Just fixing some minor TODOs in my name. Not worth splitting into
separate CLs as the changes are minor.
Bug: webrtc:9883
Change-Id: I05c54b76507a1d51b92cad080ca4e2dfe8546bf1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155520
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29377}
This improves the accuracy of the priority bitrate on IPv6 networks
and when the min frame length is longer than 20 ms. Unless either of
those are true, there's no significant change in behavior.
Bug: webrtc:11001
Change-Id: I29530655cb607a8e7e8186431cd9362ca397910b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155521
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29375}
To avoid incorrect QP sum in the reported stats and to avoid log spam
when switching from a decoder that reports QP to a decoder that does
not report QP.
Bug: None
Change-Id: Ib5ef4d6227344b0d03c3d75596b4a07ef427ae1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155444
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29373}
This CL removes some unnecessary locking, since we are already
serialized by the lock in VideoStreamEncoder. A simple RaceChecker is
used to verify this.
We also remove the usage of RegisterPayloadType() and replace it with
a parameter in SendVideo instead. This way we are prepared for removing
the payload type map and lock entirely. Some usage still exists
downstream and needs to be removed before cleaning this up.
Bug: webrtc:10809
Change-Id: Ie90163f15d11c8843f3beaf9a0df0dd2a1fd5ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154700
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29372}
This is a no-op change that just removes the AudioAllocationSettings
helper class that was previously introduced since the field trials in it
were used in several places. Those other usages has now been removed
and AudioSendStream is now the only user. By moving the trials directly
to AudioSendStream we reduce the reader overhead when trying to follow
what a particular field trial does.
The "WebRTC-Audio-ForceNoTWCC" trial was removed as it is always set
together with "WebRTC-Audio-ABWENoTWCC".
Bug: webrtc:9883
Change-Id: Ib63589255bfe7adb155ea41279bdcd153f1536c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155366
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29371}
This adds the ability to disable legacy overhead calculation so we'll
use the available data on per packet over head and frame length range
to set the min and max total allocatable bitrate.
Bug: webrtc:11001
Change-Id: I2a94499433e15bad11a08f81fe7f1dfc27982cdf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155175
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29368}
This is a reland of a21d50c1f3eab29fd9026cc67c8cb4017efda5e3
Original change's description:
> Prepares RtpSenderVideo for batch forwarding of generated packets
>
> In order to reduce contention, this CL avoids taking locks per packet
> and prepares for forwarding all packets for a frame in one call, rather
> than one at a time. This will especially reduce contention in the paced
> sender during very high packet rates.
>
> Bug: webrtc:10809
> Change-Id: Ifc5fe3759b76a2a45f418b69d29c329e876f96d0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154358
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29323}
Bug: webrtc:10809
Change-Id: I50e0a27eb3b0b1afa39f250febdd564e1e1f06eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155362
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29367}
This CL adds an optional second encoder factory to SimulcastEncoderAdapter,
that can be used to create software fallback adapter per simulcast layer.
It also adds logic to check if the encoder supports simulcast natively, if so
it only allocates a single instance and delegates the simulcast logic to that
encoder instead. This means we will be able to remove EncoderSimulcastProxy.
Bug: webrtc:11000
Change-Id: Ifd5f029cc281ee2cedf9d18efa5e7e460884d6ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155171
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29364}
The header files
api/congestion_control_interface.h
api/data_channel_transport_interface.h
api/datagram_transport_interface.h
api/media_transport_config.h
api/media_transport_interface.h
have been moved into the api/transport/ and api/transport/media
subdirectories.
Bug: webrtc:8733
Change-Id: I98752c4d1306b54559bafa71712b105932c08834
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153522
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29357}
This is a reland of 809198edfff416fce8d75b574a43afab5e67b1cd
A fix was made in https://webrtc-review.googlesource.com/c/src/+/154343
which fixed the regression issues caused by the original patch.
Original change's description:
> Fix minor regression caused by a8336d3
>
> VideoEncoder::SetRates was being called unnessesarily when the fields
> appended to RateControlParameters were changed. Since SetRates only
> cares about RateControlParameters, it should have only been called if
> the RateControlParameters themselves were actually changed.
>
> Bug: webrtc:10126
> Change-Id: Ic47d67e642a3043307fec950e5fba970d9f95167
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152829
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#29208}
Bug: webrtc:10126
Change-Id: Iecc3ab6a5cd1193a1fa8e824dcf4f0b8165f9bf8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154359
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29356}
The plugin transport parameters (a=x-opaque: lines) relate to how to create and
set up a plugin transport. When SDP bundle is used, the x-opaque line needs to
be copied into the bundled m= section. This means x-opaque can appear on a
section even if the offerer does not intend to use the transport for the media
described by that section. Consequently, the answerer cannot currently tell
whether the caller is offering an alternate transport for media, data, or both.
This change adds an a=x-alt-protocol: line to SDP. The value following this
line matches the <protocol> part of the x-opaque:<protocol>:<params> line.
However, alt-protocol is not bundled--it only ever applies to the m= section
that contains the line. This allows the offerer to express which m= sections
should actually use an alternate transport, even in the case of bundle.
Note that this is still limited by the available configuration options:
datagram transport can be used for media (audio + video) and/or data. It is
still not possible to use it for audio but not video, or vice versa.
PeerConnection places an alt-protocol line in each media (audio/video) m=
section if it is configured to use a datagram transport for media. It places
an alt-protocol line in each data m= section if it is configured to use a
datagram transport for data channels. PeerConnection leaves alt-protocol in
media (audio/video) m= sections of the answer if it is configured to use a
datagram transport for media, and in data m= sections of the answer if it is
configured to use a datagram transport for data channels.
JsepTransport now negotiates use of the datagram transport independently for
media and data channels. It only uses it for media if the m= sections for
bundled audio/video have an alt-protocol line matching the x-opaque protocol,
and only uses it for data channels if a bundled m= section for data has an
alt-protocol line matching the x-opaque protocol.
Bug: webrtc:9719
Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29351}
This started to be detected by a new version of clang and it is blocking
the roll:
../../third_party/webrtc/test/fuzzers/agc_fuzzer.cc:85:29: error: converting the result of '?:' with integer constants to a boolean always evaluates to 'true' [-Werror,-Wtautological-constant-compare]
const bool num_channels = fuzz_data->ReadOrDefaultValue(true) ? 2 : 1;
Bug: chromium:1007367
Change-Id: Ib9a6e4e3c8f109d10845a315dd0782b1498cb54e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155166
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29348}
Intended to be used for holding on to references to the java
EncodedImage and call its release method when no longer used by C++.
Bug: webrtc:9378
Change-Id: I40d917c2bb4217419ef2d609e517566c8466a274
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154740
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29347}
Renaming and splitting it into helper methods. This is to more clearly
separate the things it does and prepares for moving things to GoogCC.
Additionally, replacing calls with current_target_ as input with
ApplyTargetLimits to better reflect the intended behavior.
Bug: webrtc:9883
Change-Id: I2c47ec74a9cbc271aff91645c763373297f26acc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154425
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29346}
Passing an empty arg is working at the moment but it is not
guaranteed to continue to work in the future.
Bug: None
Change-Id: I975bc8779bac9700854de411301415338dcaf9f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154820
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29343}
This reverts commit a21d50c1f3eab29fd9026cc67c8cb4017efda5e3.
Reason for revert: Speculative revert due to unexpected perf changes.
Original change's description:
> Prepares RtpSenderVideo for batch forwarding of generated packets
>
> In order to reduce contention, this CL avoids taking locks per packet
> and prepares for forwarding all packets for a frame in one call, rather
> than one at a time. This will especially reduce contention in the paced
> sender during very high packet rates.
>
> Bug: webrtc:10809
> Change-Id: Ifc5fe3759b76a2a45f418b69d29c329e876f96d0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154358
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29323}
TBR=ilnik@webrtc.org,sprang@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10809
Change-Id: I1cbf0ce0cc06f9195b5e0716b8dd4c85f7f6bab1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155164
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29341}
If the packet can't be parsed, the buffer isn't moved to the packet.
Then, a new empty buffer is moved back from the packet.
Thus, the consequtive DCHECK fails because the data isn't the same anymore.
Bug: chromium:1009236
Change-Id: Ie27f438c40f38074d42d8491fe03df45d50eba50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155162
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29340}
Lately there were 2 separate bugs, where seeing this information in the
log could help immediately figuring out the problem.
Bug: none
Change-Id: I3f2b2d5864106cdb231715e1702edee3b9b05caa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154566
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29338}
Modifying buffers passed in to the frame buffer breaks sharing. This
cl is also a preparation for deleting
VCMEncodedFrame::VerifyAndAllocate and EncodedImage::Allocate.
Bug: None
Change-Id: I4e14bc4708bbcbcd91af2d4b764cb9b8271ec090
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154569
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29336}
actual task queue implementation for these tests is intentionally unchanged for now.
while at it, change return type of created transports to unique_ptr to note passing ownership.
Bug: webrtc:10933
Change-Id: I324597b503e647c471f43511340eb9c07ba03ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154743
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29335}
Now vp9 screenshare would enable new layers as soon as requested and will force all spatial layers present on the next frame, even if they should be dropped because of frame-rate limiting.
This might cause frame-rate liming to be exceeded if layer is toggling on and off very often, but this situation is bad itself. E.g. in realtime video it will cause too many key-frames.
Now SvcRateAllocator and VP9EncoderImpl are aware that there may be some skipped layers before the first enabled. Key-frames and ss_info triggering logic is also updated.
(This is a reland without changes after updates to downstream projects)
Original-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153483
Bug: webrtc:10977
Change-Id: I02459c5982da2e0542a837514f5753c5f96401c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154355
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29330}
