WEBRTC_VOICE_ENGINE_AGC, WEBRTC_VOICE_ENGINE_ECHO, and
WEBRTC_VOICE_ENGINE_NR are now gone.
BUG=webrtc:6506
Review-Url: https://codereview.webrtc.org/2530373002
Cr-Commit-Position: refs/heads/master@{#15309}
This change rewrites RMSLevel, making it accept an ArrayView as input,
and modify the implementation somewhat. It also makes the class keep
track of the peak RMS in addition to the average RMS over the
measurement period.
New tests are added to cover the new functionality.
BUG=webrtc:6622
Review-Url: https://codereview.webrtc.org/2535523002
Cr-Commit-Position: refs/heads/master@{#15294}
There's no longer any need to make the two arguments have the same
signedness, so we can drop the "u" suffix on literal integer
arguments.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2535593002
Cr-Commit-Position: refs/heads/master@{#15280}
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}
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.
Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.
transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.
NOTRY=True
BUG=webrtc:5589, webrtc:5878, webrtc:6785
Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
Reason for revert:
Downstream code has been updated.
Original issue's description:
> Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ )
>
> Reason for revert:
> Breaks downstream projects.
>
> Original issue's description:
> > Remove RTPPayloadStrategy and simplify RTPPayloadRegistry
> >
> > This CL removes RTPPayloadStrategy that is currently used to handle
> > audio/video specific aspects of payload handling. Instead, the audio and
> > video specific aspects will now have different functions, with linear
> > code flow.
> >
> > This CL does not contain any functional changes, and is just a
> > preparation for future CL:s.
> >
> > The main purpose with this CL is to add this function:
> > bool PayloadIsCompatible(const RtpUtility::Payload& payload,
> > const webrtc::VideoCodec& video_codec);
> > that can easily be extended in a future CL to look at video codec
> > specific information.
> >
> > BUG=webrtc:6743
> >
> > Committed: https://crrev.com/b881254dc86d2cc80a52e08155433458be002166
> > Cr-Commit-Position: refs/heads/master@{#15232}
>
> TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6743
>
> Committed: https://crrev.com/33c81d05613f45f65ee17224ed381c6cdd1c6c6f
> Cr-Commit-Position: refs/heads/master@{#15234}
TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2531043002
Cr-Commit-Position: refs/heads/master@{#15245}
Reason for revert:
Breaks downstream projects.
Original issue's description:
> Remove RTPPayloadStrategy and simplify RTPPayloadRegistry
>
> This CL removes RTPPayloadStrategy that is currently used to handle
> audio/video specific aspects of payload handling. Instead, the audio and
> video specific aspects will now have different functions, with linear
> code flow.
>
> This CL does not contain any functional changes, and is just a
> preparation for future CL:s.
>
> The main purpose with this CL is to add this function:
> bool PayloadIsCompatible(const RtpUtility::Payload& payload,
> const webrtc::VideoCodec& video_codec);
> that can easily be extended in a future CL to look at video codec
> specific information.
>
> BUG=webrtc:6743
>
> Committed: https://crrev.com/b881254dc86d2cc80a52e08155433458be002166
> Cr-Commit-Position: refs/heads/master@{#15232}
TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2528993002
Cr-Commit-Position: refs/heads/master@{#15234}
This CL removes RTPPayloadStrategy that is currently used to handle
audio/video specific aspects of payload handling. Instead, the audio and
video specific aspects will now have different functions, with linear
code flow.
This CL does not contain any functional changes, and is just a
preparation for future CL:s.
The main purpose with this CL is to add this function:
bool PayloadIsCompatible(const RtpUtility::Payload& payload,
const webrtc::VideoCodec& video_codec);
that can easily be extended in a future CL to look at video codec
specific information.
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2524923002
Cr-Commit-Position: refs/heads/master@{#15232}
The purpose with this CL is to be able to send video codec specific
information down to RTPPayloadRegistry. We already do this for audio
with explicit arguments for e.g. number of channels. Instead of
extracting the arguments from webrtc::CodecInst (audio) and
webrtc::VideoCodec, this CL sends the types unmodified all the way down
to RTPPayloadRegistry.
This CL does not contain any functional changes, and is just a
preparation for future CL:s.
In the dependent CL https://codereview.webrtc.org/2524923002/,
RTPPayloadStrategy is removed. RTPPayloadStrategy previously handled
audio/video specific aspects of payload handling. After this CL, we will
know if we get audio or video codecs without any dependency injection,
since we have different functions with different signatures for audio
vs video.
BUG=webrtc:6743
TBR=mflodman
Review-Url: https://codereview.webrtc.org/2523843002
Cr-Commit-Position: refs/heads/master@{#15231}
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}
- 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}
This change will allow for a audio source to report its sampling rate
to the audio mixer. It is needed in order to mix at a lower sampling
rate. Mixing at a lower sampling rate can in many cases lead to big
efficiency improvements, as reported by experiments.
The code affected is all implementations of the Source interface:
AudioReceiveStream and a mock class. The AudioReceiveStream now
queries its underlying voe::Channel object for the needed frequency.
Note that the changes to the mixing algorithm are done in a later CL.
BUG=webrtc:6346
NOTRY=True
TBR=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2448113009
Cr-Commit-Position: refs/heads/master@{#14839}
Reason for revert:
This CL breaks internal dependencies.
Original issue's description:
> New statistics interface for APM
>
> This adds functions to enable and retrieve statistics from APM. These functions are not yet called, which will happen in a follow-up CL.
>
> BUG=webrtc:6525
>
> Committed: https://crrev.com/8b8d3e4c30e8ea3846b58dfd36d1fd35a7799df4
> Cr-Commit-Position: refs/heads/master@{#14810}
TBR=peah@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:6525
Review-Url: https://codereview.webrtc.org/2456333002
Cr-Commit-Position: refs/heads/master@{#14814}
This adds functions to enable and retrieve statistics from APM. These functions are not yet called, which will happen in a follow-up CL.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2433153003
Cr-Commit-Position: refs/heads/master@{#14810}
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}
I added a few missing dependencies to the GN target of voice_engine while doing other
unrelated work. Currently GN's header include checker has the
following to say:
$ gn check out/gn_debug webrtc/voice_engine
ERROR at //webrtc/voice_engine/include/voe_network.h:38:11: Include not allowed.
#include "webrtc/transport.h"
^-----------------
It is not in any dependency of
//webrtc/voice_engine:voice_engine
The include file is in the target(s):
//webrtc:webrtc
which should somehow be reachable.
transport.h should probably move in to webrtc/api, since it is already
a pure virtual interface and is used in quite a few places.
BUG=webrtc:5589
NOTRY=True
Review-Url: https://codereview.webrtc.org/2421483002
Cr-Commit-Position: refs/heads/master@{#14633}
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}
Make sure that WEBRTC_VOICE_ENGINE_AGC, WEBRTC_VOICE_ENGINE_ECHO, and
WEBRTC_VOICE_ENGINE_NR are always defined.
BUG=webrtc:6506
Review-Url: https://codereview.webrtc.org/2401393002
Cr-Commit-Position: refs/heads/master@{#14587}
This CL deletes the old and not used video defines in
engine_configurations.h and pre-pends voice_ to indicate there are only
voice/audio defines left in the file.
BUG=none
R=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/2401673002 .
Cr-Commit-Position: refs/heads/master@{#14558}
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 is done to ensure GN targets are placed in the same directory as of the source files.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2365383004
Cr-Commit-Position: refs/heads/master@{#14411}
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).
After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()
See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.
NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.
BUG=webrtc:6410, chromium:630755
Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
Remove a large number of targets that are no longer built, to reduce maintenance.
Only targets that have a GN version were removed.
BUG=webrtc:6323
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2340773003
Cr-Commit-Position: refs/heads/master@{#14231}
Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.
Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.
BUG=
NOPRESUBMIT=true
Committed: https://crrev.com/ade2a038a9290ee0c85d8c682eba5447aca943cd
Review-Url: https://codereview.webrtc.org/2319583005
Cr-Original-Commit-Position: refs/heads/master@{#14191}
Cr-Commit-Position: refs/heads/master@{#14198}
