This is a reland of 2d0880b56954f57548deea51dfa678b80dbf618f
To fix perf bot issue reading of perf results file was updated. Now perf
results file will be generated by each test and then returned via output to
the python script, which will get it and put into final file.
Original change's description:
> Fix collection of audio metrics from PC test framework for audio test
>
> Bug: webrtc:10138
> Change-Id: I18a8509a0cdc4ed1db6894c7540d5c0a155d6233
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144784
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28514}
Bug: webrtc:10138
Change-Id: I1347f09427736362a2d550612b48e09c06cfb1d1
No-Presubmit: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145201
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28527}
This is a trivial CL, updating rtp_generator.cc according to changes in
APIs in other places.
Bug: webrtc:10807
Change-Id: Ie85c6283f2d78dcf742979378db0b4fb0914c96c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145209
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28526}
Users of webrtc generally should be able to choose own task queue implementation.
Poison avoids accidental dependency of a low level component on the default implementation
Android and ios apis are still de-facto forced to use the default implementation.
Bug: webrtc:10284
Change-Id: I67ecf2317f43ee32b0c9e8a6e69f1e0987cf1914
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144786
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28524}
This is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770
Patch set 1 is the original CL.
Patch set 2 introduced a trivial fix. In RtcpSender::SetSSRC(), check
if either current SSRC is 0 or if the SSRC is identical to the current
one. If so, don't schedule an early report.
This prevents a regression in which audio jitter became too low(?)
Original change's description:
> Add ability to set RTCP sender ssrc at construction time
>
> Bug: webrtc:10774
> Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28506}
Bug: webrtc:10774
Change-Id: I103dfa48719aa41d6ab633cdac8b3a5c46b54843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144565
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28520}
Ensures that newly added ChannelMixer is utilized when number of channels
is larger than two in the output mixer.
Decided to land with henrik.lundin as TBR since he has reviewed all other
changes in WebRTC related to channel mixing for multi-channel cases.
All this CL does is to ensure that the new channel mixing scheme can be used
in Chrome. The old scheme is still used for mono and stereo combinations.
TBR: henrik.lundin
Bug: webrtc:10783
Change-Id: I11c02f1b4ef60e847095efbcd5e5f5faf27a5cdd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140290
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28517}
If VideoEncoderConfig::max_bitrate_bps is unset then max bitrate of
video stream is set equal to max bitrate value recommended by encoder
for given resolution via encoder capabilities (if available).
Bug: webrtc:10796
Change-Id: I7fce9afc476b794a16956e694e891faee110048e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144526
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28515}
A cloaked window is composited but not visible to the user.
When Win10 feature 'Cortana' is enabled it creates a window
that is always invisible and its z-order is top most. Because
of that the cropping capturer detects occlusion everywhere
preventing it from capturing anything.
The solution is to ignore all cloaked windows like if
::IsWindowVisible would return false.
Bug: chromium:978885
Change-Id: Id5aa8dc81dcf4979ffb30dd808fa2a553934c6e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143980
Commit-Queue: Julien Isorce <julien.isorce@chromium.org>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28510}
Ntp timestamps are used for end-to-end delay measurements and can never
go away. The naming and number of timestamp fields in VideoFrame could
change in the future, but capture time in local clock will always be
there on the receive side.
Bug: none
Change-Id: I358689cd8a44b1da8503136b3dd898b936f2d693
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144542
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28498}
This CL makes the new code path for paced sending functionally complete.
By default, the field trial WebRTC-Pacer-ReferencePackets is Enabled,
meaning that there is no behavior change unless the field trial is
forced to Disabled. This is done in tests, and can be done on the
command line for manual testing.
Bug: webrtc:10633
Change-Id: I0d66c94ef83b5847dee437a785018f09ba3f828d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144050
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28497}
Encountering GL_OUT_OF_MEMORY is relatively common and we should give
clients a chance to deal with it in a non-fatal way.
Bug: webrtc:8154
Change-Id: Ifa9ca74392f21083692b02a5144dc5632a88d34d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144561
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28495}
There is no public API to create NetworkBehaviorInterface from
BuiltInNetworkBehaviorConfig, so this CL will add direct method, that will
allow downstream projects to use BuiltInNetworkBehaviorConfig for network
emulation.
Bug: webrtc:10138
Change-Id: Iaec3ea17c12bd06b1c0ff3e5bc2b32cc1c4f62f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144628
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28494}
Add PC level audio quality test as separate file but into the same
binary as low_bandwidth_audio_test.cc
Bug: webrtc:10138
Change-Id: I3bb7df4130ab62c21a82881cd9f57b936f3cc11d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144621
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28493}
Because of webrtc:10801, we don't actually support 4 simulcast layers but 3.
Until this is fixed, we limit the value to what we can currently handle.
Bug: webrtc:8785
Change-Id: I513b7c8d4c889fa0d80c91adc1c4f874acb86fdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144625
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28487}
Two new classes are added to WebRTC from Chrome: ChannelMixer and
ChannelMixingMatrix but they are not yet utilized in the audio path for
WebRTC.
The idea is to utilize these new classes when adding support for multi-
channel encoding/decoding in WebRTC/Chrome.
Adds support for a new enumerator call webrtc::ChannelLayout and some
helper methods which maps between channel layout and number of channels.
These parts are also copied from Chrome.
Minor (cosmetic) changes are also done on the AudioFrame to prepare
for upcoming work.
Bug: webrtc:10783
Change-Id: I6cd7a13a3bc1c8bbfa19bc974c7a011d22d19197
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141674
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28482}
Implements RTCAudioSourceStats members:
- audioLevel
- totalAudioEnergy
- totalSamplesDuration
In this CL description these are collectively referred to as the audio
levels.
The audio levels are removed from sending "track" stats (in Chrome,
these are now reported as undefined instead of 0).
Background:
For sending tracks, audio levels were always reported as 0 in Chrome
(https://crbug.com/736403), while audio levels were correctly reported
for receiving tracks. This problem affected the standard getStats() but
not the legacy getStats(), blocking some people from migrating. This
was likely not a problem in native third_party/webrtc code because the
delivery of audio frames from device to send-stream uses a different
code path outside of chromium.
A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the
send-side audio levels to the RTCAudioSourceStats, while keeping the
receive-side audio levels on the "track" stats. This allows an
implementation to report the audio levels even if samples are not sent
onto the network (such as if an ICE connection has not been established
yet), reflecting some of the current implementation.
Changes:
1. Audio levels are added to RTCAudioSourceStats. Send-side audio
"track" stats are left undefined. Receive-side audio "track" stats
are not changed in this CL and continue to work.
2. Audio level computation is moved from the AudioState and
AudioTransportImpl to the AudioSendStream. This is because a) the
AudioTransportImpl::RecordedDataIsAvailable() code path is not
exercised in chromium, and b) audio levels should, per-spec, not be
calculated on a per-call basis, for which the AudioState is defined.
3. The audio level computation is now performed in
AudioSendStream::SendAudioData(), a code path used by both native
and chromium code.
4. Comments are added to document behavior of existing code, such as
AudioLevel and AudioSendStream::SendAudioData().
Note:
In this CL, just like before this CL, audio level is only calculated
after an AudioSendStream has been created. This means that before an
O/A negotiation, audio levels are unavailable.
According to spec, if we have an audio source, we should have audio
levels. An immediate solution to this would have been to calculate the
audio level at pc/rtp_sender.cc. The problem is that the
LocalAudioSinkAdapter::OnData() code path, while exercised in chromium,
is not exercised in native code. The issue of calculating audio levels
on a per-source bases rather than on a per-send stream basis is left to
https://crbug.com/webrtc/10771, an existing "media-source" bug.
This CL can be verified manually in Chrome at:
https://codepen.io/anon/pen/vqRGyq
Bug: chromium:736403, webrtc:10771
Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28480}
