This is in preparation of merging the PLC and CNG decision logic.
Bug: webrtc:13322
Change-Id: Ica782440b0d5c43c92ad5c33631b0cb708b51b0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297861
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39580}
payload type frequency is not communicated inside an RTP packet and
thus do not belong to the RTPHeader
Bug: None
Change-Id: Ic3e48f1b0507a96ddc697503145f7c8785962926
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296763
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39515}
This is a pure noop refactor that removes duplicated state.
It also correctly keeps track of generated samples when transitioning from CNG to expand mode when CNG timeout is used.
Bug: webrtc:12790
Change-Id: Ieca559bd771c42566e5d4f8837235cb25b1420bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293862
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39334}
This is still a behavior that we want, but a more careful rollout is needed.
Bug: webrtc:12790
Change-Id: Ic74c7b4945c0cdeda2b17f52301069424ad91162
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293860
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39333}
The method and config are no longer used. This concludes the work to
break apart AcmReceiver and AudioCodingModule.
Bug: webrtc:14867
Change-Id: I87219749a1ea72a01b95e960d1f32292f7352c9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291801
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39250}
This change makes AudioCodingModule a pure sender and AcmReceiver a pure
receiver.
The Config struct is in practice no longer used by AudioCodingModule,
so a new definition is included in AcmReceiver. The old definition
remains in AudioCodingModule while downstream clients are being
updated.
Bug: webrtc:14867
Change-Id: If0d0b4214c5aa278cf6c85c5b62c6da644de20e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291533
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39244}
This is a prerequisite step to break apart AudioCodingModule and AcmReceiver.
Bug: webrtc:14867
Change-Id: Iba589c7a31b6346ff4acb727793d84077162c8c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291534
Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39235}
After having generated one second of comfort noise and not received any packets, switch to expand mode which will fade out to silence and enter the efficient muted mode.
The behavior is enabled by default but can be disabled through a field trial.
Bug: webrtc:12790
Change-Id: I1e2c1acced3e4a2c1c1595824f1303a0c339aeb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290578
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39043}
Keeping the headers to allow compatibility with current users
that expect the headers to be in that target before they are
also updated.
Bug: webrtc:9838
Change-Id: I8b1e88850958e92c043686587a37791f01860220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39031}
This was done in crbug.com/webrtc/4559 since "CELT-only mode does not have DTX", but that should not be the case anymore (support was added in Opus v1.2.1).
One exception where DTX does not work is with OPUS_APPLICATION_AUDIO (used with stereo) and low complexity settings. This should not be a common config.
Bug: None
Change-Id: I1476083b836bcabeb73df83d5bf06c3878146d28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288420
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38923}
The problem occurs when more than one call is made to the method RunToNextGetAudio. Except for the first call to that method, the clock was not properly updated on the first iteration of the inner loop in RunToNextGetAudio.
Pair: lionelk@webrtc.org
Bug: webrtc:14735
Change-Id: If6fb5c2c700b0f715f626fedf95672a56b04ab12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285942
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38843}
Add a legend when on the python plots generated with neteq_rtpplay.
Bug: None
Change-Id: I4299858bb9e8e59564c824c99272e4fabc610162
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286840
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38838}
A follow up cl will be created to better handle nullopt frame length range in AudioSendStream.
Note that maxptime is still not used for setting the frame length (only ptime is).
Bug: chromium:1109337
Change-Id: Id21fd8c76a6c4a0c85719a955116f8d16001a3d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284501
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38702}
This is a reland of commit b46c4bf27b
Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}
Bug: webrtc:14450
Change-Id: Ia22c4d7724b6022238235fede93e36e570a49376
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283843
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38665}
This reverts commit b46c4bf27b.
Reason for revert: breaks a downstream project
Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}
Bug: webrtc:14450
Change-Id: Ice138004e84e8c5f896684e8d01133d4b2a77bb7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38655}
Note: this CL has to leave behind one part of iSAC, which is its VAD
currently used by AGC1 in APM. The target visibility has been
restricted and the VAD will be removed together with AGC1 when the
time comes.
Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
Bug: webrtc:14450
Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38652}
This is a reland of commit 38fcd58429
Original change's description:
> Change default NetEq sample rate to 48k.
>
> This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k).
>
> Bug: none
> Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38536}
Bug: none
Change-Id: Id634799286f6d1f1eaf315ebe8e70de669d589db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281900
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38572}
This reverts commit 38fcd58429.
Reason for revert: Breaks downstream test
Original change's description:
> Change default NetEq sample rate to 48k.
>
> This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k).
>
> Bug: none
> Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38536}
Bug: none
Change-Id: I03181168ab14d2d99320767c3a25ba3cfb726b2c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281441
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Auto-Submit: Jakob Ivarsson <jakobi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38538}
This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k).
Bug: none
Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38536}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
New ctor added without optional and media specific fields.
Bug: webrtc:10739, b/246753278
Change-Id: I7e15849aced6ed0a7ada725ea171a15ea1e9bc5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275941
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38124}
Opus was recently updated in Chromium (https://crbug.com/1347531), resulting in these failing for a non-SSE build.
Bug: None
Change-Id: I6c4124192f98f9358e7cc2241aec16a5338e689a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274760
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Felicia Lim <flim@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38095}
This CL also removes the existing non-standard implementation of the metric.
Bug: webrtc:14147, webrtc:11789
Change-Id: I70fd1c451dfd59380fe5ce959086f37b31697c16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265360
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37441}
