This CL re-routes audio through AudioMixer instead of AudioConferenceMixer.
This is done without any modifications to VoiceEngine.
Previously, output audio was polled by an AudioDevice through an AudioTransport
pointer, which was an instance of VoEBaseImpl. VoiceEngineImpl sent the
request for data on to OutputMixer and further to AudioConferenceMixer.
This CL changes the audio flow to an AudioDevice. We reconfigure the AudioDevice
to have another AudioTransport pointer, which points to an AudioTransportProxy.
The AudioTransportProxy is responsible for feeding mixed data to the
AudioProcessing component for echo cancellation, and to resample the audio data
after AudioProcessing and before it is sent to the AudioDevice.
The set up of the audio path was previously done during VoiceEngine
initialization. Now it is changed in the AudioState constructor.
This list shows where audio-path-related VoiceEngine functionality has been
moved:
OutputMixer --> AudioTransportProxy
VoiceEngineImpl --> AudioState, AudioTransportProxy
SharedData --> AudioState
Channel --> AudioReceiveStream, ChannelProxy, Channel
AudioState owns the new mixer and connects it to AudioTransport and
AudioDevice on initialization.
The audio input source is AudioReceiveStream, which registers itself with the
mixer (which it gets from AudioState) on Start and Stop.
# Since the AudioTransport interface contains non-const references.
NOPRESUBMIT=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2436033002
Cr-Commit-Position: refs/heads/master@{#15193}
The mock is used in a dependent CL https://codereview.webrtc.org/2436033002.
There is also a goal to allow external mixing implementations
(subclasses of webrtc::AudioMixer) and inject them to
PeerConnectionFactory. We think that part of that is an official and
maintained mock.
Summary of changes:
* Created a mixer mock/stub in webrtc/api/test
* Made a target webrtc/api:mock_audio_mixer for it.
NOTRY=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2520323002
Cr-Commit-Position: refs/heads/master@{#15190}
Previously this was only collected for RTCOutboundRTPStreamStats video,
with no comment saying it was missing for Inbound. (nack_count should be
collected vor audio as well but this is currently not available - there
is already an existing comment about this in rtcstats_objects.h.)
BUG=chromium:657855, chromium:657854, chromium:627816
Review-Url: https://codereview.webrtc.org/2515293002
Cr-Commit-Position: refs/heads/master@{#15185}
This CL interfaces the SDP information (payload types and
SSRCs) about FlexFEC with the corresponding configs at the
Call layer. It also adds a field trial, which when active
will expose FlexFEC in the default codec list, thus showing
up in the default SDP.
BUG=webrtc:5654
R=magjed@webrtc.org, stefan@webrtc.orgCC=perkj@webrtc.org
Review-Url: https://codereview.webrtc.org/2511703002
Cr-Commit-Position: refs/heads/master@{#15184}
Both of these are unused legacy stuff.
BUG=webrtc:6323
NOTRY=True
Review-Url: https://codereview.webrtc.org/2520253002
Cr-Commit-Position: refs/heads/master@{#15183}
This change adds code that lets Opus increase the complexity setting
at low bitrates (only relevant for mobile where the default complexity
is not already maximum). The feature is default off.
Also adding a performance test to make sure the complexity adaptation
has desired effect.
BUG=webrtc:6708
Review-Url: https://codereview.webrtc.org/2503443002
Cr-Commit-Position: refs/heads/master@{#15182}
This improves the performance of SurfaceViewRenderer. This feature is added
behind a flag for now because it can be buggy on some devices.
BUG=webrtc:6470
Review-Url: https://codereview.webrtc.org/2510413002
Cr-Commit-Position: refs/heads/master@{#15181}
The class VideoDecoderSoftwareFallbackWrapper is an implementation
detail of webrtc/media/engine/webrtcvideoengine2.cc and should not be
directly under webrtc/video_decoder.h. The main purpose is to improve
the dependency graph in WebRTC so that VideoDecoderSoftwareFallbackWrapper
can depend on cricket::VideoCodec.
The test for VideoDecoderSoftwareFallbackWrapper is also moved from
webrtc/video/video_decoder_unittest.cc to
webrtc/media/engine/videodecodersoftwarefallbackwrapper_unittest.cc.
BUG=webrtc:6743
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2518263003
Cr-Commit-Position: refs/heads/master@{#15180}
This file fits there more naturally since it has dependencies to jni.
BUG=None
Review-Url: https://codereview.webrtc.org/2514383002
Cr-Commit-Position: refs/heads/master@{#15179}
This allows downstream dependencies can add it as a dependency.
BUG=webrtc:6499
Review-Url: https://codereview.webrtc.org/2521183002
Cr-Commit-Position: refs/heads/master@{#15178}
a == b would return true if a.member is defined even if b.member is
undefined if their values were equal. We would say that b does not have
a value in that case but its value_ member would still be initialized to
something that is being compared to. Bugfix makes sure not to do value
comparison in this case if b is undefined.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2517163002
Cr-Commit-Position: refs/heads/master@{#15172}
Now gtest-parallel is the same as in the github repo.
A wrapper script was created to handle the environment variables.
TBR=pbos@webrtc.org
BUG=chromium:497757
NOTRY=True
Review-Url: https://codereview.webrtc.org/2513073002
Cr-Commit-Position: refs/heads/master@{#15170}
Make magjed@ owner of whole webrtc/api/android/ and remove him as owner of subfolders.
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2516973004
Cr-Commit-Position: refs/heads/master@{#15168}
the previously specified setting is changed if it is specified to be changed,
and otherwise the previously specified setting is kept as it is.
This CL replicates this functionality for the way that the new APM
parameter scheme is used.
BUG=webrtc::6220, webrtc::6296, webrtc::6297, webrtc::6181, webrtc::5298
Review-Url: https://codereview.webrtc.org/2489343002
Cr-Commit-Position: refs/heads/master@{#15167}
A recent cl (https://codereview.webrtc.org/2510583002) introduced an
issue where the initial rate allocation (call to VideoBitrateAllocator
and any associated temporal layers) uses framerate = 0 fps. This may
cause issues, including having the rate control in ScreenshareLayers
ramp up too slowly.
This CL make the initial call use VideoCodec.maxFramerate as framerate.
Also expanded unit tests.
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2513383002
Cr-Commit-Position: refs/heads/master@{#15166}
They just decay to pointers anyway, so it's more honest to declare
them as pointers.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2515163002
Cr-Commit-Position: refs/heads/master@{#15165}
Previously, layout matrix was not correctly taken into account when calculating
the drawn size of the frame.
BUG=webrtc:6470
Review-Url: https://codereview.webrtc.org/2514793002
Cr-Commit-Position: refs/heads/master@{#15163}
After https://chromium-review.googlesource.com/c/412190/
we can remove all the GYP variables. The GN args are also not used
during runhooks (only GYP_DEFINES had any effect), so they're also
removed to avoid confusion. Only use_goma=True was left since
the ios recipe module uses it to decide if Goma shall be started
on the bots.
Delete unused GN/GYP-specific JSON files for bots that are now removed.
Finally, bump iOS version 10.0 and change simulator phones to
iPhone 6s to match what Chromium uses (may help solving bugs.webrtc.org/4752)
BUG=webrtc:4752, webrtc:6323
NOTRY=True
Review-Url: https://codereview.webrtc.org/2507063008
Cr-Commit-Position: refs/heads/master@{#15160}
ScreenCapturerIntegrationTest is flaky on Windows systems due to some unknown
reason. But it's do easily impacted by the environment, so this change adds more
logging (entire screenshot) to help debugging.
Meanwhile, this change also includes a nice-to-have change in ScreenDrawerWin to
always bring the window to front in each WaitForPendingDraws() function call. I
cannot quite tell whether this change can help to resolve the issue, but it is
worth trying.
BUG=webrtc:6666
Review-Url: https://codereview.webrtc.org/2492723002
Cr-Commit-Position: refs/heads/master@{#15158}
Reason for revert:
This class is used by downstream applications.
Original issue's description:
> Remove unused HttpClient class.
>
> BUG=none
>
> Committed: https://crrev.com/4a698f611a4bed4e074a26556ca9262b182955ee
> Cr-Commit-Position: refs/heads/master@{#15156}
TBR=tommi@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=none
Review-Url: https://codereview.webrtc.org/2518723002
Cr-Commit-Position: refs/heads/master@{#15157}
This was blocking swarming for memcheck.
BUG=chromium:497757, webrtc:6727
Review-Url: https://codereview.webrtc.org/2511393002
Cr-Commit-Position: refs/heads/master@{#15153}
Move the resources to //resources and upload them to Google Storage.
BUG=webrtc:6727
Review-Url: https://codereview.webrtc.org/2508943004
Cr-Commit-Position: refs/heads/master@{#15152}
In the AudioMixerImpl implementation, removing a source never fails
and the return value is always true (see audio_mixer/audio_mixer_impl.cc).
A return value of |false| signaled that removing a source failed for
some reason. We have come to the conclusion that
* we don't know how to handle a return value of |false|
* we can't think of why an alternative implementation would need to
signal failure when removing a stream.
To avoid having a status code that is never read, never acted upon and
probably never set to anything but |true|, we change ::RemoveSource to
not have a return value.
NOTRY=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2506173003
Cr-Commit-Position: refs/heads/master@{#15150}
Reason for revert:
The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change:
function removeVideoCodec(offerSdp) {
- offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n',
- 'a=rtpmap:100 XVP8/90000\r\n');
+ offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/,
+ 'a=rtpmap:$1 XVP8/90000\r\n');
return offerSdp;
}
Original issue's description:
> Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ )
>
> Reason for revert:
> Breaks chromium.fyi test:
> WebRtcBrowserTest.NegotiateUnsupportedVideoCodec
>
> Original issue's description:
> > Stop using hardcoded payload types for video codecs
> >
> > This CL stops using hardcoded payload types for different video codecs
> > and will dynamically assign them payload types incrementally from 96 to
> > 127 instead.
> >
> > This CL:
> > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in
> > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for
> > internally supported software codecs instead. The purpose is to
> > streamline the payload type assignment in webrtcvideoengine2.cc which
> > will now have two encoder factories of the same
> > WebRtcVideoEncoderFactory type; one internal and one external.
> > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec
> > instead.
> > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and
> > moves the create function to the internal encoder factory instead.
> > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an
> > interface without any static functions.
> > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies
> > the internal and external codecs and assigns them payload types
> > incrementally from 96 to 127.
> > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about
> > what payload types will be used.
> >
> > BUG=webrtc:6677,webrtc:6705
> > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org
> >
> > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b
> > Cr-Commit-Position: refs/heads/master@{#15135}
>
> TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6677,webrtc:6705
>
> Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591
> Cr-Commit-Position: refs/heads/master@{#15140}
TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6677,webrtc:6705
Review-Url: https://codereview.webrtc.org/2511933002
Cr-Commit-Position: refs/heads/master@{#15148}
Reason for revert:
Breaks downstream projects:
error: undefined reference to 'rtc::ExpFilter::kValueUndefined'
error: undefined reference to 'rtc::ExpFilter::Apply(float, float)'
error: undefined reference to 'rtc::ExpFilter::Reset(float)'
rror: undefined reference to 'rtc::ExpFilter::UpdateBase(float)'
Original issue's description:
> Move smoothing filter to common audio.
>
> This will make the smoothing filter a basic tool that is going to be used by both voice engine and ANA.
>
> BUG=webrtc:6443
>
> Committed: https://crrev.com/a82395bf7cd15b7396456df06fe952ede8db0c39
> Cr-Commit-Position: refs/heads/master@{#15146}
TBR=minyue@webrtc.org,solenberg@webrtc.org,perkj@webrtc.org,tommi@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
Review-Url: https://codereview.webrtc.org/2510373002
Cr-Commit-Position: refs/heads/master@{#15147}
This will make the smoothing filter a basic tool that is going to be used by both voice engine and ANA.
BUG=webrtc:6443
Review-Url: https://codereview.webrtc.org/2484153002
Cr-Commit-Position: refs/heads/master@{#15146}
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}
