Some frames are already marked as 'timing frames' via video-timing RTP header extension. Timestamps along full WebRTC pipeline are gathered for these frames. This CL implements reporting of these timestamps for a single
timing frame since the last GetStats(). The frame with the longest end-to-end delay between two consecutive GetStats calls is reported.
The purpose of this timing information is not to provide a realtime statistics but to provide debugging information as it will help identify problematic places in video pipeline for outliers (frames which took longest to process).
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2946413002
Cr-Commit-Position: refs/heads/master@{#18909}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
Can be enabled by setting "enable_encrypted_rtp_header_extensions" in
"crypto_options" of "PeerConnectionFactoryInterface::Options" and will
only be used if both peers support it.
BUG=webrtc:3411
Review-Url: https://codereview.webrtc.org/2761143002
Cr-Commit-Position: refs/heads/master@{#18842}
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
This CL makes the WebRTC more modular and allows the users to build
WebRTC without audio and video(DataChannel only).
The BUILD files in call/, logging/, media/ and pc/ are modified to
support modular WebRTC.
The dependencies on Call and RtcEventLog are removed from the
PeerConnection. Instead of being created internally, they would be
passed in by the PeerConnectionFactory.
Add the CreateModularPeerConnectionFactory function which allow the
users to create a PeerConnectionFactory with the modules they need.
If the users want to build WebRTC without audio and video, they can
pass in null pointers for modules they don't need. (MediaEngine,
VideoEncoderFactory etc.)
BUG=webrtc:7613
Review-Url: https://codereview.webrtc.org/2854123003
Cr-Commit-Position: refs/heads/master@{#18617}
This eliminates a thread hop in PeerConnectionFactory initialization,
and will allow some code to be simplified.
BUG=None
Review-Url: https://codereview.webrtc.org/2934103002
Cr-Commit-Position: refs/heads/master@{#18613}
WebRtcVideoChannel and and WebRtcVideoEngine seem to have been removed, and only WebRtcVideoChannel2 and WebRtcVideoEngine2 remain, which removes the need for the "2" postfix.
BUG=None
Review-Url: https://codereview.webrtc.org/2932073002
Cr-Commit-Position: refs/heads/master@{#18531}
Reason for revert:
Broken downstream projects
Original issue's description:
> Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls.
>
> Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
>
> BUG=webrtc:5079
> R=deadbeef@webrtc.org, hbos@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2863123002 .
> Cr-Commit-Position: refs/heads/master@{#18384}
> Committed: e80f4c91d0TBR=hbos@webrtc.org,deadbeef@webrtc.org,holmer@google.com,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2916793003
Cr-Commit-Position: refs/heads/master@{#18386}
Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
BUG=webrtc:5079
R=deadbeef@webrtc.org, hbos@webrtc.org
Review-Url: https://codereview.webrtc.org/2863123002 .
Cr-Commit-Position: refs/heads/master@{#18384}
Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2799033006
Cr-Commit-Position: refs/heads/master@{#17977}
With the upcoming changes to WebRtcVoiceEngine due to injectable audio
encoders, codec names will no longer be normalized against the ACM's
codec database. This not only breaks a couple of downstream tests
(which I've written fixes for) but also runs the risk of breaking
other external usage, like SDP mangling etc.
Eventually, we should change to using the spelling from the relevant
RFCs, although not as part of this other, large change.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2845603002
Cr-Commit-Position: refs/heads/master@{#17903}
When SSRCs aren't signaled, an SSRC of 0 is used internally to mean
"the default receive stream". But this wasn't working with the
implementation of GetRtpReceiveParameters in the audio/video
engines. This was breaking RtpReceiver.GetParameters in this situation,
as well as the new getStats implementation (which relies on
GetParameters).
The new implementation will fail if *no* default receive stream is
configured (meaning no default sink is set), and otherwise will return
a default RtpEncodingParameters (later it will be filled with relevant
SDP parameters as they're implemented).
BUG=webrtc:6971
Review-Url: https://codereview.webrtc.org/2806173002
Cr-Commit-Position: refs/heads/master@{#17803}
The example that brought up this issue was:
1. Do offer/answer exchange.
2. Later, remove the audio/video stream.
3. Add back a new stream, that contains only the audio track.
4. Do new offer/answer.
The new offer didn't have the new stream ID, but code elsewhere was
expecting one. As a result, the send stream is never hooked up to the
audio track, and audio packets aren't sent.
BUG=chromium:611708
Review-Url: https://codereview.webrtc.org/2810733003
Cr-Commit-Position: refs/heads/master@{#17709}
The compiler is allowed to leave a chopped-off horse's head on your
bed if you do.
Happily, we didn't even have to move this function to another namespace,
since it was unused.
BUG=webrtc:7484
Review-Url: https://codereview.webrtc.org/2815203002
Cr-Commit-Position: refs/heads/master@{#17696}
Also renames "peerconnection_unittests" to "peerconnection_integrationtests",
and moves the ICE URL parsing code to separate files.
The main problem previously was that the test assertions
occurred in various places in the main test class, and this shared test
code was overly complex and stateful. As a result, it was difficult to
tell what a test even does, let alone what assertions it's meant to be
making. And writing a new test that does what you want can be a
frustrating ordeal.
The new code still uses helper methods, but they have intuitive names
and a smaller role; all of the important parts of the test's logic are
in the test case itself.
We're planning on merging PeerConnection and WebRtcSession at some point
soon, so it seemed valuable to do this, so that the WebRtcSession tests
can be rewritten as PeerConnection tests using better patterns.
BUG=None
Review-Url: https://codereview.webrtc.org/2738353003
Cr-Commit-Position: refs/heads/master@{#17458}
In ViEEncoder, try to reduce framerate instead of resolution if the
current degradation preference is maintain-resolution rather than
balanced.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2716643002
Cr-Commit-Position: refs/heads/master@{#17327}
This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver
They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:
* You can only have one of each type of sender and receiver (audio/video) on top
of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.
Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:
ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine
And later we hope to have simply:
PeerConnection -> "Real" ORTC objects -> Media engine
See the linked bug for more context.
BUG=webrtc:7013
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}
This CL fixes issue 7211 as well as adding a test that would have
caught the issue.
BUG=webrtc:7211,webrtc:6850,b/35471214
R=sprang@webrtc.org
TBR=kthelgason, sprang
Review-Url: https://codereview.webrtc.org/2713683002 .
Cr-Commit-Position: refs/heads/master@{#16778}
It's annoying to have to re-implement this every time I need a debug
printout.
Declared inline, so that there'll be zero runtime overhead.
This CL also modifies a unit test so that it will make use of the new
operator<< in case it finds errors.
BUG=none
Review-Url: https://codereview.webrtc.org/2705203002
Cr-Commit-Position: refs/heads/master@{#16749}
This helps show where ownership is transfered between objects.
Specifically, this CL wraps cricket::VideoCapturer, MediaEngineInterface
and DataEngineInterface in unique_ptr.
BUG=None
TBR=magjed@webrtc.org
Review-Url: https://codereview.webrtc.org/2685093002
Cr-Commit-Position: refs/heads/master@{#16548}
The current method with max_pixel_count and max_pixel_count_step_up,
where only one should be used at a time and this first signaling an
inclusive upper bound and other other an exclusive lower bound, makes
for a lot of confusion.
I've updated this to have a desired target and a maximum instead. The
source should select a resolution as close to the target as possible,
but no higher than the maximum.
I intend to also add similar frame rate settings in an upcoming cl.
BUG=webrtc:4172,webrtc:6850
Review-Url: https://codereview.webrtc.org/2672793002
Cr-Commit-Position: refs/heads/master@{#16533}
These methods relate to typing detection but are not used anymore. Typing detection is enabled through the VoiceDetection module on the APM.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2684933008
Cr-Commit-Position: refs/heads/master@{#16527}
Replaced by assigning value to a local variable, followed by a DCHECK.
Also deletes dead test code under the always false TEST_DIGEST define.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2623473004
Cr-Commit-Position: refs/heads/master@{#16476}
