Commit graph

9 commits

Author SHA1 Message Date
Artem Titov
913cfa76ec Use backticks not vertical bars to denote variables in comments for /modules/rtp_rtcp
Bug: webrtc:12338
Change-Id: I52eb3b6675c4705e22f51b70799ed6139a3b46bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227164
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34686}
2021-08-09 15:51:03 +00:00
Doudou Kisabaka
fe6595f006 Include all RTP packet infos from the mix list when updating the audio frame for mixing.
Users of the mixer can use this information to determine which sources were included in the frame.

Bug: webrtc:12745
Change-Id: I11a8e3b1f4e8f95eb870336cad8dd082330bdf02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217768
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Doudou Kisabaka <doudouk@google.com>
Cr-Commit-Position: refs/heads/master@{#34035}
2021-05-18 11:05:37 +00:00
Johannes Kron
f7de74c58c Use Timestamp to represent packet receive timestamps
Before this CL, timestamps of received packets were rounded
to the nearest millisecond and stored as int64_t. Due to the
rounding it sometimes happened that timestamps later in the
pipeline that are not rounded seem to occur even before the
video frame was received.

Change-Id: I92d8f3540b23baae2d4a1dc6a7cb3f58bcdaad18
Bug: webrtc:12722
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216398
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33916}
2021-05-04 13:16:54 +00:00
Ruslan Burakov
d51cc7bd71 Add absolute capture time property to rtp sources.
This part of the effort to implement A/V sync metric.

Bug: webrtc:10739
Change-Id: I4adba1b99b37b31868168e37d9aa8e03f8ea6d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159886
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ruslan Burakov <kuddai@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29849}
2019-11-20 18:50:45 +00:00
Mirko Bonadei
1b575417b3 Always pass arguments to INSTANTIATE_TEST_SUITE_P.
Passing an empty arg is working at the moment but it is not
guaranteed to continue to work in the future.

This CL has been generated with:
git grep -l "INSTANTIATE_TEST_SUITE_P(," | xargs sed -i \
    "s/INSTANTIATE_TEST_SUITE_P(,/INSTANTIATE_TEST_SUITE_P(All,/g"

Bug: None
Change-Id: Icd2fb9d9d29aed5d692a234124bd990d0f097db4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153890
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29282}
2019-09-24 08:56:24 +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
Chen Xing
e08648dc70 Add AbsoluteCaptureTime to RtpPacketInfo.
This change stores the optional `AbsoluteCaptureTime` header extension in `RtpPacketInfo` so that we later can consume it in `SourceTracker`.

Bug: webrtc:10739
Change-Id: I975e8863117fcda134535cd49ad71079a7ff38ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148068
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28790}
2019-08-07 10:12:56 +00:00
Chen Xing
12d64deb6c Remove sequence_number from RtpPacketInfo.
This change removes sequence_number from RtpPacketInfo since it's currently not used.

Bug: webrtc:10668
Change-Id: I9b45c7476457df1d18173f37c421374108678931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141873
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28281}
2019-06-14 11:21:42 +00:00
Chen Xing
9c16af7eb7 Add a tracker for RTCRtpContributingSource and RTCRtpSynchronizationSource.
This change adds a new SourceTracker class that can do spec-compliant tracking of RTCRtpContributingSource and RTCRtpSynchronizationSource when frames are delivered to the RTCRtpReceiver's MediaStreamTrack for playout. It will replace the existing spec-incompliant ContributingSources.

Bug: webrtc:10545 webrtc:10668
Change-Id: I961adaba09d6337f2f36b301a4fabcd20de65271
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140948
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28249}
2019-06-12 12:11:55 +00:00