Wav file capturer won't repeat file or produce silence after file end and
WebRTC pipeline will crash in such case. In future we need to make it
possible to continue audio after file was ended to behalf in the same
way as video files capturer.
Bug: webrtc:10138
Change-Id: I35f5bd33790cd430a56002a44af0abb894a96d29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132795
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27609}
The former became redundant and didn't guarantee
numerical stability for variance computation.
Bug: webrtc:10412
Change-Id: Idc291abe9add41bde9e7734f179e5d6c65f2630b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132460
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27605}
Without this |ready_runners_| might still have entries left if the
yield call comes from another task queue (only done in testing).
Bug: webrtc:10365
Change-Id: I704249e00bf5e75e1f58fdda1809b955de20c304
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132713
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27596}
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}
We need to keep the lock until we have finished using the task runner
returned by GetNextReadyRunner to ensure that we don't remove it from
another thread. Additionally, we must get only one runner at a time in
case the first runner removes the second runner.
Bug: webrtc:10538
Change-Id: Idbd5610b67666238545b3a321fb79f7e86fcac56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132342
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27584}
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}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
This CL just moves code around to prepare for an upcoming
CL where more stats collection is added to scenario tests.
Bug: webrtc:10365
Change-Id: I8a960e44fd11fc36047677c4d8dfc0af96aacb22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132002
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27519}
Fuzzer test was configured in a wrong way in
https://webrtc-review.googlesource.com/c/src/+/129768
This fixes it (verified locally on libfuzzer MSAN and ASAN).
Bug: webrtc:8649, chromium:950813
Change-Id: I52647bb12c4c412252cdcd931c9e210606bdb12d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132009
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27516}
This CL does two things:
* It ensures that video quality analysis is always finished when a
Scenario test is stopped. This ensures consistency between different
task queue implementations.
* It disables one real time test that is too heavy to run on IOS
ARM64 debug builds.
Bug: webrtc:10515
Change-Id: I34b59ecde6f2b68c399734a43ecdbc7223725b17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131388
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titarenko <artit@google.com>
Cr-Commit-Position: refs/heads/master@{#27495}
See https://webrtc-review.googlesource.com/c/src/+/121764 for the
overall vision.
This CL adds a multistream Opus decoder. It's a new code-path to not
interfere with the standard Opus decoder. We introduce new SDP syntax,
which uses terminology of RFC 7845. We also set up the decoder side to
parse it. The encoder part will come in a later CL.
E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"
Bug: webrtc:8649
Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27493}
rtcp_receiver_fuzzer was running over inputs of unreasonable
length, leading to timeouts. RTCP typically runs over UDP.
This CL limits the inputs to a bit over the max UDP payload length.
Bug: chromium:948469
Change-Id: I669a5b24c265bb3b6da2503da109efed32c25182
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131393
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27482}
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}
This is to avoid inconsistent/flaky behavior on mobile bots.
Bug: webrtc:10365
Change-Id: I52ab4f9ef92b10329c1eac502adfcf2886058114
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131329
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27465}
Refactoring of quality measurement code, basing frame matching on
frame thumb likeness. This way the code is robust against variations
in timing and frame drops.
Bug: webrtc:9510
Change-Id: Ief7266e01f39ca621a529c0da736e5ed1df8560a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/124401
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27415}
The has some benefits:
* We no longer need locks to protect the emulated network node state.
* We only process when there are packets in flight.
* It makes Scenario more similar to network emulation manager.
Bug: webrtc:10365
Change-Id: I8bd1ad1edfb54b047e8109dabd9846ae451cef17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127548
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27393}
Also adding sync group for video streams.
Bug: webrtc:10365
Change-Id: I9ef92de756f06bbbcd7b67524bbf51fe1365fa85
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130508
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27390}
to decouple it from other optional parameters
and with plan to make it mandatory
Bug: webrtc:10284
Change-Id: I71c1d3d9eaf09d00b99b0bc4c811ab173ea5f01f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130473
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27385}
