274 Commits

Author SHA1 Message Date
Niels Möller
1f3206cca4 Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1
Add new method OnRtpPacket, but leave
ReceiveStatisticsImpl::IncomingPacket and most of the implementation
unchanged. Deleting the old method and converting implementation from
RTPHeader to RtpPacketreceived is planned for a followup, after
downstream code is updated.

Bug: webrtc:7135, webrtc:8016
Change-Id: I697ec12804618859f8d69415622d1b957e1d0847
Reviewed-on: https://webrtc-review.googlesource.com/100104
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24889}
2018-09-28 12:00:28 +00:00
Johannes Kron
07ba2b9445 Parse two-byte header extensions.
Bug: webrtc:7990
Change-Id: I967d2065b85d6a2ca938ac0e83035cb92b45a907
Reviewed-on: https://webrtc-review.googlesource.com/98160
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24881}
2018-09-28 08:32:17 +00:00
Danil Chapovalov
9eb6ce1cd0 Remove redundant member variable in RtpRtcp
Bug: None
Change-Id: Ia999bb4020c8f270c916074e5c58bab15f6c77d0
Reviewed-on: https://webrtc-review.googlesource.com/33300
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24848}
2018-09-26 11:34:35 +00:00
philipel
569397fec7 Reland "Added field trial WebRTC-GenericDescriptor for the new generic descriptor."
This reverts commit 6f68324adbf52b247e10b33a4e83a586e66cc6df.

Reason for revert: Removed full stack tests that cause timeout.

Original change's description:
> Revert "Added field trial WebRTC-GenericDescriptor for the new generic descriptor."
> 
> This reverts commit 3f4a4fad8cd661309ff5d9a631e89518f32e7c5e.
> 
> Reason for revert: Breaking internal tests
> 
> Original change's description:
> > Added field trial WebRTC-GenericDescriptor for the new generic descriptor.
> > 
> > Also parameterized tests to test the new generic descriptor and
> > added --generic_descriptor flag to loopback tests.
> > 
> > Bug: webrtc:9361
> > Change-Id: I2b76889606fe2e81249ecdebb0b7b61151682485
> > Reviewed-on: https://webrtc-review.googlesource.com/101900
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24835}
> 
> TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org
> 
> Change-Id: I4d4714a9f4ab0e95adf0f4130bc1a932efc448fa
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9361
> Reviewed-on: https://webrtc-review.googlesource.com/101940
> Reviewed-by: Lu Liu <lliuu@webrtc.org>
> Commit-Queue: Lu Liu <lliuu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24839}

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

Change-Id: Ibcf0a1d3aa947b84e3b891b1975d0fc2c730f2ae
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9361
Reviewed-on: https://webrtc-review.googlesource.com/102064
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24845}
2018-09-26 10:26:43 +00:00
Johannes Kron
4a8a5e7db1 Reland "Second reland of "Optimize execution time of RTPSender::UpdateDelayStatistics""
This reverts commit 8b7bc5d7010c84ac57459518fe18309ef5fee1dd.

Reason for revert: Slow RTC_DCHECK has been removed.

Original change's description:
> Revert "Second reland of "Optimize execution time of RTPSender::UpdateDelayStatistics""
>
> This reverts commit 9def3b45ef06de9e068e8f4d1644e9d508baa913.
>
> Reason for revert: webrtc_perf_tests fails on Mac-10.12.
>
> Original change's description:
> > Second reland of "Optimize execution time of RTPSender::UpdateDelayStatistics"
> >
> > The reland has a lot of additional DCHECKS for easier debugging,
> > so in debug builds it will actually be a ~2x slowdown compared to the old code.
> > The excessive DCHECKS should be removed in a followup CL.
> >
> > Bug: webrtc:9439
> > Change-Id: I8389cd84f1ca12c29cc6993f0d2cf7e6d7dd8379
> > Reviewed-on: https://webrtc-review.googlesource.com/101761
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24821}
>
> TBR=terelius@webrtc.org,asapersson@webrtc.org,kron@webrtc.org
>
> Change-Id: I98c4c96d552858d0299d49993e9b9be6a6204dfe
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9439
> Reviewed-on: https://webrtc-review.googlesource.com/101860
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24825}

TBR=terelius@webrtc.org,asapersson@webrtc.org,kron@webrtc.org

Change-Id: I260c56932710d26f9d7201c07279fef8d2150bd9
Bug: webrtc:9439
Reviewed-on: https://webrtc-review.googlesource.com/102000
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24843}
2018-09-26 09:45:25 +00:00
Lu Liu
6f68324adb Revert "Added field trial WebRTC-GenericDescriptor for the new generic descriptor."
This reverts commit 3f4a4fad8cd661309ff5d9a631e89518f32e7c5e.

Reason for revert: Breaking internal tests

Original change's description:
> Added field trial WebRTC-GenericDescriptor for the new generic descriptor.
> 
> Also parameterized tests to test the new generic descriptor and
> added --generic_descriptor flag to loopback tests.
> 
> Bug: webrtc:9361
> Change-Id: I2b76889606fe2e81249ecdebb0b7b61151682485
> Reviewed-on: https://webrtc-review.googlesource.com/101900
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24835}

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

Change-Id: I4d4714a9f4ab0e95adf0f4130bc1a932efc448fa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9361
Reviewed-on: https://webrtc-review.googlesource.com/101940
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24839}
2018-09-25 18:49:02 +00:00
Danil Chapovalov
84ffb35e74 Use minimalistic vp8 descriptor when generic descriptor is in use
which saves up to 5 bytes per packet.

Bug: webrtc:9361
Change-Id: Ia4335459fb2bd6390950ddd1d487a0f9f96f743d
Reviewed-on: https://webrtc-review.googlesource.com/101840
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24838}
2018-09-25 18:05:52 +00:00
philipel
3f4a4fad8c Added field trial WebRTC-GenericDescriptor for the new generic descriptor.
Also parameterized tests to test the new generic descriptor and
added --generic_descriptor flag to loopback tests.

Bug: webrtc:9361
Change-Id: I2b76889606fe2e81249ecdebb0b7b61151682485
Reviewed-on: https://webrtc-review.googlesource.com/101900
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24835}
2018-09-25 16:55:55 +00:00
Danil Chapovalov
f3119ef66d Add generic frame descriptor to list of supported video extensions
Update uri to include version.

Bug: webrtc:9361
Change-Id: I71164879d2a5b88c052fdab9f74202f758e85be6
Reviewed-on: https://webrtc-review.googlesource.com/101762
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24827}
2018-09-25 13:41:01 +00:00
philipel
dabfcae2d9 Use the generic descriptor information in the RtpFrameReferenceFinder.
Bug: webrtc:9361
Change-Id: I8d7e7ee1d3ca89283552c21c45950d4dba1c1927
Reviewed-on: https://webrtc-review.googlesource.com/100301
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24826}
2018-09-25 13:15:45 +00:00
Johannes Kron
8b7bc5d701 Revert "Second reland of "Optimize execution time of RTPSender::UpdateDelayStatistics""
This reverts commit 9def3b45ef06de9e068e8f4d1644e9d508baa913.

Reason for revert: webrtc_perf_tests fails on Mac-10.12.

Original change's description:
> Second reland of "Optimize execution time of RTPSender::UpdateDelayStatistics"
> 
> The reland has a lot of additional DCHECKS for easier debugging,
> so in debug builds it will actually be a ~2x slowdown compared to the old code.
> The excessive DCHECKS should be removed in a followup CL.
> 
> Bug: webrtc:9439
> Change-Id: I8389cd84f1ca12c29cc6993f0d2cf7e6d7dd8379
> Reviewed-on: https://webrtc-review.googlesource.com/101761
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24821}

TBR=terelius@webrtc.org,asapersson@webrtc.org,kron@webrtc.org

Change-Id: I98c4c96d552858d0299d49993e9b9be6a6204dfe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9439
Reviewed-on: https://webrtc-review.googlesource.com/101860
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24825}
2018-09-25 12:05:39 +00:00
Danil Chapovalov
e7e156d7a6 Support first packet reduction in H264 packetizer
Bug: webrtc:9680
Change-Id: I73c9a5acecdf8dd82347be602bbfd7412c9610c5
Reviewed-on: https://webrtc-review.googlesource.com/99804
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24823}
2018-09-25 11:43:29 +00:00
Johannes Kron
9def3b45ef Second reland of "Optimize execution time of RTPSender::UpdateDelayStatistics"
The reland has a lot of additional DCHECKS for easier debugging,
so in debug builds it will actually be a ~2x slowdown compared to the old code.
The excessive DCHECKS should be removed in a followup CL.

Bug: webrtc:9439
Change-Id: I8389cd84f1ca12c29cc6993f0d2cf7e6d7dd8379
Reviewed-on: https://webrtc-review.googlesource.com/101761
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24821}
2018-09-25 10:20:21 +00:00
Johannes Kron
6ea771933d Potential bug in RTP packet.
A potential bug was introduced in "Refactor to remove direct memory
dependency on kMaxId" due to a memory restructuring,
commit c5744b8b21b627213286f1b6f2c65da5df9ce8d0

Bug: webrtc:7990
Change-Id: I0dcaf47e1c1e361d65220c278a2326d6f2686af7
Reviewed-on: https://webrtc-review.googlesource.com/101642
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24814}
2018-09-24 23:21:25 +00:00
Johannes Kron
965e7942a3 Add sanity checks to UpdateDelayStatistics and patch unit tests.
RtpPacket::UpdateDelayStatistics was previously optimized with several
sanity checks added. These sanity checks caused many of the unit tests
in peerconnection_integration_unittests to fail and the CL was therefore
reverted. This CL contains the sanity checks along with patches so that
the unit tests pass.

Bug: webrtc:9439
Change-Id: Ia5f5e8b125e5f3f4b79d433e2282901143530a25
Reviewed-on: https://webrtc-review.googlesource.com/99802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24813}
2018-09-24 23:13:02 +00:00
Johannes Kron
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
Danil Chapovalov
55bb00e4c2 Refactor RtpPacketizerH264 tests
Add helper function to create test data,
reduce amount of unrelated details

Reduced complicated logic in tests, in particular
move most of expectation inside the tests from helpers.

Bug: None
Change-Id: I53f29a70989086c7628a0b112a45ec4567b40bf9
Reviewed-on: https://webrtc-review.googlesource.com/100380
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24760}
2018-09-18 08:57:47 +00:00
Danil Chapovalov
585d1aac17 Register video rtp header extensions in rtp_rtcp by uri
Remove function for converting uri into ExtensionType
This removes one of the lists of all supported extensions

Bug: webrtc:7472
Change-Id: I0c27239d91ef14ca4a3aa0c00588fa2b9cf10e0c
Reviewed-on: https://webrtc-review.googlesource.com/100523
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24752}
2018-09-17 10:02:30 +00:00
philipel
ef615ea7a3 Added is_last_packet_in_frame to match is_first_packet_in_frame.
Today we use |is_first_packet_in_frame| to know when a frame begins and the
|markerBit| to know when it ends, but the markerbit does not actually mark the
end of a frame, it marks the end of a picture.

Bug: webrtc:9361
Change-Id: Icc70e6075590cdc31e875a4eb9d489868adbb67c
Reviewed-on: https://webrtc-review.googlesource.com/100160
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24722}
2018-09-13 11:07:10 +00:00
philipel
b3e42a4948 Write and parse the generic video descriptor.
Bug: webrtc:9361
Change-Id: Id129a6ab7a86641c6e80827458ef0c40c5640855
Reviewed-on: https://webrtc-review.googlesource.com/99542
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24718}
2018-09-13 09:00:50 +00:00
Minyue Li
1a80018a3c Avoid wrong parsing of padding length and its use in NetEq simulation.
Bug: b/113648474, webrtc:9730
Change-Id: Ieff7ab8697f5c8742548897a9b452a20b0bd2e7c
Reviewed-on: https://webrtc-review.googlesource.com/98461
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24703}
2018-09-12 11:23:03 +00:00
Danil Chapovalov
fd5fbd0b58 Cleanup RtpPacketizerH264 constructor
Merge SetPayloadData into constructor.
Add TODO to support first_packet_reduction_len

Bug: webrtc:9680
Change-Id: I65e771848e0ffe8968cd084840e77afc0152caeb
Reviewed-on: https://webrtc-review.googlesource.com/99505
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24702}
2018-09-12 11:11:18 +00:00
Johnny Lee
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
Danil Chapovalov
c7fff58d1e Allow nullptr retransmition rate limiter
as iniditcation retransmission shouldn't be limited because of rate.

Bug: None
Change-Id: I579261749515260b972631779dadc6349dfcab46
Reviewed-on: https://webrtc-review.googlesource.com/99541
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24690}
2018-09-11 14:50:54 +00:00
Per Kjellander
e9da5f27a4 Reland "Decrease complexity of RtpPacketHistory::GetBestFittingPacket.""
This reverts commit 49b2c3c4c43359bc86d8510d29d117f3d7a621a3.

Original CL description:
Decrease complexity of RtpPacketHistory::GetBestFittingPacket.
Use a map of packet sizes in RtpPacketHistory instead of looping through the whole history for every call

patch set 1 contains the initial submit from https://webrtc-review.googlesource.com/c/src/+/98882
new patch sets contains the modification.

The problem with the initial submit was the assumption that packets are removed
from history in the same order as they are added which is not always true.

Bug: webrtc:9731
Change-Id: Ic2c8905a0f47287fc46e53f41a019a4c69c3dd8e
Reviewed-on: https://webrtc-review.googlesource.com/99460
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24687}
2018-09-11 13:54:30 +00:00
Per Kjellander
49b2c3c4c4 Revert "Decrease complexity of RtpPacketHistory::GetBestFittingPacket."
This reverts commit 54caa4b68a0acb81c7f6ef60ffec45b473a7e1a2.

Reason for revert: Crashes on some perf tests.
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fclient.webrtc.perf%2FLinux_Trusty%2F7170%2F%2B%2Frecipes%2Fsteps%2Fwebrtc_perf_tests%2F0%2Fstdout



Original change's description:
> Decrease complexity of RtpPacketHistory::GetBestFittingPacket.
> Use a map of packet sizes in RtpPacketHistory instead of looping through the whole history for every call.
> 
> Bug: webrtc:9731
> Change-Id: I44a4f6221e261a6cb3d5039edfa7556a102ee6f1
> Reviewed-on: https://webrtc-review.googlesource.com/98882
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24662}

TBR=danilchap@webrtc.org,sprang@webrtc.org,perkj@webrtc.org

Change-Id: Id183cd31a67117e9614d163e4388131fd88de07d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9731
Reviewed-on: https://webrtc-review.googlesource.com/99440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24665}
2018-09-11 05:35:09 +00:00
Per Kjellander
54caa4b68a Decrease complexity of RtpPacketHistory::GetBestFittingPacket.
Use a map of packet sizes in RtpPacketHistory instead of looping through the whole history for every call.

Bug: webrtc:9731
Change-Id: I44a4f6221e261a6cb3d5039edfa7556a102ee6f1
Reviewed-on: https://webrtc-review.googlesource.com/98882
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24662}
2018-09-10 19:09:29 +00:00
Danil Chapovalov
e4f8b38091 Allow different header extensions in 1st packet of a video frame
no behavior changes expected.
Different exension for the 1st packet will be added in a follow-up

Bug: webrtc:9680
Change-Id: I8c853b2710d58df579aeb4b029b42210310423cc
Reviewed-on: https://webrtc-review.googlesource.com/98843
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24655}
2018-09-10 12:53:10 +00:00
Danil Chapovalov
067818fe85 Move RtcpTransceiver deletion of copy and assign methods to public section
Bug: chromium:881453
Change-Id: Iff5c522b983af018c1308649887a1121519c73ea
Reviewed-on: https://webrtc-review.googlesource.com/98981
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24652}
2018-09-10 12:23:19 +00:00
Danil Chapovalov
0b445c6271 Cleanup RtpPacketizerVp9
Merge SetPayloadData into constructor.
Reuse payload size split function

Bug: webrtc:9680
Change-Id: If230a4ea901b5cdd6a376f8dd2db48e94d6dca36
Reviewed-on: https://webrtc-review.googlesource.com/98866
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24635}
2018-09-07 22:01:46 +00:00
Danil Chapovalov
792df6b4b9 Make RtcpTransceiver destructor non-blocking
At cost of removing assumption callbacks can't be used after destructor.

Bug: webrtc:8239
Change-Id: Id79f7553528cf6c102d3ee0bf7aa2de5b0437d2a
Reviewed-on: https://webrtc-review.googlesource.com/98860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24632}
2018-09-07 15:34:08 +00:00
Philip Eliasson
d52a1a6971 Reland "Remove RTPVideoHeader::vp8() accessors."
This reverts commit 1811c04f22a26da3ed2832373a5c92a9786420c3.

Reason for revert: Downstream projects fixed.

Original change's description:
> Revert "Remove RTPVideoHeader::vp8() accessors."
> 
> This reverts commit af7afc66427b0e9109e7d492f2805d63d239b914.
> 
> Reason for revert: Break downstream projects.
> 
> Original change's description:
> > Remove RTPVideoHeader::vp8() accessors.
> > 
> > Bug: none
> > Change-Id: Ia7d65148fb36a8f26647bee8a876ce7217ff8a68
> > Reviewed-on: https://webrtc-review.googlesource.com/93321
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > 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@{#24626}
> 
> TBR=danilchap@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,holmer@google.com
> 
> Change-Id: I3f7f19c0ea810c0fd988c59e6556bbea9b756b33
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: none
> Reviewed-on: https://webrtc-review.googlesource.com/98864
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24628}

TBR=danilchap@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,holmer@google.com

Change-Id: I9246f36e638108ae4fc46c1ae4559c8205d50fc1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/98841
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24629}
2018-09-07 13:04:07 +00:00
Philip Eliasson
1811c04f22 Revert "Remove RTPVideoHeader::vp8() accessors."
This reverts commit af7afc66427b0e9109e7d492f2805d63d239b914.

Reason for revert: Break downstream projects.

Original change's description:
> Remove RTPVideoHeader::vp8() accessors.
> 
> Bug: none
> Change-Id: Ia7d65148fb36a8f26647bee8a876ce7217ff8a68
> Reviewed-on: https://webrtc-review.googlesource.com/93321
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> 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@{#24626}

TBR=danilchap@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,holmer@google.com

Change-Id: I3f7f19c0ea810c0fd988c59e6556bbea9b756b33
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/98864
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24628}
2018-09-07 12:36:17 +00:00
philipel
af7afc6642 Remove RTPVideoHeader::vp8() accessors.
Bug: none
Change-Id: Ia7d65148fb36a8f26647bee8a876ce7217ff8a68
Reviewed-on: https://webrtc-review.googlesource.com/93321
Reviewed-by: Niels Moller <nisse@webrtc.org>
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@{#24626}
2018-09-07 12:01:19 +00:00
Danil Chapovalov
fa5ec8d20d Use signed integers for limiting packet size in video packetizers
Using signed integers allows to centralize checking of edge cases
in RtpPacketizer::SplitAboutEqually and
reduce chance of hitting issues with size_t underflow

Bug: webrtc:9680
Change-Id: Ic05bf0a9565a277c4608f43061ca46cf44e82d08
Reviewed-on: https://webrtc-review.googlesource.com/98602
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24618}
2018-09-07 09:24:18 +00:00
Jonas Olsson
366a50c4ef Remove simple stringstream usages.
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.

The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.

Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
2018-09-06 12:53:19 +00:00
Danil Chapovalov
af8c03673b Cleanup RtpPacketizerGeneric
merge SetPayloadData into constructor.
Reuse payload size split function

Bug: webrtc:9680
Change-Id: If8de22b2fc39c0bdf6d60154ecaaf19e2dccdcc9
Reviewed-on: https://webrtc-review.googlesource.com/97640
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24598}
2018-09-06 08:54:55 +00:00
Danil Chapovalov
f0076d31f8 Add RtcpTransceiver::Stop to allow non-blocking destruction
As downside it disallows to destroy RtcpTransceiver on the TaskQueue
without prio call to the Stop function

BUG: webrtc:8239
Change-Id: I236b9aff7a0746044dd764c619174cc5ac03d27f
Reviewed-on: https://webrtc-review.googlesource.com/98120
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24587}
2018-09-05 15:42:24 +00:00
Danil Chapovalov
bace3a4bd6 in RtpPacketizer add support for first packet reduction length
Bug: webrtc:9680
Change-Id: I68580a8a44cbbf932ed0b2448b31169b26df5044
Reviewed-on: https://webrtc-review.googlesource.com/98005
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24584}
2018-09-05 14:54:02 +00:00
Danil Chapovalov
1a6a4f3d62 Replace WeakPtr with CancelablePeriodicTask in RtcpTransceiverImpl
This allow to destroy the RtcpTransceiverImpl off the task queue
with assumption it is destroyed after task queue.
i.e. it allows to post destruction of RtcpTransceiverImpl to the TaskQueue without waiting.

BUG: webrtc:8239
Change-Id: I4bea7a6d2edc97061ebd00f2f275c1ab827bc3c5
Reviewed-on: https://webrtc-review.googlesource.com/97160
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24574}
2018-09-05 11:06:16 +00:00
Mirko Bonadei
55d5ef0048 Remove clang:find_bad_constructs suppression from logging (part 1).
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

Bug: webrtc:9251, webrtc:163
Change-Id: I443c0c77cef30a3f5712f72ce88db80d1fb29874
Reviewed-on: https://webrtc-review.googlesource.com/96642
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24567}
2018-09-05 08:33:17 +00:00
Danil Chapovalov
376e1147e6 in RtpPacketizerVp8 factor out payload splitter function
so that it can be shared between different packetizers
and thus easier to extend

Bug: webrtc:9680
Change-Id: Ie5e904ad27afb8dd2ed35ef9e009f7f408017b2f
Reviewed-on: https://webrtc-review.googlesource.com/97661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24555}
2018-09-04 15:28:05 +00:00
Danil Chapovalov
8a3c166fff Cleanup RtpPacketizerVP8 tests
Remove partition support for test helper and from tests.
Merge Init function into constructor
Replace extra macroses in favor of Bit helper function
Replace extra members in favor of local variables
Remove fixture

Bug: None
Change-Id: Ibf1600dda9f59abe5afd2bbe40c3e232a2d269ea
Reviewed-on: https://webrtc-review.googlesource.com/96940
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24508}
2018-08-31 08:13:45 +00:00
Rasmus Brandt
260b4151c8 Revert "Reland "Optimize execution time of RTPSender::UpdateDelayStatistics""
This reverts commit 7bcd2a98be3fa8c246866d6b343c7f94752977b3.

Reason for revert: peerconnection_unittests fails on downstream test runner.

Original change's description:
> Reland "Optimize execution time of RTPSender::UpdateDelayStatistics"
> 
> The reland has a lot of additional DCHECKS for easier debugging,
> so in debug builds it will actually be a ~2x slowdown compared to the old code.
> The excessive DCHECKS should be removed in a followup CL.
> 
> Bug: webrtc:9439
> Change-Id: I493de337bf20c998aa32c2532212cac85c5517fb
> Reviewed-on: https://webrtc-review.googlesource.com/96641
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24501}

TBR=terelius@webrtc.org,asapersson@webrtc.org,philipel@webrtc.org

Change-Id: Ia48444d2a7647cf826ef93b4720f6d7ff9a712c3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9439
Reviewed-on: https://webrtc-review.googlesource.com/96960
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24502}
2018-08-30 15:24:47 +00:00
Bjorn Terelius
7bcd2a98be Reland "Optimize execution time of RTPSender::UpdateDelayStatistics"
The reland has a lot of additional DCHECKS for easier debugging,
so in debug builds it will actually be a ~2x slowdown compared to the old code.
The excessive DCHECKS should be removed in a followup CL.

Bug: webrtc:9439
Change-Id: I493de337bf20c998aa32c2532212cac85c5517fb
Reviewed-on: https://webrtc-review.googlesource.com/96641
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24501}
2018-08-30 14:42:02 +00:00
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
Danil Chapovalov
8d1b582f33 Cleanup RtpPacketizerVp8
Merge SetPayloadData into constructor,
Remove payload size member because now used only during construction.
Remove member that should be constant

Bug: None
Change-Id: Ib2083439f466ad9151ce8e54fceede6cef51d955
Reviewed-on: https://webrtc-review.googlesource.com/96740
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24491}
2018-08-30 10:16:33 +00:00
Kári Tristan Helgason
55e7785686 Remove kVideoCodecUnknown completely.
Followup to cl https://webrtc-review.googlesource.com/92086

Bug: webrtc:8136
Change-Id: I077d18ac6def475d05424ff0c5e545c939cd8e61
Reviewed-on: https://webrtc-review.googlesource.com/95642
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24475}
2018-08-29 08:48:40 +00:00
Danil Chapovalov
f7f8a1f176 Cleanup RtpPacketizer interface
merge construction and call to SetPayloadData
Add NumPackets instead of SetPayloadData
Remove virtual ToString() as unused
move CHECK(rtp_video_header) from RtpPacketizer::Create to RtpSenderVideo::SendVideo

Bug: webrtc:9680
Change-Id: I074644e048c797eb836f79979df363fe1ea0075e
Reviewed-on: https://webrtc-review.googlesource.com/96543
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24474}
2018-08-29 08:44:08 +00:00
philipel
bf2b620cc8 Convert VP8 descriptor to generic descriptor.
Also adds a running picture id for the old generic format when
kVideoCodecGeneric is used (behind "WebRTC-GenericPictureId" field trial).

Bug: webrtc:9361
Change-Id: I6f232a2663bb60257c97ed3473eb07044d325b90
Reviewed-on: https://webrtc-review.googlesource.com/94842
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24449}
2018-08-27 13:03:20 +00:00