Currently media transport can't log events to event log, but it should (things like bitrate estimates, goog cc logging, etc). This change make RtcEventLog available inside media transport.
Bug: webrtc:9719
Change-Id: I89a3b727049ccadc11c26c1d26ebaee3a1172556
Reviewed-on: https://webrtc-review.googlesource.com/c/115789
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26106}
This adds a second (!) VoiceDetection instance in APM, activated via webrtc::AudioProcessing::Config and which reports its values in the webrtc::AudioProcessingStats struct.
The alternative is to reuse the existing instance, but that would require adding a proxy interface returned by AudioProcessing::voice_detection() to update the internal config of AudioProcessingImpl when calling voice_detection()->Enable().
Complexity-wise, no reasonable client will enable both interfaces simultaneously, so the footprint is negligible.
Bug: webrtc:9947
Change-Id: I7d8e28b9bf06abab8f9c6822424bdb9d803b987d
Reviewed-on: https://webrtc-review.googlesource.com/c/115243
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26101}
This reverts commit d1208c26b1cdb536fdec942207033711101d5d26.
Reason for revert: This cl causes the crashing issue as in
chromium:916961 at starting desktop capture on Windows.
Original change's description:
> Desktop capturer: Add OnDisplayChanged callback
>
> This adds support for a new DesktopCapturer::Callback method
> OnDisplayChanged that is sent at the start of a desktop capture
> session and whenever the display geometry changes.
>
> This cl adds the basic structure to call this api at the start
> of the capture session. Currently Windows only.
>
> A follow-up cl will add support to call this whenever the display
> geometry changes.
>
> Bug: webrtc:10122, chromium:915411
> Change-Id: Ie7283be5992454180daab1a60f58a3b2efdfed56
> Reviewed-on: https://webrtc-review.googlesource.com/c/114020
> Commit-Queue: Gary Kacmarcik <garykac@chromium.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26053}
TBR=jamiewalch@chromium.org,braveyao@webrtc.org,braveyao@chromium.org,garykac@chromium.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10122, chromium:915411, chromium:916961
Change-Id: Id0471e01bb90bb5accdf58262ae2b130cf343ecd
Reviewed-on: https://webrtc-review.googlesource.com/c/115433
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26095}
SSRCs are specified twice in calls to the RtpVideoSender constructor.
Once in the first argument of ssrcs, and then again in the RtpConfig
ssrcs variable. Resolving to reference the variable in the RtpConfig.
Bug: None
TBR: stefan@webrtc.org
Change-Id: I53528140166a53f3558f950d5662b7d3d6b8c822
Reviewed-on: https://webrtc-review.googlesource.com/c/114910
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26094}
The new controller behaves mostly like before, but increases the target
rate on timer update rather than when feedback is received. This makes
the behavior easier to predict. It also uses a duration parameter to
track the increase, removing the meed for the minimum rate increase
constants that exists in the previous solution.
Bug: webrtc:9718
Change-Id: Iae31a9ba2d6474a8236f8eb72f86ff434f1d1fc6
Reviewed-on: https://webrtc-review.googlesource.com/c/114681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26088}
Perf bots can be removed from this config because they will not perform compilation anymore.
Note that Linux64 Builder already exists.
Bug: chromium:908001
Change-Id: I3d2de332083bc0e7054fa09f8814c6500fad9ee4
Reviewed-on: https://webrtc-review.googlesource.com/c/115413
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26087}
This test was added in
https://webrtc-codereview.appspot.com/15789004/, which looks like it
was early on in adding support for capture and encode with native
textures on Android. Since then, the VideoFrame/VideoFrameBuffer
interfaces have emerged and texture support has changed a lot (now
using VideoFrameBuffer with type kNative).
The test only exercises the parts of the video pipeline before
VideoStreamEncoder::pre_encode_callback_, which doesn't care at all
about the type of the corresponding VideoFrameBuffer. That's not so
useful, and since it blocks removal of pre_encode_callback, let's
delete this old test.
Bug: webrtc:9864, chromium:362437
Change-Id: I2eb6c4c48557883309fd6431bc25528441c83078
Reviewed-on: https://webrtc-review.googlesource.com/c/115411
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26086}
Just ignoring single_packet_reduction_len is wrong, because if the
fragment is put in a single packet it might still be the first or the
last packet in the whole sequence.
Bug: none
Change-Id: I4a2fbebe1d49cbef9298bb32d9cecaa617e4dfc3
Reviewed-on: https://webrtc-review.googlesource.com/c/115403
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26084}
This allows rtc::scoped_refptr to be used with templates
that use element_type as the mechanism to interface with
smart pointers.
Bug: None
Change-Id: Ie742f416a78efce0b07cfa3009d939e51506ccf9
Reviewed-on: https://webrtc-review.googlesource.com/c/115100
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26078}
The BBR controller can still be injected, but the trials
will no longer work. This reduces the binary size.
Bug: webrtc:8415
Change-Id: I2c32c414d08ef0cc16bfd72651535a755cde9916
Reviewed-on: https://webrtc-review.googlesource.com/c/114120
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26077}
Set spatial index of assembled VP9 picture equal to spatial index of
its top spatial layer frame.
Bug: webrtc:10151
Change-Id: Iae40505864b14b01cc6787f8da99a9e3fe283956
Reviewed-on: https://webrtc-review.googlesource.com/c/115280
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26075}
To be able to reuse VideoBroadcaster, that class needs to be
officially threadsafe. It already had the needed locks, but thread
checkers have to be deleted to allow calls to AddOrUpdateSink on
multiple threads (worker thread + encoder thread).
Bug: webrtc:6353, webrtc:10147
Change-Id: I16128ac205c566f09402b6f22587a340d9a983c1
Reviewed-on: https://webrtc-review.googlesource.com/c/115201
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26073}
This fixes a bug where the streams are not updated if the "msid" changes
without triggering "ontrack", such as if the streams associated with a
receiver changes while the receiver is active.
Bug: webrtc:10083, chromium:916934
Change-Id: Ic7b19ad5ef648ed6880cae4157bf49f8435467ae
Reviewed-on: https://webrtc-review.googlesource.com/c/114161
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26069}
The deprecated method will instantiate the alr detector member
that is not actually used later on.
Bug: webrtc:10108
Change-Id: I78ac8f286758078b5a9351578bea44a862e499c4
Reviewed-on: https://webrtc-review.googlesource.com/c/115180
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26068}
The return value is not used. This change prepares for future
refactoring by removing the requirement that TryDeliverPacket must be
synchronous. Also renaming to DeliverPacket as we no longer need to
indicate the meaning of the return value.
Bug: webrtc:9510
Change-Id: I78536434b198fa7bf4df88b10d6add23684767f1
Reviewed-on: https://webrtc-review.googlesource.com/c/115181
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26066}
Currently the stats callback is registered too early.
For now we ignore media transport for these callbacks (it was ignored
already), and we will introduce changes to media transport in the
future.
Bug: webrtc:9719
Bug: chromium:906998
Bug: chromium:906533
Change-Id: I24c0265d46ec2eb35743de6cd96a11d8c41fefbe
Reviewed-on: https://webrtc-review.googlesource.com/c/114904
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26062}
