They are now passed as part of the Environment
Bug: webrtc:362762208
Change-Id: I02868e9f41533a546f62fe30fdc6f3a7708eb346
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362084
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43032}
This can happen when VP8 simulcast is negotiated while two-byte header
extensions are not negotiated via extmap-allow-mixed. For VP8 the
DD extension would be 23 bytes long which exceeds the maximum size
of 15 bytes for a one-byte header extension.
To test, revert
f04b52b4a7
and test using VP8.
Note that this works for VP9, AV1, H264 out of the box.
BUG=webrtc:40191093
Change-Id: I2f5d04d8b58b71d32547b06fab6b9a9006df9f1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359623
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42786}
Split out time_util.h and cc from target rtp_rtcp to its own target.
This is to avoid possible circular dependencies and not having all targets using them to depend on the full RtpRcp module.
Bug: webrtc:343076000
Change-Id: I7b3c84456b17f1920f71afdd5a644d27e28caed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352480
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42392}
This depenency is not needed and may lead to a circular dependency. The cl removes old unused functionaliy to log BWE related statistics using compile time flags.
Bug: webrtc:42225697
Change-Id: I6cc01b367c0c48ab30f34c12a10afc58d1e7822f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352142
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42386}
Implements a two-pass approach to packetization which creates
packets of an even size similar to RtpPacketizer::SplitAboutEqually.
This improves the bandwidth estimation.
The algorithm does a first pass with the existing packetizer, then
iterates through the resulting packet sizes and sums up the bytes left unused in each packet.
It then calculates a new maximum packet length as
configured_max_packet_len - ((unused_bytes - packets + 1) / packets)
adjusts for the overhead and re-runs the packetization algorithm.
For example, a list of OBUs with sizes
{1206, 1476, 1431}
currently gets packetized greedily as payload sizes
{1200, 1200, 1200, 523}
With this change, it gets packetized as
{1032, 1032, 1032, 1028}
This change is guarded by the field trial
WebRTC-Video-AV1EvenPayloadSizes
which is acting as a rollout flag.
BUG=webrtc:15927
Co-authored-by: Shyam Sadhwani <shyamsadhwani@meta.com>
Change-Id: I4f0b3c27de6f06104908dd769c4dd1f34115712c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348100
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42203}
This seems to confuse perfetto, and the data ends up on its own track
and the end event is just ignored. As it was invalid, I am assuming it
is not used, and can be simply removed.
#rtc_fixit
Bug: webrtc:15867
Change-Id: I31a814f6c2147c3ce534726bf9046a79369b9eb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342761
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41896}
Removed thread safety: for a low level helper it adds overhead that users may not need. In particular RtpSenderVideo doesn't need it because calls to SendVideo are already synchronized.
Added a feature to force producing extension as requested by downstream.
Cleanup and document api:
Changed rtp_frequency type to int as it has no reason to use uint32_t per style guide
Changed absolute_capture_time to NtpTime to clarify both units and offset of the time. NtpTime has trivial conversion to/from uint64_t
Documented all the parameters.
Cleanup tests.
Bug: b/307553606
Change-Id: I0922ca4d3c89f124eeb561742dca79ed5c2327fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325022
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/main@{#41023}
since the extension can be too large to fit the 16 bytes available
to one-byte extensions
https://www.rfc-editor.org/rfc/rfc8285#section-4.2
when including the width and height fields.
Also document when those fields are sent.
BUG=webrtc:12000
Change-Id: If17f57d40c0bde9b060f223c548e407d6c124b82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40910}
Remove support for setting one limit without another limit
because related rtp header extension doesn't support such values.
Start morphing VideoPlayouDelay into a class and stricter type: add accessors returning TimeDelta
Bug: webrtc:13756
Change-Id: If0dd02620528dc870b015beeff3a8103e04022ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315921
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40570}
contributing sources are usually decided per packet, and thus having persistent member for csrcs makes them less natural to use.
Bug: None
Change-Id: I804d58ace574368f8cdd4356a15471110e530744
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291334
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40547}
CSRCs are decided on a per frame bases, thus keeping a constant copy of
csrcs inside the rtp sender transform delegate is confusing: when transform delegate is created, csrcs list is always empty.
Bug: None
Change-Id: Id94acc76857a47ad9a1dd8254648ab9cb5d6d31d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311840
Reviewed-by: Tony Herre <herre@google.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40533}
This CL cleans up all local conversions, in favor of the common helper
function.
Bug: webrtc:15210
Change-Id: Id77e1c6b1151a2542d92e220e91d5b11285479b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311060
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40420}
Currently FecController knows about network conditions, these information can be used to control RTX settings in-call.
Change-Id: I8f84164aeac48ea13b7f1cf82fd7424431f98ada
Bug: webrtc:15167
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304800
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40192}
encrypted_video_payload already allocates enough bytes - first SetSize query such size from the frame_encryptor_
Minimizing VP9 when generic descriptor is used might be harmful in multi-participant scenario where frame needs to be send to a participant without generic descriptor support and thus require complicated restoration of the VP9 specific descriptor.
No-Try: true
Bug: None
Change-Id: I5f2c32c2c9ae745794dfaaa4aec4c5898dff78f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293820
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39327}
Allow CSRCs to be modified per-frame in an Encoded Insertable Streams
transform, to support a web API which allows per-frame CSRC
modifications to signal when a JS application has changed the source
of the video which is written into an encoded frame.
Initially only for Video, with Audio support likely to follow later.
Bug: webrtc:14709
Change-Id: Ib34f35faa9cee56216b30eaae42d7e65c78bb9f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291324
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tove Petersson <tovep@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39214}
Instead of re-using the sender task queue, a new task queue will
suffice.
Bug: webrtc:14445
Change-Id: Ia7395ace2f0bb66bf9e76e3783b208f2cd0385dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275771
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38332}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
Sends a VideoLayersAllocation header extension if frame rate change more than 5fps since the last time it was sent with valid frame rate and resolution.
Bug: webrtc:12000
Change-Id: I2572c966025cc2c22743bbe2187cec7cceb86d01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234752
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35180}
The config flag will be removed once downstream usage is gone.
Bug: webrtc:11340
Change-Id: Iee8816660009211540d9b09bb3cba514455d709b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228431
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34757}
With this turned on, packets will be sequence number after the pacing
stage rather that during packetization.
This avoids a race where packets may be sent out of order, and paves
the way for the ability to cull packets from the pacer queue without
causing sequence number gaps.
For now, the feature is off by default. Follow-ups will enable it for
video and audio separately.
Bug: webrtc:11340, webrtc:12470
Change-Id: I6d411d8c85b9047e3e9b05ff4c2c3ed97c579aa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208584
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34661}
as there are encryption schemes that preserve the payload structure
well enough and do not require those extensions.
This improves consistency as the webrtc-encoded-transform API
(which does not use this synchronous codepath) does not require those
header extensions either.
BUG=webrtc:12995
Change-Id: If237ca5d92e8871ac71c3d48fdd05127206395e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226741
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34537}
This allows callers to use timestamps generated from their own clocks
without worrying about converting to webrtc time.
No-Try because of lack of infra lack of capacity on macs.
No-Try: True
Bug: webrtc:11327
Change-Id: I7b1935654a2b23cf844c7b3622ed68763ced9da5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219785
Commit-Queue: Paul Hallak <phallak@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34076}
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.
Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
This is a reland of 5cc99570620890edc3989b2cae1d1ee0669a021c
Original change's description:
> Batch assign RTP seq# for all packets of a frame.
>
> This avoids a potential race where other call sites could assign
> sequence numbers while the video frame is mid packetization - resulting
> in a non-contiguous video sequence.
>
> Avoiding the tight lock-unlock within the loop also couldn't hurt from
> a performance standpoint.
>
> Bug: webrtc:12448
> Change-Id: I6cc31c7743d2ca75caeaeffb98651a480dbe08e2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207867
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33291}
Bug: webrtc:12448
Change-Id: I7c5a5e00a5e08330ff24b58af9f090c327eeeaa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208221
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33296}
