Commit Graph

5488 Commits

Author SHA1 Message Date
80260c226d Switch VCMRttFilter to use TimeDelta
* Moved into its own GN target
* Switched the internal buffer types to absl::InlinedVector as arrays
  are tricky to use with types that do not have default constructors.
* Update fields arnd variables to use style guide.
* Use constexpr for formerly const fields.
* Adds unit tests.

Change-Id: I476ae8491f0f9878c176e7b87a5133942c3d79f7
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36133}
2022-03-04 16:03:28 +00:00
a2ee9234b4 Migrate to Timestamp and TimeDelta types in RtpPacketHistory
Bug: webrtc:13757
Change-Id: Ie542fca50b97fe9dc450e45da40f05e2b66c7da5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252981
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36132}
2022-03-04 15:02:58 +00:00
15ee87fe0e Use VideoCodec complexity to determine AV1 encoder cpu_speed.
Bug: webrtc:13744
Change-Id: Ib6d62dcdf7346d886c0aca09735c7d5c1f3e2455
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Michael Horowitz <mhoro@google.com>
Commit-Queue: Michael Horowitz <mhoro@google.com>
Cr-Commit-Position: refs/heads/main@{#36125}
2022-03-03 19:06:17 +00:00
45623a3c0f Remove operator= from VCMJitterEstimator and VCMRttFilter
Change-Id: I70846d9cdc17d904585a18983acee7980292e62e
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253301
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36122}
2022-03-03 15:26:27 +00:00
b663cfaae4 Cleanup RtpPacketHistory from unused features
history no longer used for storing unsent packets and for legacy pacer.

Bug: None
Change-Id: I639c37de66857a64c620e80df6288fa6ce8326d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253260
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36120}
2022-03-03 14:30:27 +00:00
66557e1af3 Revert "remove NV12 to I420 conversion in webrtc AV1 Encoder."
This reverts commit 9558ab41eb4de39c62cda2dd1e559f5814a3a0c7.

Reason for revert: speculative revert: breaks downstream project

Original change's description:
> remove NV12 to I420 conversion in webrtc AV1 Encoder.
>
> libaom supports for NV12 inputs for encoding av1 stream. It will reduce
> unnecessary conversion from NV12 to I420 format.
> (https://bugs.chromium.org/p/aomedia/issues/detail?id=3232&q=3232&can=2)
>
> Bug: webrtc:13746
> Change-Id: I1407227d1690b3f63cb6581eef5d587e5f418892
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251920
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Shuhai Peng <shuhai.peng@intel.com>
> Cr-Commit-Position: refs/heads/main@{#36111}

Bug: webrtc:13746
Change-Id: Ie928f7f5b5992337a9d186fa70b7fdec20a33f00
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253122
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Owners-Override: Artem Titov <titovartem@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36114}
2022-03-02 16:01:28 +00:00
5cd7d2aa0f audioproc_f: fix AGC1 digital adaptive flag bug
- missing negation causes the opposite behavior when
  `analog_agc_disable_digital_adaptive` is used
- flag replaced with `analog_agc_use_digital_adaptive_controller`
  which is less error-prone

Bug: webrtc:7494
Change-Id: If9e0ba4fc9e539c73269faf9096ca782620dac6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251322
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36113}
2022-03-02 15:50:57 +00:00
d6cdf80072 Use Timestamp and TimeDelta in VCMTiming
* Switches TimestampExtrapolator to use Timestamp as well.

Bug: webrtc:13589
Change-Id: I042be5d693068553d2e8eb92fa532092d77bd7ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249993
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36112}
2022-03-02 15:07:25 +00:00
9558ab41eb remove NV12 to I420 conversion in webrtc AV1 Encoder.
libaom supports for NV12 inputs for encoding av1 stream. It will reduce
unnecessary conversion from NV12 to I420 format.
(https://bugs.chromium.org/p/aomedia/issues/detail?id=3232&q=3232&can=2)

Bug: webrtc:13746
Change-Id: I1407227d1690b3f63cb6581eef5d587e5f418892
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251920
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Shuhai Peng <shuhai.peng@intel.com>
Cr-Commit-Position: refs/heads/main@{#36111}
2022-03-02 14:18:36 +00:00
c6d3a7a691 Ensure returned delay based estimate from probe can be clamped by
AimdRateControl

AimdRateControl can potentially clamp bitrate in SetEstimate.
DelayBasedBwe::MaybeUpdateEstimate should therefore check the result before using the probe bitrate.
Otherwise, BWE may drop on next update.

Bug: none
Change-Id: I8b1b3549a2bcd981e941b1cc802c984828d68261
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252444
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36099}
2022-03-01 09:45:30 +00:00
141007668c Add field trial for limiting probes and delay based estimates to link
capacity.

Allow delay based estimate to increase up to 85% of the current NetworkStateEstimate
even if in ALR. The estimate may not increase higher than that.
WebRTC-Bwe-EstimateBoundedIncrease/ratio:0.85,ignore_acked:true

Bug: none
Change-Id: I6f34af7fab03082ca168e624ddea06f216790fbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252442
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36096}
2022-02-28 17:29:37 +00:00
c2b1bad4c8 In RtcpTransceiver use TimeDelta instead of raw int to represent time
Bug: webrtc:8239, webrtc:13757
Change-Id: Idda3fe5761665b4b3fedaf2dd1a28bb0119ae1f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252287
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36094}
2022-02-28 11:21:17 +00:00
9af4aa7cf4 Reland "Represent RtpPacketToSend::capture_time with Timestamp"
This reverts commit 56db8d09529d5ba92d24954a1d78a90c8ea2cd85.

Reason for revert: downstream problem addressed

Original change's description:
> Revert "Represent RtpPacketToSend::capture_time with Timestamp"
>
> This reverts commit 385eb9714daa80306d2f92d36678c42892dab555.
>
> Reason for revert: Causes problems downstream:
>
> #
> # Fatal error in: rtc_base/units/unit_base.h, line 122
> # last system error: 0
> # Check failed: value >= 0 (-234 vs. 0)
>
> Original change's description:
> > Represent RtpPacketToSend::capture_time with Timestamp
> >
> > Bug: webrtc:13757
> > Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#36083}
>
> Bug: webrtc:13757
> Change-Id: I8442abd438be8726cf671d0f372d50ecfac6847e
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252720
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36087}

Bug: webrtc:13757
Change-Id: I1fa852757480116f35deb2b6c3c27800bdf5e197
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252781
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36093}
2022-02-28 10:04:37 +00:00
56db8d0952 Revert "Represent RtpPacketToSend::capture_time with Timestamp"
This reverts commit 385eb9714daa80306d2f92d36678c42892dab555.

Reason for revert: Causes problems downstream:

#
# Fatal error in: rtc_base/units/unit_base.h, line 122
# last system error: 0
# Check failed: value >= 0 (-234 vs. 0)

Original change's description:
> Represent RtpPacketToSend::capture_time with Timestamp
>
> Bug: webrtc:13757
> Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36083}

Bug: webrtc:13757
Change-Id: I8442abd438be8726cf671d0f372d50ecfac6847e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252720
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36087}
2022-02-26 10:35:13 +00:00
385eb9714d Represent RtpPacketToSend::capture_time with Timestamp
Bug: webrtc:13757
Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36083}
2022-02-25 16:44:07 +00:00
0825daf2ed PipeWire capturer: search for epoxy headers
We actually use headers from libepoxy (it's part of the sysroot) so this
was removed accicentally in one of previous changes and it just
magically worked as we include those headers with their full path

Bug: webrtc:13429
Change-Id: I4f5684521a76287a725272ce3833daae673d9332
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252002
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36073}
2022-02-25 01:20:32 +00:00
ecd5ba15cb Fix missing include of rtc_base/system/no_unique_address.h
Bug: None
Change-Id: I047c456cde647282824e8c51122ae53bef7cb7b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252440
Auto-Submit: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36070}
2022-02-24 19:25:12 +00:00
3ed1dbb56e In RtpPacketReceived delete deprecated accessors for handling time in ms
Bug: None
Change-Id: I02a43a16e8d9bf3a1e2c9f6442a1c119620e1288
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252286
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36067}
2022-02-24 15:37:51 +00:00
3147e29c4e Refactor encoder-complexity param in VideoCodec w/backward compatibility
Move complexity parameter to the main VideoCodec class to enable
additional video codecs to use the parameter without creating a new
codec-specific structure.

Bug: webrtc:13694
Change-Id: Icb7cf640b178875d799f39ade8b5084e3222bb1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251921
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@google.com>
Cr-Commit-Position: refs/heads/main@{#36040}
2022-02-21 19:40:44 +00:00
808531653e In RtcpTransceiver implement handling incoming RRTR
Bug: webrtc:8239
Change-Id: I4a469b6a0c2e387e35262798f4686fbf310d00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251902
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36037}
2022-02-21 14:07:34 +00:00
c4ed5f0b1a Adding fuzzer for G711/PCM u/A decoders and fixing a fuzzer problem
Bug: chromium:1279775
Change-Id: I8cc3f5fe25b9e707e9d171251026bd5a8bad5da5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251844
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36036}
2022-02-21 13:35:24 +00:00
ba2677061a Add fuzzer test for G722 and fix a fuzzer problem
The problem was fixed by implementing the methid PacketDuration() in
AudioDecoderG722StereoImpl, which catches the issue in
AudioDecoder::Decode().


Bug: chromium:1280851
Change-Id: I31f974b9999f3c1c62b0e5dc39bb3e56a9a9388d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251842
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36034}
2022-02-21 10:16:47 +00:00
feac97bb25 AgcManagerDirect: improve AgcMinMicLevelExperimentEnabled50 test
Also test the field trial with valid parameter and non-empty suffix.

Bug: webrtc:7494
Change-Id: I3d871b41dd71c951ac56e180b3c09cda4c3627d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251441
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36031}
2022-02-18 17:52:01 +00:00
8968bcae8d In RtcpTransceiver avoid generating rtcp sender reports for inactive senders
Bug: webrtc:8239
Change-Id: I97d50c628db04c56669179ab7039a3fe3bd61d34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251901
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36030}
2022-02-18 16:44:52 +00:00
27d5f14cf2 in RTPSender disallow enabling misconfigured rtx
Bug: None
Change-Id: Id94771626ef723212e4d92d9093af3ec9e647990
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251780
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36020}
2022-02-16 16:08:40 +00:00
454d2309de Add bitrate adaptation tests
Bug: none
Change-Id: I3e2c503efc7a85a3daaa40cd8118c1b02d3b81cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251680
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36013}
2022-02-16 10:25:13 +00:00
1db0a261ca Reland "Reland "Remove unused APM voice activity detection sub-module""
This reverts commit 09aaf6f7bcfb4da644bd86c76896a04a41f776e1.

Reason for revert: downstream fixed (see https://chromium-review.googlesource.com/c/chromium/src/+/3461371)

Original change's description:
> Revert "Reland "Remove unused APM voice activity detection sub-module""
>
> This reverts commit 54d1344d985b00d4d1580dd18057d4618c11ad1f.
>
> Reason for revert: Breaks chromium roll, see 
> https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_tsan_rel_ng/1080583/overview
>
> https://chromium-review.googlesource.com/c/chromium/src/+/3461512
>
> Original change's description:
> > Reland "Remove unused APM voice activity detection sub-module"
> >
> > This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.
> >
> > Reason for revert: dependency in a downstream project removed
> >
> > Original change's description:
> > > Revert "Remove unused APM voice activity detection sub-module"
> > >
> > > This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.
> > >
> > > Reason for revert: breaking downstream projects
> > >
> > > Original change's description:
> > > > Remove unused APM voice activity detection sub-module
> > > >
> > > > API changes:
> > > > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > > > - webrtc::AudioProcessingStats::voice_detected deprecated
> > > > - cricket::AudioOptions::typing_detection deprecated
> > > > - webrtc::StatsReport::StatsValueName::
> > > >   kStatsValueNameTypingNoiseState deprecated
> > > >
> > > > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> > > >
> > > > Bug: webrtc:11226,webrtc:11292
> > > > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/main@{#35975}
> > >
> > > TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
> > >
> > > Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: webrtc:11226,webrtc:11292
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#35977}
> >
> > # Not skipping CQ checks because this is a reland.
> >
> > Bug: webrtc:11226,webrtc:11292
> > Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
> > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35984}
>
> TBR=mbonadei@webrtc.org,gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: Ib308a3af2dcce85a0074ef5a4680ccec3f82712f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11226,webrtc:11292
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251688
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35990}

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11226,webrtc:11292
Change-Id: Idfda6a517027ad323caf44c526a88468e5b52b65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251762
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36012}
2022-02-16 08:41:30 +00:00
5ae9b260ff Implement MouseCursorMonitorPipeWire to track cursor changes separately
Current implementation has mouse cursor as part of the screen itself
which means that everytime a cursor changes location, we have to update
whole screen content, which brings unnecessary load overhead. Using our
own mouse cursor monitor implementation allows us to track only mouse
cursor changes and update them separately for much better performance.

Bug: webrtc:13429
Change-Id: I224e9145f0bc7e45eafe4490de160f2ad4c8b545
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244507
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36011}
2022-02-15 23:03:41 +00:00
0c6e34ce5c Ensure PipeWire doesn't use a Null SourceId
This has mostly seemed to work fine until now; but there's a collision
happening in chromium where if the source is being shown in the Window
Picker it collides with the (also null) Dialog ID and is ignored. While
we could patch that code to not count Null as a collision, there's the
potential for other (future) code to simply ignore a capture source
that it thinks is Null.

Fixed: chromium:1295375
Change-Id: I4356084f0af97f4d56632938b0d9a24d327f7107
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251500
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36008}
2022-02-15 20:03:33 +00:00
0b02d637c0 Calculate max/avg encode/decode latency in codec tests
Bug: none
Change-Id: Ie42461dd06b1764c99308393477921ea25319ab4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251687
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36007}
2022-02-15 18:14:41 +00:00
ac341df436 Adding fuzzer for PCM16b decoder and fixing a fuzzer problem
Bug: chromium:1280852
Change-Id: I7f6c5de86ceee01156743c0389c59f875e53bb5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251580
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36005}
2022-02-15 15:59:01 +00:00
f7a1937e70 Add FrameBufferProxy test for low-latency renderer
Ensures that frames are decoded instantly when in low-latency render
mode. This also tests the max queue size behaviour. Adds a new test
suite for FrameBufferProxy that sets the appropriate field trials.

* Fixes FrameDecodeTiming to never use negative wait times for decode
timestamps.

R=kron@webrtc.org

Change-Id: I06cbec52e1e866e21aa964b24c4fd0163c26961b
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251601
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35999}
2022-02-15 08:30:51 +00:00
405ac4e840 Add objc_class_prefix to the Audio Network Adaptor proto.
WANA: WebRTC Audio Network Adaptor

No-Try: True
Bug: None
Change-Id: I291e02ab70323ecc45d87cea0ea8d7e8cb62db9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249784
Reviewed-by: Minyue Li <minyue@google.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35998}
2022-02-14 21:04:20 +00:00
2f194e0325 PipeWire capturer: Import DMA-BUFs with correct render node
With more GPUs it might happen that server used different render
node from the one we pick from the list. This would cause DMA-BUF to
fail to import so we use Wayland client library to obtain wl_display in
order to initialize EGLDisplay using same render node and have previous
approach as a fallback. Also everyone else uses EGL_LINUX_DMA_BUF_EXT
target for importing EGLImages from DMA-BUF file descriptors so use it
as well to be sure we import buffers same way as they are produced.

Bug: chromium:1290566
Bug: webrtc:13429
Change-Id: I32bbb0bdb28c08b6e7fcb3f94009f82a2041b6ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250661
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35997}
2022-02-14 20:08:50 +00:00
f2b987377b in RtcpTransceiver implement sending rtcp sender reports
Bug: webrtc:8239
Change-Id: Id3298bf4e0eb18a3fc8072fb19416e67a126705f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249788
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35995}
2022-02-14 15:58:40 +00:00
09aaf6f7bc Revert "Reland "Remove unused APM voice activity detection sub-module""
This reverts commit 54d1344d985b00d4d1580dd18057d4618c11ad1f.

Reason for revert: Breaks chromium roll, see 
https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_tsan_rel_ng/1080583/overview

https://chromium-review.googlesource.com/c/chromium/src/+/3461512

Original change's description:
> Reland "Remove unused APM voice activity detection sub-module"
>
> This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.
>
> Reason for revert: dependency in a downstream project removed
>
> Original change's description:
> > Revert "Remove unused APM voice activity detection sub-module"
> >
> > This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.
> >
> > Reason for revert: breaking downstream projects
> >
> > Original change's description:
> > > Remove unused APM voice activity detection sub-module
> > >
> > > API changes:
> > > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > > - webrtc::AudioProcessingStats::voice_detected deprecated
> > > - cricket::AudioOptions::typing_detection deprecated
> > > - webrtc::StatsReport::StatsValueName::
> > >   kStatsValueNameTypingNoiseState deprecated
> > >
> > > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> > >
> > > Bug: webrtc:11226,webrtc:11292
> > > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#35975}
> >
> > TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
> >
> > Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:11226,webrtc:11292
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35977}
>
> # Not skipping CQ checks because this is a reland.
>
> Bug: webrtc:11226,webrtc:11292
> Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35984}

TBR=mbonadei@webrtc.org,gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Ib308a3af2dcce85a0074ef5a4680ccec3f82712f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251688
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35990}
2022-02-14 12:25:51 +00:00
eb6c6fcf27 Fix delta frame delay calculation
The issue was introduced in https://webrtc-review.googlesource.com/c/src/+/132460.

Bug: webrtc:10412
Change-Id: I92d1bd2be63ea34d150145cec63c282f7aa49ce8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251683
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35989}
2022-02-14 11:15:50 +00:00
54d1344d98 Reland "Remove unused APM voice activity detection sub-module"
This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.

Reason for revert: dependency in a downstream project removed

Original change's description:
> Revert "Remove unused APM voice activity detection sub-module"
>
> This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.
>
> Reason for revert: breaking downstream projects
>
> Original change's description:
> > Remove unused APM voice activity detection sub-module
> >
> > API changes:
> > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > - webrtc::AudioProcessingStats::voice_detected deprecated
> > - cricket::AudioOptions::typing_detection deprecated
> > - webrtc::StatsReport::StatsValueName::
> >   kStatsValueNameTypingNoiseState deprecated
> >
> > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> >
> > Bug: webrtc:11226,webrtc:11292
> > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35975}
>
> TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11226,webrtc:11292
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35977}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:11226,webrtc:11292
Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35984}
2022-02-13 14:02:08 +00:00
a751f167c6 Revert "Remove unused APM voice activity detection sub-module"
This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.

Reason for revert: breaking downstream projects

Original change's description:
> Remove unused APM voice activity detection sub-module
>
> API changes:
> - webrtc::AudioProcessing::Config::VoiceDetection removed
> - webrtc::AudioProcessingStats::voice_detected deprecated
> - cricket::AudioOptions::typing_detection deprecated
> - webrtc::StatsReport::StatsValueName::
>   kStatsValueNameTypingNoiseState deprecated
>
> PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
>
> Bug: webrtc:11226,webrtc:11292
> Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35975}

TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35977}
2022-02-11 12:15:44 +00:00
9cc5fffee1 Convert a few more uses of rtc::split to use string_view
Bug: webrtc:13579
Change-Id: I84bdb908bf390924c6d67cd1c5aabcc9e62f33da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251581
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35976}
2022-02-11 11:31:54 +00:00
b4e06d032e Remove unused APM voice activity detection sub-module
API changes:
- webrtc::AudioProcessing::Config::VoiceDetection removed
- webrtc::AudioProcessingStats::voice_detected deprecated
- cricket::AudioOptions::typing_detection deprecated
- webrtc::StatsReport::StatsValueName::
  kStatsValueNameTypingNoiseState deprecated

PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0

Bug: webrtc:11226,webrtc:11292
Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35975}
2022-02-11 10:47:39 +00:00
1a41178e33 scoped_glib: Fix ODR violation
Moving the template specialization into the header causes ODR
violation when the header file is included in other units. Making
the specialization inline to avoid this problem.

Bug: chromium:1291247
Change-Id: I090548c1c3dd07a8c46b87ae90ebdd45a60a5cde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251200
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35969}
2022-02-09 21:52:26 +00:00
9b3c792f67 screencast_portal.h: Remove unused typedef
Minor cleanup to remove unused typedef.

Bug: chromium:1291247
Change-Id: Idbbe8dba13d4d14888f843ae170a898ff604852b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249700
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Salman Malik <salmanmalik@google.com>
Cr-Commit-Position: refs/heads/main@{#35968}
2022-02-09 18:52:55 +00:00
ffdc6804bf Reland: Added support for H264 YUV444 (I444) decoding.
PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/235340

Bug: chromium:1251096
Change-Id: Icd997c7f7732229954d5494343b4e7a70deb09d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251303
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35964}
2022-02-09 11:57:55 +00:00
b02220d1a0 Reland "Mark all bool conversion operators as explicit"
This is a reland of 325789c4576b60147ee1ef225d438cbb740f65ff

Original change's description:
> Mark all bool conversion operators as explicit
>
> An explicit bool conversion operator will still be used implicitly
> when an expression appears in "bool context", e.g., as the condition
> in an if statement, or as argument to logical operators. The
> `explicit` annotation prevents conversion in other contexts, e.g.,
> converting both a and b to bool in an expression like `a == b`.
>
> Bug: None
> Change-Id: I79ef35b1ea831e6011ae472900375ae8a3e617ab
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250664
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35927}

Bug: None
Change-Id: Ie057dfc8c0b5c498e2c8daff7620172c89f0e011
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251380
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35962}
2022-02-09 09:40:05 +00:00
f4cad8ac51 PipeWire capturer: drop DMA-BUF modifier and renegotiate parameters on failure
In case we fail to import a DMA-BUF with given modifier, we can try to
drop the modifier we failed to use and renegotiate stream parameters
in order to use a different modifier or fallback to shared memory buffers.

Bug: chromium:1290566
Change-Id: I617513bdd67a43f62b647a172e0c166af138b3f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249798
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35957}
2022-02-08 20:38:54 +00:00
49e0e77e40 PipeWire capturer: make use of ScreenCaptureFrameQueue
This allows us to keep always some frame around so we can return it
everytime consumer asks us to capture a frame as before we either
returned current frame or nothing as there was no new frame available.
This will be needed in order to support mouse cursor separately as
DesktopAndCursorComposer requires frame everytime, even if it's the
same one as before so we can combine it with the mouse cursor.

Bug: webrtc:13429
Change-Id: Ice87968846870c0a880ab469d9e052b4978e658c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239362
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35956}
2022-02-08 18:02:25 +00:00
8a9aa55561 Remove AudioProcessing::ChannelLayout
This enum is no longer needed. Also moving the last piece of code from
common.h to audio_processing_impl.h, allowing to delete common.h.

Bug: chromium:1271981, b/217349489
Change-Id: If115336c36d6d7b5845a903e421c18aebfe434ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251242
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35946}
2022-02-08 10:07:36 +00:00
1a58a3fe3f Reland "Delete implicit conversion from raw pointer to scoped_ref_ptr"
This is a reland of 7b370b935ec0dac991da08f9da227df9ce245fd5

Original change's description:
> Delete implicit conversion from raw pointer to scoped_ref_ptr
>
> Followup to https://webrtc-review.googlesource.com/c/src/+/242363
>
> Bug: webrtc:13464
> Change-Id: I44358e8cfedeea92aac4ef47c540aff9a4865cdc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247362
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35897}

Bug: webrtc:13464
Change-Id: Ia0da558adb65852a900030ca7c2f2310a275188e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35943}
2022-02-08 08:40:44 +00:00
092d776b7b Make WebRtcAudioRecord save timestamps
Add timestamps to audio_record_jni DataIsRecorded() function, and make
WebRtcAudioRecord find and send the time stamp to that function.

This CL is an continuation of
https://webrtc-review.googlesource.com/c/src/+/249085

Bug: webrtc:13609
Change-Id: I63ab89f1215893cbe1d11d9d8948f5639fc5cdfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249951
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#35933}
2022-02-07 13:30:54 +00:00