudioReceiveStream -> voe::Channel -> RtpRtcp module
> >
> > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
> >
> > BUG=chromium:703122
> > TBR=stefan@webrtc.org, danilchap@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2770233003
> > Cr-Commit-Position: refs/heads/master@{#17591}
> > Committed: 292084c376
>
> TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:703122
>
> Review-Url: https://codereview.webrtc.org/2809613002
> Cr-Commit-Position: refs/heads/master@{#17616}
> Committed: fbcc5cb386TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122
Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
Reason for revert:
Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
Original issue's description:
> Added the GetSources() to the RtpReceiverInterface and implemented
> it for the AudioRtpReceiver.
>
> This method returns a vector of RtpSource(both CSRC source and SSRC
> source) which contains the ID of a source, the timestamp, the source
> type (SSRC or CSRC) and the audio level.
>
> The RtpSource objects are buffered and maintained by the
> RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> the info of the contributing source will be pulled along the object
> chain:
> AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> AudioReceiveStream -> voe::Channel -> RtpRtcp module
>
> Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
>
> BUG=chromium:703122
> TBR=stefan@webrtc.org, danilchap@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2770233003
> Cr-Commit-Position: refs/heads/master@{#17591}
> Committed: 292084c376TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:703122
Review-Url: https://codereview.webrtc.org/2809613002
Cr-Commit-Position: refs/heads/master@{#17616}
This CL contains all the changes made to audio_coding while making
audio encoders injectable. Apart from some small changes to
webrtcvoiceengine, nothing here is hooked up to the outside
world. Those changes will be added to a follow-up CL.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2695243005
Cr-Commit-Position: refs/heads/master@{#17569}
This reverts to previous behavior where b=AS only affects the codec bitrate for audio streams, and not the max bandwidth estimate.
BUG=chromium:703903
Review-Url: https://codereview.webrtc.org/2774123002
Cr-Commit-Position: refs/heads/master@{#17386}
Reason for revert:
Makes perf and Chromium FYI bots unhappy.
Original issue's description:
> WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
>
> This removes one more place where we were unable to handle codecs not
> in the built-in set.
>
> BUG=webrtc:5805
>
> Review-Url: https://codereview.webrtc.org/2686043006
> Cr-Commit-Position: refs/heads/master@{#17370}
> Committed: 1724cfbdbaTBR=ossu@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:5805
Review-Url: https://codereview.webrtc.org/2772043002
Cr-Commit-Position: refs/heads/master@{#17374}
This removes one more place where we were unable to handle codecs not
in the built-in set.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2686043006
Cr-Commit-Position: refs/heads/master@{#17370}
Reason to go back is that we may end up with a bunch of streams that are never cleaned up and consume resources.
BUG=webrtc:7175, b/35863246
Review-Url: https://codereview.webrtc.org/2746763002
Cr-Commit-Position: refs/heads/master@{#17210}
It seems to me that we're currently just picking the first CN codec, rather than the one that matches the clock rate of the voice codec. The only test I've gotten to fail by changing this behavior is the one that's also changed in this CL, which explicitly expects a CN codec to be chosen even though there's none matching.
BUG=webrtc:7282
Review-Url: https://codereview.webrtc.org/2707133007
Cr-Commit-Position: refs/heads/master@{#16979}
These methods relate to typing detection but are not used anymore. Typing detection is enabled through the VoiceDetection module on the APM.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2684933008
Cr-Commit-Position: refs/heads/master@{#16527}
It's currently only used to ensure transport-cc is enabled for the format in question. It might be used to toggle more things in future.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2669583002
Cr-Commit-Position: refs/heads/master@{#16514}
These structs will be used for ORTC objects (and their WebRTC
equivalents).
This CL also introduces some minor changes to the existing implemented
structs:
- max_bitrate_bps uses rtc::Optional instead of "-1 means unset"
- "mime_type" turned into "name"/"kind" (which can be used to form the
MIME type string, if needed).
- clock_rate and channels changed to rtc::Optional, since they will
need to be for RtpSender.send().
- Renamed "channels" to "num_channels" (the ORTC name, which I prefer).
BUG=webrtc:7013, webrtc:7112
Review-Url: https://codereview.webrtc.org/2651883010
Cr-Commit-Position: refs/heads/master@{#16437}
BUG=webrtc:7097
TEST=Set "ptime=120", try WebRTC calls over custom build Chromium with and without Opus 120ms. Try both Chromium w <-> Chromium w and Chromium w <-> Chromium w/o
Review-Url: https://codereview.webrtc.org/2668633004
Cr-Commit-Position: refs/heads/master@{#16408}
Bulk of the changes were done using
git grep -l '#include "webrtc/base/common.h"' | \
xargs sed -i '\,^#include.*webrtc/base/common\.h,d'
followed by adding back the include in the few places where it is
still needed, and in one case (pseudotcp.cc) instead deleting its use
of RTC_UNUSED.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2644103002
Cr-Commit-Position: refs/heads/master@{#16263}
webrtcvoiceengine.cc ensured that if the bitrate set for ISAC was 0,
it was changed to -1 so that the codec could manage the bitrate
itself.
webrtcsdp.cc ensured that if the bitrate set for ISAC was 0, it was
explicitly set to default values to avoid the codec's built in bitrate
management.
Eventually, there'll be no codec specific code like this in these
layers. This is one step towards that goal.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2642923003
Cr-Commit-Position: refs/heads/master@{#16220}
