It hasn't been changed in any meaningful way since 2013, the same year
it was created.
Bug: webrtc:8396
Change-Id: I5633188134f71f24311fbd3098d046632fc4ee3a
Reviewed-on: https://webrtc-review.googlesource.com/101563
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24816}
A potential bug was introduced in "Refactor to remove direct memory
dependency on kMaxId" due to a memory restructuring,
commit c5744b8b21b627213286f1b6f2c65da5df9ce8d0
Bug: webrtc:7990
Change-Id: I0dcaf47e1c1e361d65220c278a2326d6f2686af7
Reviewed-on: https://webrtc-review.googlesource.com/101642
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24814}
RtpPacket::UpdateDelayStatistics was previously optimized with several
sanity checks added. These sanity checks caused many of the unit tests
in peerconnection_integration_unittests to fail and the CL was therefore
reverted. This CL contains the sanity checks along with patches so that
the unit tests pass.
Bug: webrtc:9439
Change-Id: Ia5f5e8b125e5f3f4b79d433e2282901143530a25
Reviewed-on: https://webrtc-review.googlesource.com/99802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24813}
This reverts commit 3a9731ff2f871bcc798ce9760fa51d81cfd5e755.
Reason for revert: Seems to cause crashes in Chrome browser tests, see for example https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8934487169011818016/+/steps/browser_tests__retry_with_patch_/0/logs/WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsOfferEcdsaAnswerEcdsa/0
Original change's description:
> Bug in histogram metric reporting.
>
> A (actually several weeks) while ago, we noticed an error with the
> WebRTC.Audio.Agc2.EstimatedNoiseLevel histogram. It always reported
> the value 0. Here is why:
>
> The histogram bins go from 0 to 100. But the value logged is dBFS. It is
> always less than or equal to 0. This CL changes the bins.
>
> Bug: webrtc:7494
> Change-Id: I45fd122e98f9396f9871bc965a708987bd1815f6
> Reviewed-on: https://webrtc-review.googlesource.com/101340
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24800}
TBR=saza@webrtc.org,aleloi@webrtc.org
Change-Id: I84883f73710b7e13aa90ee29b140acfc417f109f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7494
Reviewed-on: https://webrtc-review.googlesource.com/101701
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24809}
This avoids crashing if an application resets the starting bitrate
before adding streams to the call.
Bug: webrtc:9586
Change-Id: I8d31aba1f4fee40c67c8930f5a32d17700ccadc3
Reviewed-on: https://webrtc-review.googlesource.com/101680
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24808}
It will soon lose the ability to do so.
Also, the ACM no longer creates comfort noise encoders for us, so
don't bother testing that.
Bug: webrtc:8396
Change-Id: I24a12e26bef142f9f8e7532b764f28572e0c6ace
Reviewed-on: https://webrtc-review.googlesource.com/101640
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24803}
Speeds up adaptation of the matched filter of the delay estimator by
allowing the estimated echo and the error signal (microphone minus
estimated echo) to be saturated. Only microphone saturation pauses
the filter adaptation.
Bug: webrtc:9773
Change-Id: I8b8400539fde3ee821f36a95818bece02ddd626b
Reviewed-on: https://webrtc-review.googlesource.com/101341
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24802}
When two-byte header extensions are enabled, kMaxId will change from 15
to 255. This CL is a refactor to remove the direct dependency between
memory allocation and kMaxId.
Bug: webrtc:7990
Change-Id: I38974a9c705eb6a0fdba9038a7d909861587d98d
Reviewed-on: https://webrtc-review.googlesource.com/101580
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24801}
A (actually several weeks) while ago, we noticed an error with the
WebRTC.Audio.Agc2.EstimatedNoiseLevel histogram. It always reported
the value 0. Here is why:
The histogram bins go from 0 to 100. But the value logged is dBFS. It is
always less than or equal to 0. This CL changes the bins.
Bug: webrtc:7494
Change-Id: I45fd122e98f9396f9871bc965a708987bd1815f6
Reviewed-on: https://webrtc-review.googlesource.com/101340
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24800}
Right after a volume decrease, the echo path estimate is overestimated and, as a side effect, the nearend signal is also overestimated. Due to that, the suppression gains are kept high avoiding the suppression of echoes. In this CL the neared power spectrum estimation is limited to a level given by the power spectrum or the microphone input signal. Additionally, the minimum gain that is computed inside the suppressor is also modified. Instead of using the nearend power spectrum that is now bounded, the power spectrum of the signal after the linear echo canceler is used.
Bug: webrtc:9762
Change-Id: Ia24cd2ce248f2c2ba124711b75acff3b8c5cfa9f
Reviewed-on: https://webrtc-review.googlesource.com/100720
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24796}
Replace calls to .str() which copies with .Release which moves in cases where that's safe.
This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"
Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
VP9 frame rate controller is supposed to be used in screen mode only
but it was partially enabled in normal video mode. This restricts use
of VP9 frame rate controller to screen mode.
Bug: chromium:884164
Change-Id: Ie2eaa31f3364a8abccbc4171007708cf7040fc38
Reviewed-on: https://webrtc-review.googlesource.com/100424
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24769}
Add helper function to create test data,
reduce amount of unrelated details
Reduced complicated logic in tests, in particular
move most of expectation inside the tests from helpers.
Bug: None
Change-Id: I53f29a70989086c7628a0b112a45ec4567b40bf9
Reviewed-on: https://webrtc-review.googlesource.com/100380
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24760}
Uninline RTPFragmentaion functions
fix RTPFragmentation move constructor and assign operators (was recursive for win)
replace assert with rtc::dchecked_cast
Remove unused includes and dependencies.
Fix other targets that used those includes transitively instead of directly
Bug: None
Change-Id: I647cb1eda107dc7d87d25234095545bc2842fa40
Reviewed-on: https://webrtc-review.googlesource.com/100500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24759}
Remove function for converting uri into ExtensionType
This removes one of the lists of all supported extensions
Bug: webrtc:7472
Change-Id: I0c27239d91ef14ca4a3aa0c00588fa2b9cf10e0c
Reviewed-on: https://webrtc-review.googlesource.com/100523
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24752}
Original CL:
https://webrtc-review.googlesource.com/c/src/+/97603
- Changes EchoCancellationImpl to inherit privately from
EchoCancellation.
- Removes usage of AudioProcessing::echo_cancellation() inside most of
the audio processing module and unit tests.
- Default-enables metrics collection in AEC2.
The CL breaks audioproc_f backwards compatibility: It can no longer
use all recorded settings (drift compensation, suppression level), but
prints an error message when such settings are encountered.
Revert CL:
https://webrtc-review.googlesource.com/c/src/+/100305
Bug: webrtc:9535
TBR: gustaf@webrtc.org
Change-Id: I9248046b3a6a82df6221e502481836948643a991
Reviewed-on: https://webrtc-review.googlesource.com/100461
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24749}
If the bandwidth is just on the edge of being able to enable a new
stream, the keyframe generated when it is enabled might be large enough
to trigger an overuse and force the stream off again.
To avoid toggling, this CL adds hysteresis so that the available
bandwidth needs to be above X% to start bitrate in order to enable the
stream. It will be shut down once available bitrate falls below the
original enabling bitrate.
For screen content, X defaults to 35.
For realtime content, X defaults to 0.
Both can be individually modified via field trials.
Bug: webrtc:9734
Change-Id: I941332d7be7f2a801d13d9202b2076d330e7df32
Reviewed-on: https://webrtc-review.googlesource.com/100308
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24745}
Also cleans up usage of the new RoundRobinPacketQueue to reduce code
bloat.
Bug: webrtc:8288
Change-Id: I90f17a4422b32c1d4e2d7d5065573157346d6a0b
Reviewed-on: https://webrtc-review.googlesource.com/100306
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24744}
Now checks the preferred channel configuration and requests implicit channel
upmixing (audio engine extends from 2 to N channels internally) if the
preferred number of channels is larger than two; i.e., initialize the
stream in stereo even if the preferred configuration is multi-channel.
To summarize: with this CL, it is now possible to use e.g. a 7.1 headset
with a native WebRTC client. All internal processing in WebRTC will be in
stereo, and the audio device will be opened up in stereo as well to match
WebRTC. Before this change, we would open up the audio device using 8
channels but that was not supported by WebRTC.
Bug: webrtc:9265
Change-Id: I1530fee28c4b8b5cda29ab6baf8d65fd391d935d
Reviewed-on: https://webrtc-review.googlesource.com/98421
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24743}
This CL:
-Adds the option to log the aec3 parameters used for a simulation.
-Cleans up the logging of the custom setting of aec3 parameters to
instead rely on the newly added logging.
Bug: webrtc:8671
Change-Id: If73a73d08e5a5077416033ded598a83fb1ade3e0
Reviewed-on: https://webrtc-review.googlesource.com/100381
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24742}
The test is refitted to use the AudioProcessingStats struct to get
reference data.
The old metrics do not map entirely injectively to the new ones, so the
reference protobuf and files are updated as well.
Bug: webrtc:9535
Change-Id: I546dca2979380e03895af0077bfc77ffd24abe36
Reviewed-on: https://webrtc-review.googlesource.com/100100
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24740}
This change enables NetEq to use the packet concealment audio (aka
PLC) produced by a decoder. The change also includes a new API to the
AudioDecoder interface, which lets the decoder implementation generate
and deliver concealment audio.
Bug: webrtc:9180
Change-Id: Icaacebccf645d4694b0d2d6310f6f2c7132881c4
Reviewed-on: https://webrtc-review.googlesource.com/96340
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24738}
This CL removes some indirection and moves some constants. This
is done to simplify understanding and debugging of the code.
Bug: webrtc:9718
Change-Id: Ibe2b1da0163b4c97ffd1a5bc157f6aa59582d697
Reviewed-on: https://webrtc-review.googlesource.com/98240
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24732}
This CL removes some indirection This is done to simplify
understanding and debugging of the code.
Bug: webrtc:9718
Change-Id: I48974d161213b9ef8fc5912bd3dc3f9d85ddfa66
Reviewed-on: https://webrtc-review.googlesource.com/100302
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24730}
This reverts commit 1a03960e632a04e2ff866f2048cc36146af83e41.
Reason for revert: breaks downstream projects.
Original change's description:
> Remove APM internal usage of EchoCancellation
>
> This CL:
> - Changes EchoCancellationImpl to inherit privately from
> EchoCancellation.
> - Removes usage of AudioProcessing::echo_cancellation() inside most of
> the audio processing module and unit tests.
> - Default-enables metrics collection in AEC2.
>
> This CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (drift compensation, suppression level), but
> prints an error message when such settings are encountered.
>
> Some code in audio_processing_unittest.cc still uses the old interface.
> I'll handle that in a separate change, as it is not as straightforward
> to preserve coverage.
>
> Bug: webrtc:9535
> Change-Id: Ia4d4b8d117ccbe516e5345c15d37298418590686
> Reviewed-on: https://webrtc-review.googlesource.com/97603
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24724}
TBR=gustaf@webrtc.org,saza@webrtc.org
Change-Id: Ifdc4235f9c5ee8a8a5d32cc8e1dda0853b941693
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/100305
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24729}
Currently we use the NetworkStatistics to monitor these metrics, but because these get reset on every call, this makes it impossible to use them for other purposes.
Bug: webrtc:9667
Change-Id: If648085f04d2d58aae263cff5b9491bcad373a96
Reviewed-on: https://webrtc-review.googlesource.com/99740
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24727}
This CL:
- Changes EchoCancellationImpl to inherit privately from
EchoCancellation.
- Removes usage of AudioProcessing::echo_cancellation() inside most of
the audio processing module and unit tests.
- Default-enables metrics collection in AEC2.
This CL breaks audioproc_f backwards compatibility: It can no longer
use all recorded settings (drift compensation, suppression level), but
prints an error message when such settings are encountered.
Some code in audio_processing_unittest.cc still uses the old interface.
I'll handle that in a separate change, as it is not as straightforward
to preserve coverage.
Bug: webrtc:9535
Change-Id: Ia4d4b8d117ccbe516e5345c15d37298418590686
Reviewed-on: https://webrtc-review.googlesource.com/97603
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24724}
This CL removes a set of DCHECKs in AudioDeviceBuffer (ADB) where the goal has been
to ensure that some methods are called on one and the same native I/O thread.
The implementation of the ADB is platform independent but the underlying (driving)
audio components differ between platforms. This combination has shown to generate complex
corner cases such as:
- OS dependent I/O-thread(s) changes while audio is active
- OS dependent audio device changes and it leads to restart of native I/O threads
- Start/Stop of audio has different timing depending on platform and possibly also usage of
JNI and/or emulators.
To summarize: the gain of maintaining the current strict thread checking (in Debug mode)
is not worth all the efforts trying to resolve complex dynamic cases where the native
I/O threads changes ID.
TBR=glaznev
Bug: b/115385789
Change-Id: I681c89adec497a18b97d2a40421c04ea218fd919
Reviewed-on: https://webrtc-review.googlesource.com/100200
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24723}
Today we use |is_first_packet_in_frame| to know when a frame begins and the
|markerBit| to know when it ends, but the markerbit does not actually mark the
end of a frame, it marks the end of a picture.
Bug: webrtc:9361
Change-Id: Icc70e6075590cdc31e875a4eb9d489868adbb67c
Reviewed-on: https://webrtc-review.googlesource.com/100160
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24722}
as iniditcation retransmission shouldn't be limited because of rate.
Bug: None
Change-Id: I579261749515260b972631779dadc6349dfcab46
Reviewed-on: https://webrtc-review.googlesource.com/99541
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24690}
This reverts commit 49b2c3c4c43359bc86d8510d29d117f3d7a621a3.
Original CL description:
Decrease complexity of RtpPacketHistory::GetBestFittingPacket.
Use a map of packet sizes in RtpPacketHistory instead of looping through the whole history for every call
patch set 1 contains the initial submit from https://webrtc-review.googlesource.com/c/src/+/98882
new patch sets contains the modification.
The problem with the initial submit was the assumption that packets are removed
from history in the same order as they are added which is not always true.
Bug: webrtc:9731
Change-Id: Ic2c8905a0f47287fc46e53f41a019a4c69c3dd8e
Reviewed-on: https://webrtc-review.googlesource.com/99460
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24687}
Set target frame rate of spatial layer equal to minimum of two: maximum
frame rate of layer (SpatialLayer::maxFramerate) and maximum frame rate
of codec (VideoCodec::maxFramerate).
Bug: webrtc:9740, webrtc:9739, chromium:882358
Change-Id: I34f36e7fd2889f0417474347abab5327fa2d9d7c
Reviewed-on: https://webrtc-review.googlesource.com/99501
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24686}
