14 Commits

Author SHA1 Message Date
minyue
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
magjed
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
magjed
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
ilnik
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
Edward Lemur
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
ilnik
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
Henrik Kjellander
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
kjellander
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
ilnik
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
ilnik
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
sprang
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
zstein
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
asapersson
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
eladalon
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