Commit Graph

25 Commits

Author SHA1 Message Date
a35df4260f WebRtcVideoEngine: Encapsulate logic for unifying internal and external video codecs
This CL encapsulates the logic for unifying the internal and external
video encoders into a helper class. The purpose is to prepare for
introducing a new video encoder factory interface that inherently
represents all encoders (i.e. both internal and external). A helper
interface EncoderFactoryAdapter is introduced that both the old
WebRtcVideoEncoderFactory and the new factory interface can implement
and serves as common point to leave the rest of the code unchanged.

BUG=webrtc:7925

Review-Url: https://codereview.webrtc.org/3006713002
Cr-Commit-Position: refs/heads/master@{#19600}
2017-08-30 11:21:30 +00:00
f52d34d682 Let VideoEncoderSoftwareFallbackWrapper own the wrapped encoder
Currently, ownership of the wrapped hardware encoder is handled outside
VideoEncoderSoftwareFallbackWrapper. It's easier if
VideoEncoderSoftwareFallbackWrapper owns and relases it instead.

BUG=webrtc:7925

Review-Url: https://codereview.webrtc.org/3007683002
Cr-Commit-Position: refs/heads/master@{#19572}
2017-08-29 07:58:52 +00:00
3f89758398 Clean up ownership of webrtc::VideoEncoder
Currently, webrtc::VideoEncoders are supposed to be deleted through the
factory that created them with the
WebRtcVideoEncoderFactory::DestroyVideoEncoder method. In practice,
we sometimes use this method and sometimes we just call delete on the
webrtc::VideoEncoder pointer. We want to be able to consistently use the
normal destructor of webrtc::VideoEncoder instead of having to call
DestroyVideoEncoder so that we can put webrtc::VideoEncoder inside
an std::unique_ptr and make ownership more clear. As part of webrtc:7925
we also want to make a new encoder factory class that does not have the
DestroyVideoEncoder() method, and this CL is a step in that direction.

This CL introduces a helper function CreateScopedVideoEncoder that takes
a webrtc::VideoEncoder and a WebRtcVideoEncoderFactory pointer, and
returns a new webrtc::VideoEncoder instance that can be deleted through
the regular destructor.

This CL also removes WebRtcSimulcastEncoderFactory that almost only
contains logic for handling the DestroyVideoEncoder calls that we no
longer need, and inlines the rest of the logic inside the
WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoder method.

BUG=webrtc:7925

Review-Url: https://codereview.webrtc.org/3007643002
Cr-Commit-Position: refs/heads/master@{#19564}
2017-08-28 15:05:42 +00:00
26e3abbb40 Reverse |rtx_payload_types| map, and rename.
New name is |rtx_associated_payload_types|.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/3000273002
Cr-Commit-Position: refs/heads/master@{#19514}
2017-08-25 11:44:25 +00:00
1cdddc96fa Make CodecType conversion functions non-optional.
We can't handle no value here anyway and end up setting a default
at each call site. The defaults aren't even the same in each place.

BUG=None

Review-Url: https://codereview.webrtc.org/2998293002
Cr-Commit-Position: refs/heads/master@{#19485}
2017-08-24 10:52:48 +00:00
a79cc28de1 Report max interframe delay over window insdead of interframe delay sum
Maximum of interframe delay is calculated over moving window in
ReceiveStatistics proxy now and reported via GetStats. Name of a metric
is also changed.

BUG=none

Review-Url: https://codereview.webrtc.org/2995143002
Cr-Commit-Position: refs/heads/master@{#19463}
2017-08-23 12:24:10 +00:00
eb13f5e400 Add video timing frames to set of default RTP header extensions
BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2994893002
Cr-Commit-Position: refs/heads/master@{#19449}
2017-08-22 14:05:47 +00:00
2bf9e73e6b Delete unneeded Start and Stop methods on FlexfecReceiveStream.
Bug: None
Change-Id: I3013cfc54ed357901f175dd408127eda75e5ba99
Reviewed-on: https://chromium-review.googlesource.com/542735
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19363}
2017-08-16 09:41:27 +00:00
ee21f374ca Default enable content type rtp header extension
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2998843002
Cr-Commit-Position: refs/heads/master@{#19346}
2017-08-15 08:32:51 +00:00
186d9c3873 Renamed fields in common_types.h/RtcpStatistics.
BUG=webrtc:8033

Review-Url: https://codereview.webrtc.org/2992043002
Cr-Commit-Position: refs/heads/master@{#19247}
2017-08-04 12:03:53 +00:00
c0d481a4a6 Protected streams report RTP messages directly to the FlexFec streams
In preparation of making RTP packet demuxing many-to-one (one SSRC goes to one sink, but one sink may have multiple SSRCs), we need to remove FlexFEC's dependence on being able to register itself with the demuxer. Instead, we register FlexFEC streams with the streams they protect; when those streams get packets, they'll forward them to their associated FlexFEC streams, too.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2974453002
Cr-Commit-Position: refs/heads/master@{#19219}
2017-08-02 14:39:07 +00:00
f032e4041c Revert "Prefer external video codecs over internal in SDP"
This reverts commit 06f3aae345854ba9dcc5ae3b603de1f86505acf9.

The reason for reverting is that it seems to break Chromium importer. See https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/17862

BUG=None

TBR=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2982053002
Cr-Commit-Position: refs/heads/master@{#19058}
2017-07-17 15:45:17 +00:00
06f3aae345 Prefer external video codecs over internal in SDP
Currently, when we generate the list of supported video codecs that will
be signaled in SDP, we start with the internal video codecs and then
append the external video codecs. When we create a video encoder for a
given codec, we prefer an external encoder over an internal encoder.

This CL lists the external video codecs first in SDP instead, so that we
consistently prefer external video codecs over internal.

The reason for doing this is that we will otherwise prefer an internal
SW H264 encoder over an external HW H264 encoder if the H264 profiles
differs.

BUG=chromium:688541

Review-Url: https://codereview.webrtc.org/2974383002
Cr-Commit-Position: refs/heads/master@{#19026}
2017-07-14 17:36:23 +00:00
6cc25614a9 Remove webrtc::VideoEncoderFactory
Replace the use of webrtc::VideoEncoderFactory with
cricket::WebRtcVideoEncoderFactory and remove the adapter classes
between these two factory types.

Some code changes were necessary in order to accomplish this:
 * Move SimulcastEncoderAdapter from
   webrtc/modules/video_coding/codecs/vp8 to webrtc/media/engine (that's
   where it's used).
 * Rename simulcast_unittest.h to simulcast_test_utility.h and make it
   into it's own target, because it's used from both
   simulcast_unittest.cc and simulcast_encoder_adapter_unittest.cc.
 * Remove ownership of the encoder factory from SimulcastEncoderAdapter,
   and make the necessary changes in surrounding code.

The goal with this CL is to clean up the code, and also to free up
the name webrtc::VideoEncoderFactory for future use.

BUG=webrtc:7925

Review-Url: https://codereview.webrtc.org/2964953002
Cr-Commit-Position: refs/heads/master@{#18945}
2017-07-10 10:26:36 +00:00
f04afde85a Report interframe delay sum in old GetStats
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2965033002
Cr-Commit-Position: refs/heads/master@{#18924}
2017-07-07 08:26:24 +00:00
c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00
2edc6845ac Report timing frames info in GetStats.
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}
2017-07-06 10:06:50 +00:00
a80c16a67c Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
This reverts commit c3771cc4d37f5573fe53b7c7cff295a4f0f9560f.
(breaks downstream internal project)

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2972463002 .
Cr-Commit-Position: refs/heads/master@{#18873}
2017-07-01 14:48:18 +00:00
c3771cc4d3 Update includes for webrtc/{base => rtc_base} rename (2/3)
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}
2017-06-30 20:42:44 +00:00
04f4d126f8 Implement timing frames.
Timing information is gathered in EncodedImage,
starting at encoders. Then it's sent using RTP header extension. In the
end, it's gathered at the GenericDecoder. Actual reporting and tests
will be in the next CLs.

BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2911193002
Cr-Commit-Position: refs/heads/master@{#18659}
2017-06-19 14:18:55 +00:00
6b826ef66d Add cropping to VIEEncoder to match simulcast streams resolution
Detect when simulcaststreamfactory adjust resolution and remeber cropping
parameters in VIEEncoder.
Expose EncoderStreamFactory in webrtcvideoengine2.

BUG=webrtc:7375, webrtc:6958

Review-Url: https://codereview.webrtc.org/2936393002
Cr-Commit-Position: refs/heads/master@{#18632}
2017-06-16 13:53:48 +00:00
67561a6411 Use the same QP max for tests as in production
BUG=webrtc:7664

Review-Url: https://codereview.webrtc.org/2941023002
Cr-Commit-Position: refs/heads/master@{#18611}
2017-06-15 13:34:42 +00:00
a5e0df6438 Move MinPositive to call.h as discussed here: https://codereview.chromium.org/2888303005/#msg19
TBR=stefan@webrtc.org
BUG=webrtc:7395

Review-Url: https://codereview.webrtc.org/2924393002
Cr-Commit-Position: refs/heads/master@{#18599}
2017-06-14 18:41:48 +00:00
3c81a1afd8 Add field trial for balanced degradation preference.
BUG=webrtc:7607

Review-Url: https://codereview.webrtc.org/2923563002
Cr-Commit-Position: refs/heads/master@{#18589}
2017-06-14 12:52:21 +00:00
f184138a5f s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine
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}
2017-06-12 08:16:46 +00:00