This is a reland of 18c4261339dc76b220e7c805e36b4ea6f3dd161d
Original change's description:
> Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer.
>
> Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
>
> Bug: webrtc:8653
> Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
> Reviewed-on: https://webrtc-review.googlesource.com/37740
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21646}
TBR=sprang@webrtc.org,stefan@webrtc.org,deadbeef@webrtc.org
Bug: webrtc:8630
Change-Id: Ib3df6f9b7158bff362a7ec66fc57e368682c5846
Reviewed-on: https://webrtc-review.googlesource.com/40980
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21688}
Current checks do not allow first frame after fallback to be sent for encode
because of the native checks. However, the rest of the frames are sent and
encoded correctly.
Fallback encoder, although SupportsNativeHandle() is false, can call ToI420()
which converts from native handles, and accordingly can return error or
success. This higher level check seems unnecessary.
Bug: webrtc:8021
Change-Id: I69780cc25eb1e06317ff213e9b80288064e9f1e3
Reviewed-on: https://webrtc-review.googlesource.com/40441
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21685}
It's reference counted, yet we aren't taking a reference to it for some
reason. This could be causing it to be dereferenced after deletion in
some cases in chromium.
Bug: chromium:798251
Change-Id: I0b91451e38ed611d2ea8a477f1e7db482a790f79
Reviewed-on: https://webrtc-review.googlesource.com/37283
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21684}
This also changes RtpReceiver and RemoteAudioSource to have two-step
initialization, since in Unified Plan RtpReceivers are created much
earlier than in Plan B.
Bug: webrtc:7600
Change-Id: Ia135d25eb8bcab22969007b3a825a5a43ce62bf4
Reviewed-on: https://webrtc-review.googlesource.com/39382
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21681}
This used to be used by the linter, but now it's actually using
build/android/AndroidManifest.xml to determine the minimum SDK
level we support.
Remove this file to avoid confusion.
Bug: None
Change-Id: I2ea60854d52276877ae9d51a9c1f379cfaa3ed5d
Reviewed-on: https://webrtc-review.googlesource.com/40661
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21677}
GetPacingFactor exposed internal details that should not be relied upon.
In a later CL theese won't be available any more, this CL is in
preparation for that change.
The only usage was in video send stream tests. To keep the tests
working, they now access the internal video send stream directly. The
test code retrieves an optional that indicates whether the send stream
has overridden the pacing factor. This means the implementation
dependency between video send stream and video send stream tests is
increased. This is an improvement compared to depending on the paced
sender implementation.
Bug: webrtc:8415
Change-Id: Id357553692b3ff3283fa3b64da1b1ebb3c97f04d
Reviewed-on: https://webrtc-review.googlesource.com/39265
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21675}
These features are supported via desugar, but we started getting lint errors:
"Static interface method requires API level 24 (current min is 16)"
60990c11ce
Bug: webrtc:8084
Change-Id: Iba8b5825a52bd57da7ee4ab52cde2d7bd9ebce30
Reviewed-on: https://webrtc-review.googlesource.com/40421
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21672}
And wire it up to methods on RTCConfiguration, via MediaConfig::Video.
Bug: webrtc:8504
Change-Id: I30805ee20c11d1d2fe552eb81f16d514db0ba4a8
Reviewed-on: https://webrtc-review.googlesource.com/39786
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21670}
This is a reland of 1880c7162bd3637c433f9421c798808cd6eacaf7
Original change's description:
> Updated analysis in videoprocessor.
>
> - Run analysis after all frames are processed. Before part of it was
> done at bitrate change points;
> - Analysis is done for whole stream as well as for each rate update
> interval;
> - Changed units from number of frames to time units for some metrics
> and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
> 'time to reach target bitrate, sec';
> - Changed data type of FrameStatistic::max_nalu_length (renamed to
> max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
> use such advanced data type in such low level data structure.
>
> Bug: webrtc:8524
> Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
> Reviewed-on: https://webrtc-review.googlesource.com/31901
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21653}
TBR=brandtr@webrtc.org, stefan@webrtc.org
Bug: webrtc:8524
Change-Id: Ie0ad7790689422ffa61da294967fc492a13b75ae
Reviewed-on: https://webrtc-review.googlesource.com/40202
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21668}
This changes PeerConnectionObserver to not default the stream
events as pure virtual. Applications which have switched to using
OnAddTrack and OnRemoveTrack will no longer need to implement
these callbacks.
Bug: webrtc:8587
Change-Id: I659ce7b5a208ebfcb29e899dd17916ae0072d3cc
Reviewed-on: https://webrtc-review.googlesource.com/39384
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21665}
rtc_build_ssl is defined to be !build_with_mozilla, but we have
tautological if conditions involving both. This removes the offending
clauses for now. It makes more sense to address the problem during the
next import of webrtc.org code into Firefox.
Bug: webrtc:8670
Change-Id: I586deb7c33566efae6bb6380c9d4ae61a0fe1c37
Reviewed-on: https://webrtc-review.googlesource.com/39960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Dan Minor <dminor@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21662}
This will be overridden in Chromium in a future CL and following that,
used inside of WebRTC as needed while issues with blocking events are
being fixed.
Bug: chromium:796889, chromium:795340
Change-Id: I098a03d9bad621f1349ef483b82e4d8e786a8a75
Reviewed-on: https://webrtc-review.googlesource.com/40140
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21659}
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.
NOPRESUBMIT=true
Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
This reverts commit 1880c7162bd3637c433f9421c798808cd6eacaf7.
Reason for revert: breaks internal tests
Original change's description:
> Updated analysis in videoprocessor.
>
> - Run analysis after all frames are processed. Before part of it was
> done at bitrate change points;
> - Analysis is done for whole stream as well as for each rate update
> interval;
> - Changed units from number of frames to time units for some metrics
> and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
> 'time to reach target bitrate, sec';
> - Changed data type of FrameStatistic::max_nalu_length (renamed to
> max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
> use such advanced data type in such low level data structure.
>
> Bug: webrtc:8524
> Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
> Reviewed-on: https://webrtc-review.googlesource.com/31901
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21653}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org
Change-Id: Id0b7d387bbba02e71637b229aeed6f6cf012af46
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8524
Reviewed-on: https://webrtc-review.googlesource.com/40220
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21656}
This reverts commit 16beea7c90e161ce82764bf046d588e87d38191f.
Reason for revert: breaks internal tests.
Original change's description:
> Set encodedWidth/encodedHeight to actual coded resolution.
>
> This fixes mismatch between resolution of coded frame indicated by VP9
> encoder wrapper and actual coded resolution. If internal resize is
> enabled VP9 encoder might downscale input frame when bitrate is too low
> to keep good spatial quality. Before this fix VP9 wrapper always set
> coded resolution equal to input resolution. Now it sets it to actual
> coded resolution which it reads from frame pkt.
>
> Bug: webrtc:5749
> Change-Id: I7dc8ba89947e99213a3b4c3cd4d974b662f090c4
> Reviewed-on: https://webrtc-review.googlesource.com/39661
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21651}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,ssilkin@webrtc.org
Change-Id: I32cb1271c863cf435f3fc3b465e11232cbd6a888
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5749
Reviewed-on: https://webrtc-review.googlesource.com/40161
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21655}
When https://crbug.com/webrtc/8734 is resolved this setup should be
valid and CreateOffer() and SetLocalDescription() should work, but
currently it doesn't. It probably fails because both senders are
assigned the same ID (the track ID).
EXPECT-ing the current behavior with a TODO referencing the bug.
Bug: webrtc:8734
Change-Id: If2a9cc9b0be12c39def83b0e219e1ca82dbd7d65
Reviewed-on: https://webrtc-review.googlesource.com/39041
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21654}
- Run analysis after all frames are processed. Before part of it was
done at bitrate change points;
- Analysis is done for whole stream as well as for each rate update
interval;
- Changed units from number of frames to time units for some metrics
and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
'time to reach target bitrate, sec';
- Changed data type of FrameStatistic::max_nalu_length (renamed to
max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
use such advanced data type in such low level data structure.
Bug: webrtc:8524
Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
Reviewed-on: https://webrtc-review.googlesource.com/31901
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21653}
This fixes mismatch between resolution of coded frame indicated by VP9
encoder wrapper and actual coded resolution. If internal resize is
enabled VP9 encoder might downscale input frame when bitrate is too low
to keep good spatial quality. Before this fix VP9 wrapper always set
coded resolution equal to input resolution. Now it sets it to actual
coded resolution which it reads from frame pkt.
Bug: webrtc:5749
Change-Id: I7dc8ba89947e99213a3b4c3cd4d974b662f090c4
Reviewed-on: https://webrtc-review.googlesource.com/39661
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21651}
This reverts commit 18c4261339dc76b220e7c805e36b4ea6f3dd161d.
Reason for revert: Broke internal tests
Original change's description:
> Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer.
>
> Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
>
> Bug: webrtc:8653
> Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
> Reviewed-on: https://webrtc-review.googlesource.com/37740
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21646}
TBR=deadbeef@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,shampson@webrtc.org
Change-Id: I0aeb743cbd2e8d564aa732c937587c25a4c49b09
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8653
Reviewed-on: https://webrtc-review.googlesource.com/39883
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21647}
Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
Bug: webrtc:8653
Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
Reviewed-on: https://webrtc-review.googlesource.com/37740
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21646}
Because they usually happen for all or for no frames in a stream.
Bug: chrome:801327
Change-Id: Ie09b93a0822e821076ff2743cf1223c29fcf44a6
Reviewed-on: https://webrtc-review.googlesource.com/39785
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21644}
This patch exposes the network_thread so that
a custom PortAllocator can use it instead of e.g
creating own thread.
Bug: webrtc:8640
Change-Id: I705629e4f1a4d0a4fed7d53a774ba9564ba076fe
Reviewed-on: https://webrtc-review.googlesource.com/39925
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21643}
To align behavior with video receive streams. Configuration of header
extensions happen outside the stream classes (i.e. in Call). Recent
changes stopped recreating streams when extensions changed, but relied
on reconfiguring the stream instead.
Bug: webrtc:4690
Change-Id: I9efe944f94b811c353628d3be34f548f998d0efc
Reviewed-on: https://webrtc-review.googlesource.com/39664
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21642}
AGC APIs have recently been removed from the ADM.
This CL ensures that usage of the analog part of the AGC is removed
as well (since this code is dead today anyhow).
Bug: webrtc:7306, webrtc:8598
Change-Id: I144f01cd545e5ff6900707c9308906081914e413
Reviewed-on: https://webrtc-review.googlesource.com/36120
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21640}
