class to pass packet to RtcpPacketParser
This helpers make tests setup cleaner and
makes explicit expectation on number of packets passed to the transport.
Bug: webrtc:8239
Change-Id: I2d5975be59327cee440e87dbd0701b93514c9726
Reviewed-on: https://webrtc-review.googlesource.com/22460
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20911}
This tests NetEq with a stream encoded with Opus using it's internal
DTX/CNG.
Also adding a new resource file which is encoded using Opus with DTX.
Bug: webrtc:8488
Change-Id: Icfba5bc5dc7f9c9d0e637a90f4df674e8ba40358
Reviewed-on: https://webrtc-review.googlesource.com/26028
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20905}
The root cause of the flakiness is unknown, the possible issue is that the
console window running the test case is hidden or minimized. So this change
adds a SetWindow(SW_MAXIMIZE) to ensure the console window is showing.
I have run the tests against win_asan for hundreds times during the
thanksgiving. So far, no flakiness were caught.
Bug: webrtc:8568
Change-Id: Ib2c93e9bd511257213254bdaa0079c14ea50f3e4
Reviewed-on: https://webrtc-review.googlesource.com/25286
Reviewed-by: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20902}
To make desktopCapture autotest in chromium more meaningful, it's better
to creake fake capturer of each capture type. Here we can reuse the
FakeDesktopCapturer for window capture in chromium.
Bug: chromium:699201
Change-Id: Icbe134d99cbd4980bf27fe74c1c629a1469836ea
Reviewed-on: https://webrtc-review.googlesource.com/26360
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20901}
This reverts commit 2a9dbe6e7e722cf069b72ccff0051b1517706996.
Reason for revert: The fix does not help, it still fails at runtime
Original change's description:
> Adding -Wno-deprecated-declarations to declarations deprecated in iOS 10
>
> This CL will unblock the Chromium Roll while deprecated declarations
> will be removed from the WebRTC codebase.
>
> Bug: webrtc:8570
> Change-Id: I55cf78040758369ce45176cf0a00df50a87eb972
> No-Try: True
> Reviewed-on: https://webrtc-review.googlesource.com/25641
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20858}
No-Try: True
Bug: webrtc:8570
Change-Id: Ib17ed05f8f239912661281329ecab175c23491d6
Reviewed-on: https://webrtc-review.googlesource.com/25964
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20890}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
Bug: None
Change-Id: If2a98dc714d1755f07af1f70248cf41e4a9db750
Reviewed-on: https://webrtc-review.googlesource.com/23612
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20887}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
Bug: None
Change-Id: Iedebf4dc56a973306e7d7e7649525879808dc72b
Reviewed-on: https://webrtc-review.googlesource.com/23578
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20878}
The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM.
Bug: webrtc:8563, b/67926135
Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24
Reviewed-on: https://webrtc-review.googlesource.com/25621
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20877}
This test produces a consistent stream of false positive alerts, and I have been unable to make it more robust, despite several attempts. It also has never managed to catch a real regression, so I think it is better to remove it.
Bug: chromium:788318
Change-Id: I7e9731834f67af1ef2fa15a655e620bd64a4cfde
Reviewed-on: https://webrtc-review.googlesource.com/25824
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20874}
GetBandwidthObserver should be used instead as it exposes a smaller interface.
Bug: webrtc:8415
Change-Id: I29ca795657e205186d7ebd929e756038a294b5f7
Reviewed-on: https://webrtc-review.googlesource.com/23900
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20871}
The audio processing module reports the metrics 'echo return loss'
and 'echo return loss enhancement' for AEC3 via the old, soon to be
replaced, GetStatistics fuction.
Bug: webrtc:8533
Change-Id: I6b2286b5cdf8f20ebf14f82f1180f6bfb6c00c68
Reviewed-on: https://webrtc-review.googlesource.com/25642
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20869}
This new tool provides the similar functionality as the legacy tool, but
is implemented using less legacy helpers. It also replaces RTPtimeshift
and RTPchange. The most significant change versus the old RTPjitter tool
is that the new tool takes the timing data in the form of integers in a
text file (instead of the binary data file used by the old tool). This
should make it easier to create custom timing files when needed.
Bug: webrtc:2692
Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287
Reviewed-on: https://webrtc-review.googlesource.com/25700
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20868}
This CL will unblock the Chromium Roll while deprecated declarations
will be removed from the WebRTC codebase.
Bug: webrtc:8570
Change-Id: I55cf78040758369ce45176cf0a00df50a87eb972
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/25641
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20858}
This new tool provides the same functionality as the legacy tool, but it
is implemented using AudioCodingModule and AudioEncoder APIs instead of
the naked codecs.
Bug: webrtc:2692
Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc
Reviewed-on: https://webrtc-review.googlesource.com/24861
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20857}
This can be used to determine whether the bitrate of a given spatial and temporal layer has been set in the allocation, even if the value it's set to is zero.
GetBitrate still returns 0 if the queried layer does not have the bitrate set.
Bug: webrtc:8479
Change-Id: I1d982e211da9b052fcccdbf588b67da1a4550c60
Reviewed-on: https://webrtc-review.googlesource.com/17440
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Varga <erikvarga@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20852}
It will be easier to make perf results output to JSON if the PrintResult*
functions receive doubles instead of strings.
I'll make follow-up CLs for PrintResultMeanAndError and PrintResultList.
Bug: webrtc:8566
Change-Id: I198e422a7bb8cd237c6364af98d2f67f0858452e
Reviewed-on: https://webrtc-review.googlesource.com/25300
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20848}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=danilchap@webrtc.org
Bug: None
Change-Id: Ib4694d183f04d675f2ea66d39661fdffb2a984f1
Reviewed-on: https://webrtc-review.googlesource.com/23604
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20846}
The APM-QA tool produces clean-speech + noise + echo mixes with the
--additive_noise_tracks_path, --test_data_generators,
--echo_path_simulator flags. From this CL, it automatically produces
compressed Numpy annotations for the mixes. Annotations are placed in
the same folder as the mixes with name '${basename}-annotations.npz'.
TBR=alessiob@webrtc.org
NOTRY=True
Bug: webrtc:7494
Change-Id: I71941a4283594ef813de3ed65be31623bce5d7de
Reviewed-on: https://webrtc-review.googlesource.com/24960
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20844}
This type of check should instead be performed by the application/client.
If the app does not have mic permissions, construction of the AudioRecord
object will fail and the user will receive an error callback anyhow.
Bug: b/69434512
Change-Id: If1d7eff488f7c693697e048a567c17ed0c51f040
Reviewed-on: https://webrtc-review.googlesource.com/25261
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20839}
Final stage since these APIs are no longer used in Chrome.
Bug: webrtc:7306
Change-Id: Ia116671bc888daa75c4105ad1ebeb21833f5d090
Reviewed-on: https://webrtc-review.googlesource.com/25220
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20836}
The audio processing module reports the metrics 'echo return loss'
and 'echo return loss enhancement' for AEC3.
Bug: webrtc:8533
Change-Id: I166c504adf013d6cb5d6d3c9717d0622c3454bb7
Reviewed-on: https://webrtc-review.googlesource.com/24880
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20835}
Specifically, I'm moving
histogram_percentile_counter.h
mathutils.h
mod_ops.h
moving_max_counter.h
They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.
BUG=webrtc:8445
Change-Id: I78a999984a27ef935be2d7c3136475d5f209adda
Reviewed-on: https://webrtc-review.googlesource.com/20870
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20832}
Specifically, I'm moving
safe_compare.h
safe_conversions.h
safe_minmax.h
They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.
BUG=webrtc:8445
Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
There is now an 'ExternalVad' class in the AnnotationsExtractor. The
Extractor takes an extra list of these in addition to the other
VADs. The external VAD runs an external program to generate the
annotations. Annotations are loaded and saved to a compressed Numpy format.
Also made a small fix to name a mixed file in a way so that files will not
be overwritten.
Also did some minor changes to the unittests.
TBR=alessiob@webrtc.org
Bug: webrtc:7494
Change-Id: I7816b04466be16cd635ac6ceab18cd7aad5325a4
Reviewed-on: https://webrtc-review.googlesource.com/23623
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20819}
Calculation of quality metrics required writing of decoded video
to file. There were two drawbacks with that approach. First, frame
drops significantly affected metrics because comparison was done
against the last decoded frame. Second, simulcast/SVC required
writing of multiple files. This might be too much data to dump.
On-fly metrics calculation is done in frame decoded callback.
Calculation time is excluded from encoding/decoding time. If CPU
usage measurement is enabled metrics calculation is disabled since
it affects CPU usage. The results are reported in Stats::PrintSummary.
Bug: webrtc:8524
Change-Id: Id54fb21f2f95deeb93757afaf46bde7d7ae18dac
Reviewed-on: https://webrtc-review.googlesource.com/22560
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20798}
Background: After 20 consecutive DTX frames, Opus encodes the background
noise in a normal frame and then goes back to outputting DTX frames.
Currently all Opus frames are flagged as containing speech.
This CL is has two effects on outgoing Opus packets:
1. DTX frames are flagged as non-speech.
2. A non-DTX frame that follows 20 consecutive DTX frames is flagged as
non-speech.
Bug: webrtc:8088
Change-Id: Ic36cf8c9d0a34f55ed4e57858362ad91e3897dda
Reviewed-on: https://webrtc-review.googlesource.com/23760
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20794}
This new parallel GetStatistics function uses Optionals to indicate if stats are valid or not, and no longer relies on default values. It also takes an argument to indicate if receive streams are present, and if not several stats will not be set.
Bug: b/67926135
Change-Id: I175de1c65c414bea6ec9ca8b0b16f07cb2308d9f
Reviewed-on: https://webrtc-review.googlesource.com/17942
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20789}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=henrik.lundin@webrtc.org
Bug: None
Change-Id: I733a83f702fe11884d229a1713cfac952727bde8
Reviewed-on: https://webrtc-review.googlesource.com/23601
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20786}
This CL refactors the delay estimator in AEC3.
Furthermore, it adds:
1. Allow for a customized delay estimator behavior to simplify
development.
2. Exposes that behavior to clear configuration settings.
3. Adds logging of the delay range supported by the delay
estimator.
Bug: webrtc:8519
Change-Id: I1764a090519a78b021b2e7de565c52a6c02c848e
Reviewed-on: https://webrtc-review.googlesource.com/21166
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20733}
Because some hardware encoders may not preserve capture timestamp, use
rtp timestamp as a key to capture time storage and restore corrupted
capture timestamps whenever possible.
Bug: webrtc:8497
Change-Id: Ib3449313c65e864a89cd645a3f5dec99aa9cf8e4
Reviewed-on: https://webrtc-review.googlesource.com/23620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20732}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=kwiberg@webrtc.org
Bug: None
Change-Id: I055411a3e521964c81100869a197dd92f5608f1b
Reviewed-on: https://webrtc-review.googlesource.com/23619
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20728}
This makes it easier to import cleanly downstream, and makes it
a lot easier to reason about.
Bug: webrtc:6828
Change-Id: I953e129de73053f8619333fe7e318b36e3a1fffa
Reviewed-on: https://webrtc-review.googlesource.com/22722
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20727}
