This allows forcing a minimum frame rate if the producer doesn’t update
the SurfaceTexture often.
This needs to be done in SurfaceTextureHelper to keep the
synchronization of the texture access consistent.
Bug: b/149383039
Change-Id: I0e3c82dd51d486b931bd8dda0fd9d5cdb1a90901
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177001
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@google.com>
Cr-Commit-Position: refs/heads/master@{#31504}
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 adds priority to the API configuration of datachannels,
and passes the value in the OPEN message.
It does not yet influence SCTP prioritization of messages.
Bug: chromium:1083227
Change-Id: I46ddd1eefa0e3d07c959383788b9e80fcbfa38d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175107
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31287}
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 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 CL introduced 2 new macros that affect the WebRTC OBJC API symbols:
- RTC_OBJC_TYPE_PREFIX:
Macro used to prepend a prefix to the API types that are exported with
RTC_OBJC_EXPORT.
Clients can patch the definition of this macro locally and build
WebRTC.framework with their own prefix in case symbol clashing is a
problem.
This macro must only be defined by changing the value in
sdk/objc/base/RTCMacros.h and not on via compiler flag to ensure
it has a unique value.
- RCT_OBJC_TYPE:
Macro used internally to reference API types. Declaring an API type
without using this macro will not include the declared type in the
set of types that will be affected by the configurable
RTC_OBJC_TYPE_PREFIX.
Manual changes:
https://webrtc-review.googlesource.com/c/src/+/173781/5..10
The auto-generated changes in PS#5 have been done with:
https://webrtc-review.googlesource.com/c/src/+/174061.
Bug: None
Change-Id: I0d54ca94db764fb3b6cb4365873f79e14cd879b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31153}
Currently, ScopedJavaGlobalRef can only be set at creation and never
changed. This CL makes it possible to re-set these.
Bug: b/153389044
Change-Id: I6be92dae83a9f5f3d87aa44dde226b874f4ca0a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174041
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31145}
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}
We have observed an internal deadlock in libGLESv2_adreno where one
thread is in eglCreateContext and another thread in glUseProgram. We
have observed similar deadlocks before and started to synchronize all
access to the offending GL functions. Calls to eglCreateContext are
already synchronized, and this CL synchronizes calls to glUseProgram as
well.
Bug: b/153513005
Change-Id: I576e564aab44c9e429f2b1407105ed72942c309e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173742
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31118}
We have seen crashes originating from derefencing nullptrs in this code,
for unknown reasons. This CL adds null checks to protect against this.
The stacktraces will be missing or truncated when this happens.
Bug: b/147338449
Change-Id: Ieb006f0f8dec4f9621e4df2e2c1a9641f086df86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173593
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31079}
network_priority was already exposed, but without the ability to set
enable_dscp, it wasn't actually doing anything.
Bug: webrtc:5658
Change-Id: I092bc3dd46e3e7be363313203428bccfccccf3c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171641
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30951}
This reverts commit 86e0ea5711cfef95960ffcc8b6d918c67576e5c9.
Reason for revert: The reasons for removing bitratePriority are unclear. Aside from the fact that you can't yet use it for the relative bitrate of simulcast streams, only the relative bitrate of entire tracks, it's working as intended. It differs from the standard, but only in that it's more flexible; the web standard only allows values of 0.5, 1.0, 2.0, and 4.0 while for the native API we allow any ratio.
Original change's description:
> Remove bitratePriority from the Obj-C RTCRtpEncodingParameters wrapper.
>
> This was added in CL 135122, but the bitratePriority parameter is not
> standard and not implemented in a way users would expect. So it should
> actually not be exposed in the Obj-C SDK.
>
> Bug: webrtc:10438
> Change-Id: I801ce940a32701d2703e951ef2b601c606aa2111
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135287
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27861}
TBR=andersc@webrtc.org,kthelgason@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10438
Change-Id: Ibc16b6054a1583de43a868d98683ea114bd89435
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171140
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30863}
Removes support for parsing and serializing
a=ssrc:1 mslabel:stream
a=ssrc:1 label:track
which have been superceeded by
a=ssrc:1 msid:stream track
a long time ago.
Bug: webrtc:7110
Change-Id: I3aca47728098b6e7e049b82ed34c59426d411c41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168244
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30801}
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}
It can only be one of four possible values, so it never made sense
for it to be a double. Other than the fact that its neighbor
bitrate_priority is a double, and they're both defined as the same enum
in the web spec. However, while bitrate_priority being a double
offers more flexibility than the web spec, network_priority being a
double is only confusing.
Bug: webrtc:5658
Change-Id: I0784c116f3260c4b3a8b99a3cd85c8d66017e46f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168840
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30685}
