Currently, if LNTF and NACK messages are both created, they will
be sent out in separate RTCP messages. This is wasteful.
This CL is the first of in a series of CLs that will ensure that
these feedback messages can be buffered together, without introducing
more of a delay than the CPU time required to process both messages.
Bug: webrtc:10336
Change-Id: I950324112ee346695a12a17d025483ea5e99c732
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139112
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28136}
This is a partial revert of
https://webrtc-review.googlesource.com/c/src/+/130101.
The KeyFrameRequestSender argument is added back to the constructor of
RtpVideoStreamReceiver. It is optional; if a null pointer is passed,
key frame requests are sent via the internal RtpRtcp module, and this is
how the class is used by VideoReceiveStream. An injectable
KeyFrameRequestSender is useful for tests, for downstream applications
that want to route key frame requests elsewhere, and should also aid
later migration to RtcpTransciever.
Bug: None
Change-Id: Idf9baeed21570625ad74e9afbe38f7ea5bf79feb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139107
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28102}
LNTF (loss notifications) are no longer mutually exclusive with
NACK and key frames; a receiver may send both, and the sender would
be allowed to choose which to regard.
Bug: webrtc:10336
Change-Id: I1ae7d972f9f47b07fe2f493bb31c1916456d6a3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138828
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28081}
RTP timestamp was recently added to contributing sources in the WebRTC
specification. This CL implements that change in WebRTC.
Bug: webrtc:10650
Change-Id: Ic0ccfbea7049a5b66063fa6cf60d01d5bd713132
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137515
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28020}
RtpVideoStreamReceiver used to pass the PacketRouter when creating its
RtpRtcp module, but it's not needed for a receive-only module. Make the
PacketRouter optional to the constructor; it's used only for registering
the created RtpRtcp module as a candidate for sending rtcp feedback.
Bug: None
Change-Id: I371a0bdb9d68ac48b16f52e1d7939f8c177dc528
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137429
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27984}
The latter is also a member of the former. This cleanup is also
a preparation for dropping WebRtcRTPHeader::frameType (or deleting
WebRtcRTPHeader right away), now that it's a video-specific member.
Tbr: kwiberg@webrtc.org # Comment change in modules/include/
Bug: None
Change-Id: I5c1f3f981f0d750713fc9b9b145278150fe32b5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133024
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27740}
Color space should only be transmitted in the last packet of a key frame,
therefore, neglect it otherwise so that |last_color_space_| is not reset by
mistake.
Bug: webrtc:10543
Change-Id: I374f9e52739292b18f510cc2941666fe6ba6951e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132553
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27717}
This change refactors the FrameDecryptorInterface to use the new API. The new
API surface simply moves bytes_written to the return type and implements a
simple Status type.
Bug: webrtc:10512
Change-Id: I622c5d344d58e618853c94c2f691cf7c8fb73a36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131460
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27497}
This change introduces new logic to allow the injection of the FrameDecryptor
into an arbitrary already running VideoReceiveStream without resetting it. It
does this by taking advantage of the BufferedFrameDecryptor which will
forcefully be created regardless of whether a FrameDecryptor is passed in
during construction of the VideoReceiver if the
crypto_option.require_frame_encryption is true. By allowing the
BufferedFrameDecryptor to swap out which FrameDecryptor it uses this allows the
Receiver to switch decryptors without resetting the stream.
This is intended to mostly be used when you set your FrameDecryptor at a point
post creation for the first time.
Bug: webrtc:10416
Change-Id: If656b2acc447e2e77537cfa394729e5c3a8b660a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130361
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27458}
VCMReceiveStatisticsCallback originates in the old jitter buffer, and
is no longer used.
VCMFrameTypeCallback originates in VideoReceiver::RequestKeyFrame,
which is called from OncomingPacket, Process, Decode(uint16_t
maxWaitTimeMs), all of which are unused by VideoReceiveStream.
So delete the code to wire them up via VideoStreamDecoder.
Bug: webrtc:7408
Change-Id: I173bc94eb32f2641f943c125083db038c3bcaeb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128870
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27277}
to clearly signal passed ownership.
Drop support for accepting nullptr clock to avoid copying the Configuration structure.
Update all calls in webrtc to the new factory function
Bug: None
Change-Id: Ic5a78da8e59ba3988a757a9d9634fa31499ce0db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125901
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26994}
The current Key Frame request system doesn't take into account failed
decryptions and this can lead to WebRTC spamming new key frame requests when
the issue is actually in the decryptor layer. To prevent this if frame
decryption is required for the PeerConnection key frame requests will not be
sent at 200ms intervals but will wait until the stream is decryptable before
utilizing this logic.
Bug: webrtc:10330
Change-Id: I188a21dfd142dec6175d9def95f39a2bc517017c
Reviewed-on: https://webrtc-review.googlesource.com/c/123414
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26931}
* LossNotificationController is the class that decides when to issue
LossNotification RTCP messages.
* RtpRtcp handles the technicalities of producing RTCP messages.
Bug: webrtc:10336
Change-Id: I292536257a984ca85d21d9cfa38e7ff2569cbb39
Reviewed-on: https://webrtc-review.googlesource.com/c/124123
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26840}
The discardability flag denotes whether the frame may be dropped by
the decoder with no effect on the decodability of subsequent frames.
Bug: webrtc:10214
Change-Id: I3654951d8863b50effe9670b8d1d7eb051240039
Reviewed-on: https://webrtc-review.googlesource.com/c/122241
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26763}
- Enable vp9 flexible mode in VideoEngine if 3 spatial layers are set.
- Enable flexible mode in loopback tools and quality tests.
- Reset first active spatial layer on keyframe in encoder.
- Ensure duplicate references are not set by the sender in video header.
- Set references manually for flexible mode in vp9 encoder.
- Delay new activated layers until next base layer frame.
- On receive side put each spatial layer as a separate frame to FrameBuffer
and return several frames combined from FrameBuffer.
Bug: webrtc:10049,webrtc:9794,webrtc:9784
Change-Id: I01e69f134cc145deba666ccc92deb1d37a324ede
Reviewed-on: https://webrtc-review.googlesource.com/c/112289
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25895}
This change introduces a new class BufferedFrameDecryptor that is responsible
for decrypting received encrypted frames and passing them on to the
RtpReferenceFinder. This decoupling refactoring was triggered by a new
optimization also introduced in this patch to stash a small number of
undecryptable frames if no frames have ever been decrypted. The goal of this
optimization is to prevent re-fectching of key frames on low bandwidth networks
simply because the key to decrypt them had not arrived yet.
The optimization will stash 24 frames (about 1 second of video) in a ring buffer
and will attempt to re-decrypt previously received frames on the first valid
decryption. This allows the decoder to receive the key frame without having
to request due to short key delivery latencies. In testing this is actually hit
quite often and saves an entire RTT which can be up to 200ms on a bad network.
As the scope of frame encryption increases in WebRTC and has more specialized
optimizations that do not apply to the general flow it makes sense to move it
to a more explicit bump in the stack protocol that is decoupled from the WebRTC
main flow, similar to how SRTP is utilized with srtp_protect and srtp_unprotect.
One advantage of this approach is the BufferedFrameDecryptor isn't even
constructed if FrameEncryption is not in use.
I have decided against merging the RtpReferenceFinder and EncryptedFrame stash
because it introduced a lot of complexity around the mixed scenario where some
of the frames in the stash are encrypted and others are not. In this case we
would need to mark certain frames as decrypted which appeared to introduce more
complexity than this simple decoupling.
Bug: webrtc:10022
Change-Id: Iab74f7b7d25ef1cdd15c4a76b5daae1cfa24932c
Reviewed-on: https://webrtc-review.googlesource.com/c/112221
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25865}
Deleted from subclass video_coding::EncodedFrame. Also delete Length
and SetLength methods on the intermediate class
video_coding::VCMEncodedFrame.
Bug: webrtc:9378
Change-Id: I3c90b14735f622f50b2f403f79072e22fc025d11
Reviewed-on: https://webrtc-review.googlesource.com/c/112131
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25838}
There have been several bugs where the members of PlayoutDelay were
zero initialized when handling RTP packets without the corresponding
extensions. Initializing to {-1, -1} (meaning not provided) is less
brittle.
Bug: None
Change-Id: I196850377128d5e67a19bdaf9298403b2e9f5a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/111181
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25670}
This patch adds (optional) csrc to ContributingSources.
This will be used if using virtual audio ssrc, since
the audio level is otherwise unaccessible in that configuration.
BUG=webrtc:3333
Change-Id: Ied263b8f0850553cd637fd6bead373ed4252fd1e
Reviewed-on: https://webrtc-review.googlesource.com/c/109281
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25516}
When FlexFEC is enabled, sometimes media packet will be recovered by FEC before the actual media packet's arrival. In current implementation this will be considered as packet out of order and nack will be sent, thus cause large increase in retransmit bitrate.
This fix:
1. Avoid sending nack for packet out of order caused by "early" recovered media packets.
2. Save recovered media packet in a set, and do not send nack for these packets.
Bug: None
Change-Id: I008ef4e33668bce6d2cb9ff52b4b5c8e3f349965
Reviewed-on: https://webrtc-review.googlesource.com/c/108090
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25444}
This change enables the FrameDecryptor attached to an RtpVideoReceiver to do
an initial request for a KeyFrame if the first successfully decrypted payload
is not a key frame.
Bug: webrtc:9795
Change-Id: I401ce1f513cb51ce520b60dcaf8b825a68d00c7f
Reviewed-on: https://webrtc-review.googlesource.com/c/107246
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25295}
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 CL makes the RtpGenericFrameDescriptor available in
RTPSenderVideo::SendVideo for encryption and in
RtpVideoStreamReceiver::OnReceivedFrame for decryption.
Bug: webrtc:9361
Change-Id: I5b6d10138c0874657862f103c8c9a2328e6d4a66
Reviewed-on: https://webrtc-review.googlesource.com/102720
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24929}
Add new method OnRtpPacket, but leave
ReceiveStatisticsImpl::IncomingPacket and most of the implementation
unchanged. Deleting the old method and converting implementation from
RTPHeader to RtpPacketreceived is planned for a followup, after
downstream code is updated.
Bug: webrtc:7135, webrtc:8016
Change-Id: I697ec12804618859f8d69415622d1b957e1d0847
Reviewed-on: https://webrtc-review.googlesource.com/100104
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24889}
