FlexfecSender is owned and configured by VideoSendStream.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2501503003
Cr-Commit-Position: refs/heads/master@{#15082}
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}
The only difference is that the F and R bits have changed place.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2495253002
Cr-Commit-Position: refs/heads/master@{#15064}
Reason for revert:
Broke a lot of tests in chromium.webrtc browser_tests. See e.g. https://build.chromium.org/p/chromium.webrtc/builders/Mac%20Tester/builds/62228 and https://build.chromium.org/p/chromium.webrtc/builders/Win8%20Tester/builds/30102.
[ RUN ] WebRtcVideoQualityBrowserTests/WebRtcVideoQualityBrowserTest.MANUAL_TestVideoQualityH264/1
...
#
# Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\modules\rtp_rtcp\source\rtp_format_h264.cc, line 170
# last system error: 0
# Check failed: packetization_mode_ == kH264PacketizationMode1 (0 vs. 2)
#
Original issue's description:
> Implement H.264 packetization mode 0.
>
> This approach extends the H.264 specific information with
> a packetization mode enum.
>
> Status: Parameter is in code. No way to set it yet.
>
> Rebase of CL 2009213002
>
> BUG=600254
>
> Committed: https://crrev.com/3bba101f36483b8030a693dfbc93af736d1dba68
> Cr-Commit-Position: refs/heads/master@{#15032}
TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=600254
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2500743002
Cr-Commit-Position: refs/heads/master@{#15050}
In older Chrome versions, the associated payload type in the RTX header
of retransmitted packets was always set to be the original media payload type,
regardless of the actual payload type of the packet. This meant that packets
encapsulated with RED headers had incorrect payload type information in the
RTX header. Due to an assumption in the receiver, this incorrect payload type
information would effectively be undone, leading to a working system.
Albeit working, this behaviour was undesired, and thus removed. In the interim,
several workarounds were introduced to not destroy interop between old and
new Chrome versions:
(1) https://codereview.webrtc.org/1649493004
- If no payload type mapping existed for RED over RTX, the payload type
of the underlying media would be used.
- If RED had been negotiated, received RTX packets would always be
assumed to contain RED.
(2) https://codereview.webrtc.org/1964473002
- If RED was removed from the remote description answer, it would be
disabled in the local receiver as well.
(3) https://codereview.webrtc.org/2033763002
- If RED was negotiated in the SDP, it would always be used, regardless
if ULPFEC was negotiated and used, or not.
Since the Chrome versions that exhibited the original bug now are very old,
this CL removes the workarounds from (1) and (2). In particular, after this
change, we will have the following behaviour:
- We assume that a payload type mapping for RED over RTX always is set.
If this is not the case, the RTX packet is not sent.
- The associated payload type of received RTX packets will always be obeyed.
- The (non)-existence of RED in the remote description does not affect the
local receiver.
The workaround in (3) still needs to exist, in order to interop with receivers
that did not have the workarounds in (1) and (2) removed. The change in (3)
can be removed in a couple of Chrome versions.
TESTED=Using AppRTC between patched Chrome (connected to ethernet) and standard Chrome M54 (connected to lossy internal Google WiFi), with and without FEC turned off using AppRTC flag. Also using "Munge SDP" sample on patched Chrome over loopback interface, with 100ms delay and 5% packet loss simulated using tc.
BUG=webrtc:6650
Review-Url: https://codereview.webrtc.org/2469093003
Cr-Commit-Position: refs/heads/master@{#15038}
This approach extends the H.264 specific information with
a packetization mode enum.
Status: Parameter is in code. No way to set it yet.
Rebase of CL 2009213002
BUG=600254
Review-Url: https://codereview.webrtc.org/2337453002
Cr-Commit-Position: refs/heads/master@{#15032}
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}
Half of the function's code was reformatted and reindented in
https://codereview.webrtc.org/1307663004, but the bottom half was still
adhering to an old coding style and using different indentation values.
Not only does this make the code look confusing, but it can cause build
issues on certain compilers: for example, GCC 6.2.0 with -Wall causes
the build to fail because -Wmisleading-indentation is enabled.
BUG=None
R=asapersson@webrtc.org,danilchap@webrtc.org
Review-Url: https://codereview.webrtc.org/2479193002
Cr-Commit-Position: refs/heads/master@{#14957}
- 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}
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}
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}
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}
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}
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}
