Make the argument speech_probability non-optional in
InputVolumeController::Process() and
MonoInputVolumeController::Process().
Additional clean-up: Remove the flag enabled in the
config. Add unit tests for MonoInputVolumeController.
Bug: webrtc:7494
Change-Id: Ie28af77dc628bf71d09ce1ff033d39031f77a21e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283700
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38710}
and set the underlying type to unknown for mac-based detection.
BUG=webrtc:13097,webrtc:13172
Co-authored by: amir.pirogovsky@microsoft.com
Change-Id: I177214f9eb5c2990faa65262f77b3d0ce9d5eab7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284500
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38709}
The expected behavior is to have something similar than python:
https://docs.python.org/dev/library/argparse.html#dest:
"Any internal - characters will be converted to _ characters to make sure the string is a valid attribute name".
This allows to catch chromium arguments like 'isolated-script-test-output' that previously needed some preprocessing done for example in flags_compatibility.py.
This CL also fixes a fuchsia specific issue where the test runner needs a 'isolated-script-test-output' argument but then pass the argument to WebRTC that expects a 'isolated_script_test_output' argument. Thus calling flags_compatibility before the test_runner fails and there is not much room to change the argument in between the test runner and the test.
Change-Id: I48a591743fa50484a0ec584a3f9e97d9e0fd25ef
Bug: webrtc:14694
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284541
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38707}
A follow up cl will be created to better handle nullopt frame length range in AudioSendStream.
Note that maxptime is still not used for setting the frame length (only ptime is).
Bug: chromium:1109337
Change-Id: Id21fd8c76a6c4a0c85719a955116f8d16001a3d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284501
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38702}
VideoSendStreamImpl::Start and VideoSendStream::Start are not used by PeerConnections, only StartPerRtpStream.
Therefore this cl:
- Change implementation of VideoSendStream::Start to use VideoSendStream::StartPerRtpStream. VideoSendstream::Start is kept for convenience.
- Remove VideoSendStreamImpl::Start() since it was only used by tests that use call and is confusing.
- RtpVideoSender::SetActive is removed/changed to RtpVideoSender::Stop(). For normal operations RtpVideoSender::SetActiveModules is used.
Bug: none
Change-Id: I43b153250b07c02fe63c84e3c4cec18d4ec0d47a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283660
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38698}
Created probes are currently timed out after 5s. But to be safe, also limit the number of pending probes to 5.
Bug: webrtc:14392, b/259541308
Change-Id: Ibf630704ffe14cb165ab849b881cf75857376f82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284080
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38697}
when volume emulation is used or when neither an input volume
controller nor volume emulation are used.
This CL adds 3 tests, 2 of which currently fail because APM
behaves in an undesired way. In [1] the behavior is fixed and
the tests are enabled.
A DCHECK in `AudioProcessingImpl::set_stream_analog_level` has
been removed since a more robust behavior can be obtained - namely,
that expected in the disabled unit tests added in this CL.
[1] https://webrtc-review.googlesource.com/c/src/+/281185
Bug: webrtc:14581
Change-Id: I29d2c000cd1baf90606487afd9a4042e6f487834
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281184
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38696}
Make speech probability threshold configurable by replacing
kSpeechProbabilitySilenceThreshold with speech_probability_threshold in
InputVolumeController::Config.
Make the processing more robust against outliers in speech probability
estimaton by computing an aggregate speech activity over a speech
segment. In MonoInputVolumeController::Process(), use the passed
non-empty speech probabilities to compute the speech activity over the
speech segment and only allow updates for segments with a high enough
ratio of speech frames. Pass RMS error and speech probability for every
frame in Process(): If rms_error_dbfs is empty, volume updates are not
allowed; if speech_probability is empty, the frame counts as a non-
speech frame.
Remove startup_min_volume from the config since it's no longer used
after https://webrtc-review.googlesource.com/c/src/+/282821.
Bug: webrtc:7494
Change-Id: I0ab81b03371496315348f552133aa9909bd36f26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283523
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38685}
1. Use ComponentContext::Create instead of
ComponentContext::CreateAndServeOutgoingDirectory. We're not
actually serving an outgoing directory here, and trying to causes
conflicts when this code is linked into a Fuchsia component.
2. Mark the whole screen as having been updated on each frame. Some
codecs were assuming that nothing on the screen was changing, and
so only the first frame would be shared.
Change-Id: Icb02a2cc097947b85cceddec49291e666257ed81
Bug: webrtc:14681
Bug: webrtc:14682
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283920
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Sarah Pham <smpham@google.com>
Commit-Queue: Hunter Freyer <hjfreyer@google.com>
Cr-Commit-Position: refs/heads/main@{#38682}
iSAC has been removed, the tests now use Opus which requires min/max
bitrate to be set.
Bug: webrtc:14450
Change-Id: I872764b1ebb9115e314f146749fe710a7665ad62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284060
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38680}
When best candidate estimate increases above the delay based estimate, the state should be DelayBasedEstimate because the final esimate is bounded by delay based bwe anyway.
Bug: webrtc:12707
Change-Id: I0bcae684b33e5f1e9a7c57cb32c431b4eb58fd35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283802
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38677}
Cache target bit- and framerate in a frame_num -> rates map and fetch
the rates accociated with the current frame when needed. This solves
the issue when wrong target rates may be used due to frames buffering
in encoder.
Bug: b/254447893
Change-Id: I369c8d8e71234c957dc2362b055061d12cec818f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283841
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38673}
This is a reland of commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1
Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}
Bug: webrtc:14450
Change-Id: Ia22c4d7724b6022238235fede93e36e570a49376
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283843
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38665}
We have seen a few instances in a down-stream project where the circuit
breaker is still triggering and causing issues.
This CL makes the threshold configurable and adds more debug logging to
try and get to the bottom of this rarely occuring bug.
Bug: webrtc:11340, b/258509536
Change-Id: I92674d446b926ad66538ff9c8be2a32a3d95b057
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283762
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38664}
This CL adds more explicit tests for unsupported sample rates in the WebRTC audio processing module (APM). Rates are restricted to the range [8000, 384000] Hz. Rates outside this range are handled as best as possible, depending on the format.
Tested: bitexact on a large number of aecdumps
Bug: chromium:1332484, chromium:1334991
Change-Id: I9639d03dc837e1fdff64d1f9d1fff0edc0fb299f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276920
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38663}
