This makes the code path where packets are directly owned by PacedSender
rather that being temporarily put in the RtpPacketHistory the default.
Functionally, this should essentially be a noop, with only minor timing
differences.
The old code-path will stay around for a short while and then be
removed once we are certain there are no regressions.
Bug: webrtc:10633
Change-Id: Id6360dea48fd0c9d46fde6f5eee93726d4f11d13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146212
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28660}
If the SSRC of an RTP module is changed at runtime, we may get conflicts
with packets already there. Eg:
* Put seq# 123 in the history for SSRC 1.
* Change the SSRC to 2.
* Send a NACK for seq# 123 from SSRC 2.
Currently, we will respond with the packet belonging to SSRC 1 (and not
if the NACK specifies SSRC 1, to boot).
We can gen similar issues if the sequence number is changed, where
half frame are left in the buffer.
In these cases, the stream is likely being reset so we should just
clear the packet history too.
Bug: webrtc:10794
Change-Id: I28147c2532cf1c78840d4808c4366d4a647541f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145729
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28658}
This CL fixes two issues related to the TransmissionOffset header
extension and the new (not yet active) pacer mode.
Previously capture time (if unset) would be populated when put into the
packet history before entering the pacer. Since the pacer now owns the
packets, this does not occur until packet is actually sent, if at all.
Capture has really nothing to do with the packet history, this should
be set by the RtpSender pre-pacing instead.
Furthermore, for retransmissions the old path would take the capture
time from the original packet, build the RTX-wrapped retransmission and
set the toffset extension of the RTX packet using that captured capture
time. Since RTX packets are now fully built before the pacer, this does
not work, and we need to transfer the capture time from the original to
the RTX packet instead.
Bug: webrtc:10633
Change-Id: I031e8b6cc4ab20fb094dbd46720829b78951e7f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146218
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28657}
This is intended to by used for visualizing catagorical data, i.e. mapping
numerical enum values to string labels.
Bug: webrtc:10623
Change-Id: Ic9c3da9a3874f479c07412f394a774ae90fd3d7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145408
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28656}
This adds the RemoteEstimate rtcp packet and wires it up to GoogCC where
it's used to improve congestion controller behavior.
The functionality is negotiated using SDP.
It's added with a field trial that allow disabling the functionality in
case there's any issues.
Bug: webrtc:10742
Change-Id: I1ea8e4216a27cd2b00505c99b42d1e38726256c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146602
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28654}
We rather have an unmet expectation than let the test crash.
Bug: webrtc:10827
Change-Id: I9e3d2dfb7cb856976305cd50377a71a2ed2ab4b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146700
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28651}
According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and
https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent
statistic should not include headers or padding.
Similarly, according to
https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes
received are calculated the same way as bytes sent (eg. not including
padding or headers).
This change stops adding padding and headers to these statistics.
Bug: webrtc:8516,webrtc:10525
Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28647}
This reverts commit d978cb43c238ca24b2320acd7b656f446b906101.
Reason for revert: It breaks perf tests: https://ci.chromium.org/p/webrtc/builders/perf/Perf%20Android32%20(L%20Nexus4)/1561
Original change's description:
> Record audio/video bytes sent in analyzer stream stats.
>
> For each SSRC report, record the number of bytes sent for that stream
> and expose them in analyzer stats. These numbers can be used to
> determine useful metrics such as total media throughput (by adding the
> bytes sent for all streams) and overhead (by subtracting that amount
> from the total bytes sent to the network).
>
> Bug: webrtc:9719
> Change-Id: I977bbd40acdd0a1ec64763ddd55a642b9a50f309
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146240
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28637}
TBR=mbonadei@webrtc.org,mellem@webrtc.org,titovartem@webrtc.org
Change-Id: I3e46307dd6ef121b9377b93fc8d9fa788245ea5f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146605
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28646}
For each SSRC report, record the number of bytes sent for that stream
and expose them in analyzer stats. These numbers can be used to
determine useful metrics such as total media throughput (by adding the
bytes sent for all streams) and overhead (by subtracting that amount
from the total bytes sent to the network).
Bug: webrtc:9719
Change-Id: I977bbd40acdd0a1ec64763ddd55a642b9a50f309
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146240
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28637}
The code was doing nothing except for triggering thread sanitizer,
since concurrent writes weren't guarded:
* ReadRecordedData() through webrtc_audio_module_rec_thread
* InitPlayout() through main thread
Bug: webrtc:9751
Change-Id: I7ecf4fa436ff0695e5b998d7e3f159fb6c7e9214
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146216
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28636}
Lifetime issue: "webrtc_audio_module_rec_thread" was still accessing
AudioTransport mock at and after its destruction.
Bug: webrtc:9751
Change-Id: I24308077cdeb77e570b8ec74098f1ae3397b7155
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146217
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28635}
It's not currently used and it complicates receive side estimation.
Bug: webrtc:10742
Change-Id: Iaa3c86807c7b637aea3ff393e728dc91eac23db6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145724
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28623}
Emscripten does not support C++11 thread_local but does support
the pthread TLS API.
Bug: None
Change-Id: Ia21895148d1df7652579d086d9e1c0c53d7a85f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145441
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28621}
WebRTC has been migrated to Abseil Flags.
Bug: webrtc:10616
Change-Id: Id4a363429ccd2dd55c0dff00c9490c15124fdccc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144631
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28620}
This tool is unused, this CL removes it in order to reduce the cost
of the maintenance (in the last 2 years only maintenance commits have
been landed in this directory).
Bug: None
Change-Id: Ieec113bc25c480405d32e284a0456572758352e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146204
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28619}
Rationale:
* More explicit (you won't miss that when glancing at the code).
* More consistent (see MAYBE_* in other tests).
* Allow to re-activate tests via CLI (--gtest_also_run_disabled_tests).
* Tests won't wrongly show up as PASSING (bug/webrtc:10819),
since they won't show up at all.
Bug: webrtc:9778
Change-Id: Ic32e18cb8ee2352def95206c2aa66e1dea0cc1e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146200
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28617}
This CL removes the field trial left in place as a kill-switch in case
there were any regressions related to selecting payload padding based
on the likelihood of being useful instead of matching size.
It also removes the functionality that was only enabled with the
kill-switch active.
The feature has been default-on since June 23rd 2019:
https://webrtc.googlesource.com/src.git/+/214f54365ec210db76218a35ead66c9ce23e068e
Since we have not observed any issues, let's clean this code up.
Bug: webrtc:8975
Change-Id: I7f49fe354227b3f6566a250332e56b6d70fe2f09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145821
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28616}
This CL fixes two things related to the (not yet active) new
PacedSender code path:
1. Make sure BWE header extensions are properly populated for all
padding packets.
2. When generating padding, don't hold the RtpSender critsect when
accessing the RtpPacketHistory as this may lead to a lock order
inversion.
Bug: webrtc:10633
Change-Id: I8650fbf5dafddbeae61837d2137338163e1c48ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145723
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28613}
SVC support is limited:
During SVC testing there is no SFU, so framework will try to emulate SFU
behavior in regular p2p call. Because of it there are such limitations:
* if |target_spatial_index| is not equal to the highest spatial layer
then no packet/frame drops are allowed.
If there will be any drops, that will affect requested layer, then
WebRTC SVC implementation will continue decoding only the highest
available layer and won't restore lower layers, so analyzer won't
receive required data which will cause wrong results or test failures.
Bug: webrtc:10138
Change-Id: I079566260ca9f1815935bce365d1bca10766663a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144882
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28612}
