Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.
BUG=chromium:621691
Committed: https://crrev.com/c681250aaa2025836db7669694e323898e5c2ca7
Review-Url: https://codereview.webrtc.org/2405173006
Cr-Original-Commit-Position: refs/heads/master@{#14923}
Cr-Commit-Position: refs/heads/master@{#14935}
Reason for revert:
Still breaks internal downstream project.
Sergey: Please update internal project before relanding this.
Original issue's description:
> Remove deprected functions from EncodedImageCallback and RtpRtcp
>
> Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> These methods should no longer be used anywhere and it's safe to remove
> them.
>
> BUG=chromium:621691
>
> Committed: https://crrev.com/c681250aaa2025836db7669694e323898e5c2ca7
> Cr-Commit-Position: refs/heads/master@{#14923}
TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2479643002
Cr-Commit-Position: refs/heads/master@{#14925}
Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2405173006
Cr-Commit-Position: refs/heads/master@{#14923}
This class will interface RTPSenderVideo with the underlying
erasure code. It is functionally similar to ProducerFec
(to be renamed UlpfecGenerator). In fact, the FlexfecSender is a
friend of ProducerFec, and reuses most of its implementation.
Besides the fact that FlexfecSender outputs FlexFEC packets,
the main difference with ProducerFec is that FlexfecSender
allocates RTP sequence numbers, whereas ProducerFec does not
do this for the RED-encapsulated ULPFEC packets.
This class is split as interface/implementation, since it will
be owned by VideoSendStream initially. Further along, it may be
owned by PacedSender.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2441613002
Cr-Commit-Position: refs/heads/master@{#14922}
There is no need for it to be an interface.
In this CL, I also took the opportunity to make two small fixes:
- remove the 'flexfec_' prefix from some member variables
- remove unnecessary use of a stringstream object
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2471073003
Cr-Commit-Position: refs/heads/master@{#14919}
Reason for revert:
Breaks everything
Original issue's description:
> Revert of Remove deprected functions from EncodedImageCallback and RtpRtcp (patchset #4 id:100001 of https://codereview.webrtc.org/2405173006/ )
>
> Reason for revert:
> This might be breaking projects downstream.
>
> Original issue's description:
> > Remove deprected functions from EncodedImageCallback and RtpRtcp
> >
> > Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> > These methods should no longer be used anywhere and it's safe to remove
> > them.
> >
> > BUG=chromium:621691
> >
> > Committed: https://crrev.com/fa565842718ad178a7562721b25d916fbabc2b92
> > Cr-Commit-Position: refs/heads/master@{#14902}
>
> TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:621691
>
> Committed: https://crrev.com/6c78307a21252c2dbd704f6d5e92a220fb722ed4
> Cr-Commit-Position: refs/heads/master@{#14914}
TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2467373003
Cr-Commit-Position: refs/heads/master@{#14915}
Reason for revert:
This might be breaking projects downstream.
Original issue's description:
> Remove deprected functions from EncodedImageCallback and RtpRtcp
>
> Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> These methods should no longer be used anywhere and it's safe to remove
> them.
>
> BUG=chromium:621691
>
> Committed: https://crrev.com/fa565842718ad178a7562721b25d916fbabc2b92
> Cr-Commit-Position: refs/heads/master@{#14902}
TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2474433008
Cr-Commit-Position: refs/heads/master@{#14914}
The H264SpsPpsTracker class:
- Keeps track of all received SPS/PPS.
- Decides whether a packet should be inserted into the PacketBuffer or not.
- Don't insert if this packet only contains SPS and/or PPS.
- Don't insert if this is the first packet of and IDR and we have not
received the required SPS/PPS.
- Insert start codes, and in the case of the first packet of an IDR prepend
the bitstream with the given SPS/PPS for this IDR.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2466993003
Cr-Commit-Position: refs/heads/master@{#14906}
Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2405173006
Cr-Commit-Position: refs/heads/master@{#14902}
- Add histogram: "WebRTC.Video.RtpToNtpFreqOffsetInKhz"
The absolute value of the difference between the estimated frequency during RTP timestamp to NTP time conversion and the actual value (i.e. 90 kHz) is measured per received video frame. The max offset during 40 second intervals is stored. The average of these stored offsets per received video stream is recorded when a stream is removed.
Updated rtp_to_ntp.cc:
- Add validation for only inserting newer RTCP sender reports to the rtcp list.
- Move calculation of frequency/offset (from RTP/NTP timestamp pairs) to UpdateRtcpList. Calculated when a new RTCP SR in inserted (and not in RtpToNtpMs per packet).
BUG=webrtc:6579
Review-Url: https://codereview.webrtc.org/2385763002
Cr-Commit-Position: refs/heads/master@{#14891}
This is a proposal for a new RTCP message. Feel free to comment on the
message structure, selected type ids etc, as well as code for
serialization/deserialization. Once we agree on this, I'll continue
with wiring it up in the actual rtcp sender and receiver.
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2306873003
Cr-Commit-Position: refs/heads/master@{#14867}
Design of individual block in ExtendedReports packet suggest there is
no point to have more than one block per type.
This CL reduce complexity of having several blocks of the same type in
same report.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2378113002
Cr-Commit-Position: refs/heads/master@{#14855}
renamed kName to kUri and make it more const.
remove IsSupportedBy to reduce header dependency.
BUG=webrtc:1994
Review-Url: https://codereview.webrtc.org/2457783005
Cr-Commit-Position: refs/heads/master@{#14825}
Include CVO in key frame.
Include CVO in delta frame when rotation changes.
Include CVO when it is non zero to support current receiver implementation.
BUG=webrtc:6600
Review-Url: https://codereview.webrtc.org/2452583002
Cr-Commit-Position: refs/heads/master@{#14784}
Rename kFullSendSideEstimator -> kSendSideEstimator and add new class SendSideBweSender (controlled by kSendSideEstimator) that actually uses the send side BWE.
Move the mock to logging/rtc_event_log/mock.
Allow congestion_controller, remote_bitrate_estimator and audio to depend on loggging/rtc_event_log
BUG=webrtc:6526
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2431093003
Cr-Commit-Position: refs/heads/master@{#14772}
Remove functions to enumerate all extensions,
Remove concept of the inactive extension.
Decision if extension should be included into rtp header is done by rtp_sender
GetTotalLengthInBytes now calculates all extension, included or not.
That is used only for calculating how much space to reserve for fec.
Since extension might suddenly be included in the next packet (which still might belong to same fec group), it is safer to calculate all registered extension.
BUG=webrtc:5565, webrtc:1994
Review-Url: https://codereview.webrtc.org/2431253003
Cr-Commit-Position: refs/heads/master@{#14763}
Separate the null implementation from rtp_rtcp_defines.h, and follow chromium style guide for virtual functions.
BUG=webrtc:163
Review-Url: https://codereview.webrtc.org/2400993002
Cr-Commit-Position: refs/heads/master@{#14738}
for consistency with other rtcp packet classes.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2361853002
Cr-Commit-Position: refs/heads/master@{#14648}
and thus IP_PACKET_SIZE constant:
Build() use BlockLength() instead of constant IP_PACKET_SIZE for packet
capacity, adding extra checks about packet generation in tests.
Build(callback) removed as unused.
definitions reordered to follow style.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2270753002
Cr-Commit-Position: refs/heads/master@{#14647}
When the FlexfecReceiver recovers media packets, it inserts these into
internal::Call, which then distributes them to the appropriate
VideoReceiveStream/RtpStreamReceiver.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2390823009
Cr-Commit-Position: refs/heads/master@{#14642}
This class is split in interface/implementation classes, since it
will be referenced from the Call level. Its purpose is to interface
the erasure code decoder with a new class FlexfecReceiveStream
(for received packets), as well as with the main RTP pipeline (for
recovered packets).
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2392663006
Cr-Commit-Position: refs/heads/master@{#14594}
The renaming is to reflect this class is only used for RTCP interaction
and not for other transports.
This Cl will be followed by multiple CLs moving all send-side RTP
functionality to a separate class, rtp module ownership away from
VideoSendStream and use TaskQueue instead of ProcessThread for RTP.
BUG=webrtc:6456
Review-Url: https://codereview.webrtc.org/2390463002
Cr-Commit-Position: refs/heads/master@{#14556}
CheckPayloadChanged.
Removed last_received_frequency_, cng_payload_type_,
g722_payload_type_ and last_received_g722_ from RTPReceiverAudio and
cleaned up most of the related, now dead code.
Since g722_payload_type_ was never set, neither was
last_received_g722_, which means the frequency change in
CNGPayloadType was never done. Setting the frequency to the standard
values also proved unnecessary, since they were already set before the
call. Even if frequency would have been changed by RTPReceiverAudio, I
was not able to find a place where that would actually have
mattered. The ACM and NetEq, for example, which eventually gets these
packages, don't care about that value.
Also, GetPayloadTypeFrequency was never called, so keeping track of
last_received_frequency_ proved unnecessary.
cng_payload_type_ was stored to be able to check in CNGPayloadType if
cng_payload_type_has_changed. This flag was also never read, so these
all disappear.
The main reason for starting this change was to root out any G722
specific code we have sprinkled around the code base (specifically
dealing with the fact that for G722 clock rate != sample rate). In
this case, once I started pulling at one end of the string, the whole
thing came unraveled.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2383103002
Cr-Commit-Position: refs/heads/master@{#14530}
I'll be doing some changes to code it tests (rtp_receiver_audio,
specifically) and want to make sure there are tests in place before I
touch anything.
Fixed test_api_audio not properly checking payload data. Required a
fix to LoopBackTransport in test_api to as to act like the regular
audio and video parts of WebRTC and separate payload from header data.
Also added a test for CNG and cleaned up constants.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2378403004
Cr-Commit-Position: refs/heads/master@{#14529}
This CL introduces changes that clearly demarcate
where we disable Unequal Protection in the FEC.
No functional changes are expected.
BUG=webrtc:5654
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2314743002 .
Cr-Commit-Position: refs/heads/master@{#14496}
- Change some member functions to be private. These were only
called by other private member functions.
- Replace DeleteMediaPackets() with direct calls to
media_packets_.clear()
- Rename GetFecPacketsAsRed to GetUlpfecPacketsAsRed.
No functional changes are intended by this CL.
BUG=webrtc:5654
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2305793003 .
Cr-Commit-Position: refs/heads/master@{#14491}
- Place all member function definitions between test fixture
declaration and unit tests.
- Rename GenerateFrame -> PacketizeFrame.
No functional changes are intended by this CL.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2275303003
Cr-Commit-Position: refs/heads/master@{#14488}
Helper class for FlexFEC unit tests to come.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2282473002
Cr-Commit-Position: refs/heads/master@{#14487}
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.
After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).
The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).
This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.
BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2380683005 .
Cr-Commit-Position: refs/heads/master@{#14485}
Main changes:
- Split out general functionality from UlpfecPacketGenerator
into a new class AugmentedPacketGenerator. This will allow
for the addition of a FlexfecPacketGenerator that inherits
from AugmentedPacketGenerator.
- Rename RawRtpPacket to AugmentedPacket. This name is more
reflective of its purpose, i.e., an FEC packet with an
additional WebRtcRTPHeader.
- Return std::unique_ptr's instead of raw pointers.
Minor changes:
- Update names, based on RawRtpPacket -> AugmentedPacket name
change, in FEC unit tests.
- Rename |generator_| to |packet_generator_|, in FEC unit tests.
- Change some int's to size_t's in the packet generator classes.
- Use std::unique_ptr in ProducerFec unittest.
No functionality or performance changes are expected
due to this CL.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2271273004
Cr-Commit-Position: refs/heads/master@{#14477}
- Change some types to size_t.
- Update parameter ordering.
- Run 'git cl format'
- Moved 'using declarations' into unnamed namespaces.
- Removed "::webrtc::" prefix from some using declarations.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2273353002
Cr-Commit-Position: refs/heads/master@{#14475}
