This refactoring helps to reduce unnecessary memcpy calls on the receive
side.
This CL is the first stage of refactoring: it only replaces
|uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| and does
necessary changes.
A follow-up CL will remove length field of the Packet class.
Bug: webrtc:10750
Change-Id: Ie233da83ff33f6370f511955e4c65d59522389a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144881
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28539}
The (currently unused) new pacer code path was implemented with what
was intended as a more careful padding strategy.
Unfortunately this doesn't work as well as expected due to the fact
that the padding budget cannot build up underuse.
I made another CL that could fix that, but I think it adds complexity
for dubious gains. It also will make it more difficult to find any
potential regression when switching to the new path, should one occur.
See https://webrtc-review.googlesource.com/c/src/+/144563
Therefore, this CL makes the new code path choose RTX payload in the
same way as is currently done.
Bug: webrtc:10633
Change-Id: If2115d4fa7463add959faa77c63101286c27e0f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145202
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28537}
The legacy code-path uses a hashmap as cache in order to speed up
finding the right rtp module to send on. The new path should use that
as well.
In addition, there are checks that verify if an RTP module can send
padding, in some cases payload based. These result in a number of
calls to methods in RTPSender requiring its lock to be taken. This CL
introduces a combined SupportsPadding() check method which performs
all those checks in one go.
Bug: None
Change-Id: I2d18d0d6e7d8cfe92c81d08cef248a4daa7dcd4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144780
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28535}
Datagram_dtls_adaptor needs access to rtp_rtcp modules and this moves helps to keep p2p/base/ without dependency on rtp_rtcp.
Bug: webrtc:9719
Change-Id: Ic337be3fb9f68106187a84efa815eefbe5b0fcd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145267
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28533}
Removing extension will be used in DatagramDtlsAdaptor to remove transport sequence number to avoid having both datagram and RTP feedback loops. The sequence number will be stored in temporary map and used to re-create RTCP fdeedback packed when we receive datagram ACK. It would enable integration of Datagram transport without any changes in the upper layers of RTP stack. Note that Datagram adaptor changes will be implemented in a separate changelist.
In this change:
- Implement method to remove extension by rebuilding entire packet without given extension type.
- Fails if extension is not registered or not set.
- Unit test
Bug: webrtc:9719
Change-Id: I9d3811d5d97fadde5a294d5da643b2ebc6a8196e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145100
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28530}
This is a reland of 2d0880b56954f57548deea51dfa678b80dbf618f
To fix perf bot issue reading of perf results file was updated. Now perf
results file will be generated by each test and then returned via output to
the python script, which will get it and put into final file.
Original change's description:
> Fix collection of audio metrics from PC test framework for audio test
>
> Bug: webrtc:10138
> Change-Id: I18a8509a0cdc4ed1db6894c7540d5c0a155d6233
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144784
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28514}
Bug: webrtc:10138
Change-Id: I1347f09427736362a2d550612b48e09c06cfb1d1
No-Presubmit: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145201
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28527}
This is a trivial CL, updating rtp_generator.cc according to changes in
APIs in other places.
Bug: webrtc:10807
Change-Id: Ie85c6283f2d78dcf742979378db0b4fb0914c96c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145209
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28526}
Users of webrtc generally should be able to choose own task queue implementation.
Poison avoids accidental dependency of a low level component on the default implementation
Android and ios apis are still de-facto forced to use the default implementation.
Bug: webrtc:10284
Change-Id: I67ecf2317f43ee32b0c9e8a6e69f1e0987cf1914
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144786
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28524}
This is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770
Patch set 1 is the original CL.
Patch set 2 introduced a trivial fix. In RtcpSender::SetSSRC(), check
if either current SSRC is 0 or if the SSRC is identical to the current
one. If so, don't schedule an early report.
This prevents a regression in which audio jitter became too low(?)
Original change's description:
> Add ability to set RTCP sender ssrc at construction time
>
> Bug: webrtc:10774
> Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28506}
Bug: webrtc:10774
Change-Id: I103dfa48719aa41d6ab633cdac8b3a5c46b54843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144565
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28520}
Ensures that newly added ChannelMixer is utilized when number of channels
is larger than two in the output mixer.
Decided to land with henrik.lundin as TBR since he has reviewed all other
changes in WebRTC related to channel mixing for multi-channel cases.
All this CL does is to ensure that the new channel mixing scheme can be used
in Chrome. The old scheme is still used for mono and stereo combinations.
TBR: henrik.lundin
Bug: webrtc:10783
Change-Id: I11c02f1b4ef60e847095efbcd5e5f5faf27a5cdd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140290
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28517}
If VideoEncoderConfig::max_bitrate_bps is unset then max bitrate of
video stream is set equal to max bitrate value recommended by encoder
for given resolution via encoder capabilities (if available).
Bug: webrtc:10796
Change-Id: I7fce9afc476b794a16956e694e891faee110048e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144526
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28515}
A cloaked window is composited but not visible to the user.
When Win10 feature 'Cortana' is enabled it creates a window
that is always invisible and its z-order is top most. Because
of that the cropping capturer detects occlusion everywhere
preventing it from capturing anything.
The solution is to ignore all cloaked windows like if
::IsWindowVisible would return false.
Bug: chromium:978885
Change-Id: Id5aa8dc81dcf4979ffb30dd808fa2a553934c6e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143980
Commit-Queue: Julien Isorce <julien.isorce@chromium.org>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28510}
Ntp timestamps are used for end-to-end delay measurements and can never
go away. The naming and number of timestamp fields in VideoFrame could
change in the future, but capture time in local clock will always be
there on the receive side.
Bug: none
Change-Id: I358689cd8a44b1da8503136b3dd898b936f2d693
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144542
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28498}
This CL makes the new code path for paced sending functionally complete.
By default, the field trial WebRTC-Pacer-ReferencePackets is Enabled,
meaning that there is no behavior change unless the field trial is
forced to Disabled. This is done in tests, and can be done on the
command line for manual testing.
Bug: webrtc:10633
Change-Id: I0d66c94ef83b5847dee437a785018f09ba3f828d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144050
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28497}
Encountering GL_OUT_OF_MEMORY is relatively common and we should give
clients a chance to deal with it in a non-fatal way.
Bug: webrtc:8154
Change-Id: Ifa9ca74392f21083692b02a5144dc5632a88d34d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144561
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28495}
There is no public API to create NetworkBehaviorInterface from
BuiltInNetworkBehaviorConfig, so this CL will add direct method, that will
allow downstream projects to use BuiltInNetworkBehaviorConfig for network
emulation.
Bug: webrtc:10138
Change-Id: Iaec3ea17c12bd06b1c0ff3e5bc2b32cc1c4f62f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144628
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28494}
Add PC level audio quality test as separate file but into the same
binary as low_bandwidth_audio_test.cc
Bug: webrtc:10138
Change-Id: I3bb7df4130ab62c21a82881cd9f57b936f3cc11d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144621
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28493}
