31 Commits

Author SHA1 Message Date
Niels Möller
5304a32a94 Delete StreamStatistician::IsRetransmitOfOldPacket
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}
2018-08-30 11:00:13 +00:00
Åsa Persson
a765c8208a Change some pointers to std::unique_ptr in rtp_rtcp tests.
Bug: none
Change-Id: Ia4e69e44bbda7b5b633b8be1779d105649f44930
Reviewed-on: https://webrtc-review.googlesource.com/94844
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24419}
2018-08-24 07:26:04 +00:00
Åsa Persson
315ce5b308 Remove unused members in rtp_rtcp tests and make some members const.
Bug: none
Change-Id: I5f92899742406157d94de235e7c1a50755b3ac61
Reviewed-on: https://webrtc-review.googlesource.com/92393
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24303}
2018-08-16 07:53:16 +00:00
Sebastian Jansson
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
Niels Möller
d3b62cfb02 Delete unused method RtpReceiver::CSRCs.
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}
2018-08-10 10:01:11 +00:00
Sami Kalliomäki
426a80ce08 Add extended header containing frame ID to the generic packetizer.
Also changes default value of frame ID in RTPVideoHeader to
kNoPictureId. Special care should be take so that picture ID will not
be set in RTPVideoHeader unless the client on the end supports
deserializing extended generic header.

Bug: webrtc:9582
Change-Id: Ib096373ed187f31e51d481193a2bda56de68f167
Reviewed-on: https://webrtc-review.googlesource.com/92084
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24250}
2018-08-09 14:05:39 +00:00
Niels Möller
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
Joachim Bauch
d3b7ec2e91 Allow all "token" chars from RFC 4566 when checking for legal mid names.
Previously only alphanumeric characters were allowed.

Bug: webrtc:9537
Change-Id: I3fd793ad88520b25ecd884efe3a698f2f0af4639
Reviewed-on: https://webrtc-review.googlesource.com/89388
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24167}
2018-08-01 18:20:42 +00:00
Niels Möller
1bd66642c3 Set RtpReceiverAudio::telephone_event_forward_to_decoder_ true on construction.
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}
2018-07-30 12:24:49 +00:00
philipel
0a5fe77d23 Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
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}
2018-06-19 16:44:19 +00:00
Yves Gerey
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
Niels Möller
a46bd4b9c7 Reland "Move class VideoCodec from common_types.h to its own api header file."
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}
2018-06-11 19:23:20 +00:00
Danil Chapovalov
350531e2a3 Revert "Move class VideoCodec from common_types.h to its own api header file."
This reverts commit efc71e565e9b36bcdfb4571f59e34bbd8fabd0cd.

Reason for revert: probably breaks downstream test

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}

TBR=danilchap@webrtc.org,brandtr@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org

Change-Id: Id8bd37c79c2f8d09a4d88368765230103f1db2c8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7660
Reviewed-on: https://webrtc-review.googlesource.com/82101
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23547}
2018-06-08 11:04:23 +00:00
Niels Möller
efc71e565e 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}
2018-06-08 07:55:04 +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
Niels Möller
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
Danil Chapovalov
dd7e284ce8 Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets"
This reverts commit 01aa210fad68f1006528d32d388b307c22990734.

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 9486b117daac09c9f7ac8450ccda835938cf3150.
> 
> 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 9486b117daac09c9f7ac8450ccda835938cf3150.

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
2e1d784956 Delete the VideoCodec::plName string.
It holds the same information as codecType, but in different format.

Bug: webrtc:8830
Change-Id: Ia83e2dff4fd9a5ddb489501b7a1fe80759fa4218
Reviewed-on: https://webrtc-review.googlesource.com/56100
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22307}
2018-03-06 11:17:41 +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
Niels Möller
c3fa8e1ce7 New method RtpReceiver::GetLatestTimestamps.
The two timestamps, rtp time and corresponding system time, are always
used together, for audio/video sync. The new method reads both
timestamps, without releasing a lock in between. Ensures that the
caller gets values corresponding to the same packet.

Bug: webrtc:7135
Change-Id: I25bdcbe9ad620016bfad39841b339c266efade14
Reviewed-on: https://webrtc-review.googlesource.com/4062
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20120}
2017-10-03 16:14:29 +00:00
Karl Wiberg
73b60b82ee Remove the redundant method GetPayloadSpecifics
It's in the way of a refactoring.

Also change PayloadTypeToPayload---the method all callers can use instead---to return Optional<Payload> instead of const Payload* (for thread safety reasons: an object that protects itself with a mutex shouldn't be handing out pointers to parts of itself). 

BUG=webrtc:8159

Change-Id: I7ef0d545077ffdea016b309f2165e3c4955a2928
Reviewed-on: https://webrtc-review.googlesource.com/2360
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19917}
2017-09-21 20:19:55 +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
niklase@google.com
5adc73aad3 git-svn-id: http://webrtc.googlecode.com/svn/trunk@166 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:46:41 +00:00
niklase@google.com
ff72b0d8f3 Review URL: http://webrtc-codereview.appspot.com/40002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@89 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-15 23:43:03 +00:00
niklase@google.com
89714f2880 Review URL: http://webrtc-codereview.appspot.com/33009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@88 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-15 23:07:17 +00:00
niklase@google.com
77ae29bc81 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:22:19 +00:00