Commit graph

11 commits

Author SHA1 Message Date
Niels Möller
af17595879 Refactor RtpReceiverImpl, extracting CSRC book-keeping to its own class
Bug: webrtc:7135
Change-Id: I7ce9afe575241542e4e3f7e2e8459ee3257eec76
Reviewed-on: https://webrtc-review.googlesource.com/93466
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24271}
2018-08-13 11:59:08 +00:00
Niels Möller
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
Danil Chapovalov
dd7e284ce8 Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets"
This reverts commit 01aa210fad.

Reason for revert: downstream project adjusted

Original change's description:
> Revert "Enable and fix chromium clang warnings in rtp_rtcp test targets"
> 
> This reverts commit 9486b117da.
> 
> Reason for revert: Breaks downstream project
> 
> Original change's description:
> > Enable and fix chromium clang warnings in rtp_rtcp test targets
> > 
> > Bug: webrtc:163
> > Change-Id: I4ed3e63296d8bf06536a83196d597c7a906ba11c
> > Reviewed-on: https://webrtc-review.googlesource.com/60802
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22357}
> 
> TBR=danilchap@webrtc.org,phoglund@webrtc.org,terelius@webrtc.org
> 
> Change-Id: I2c3777ea9f26813bdb395e7fd68f6b49443586ea
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:163
> Reviewed-on: https://webrtc-review.googlesource.com/61060
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22365}

TBR=danilchap@webrtc.org,phoglund@webrtc.org,oprypin@webrtc.org,terelius@webrtc.org

Change-Id: I0b4cb6d05b37caeb52cca9abf95417ad3ad6f76b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:163
Reviewed-on: https://webrtc-review.googlesource.com/61080
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22368}
2018-03-09 16:04:35 +00:00
Oleh Prypin
01aa210fad Revert "Enable and fix chromium clang warnings in rtp_rtcp test targets"
This reverts commit 9486b117da.

Reason for revert: Breaks downstream project

Original change's description:
> Enable and fix chromium clang warnings in rtp_rtcp test targets
> 
> Bug: webrtc:163
> Change-Id: I4ed3e63296d8bf06536a83196d597c7a906ba11c
> Reviewed-on: https://webrtc-review.googlesource.com/60802
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22357}

TBR=danilchap@webrtc.org,phoglund@webrtc.org,terelius@webrtc.org

Change-Id: I2c3777ea9f26813bdb395e7fd68f6b49443586ea
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:163
Reviewed-on: https://webrtc-review.googlesource.com/61060
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22365}
2018-03-09 14:49:15 +00:00
Danil Chapovalov
9486b117da Enable and fix chromium clang warnings in rtp_rtcp test targets
Bug: webrtc:163
Change-Id: I4ed3e63296d8bf06536a83196d597c7a906ba11c
Reviewed-on: https://webrtc-review.googlesource.com/60802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22357}
2018-03-09 12:27:35 +00:00
Niels Möller
22ec952829 Delete in_order argument to RtpReceiver::IncomingRtpPacket
Bug: webrtc:7135
Change-Id: I35fbc76a5ca8d50caff918bbfd2cb13dce4cbd21
Reviewed-on: https://webrtc-review.googlesource.com/4141
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20154}
2017-10-05 07:19:20 +00:00
Karl Wiberg
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
Karl Wiberg
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
Mirko Bonadei
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
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/rtp_rtcp/source/rtp_receiver_unittest.cc (Browse further)