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}
This reverts commit 5cc99570620890edc3989b2cae1d1ee0669a021c.
Reason for revert: Seems this CL breaks the below test when being imported in google3
https://webrtc-review.googlesource.com/c/src/+/207867
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: I2547f946a5ba75aa09cdbfd902157011425d1c30
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208220
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/master@{#33294}
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}
increase.
VP9 and other codecs can in theory add spatial layers without a key
frame.
Bug: webrtc:12000
Change-Id: I27461af2e34c855203a130e400a6aa01144d3cf7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198781
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32883}
This CL:
1) Updates RtpSenderVideo to actually populate the is_key_frame field
properly.
2) Updates UlpfecGenerator to:
* Allow updating the protection parameters before adding any packet.
* Apply keyframe protection parameter when at least one buffered
media packet to be protected belongs to a keyframe.
Updating the parameters in the middle of a frame is allowed, at that
point they only determine how many _complete_ frames are needed in order
to trigger FEC generation. Only that requirement is met, will the
protection parameters (e.g. FEC rate and mask type) actually be applied.
This means that delta-frames adjecent to a key-frame (either ahead of
or after) may be protected in the same way as the key-frame itself.
Bug: webrtc:11340
Change-Id: Ieb84d0ae46de01c17b4ef72251a4cb37814569da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195620
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32787}
This adds a method to allow VideoLayersAllocation to be sent using the header extension RtpVideoLayersAllocationExtension.
Bug: webrtc:12000
Change-Id: Iafdc1e16911c57ca55d7cc0559a0b45774211e92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187495
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32397}
Video playout delay is used to give a hint to the receiver
how the video should be played out.
Add the field trial WebRTC-ForceSendPlayoutDelay to set the video playout
delay of outgoing RTP packets to enable experimentation with this feature.
Bug: webrtc:11896
Change-Id: Ie6123b5967763bde6a830f4c5e5a963e73fb0acb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185042
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32169}
We can then finally delete the top-level common_types.h, and the
corresponding build target webrtc_common.
Bug: webrtc:7660
Change-Id: I1c1096541477586d90774c7a3405b9d36edec14a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182800
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32044}
