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}
The Android HW encoder is currently not setting any H264 codec parameters or profile information. No profile-level-id means Baseline Level 1, but we are actually using Contrained Baseline Level 3.1. This CL sets the correct codec parameters.
BUG=webrtc:6337
Review-Url: https://codereview.webrtc.org/2497163002
Cr-Commit-Position: refs/heads/master@{#15247}
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}
Add a fallback path to tools/valgrind-webrtc/webrtc_tests.sh if locate_valgrind.sh fails.
This is a workaround to run memcheck on swarming, since locate_valgrind.sh fails even though the files are present. This is almost certainly because the way we use symlinks.
A warning message is displayed to warn the developers to follow the instructions to get the valgrind binaries.
Some extra suppressions were needed. The bug tracking them is https://bugs.webrtc.org/6773R=kjellander@chromium.org
BUG=chromium:497757
Review-Url: https://codereview.webrtc.org/2531573003
Cr-Commit-Position: refs/heads/master@{#15244}
It will be necessary to keep the H264 profile information in
VideoReceiveStream::Decoder. I think it will be easier now and for the
future to just store all of the codec parameters unmodified in
VideoReceiveStream::Decoder instead of extracting a subset of them to an
ad hoc class.
BUG=webrtc:6743,webrtc:5948
Review-Url: https://codereview.webrtc.org/2523773003
Cr-Commit-Position: refs/heads/master@{#15239}
WebRTC.Video.BitrateSentInKbps
WebRTC.Video.MediaBitrateSentInKbps
WebRTC.Video.PaddingBitrateSentInKbps
WebRTC.Video.RetransmittedBitrateSentInKbps
WebRTC.Video.FecBitrateSentInKbps
RtpSender has two StreamDataCounters: for the non-RTX and the RTX stream.
The same counter (for the non-RTX stream) is reported for both the media SSRC and the FlexFEC SSRC.
Bitrate stats are summed for all SSRCs, thus the counter for the non-RTX stream is counted twice.
Do not store the counter for the FlexFEC SSRC.
Do not include info from FlexFEC substreams in VideoSendStream::Stats::ToString (periodically logged during a call).
BUG=webrtc:6774
Review-Url: https://codereview.webrtc.org/2525293002
Cr-Commit-Position: refs/heads/master@{#15238}
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:
Include fix for downstream import.
Original issue's description:
> Revert of Split out target rtc_media_base from rtc_media (patchset #3 id:40001 of https://codereview.webrtc.org/2471573003/ )
>
> 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
>
> Committed: https://crrev.com/0d0d7531b50a78efe7468610395e9dc5f496e2e9
> Cr-Commit-Position: refs/heads/master@{#15139}
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/2509123003
Cr-Commit-Position: refs/heads/master@{#15235}
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}
When set to true, this adds the files necessary to run memcheck as data dependencies, listed in the .gni files.
This will enable us to run memcheck on swarming.
R=kjellander@chromium.org
BUG=chromium:497757
NOTRY=True
Review-Url: https://codereview.webrtc.org/2510033004
Cr-Commit-Position: refs/heads/master@{#15219}
Change the default value of rtcp-mux policy in RTCConfiguration.
Refactor the peerconnectioninterface and webrtcsession unit tests.
BUG=webrtc:6030
Review-Url: https://codereview.webrtc.org/2043193003
Cr-Commit-Position: refs/heads/master@{#15217}
I forget to remove these when fixing them.
BUG=chromium:636818
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2522023003
Cr-Commit-Position: refs/heads/master@{#15215}
There shouldn't be a need to bind the framebuffer to modify the texture
size. Binding the framebuffer causes a crash on Samsung Galaxy S3 mini
for some reason.
BUG=webrtc:6470
Review-Url: https://codereview.webrtc.org/2524003002
Cr-Commit-Position: refs/heads/master@{#15210}
RTCStatsCollector supports "payloadType", "codec" and "clockRate".
"channels", "parameters" and "implementation" need to be supported
before closing crbug.com/659117.
[1] https://w3c.github.io/webrtc-stats/#codec-dict*
BUG=chromium:659117, chromium:627816, chromium:657854
NOTRY=True
Review-Url: https://codereview.webrtc.org/2509803004
Cr-Commit-Position: refs/heads/master@{#15207}
Increases readibility and helps with the formating.
BUG=webrtc:6680
Review-Url: https://codereview.webrtc.org/2488973002
Cr-Commit-Position: refs/heads/master@{#15206}
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}
