Previously a histogram was added to track the requested buffer size,
this CL adds a histogram for the actually used buffer size.
Bug: b/157429867
Change-Id: I04016760982a4c43b8ba8f0e095fe1171b705258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176227
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31385}
The Android native audio code asks the OS to provide an appropriate
buffer size for real-time audio playout. We should add logging for this
value so we can see what values are used in practice.
Bug: b/157429867
Change-Id: I111a74faefc0e77b5c98921804d6625cba1b84af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176126
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreasson <henrika@chromium.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31368}
This is a reland of 1b8ef63876ebfa55a51c8ca9b1d8206bf8233e01. It was
previously reverted (https://webrtc-review.googlesource.com/c/src/+/175008)
but the revert was found to be unnecessary.
Original change's description:
> Add an optional override for AudioRecord device
>
> This is important when we have multiple named devices connected over
> USB (eg. "Webcam", "Microphone", "Headset") and there is some way to
> choose a specific input device to route from.
>
> Bug: b/154440591
> Change-Id: I8dc1801a5e4db7f7bb439e855d43897c1f7d8bc4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173748
> Commit-Queue: Robin Lee <rgl@google.com>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31130}
TBR=henrika@webrtc.org,sakal@webrtc.org,rgl@google.com
Bug: b/154440591, b/155256727
Change-Id: Ic9bf8305c85552a0dc0d2cde6190988423e7fc70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175084
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31255}
This reverts commit 1b8ef63876ebfa55a51c8ca9b1d8206bf8233e01.
Reason for revert: Breaks downstream projects. b/155256727
Original change's description:
> Add an optional override for AudioRecord device
>
> This is important when we have multiple named devices connected over
> USB (eg. "Webcam", "Microphone", "Headset") and there is some way to
> choose a specific input device to route from.
>
> Bug: b/154440591
> Change-Id: I8dc1801a5e4db7f7bb439e855d43897c1f7d8bc4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173748
> Commit-Queue: Robin Lee <rgl@google.com>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31130}
TBR=henrika@webrtc.org,sakal@webrtc.org,rgl@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: b/154440591, b/155256727
Change-Id: I6836676096d47d9da5702a40b9d127569ad50dda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175008
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31238}
This field trial will be used to rollout the cellular types added
in https://webrtc-review.googlesource.com/c/src/+/174500 in
a controlled fashion.
Bug: webrtc:11473
Change-Id: I371d13d6935f6e0273a023657ce1b11b32bef346
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174831
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31224}
This patch adds detection of 5G to andoird_network_monitor
using the TelephonyManager.NETWORK_TYPE_NR.
It also adds
- TelephonyManager.NETWORK_TYPE_GSM as 2G
- TelephonyManager.NETWORK_TYPE_TD_SCDMA as 3G
- TelephonyManager.NETWORK_TYPE_IWLAN as 4G
note: AdapterTypeFromNetworkType still return rtc::ADAPTER_TYPE_CELLULAR
for all cellular connections (changing that is a next step).
Bug: webrtc:11473
Change-Id: If2e681e10b24f46ea0071db0cdba758a8c4e7ee2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174500
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31171}
This is important when we have multiple named devices connected over
USB (eg. "Webcam", "Microphone", "Headset") and there is some way to
choose a specific input device to route from.
Bug: b/154440591
Change-Id: I8dc1801a5e4db7f7bb439e855d43897c1f7d8bc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173748
Commit-Queue: Robin Lee <rgl@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31130}
to avoid collission and confusion with VideoCodeType based on
c++ enum with the same name.
Bug: b/148146536
Change-Id: I049cce21d59f454c7ce507fdfc3a85d168f96223
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170048
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30728}
The degradation preference is now based on the content hint of the track
if it's unspecified.
Bug: webrtc:11164
Change-Id: Iaa0dbf1c1bf68a46fc5131e534d423c30c5439c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161233
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30691}
This makes it safe to deliver frames to the sink from VideoProcessor
even after setSink has been called with null reference without danger
of use after free.
Bug: b/148063550
Change-Id: Ib78f75ac49fc6117f744c55da1a4e671bbdcdf22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168160
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30455}
This was an ICE configuration experiment added a couple years ago that did not end up being used.
Bug: webrtc:11316
Change-Id: Iafb7e1c4f7b4598815f045808dbf6e470172f119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167680
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30395}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
Since rtc::Thread is the only class inheriting from rtc::MessageQueue
and most members of MessageQueue are public or protected the split is
not adding much value. In preparation for future cleanup, this cl merges
the two classes.
Bug: webrtc:9883
Change-Id: Ia0efb4349f66f653aa34fa4d244998f187e3ce36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165340
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30235}
This is part of a CL series merging rtc::MessageQueue into rtc::Thread.
Bug: webrtc:9883
Change-Id: I4a1bcd44c9523b6402b3f05b50597bdc2e6615e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165345
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30216}
The current camera switch API sequentially cycles through each
camera name for each method invocation. This policy provides
reasonable behavior for devices with 2 or 3 cameras, but
presents challenges with devices that contain several cameras.
For example in a scenario where the current camera is oriented
on the same side as the next camera name, a developer would need to
call switchCamera multiple times to capture from a camera oriented on
a different side of the device.
This commit allows a developer to specify a camera name when switching
cameras. This flexibility allows developers to have more control over
which device they switch to in cases where a device contains several cameras.
Bug: webrtc:11261
Change-Id: I93d46d70b2c7cf735a411a4ef4f33e926bf3a5ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165040
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30199}
This wires the current degradation preference in the SDK, it will later
be nullable in a follow up change once the native API supports it.
Bug: webrtc:11164
Change-Id: I8324e6e0af996dfddfa07e3aff4ba242d9533388
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161321
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30170}
Both cameraThreadHandler and surfaceHelper shouldn't be null.
Bug: None
Change-Id: I3c239c4275c53b836bbc2e9d6af71bf2b1b65387
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161480
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30047}
If the task to call OnEncodedImage is posted to the encoder task queue
just after VideoStreamEncoder::Stop post the task to release the
encoder, the destruction sequence of java HardwareVideoEncoder
deadlocks in outputBuffersBusyCount.waitForZero();
Encoders are generally allowed to call OnEncodedImage on any internal
encoder thread, so posting to the encoder task queue seems unnecessary.
Bug: webrtc:9378
Change-Id: Iee14f151d9efdc5ab348f9c86069fdb762e6a0dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161447
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30035}
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.
Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
Some ErrorProne warnings have been enabled by [1], that broke the
Chromium Roll into WebRTC, this CL should have taken care of all the
problems.
[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1935889
Bug: None
Change-Id: I2670e948c320984a122fdb774b891c98e05f582e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160862
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29933}
Implementers of Java wrappers for native encoders need to have the same
implementation of all the unsupported methods, as mentioned in the
documentation of VideoEncoder.createNativeVideoEncoder (and its decoder
equivalent).
This simplifies implementation of such encoders/decoders, and also make sure
they don’t override unsupported methods, as they are guaranteed not to be
called.
Bug: None
Change-Id: Iaa8499eda1b52cc14b04622bea2766cd09ba43e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160186
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@google.com>
Cr-Commit-Position: refs/heads/master@{#29866}
This CL effectively expands the zone of influence of
https://webrtc-review.googlesource.com/64160,
forcing 16-byte stack alignment of generated JNI methods
for the Android x86 platform.
Bug: webrtc:9085
Change-Id: Idc40c00ea3fb52dbbbeac7b58ceda2a9a44733d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159928
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29858}
Those are preventive annotations to prepare for incoming android update
(coming with Chromium roll).
Currently the roll is blocked partly because errorprone complains!
Bug: webrtc:11095, chromium:1003532
Change-Id: If4e2879a522e895ce7fb1f2a9ad36d06f98f2a61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160002
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29830}
