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}
Since the pacer is always enabled, removing enable/disable which makes
all packet queueing succeed. Also renaming one of the ::SendPackets
::InsertPacket to avoid confusion.
BUG=webrtc:1695, webrtc:2629
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1392513002 .
Cr-Commit-Position: refs/heads/master@{#10211}
In video_sender.cc, properly read the number of temporal layers for VP9 too.
Also, some cleanup in video_loopback.cc and video_quality_test.h.
Review URL: https://codereview.webrtc.org/1351693005
Cr-Commit-Position: refs/heads/master@{#10201}
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}
When fetching a packet from the rtp packet history, cuased by a
retransmission, the transport seq extension header is enabled but the
sequence number is set to 0. A new transport seq should be assigned in
this case.
BUG=
Review URL: https://codereview.webrtc.org/1385563005
Cr-Commit-Position: refs/heads/master@{#10183}
Since padding is no longer sent on Encoded() callbacks, dummy callbacks
aren't required to generate padding. This skip-frame behavior can then
be removed to get rid of dummy callbacks though nothing was encoded. As
frames don't have to be generated for frames that don't have to be sent
we skip encoding frames that aren't intended to be sent either, reducing
CPU load.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1369923005 .
Cr-Commit-Position: refs/heads/master@{#10181}
Poller thread is currently started in the constructor, so the first call
to PollStats() may happen even before the streams have been configured.
This will blow up on RTC_DCHECK_GT(expected_bitrate_bps_, 0);
Thread should instead be started on PerformTest() call.
BUG=
Review URL: https://codereview.webrtc.org/1378303004
Cr-Commit-Position: refs/heads/master@{#10149}
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 addition to this the ramp-up tests are refactored to use a receive call instead of only a remote bitrate estimator, and to make use of BaseTest.
BUG=webrtc:4836
Review URL: https://codereview.webrtc.org/1368943002
Cr-Commit-Position: refs/heads/master@{#10087}
The BWE expects arrival timestamps in ms, while the audio path delivered
them in us, causing the BWE to break down under the combined audio/video
BWE experiment. This was introduced in r9892 (68786d2040).
BUG=webrtc:4758
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1360913004 .
Cr-Commit-Position: refs/heads/master@{#10032}
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}
While refactoring, I incorrectly set the target bitrate for one of the tests to
500k instead of 2000k.
This does not fix all the perf regressions.
BUG=534220
Review URL: https://codereview.webrtc.org/1356123002
Cr-Commit-Position: refs/heads/master@{#10008}
Moved the build target into a section in the gyp file that is conditional on 'include_test==1', as well as on 'enable_protobuf==1'.
Original review: https://codereview.webrtc.org/1297653002/
Reverted in be4959535a39262e1508cc4223b78b8db677cb94
BUG=webrtc:4741
TBR=kjellander@webrtc.org,stefan@webrtc.org,henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1353083003 .
Cr-Commit-Position: refs/heads/master@{#9990}
Reason for revert:
Breaks Chromium WebRTC FYI bots.
Updating projects from gyp files...
gyp: /b/build/slave/linux/build/src/third_party/gflags/gflags.gyp not found (cwd: /b/build/slave/linux/build)
Error: Command '/usr/bin/python src/build/gyp_chromium' returned non-zero exit status 1 in /b/build/slave/linux/build
Original issue's description:
> Tool to convert RtcEventLog files to RtpDump format.
>
> This is a small utility that reads RtcEventLog files, and converts the RTP headers within it to RtpDump format. All other types of events are ignored. Three command-line flags are supported, --audio-only, --video-only and --data-only. When one of these flags is supplied, only RTP packets that match the requested type are converted.
>
> BUG=webrtc:4741
> R=henrik.lundin@webrtc.org, kjellander@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/35624c2c3686a2ad40daffe073aa78507b0ef88e
> Cr-Commit-Position: refs/heads/master@{#9980}
TBR=henrik.lundin@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,kjellander@webrtc.org,kjellander@google.com,ivoc@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1345983009
Cr-Commit-Position: refs/heads/master@{#9987}
To make this possible padding only packets will have the same timestamp
as the previously sent media packet, as long as RTX is not enabled. This
has the side effect that if we send only padding for a long time without
sending media, a receive-side jitter buffer could potentially overflow.
In practice this shouldn't be an issue, partly because RTX is recommended and
used by default, but also because padding typically is terminated before being
received by a client. It is also not an issue for bandwidth estimation as long
as abs-send-time is used instead of toffset.
BUG=chromium:425925
R=mflodman@webrtc.org, sprang@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1327933003 .
Cr-Commit-Position: refs/heads/master@{#9984}
This is a small utility that reads RtcEventLog files, and converts the RTP headers within it to RtpDump format. All other types of events are ignored. Three command-line flags are supported, --audio-only, --video-only and --data-only. When one of these flags is supplied, only RTP packets that match the requested type are converted.
BUG=webrtc:4741
R=henrik.lundin@webrtc.org, kjellander@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/1297653002 .
Cr-Commit-Position: refs/heads/master@{#9980}
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.
IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately
BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1335353005 .
Cr-Commit-Position: refs/heads/master@{#9978}
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}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
Helps differentiate between different instances when debugging.
Review URL: https://codereview.webrtc.org/1337003003
Cr-Commit-Position: refs/heads/master@{#9927}
An option was added to voe_cmd_test to make a RtcEventLog dump.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1267683002
Cr-Commit-Position: refs/heads/master@{#9901}
