Commit graph

24 commits

Author SHA1 Message Date
Paul Hallak
47ed99872d Use the clock to convert absolute capture timestamps to NTP times.
This allows callers to use timestamps generated from their own clocks
without worrying about converting to webrtc time.

No-Try because of lack of infra lack of capacity on macs.

No-Try: True
Bug: webrtc:11327
Change-Id: I7b1935654a2b23cf844c7b3622ed68763ced9da5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219785
Commit-Queue: Paul Hallak <phallak@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34076}
2021-05-21 12:41:50 +00:00
Niels Moller
2accc7d6e0 Revert "Add task queue to RtpRtcpInterface::Configuration."
This reverts commit f23e2144e8.

Reason for revert: Need further discussion on appropriate thread/tq requirements.

Original change's description:
> Add task queue to RtpRtcpInterface::Configuration.
>
> Let ModuleRtpRtcpImpl2 use the configured value instead of
> TaskQueueBase::Current().
>
> Intention is to allow construction of RtpRtcpImpl2 on any thread.
> If a task queue is provided (required for periodic rtt updates), the
> destruction of the object must be done on that same task queue.
>
> Also, delete ModuleRtpRtcpImpl2::Create, callers updated to use std::make_unique.
>
> Bug: None
> Change-Id: I412b7b1e1ce24722ffd23d16aa6c48a7214c9bcd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199968
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32949}

TBR=danilchap@webrtc.org,ilnik@webrtc.org,saza@webrtc.org,nisse@webrtc.org,srte@webrtc.org

Change-Id: I7e5007f524a39a6552973ec9744cd04c13162432
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201420
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32953}
2021-01-12 17:47:32 +00:00
Niels Möller
f23e2144e8 Add task queue to RtpRtcpInterface::Configuration.
Let ModuleRtpRtcpImpl2 use the configured value instead of
TaskQueueBase::Current().

Intention is to allow construction of RtpRtcpImpl2 on any thread.
If a task queue is provided (required for periodic rtt updates), the
destruction of the object must be done on that same task queue.

Also, delete ModuleRtpRtcpImpl2::Create, callers updated to use std::make_unique.

Bug: None
Change-Id: I412b7b1e1ce24722ffd23d16aa6c48a7214c9bcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199968
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32949}
2021-01-12 12:42:58 +00:00
Minyue Li
e64b3d0159 Send estimated capture clock offset when sending Abs-capture-time RTP header extension.
Bug: webrtc:10739
Change-Id: I4e3c46c749b9907ae9d212651b564add91c56958
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182004
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#31973}
2020-08-20 16:23:22 +00:00
Tomas Gunnarsson
f25761d798 Remove dependency from RtpRtcp on the Module interface.
The 'Module' part of the implementation must not be
called via the RtpRtcp interface, but is rather a part of
the contract with ProcessThread. That in turn is an
implementation detail for how timers are currently implemented
in the default implementation.

Along the way I'm deprecating away the factory function which
was inside the interface and tied it to one specific implementation.
Instead, I'm moving that to the implementation itself and down the
line, we don't have to go through it if we just want to create an
instance of the class.

The key change is in rtp_rtcp.h and the new rtp_rtcp_interface.h
header file (things moved from rtp_rtcp.h), the rest falls from that.

Change-Id: I294f13e947b9e3e4e649400ee94a11a81e8071ce
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176419
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31440}
2020-06-04 08:11:21 +00:00
Tomas Gunnarsson
fae05624ec Deprecate the static RtpRtcp::Create() method.
The method is being used externally to create instances
of the deprecated internal implementation.

Instead, I'm moving how we instantiate the internal implementation into
the implementation itself and move towards keeping the interface
separate from a single implementation.

Change-Id: I743aa86dc4c812b545699c546c253c104719260e
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176404
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31420}
2020-06-03 09:41:34 +00:00
Ruslan Burakov
d74c56fcd0 Add absolute capture time to audio sender path.
WebRTC prototype:
https://webrtc-review.googlesource.com/c/src/+/158520

Bug: webrtc:10739
Change-Id: I07b7a60602b41dc04292a91923e878a8d753486f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161732
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30335}
2020-01-21 13:06:18 +00:00
Erik Språng
77b7529515 Reland "Use RtpSenderEgress directly instead of via RTPSender"
This is a reland of b533010bc6

Patchset 1 is identical to previously landed CL.
Patchset 2 contains a workaround to migrate downstream tests.

Original change's description:
> Use RtpSenderEgress directly instead of via RTPSender
>
> Bug: webrtc:11036
> Change-Id: Ida4e8bc705ae43ceb1b131114707b30d10ba8642
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158521
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29626}

Bug: webrtc:11036
Change-Id: I8054169036a7f9f262308cac59f12ac8f9c73c17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158531
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29635}
2019-10-28 17:13:30 +00:00
Erik Språng
a81e2b4510 Revert "Use RtpSenderEgress directly instead of via RTPSender"
This reverts commit b533010bc6.

Reason for revert: Breaks downstream tests.

Original change's description:
> Use RtpSenderEgress directly instead of via RTPSender
> 
> Bug: webrtc:11036
> Change-Id: Ida4e8bc705ae43ceb1b131114707b30d10ba8642
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158521
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29626}

TBR=ilnik@webrtc.org,sprang@webrtc.org

Change-Id: Ib3354f6907d21462a8ad0c37eb8f6e94c48af217
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11036
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158526
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29627}
2019-10-28 11:17:18 +00:00
Erik Språng
b533010bc6 Use RtpSenderEgress directly instead of via RTPSender
Bug: webrtc:11036
Change-Id: Ida4e8bc705ae43ceb1b131114707b30d10ba8642
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158521
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29626}
2019-10-28 10:38:14 +00:00
Danil Chapovalov
a74e47759e Deprecate legacy RtpHeaderExtensionMap::Register function
Bug: None
Change-Id: Ia27ecf4d316563c5f7693162aedff535855c403b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152667
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29170}
2019-09-12 17:04:01 +00:00
Danil Chapovalov
59e1464fcd Fix 28 ClangTidy - Readability findings in modules/rtp_rtcp/
These fixes are automatically created by various analysis tools, but have been manually triggered to be applied.
 * the 'empty' method should be used to check for emptiness instead of 'size' (3 times)
 * using decl 'Return' is unused (4 times)
 * using decl '_' is unused (3 times)
 * using decl 'DoAll' is unused (2 times)
 * using decl 'SetArgPointee' is unused
 * using decl 'Dlrr' is unused
 * using decl 'IsEmpty' is unused
 * redundant get() call on smart pointer
 * using decl 'Invoke' is unused (2 times)
 * using decl 'SizeIs' is unused (3 times)
 * using decl 'make_tuple' is unused
 * using decl 'NiceMock' is unused
 * using decl 'SaveArg' is unused (2 times)
 * using decl 'AtLeast' is unused
 * using decl 'ElementsAre' is unused
 * using decl 'Gt' is unused

Bug: None
Change-Id: I97658fb0e94620b8319d7c3da29b15e27ec23188
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151133
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29056}
2019-09-04 07:38:20 +00:00
Erik Språng
54d5d2c75b Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc
The name media_send_ssrc makes less sense when used mostly for the
RtcpReceiver functionality.

The old member is still there and used as a fallback. That will be
cleaned away after downstream code is fixed.

Bug: webrtc:10774
Change-Id: I4ec18db76910f31dfe76bc9b137ffe89220d3fa8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149836
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28923}
2019-08-21 09:45:21 +00:00
Jonas Olsson
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
Erik Språng
4580ca2e99 Reland: Add ability to set ssrcs of RtpSender at construction time
Patch set 1 is identical to original CL. Next one adds fix for
backwards compatibilit.

Original cl: https://webrtc-review.googlesource.com/c/src/+/144037

Bug: webrtc:10774
Change-Id: Ib72e3723c7a07e9ee83f97560a85367becd868a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144601
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28485}
2019-07-04 11:50:19 +00:00
Amit Hilbuch
02d7d353a9 Revert "Add ability to set ssrcs of RtpSender at construction time"
This reverts commit e9d6e658c3.

Reason for revert: breaks downstream project

Original change's description:
> Add ability to set ssrcs of RtpSender at construction time
> 
> Bug: webrtc:10774
> Change-Id: I7147a75ccbcd1093dcd2e08047da8900843fdd8d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144037
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28447}

TBR=asapersson@webrtc.org,sprang@webrtc.org

Change-Id: I8b0cba0836e7d86ae1718055196c2c89860b97ff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144368
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28453}
2019-07-02 21:05:07 +00:00
Erik Språng
e9d6e658c3 Add ability to set ssrcs of RtpSender at construction time
Bug: webrtc:10774
Change-Id: I7147a75ccbcd1093dcd2e08047da8900843fdd8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144037
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28447}
2019-07-02 13:03:25 +00:00
Niels Möller
c936cb6a86 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h
Bug: webrtc:5876
Change-Id: I0c92f9410fcf0832bfa321229b3437134255dba6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128085
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@{#27190}
2019-03-19 16:59:27 +00:00
Per Kjellander
e11b7d2e80 Replace field trials with WebRtcKeyValueConfig in RtpRtcpModule
Replaces use of field trials in RtpSender and RtpVideoSender with injectable WebRtcKeyValueConfig.
Implementation still defaults to field trials.

BUG: webrtc:10335
Change-Id: I5fc6d327ee5d011ccc41385734b38344df172627
Reviewed-on: https://webrtc-review.googlesource.com/c/123447
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26795}
2019-02-21 14:25:34 +00:00
Elad Alon
d8d3248d95 Reland "Delete test/constants.h"
This reverts commit 4f36b7a478.

Reason for revert: Failing tests fixed.

Original change's description:
> Revert "Delete test/constants.h"
>
> This reverts commit 389b1672a3.
>
> Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate
>
> Original change's description:
> > Delete test/constants.h
> >
> > It's not possible to use constants.h for all RTP extensions
> > after the number of extensions exceeds 14, which is the maximum
> > number of one-byte RTP extensions. This is because some extensions
> > would have to be assigned a number greater than 14, even if the
> > test only involves 14 extensions or less.
> >
> > For uniformity's sake, this CL also edits some files to use an
> > enum as the files involved in this CL, rather than free-floating
> > const-ints.
> >
> > Bug: webrtc:10288
> > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> > Commit-Queue: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26728}
>
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
>
> Bug: webrtc:10288, chromium:933127
> Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/123381
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26744}

TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org

Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954
Bug: webrtc:10288, chromium:933127
Reviewed-on: https://webrtc-review.googlesource.com/c/123384
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26750}
2019-02-19 08:51:20 +00:00
Oleh Prypin
4f36b7a478 Revert "Delete test/constants.h"
This reverts commit 389b1672a3.

Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate

Original change's description:
> Delete test/constants.h
>
> It's not possible to use constants.h for all RTP extensions
> after the number of extensions exceeds 14, which is the maximum
> number of one-byte RTP extensions. This is because some extensions
> would have to be assigned a number greater than 14, even if the
> test only involves 14 extensions or less.
>
> For uniformity's sake, this CL also edits some files to use an
> enum as the files involved in this CL, rather than free-floating
> const-ints.
>
> Bug: webrtc:10288
> Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26728}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org

No-Presubmit: True
Bug: webrtc:10288, chromium:933127
Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
Reviewed-on: https://webrtc-review.googlesource.com/c/123381
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26744}
2019-02-18 18:09:22 +00:00
Elad Alon
389b1672a3 Delete test/constants.h
It's not possible to use constants.h for all RTP extensions
after the number of extensions exceeds 14, which is the maximum
number of one-byte RTP extensions. This is because some extensions
would have to be assigned a number greater than 14, even if the
test only involves 14 extensions or less.

For uniformity's sake, this CL also edits some files to use an
enum as the files involved in this CL, rather than free-floating
const-ints.

Bug: webrtc:10288
Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
Reviewed-on: https://webrtc-review.googlesource.com/c/123048
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26728}
2019-02-17 21:47:41 +00:00
Niels Möller
59ab1cf081 Move ownership of RTPSenderVideo and RTPSenderAudio one level up
From RTPSender to RtpRtcpImpl. Makes RTPSender operate on packets
only, not frames.

Bug: webrtc:7135
Change-Id: Ia9a11456404c3b322d873d4f8fb828742296b26d
Reviewed-on: https://webrtc-review.googlesource.com/c/120044
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26586}
2019-02-07 13:31:48 +00:00
Niels Möller
a34d7766c5 Move RtpSenderAudioTest to its own file
Update RtpSenderAudioTest to call methods on RTPSenderAudio rather
than RTPSender, when possible. In particular, avoid
RTPSender::SendOutgoingData. Drop parameterization on the
WebRTC-SendSideBwe-WithOverhead field trial, since that appears
unrelated to these tests.

Also delete some unused parts of the RtpSender test.

Bug: webrtc:7135
Change-Id: I535bf48bb1720e2727f4a62fa3e49b2bb84394a0
Reviewed-on: https://webrtc-review.googlesource.com/c/120920
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26516}
2019-02-01 15:15:56 +00:00