Bug: webrtc:9719
Change-Id: I9ec89fca7d4555f31b5192980f193b58d99e3b71
Reviewed-on: https://webrtc-review.googlesource.com/c/125100
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26910}
This involves inserting an extra layer between jsep_transport_controller
and the cricket::SctpTransportInternal layer. The objects at this layer
are reference counted.
Bug: chromium:818643
Change-Id: Ibed57c4a538de981cee63e0f7f1f319f029cab39
Reviewed-on: https://webrtc-review.googlesource.com/c/123884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26889}
This replaces the use of command-line flags with the use of a config
struct. This makes it easier for non command-line applications to use
the NetEqTestFactory to run simulations.
Bug: webrtc:10337
Change-Id: I24533bf206e70e12db9af8d9675769c1ff7c7d48
Reviewed-on: https://webrtc-review.googlesource.com/c/123600
Reviewed-by: Pablo Barrera González <barrerap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26887}
If x-mt line is present (one or more), and the first line is dedicated
for the media transport that we support, pass the config down to this
media transport.
In the future we will do 3 changes:
1) Add MediaTransportFactory::IsSupported(config) to let the
implementation decide whether the current factory can support a given
setting
2) Add support for multiple x-mt lines. Right now the support is
minimal: we only look at the first line (because we only allow single
media transport factory). In the future, when RtpMediaTransport is
introduced, this may and will change.
3) Allow multiple MediaTransportFactories and add fallback to RTP if
media transport is not supported.
Current solution provides backward compatibility for the 2 above
extensions.
Bug: webrtc:9719
Change-Id: I82a469fecda57effc95d7d8191f4a9e4a01d199c
Reviewed-on: https://webrtc-review.googlesource.com/c/124800
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26882}
On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.
Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}
One-byte RTP extensions may only have IDs in the range 1-14.
For higher IDs, the two-byte format must be used.
If default IDs are set for all extensions, once 15 extensions are
defined by the code, some extensions will have IDs greater than 14.
This will happen even if only one extension actually ends up being
offered, so long as it's that unfortunate RTP extension.
It's better to dynamically assign the IDs to those extensions we
actually offer. The code that assigns the IDs is currently
distributed ( WebRtcVoiceEngine::GetCapabilities() and
WebRtcVideoEngine::GetCapabilities()), and without a bigger
refactoring effort would produce some ID collisions and mismatches.
Those are already handled by MergeRtpHdrExts(), so so that
should not be a problem.
Bug: webrtc:10288
Change-Id: I087f1ed5baa9fd61fd5556f1d82f540304ec6b93
Reviewed-on: https://webrtc-review.googlesource.com/c/122480
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26876}
In order to enable ::Connect method, we also need to split the factory and create a method that creates media transport, but doesn't connect it.
So far media transport was connecting right away after creation. We would however want to expose some of the settings in SDP. SDP is created before connection is connected (and before ICE transport is created), and so we would like to be able to get the settings from the caller to the callee.
Bug: webrtc:9719
Change-Id: I1dc2f30c9a2dae8b3db04f14c8b334cd1b3ab5ab
Reviewed-on: https://webrtc-review.googlesource.com/c/124517
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26863}
Bug: webrtc:9719
Change-Id: I4aef407c4770fc98abcbc114b87e73bbf13d8f56
Reviewed-on: https://webrtc-review.googlesource.com/c/124021
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26860}
Adds an implementation of the CoDel active queue management algorithm
to the network simulation. It is loosely based on CoDel pseudocode
from ACMQueue: https://queue.acm.org/appendices/codel.html
Bug: webrtc:9510
Change-Id: Ice485be35a01dafa6169d697b51b5c1b33a49ba6
Reviewed-on: https://webrtc-review.googlesource.com/c/123581
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26834}
This removes the temporary non-const method that was kept in the code to
enable backwards compatibility while we fix downstream project dependencies.
Bug: webrtc:10251
Change-Id: Ie221af1d3b0f19112449d61e0f357a833f7a8b18
Reviewed-on: https://webrtc-review.googlesource.com/c/123561
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26824}
This informs the media transport that PeerConnection wants to use a data channel
and gives it a chance to set up before the data channel sends the first message.
Bug: webrtc:9719
Change-Id: I6ea905a74b29b8735e77ac68bc8606e7bca77f18
Reviewed-on: https://webrtc-review.googlesource.com/c/124020
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26823}
TransportSequenceNumberV2 is an experimental feature that should
not be part of the default offer. However, if we receive an offer
with this extension we should respond that we support it.
Bug: webrtc:10264
Change-Id: Id2424d421361e5d71f3a608cb8f74b63645c264a
Reviewed-on: https://webrtc-review.googlesource.com/c/123783
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26817}
The difference to the original is new bitexactness strings. The
reason for reland is breaking downstream projects.
Original CL 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.
TBR=ossu@webrtc.org
Bug: webrtc:8649
Change-Id: I6261b18c69fd666d43ab34ed8f1bc9d5cc82b21f
Reviewed-on: https://webrtc-review.googlesource.com/c/123882
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26809}
This reverts commit 5341aaccdb64e3336abf5875e8828222446adffa.
Reason for revert: Order of initialization of global static strings.
Original change's description:
> Reland of https://webrtc-review.googlesource.com/c/src/+/114883
>
> The difference to the original is new bitexactness strings AND
> global static file string constants. The reason for reland is breaking
> downstream projects.
>
> Original CL 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: I9fd47c790c241c1876c4a731b0840bec30b4f1b2
> Reviewed-on: https://webrtc-review.googlesource.com/c/123387
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26774}
TBR=aleloi@webrtc.org,ossu@webrtc.org
Change-Id: I88060f2050ccee83d6091b042a10f79b3c4edc47
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/123580
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26777}
The difference to the original is new bitexactness strings AND
global static file string constants. The reason for reland is breaking
downstream projects.
Original CL 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: I9fd47c790c241c1876c4a731b0840bec30b4f1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/123387
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26774}
The discardability flag denotes whether the frame may be dropped by
the decoder with no effect on the decodability of subsequent frames.
Bug: webrtc:10214
Change-Id: I3654951d8863b50effe9670b8d1d7eb051240039
Reviewed-on: https://webrtc-review.googlesource.com/c/122241
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26763}
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}
Latency corresponds to base minimum delay on NetEq.
Bug: webrtc:10287
Change-Id: I538d202e3e4fe07b779c46bf560e2fde38e0468e
Reviewed-on: https://webrtc-review.googlesource.com/c/121704
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26724}
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}
