NetEq currently only passes `jitterBufferDelay` to `getStats()`. We need its paired `jitterBufferEmittedCount` denominator stat for the calculations to be accurate.
Bug: webrtc:10192
Change-Id: I655aea629026ce9101409c2e0f18c2fa57a1c3ab
Reviewed-on: https://webrtc-review.googlesource.com/c/117320
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#26276}
Configuration for AudioProcessing::voice_detection() is moving into
AudioProcessing::Config, to get rid of the pointer-to-submodule
interfaces (such as voice_detection()).
Bug: webrtc:9947
Change-Id: Ia64ae996a43d44423aa0d612a3f1185b52a3e534
Reviewed-on: https://webrtc-review.googlesource.com/c/116067
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26216}
The following APIs on AudioCodingModule are deprecated with this CL:
static int NumberOfCodecs();
static int Codec(int, CodecInst*);
static int Codec(const char*, CodecInst*, int, size_t);
static int Codec(const char*, int, size_t);
absl::optional<CodecInst> SendCodec() const;
bool RegisterReceiveCodec(int, const SdpAudioFormat&);
int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&);
int UnregisterReceiveCodec(uint8_t);
int32_t ReceiveCodec(CodecInst*);
absl::optional<SdpAudioFormat> ReceiveFormat();
As well as this method on RtpRtcp module:
int32_t RegisterSendPayload(const CodecInst&);
Bug: webrtc:7626
Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/113816
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26025}
setting max reordering recently has been fix to actually set it.
(https://webrtc-review.googlesource.com/c/src/+/111752)
Another recent change fix stats to skip counting large sequence number jumps as packet loss
(https://webrtc-review.googlesource.com/c/src/+/111962)
max reordering thresholds affects how packet loss is calculated.
Packet loss is then reported to remote sending participant in rtcp receiver reports.
Sender uses packet loss mostly for stats, but also e.g. for opus fec adjustment.
Setting threshold to zero de-facto imply all packets should be considered in order.
That bug was mitigated by two other bugs mentioned above
This change increase threshold to default 50 packets aligning it with Video receiver
and unblocks (re)landing 2nd fix
Bug: b/120482366
Change-Id: Iadda0c2148ed84dd83c01183cfe9285568db4e29
Reviewed-on: https://webrtc-review.googlesource.com/c/113064
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25905}
Note that this value will override the minimum delay that is used for audio/video sync.
Bug: webrtc:10053
Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33
Reviewed-on: https://webrtc-review.googlesource.com/c/112121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25805}
This CL decouples //rtc_base:rtc_base_tests_utils from gunit by
moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h
(which depends on gunit helpers) to their own build target.
It also removes some unused dependencies in the WebRTC build graph.
Bug: None
Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623
Reviewed-on: https://webrtc-review.googlesource.com/c/111861
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25769}
This will be used in a later CL to use the link capacity field in the
update to control the Opus encoder.
Bug: webrtc:9718
Change-Id: If2ad16a8f4656e8cdf10c33f5fb060ef7ca5caba
Reviewed-on: https://webrtc-review.googlesource.com/c/111510
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25761}
Replaced by interface ChannelSendInterface, implemented by ChannelSend
and mock class.
Thread checkers are moved to ChannelSend, which is also moved into
the anonymous namespace and exposed only via a function CreateChannelSend.
Bug: webrtc:9801
Change-Id: I73b2e2bfb67c1a5077709f2379533bf315babad9
Reviewed-on: https://webrtc-review.googlesource.com/c/111240
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25684}
So far ANA was not available for media transport interface. With recent changes to media transport, we can now account for packet overhead, network route (ip/tcp/udp/turn overheads) and we can also use bandwidth estimate from the media transport.
Bug: webrtc:9719
Change-Id: I98c9a09dd418b763c339ee2ee05592e164cf9199
Reviewed-on: https://webrtc-review.googlesource.com/c/110367
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25677}
'configured_bitrate_bps_' is accessed from different threads in
SetBitrate and GetBitrate (one comes back from OnNetworkRouteChange
callback, the other one is used in GetStats()) and so it should be
protected by a critical section.
Bug: webrtc:10010
Change-Id: I029baa729e0203b9f2d180d8835d61add26e6cef
Reviewed-on: https://webrtc-review.googlesource.com/c/111281
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25675}
This is a preparation for deleting ChannelSendProxy. Signature is
changed on a couple of methods. Unused methods
EnableAudioNetworkAdaptor, DisableAudioNetworkAdaptor,
SetReceiverFrameLengthRange and RtpRtcpModulePtr are deleted. Some
methods are demoted to private: SendData, SendRtp, SendRtcp,
PreferredSampleRate, Sending, and OnOverheadChanged.
Bug: webrtc:9801
Change-Id: I982e72418a32e66fb5de410350b1bfebd9a3219c
Reviewed-on: https://webrtc-review.googlesource.com/c/110605
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25666}
Replaced by an interface ChannelReceiveInterface, implemented
by ChannelReceive and the corresponding mock class.
Moved thread checkers to ChannelReceive. That class is moved to the
anonymous namespace in the .cc file, and exposed only via a function
CreateChannelReceive.
Bug: webrtc:9801
Change-Id: Iecacbb1858885bf86da9484f2422e53323dbe87a
Reviewed-on: https://webrtc-review.googlesource.com/c/110610
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25665}
This is a follow up of https://webrtc-review.googlesource.com/c/src/+/43201.
Issue 43201 didn't do the job properly.
1. The audio rtcp report interval is not properly hooked up.
2. We don't need to propagate audio rtcp interval into video send stream or vice versa.
3. We don't need to propagate rtcp report interval to any receiving streams.
Bug: webrtc:8789
Change-Id: I1f637d6e5173608564ef0702d7eda6fc93b3200f
Reviewed-on: https://webrtc-review.googlesource.com/c/110105
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25610}
This is a preparation for deleting ChannelReceiveProxy, Changes
signature of some methods, and demotes methods OnData and
OnReceivedPayloadData to private.
Bug: webrtc:9801
Change-Id: Ib00a80c6482ed5238f3cc8233860c70f11484df9
Reviewed-on: https://webrtc-review.googlesource.com/c/110606
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25599}
This reverts commit 61c6e5643e7ea058e653956980a90e033249c055.
Reason for revert: downstream projects prepared for this change
Original change's description:
> Revert "Isolating APM API build target: making :api an actual target."
>
> This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.
>
> Reason for revert: breaking downstream
>
> Original change's description:
> > Isolating APM API build target: making :api an actual target.
> >
> > This CL is part of a refactoring work to unblock other CLs
> > that would generate a circular dependency when including
> > modules/audio_processing. It will also allow to easily move
> > the APM interface part under //api.
> >
> > More in detail, this change moves the APM interface files from
> > the build target modules/audio_processing to
> > modules/audio_processing:api. It also adds :api as dependency
> > where needed.
> >
> > Bug: webrtc:9535
> > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25539}
>
> TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
>
> Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9535
> Reviewed-on: https://webrtc-review.googlesource.com/c/109820
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25540}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109884
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25547}
This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.
Reason for revert: breaking downstream
Original change's description:
> Isolating APM API build target: making :api an actual target.
>
> This CL is part of a refactoring work to unblock other CLs
> that would generate a circular dependency when including
> modules/audio_processing. It will also allow to easily move
> the APM interface part under //api.
>
> More in detail, this change moves the APM interface files from
> the build target modules/audio_processing to
> modules/audio_processing:api. It also adds :api as dependency
> where needed.
>
> Bug: webrtc:9535
> Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25539}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109820
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25540}
This CL is part of a refactoring work to unblock other CLs
that would generate a circular dependency when including
modules/audio_processing. It will also allow to easily move
the APM interface part under //api.
More in detail, this change moves the APM interface files from
the build target modules/audio_processing to
modules/audio_processing:api. It also adds :api as dependency
where needed.
Bug: webrtc:9535
Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
Reviewed-on: https://webrtc-review.googlesource.com/c/109501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25539}
This patch adds (optional) csrc to ContributingSources.
This will be used if using virtual audio ssrc, since
the audio level is otherwise unaccessible in that configuration.
BUG=webrtc:3333
Change-Id: Ied263b8f0850553cd637fd6bead373ed4252fd1e
Reviewed-on: https://webrtc-review.googlesource.com/c/109281
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25516}
This enables PeerConnection tests to use LoopbackMediaTransport to test
data-channel-over-media-transport code.
Also changes LoopbackMediaTransport to invoke callbacks asynchronously.
This is more accurate, as these callbacks are triggered by network
events. The caller should not block while the callback executes.
Since LoopbackMediaTransport is used for testing, it provides a
FlushAsyncInvokes() method which may be used to ensure that callbacks
occur deterministically (eg. before checking that data has been
received).
Bug: webrtc:9719
Change-Id: Ib8ea9aebf4a0ad3d5934a6fe4ab33432c68523fd
Tbr: stefan@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/109060
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25489}
And put codecs/cng/webrtc_cng.h in a non-public build target while
we're at it.
Bug: webrtc:8396
Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c
Reviewed-on: https://webrtc-review.googlesource.com/c/108782
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25486}
It was previously logged in Call, but streams are not always created
with the full configuration, which caused header extensions to be
missing from the log.
Bug: webrtc:9885
Change-Id: I86c0424004c4629ebab0f6b155b83fb90e15b131
Reviewed-on: https://webrtc-review.googlesource.com/c/108601
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25483}
This CL consistently use:
* relative paths for WebRTC dependent targets (test_support)
* absolute paths for shared dependent targets (abseil)
This is a necessary (but insufficient) step to build WebRTC tests
from Chromium tree (rtc_include_tests=true), since test/ doesn't
sit anymore in the top level directory.
We also make sure that target declarations and uses are
consistent in regard to build_with_chromium flag.
Bug: webrtc:9943
Bug: webrtc:9855
Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108720
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25445}
This reverts commit 9a0662ac7e4a3bc6b3a316397a7fdf25f0025d35.
Reason for revert: breaks some av sync perf tests
Original change's description:
> Use only first payload timestamp for RTCP SR generation for audio
>
> Since now RTP rate is set correctly for audio, there's no need to
> use the very last data packet rtp/capture timestamps for generating
> RTCP SR packets.
>
> Using only one (first) packet timestamp eliminates the jitter between
> rtp and capture timestamps for audio. This jitter comes from the fact
> that capture timestamp for audio is unknown and we generate bogus
> timestamp at arbitrary, non-constant offset from the real capture time.
>
> Bug: webrtc:9905
> Change-Id: I855556184cfe994be39ab7780836a050f5a38c35
> Reviewed-on: https://webrtc-review.googlesource.com/c/108580
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25430}
TBR=danilchap@webrtc.org,ilnik@webrtc.org,ossu@webrtc.org
Change-Id: I208a659379b1075258ee94613e42afd9aebe4754
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9905
Reviewed-on: https://webrtc-review.googlesource.com/c/108623
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25435}
