* Both timestamps must be unwrapped before comparing
* rtp timestamp delta must be subtracted before unwrapping
BUG=webrtc:5637, webrtc:5537
Review URL: https://codereview.webrtc.org/1774123003
Cr-Commit-Position: refs/heads/master@{#11926}
Removes addition of at least one zero sample in webrtc_perf_tests that
can skew stats differently depending on how often these stats are
updated. Unclear if this skewing is different between now and before.
BUG=chromium:585071, chromium:586216
R=sprang@google.com, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1727583003 .
Cr-Commit-Position: refs/heads/master@{#11720}
Permits measuring encoding time even when performed on another thread,
typically for hardware encoding, instead of assuming that encoding is
blocking the calling thread.
Permitted encoding time is increased for hardware encoders since they
can be timed to keep 30fps, for instance, without indicating overload.
Merges EncodingTimeObserver into EncodedFrameObserver to have one post-encode
callback.
BUG=webrtc:5042, webrtc:5132
R=asapersson@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1569853002 .
Cr-Commit-Position: refs/heads/master@{#11499}
Adds negotiation of rtx codecs for red and vp9. To keep backwards
compatibility with older Chrome versions, this change includes two
hacks:
1. Red packets will be retransmitted over the rtx codec associated with
vp8 if no rtx codec is associated with red. This is how Chrome does
it today and ensures that we still can send red over rtx to older
versions.
2. If rtx packets associated with the media codec (vp8/vp9 etc) are
received and red has been negotiated, we will assume that the sender
incorrectly has packetized red inside the rtx header associated with
media. We will therefore restore it with the red payload type
instead, which ensures that we can still receive rtx associated with
red from old versions.
Offering multiple rtx codecs to older versions should not be a problem
since old versions themselves only try to negotiate rtx for vp8.
R=pbos@webrtc.orgTBR=mflodman@webrtc.org
BUG=webrtc:4024
TEST=Verified by running apprtc and emulating packet loss between Chrome with and without the patch.
Review URL: https://codereview.webrtc.org/1649493004 .
Cr-Commit-Position: refs/heads/master@{#11472}
We have seen an instance of flakiness of the perf tests where it looked
like timestamp wraparound could be an issue. Better safe...
BUG=
Review URL: https://codereview.webrtc.org/1645463002
Cr-Commit-Position: refs/heads/master@{#11440}
It works on all platforms except Android and iOS (FFmpeg limitation).
Implemented behind compile time flags, off by default.
The plan is to have it enabled in Chrome (see bug), but not in Chromium/webrtc by default.
Flags to turn it on:
- rtc_use_h264 = true
- ffmpeg_branding = "Chrome" (or other brand that includes H.264 decoder)
Tests using H264:
- video_loopback --codec=H264
- screenshare_loopback --codec=H264
- video_engine_tests (EndToEndTest.SendsAndReceivesH264)
NOTRY=True
BUG=500605, 468365
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424
Review URL: https://codereview.webrtc.org/1306813009
Cr-Commit-Position: refs/heads/master@{#11390}
Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers.
Audio streams are using a fake audio device with file input.
The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code.
R=pbos@webrtc.orgTBR=kjellander@webrtc.org
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1542653002 .
Cr-Commit-Position: refs/heads/master@{#11171}
Also move (and clean up includes) rampup_tests.* to webrtc/call in preparation for combined audio/video ramp-up tests.
No functional changes.
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1537273003
Cr-Commit-Position: refs/heads/master@{#11101}
Makes use of rtc::Event which is simpler and can be used without
allocating additional objects on the heap.
Does not modify test/channel_transport/.
BUG=
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1487893004 .
Cr-Commit-Position: refs/heads/master@{#10968}
* Move PlatformThread to rtc::.
* Remove ::CreateThread factory method.
* Make non-scoped_ptr from a lot of invocations.
* Make Start/Stop void.
* Remove rtc::Thread priorities, which were unused and would collide.
* Add ::IsRunning() to PlatformThread.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1476453002 .
Cr-Commit-Position: refs/heads/master@{#10812}
Also removes all virtual methods. Permits using a thread from
rtc_base_approved (namely event tracing).
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1469013002
Cr-Commit-Position: refs/heads/master@{#10760}
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}
