This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
This is a re-land of https://codereview.webrtc.org/1353263005/
which was reverted because of perf-regressions. Changes since that CL:
* Change LayerFilteringTransport to send a padding packet instead of
dropping it for data that should be filtered out. This prevents
confusion due to changed sequence numbers.
* Changed timing of stats poller thread in VideoAnalyzer. Startup was
racy wrt initializion of send_stream_.
* Minor formatting issues.
PERF NOTE: This change will affect some performance numbers slightly.
In particular, {encode_frame_rate, encode_time_ms,
encode_usage_percent, media_bitrate_bps} will change due to timing
of the measurements.
BUG=
R=pbos@webrtc.orgTBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1412233003
Cr-Commit-Position: refs/heads/master@{#10483}
Required a bit of refactoring to make it possible to pass a Call to DirectTransport on construction. This also lead to me having to remove the shared lock between PacketTransport and RtpRtcpObserver. Now RtpRtcpObserver has a SetTransports method instead of a SetReceivers method.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1419193002
Cr-Commit-Position: refs/heads/master@{#10430}
Reason for revert:
Temporarily reverting as this causes some issues with perf tests. Especially tests with packet loss no longer works.
Original issue's description:
> Adding support for simulcast and spatial layers into VideoQualityTest
>
> The CL includes several changes:
> - Adding flags describing the streams and spatial layers.
> - Reorganizing the order of the flags, to make them easier to maintain.
> - Adding a member .params_ to VideoQualityAnalyzer.
> (instead of passing it to every member function manually)
> - Updating VideoAnalyzer to support simulcast.
> (select appropriate ssrc and fix timestamps which are sometimes increased by 1)
> - VP9EncoderImpl already had code for automatic calculation of bitrate for each layer.
> Changing to first read bitrates and resolution ratios from the flags, if specified.
> If not specified, reverting to the old code are setting the values automatically.
> - Changing the parameters in LayerFilteringTransport, replacing
> xx_discard_thresholds with selected_xx, to make it easier to use for the end user.
>
> Committed: https://crrev.com/87f83a9a27d657731ccb54025bc04ccad0da136e
> Cr-Commit-Position: refs/heads/master@{#10215}
TBR=pbos@webrtc.org,mflodman@webrtc.org,ivica@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1397363002
Cr-Commit-Position: refs/heads/master@{#10252}
The CL includes several changes:
- Adding flags describing the streams and spatial layers.
- Reorganizing the order of the flags, to make them easier to maintain.
- Adding a member .params_ to VideoQualityAnalyzer.
(instead of passing it to every member function manually)
- Updating VideoAnalyzer to support simulcast.
(select appropriate ssrc and fix timestamps which are sometimes increased by 1)
- VP9EncoderImpl already had code for automatic calculation of bitrate for each layer.
Changing to first read bitrates and resolution ratios from the flags, if specified.
If not specified, reverting to the old code are setting the values automatically.
- Changing the parameters in LayerFilteringTransport, replacing
xx_discard_thresholds with selected_xx, to make it easier to use for the end user.
Review URL: https://codereview.webrtc.org/1353263005
Cr-Commit-Position: refs/heads/master@{#10215}
Also, in Sample struct, replacing double with the original type.
It makes more sense to save the original data as truthful as possible, and then
convert it to double later if necessary (in the plot script).
Review URL: https://codereview.webrtc.org/1374233002
Cr-Commit-Position: refs/heads/master@{#10184}
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.
BUG=4173
Review URL: https://codereview.webrtc.org/1376673004
Cr-Commit-Position: refs/heads/master@{#10144}
In the middle of refactoring, I replaced the VideoCapturer with
FrameGeneratorCapturer, to reuse the code, and with that disabled the camera.
Now adding capturer_ element to VideoQualityTest and ignoring
frame_generator_capturer_ from the parent class test::CallTest.
Review URL: https://codereview.webrtc.org/1356933005
Cr-Commit-Position: refs/heads/master@{#10023}
In screensharing full stack tests, instead of using YuvFileGenerator by default
when no scrolling is used, I always used ScrollingImageFileGenerator.
That possibly slowed down the test a little bit, at least for the slowed
devices, as it unnecessarily copied few MBs per frame.
BUG=chromium:534220
Review URL: https://codereview.webrtc.org/1359783002
Cr-Commit-Position: refs/heads/master@{#10014}
Refactoring full stack, video and screenshare tests to use the same code basis
for parametrization and initialization. This patch is done on top of recently
commited full stack graphs CL https://codereview.webrtc.org/1289933003/, but
virtually no changes have been made to full_stack_plot.py nor to the VideoAnalyzer
in full stack, except moving it to video_quality_test.cc.
Also, full_stack_samples.cc (build target) was removed and replaced with
-output_filename and -duration cmdline arguments in video_loopback and
screenshare_loopback.
The important things to review:
- video_quality_test.h
Is the structure of Params good? (examples of usage can be found in
full_stack.cc, video_loopback.cc and screenshare_loopback.cc)
- video_quality_test.cc
Is the initialization correct? The case for using Analyzer and using local
renderer are different, can they be further merged?
- webrtc_tests.gypi
Reproducing the different bitrate settings the full stack and loopback tests had
was a little bit tricky. To support both simultaneously, I added BitrateConfig
to the Params struct, as well as separate start_bitrate and target_bitrate flags
for loopback tests.
Note: Side-by-side diff for video_quality_test.cc compares that file directly
with the old full_stack.cc, so changes to VideoAnalyzer are clearly visible.
Note: Recent CL I've committed added -num_temporal_layers and -sl_discard_threshold
args to loopback tests. This was removed here. Support for streams and SVC
will be added in a CL following this one.
Review URL: https://codereview.webrtc.org/1308403003
Cr-Commit-Position: refs/heads/master@{#9969}
