This is the first change in a series of changes to get new functionality
into the VP8 packetizer.
This first refactors the RtpFormatVp8Test class, without changing the
operation of the tested RtpFormatVp8 class. A test helper class
RtpFormatVp8TestHelper is introduced to reduce code duplication.
Review URL: http://webrtc-codereview.appspot.com/304009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1258 4adac7df-926f-26a2-2b94-8c16560cd09d
This change avoids having old packets end up on the current packet list for FEC decoding, and so they are immediately sent out to jitter buffer.
The current list of packets for FEC decoding are sent out only when new packet arrives (with time-stamp greater than current).
Review URL: http://webrtc-codereview.appspot.com/322009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1222 4adac7df-926f-26a2-2b94-8c16560cd09d
Updated RtpFormatVp8, ModuleRTPUtility, VP8Encoder and VP8Decoder
to support a new LayerSync ("Y") bit. Note, in VP8Encoder the bit
must be used together with a non-negative value for temporalIdx.
Fixing the plumbing between RTP module and and from VP8 wrapper.
Updating unit tests; all pass.
The new bit is yet to be used by the VP8 wrapper.
Review URL: http://webrtc-codereview.appspot.com/323008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1169 4adac7df-926f-26a2-2b94-8c16560cd09d
A set of the packet masks (up 10x10 size) are modified for the following reasons:
1) have more even column and row degree (number of 1 bits), when possible.
2) if cases where the column degree cannot be constant across source packets, placed the extra 1 bit in the first packet column (so little more protection on 1st partition), as opposed to having some ~middle source packet have the extra bit.
3) in some cases, made the mask a little more sparse/reduced the overlap.
Overall the average recovery is a little better with these masks.
Mask sizes above 10 will be updated in future changelist.
Review URL: http://webrtc-codereview.appspot.com/305001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1113 4adac7df-926f-26a2-2b94-8c16560cd09d
A new RtpRtcpClock interface has been added to rtp_rtcp_defines.h
and provides time facilities used by an RTP/RTCP module. Also,
NTP constants have been made public in the
webrtc::ModuleRTPUtility namespace to make implementation of
external clocks easier.
An overloaded version of CreateRtpRtcp() accepts a clock argument. By
default, if no clock is provided, the module uses the system clock
(old ModuleRTPUtility implementation).
Throughout the RTP/RTCP module code, calls to TickTime and
ModuleRTPUtility time functions have been replaced with calls to time
methods on a clock object.
The following classes take a clock object in their constructor and
hold a _clock field (either directly, or inherited from a parent):
Bitrate
ModuleRtpRtcpImpl
RTCPReceiver
RTCPSender
RTPReceiver
RTPSender
RTPSenderAudio
RTPSenderVideo
Methods from other classes that do not derive any of those and
require a time take an additional nowMS parameter, that should be
the result of calling GetTimeInMS() on a clock object.
Review URL: http://webrtc-codereview.appspot.com/268017
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1076 4adac7df-926f-26a2-2b94-8c16560cd09d
Updating the VP8 packetizer class (RtpFormatVp8) and VP8 parser
(in class RTPPayloadParser) to follow the -02 revision of the spec.
See http://tools.ietf.org/html/draft-ietf-payload-vp8-02.
Updating the unit tests, too. Finally, updating the tests to
follow the recommendations from the test team; specifically
including the test code in the webrtc namespace, and omitting
the main function at the end of each test file.
Review URL: http://webrtc-codereview.appspot.com/296003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1013 4adac7df-926f-26a2-2b94-8c16560cd09d
Changing how the max payload length is calculated. Instead
of handling RTP and FEC header overhead explicitly, call the
MaxDataPayloadLength method which already does it. Avoid redundant code. Had to move MaxDataPayloadLength to the
RTPSenderInterface.
Review URL: http://webrtc-codereview.appspot.com/269002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@901 4adac7df-926f-26a2-2b94-8c16560cd09d
Issues solved:
1. Possible overflow when reducing the bandwidth estimate at the send-side
2. A burst of loss reports could make us reduce the rate way too far since
we reduced the rate relative the current estimate and not the actual
rate sent.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/244011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@822 4adac7df-926f-26a2-2b94-8c16560cd09d
The VP8 decoder wrapper will request key frames 30 frames after seeing
a packet loss, if it hasn't received a state refresh (only possible
through key frames in this version).
For this to be possible the jitter buffer has been made aware of
picture ids to be able to detect frame losses. Legacy JB code to
handle streams without marker bits was also removed since it
conflicts with streams with FEC.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/239002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@774 4adac7df-926f-26a2-2b94-8c16560cd09d
