This is a preparatory step in deleting the ChannelManager class.
Also delete some declarations whose implementation was previously removed.
Bug: webrtc:13931
Change-Id: I8764c00fa696932e79fcfe17550ef2490d6a1ed1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262804
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36923}
implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)
This is similar to totalProcessingDelay
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.
This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.
Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as
totalAssemblyTime of type double
Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.
This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.
framesAssembledFromMultiplePacket of type unsigned long
Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
For such frames the totalAssemblyTime is incremented.
BUG=webrtc:13986
Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36922}
(Support parsing TWCC which nack sequnce numbers greater than the last
one received. Don't silently drop unrecognized/malformed RTCP packets.)
Bug: webrtc:14078
Change-Id: I34a0deabfdb5f36b988919cfcc9159197435756c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262800
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36921}
for video dealing with both the case where there is no common media
codec as well as only a red/ulpfec/flexfec codec in common for video
and only RED/CN in common for audio
BUG=webrtc:4957,webrtc:14069
Change-Id: I1c888b4f77199aade8122051c31b690dc2fd5925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262642
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36920}
Structs with user-declared constructors are not aggregates and cannot
be initialized with designated initializers. Remove declarations that
don't actually affect anything.
Bug: chromium:1284275
Change-Id: Ib45ea334d7be28bfa7bbce132985612f0e6ecd10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262820
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36916}
Instead of using field trials in BitrateProber for probe duration, use values provided in ProbeClusterConfig from GoogCC.
Field trials are instead read in ProbeController.
To avoid having to do a thread jump for every ProbeClusterConfig, RtpPacketPacer interface is changed to RtpPacketPacer::CreateProbeClusters(std::vector<ProbeClusterConfig>
Deprecates field trial "WebRTC-Bwe-ProbingConfiguration"
Change-Id: I3991e4b54770601855a3af2d6a16678f11d41c31
Bug: webrtc:14027
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261265
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36911}
RTCPReceiver::main_ssrc_ and local_media_ssrc() represent the same
value but could get out of sync when `set_media_ssrc()` was called.
Instead of using main_ssrc_, just use the local_media_ssrc() accessor.
Bug: webrtc:11993
Change-Id: I2b034287e6b6025d9b0d2affa391a168896a614b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262663
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36905}
A packet arrival history is used to store the timing of incoming packets and tracks the earliest and latest packets by taking the difference between rtp timestamp and arrival time. The history is windowed to 2 seconds by default. The packet arrival history will replace the relative arrival delay tracker in a follow up cl.
The playout delay is estimated by taking the difference between the current playout timestamp and the earliest packet arrival in the history. This method works better when DTX is used compared to the buffer level filter that it replaces.
The threshold for acceleration is changed to be the maximum of the target delay and the maximum packet arrival delay in the history. This prevents any acceleration immediately after an underrun and gives some time to adapt the target delay to new network conditions.
The logic when to decode the next packet after a packet loss is also changed to do concealment for the full loss duration unless the delay is too high.
The new mode is default disabled and can be enabled using a field trial.
Bug: webrtc:13322,webrtc:13966
Change-Id: Idfa0020584591261475b9ca350cc7c6531de9911
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259820
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36899}
VP9 automaticResizeOn is disabled if more than one spatial layer is configured via scalability mode.
Bug: webrtc:13960
Change-Id: I7c6351bca6d2f32bcc7391894e8dcc9e74ca2050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261315
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36897}
When ADM is terminated and its state requires reinitialized, VoipCore::initialized_ field will falsely skip required reinitializing.
Bug: webrtc:14054
Change-Id: Ibeb4987a7e9763b8e40926acc4d7eaabde7a3478
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261924
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Tim Na <natim@google.com>
Commit-Queue: Tim Na <natim@google.com>
Cr-Commit-Position: refs/heads/main@{#36893}
The new TaskQueuePacedSender has been default-on in code since M97, and
there are no further usages of it that I can find. Let's clean this up!
The PacingController and associated tests will be cleaned up in a
follow-up cl.
Bug: webrtc:10809
Change-Id: I0cb888602939add953415977ee79ff0b3878fea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258025
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36890}
This CL first restricts Metrics to be retrievable when the socket is
created. This avoids having most fields as optional and makes it
easier to add more metrics.
Secondly, the peer implementation is moved into Metrics.
Bug: webrtc:13052
Change-Id: I6cb53eeef3f84ac34f3efc883853338f903cc758
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262256
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36888}
This reverts commit 181ea6e414c5982015ce161e6368120be3658ec4.
Reason for revert: Breaks downstream project. Kári will help to land it next week.
Original change's description:
> Add a prefix for objc category.
>
> According to the Google Objective-C style [1], category names should
> start with an appropriate prefix. WebRTC has some category definitions
> for system interfaces, but it doesn't use prefixes.
>
> $ otool -ov WebRTC.framework/WebRTC | grep -E "^[0-9a-z]{16} 0x[0-9a-z]+ __OBJC_._CATEGORY" | grep -v "_RTC"
> 0000000002160840 0x217c3c0 __OBJC_$_CATEGORY_UIDevice_$_H264Profile
> 0000000002160850 0x21808b8 __OBJC_$_CATEGORY_AVCaptureSession_$_DevicePosition
> 0000000002160858 0x2180968 __OBJC_$_CATEGORY_NSString_$_StdString
> 0000000002160860 0x21809c8 __OBJC_$_CATEGORY_NSString_$_AbslStringView
>
> To avoid conflicts, prefix the names and methods of those categories.
> Also remove sdk/objc/Framework/Classes/Common/NSString+StdString.h as
> it is not used by any other files.
>
> [1] https://google.github.io/styleguide/objcguide.html#category-naming
>
> Bug: webrtc:13884
> Change-Id: I2cf2742af198ab4e0bfb15c0476d72971e50ceee
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262341
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36880}
Bug: webrtc:13884
Change-Id: I85257088e4a3a62e01ff925ab5e77af83b078ef3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262420
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Auto-Submit: Artem Titov <titovartem@webrtc.org>
Owners-Override: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36885}
According to the Google Objective-C style [1], category names should
start with an appropriate prefix. WebRTC has some category definitions
for system interfaces, but it doesn't use prefixes.
$ otool -ov WebRTC.framework/WebRTC | grep -E "^[0-9a-z]{16} 0x[0-9a-z]+ __OBJC_._CATEGORY" | grep -v "_RTC"
0000000002160840 0x217c3c0 __OBJC_$_CATEGORY_UIDevice_$_H264Profile
0000000002160850 0x21808b8 __OBJC_$_CATEGORY_AVCaptureSession_$_DevicePosition
0000000002160858 0x2180968 __OBJC_$_CATEGORY_NSString_$_StdString
0000000002160860 0x21809c8 __OBJC_$_CATEGORY_NSString_$_AbslStringView
To avoid conflicts, prefix the names and methods of those categories.
Also remove sdk/objc/Framework/Classes/Common/NSString+StdString.h as
it is not used by any other files.
[1] https://google.github.io/styleguide/objcguide.html#category-naming
Bug: webrtc:13884
Change-Id: I2cf2742af198ab4e0bfb15c0476d72971e50ceee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262341
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36880}
The tests in rtc_unittests_objc are old gtest based tests and
The tests in sdk_unittests are XCTest based tests.
The objc tests in rtc_unittest are causing problems [1],
so I think it's time to combine the two types of objc tests.
Renaming the files to match the existing sdk_unittests and
removing the use of gtest helper functions (eg, EXPECT_EQ)
are planned for follow-up CLs.
[1] https://webrtc-review.googlesource.com/c/src/+/261724/9
Bug: webrtc:8382
Change-Id: I9308b35f654c4479c9df72bf7cd7f4032e267eb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262380
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36878}
When Apple introduced Swift, they renamed all the AppKit Objective-C
constants to have a shared prefix for ease of bridging, and deprecated
all the old names. These new names were introduced in the macOS 10.12
and 10.14 SDKs, but compile down to the same constant values so they
can be deployed all the way back to 10.0.
Bug: chromium:1322548
Change-Id: Ib3cacbe493e97ca43dd180047eacbd2866d70016
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262180
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Auto-Submit: Avi Drissman <avi@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36874}
