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}
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}
PayloadRouter::OnEncodedImage() was casing boolean result from
SendOutgoingData() to int, and then not handling it correctly, which
results in all errors in SendOutgoingData() being ignored. This issue
was introduced in
https://chromium.googlesource.com/external/webrtc/+/ad34dbe934
This bug masked another issue with VP9 codec (see
crbug.com/webrtc/6723 ) and that increased number of dropped frames.
BUG=634816
Review-Url: https://codereview.webrtc.org/2512543002
Cr-Commit-Position: refs/heads/master@{#15143}
Added the callback in native c++ API.
The callback function is called when a track is added and a new RtpReceiver is created.
The application can tell when tracks are added to the streams by listening to this callback.
BUG=webrtc:6112
Review-Url: https://codereview.webrtc.org/2505173002
Cr-Commit-Position: refs/heads/master@{#15142}
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
Review-Url: https://codereview.webrtc.org/2513633002
Cr-Commit-Position: refs/heads/master@{#15140}
Reason for revert:
Breaks downstream import.
Original issue's description:
> Split out target rtc_media_base from rtc_media
>
> The purpose with this CL is to be able to depend on
> cricket::VideoCodec (webrtc/media/base/codec.h) from other targets
> without getting cyclic dependencies.
>
> BUG=webrtc:6402,webrtc:6337
>
> NOTRY=True
>
> Committed: https://crrev.com/aae7e7cf35a5bb43ebbaf75396aa7ccc544e920a
> Cr-Commit-Position: refs/heads/master@{#15137}
TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6402,webrtc:6337
Review-Url: https://codereview.webrtc.org/2508163002
Cr-Commit-Position: refs/heads/master@{#15139}
Add new perf desktop bots.
Remove dcheck_always_on=true for all perf bots.
Cleanup some more GYP traces.
Remove gn_ prefix for all mixins for readability.
BUG=chromium:665874
NOTRY=True
Review-Url: https://codereview.webrtc.org/2505183003
Cr-Commit-Position: refs/heads/master@{#15138}
The purpose with this CL is to be able to depend on
cricket::VideoCodec (webrtc/media/base/codec.h) from other targets
without getting cyclic dependencies.
BUG=webrtc:6402,webrtc:6337
NOTRY=True
Review-Url: https://codereview.webrtc.org/2471573003
Cr-Commit-Position: refs/heads/master@{#15137}
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
Review URL: https://codereview.webrtc.org/2493133002 .
Cr-Commit-Position: refs/heads/master@{#15135}
This is a refactoring change in preparation for enabling AudioMixer
with the goal to have a small CL as possible for passing audio through
the new audio mixer in WebRTC. The dependent CL https://codereview.webrtc.org/2436033002/
enables the mixer.
An object of class AudioState is shared across different webrtc audio
connections. It is created in tests and in
WebRTCVoiceEngine. AudioState is constructed by passing a Config
struct, where one argument is scoped_refptr<AudioMixer>.
Populating this field has now been mandatory. Tests and
WebRTCVoiceEngine create and pass either a AudioMixerImpl.
WebRTCVoiceEngine passes a real AudioMixer, which is
currently unused.
An alternative would have tests pass a mocked audio mixer. We
chose not to do that, because we believe that tests should use
the real thing unless there are reasons against it. Construction
time is not an issue, because the real mixer is relatively
lightweight.
We couldn't find a way to test any mixer-related changes in AudioState
before the mixes is connected. The next dependent CL
https://codereview.webrtc.org/2436033002/ contains unit tests for
mixer usage.
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2469743002
Cr-Commit-Position: refs/heads/master@{#15134}
All audio in calls is now routed through AudioTransportProxy. The
AudioTransport implemented by VoEBaseImpl is disconnected from
AudioDevice and replaced by an empty proxy layer that forwards calls
to the old Transport. This is a refactoring CL in preparation for
landing https://codereview.webrtc.org/2436033002/, which will connect
the new AudioMixer.
In the planned configuration, the currently empty AudioTransportProxy
will query the new mixer for audio instead of polling data from the
old Transport. Mixed audio will be passed to an AudioProcessing
interface. AudioTransportProxy is initialized with an AudioProcessing*,
which is currently unused.
No presubmit since we implement an interface with non-const references.
NOPRESUBMIT=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2454373002
Cr-Commit-Position: refs/heads/master@{#15133}
This will ensure that the estimated likelihood starts at a low value and prevents initial spikes.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2503843004
Cr-Commit-Position: refs/heads/master@{#15131}
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}
Parse the estimation parameters from the field trial string.
BUG=webrtc:6690
Review-Url: https://codereview.webrtc.org/2489323002
Cr-Commit-Position: refs/heads/master@{#15126}
To be able to smooth the bandwidth estimation according to the probing interval.
BUG=webrtc:6443
Review-Url: https://codereview.webrtc.org/2380883003
Cr-Commit-Position: refs/heads/master@{#15123}
The red acronym is already in use in the context of audio coding, so it is better to avoid reusing it here because it could be confusing.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2505993002
Cr-Commit-Position: refs/heads/master@{#15121}
This CL wires everything up and enables actual setting of the max bitrate encoding parameter
on the video RTP sender.
The following changes were made
* Add maxbitrate property to the settings model and settings store. Make sure to store and
read the maxbitrate from storage (to persist between app launches and make testing easier)
* Fix setup of encoding parameters for the rtp sender as previous timing was not right.
* Fix header of RTCRtpSender to expose needed parameter
BUG=webrtc:6654
Review-Url: https://codereview.webrtc.org/2492693003
Cr-Commit-Position: refs/heads/master@{#15120}
This CL adds full stack tests that are used to measure the performance
of H264 with and without FlexFEC. In order to not increase the bot
run time, the CL also reduces the test time to 45 secs. This should
be OK, since the BWE is faster to ramp up nowadays.
Due to the test time change, there may be some performance alerts.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2499273002
Cr-Commit-Position: refs/heads/master@{#15118}
Will be used by full stack tests and video_loopback.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2500373002
Cr-Commit-Position: refs/heads/master@{#15114}
Just returns the configuration the PC was constructed with, or the last
one passed into SetConfiguration.
BUG=chromium:587453
Review-Url: https://codereview.webrtc.org/2504103002
Cr-Commit-Position: refs/heads/master@{#15111}
Previously AlrDetector was measuring amount of data sent in each 100ms
interval and would enter ALR mode after 5 consecutive intervals when
average bandwidth usage doesn't exceed 30% of the current estimate
estimate. This meant that an application that uses only slightely more
than 6% of total bandwidth may stay out of ALR mode, e.g. if it sends
a frame of size BW*30ms every 0.5 seconds. 100ms is too short interval
to average over, particularly when frame-rate falls below 10fps.
With this change AlrDetector averages BW usage over last 500ms. It then
enters ALR state when usage falls below 30% and exits it when usage
exceeds 50%.
BUG=webrtc:6332
R=philipel@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2503643003 .
Cr-Commit-Position: refs/heads/master@{#15109}
