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}
This function has no public use,
removed tests calling it: effect of registering extension is better
tested in AllocatePacket and SendPacket tests.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2530363002
Cr-Commit-Position: refs/heads/master@{#15258}
Reason for revert:
Include fix; set profile information in CreatePayloadType for video.
Original issue's description:
> Revert of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:40001 of https://codereview.webrtc.org/2525693003/ )
>
> Reason for revert:
> The CL doesn't actually set profile information in VideoPayload.
>
> Original issue's description:
> > Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload
> >
> > It's necessary to check H264 profile information as well as payload name
> > in PayloadIsCompatible in RTPPayloadRegistry.
> >
> > BUG=webrtc:6743
> >
> > Committed: https://crrev.com/bdbc4b7ef578ba1d61ceec351bc47c33da115329
> > Cr-Commit-Position: refs/heads/master@{#15248}
>
> TBR=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/d7e6ccbc53fc24acdcc7507a6f3a155626473d54
> Cr-Commit-Position: refs/heads/master@{#15251}
TBR=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/2529153002
Cr-Commit-Position: refs/heads/master@{#15252}
Reason for revert:
The CL doesn't actually set profile information in VideoPayload.
Original issue's description:
> Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload
>
> It's necessary to check H264 profile information as well as payload name
> in PayloadIsCompatible in RTPPayloadRegistry.
>
> BUG=webrtc:6743
>
> Committed: https://crrev.com/bdbc4b7ef578ba1d61ceec351bc47c33da115329
> Cr-Commit-Position: refs/heads/master@{#15248}
TBR=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/2529143002
Cr-Commit-Position: refs/heads/master@{#15251}
A recent cl (https://codereview.webrtc.org/2510583002) introduced an
issue where temporal layers may return incorrect bitrates, given that
they are stateful and that the GetPreferredBitrateBps is called.
The fix is to use a temporary simulcast rate allocator instance, without
temporal layers, and get the preferred bitrate from that.
Additionally, some regression in bitrate allocated stems from overly
often reconfiguring the encoder, which yields suboptimal rate control.
The fix here is to limit encoder updates to when values have actually
changed.
As a bonus, dchecks added by this cl found a bug in the (unused) RealtimeTemporalLayers implementation. Fixed that as well.
BUG=webrtc:6301, chromium:666654
Review-Url: https://codereview.webrtc.org/2529073003
Cr-Commit-Position: refs/heads/master@{#15250}
It's necessary to check H264 profile information as well as payload name
in PayloadIsCompatible in RTPPayloadRegistry.
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2525693003
Cr-Commit-Position: refs/heads/master@{#15248}
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}
Turns out this function is needed by external code.
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2532663002
Cr-Commit-Position: refs/heads/master@{#15237}
- Out from modules/utility/ and into modules/audio_device/ios/ - there they are used.
BUG=none
Review-Url: https://codereview.webrtc.org/2526273002
Cr-Commit-Position: refs/heads/master@{#15236}
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}
This perf tests the echo detector in 3 scenarios: standalone, as part of APM with only the echo detector enabled and as part of a normally configured APM.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2517523003
Cr-Commit-Position: refs/heads/master@{#15224}
RunPlayoutAndRecordingInFullDuplex fails sometimes on Android swarming
bots, presumably because the timing is hardware dependent.
This test ensures that audio starts pumping. The exact performance is
not that important.
R=kjellander@webrtc.org, henrika@webrtc.org
BUG=webrtc:6464
NOTRY=True
Review-Url: https://codereview.webrtc.org/2525943003
Cr-Commit-Position: refs/heads/master@{#15223}
The histogram will log a new value every time the AGC changes level_.
BUG=webrtc:6622
Review-Url: https://codereview.webrtc.org/2525963002
Cr-Commit-Position: refs/heads/master@{#15222}
This change adds RotateDesktopFrame(), RotateRect(), RotateSize(),
ReverseRotate() functions, so an implementation can use these free functions to
rotate and copy pixels from one DesktopFrame to another at the same time.
This is the first part of the change to support rotation in DirectX capturer. In
a coming change, these functions will be used in DxgiOutputDuplicator to do the
rotation and copying.
Background,
DirectX APIs always return unrotated data buffer, so we need to rotate it to
match the user-selected rotation. What worse is except for the data buffer,
other variables return by these APIs are all rotated, e.g. output size, monitor
position. So we will eventually not be able to capture the rotated monitors,
because we cannot set their position and size correctly. Though
DXGI_OUTDUPL_DESC provides a DXGI_MODE_ROTATION enumeration to indicate the
output rotation, it does not provide a simple way to rotate an IDXGIResource,
which is the only thing we can get from duplication APIs. A typical user case
here is to use a matrix to transform the IDXGIResource and render it to a
surface. But since we do not render the IDXGIResource at all, we need to
manually rotate it.
BUG=314516
Review-Url: https://codereview.webrtc.org/2500883004
Cr-Commit-Position: refs/heads/master@{#15205}
Previously ProbeController was starting probing as soon as SetBitrates()
is called. As result these probes would often timeout while connection
is being established. Now ProbeController receives notifications about
network route changes. This allows to start probing only when transport
is connected. This also makes it possible to restart probing whenever
transport route changes (will be done in a separate change).
BUG=webrtc:6332
R=honghaiz@webrtc.org, philipel@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2458863002 .
Committed: https://crrev.com/5c99c76255ee7bface3c742c25fb5617748ac86e
Cr-Original-Commit-Position: refs/heads/master@{#15094}
Cr-Commit-Position: refs/heads/master@{#15204}
Set the |qp_| field in EncodedImage before calling OnEncodedImage.
BUG=webrtc:6541
Review-Url: https://codereview.webrtc.org/2499003002
Cr-Commit-Position: refs/heads/master@{#15199}
the APM parameters to the high-pass filter.
The introduction will be done in three steps:
1) This CL which introduces the new scheme and
changes the code in webrtcvoiceengine.cc to use it.
2) Introduce the scheme into upstream code.
3) Remove the HighPassFilter interface in APM.
BUG=webrtc::6220, webrtc::6296, webrtc::6297, webrtc::6181, webrtc::5298
Review-Url: https://codereview.webrtc.org/2415403002
Cr-Commit-Position: refs/heads/master@{#15197}
This new histogram will log a value every time input clipping is
detected. The value is a boolean, with "true" meaning that the gain
was in fact adjusted in response to the detected clipping, and "false"
meaning that adjustment was not allowed due to kClippedLevelMin.
BUG=webrtc:6622
Review-Url: https://codereview.webrtc.org/2522543006
Cr-Commit-Position: refs/heads/master@{#15196}
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}
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}
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}
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:
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}
