Prior to this change, FlexFEC packets that were paced would be lost in
the RTPSender, since they were not stored in a packet history. This CL
introduces such a packet history, as well as the needed wireup for
higher layers to be aware that the particular RTPSender is able to
send FlexFEC packets with a particular SSRC.
Updated RTPSender unit test to reflect the fact that paced packets
are now actually sent.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2491293002
Cr-Commit-Position: refs/heads/master@{#15066}
This is not the right place for a SequencedTaskChecker, as we can
not make any guarantees about the thread this method runs on.
We were hitting this check on Android and iOS whenever the encoder
would be reconfigured. Access to these ivars should be guarded
by a lock.
As a bonus, an unused method declaration was removed.
BUG=webrtc:6686
Review-Url: https://codereview.webrtc.org/2495483002
Cr-Commit-Position: refs/heads/master@{#15019}
This CL adds the ability for RTPSenderVideo to generate and send
FlexFEC packets.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2490523002
Cr-Commit-Position: refs/heads/master@{#15016}
- Change const ptr to const ref in parameter list.
Using nullptr as argument was invalid, so no need to send
pointer instead of reference.
- Change return type to void or bool, where appropriate
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2455963003
Cr-Commit-Position: refs/heads/master@{#14945}
Prior to this change, we signalled that ULPFEC was disabled
through a bool, but that RED was disabled by setting its
payload type to -1. The latter is consistent with how we
disable RED/ULPFEC in the config, so this CL removes the
ULPFEC bool from the {,Set}UlpfecConfig chain of member
functions.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2460533002
Cr-Commit-Position: refs/heads/master@{#14944}
At the same time, change to using int's instead of uint8_t's for the payload type.
This allows us to signal disabled FEC or RED using the sentinel value -1, which
is commonplace in other parts of the code.
These APIs will be deprecated when ULPFEC is deprecated.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2448463003
Cr-Commit-Position: refs/heads/master@{#14942}
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}
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}
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}
- Rename GetNumberOfFecPackets -> NumFecPackets and
PacketOverhead -> MaxPacketOverhead in ForwardErrorCorrection.
- Rename FECPacketOverhead -> FecPacketOverhead in ProducerFec.
- Move ownership of ForwardErrorCorrection from RTPSenderVideo
to ProducerFec.
- Make MaxPacketOverhead a member function of ForwardErrorCorrection.
This will allow for changing it, based on FEC header types, later on.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2275443002
Cr-Commit-Position: refs/heads/master@{#14194}
- Make more use of std::unique_ptr.
- Auto type deduction for iterator type names.
- More extensive comments.
- Variable renaming.
- Make ProducerFec::BuildRedPacket() static.
- Avoid dynamic allocation of ProducerFec::fec_.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2110763002
Cr-Commit-Position: refs/heads/master@{#13700}
double check rtp_sender in sending mode when altering sequence_number
adjust test to skip validating timestamp on rtx streams
fix test by waiting for all 3 media streams instead of 3 out 6 media and rtx streams.
BUG=webrtc:4332
Review-Url: https://codereview.webrtc.org/2177523002
Cr-Commit-Position: refs/heads/master@{#13587}
- Renamed variables and some function to comply with style guide.
- Removed default argument values.
- Removed some dead code.
- Cleaned up comments formatting in rtp_rtcp.h
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2067673004 .
Cr-Commit-Position: refs/heads/master@{#13565}
Reason for revert:
Breaks upstream code.
Original issue's description:
> Refactor NACK bitrate allocation
>
> Nack bitrate allocation should not be done on a per-rtp-module basis,
> but rather shared bitrate pool per call. This CL moves allocation to the
> pacer and cleans up a bunch if bitrate stats handling.
>
> BUG=
> R=danilchap@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
>
> Committed: 5fc59e810bTBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2131913003
Cr-Commit-Position: refs/heads/master@{#13417}
This CL adds support for an extension on RTP frames to allow the sender
to specify the minimum and maximum playout delay limits.
The receiver makes a best-effort attempt to keep the capture-to-render delay
within this range. This allows different types of application to specify
different end-to-end delay goals. For example gaming can support rendering
of frames as soon as received on receiver to minimize delay. A movie playback
application can specify a minimum playout delay to allow fixed buffering
in presence of network jitter.
There are no tests at this time and most of testing is done with chromium
webrtc prototype.
On chromoting performance tests, this extension helps bring down end-to-end
delay by about 150 ms on small frames.
BUG=webrtc:5895
Review-Url: https://codereview.webrtc.org/2007743003
Cr-Commit-Position: refs/heads/master@{#13059}
This shouldn't be needed, but because the receiver assumes RTX packets
contain RED if configured to receive them (due to an incompatibility
issue), we also have to make sure we send them for now.
BUG=webrtc:5675
Review-Url: https://codereview.webrtc.org/2033763002
Cr-Commit-Position: refs/heads/master@{#13024}
- First audio RTP packet sent / received
- First RTP packet of the first video frame sent / received
- Last RTP packet of the first video frame sent / received
These timestamps should make it easier to measure how fast the call
becomes established from the user's perspective.
Review URL: https://codereview.webrtc.org/1765443002
Cr-Commit-Position: refs/heads/master@{#12287}
There were two different structures named RtpPacket in webrtc namespace:
RtpPacket defined in fec_test_helper renamed to test::RawRtpPacket
RtpPacket defined in rtp_sender_video and producer_fec removed as unused
BUG=webrtc:5261
R=sprang@google.com, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1710103004 .
Cr-Commit-Position: refs/heads/master@{#11682}
TMMBN was capped by configured max bitrate for no apparent reason.
Removing this to not require payload-type reconfiguration on new
video-codec settings. Actual removal of payload-type reconfiguration
will happen in a pending CL.
BUG=webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1702043002 .
Cr-Commit-Position: refs/heads/master@{#11639}
Since the address of the dereference is taken this inputs a garbage
almost-null pointer into RtpPacketizer. Not likely that a load/store is
performed on the address, but UBSan fires and it's a source of potential
future errors.
BUG=webrtc:5124, webrtc:5490
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1677003002 .
Cr-Commit-Position: refs/heads/master@{#11528}
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.
BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1479023002 .
Cr-Commit-Position: refs/heads/master@{#10909}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
Since the pacer is always enabled, removing enable/disable which makes
all packet queueing succeed. Also renaming one of the ::SendPackets
::InsertPacket to avoid confusion.
BUG=webrtc:1695, webrtc:2629
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1392513002 .
Cr-Commit-Position: refs/heads/master@{#10211}
