Intended to make copy construction and assignment of EncodedImage
cheaper, but otherwise not have any effect on users of the class.
Bug: webrtc:9378, chromium:931692
Change-Id: I22cf8c05f6ef7b7b5cf7ef08fd0dfc5c61211196
Reviewed-on: https://webrtc-review.googlesource.com/c/123442
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26760}
Added a layer in RtpSender that bridges the gap between the layers
that the user sees and the layer that the media engine sees.
Media engine still maintains the invariant that the number of layers
cannot be changed, while RtpSender adds and removes layers between
the user GetParameters and SetParameters calls and the media engine.
Bug: webrtc:10251
Change-Id: I33839c1f9a9052cb6130253e5a582606f2cbe54a
Reviewed-on: https://webrtc-review.googlesource.com/c/122641
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26756}
This reverts commit 9c31ac23231a3494a794b3ba0a6b018969eaa7aa.
Reason for revert: Breaks downstream project.
Original change's description:
> Tests for multi-stream Opus.
>
> This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
> tests are in audio_coding_unittest.cc. Some refactoring of
> AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
> possible. A few checks for "channels \in {1, 2}" are replaced with
> "channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
> other changes are made to be able to write and read multi-channel WAV
> files.
>
> The SDP changes are NOT included; as of this CL there is no way to set
> up a multi-stream opus en/de-coder from SDP strings.
>
> Bug: webrtc:8649
> Change-Id: I1d93a9b8eecc3f6e19896ff2e2ce9b63da77a23c
> Reviewed-on: https://webrtc-review.googlesource.com/c/114883
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26742}
TBR=aleloi@webrtc.org,ossu@webrtc.org
Change-Id: I0ac48b7320d31d3e7af33bf8714c3db6c807b82e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/123385
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26747}
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.
The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.
Bug: webrtc:8649
Change-Id: I1d93a9b8eecc3f6e19896ff2e2ce9b63da77a23c
Reviewed-on: https://webrtc-review.googlesource.com/c/114883
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26742}
The needed constant kMaxSpatialLayers is now moved to
video_codec_constants.h.
Bug: webrtc:8311
Change-Id: Iefde2e0668ce6a8d262d2747ad497ae8891873f9
Reviewed-on: https://webrtc-review.googlesource.com/c/123181
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26739}
Reland with fixes for failing chromium tests.
Propagate VideoFrame::UpdateRect to encoder
Accumulate it in all places where frames can be dropped before they reach the encoder.
Reset UpdateRect in VideoBroadcaster if frame the previous frame is dropped.
No accumulation is done here since it's supposed to be a brief occasion then configuration have changed.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/123102
Bug: webrtc:10310
Change-Id: I18be73f47f227d6392bf9cb220b549ced225714f
Reviewed-on: https://webrtc-review.googlesource.com/c/123230
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26738}
Originally RtcEventProbeClusterCreated was logged in bitrate prober. This means that anyone who was using GoogCcNetworkControl wasn't logging it, and the NetworkControl wasn't self-contained.
This changes moves the responsibility for logging ProbeClusterCreated to ProbeController (where the probe is created), it also moves the responsibility for assigning probe ids to the probe controller.
Bug: None
Change-Id: If0433cc6d311b5483ea3980749b03ddbcd2bf041
Reviewed-on: https://webrtc-review.googlesource.com/c/122927
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26713}
Accumulate it in all places where frames can be dropped before they reach
the encoder.
Reset UpdateRect in VideoBroadcaster if frame the previous frame is dropped.
No accumulation is done here since it's supposed to be a brief occusion then
configuration have changed.
Bug: webrtc:10310
Change-Id: I2813ecd009eb730bd99ffa0a02f979091b56bf80
Reviewed-on: https://webrtc-review.googlesource.com/c/123102
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26711}
This works around https://gcc.gnu.org/bugzilla/show_bug.cgi?id=89305, which
causes GCC to fail to build the code due to |thread_checker_| being const
there and not having a declared constructor.
../../api/ice_transport_factory.cc: In constructor ‘webrtc::{anonymous}::IceTransportWithTransportChannel::IceTransportWithTransportChannel(std::unique_ptr<cricket::IceTransportInternal>)’:
../../api/ice_transport_factory.cc:31:3: error: uninitialized const member in ‘const class rtc::ThreadChecker’ [-fpermissive]
IceTransportWithTransportChannel(
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../api/ice_transport_factory.cc:45:28: note: ‘const rtc::ThreadChecker webrtc::{anonymous}::IceTransportWithTransportChannel::thread_checker_’ should be initialized
const rtc::ThreadChecker thread_checker_;
^~~~~~~~~~~~~~~
Bug: chromium:819294
Change-Id: I750e8cdd796b3b0e076de01194cf7de988ac4ce2
Reviewed-on: https://webrtc-review.googlesource.com/c/122820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Raphael Kubo da Costa (rakuco) <raphael.kubo.da.costa@intel.com>
Cr-Commit-Position: refs/heads/master@{#26662}
This reverts commit 12e5d392cc8fc0ba7a04587c190daa4232e412bb.
Reason for revert: Partial Capture API is not needed, according to new info from the Chrome team.
Original change's description:
> Reland "Partial frame capture API part 1"
>
> Reland with fixes to undefined behavior.
>
> Define new optional struct in VideoFrame to signal that the frame is a
> changed part of a whole picture and add a flag to signal that partial
> update may be issued by the VideoFrame source.
>
> Also, fix too strict assumptions in FrameBuffers PasteFrom methods.
> Also, add ability to set a new buffer in video frame.
>
> Original Reviewed-on: https://webrtc-review.googlesource.com/c/120405
>
> Bug: webrtc:10152
> Change-Id: I85790dfc7cec2f23abfe9d6cd18dc76a0c343bc0
> Reviewed-on: https://webrtc-review.googlesource.com/c/120780
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26493}
TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10152
Change-Id: I1c1dd51a8b5a09f743f212336beb01447f60f26e
Reviewed-on: https://webrtc-review.googlesource.com/c/122092
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26638}
This reverts commit 126648763184b7e224d6c4a2f85efb4a9307378f.
Reason for revert: Partial Capture API is not needed, according to new info from the Chrome team.
Original change's description:
> Partial frame capture API part 3
>
> Implement utility for applying partial updates to video frames.
>
> Bug: webrtc:10152
> Change-Id: I295fa9f792b96bbf1140a13f1f04e4f9deaccd5c
> Reviewed-on: https://webrtc-review.googlesource.com/c/120408
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26522}
TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10152
Change-Id: I9d7c79ca571a44a419102871d3106e7065638433
Reviewed-on: https://webrtc-review.googlesource.com/c/122089
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26630}
This reverts commit 1f0a84a2ecea59f86adc1af70eed974a3c6d59ac.
Reason for revert: Partial Capture API is not needed, according to new info from the Chrome team.
Original change's description:
> Partial frame capture API part 5
>
> Wire up partial video frames in video quality tests
>
> Bug: webrtc:10152
> Change-Id: Ifa13bb308258c8d3930a6cfbcc97c95b132cecf3
> Reviewed-on: https://webrtc-review.googlesource.com/c/120410
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26549}
TBR=ilnik@webrtc.org,sprang@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10152
Change-Id: I32017b1a7109a3615598a976f4b0e61edf4e8757
Reviewed-on: https://webrtc-review.googlesource.com/c/122088
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26628}
Logging will not stop after 10 minutes but after the max size of the
file is reached.
Bug: None
No-Try: True
Change-Id: I18c064cc60b52d892d5070b3eddf08749b60604c
Reviewed-on: https://webrtc-review.googlesource.com/c/121959
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26625}
Hard-coding default values forces IDs over 14 to be used even
when we offer less than 15 different extensions.
Note that the code relies on MergeRtpHdrExts for making sure
that extension IDs are kept consistent and non-colliding between
different streams (audio/video).
Bug: webrtc:10288
Change-Id: I3e59f7ddc8ca43cea91084a6b7f36df70fb6be4a
Reviewed-on: https://webrtc-review.googlesource.com/c/121646
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26622}
The new parameter simplify TaskQueueFactory lifetime and
allows insertion of per TestCase state
Bug: webrtc:10191
Change-Id: If4948df2756580957052b2b333b5c12cf4914d55
Reviewed-on: https://webrtc-review.googlesource.com/c/121648
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26583}
Adds methods AddNetworkChangeCallback and RemoveNetworkChangeCallback,
to replace SetNetworkChangeCallback. Needed because both VideoChannel and
VoiceChannel register such a callback.
Bug: webrtc:9719
Change-Id: Ic592b2d775d721a0f44ba0af88ed963bf02d73a3
Reviewed-on: https://webrtc-review.googlesource.com/c/121460
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26575}
Wire up partial video frames in video quality tests
Bug: webrtc:10152
Change-Id: Ifa13bb308258c8d3930a6cfbcc97c95b132cecf3
Reviewed-on: https://webrtc-review.googlesource.com/c/120410
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26549}
Adds methods AddNetworkChangeCallback and RemoveNetworkChangeCallback,
to replace SetNetworkChangeCallback. Needed because both VideoChannel
and VoiceChannel register such a callback.
This cl is step 1, it just adds the methods to the interface, without
calling them.
Bug: webrtc:9719
Change-Id: I39f1748706d4369ca71d594ca5e2f1380de5ce66
Reviewed-on: https://webrtc-review.googlesource.com/c/121462
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26541}
