It was introduced, together with the PacketLossStats class, in cl
https://codereview.webrtc.org/1198853004 (#9568). It is unused in webrtc,
but there's downstream usage of the PacketLossStats class, which
should perhaps be moved or deleted in a later cl.
Bug: None
Change-Id: I17a3d5c8748f2cc9809c438630cbe8ab680466c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140042
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28153}
This is a standardized metric. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
It is meant to replace the legacy googBucketDelay. The average
packet delay over any interval can be calculated as the delta
totalPacketSendDelay divided by the delta packetsSent between two
calls to getStats().
Bug: webrtc:10506
Change-Id: I3d6c6d66e5a06937d0ea8d182a82cd255084ad19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137044
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27979}
This CL introduces three-value enum, in order to be able to distinguish
between send success, send failure, and invalid states such as missing
packet or invalid ssrc.
The behavior is unchanged in this CL, a follow-up will change the pacer
to not consume media budget on invalid states.
Bug: webrtc:8052,webrtc:8975
Change-Id: I1c9e2226f995356daa538d3d3cf44945f35e0133
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135165
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27923}
If the receiver has indicated that a packet has been received, via a
TransportFeedback RTCP message, it is safe to remove it from the
RtpPacketHistory as we can be sure it won't be needed anymore.
This will reduce memory usage, reduce the risk of overflow in the
history at very high bitrates, and hopefully make payload based padding
a little more useful.
This is code stems partly from
https://webrtc-review.googlesource.com/c/src/+/134208
but without the RtpPacketHistory changes which were landed in
https://webrtc-review.googlesource.com/c/src/+/134307
Bug: webrtc:8975
Change-Id: Iea9d3d32bee5512473744e9ef3a18018567fc272
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135160
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27868}
This reverts commit 3890e99b70.
Reason for revert: Seems to be causing unexpected perf regressions.
Original change's description:
> Remove packets from RtpPacketHistory if acked via TransportFeedback
>
> If the receiver has indicated that a packet has been received, via a
> TransportFeedback RTCP message, it is safe to remove it from the
> RtpPacketHistory as we can be sure it won't be needed anymore.
> This will reduce memory usage, reduce the risk of overflow in the
> history at very high bitrates, and hopefully make payload based padding
> a little more useful.
>
> Bug: webrtc:8975
> Change-Id: I703a353252943f63d7d6edda68f03bc482633fd6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133028
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27745}
TBR=danilchap@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: I68ea6cf5c8988d4b625f14a1a9bc556c06a39368
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8975
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134161
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27752}
If the receiver has indicated that a packet has been received, via a
TransportFeedback RTCP message, it is safe to remove it from the
RtpPacketHistory as we can be sure it won't be needed anymore.
This will reduce memory usage, reduce the risk of overflow in the
history at very high bitrates, and hopefully make payload based padding
a little more useful.
Bug: webrtc:8975
Change-Id: I703a353252943f63d7d6edda68f03bc482633fd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133028
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27745}
With this change, both the normal RTP and the transport-wide sequence
numbers are propagated with with AddPacket() call via a new
RtpPacketSendInfo struct, replacing the previous set of parameters.
The intent with this is that SendTimeHistory can hold a mapping from
transport-wide to rtp sequence numbers, and then via callbacks let the
RTP modules know when packets have been received by the remote end.
Bug: webrtc:8975
Change-Id: I6a24fc6282cbb041393752d39593c2867b242192
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133021
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27708}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
This is a preparation for deleting the class StringRtpHeaderExtension.
Bug: webrtc:10440
Change-Id: I3480e58d96e67d10c4d78597c8ab7f01b63e37ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128761
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27228}
This is not used in practice as there's functionality on
other levels that serves the same purpose.
Bug: None
Change-Id: I0488dc42459b07607363eba0f2b06f4c50f7cda4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27061}
The idea is to let the RtpRtcp and RTPSender classes be responsible for
media-agnostic RTP transport, and move out the media-specific processing,
such as packetization and media-specific headers.
Bug: webrtc:7135
Change-Id: Ib0ce45bf06713b3eb6c06acd91c5168856874e4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123187
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26954}
The discardability flag denotes whether the frame may be dropped by
the decoder with no effect on the decodability of subsequent frames.
Bug: webrtc:10214
Change-Id: I3654951d8863b50effe9670b8d1d7eb051240039
Reviewed-on: https://webrtc-review.googlesource.com/c/122241
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26763}
Configurability via SetSelectiveRetransmissions was deleted in
https://webrtc-review.googlesource.com/c/119920.
Delete constants kRetransmitFECPackets and kRetransmitAllPackets,
which are never enabled in production code. Also move the declaration
of RetransmissionMode from rtp_rtcp_defines.h to rtp_sender_video.h,
to reduce visibility to applications.
Bug: None
Change-Id: I70dcf7532aa3415a2449d8d807c500c1f149bf6d
Reviewed-on: https://webrtc-review.googlesource.com/c/120053
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26570}
This is a reland of 171df93262
Original change's description:
> Delete RtpUtility::Payload, and refactor RTPSender to not use it
>
> Replaced by a payload type --> video codec map in RTPSenderVideo,
> where it is used to select the right packetizer.
>
> Bug: webrtc:6883
> Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
> Reviewed-on: https://webrtc-review.googlesource.com/c/119263
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26380}
Tbr: danilchap@webrtc.org
Bug: webrtc:6883
Change-Id: I30771b86bbe50de609353e23e80dc532dc884ad4
Reviewed-on: https://webrtc-review.googlesource.com/c/119661
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26394}
Create a new RTCP feedback message for reporting the loss and/or non-decodability of video frames, to be used by the upcoming injectable VideoFrameBufferController. The new feedback message should report:
1. The sequence number of the last decoded non-discardable video frame. (TBD: If a multi-packet frame, should it be the sequence number of the first, last, or any of the packets?)
2. The sequence number of the last received RTP packet in the stream.
3. A decodability flag, whose specific meaning depends on the last-received
RTP sequence number. The decodability flag is true if and only if all of
the frame's dependencies are known to be decodable, and the frame itself
is not yet known to be unassemblable.
* Clarification #1: In a multi-packet frame, the first packet's
dependencies are known, but it is not yet known whether all parts
of the current frame will be received.
* Clarification #2: In a multi-packet frame, the dependencies would be
unknown if the first packet was not received. Then, the packet will
be known-unassemblable.
Bug: webrtc:10226
Change-Id: I1563c944477e3ed40235e82ab99a439414632aff
Reviewed-on: https://webrtc-review.googlesource.com/c/118931
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26387}
This reverts commit 171df93262.
Reason for revert: Breaks downstream project
Original change's description:
> Delete RtpUtility::Payload, and refactor RTPSender to not use it
>
> Replaced by a payload type --> video codec map in RTPSenderVideo,
> where it is used to select the right packetizer.
>
> Bug: webrtc:6883
> Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
> Reviewed-on: https://webrtc-review.googlesource.com/c/119263
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26380}
TBR=danilchap@webrtc.org,brandtr@webrtc.org,nisse@webrtc.org
Change-Id: I76489c29541827aaba72515a76db54bdb7495e28
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6883
Reviewed-on: https://webrtc-review.googlesource.com/c/119640
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26385}
Replaced by a payload type --> video codec map in RTPSenderVideo,
where it is used to select the right packetizer.
Bug: webrtc:6883
Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
Reviewed-on: https://webrtc-review.googlesource.com/c/119263
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26380}
This is a follow up of a comment in
https://webrtc-review.googlesource.com/c/src/+/110105
It was not very useful to split the audio and video report interval since the RTCP module can only either be audio or video.
The recent it was written that way in https://webrtc-review.googlesource.com/c/src/+/43201/ was because that was a straightforward transition from two global constants to two variable.
Bug: webrtc:8789
Change-Id: I2293de14ba5f363351f379a02022ed5dc7b8d458
Reviewed-on: https://webrtc-review.googlesource.com/c/110824
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25741}
Delete the deprecated IncomingPacket method, and convert implementation
to use RtpPacketReceived rather than RTPHeader.
Part 1 was https://webrtc-review.googlesource.com/c/src/+/100104
Bug: webrtc:7135, webrtc:8016
Change-Id: Ib4840d947870403deea2f9067f847e4b0f182479
Reviewed-on: https://webrtc-review.googlesource.com/c/6762
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25648}
Process video rtp frequency in the same way.
Bug: webrtc:6458
Change-Id: Ia22768e1242d686c2b3e2b911f3e5e492cf8b895
Reviewed-on: https://webrtc-review.googlesource.com/c/107651
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25334}
This CL adds fields to packet feedback structs to indicate the amount of
data that was sent prior to the represented packet without being part
packet feedback, but part of bitrate allocation.
This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.
Bug: webrtc:9796
Change-Id: I716a5325e2b7022ba6b3f90653542caafb056793
Reviewed-on: https://webrtc-review.googlesource.com/c/104921
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25067}
When two-byte header extensions are enabled, kMaxId will change from 15
to 255. This CL is a refactor to remove the direct dependency between
memory allocation and kMaxId.
Bug: webrtc:7990
Change-Id: I38974a9c705eb6a0fdba9038a7d909861587d98d
Reviewed-on: https://webrtc-review.googlesource.com/101580
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24801}
Uninline RTPFragmentaion functions
fix RTPFragmentation move constructor and assign operators (was recursive for win)
replace assert with rtc::dchecked_cast
Remove unused includes and dependencies.
Fix other targets that used those includes transitively instead of directly
Bug: None
Change-Id: I647cb1eda107dc7d87d25234095545bc2842fa40
Reviewed-on: https://webrtc-review.googlesource.com/100500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24759}
to list of extensions supported by RtpPacket.
Bug: webrtc:9361
Change-Id: Iabee824381be3776e17e95f32507058607fc0bc8
Reviewed-on: https://webrtc-review.googlesource.com/85346
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23788}
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}
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}
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}
Instead of modifying the API, we'll add a new function to return
the true value, and have a shim that returns what other code expects.
> This reverts commit 4c34f435db.
>
> Reason for revert: Broke internal projects. Type mismatch.
>
> Original change's description:
> > Make RTCP cumulative_lost be a signed value
> >
> > This is formally defined as a signed 24-bit value in RFC 3550 section 6.4.1.
> > See RFC 3550 Appendix A.3 for the reason why it may turn negative.
> >
> > Noticed on discuss-webrtc mailing list.
> >
> > BUG=webrtc:8626
> >
> > Change-Id: I7317f73e9490a876e8445bd3d6b66095ce53ca0a
> > Reviewed-on: https://webrtc-review.googlesource.com/30901
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21142}
>
> TBR=stefan@webrtc.org,hta@webrtc.org
>
> Change-Id: I544f7979d584cfb72a2d0d526f4fef84aebeecb3
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8626
> Reviewed-on: https://webrtc-review.googlesource.com/31040
> Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21144}
Change-Id: I95c8c248f4f85c4d1aa2a47424d8c4d954d4ae7a
Bug: webrtc:8626
Reviewed-on: https://webrtc-review.googlesource.com/31220
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21154}
This reverts commit 54d1da13a5.
Reason for revert: Breaking tests
Original change's description:
> BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.
>
> This CL implements the main logic and IOS appRTC integration.
>
> Unit tests and Android appRTC will be in separate CL.
>
> Bug: webrtc:8243
> Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
> Reviewed-on: https://webrtc-review.googlesource.com/4860
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20329}
TBR=deadbeef@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,alexnarest@webrtc.org
Change-Id: I5be1da78f360f72be66f9d56dd6b88c1cc13e963
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/12560
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20330}
This CL implements the main logic and IOS appRTC integration.
Unit tests and Android appRTC will be in separate CL.
Bug: webrtc:8243
Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
Reviewed-on: https://webrtc-review.googlesource.com/4860
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20329}
This is needed in the general case, now that we aim to support codecs
other than those built-in to WebRTC.
BUG=webrtc:8159
Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706
Reviewed-on: https://webrtc-review.googlesource.com/5380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20136}
Remove PayloadUnion's public member variables, so that the outside
world has to go through the accessors.
This is good code hygiene in general. For example, it makes it
possible to make the audio and video states Optional, so that exactly
one of them can be live at any one time.
BUG=webrtc:8159
Change-Id: Ie617b9038f961b329bd67b45478ff33d97148447
Reviewed-on: https://webrtc-review.googlesource.com/4428
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20064}
I need to replace the audio part of PayloadUnion with SdpAudioFormat,
but that's a non-trivially-deletable class and those just don't work
well in unions, especially unions that don't have a discriminator that
says which member is currently active.
This CL converts the union to a class, adds a discriminator, and
provides accessor functions. CL #2 in the series will change all
outsiders to use the accessors instead of the public member variables
directly, and CL #3 will remove the public member variables. (It needs
to be done in separate steps like this because PayloadUnion is
unfortunately part of the API, and just changing it all in one go
would break users.)
BUG=webrtc:8159
Change-Id: I38c44bbb21a2d38600cff59bf37d8d47dfdbce21
Reviewed-on: https://webrtc-review.googlesource.com/4340
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20025}
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.
The cpplint complaint is:
Include the directory when naming .h files [build/include] [4]
This CL disables the error but we have to remove these two headers
from the root directory.
NOPRESUBMIT=true
Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
