Before this CL for SVC streams (e.g VP9) still 3 separate RTP_RTCP senders
were created. The RTCP target bitrate messages were treated as simulcast
and were split and send for each separate spatial layer in a separate SSRC.
To fix that an svc flag is now wired to VideoSendStream config
and filled based on the encoder config in WebrtcVideoEngine. This flag is
used to differentiate between simulcast and SVC mode in RtpVideoSender.
Bug: webrtc:10485
Change-Id: Ifa01d12a7d4f01fcbe448ad11e0cc39ab2d1df55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129929
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27345}
The idea is to let the RtpRtcp and RTPSender classes be responsible for
media-agnostic RTP transport, and move out the media-specific processing,
such as packetization and media-specific headers.
Bug: webrtc:7135
Change-Id: Ib0ce45bf06713b3eb6c06acd91c5168856874e4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123187
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26954}
That is, the payload packetization overhead (eg. vp8 payload header),
not the RTP headers, extensions, etc.
The encoder and pacer both look at payload rate, but are currently not
aware of the bytes that are added in between them.
Bug: webrtc:10155
Change-Id: I4cdb04849d762360374d47a496983c8c6df191d2
Reviewed-on: https://webrtc-review.googlesource.com/c/115410
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26163}
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 is a follow up of https://webrtc-review.googlesource.com/c/src/+/43201.
Issue 43201 didn't do the job properly.
1. The audio rtcp report interval is not properly hooked up.
2. We don't need to propagate audio rtcp interval into video send stream or vice versa.
3. We don't need to propagate rtcp report interval to any receiving streams.
Bug: webrtc:8789
Change-Id: I1f637d6e5173608564ef0702d7eda6fc93b3200f
Reviewed-on: https://webrtc-review.googlesource.com/c/110105
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25610}
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 using the generic descriptor we want all simulcast streams to share one
frame id space (so that the SFU can switch stream without having to rewrite the
frame id). The state also needs to be restored when the RtpVideoSender is
recreated.
Note that |shared_simulcast_frame_id_| is only added, but not used in this CL.
Actually using it will be part of the next CL.
Bug: webrtc:9361
Change-Id: I7192a06d6ae4cab118ca5996ed99a56888ad1d97
Reviewed-on: https://webrtc-review.googlesource.com/92803
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24233}
