With LTR and SVC etc., H.264 should be able to skip lost frames, and continue to play from the new frames. With DependencyDescriptor, it is allowed to reference the previous frames, even there is a gap in the middle. However, we found there is a special logic for H.264 in packet_buffer.cc, which requires no gap for H.264.
We should allow gaps if the packet has GenericDescriptorInfo (either GenericDescriptor or DependencyDescriptor header extension).
Bug: webrtc:14887
Change-Id: Id66726bab33229bd883f257136ff2e8523fb44c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294062
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39370}
Step 1 of combining the sender and receiver types
Also moved the RtpFrameObject to rtp_rtcp/source, as it's heavily used
by the transformable receiver frame, I couldn't work out a better way
of managing the dependencies, and everything else seemed to work fine.
Bug: chromium:1412687
Change-Id: I55e816a0d7aa2962560ff9ebaf30ad63ab0b9810
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291710
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39255}
This is in prep for the migration of all unwrappers to
SequenceNumberUnwrapper as a standard implementation.
This moves the SeqNumUnwapper to its own header and adds 2 methods to
SeqNumUnwrapper which are defined by other unwrappers:
* PeekUnwrap
* Reset
It also adds two implementations for RtpTimestamps and
RtpSequenceNumbers.
Bug: webrtc:13982
Change-Id: I5baefb2de1db92fe1bb600760bd63b71e9310eb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288742
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39030}
In H264, reordered packets can cause a frame following padding to become stuck in the packet buffer.
A minimal example:
_, P, 1 - padding packet p and frame 1. Frame 1 has not been returned because of missing packet 0
0, P, 1 - when packet 0 arrives, FindFrames will stop incrementing i when it sees padding packet P, and frame 1 will never be returned
Bug: webrtc:14216
Change-Id: I78b76df9709fa8593c5025d647e2b868af3749f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266465
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37357}
this is a step towards resolving own todo: making AssembleFrame part of
the VideoRtpDepacketizer interface and replacing codec check with a
call to a virtual function.
RtpVideoStreamReceiver has access to the VideoRtpDepacketizers,
PacketBuffer - hasn't.
Bug: None
Change-Id: I83df09975c092bdb71bab270ced356d79a50683d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168056
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30833}
to avoid expensive move of the Packet and prepare PacketBuffer
to return list of packets as a frame.
Bug: None
Change-Id: I19f0452c52238228bbe28284ebb197491eb2bf4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167063
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30404}
to return time of the last receieved packet of a key frame rather than
last received first packet of a key frame.
To match VideoReceiveStream expectation and prevent requesting
a new key frame if a large key frame is currently on the way.
Bug: None
Change-Id: I443a60872a3580d324f050080a9868f7b90d71a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161730
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30084}
Together with RtpDepacketizer refactoring that would reduce
number of memcpy while handling an rtp packet
Bug: webrtc:11152
Change-Id: I6f4e09c93af5e2a9314967a15eac8ced57ec712e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161087
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29985}
it is easier to reduce and eliminate it when it is not bound to legacy video code
Bug: webrtc:10979
Change-Id: I517e298501b3358a914a23ddce40fcb3075d672d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159707
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29821}
delete traces of the OnAssembledFrame callback
relax some expectation to better match test purpose,
in particular avoid verifying every test buffer is not cleared on new packet.
switch comparisons from EXPECT_EQ(constant, value) to more natural EXPECT_EQ(value, constant)
Bug: None
Change-Id: I81e2c9d0133221435cb2bb02c9190d9f32abd548
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158140
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29610}
Rename file with tests to match code under test.
Rename fixture by moving 'Test' from prefix to suffix
Bug: None
Change-Id: I54c36d3b517bde7cdffa3a7e74528cc464ea7ad7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157301
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29532}
2019-10-18 09:05:06 +00:00
Renamed from modules/video_coding/video_packet_buffer_unittest.cc (Browse further)
