Non-paced audio should be sent "immediately", but in several places that
was determined by looking at the current time - which can lead to
inconsistencies.
E.g. if a packet is enqueued and ProcessPackets() is called 1ms later,
the pacer should see NextSendTime() as 1ms ago, so that buffer levels
are cleared at the right pace.
Bug: webrtc:10809
Change-Id: I04a169f3df3e28a5c8ef7fa8a042b9c482c307ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172845
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31002}
Especially with the TaskQueuePacerSender, it is very common that a
single packet is added to the queue and then immediately removed as
it gets sent to the network.
This CL adds a fast-path for that case, that avoid creating book-
keeping in the form of stream-priorities and timestamp sets etc.
Functionally, it should be a noop, but hopefully it can save a few
CPU cycles.
Bug: webrtc:10809
Change-Id: Idaa06b4f8d1da444fce78cc742e2ab52f9efe815
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172090
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31001}
This CL adds two kill-switches to the AEC3 code to be used as
safe fallbacks to increase AEC transparency.
The changes have been shown to be bitexact for a test dataset.
Bug: webrtc:11487,chromium:1067597
Change-Id: I7f9f78db4964990bcdfa9adae6ef36a56bce7224
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172840
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30998}
This allows clients to exclude the transient suppression submodule from WebRTC builds, by defining WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR.
The changes have been shown to be bitexact for a test dataset (when the flag is _not_ defined.)
No-Try: True
Bug: webrtc:11226, webrtc:11292
Change-Id: I6931c82a280a9b40a53ee1c2a9820ed9e674a9a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171421
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30978}
This allows the user to run audioproc_f with various field trials set.
The approach is copied from test/test_main_lib.cc.
Tested:
1. Verified bitexactness vs ToT audioproc_f on a large dataset of aecdumps
2. Ran it with flags --aec=1 --force_fieldtrials="WebRTC-Aec3ClampInstQualityToZeroKillSwitch/Enabled/WebRTC-Aec3ClampInstQualityToOneKillSwitch/Enabled/" and verified in GDB that the AEC3 config was changed accordingly.
No-Try: True
Bug: webrtc:5298
Change-Id: I70eec7777f70893b36af33794a5842f67d56af31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172623
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30976}
Currently, audioproc_f crashes on a DCHECK as the data vector of Int16Frame is not resized.
Bug: webrtc:5298
Change-Id: I897cf0fce07e0ed2c0a365a965fa50fd3d8ddd18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172624
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30969}
This CL adds a number of kill-switches to the AEC3 code to be used as
safe fallbacks to increase AEC transparency.
The changes have been shown to be bitexact for a test dataset.
Bug: webrtc:11475,chromium:1066836
Change-Id: Ibebcbbfbbd958cb6fcc6993247e3030fa65b582c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172600
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30964}
r30936 accidentally made it defualt off. This reverts to the old
behavior by default.
Bug: webrtc:8975, chromium:1066442
Change-Id: I415d2f74bb7321f17b4039ca43cbd53c3e3725f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172445
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30954}
EnqueuePackets() would reset the last process time if the queue
and media budgets were empty. This was done without reducing the
padding debt.
The result of this was that, given an existing debt, and an interval
between audio packets that is less than the drain time for the padding
debt, padding would not be sent at all.
Now, before adding a new packet, we reduce the padding debt if the
packet queue is empty.
Bug: webrtc:10809
Change-Id: I116169522c215257febd32e17abab45f1a7d609f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171808
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30949}
Add a new frame interface to be used by frame transformers in Insertable
Streams. TransformableFrameInterface will replace
video_coding::EncodedFrame in a follow up CL, once downstream
dependecies are updated to use the new interface.
Until the functions using video_coding::EncodedFrame are removed from
the API, the video sender and receiver frame transformer delegates call
both function versions to avoid breaking tests downstream.
The TransformableFrameInterface will be used for both audio and video
frame transformers in follow-up CLs.
Bug: webrtc:11380
Change-Id: I9389a8549c156e13b1d8c938ff51eaa69c502f33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171863
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30941}
Specifically, if dynamic pacer (i.e. TaskQueuePacer) was enabled while
AccountForAudio was set to true, the pacer would pace audio packets.
This should only happen when the WebRTC-Pacer-BlockAudio field trial is
enabled.
Bug: webrtc:10809
Change-Id: If5edc77de88ca9866abeb3b47e171df50673299e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172082
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30938}
This started to happen after turning on "gn analyze" on trybots. It
looks like this code was never built on MSVC trybots.
This CL tries to avoid the type deduction.
Error:
quality_assessment/sound_level.cc(103):
error C3535: cannot deduce type for 'const auto *' from '_FwdIt'
with
[
_FwdIt=std::_Array_iterator<int16_t,1440>
]
Bug: webrtc:11262
Change-Id: Iea7cf2ec62f1d0edfcf6ceac169c92050339d3c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172088
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30933}
This reverts commit 8e8b36a94a7a7a1fd0f8093979a406afa56e18c1.
Reason for revert: The CL has been improved with the following changes,
- Fixed negotiation of send/receive only clients.
- Handles the implicit assumption that any H264 decoder also can
decode H264 constraint baseline.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}
Change-Id: I834ed48ee78d04922c73e2836165e476925e1cc5
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168605
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30932}
This reverts commit 7b201012bc7f7cd95249b8314d0d7ebabe966d8b.
Reason for revert: Seems to work, but need to get low bw tests working first
Original change's description:
> Flip histograms to true by default, fix unit in isac_fix_test.
>
> Requires downstream changes for all WebRTC perf tests, and
> a corresponding recipe change so isac_fix_test starts using the new
> flow.
>
> Bug: chromium:1029452
> Change-Id: I8918fca9bef003d365037c1c6bf7c55747dfed99
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170633
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30906}
TBR=phoglund@webrtc.org,mbonadei@webrtc.org
Change-Id: I96c2309cd71be14c5a27b515736a32f1b256453c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029452
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171865
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30913}
Requires downstream changes for all WebRTC perf tests, and
a corresponding recipe change so isac_fix_test starts using the new
flow.
Bug: chromium:1029452
Change-Id: I8918fca9bef003d365037c1c6bf7c55747dfed99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170633
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30906}
Instead of passing only the local- and remote network IDs the whole
NetworkRoute is forwarded to TransportFeedbackAdapter that can then
use more detailed information to distinguish routes.
Bug: webrtc:11434
Change-Id: I48f36aa1177822d20c2b556dcc2275f6145ed845
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171581
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30895}
The WebRTC-SendSideBwe-WithOverhead field trial requires audio
encoders to properly implement the
AudioEncoder::GetFrameLengthRange() function. Thic CL implements
the function for all audio encoders in WebRTC in preparation for
making that function pure virtual in the interface.
Bug: webrtc:11427
Change-Id: Ieab6b6c72c62af6ac9525a20fcb39bd477079551
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171503
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30890}
Pulse related code should still be disabled unless
WEBRTC_ENABLE_LINUX_PULSE is defined but it will always be
compiled.
Bug: None
Change-Id: If8a03aae445a8c73c3c347e275c5996368fe3088
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171513
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30886}
This CL renames the internal macros LINUX_ALSA and LINUX_PULSE and adds
the prefix WEBRTC_. Since these macros are internal to WebRTC, it is
better to use a prefix.
Bug: None
Change-Id: I2a07fa569a4da168006cc36f32e4dbb98a75814b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171514
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30885}
This CL fixes a few issues where the reported fraction of frames
allocated to various temporal layers could be incorrect:
* In LibvpxVp8Encoder, calling GetEncoderInfo() while not initialized,
or when first configuring with temporal layers and then without,
could trigger incorrect fps allocations.
* In VP9 when different spatial layers have different max framerates,
the layer fps should be compared to the layer with the highest
configured fps, not codec_.maxFramerate which is updated to the
current input fps on SetRates().
* In EncoderBitrateAdjuster, just warn and ignore if a layer has
non-zero bps but zero fps, rather than passing down the chain and
risk weird behavior or divide by zero.
Bug: b/152040235
Change-Id: I548fb3e099b1ec9f536a7b93313fb40c4d32e596
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171516
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30880}
--- Background ---
The webrtc::VideoSendStream::StreamStats are converted into
VideoSenderInfo objects which turn into "outbound-rtp" stats objects in
getStats() (or "ssrc" objects in legacy getStats()).
StreamStats are created for each type of substream: RTP media streams,
RTX streams and FlexFEC streams - each with individual packet counters.
The RTX stream is responsible for retransmissions of a referenced media
stream and the FlexFEC stream is responsible for FEC of a referenced
media stream. RTX/FEC streams do not show up as separate objects in
getStats(). Only the media streams become "outbound-rtp" objects, but
their packet and byte counters have to include the RTX and FEC counters.
--- Overview of this CL ---
This CL adds MergeInfoAboutOutboundRtpSubstreams(). It takes
StreamStats of all kinds as input, and outputs media-only StreamStats
- incorporating the RTX and FEC counters into the relevant media
StreamStats.
The merged StreamStats objects is a smaller set of objects than the
non-merged counterparts, but when aggregating all packet counters
together we end up with exact same packet and count as before.
Because WebRtcVideoSendStream::GetVideoSenderInfo() currently aggregates
the StreamStats into a single VideoSenderInfo (single "outbound-rtp"),
this CL should not have any observable side-effects. Prior to this CL:
aggregate StreamStats. After this CL: merge StreamStats and then
aggregate them.
However, when simulcast stats are implemented (WIP CL:
https://webrtc-review.googlesource.com/c/src/+/168120) each RTP media
stream should turn into an individual "outbound-rtp" object. We will
then no longer aggregate all StreamStats into a single "info". This CL
unblocks simulcast stats by providing StreamStats objects that could be
turned into individual VideoSenderInfos.
--- The Changes ---
1. Methods added to RtpConfig to be able to easily tell the relationship
between RTP, RTX and FEC ssrcs.
2. StreamStats gets a StreamType (kMedia, kRtx or kFlexfec) that
replaces the booleans (is_rtx, is_flexfec).
3. "referenced_media_ssrc" is added to StreamStats, making it possible
to tell which kRtx/kFlexFec stream stats need to be merged with which
kMedia StreamStats.
4. MergeInfoAboutOutboundRtpSubstreams() added and used.
Bug: webrtc:11439
Change-Id: Iaf9002041169a054ddfd32c7ea06bd1dc36c6bca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170826
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30869}
As far as I can tell, every call site already populates this field, so
we can now remove it.
Bug: webrtc:8975
Change-Id: I58515dd16d4ad8bf8872077b67a67f6e92e7b542
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171222
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30857}
The error paths free the memory referenced by each pointer in the
struct, but if the pointers are not initialized, random memory belonging
to other parts of the program could be freed instead. Zero out the
entire struct as soon as it is allocated to ensure that nothing is freed
if there is nothing to free.
Bug: webrtc:11446
Change-Id: I8a2985d1388477339351aa03107ee68925372d49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171121
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30852}
This CL adds namespaces to those files remaining within APM that do not
have any such.
BUG=webrtc:5298
Change-Id: I710b3d2a3644bea9d4bdffef0d77883b30303338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171111
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30850}
This CL changes so that the AGC legacy C code is built as C++.
The CL also
-removes #defines from the header files.
-adds namespaces
-removes unused code.
To simplify the review, the CL is partitioned into different patchsets
where each comprising of one step in the modification of the code
(e.g., patch set 1 performs the renaming of the .c files to .cc).
Bug: webrtc:5298
Change-Id: I362b17bde91142b2f2166acba4f2f888efd50fa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171064
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30847}
This CL renames the shadow filter in AEC3 to have the more accurate name
coarse filter.
The CL consists of 3 main initial patch sets, designed to simplify
the review:
1) Replaces "shadow" with "coarse" and adds a fall-back functionality
to support the old filter naming.
2) Renames the files according to the new naming.
3) Performs a "git cl format"
Bug: webrtc:8671
Change-Id: I28d6041d0d34e85f8f8048d004b44a1a5f07bb07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170981
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30846}
This CL renames the main filter in AEC3 to have the more accurate name
refined filter.
The CL consists of 3 main initial patch sets, designed to simplify
the review:
1) Replaces "main" with "refined" and adds a fall-back functionality
to support the old filter naming.
2) Renames the files according to the new naming.
3) Performs a "git cl format"
Bug: webrtc:8671
Change-Id: Ifd0aab34e291736a2250e0986348404618630b1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170825
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30843}
A follow-up CL will move the rms_level.* files into the new target.
Bug: webrtc:11226
Change-Id: I59579b026346f627c0a2739d25f90c12bffbf248
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171102
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30840}
