Commit Graph

37 Commits

Author SHA1 Message Date
c5744b8b21 Refactor to remove direct memory dependency on kMaxId
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}
2018-09-24 13:26:46 +00:00
db1285676b Cleanup modules_common_types
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}
2018-09-18 08:08:33 +00:00
e0c8b230e7 Frame marking RTP header extension (PART 1: implement extension)
Bug: webrtc:7765
Change-Id: I23896d121afd6be4bce5ff4deaf736149efebcdb
Reviewed-on: https://webrtc-review.googlesource.com/85200
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24695}
2018-09-11 22:35:30 +00:00
9701e0ce2f Makes treatment of received reports of packets lost signed.
Bug: webrtc:9598
Change-Id: I0f6ffe348585b8ec69753089652812da516d33d8
Reviewed-on: https://webrtc-review.googlesource.com/93021
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24291}
2018-08-15 14:27:23 +00:00
3ed46bd83b Delete RTPReceiverStrategy::OnNewPayloadTypeCreated and related code.
Bug: webrtc:7135
Change-Id: Ic20d98cbfb8154f5abbc2501cbcccb950148e732
Reviewed-on: https://webrtc-review.googlesource.com/92396
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@{#24219}
2018-08-08 08:01:32 +00:00
ab4a530b87 Delete telephone-event handling from RTPReceiverAudio.
Bug: webrtc:7135
Change-Id: Ic8b96f44ba25ff9265570dd43d3c76ed0177abfb
Reviewed-on: https://webrtc-review.googlesource.com/91125
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@{#24172}
2018-08-02 12:55:40 +00:00
a12c42a6b2 Delete root header file typedef.h.
Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 14:59:26 +00:00
916ec7dadf Add Generic frame descritpor header extension
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}
2018-06-29 15:02:44 +00:00
fb8e7ef842 Implement PayloadUnion as variant instead of pair of optionals
Bug: None
Change-Id: I2e54f5a0561804bc59c4d4c8e35ccdaa9536b8e4
Reviewed-on: https://webrtc-review.googlesource.com/85366
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23745}
2018-06-26 15:58:06 +00:00
665174fdbb Reformat the WebRTC code base
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}
2018-06-19 14:00:39 +00:00
d264df587f Replace rtc::Optional with absl::optional in modules/rtp_rtcp
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}
2018-06-15 09:53:35 +00:00
520ca4e3b8 Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.
Bug: webrtc:8995
Change-Id: I0b44aa26f2f6a81aec7ca1281b8513d8e03228b8
Reviewed-on: https://webrtc-review.googlesource.com/79561
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23507}
2018-06-04 11:53:17 +00:00
92f83cec12 Remove deprecated rtcp SLI/RPSI observers
Bug: webrtc:7338
Change-Id: I39247a3d969637856496b630cadaacac16ef8d09
Reviewed-on: https://webrtc-review.googlesource.com/79260
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23413}
2018-05-28 13:10:54 +00:00
f782492948 Delete RtpFeedback. The ssrc for a receive stream should be known at
configuration time.

Bug: webrtc:8995
Change-Id: I3d63a76e472a8948c98c98450e96d3301fa2688b
Reviewed-on: https://webrtc-review.googlesource.com/78701
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23409}
2018-05-28 11:05:19 +00:00
02c65869c3 Adds unwrapped sequence number to feedback info.
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}
2018-05-22 16:28:19 +00:00
ef99888bca Delete OnIncomingCSRCChanged and related code.
Bug: webrtc:8995
Change-Id: I1987d1527cce5a0c315b2d15cfffa76e7343b1fa
Reviewed-on: https://webrtc-review.googlesource.com/64220
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22626}
2018-03-27 13:18:35 +00:00
9cfb18c5b3 Delete obsolete method RtpFeedback::OnInitializeDecoder.
Bug: None
Change-Id: I55e01e5ff1c54c76c43b378414a31fc43c9aa444
Reviewed-on: https://webrtc-review.googlesource.com/62142
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22561}
2018-03-22 12:06:54 +00:00
8493594dc2 Cleanup of TransportFeedbackObserver interface
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}
2018-03-08 22:51:53 +00:00
3587b8302a Make RTCP report interval configurable
Bug: webrtc:8789
Change-Id: I79c9132123c946b030ed79c647b4329e81d6e6ae
Reviewed-on: https://webrtc-review.googlesource.com/43201
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21837}
2018-02-01 10:12:11 +00:00
eb0edd832a Narrow interface PacketRouter use to send Remb and TransportFeedback
This allows to use RtcpTransceiver implementation instead of RtpRtcp.
No functional changes.

Bug: webrtc:8239
Change-Id: I3c5bd23ff2136eb844e85b567b70380fc2a65929
Reviewed-on: https://webrtc-review.googlesource.com/33005
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21298}
2017-12-15 15:58:17 +00:00
3e113438b1 Fix circular dependencies in webrtc_common.
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}
2017-12-15 14:33:26 +00:00
70206d6608 Reland "Make RTCP cumulative_lost be a signed value"
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 4c34f435db2b921b82b8be19ee5c1746f46cb188.
>
> 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}
2017-12-08 08:47:09 +00:00
e0572e5c16 Reland "Replaced magic numbers with constants for PacketFeedback."
This is a reland of 37b52232895fc200188c0e3ded261aedcb558b7b
Original change's description:
> Replaced magic numbers with constants for PacketFeedback.
> 
> Bug: None
> Change-Id: Ie22475227406f4e800052b52fa644ea6966db3f1
> Reviewed-on: https://webrtc-review.googlesource.com/27100
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20938}

Bug: None
Change-Id: I131b509212345a620519b17c1c17e84532ac089c
Reviewed-on: https://webrtc-review.googlesource.com/27401
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20957}
2017-11-30 16:04:20 +00:00
575ceefc6d Revert "Replaced magic numbers with constants for PacketFeedback."
This reverts commit 37b52232895fc200188c0e3ded261aedcb558b7b.

Reason for revert: Breaking internal builds

Original change's description:
> Replaced magic numbers with constants for PacketFeedback.
> 
> Bug: None
> Change-Id: Ie22475227406f4e800052b52fa644ea6966db3f1
> Reviewed-on: https://webrtc-review.googlesource.com/27100
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20938}

TBR=stefan@webrtc.org,srte@webrtc.org

Change-Id: I891977c9535c4c887013f3f5badc83666c29e3f8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/27220
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20943}
2017-11-29 21:15:01 +00:00
37b5223289 Replaced magic numbers with constants for PacketFeedback.
Bug: None
Change-Id: Ie22475227406f4e800052b52fa644ea6966db3f1
Reviewed-on: https://webrtc-review.googlesource.com/27100
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20938}
2017-11-29 16:56:19 +00:00
84c1a15d3c Remove deprecated field names in struct RTCPReportBlock
Bug: webrtc:8033
Change-Id: Ic7ea4fdd6cd30a2a490f1bd9fdd9a4f0a4d51f4a
Reviewed-on: https://webrtc-review.googlesource.com/23262
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20706}
2017-11-16 11:15:43 +00:00
78609d5b6b Reland of BWE allocation strategy
TBR=stefan@webrtc.org,alexnarest@webrtc.org

Bug: webrtc:8243
Change-Id: Ie68e4f414b2ac32ba4e64877cb250fabcb089a07
Reviewed-on: https://webrtc-review.googlesource.com/13940
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20369}
2017-10-20 10:16:15 +00:00
dc9ca9329b Revert "BWE allocation strategy"
This reverts commit a5fbc23379823d74b8cf4bc18887ff40237989e8.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> BWE allocation strategy
> 
> This is reland of https://webrtc-review.googlesource.com/c/src/+/4860 with the fixed RampUpTest test
> 
> Bug: webrtc:8243
> Change-Id: I4b90a449b00dd05feee974001e08fb40710b59ac
> Reviewed-on: https://webrtc-review.googlesource.com/13124
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20345}

TBR=stefan@webrtc.org,alexnarest@webrtc.org

Change-Id: I8ed12cd2115ef63204e384cc93c9f4473daa54d1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/14020
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20361}
2017-10-19 15:34:52 +00:00
a5fbc23379 BWE allocation strategy
This is reland of https://webrtc-review.googlesource.com/c/src/+/4860 with the fixed RampUpTest test

Bug: webrtc:8243
Change-Id: I4b90a449b00dd05feee974001e08fb40710b59ac
Reviewed-on: https://webrtc-review.googlesource.com/13124
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20345}
2017-10-19 09:30:00 +00:00
39260c4a6b Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic."
This reverts commit 54d1da13a584680ae80a1f229291e5bb7e76e6e1.

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}
2017-10-17 19:59:04 +00:00
54d1da13a5 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}
2017-10-17 18:22:15 +00:00
c62f6c7121 RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs
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}
2017-10-04 11:30:14 +00:00
884e49f9d6 Convert PayloadUnion from a union to a class, step 3
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}
2017-10-02 08:53:30 +00:00
83d3ec177c Convert PayloadUnion from a union to a class, step 1
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}
2017-09-28 18:32:37 +00:00
7120742701 Adding NOLINT for typedefs.h and common_types.h
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}
2017-09-15 13:03:51 +00:00
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
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}
2017-09-15 05:02:56 +00:00
bb547203bf Moving src/webrtc into src/.
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}
2017-09-15 04:25:06 +00:00