Return value always passed as the |retransmitted| argument to
ReceiveStatistics::IncomingPacket. The implementation of this method,
StreamStatisticianImpl::IncomingPacket, can call its own
IsRetransmitOfOldPacket, which is demoted to a private method.
Bug: webrtc:7135
Change-Id: I904db676738689c7a1db4caa588f70e64e3c357d
Reviewed-on: https://webrtc-review.googlesource.com/95649
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24494}
This reverts commit 84916937b70472715efe5682bc273e91c3a72695.
Reason for revert: breaking downstream projects.
Original change's description:
> Update packetsLost and jitter stats any time a packet is received.
>
> Before this CL, the packetsLost and jitter stats (as returned by
> GetStats, at the API level) were only being updated when an RTCP SR or
> RR is generated. According to the stats spec, "local" stats like this
> should be updated any time a packet is received.
>
> This CL also fixes some minor issues with the calculation of packetsLost
> (and fractionLost):
> * Packets weren't being count as lost if lost over a sequence number
> rollover.
> * Temporary periods of "negative" loss (caused by duplicate or out of
> order packets) weren't being accumulated into the cumulative loss
> counter. Example:
> Period 1: Received packets 1, 2, 4
> Loss over that period: 1 (expected 4 packets, got 3)
> Reported cumulative loss: 1
> Period 2: Received packets 3, 5
> Loss over that period: -1 (expected 1 packet, got 2)
> Reported cumulative loss: 1 (should be 0!)
>
> Landing with NOTRY because Android compile bots are broken for an
> unrelated reason.
> NOTRY=True
>
> Bug: webrtc:8804
> Change-Id: I840ba34de8957b1276f6bdaf93718f805629f5c8
> Reviewed-on: https://webrtc-review.googlesource.com/50020
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23731}
TBR=danilchap@webrtc.org,deadbeef@webrtc.org,ossu@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Landing with NOTRY because ios64_sim_ios10_dbg bot is broken.
Passing all other bots.
NOTRY=True
Bug: webrtc:8804
Change-Id: I07bd6b1206d5a8d211792ad392842f9ed6c505e9
Reviewed-on: https://webrtc-review.googlesource.com/95280
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24370}
This is a preparation for extracting CSRC book-keeping to its own
class.
Bug: webrtc:7135
Change-Id: Ic51ceb57ec53a43064a3d0392de8baa978a4e8cf
Reviewed-on: https://webrtc-review.googlesource.com/93463
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24257}
Also delete related code in RtpReceiverAudio, RtpReceiverVideo and
RtpPayloadRegistry.
Only intended change in behavior is that packets with unknown payload
types are not discarded at this level of the stack. They are discarded
higher up, in Channel::ReceivePacket (audio) and
RtpVideoStreamReceiver::ReceivePacket (video).
Bug: webrtc:8995
Change-Id: I807997120bb40a95b0575c55db6e20a0cac651bf
Reviewed-on: https://webrtc-review.googlesource.com/92087
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24196}
All users call SetTelephoneEventForwardToDecoder(true). Setting the
flag to true on construction, enables deletion of those calls,
followed by deletion of the flag itself.
The unused getter method TelephoneEventForwardToDecoder() is deleted
right away.
Bug: webrtc:7135
Change-Id: I8c52c957b3f074be7ffc425b3588402d1e42b844
Reviewed-on: https://webrtc-review.googlesource.com/90402
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24141}
This prevents a lot of unnecessary processing taking place when we are
not using FEC.
This CL also removes the FieldTrial that was used to disable ulpfec, as it's no longer used.
Bug: webrtc:9514
Change-Id: I8285b933f71eea971f5932cd19833455a42c8639
Reviewed-on: https://webrtc-review.googlesource.com/87848
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23952}
this function is now only used in combination with StreamStatistician::IsRetransmitOfOldPacket
but IsRetransmitOfOldPacket internally checks if packet is in_order, thus making extra check unnecessary
In addition to making code simpler, removing this checks avoids
taking two extra CritSection on common code path of incoming rtp packet.
Bug: webrtc:8016
Change-Id: I050004e256b5698ce700e3416aa86b55f446a270
Reviewed-on: https://webrtc-review.googlesource.com/85361
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23762}
Before this CL, the packetsLost and jitter stats (as returned by
GetStats, at the API level) were only being updated when an RTCP SR or
RR is generated. According to the stats spec, "local" stats like this
should be updated any time a packet is received.
This CL also fixes some minor issues with the calculation of packetsLost
(and fractionLost):
* Packets weren't being count as lost if lost over a sequence number
rollover.
* Temporary periods of "negative" loss (caused by duplicate or out of
order packets) weren't being accumulated into the cumulative loss
counter. Example:
Period 1: Received packets 1, 2, 4
Loss over that period: 1 (expected 4 packets, got 3)
Reported cumulative loss: 1
Period 2: Received packets 3, 5
Loss over that period: -1 (expected 1 packet, got 2)
Reported cumulative loss: 1 (should be 0!)
Landing with NOTRY because Android compile bots are broken for an
unrelated reason.
NOTRY=True
Bug: webrtc:8804
Change-Id: I840ba34de8957b1276f6bdaf93718f805629f5c8
Reviewed-on: https://webrtc-review.googlesource.com/50020
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23731}
By removing it we can in turn (next CL) get rid of RTPTypeHeader, which is a
union that cause some problems.
Bug: none
Change-Id: I9246ecbfe2c8b7eda27497cccbc5f438958b64bf
Reviewed-on: https://webrtc-review.googlesource.com/83985
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23666}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated using script:
#!/bin/bash
dir=modules/rtp_rtcp
find $dir -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $dir -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc
Reviewed-on: https://webrtc-review.googlesource.com/83584
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23624}
This is a reland of efc71e565e9b36bcdfb4571f59e34bbd8fabd0cd
Differs from the original cl by not widening the type of
VideoCodec::width and VideoCodec::height.
Original change's description:
> Move class VideoCodec from common_types.h to its own api header file.
>
> Bug: webrtc:7660
> Change-Id: I91f19bfc2565461328f30081f8383e136419aefb
> Reviewed-on: https://webrtc-review.googlesource.com/79881
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23544}
Bug: webrtc:7660
Change-Id: I7cf74a85a61ea2b831e6f32b3b3e17514ebefec8
Reviewed-on: https://webrtc-review.googlesource.com/82140
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23569}
This is a reland of 3409cfa378e75c0c08d900e0848147929249a62b
Needed to change RtpVideoStreamReceiver to stop deregistering a payload
type if two payload types refer to the same codec (which now happens,
with the packetization mode 0/1 payload types). It's not clear why this
was being done in the first place.
Original change's description:
> Start supporting H264 packetization mode 0.
>
> The work was already done to support it, but it wasn't being negotiated
> in SDP.
>
> This means we'll now see 8 H264 payload types instead of 4; one for each
> combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX.
> This could be problematic in the future, since we're starting to run
> out of dynamic payload types (using 25 of 32).
>
> Bug: chromium:600254
> Change-Id: Ief2340db77c796f12980445b547b87e939170fae
> Reviewed-on: https://webrtc-review.googlesource.com/77264
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23372}
Bug: chromium:600254
Change-Id: Ice1acc05acd1543d9b46e918de2bba0694d86259
Reviewed-on: https://webrtc-review.googlesource.com/78399
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23494}
The Quic BBR implementation uses packet sequence numbers to keep track
of the time slots used for calculation of send receive rates. To avoid
protocol dependence the port were initially written to use send times
instead.
As there are issues with running BBR in WebRTC, it makes sense to
use an identical implementation as in Quic to ensure that there
aren't implementation issues causing bad behavior. This requires
providing sequence numbers.
This prepares for making the BBR implementation more identical to the
implementation in Quic, this is to ensure that results are comparable.
Bug: webrtc:8415
Change-Id: I2cd96bc6ffb88042bb2b91421bfe6cbf7c1ff8ac
Reviewed-on: https://webrtc-review.googlesource.com/76583
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23353}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.
Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.
Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
This CL adds the ability to configure RTPSender to include the
MID header extension when sending packets. The MID will be
included on every packet at the start of the stream until an RTCP
acknoledgment is received for that SSRC at which point it will
stop being included. The MID will be included on regular RTP
streams as well as RTX streams.
Bug: webrtc:4050
Change-Id: Ie27ebee1cd00a67f2b931f5363788f523e3e684f
Reviewed-on: https://webrtc-review.googlesource.com/60582
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22574}
Also delete the method RTPPayloadRegistry::red_payload_type() and
remnants of RED support in RTPReceiverAudio.
Bug: webrtc:8995,webrtc:5922
Change-Id: Iee310f5a8628ba70942e8c0277a856d2ca1f9b35
Reviewed-on: https://webrtc-review.googlesource.com/61500
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22425}
Delete method RTPPayloadRegistry::ulpfec_payload_type().
RtpVideoStreamReceiver can check its own config to know what the
payload type is.
Bug: webrtc:8995
Change-Id: Idc2bc7d747d77127f2b2261ff50610422e5686a6
Reviewed-on: https://webrtc-review.googlesource.com/61501
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22414}
The GetTransportFeedbackVector() method is only used in tests, and they
can access the class directly anyway. Keeping it is adding code bloat
and is also making upcoming refactoring more difficult.
Bug: webrtc:8975
Change-Id: I8323addb3c1461dd73b30353c8d9fe9410471c15
Reviewed-on: https://webrtc-review.googlesource.com/60860
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22349}
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
