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10 commits

Author SHA1 Message Date
Stefan Holmer
dbdb3a0079 Refactoring PayloadRouter. 
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
  VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
  of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
  renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.

Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
 2018-07-17 14:46:15 +00:00
JT Teh
5daeff9c1f Revert "Remove RTPVideoHeader::h264() accessors." 
This reverts commit dfbced6504.

Reason for revert: Crashes when making a video call.

#9	0x00000001043dd8d8 in webrtc::RTPVideoHeaderH264& absl::variant_internal::TypedThrowBadVariantAccess<webrtc::RTPVideoHeaderH264&>() at /third_party/absl/types/internal/variant.h:315
#10	0x00000001043dd8ac in absl::variant_internal::VariantAccessResultImpl<2ul, absl::variant<webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>&&&>::type absl::variant_internal::VariantCoreAccess::CheckedAccess<2ul, absl::variant<webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>&>(absl::variant<webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>&&&) at /third_party/absl/types/internal/variant.h:597
#11	0x00000001043db778 in webrtc::RTPVideoHeaderH264& absl::get<webrtc::RTPVideoHeaderH264, webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>(absl::variant<webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>&) at /third_party/absl/types/variant.h:299
#12	0x0000000104558bcc in webrtc::RtpPacketizer::Create(webrtc::VideoCodecType, unsigned long, unsigned long, webrtc::RTPVideoHeader const*, webrtc::FrameType) at webrtc/modules/rtp_rtcp/source/rtp_format.cc:30

Original change's description:
> Remove RTPVideoHeader::h264() accessors.
>
> Bug: none
> Change-Id: I043bcaf358575688b223bc3631506e148b47fd58
> Reviewed-on: https://webrtc-review.googlesource.com/88220
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23971}

TBR=danilchap@webrtc.org,stefan@webrtc.org,philipel@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: none
Change-Id: If99bcabdfe3cae7094f24e407bbe2f47233e46e3
Reviewed-on: https://webrtc-review.googlesource.com/88820
Commit-Queue: JT Teh <jtteh@webrtc.org>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Reviewed-by: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23993}
 2018-07-16 21:36:12 +00:00
philipel
dfbced6504 Remove RTPVideoHeader::h264() accessors. 
Bug: none
Change-Id: I043bcaf358575688b223bc3631506e148b47fd58
Reviewed-on: https://webrtc-review.googlesource.com/88220
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23971}
 2018-07-13 13:53:05 +00:00
philipel
5ab67a5d71 Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader. 
This CL is in preparation to change the RTPVideoTypeHeader into an absl::variant.

Bug: none
Change-Id: I1672d866df0395f3417d8e278cc67f017ab0ff98
Reviewed-on: https://webrtc-review.googlesource.com/87261
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23856}
 2018-07-05 14:29:07 +00:00
Ilya Nikolaevskiy
8643b78750 Moved NackModule and VCMPacket to their own targets 
Bug: webrtc:9373
Change-Id: I1e882b734dcafb5c633eabf08bb8a1a6a407a251
Reviewed-on: https://webrtc-review.googlesource.com/81744
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23621}
 2018-06-15 09:00:25 +00:00
Niels Möller
425f713d24 Delete unused methods in VCMFrameBuffer and VCMSessionInfo. 
Bug: None
Change-Id: Ia97bb14ac9fa1a31dae248fc5a0f58e07b588ec7
Reviewed-on: https://webrtc-review.googlesource.com/82164
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23566}
 2018-06-11 18:39:50 +00:00
Niels Möller
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
Mirko Bonadei
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros. 
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
 2017-11-09 11:56:32 +00:00
Mirko Bonadei
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
Mirko Bonadei
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
Renamed from webrtc/modules/video_coding/session_info.cc (Browse further)