SSRCs are specified twice in calls to the RtpVideoSender constructor.
Once in the first argument of ssrcs, and then again in the RtpConfig
ssrcs variable. Resolving to reference the variable in the RtpConfig.
Bug: None
TBR: stefan@webrtc.org
Change-Id: I53528140166a53f3558f950d5662b7d3d6b8c822
Reviewed-on: https://webrtc-review.googlesource.com/c/114910
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26094}
This prepares for moving BitrateAllocationUpdate to API.
Bug: webrtc:9718
Change-Id: Ib2bcedb6b68fde33b6a2466f40829e86438aa973
Reviewed-on: https://webrtc-review.googlesource.com/c/111507
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25737}
This prepares for adding parameters to OnBitrateUpdated. By using a
struct, additional fields doesn't require a change in the signature and
only the obeservers that use the new fields will be affected by the
change.
Bug: webrtc:9718
Change-Id: I7dd6c9577afd77af06da5f56aea312356f80f9c0
Reviewed-on: https://webrtc-review.googlesource.com/c/107727
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25366}
This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into
the video send and receive path. If a FrameEncryptorInterface is set on an outgoing video RTPSender
then each outgoing video frame will first pass through the provided FrameEncryptor which
will have a chance to modify the payload contents for the purposes of encryption. In addition to
this the new GenericFrameDescriptor will be added as additional data.
If a FrameDecryptorInterface is set on an incoming video RtpReceiver then each incoming
video payload will first pass through the provided FrameDecryptor which have a chance to
modify the payload contents for the purpose of decryption.
Bug: webrtc:9795
Change-Id: I9f743ce0cb63df0cf070f6144be7ada078b4e5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/103920
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25258}
This also moves the packet feedback tracking to RtpVideoSender.
Bug: webrtc:9517
Change-Id: Ifb1ff85051730108a0b0d1dd30f6f8595ad2af6e
Reviewed-on: https://webrtc-review.googlesource.com/c/95920
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25019}
When the bandwidth estimate is volatile, and the frame rate is high,
each new frame might trigger a new video bitrate allocation that is very
close to the previous one, during BWE rampup.
This might cause unnecessarily high RTCP traffic.
This CL throttles those updates, if the allocation fullfills all of:
* Larger or the same total bitrate as the previously sent one
* Less than 10% larger bitrate compared to the previous one
* Same layers enables as the previous one
* Less than 500ms has passed since the previous one
Additionally, a call to OnEncodedImage can cause a throttled allocation
to be sent if 500ms has passed but no new call to OnBitrateUpdated has
been seen.
Bug: webrtc:9734
Change-Id: I2a17c2e512387e273e9c22bffcebd290727dc883
Reviewed-on: https://webrtc-review.googlesource.com/100560
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24751}
Especially for simulcast screensharing, we don't want to send constant
high bitrates of padding just to keep the bwe up since ALR probing
already handles that case.
Bug: webrtc:9734
Change-Id: I79a08fc073844628d8ad0561edd8bfcffed83fde
Reviewed-on: https://webrtc-review.googlesource.com/99120
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24734}
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
