This patch modifies AudioState to always call InitRecording
before StartRecording(). This makes it possible to do
SetRecording(false) + SetRecording(true), which before this
patch would not actually work if there was sending streams.
The only way was to add/remove streams...via SDP operations, puh :(.
Bonus: We also needed to modifu AndroidAudioDeviceModule
(which is a thin wrapper) so that StopRecording() will
call AudioInput->StopRecording() even when recording is not
enabled.
BUG=b/397376626
Change-Id: I954b5caab11225b544c3e6a78c5dde357d4eedb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378140
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Auto-Submit: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43946}
It is a bit more flexible than the current implementation.
Also cleanup ChannelSend::GetRTT since it is not called from the receive stream anymore.
Bug: none
Change-Id: I4403c8b1840012f2287d189be934fd1069de85fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374160
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43728}
It was previously used to get RTT for a receive stream, but it is no longer used.
Also some minor cleanup: fixed includes and removed comments about network thread.
Bug: none
Change-Id: Ia2612ea04be5df82cfe6528c0226095827ea3c77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374042
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43704}
This is a reland of commit 38ddea5ee3320bf3441aeb3654e099b3695c9789
Original change's description:
> Add InsertPacket method that takes RtpPacketInfo.
>
> The version which only passes receive_time will be removed (once migrated).
> Keeping the version that only passes header and payload for convenience.
>
> This will allow us to attach more metadata on the worker thread before InsertPacket, instead of on the playout thread after GetAudio. Eventually, the plan is to split the RTP handling on the worker thread into a separate class.
>
> Bug: webrtc:42223109
> Change-Id: I5399b53b9fc5c2f1c996e109054b1b0877ecca05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369000
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43445}
Bug: webrtc:42223109
Change-Id: I97d1d3d390e6d3de8bf9355b895ec336339d079f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369260
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43454}
This reverts commit 38ddea5ee3320bf3441aeb3654e099b3695c9789.
Reason for revert: not backwards compatible
Original change's description:
> Add InsertPacket method that takes RtpPacketInfo.
>
> The version which only passes receive_time will be removed (once migrated).
> Keeping the version that only passes header and payload for convenience.
>
> This will allow us to attach more metadata on the worker thread before InsertPacket, instead of on the playout thread after GetAudio. Eventually, the plan is to split the RTP handling on the worker thread into a separate class.
>
> Bug: webrtc:42223109
> Change-Id: I5399b53b9fc5c2f1c996e109054b1b0877ecca05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369000
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43445}
Bug: webrtc:42223109
Change-Id: Ie7cf397cfbe5dedca009f16e5e9e3af40adbe99b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369200
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43450}
The version which only passes receive_time will be removed (once migrated).
Keeping the version that only passes header and payload for convenience.
This will allow us to attach more metadata on the worker thread before InsertPacket, instead of on the playout thread after GetAudio. Eventually, the plan is to split the RTP handling on the worker thread into a separate class.
Bug: webrtc:42223109
Change-Id: I5399b53b9fc5c2f1c996e109054b1b0877ecca05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369000
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43445}
The clock rate is already known by the RTP statistician.
Also included some minor code cleanup.
Bug: b/331602608
Change-Id: I335fa2a1cfd7dcceb286706d295a175a92f6797c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368920
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43436}
This seems to have no effect on tests, so it appears that these were
not used after all.
The goal is to make transport-cc a media-section-level attribute.
Bug: webrtc:378698658
Change-Id: Ia20ca5b91472b02db30f911ad1a1892cf36cd682
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43411}
Increased the number of errors the automation is fixing to 150 from
75 in this commit.
Bug: webrtc:370878648
Change-Id: If6e6a5f40db7eb54c27c1a85fb7031838e478c70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366205
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#43337}
danilchap@webrtc.org suggested to add a converter for NtpTime <-> UTC Timestamp for in https://webrtc-review.googlesource.com/c/src/+/365641.
This CL add a NtpTime -> UTC Timestamp in Clock, and change code to start to use the new function.
Bug: None
Change-Id: If4af6cb8e31c1731692edfb8358e67b7a43226a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366001
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43293}
We should use the Timestamp type, rather then int64, to store timestamps. In https://webrtc-review.googlesource.com/c/src/+/365001/ an additional int64 timestamp was added (last_sender_report_timestamp_ms).
This CL fixes the new timestamp, as well as other similar timestamps in MediaReceiverInfo (last_sender_report_utc_timestamp_ms and last_sender_report_remote_utc_timestamp_ms).
Bug: webrtc:372393493
Change-Id: I0e473730e85a69ec595b421e2c3db920364008eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43248}
Add an environment clock timestamp to SenderReportStats and make it visible in rtc_stats_collector.cc. This make it possible to use the pc->GetConfiguration().stats_timestamp_with_environment_clock() flag to decide which timestamp to use when creating a RTCRemoteOutboundRtpStreamStats object.
This CL is the third (and possible the last) of a series of CLs that aim to replace the UTC timestamps in RTCStats objects to Environment clock timestamps. The other CLs where https://webrtc-review.googlesource.com/c/src/+/363946 and https://webrtc-review.googlesource.com/c/src/+/364782.
When Chromium and Google internal uses of RTCStats are updated to set the stats_timestamp_with_environment_clock configuration, the flag can be deleted.
Bug: chromium:369369568
Change-Id: Ic0b07d7b012505267bd6516f19a9ba90df4cafab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365001
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43206}
I missed one timestamp in https://webrtc-review.googlesource.com/c/src/+/363946, meaning that the config flag that was added do not yet work for all timestamps in RTCStats objects. The RTCRemoteOutboundRtpStreamStats still has UTC timestamps even if the config flag is set.
I will solve this by saving both an UTC (existing) and env (to be added) timestamp, and then let rtc_stats_collector choose timestamp based on the value of the config flag (just like RTCRemoteInboundRtpStreamStats is done in the 363946 commit).
Before adding the new env_ timestamp I want to make this change. I rename the existing timestamp to show what epoch it uses (NTP or UTC). This will later make it clear which timestamp is which.
So this CL will make no logical change, just renaming members.
I only need to rename the last_sender_report_timestamp_ms, but opted to rename the remote timestamp as well, to be consistent with the naming convention I add in this CL.
Bug: chromium:369369568
Change-Id: Icfe7cf274995b39799e1478a1bb8cdf5134f0b16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364782
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43194}
This is a reland of commit 4334cdfc5c0619a5f06125ea1f039cb123ccf21e
Original change's description:
> Reland "Return audio stats regarless if we have a codec."
>
> This is a reland of commit 7fff587a096c6ef40f5601f47ef50b221b3a4abf
>
> Original change's description:
> > Return audio stats regarless if we have a codec.
> >
> > Bug: b/331602608
> > Change-Id: I2d12a3ed83645fe1e7cbd8950fd86d5ba2d7c94d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361743
> > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42964}
>
> Bug: b/331602608
> Change-Id: I95c89e7059005bc8dd8569ef41bfe9e863b4082f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361762
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42969}
Bug: b/331602608
Change-Id: I743f0d623230bf871de262792981de35c156ba3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364461
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43142}
The class itself and its unit test remains, for now, but will be removed
later.
Bug: webrtc:14867
Change-Id: I36cec8fca7913663f63c53622ed2760e5e048c2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362580
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43023}
ChannelReceive is now owning and interfacing with NetEq directly.
A new ResamplerHelper is added to acm_resampler.cc/.h, to do the
audio resampling that was previously done inside AcmReceiver.
AcmReceiver still remains, since it is used in other places for now.
Bug: webrtc:14867
Change-Id: If3eb6415e06b9b5e729d393713f3fccb31b0570f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361820
Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42974}
This is a reland of commit 7fff587a096c6ef40f5601f47ef50b221b3a4abf
Original change's description:
> Return audio stats regarless if we have a codec.
>
> Bug: b/331602608
> Change-Id: I2d12a3ed83645fe1e7cbd8950fd86d5ba2d7c94d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361743
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42964}
Bug: b/331602608
Change-Id: I95c89e7059005bc8dd8569ef41bfe9e863b4082f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361762
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42969}
This makes it simpler to use in more contexts.
Bug: b/364184684
Change-Id: I1b08ebd24e51ba1b3f85261eed503a78cd006fd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361480
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42956}
When Reconfiguring there's a call to ResetSenderCongestionControlObjects followed by a later call to
RegisterSenderCongestionControlObjects which happen on the worker thread, while enqueuing packets is
happening on a different thread.
If packets are enqueued in between these calls we get a null dereference of the `rtp_packet_pacer_`
This change fixes it by pausing processing of incoming audio in the interim state
Bug: webrtc:358290775
Change-Id: I77cebfb131545ce2a6fdb26105dd999da3f7c443
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359080
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42815}
Part of a set of CL to allow video to borrow underused audio bitrate.
Bug: webrtc:35055527
Change-Id: Idb504cbbc5794c06b28bdc21b3d860c9da9df175
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358202
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Dan Tan <dwtan@google.com>
Cr-Commit-Position: refs/heads/main@{#42733}
Passing Environment would allow to propage field trials with it further to NetEq and AudioDecoders
Bug: webrtc:356878416
Change-Id: Ic68420df3b157ed341146207a2c45cb49e59a931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358501
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42728}
To make Environment available for creating AudioDecoders to use propagated field trials
Bug: webrtc:356878416
Change-Id: I5dc4a3514d1182db6a7a5aa770b87daba529a5c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357442
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42719}
To make Environment available for creating AudioDecoders to use propagated field trials
Bug: webrtc:356878416
Change-Id: Ic2371f038b75402bbd007c948f43c60cc6cca8a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358400
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42718}
This CL only adds variables necessary for the feature, which will be
implemented in later CLs.
Bug: webrtc:350555527
Change-Id: I71e56666e629f56168d316bf693150c0df0e2ecf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356740
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Dan Tan <dwtan@google.com>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42698}
MakeAudioEncoder planned to be removed, and Create planned to become pure virtual
While at it, cleanup nearby mock usage:
Remove ON_CALL that by default return default constructed result
Remove EXPECT_CALL().Times(AnyNumber()) for a NiceMock
Remove parameters in EXPECT_CALL when all are wildcard
Remove redundant get to deference a smart pointer
Bug: webrtc:343086059
Change-Id: Ica90a4980350cb82bcebd11df6c63a01b828bb9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356884
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42622}
This switches from accepting a sample rate and convert to channel
size over to accepting the channel size.
Instead of InitializeIfNeeded:
* Offer a way to explicitly initialize PushResampler via the ctor
(needed for VoiceActivityDetectorWrapper)
* Implicitly check for the right configuration from within Resample().
(All calls to Resample() were preceded by a call to Initialize)
As part of this, refactor VoiceActivityDetectorWrapper (VADW):
* VADW is now initialized in the constructor and more const.
* Remove VADW::Initialize() and instead reconstruct VADW if needed.
Add constants for max sample rate and num channels to audio_util.h
In many cases the numbers for these values are embedded in the code
which has led to some inconsistency.
Bug: chromium:335805780
Change-Id: Iead0d52eb1b261a8d64e93f51401147c8fba32f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353360
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42587}
Also adding more checks around parameter valuesm since the
`sample_rate_hz` argument is technically not needed, but will be
removed in a follow-up CL.
Bug: chromium:335805780
Change-Id: Ia7e50658f8a686ab71980f9c59cce5f097b0af40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353340
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42581}
