The transport time metric was accidentially changed by the CL
https://webrtc-review.googlesource.com/c/src/+/153660
This CL restore the transport time metric to how it has been
measured before, that is, time from encoder output to decoder input.
Bug: webrtc:10975
Change-Id: I66f022f26976451d28c0374b22849f14f9c02378
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153886
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29259}
Add ability to export internal state of SamplesStatsCounter to be able
then to plot that data.
Bug: webrtc:10138
Change-Id: I5aae5b7dea2989e9f82820933a9ab6f21db17556
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152542
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29173}
Refactor it one more time to partly roll back previous change and unify
approach between capturer and renderer. Now we will be able to add single
screen shower listener to display video during the test on the screen.
Bug: webrtc:10138
Change-Id: Ib19117b0943e7c6dfc14630faca1f0e4ee2d038f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151649
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29093}
Separate renderer part into steps and make it easier to add more steps
as separate interceptors.
Bug: webrtc:10138
Change-Id: I667fc85d0da4fb59090e69caa4c32bd4afc3bd05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151645
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29082}
Add frames_in_flight metric into PC framework to catch frames that were
captured but weren't delivered to the other side. Right now they won't
be reported as dropped, because it's unclear were they dropped or will
they be delivered. So the new metric is introduced. The lower value is
better for it.
Bug: webrtc:10138
Change-Id: Ide26b362a6b862bd961793cb53293becd92cfaa8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148599
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28834}
This reverts commit d978cb43c238ca24b2320acd7b656f446b906101.
Reason for revert: It breaks perf tests: https://ci.chromium.org/p/webrtc/builders/perf/Perf%20Android32%20(L%20Nexus4)/1561
Original change's description:
> Record audio/video bytes sent in analyzer stream stats.
>
> For each SSRC report, record the number of bytes sent for that stream
> and expose them in analyzer stats. These numbers can be used to
> determine useful metrics such as total media throughput (by adding the
> bytes sent for all streams) and overhead (by subtracting that amount
> from the total bytes sent to the network).
>
> Bug: webrtc:9719
> Change-Id: I977bbd40acdd0a1ec64763ddd55a642b9a50f309
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146240
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28637}
TBR=mbonadei@webrtc.org,mellem@webrtc.org,titovartem@webrtc.org
Change-Id: I3e46307dd6ef121b9377b93fc8d9fa788245ea5f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146605
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28646}
For each SSRC report, record the number of bytes sent for that stream
and expose them in analyzer stats. These numbers can be used to
determine useful metrics such as total media throughput (by adding the
bytes sent for all streams) and overhead (by subtracting that amount
from the total bytes sent to the network).
Bug: webrtc:9719
Change-Id: I977bbd40acdd0a1ec64763ddd55a642b9a50f309
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146240
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28637}
SVC support is limited:
During SVC testing there is no SFU, so framework will try to emulate SFU
behavior in regular p2p call. Because of it there are such limitations:
* if |target_spatial_index| is not equal to the highest spatial layer
then no packet/frame drops are allowed.
If there will be any drops, that will affect requested layer, then
WebRTC SVC implementation will continue decoding only the highest
available layer and won't restore lower layers, so analyzer won't
receive required data which will cause wrong results or test failures.
Bug: webrtc:10138
Change-Id: I079566260ca9f1815935bce365d1bca10766663a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144882
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28612}
The purpose of this interface is to allow VideoEncoder to override
the bandwidth allocation set by FecController in RtpVideoSender.
This CL defines the interface and sends it down to VideoSender.
Two upcoming CLs will:
1. Make LibvpxVp8Encoder pass it on to the (injectable)
FrameBufferController, where it might be put to good use.
2. Modify RtpVideoSender to respond to the message sent to it
via this API.
TBR=kwiberg@webrtc.org
Bug: webrtc:10769
Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28416}
This CL replaces all uses of Timestamp::us(Clock::TimeInMicroseconds())
with Clock::CurrentTime() which should be a no-op apart from slight
changes in checks.
Additionally instances of Timestamp::ms(Clock::TimeInMilliseconds()) in
test code is replaced. This slightly changes the behavior since the
timestamp will get increased resolution.
Timestamp::ms(Clock::TimeInMilliseconds()) in non-test code is untouched
to avoid changing behavior of production code.
Bug: webrtc:9883
Change-Id: I8047f4cb2ca735f44f11d32f9367aa3eb376ec04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142803
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28321}
Add support of negotiating simulcast offer/answer. Also fix some minor
issues around to make it finally work.
Bug: webrtc:10138
Change-Id: I382f5df04ca6ac04d8ed1e030e7b2ae5706dd10c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137425
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28274}
Report time_between_freezes as test duration when there were no freezes
in the call.
Bug: webrtc:10138
Change-Id: I3d99be4b714f1b1d13e7b8b7055b368a20859490
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141665
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28248}
This is a reland of 11dfff0878c949f2e19d95a0ddc209cdad94b3b4
Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org
Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
This reverts commit 11dfff0878c949f2e19d95a0ddc209cdad94b3b4.
Reason for revert: Downstream import failure.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org
Change-Id: I7f833055c67f1f879b01dd8c156ba7b8840e8747
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141411
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28225}
After this CL lands, an announcement will be made to
discuss-webrtc about the deprecation of one version
of InitEncode().
Bug: webrtc:10720
Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28224}
This CL removes the usage of absl::flat_hash_map because it transitively
depends on CCTZ which fails to link with lld-link after the switch to
libc++.
Since std::map doesn't support heterogeneous lookup until C++14, this
CL also stops using absl::string_view and switches to
`const std::string&`.
Bug: webrtc:10605
Change-Id: I4fc93969c6fc0cc7e7e62b4d2f801bdd27cff0f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135566
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27877}
Before if there is no video in PC quality test video quaity analyzer
failed on RTC_CHECK becuase of empty counter. Now it will report no
metrics and print 0 in debug logging.
Bug: webrtc:10138
Change-Id: If6656a613465c522cac1d4b2e4dd455e409229ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133886
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27731}
This is a reland of 7ac0d5f348f0b956089c4ed65c46e65bac125508
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,perkj@webrtc.org
Bug: webrtc:10481
Change-Id: I2978d5c527a18e885b7845c4e53a2424e8ad5b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132551
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27593}
This reverts commit 7ac0d5f348f0b956089c4ed65c46e65bac125508.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org
Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.
Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
The DefaultAudioQualityAnalyzer will read stats reports (temporarily
using the old PeerConnectionInterface::GetStats) and for each audio
stream it will collect some NetEq related stats.
When DefaultAudioQualityAnalyzer::Stop is invoked by the framework,
it will report the following metrics:
- expand_rate
- accelerate_rate
- preemptive_rate
- speech_expand_rate
- preferred_buffer_size_ms
Bug: webrtc:10138
Change-Id: Ie493456fcb9ed86455b12dabdab98a317387ef46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125980
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27474}
