212 Commits

Author SHA1 Message Date
Evan Shrubsole
418a8c2c83 Move string_format.h to webrtc namespace
Bug: webrtc:42232595
Change-Id: I208257358150eeb97304946929649414af5eb2ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377542
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43915}
2025-02-18 06:34:45 -08:00
Philipp Hancke
740d726739 Move DTLS related code from p2p/base to p2p/dtls
BUG=webrtc:367395350

Change-Id: I3fd1551f974705ce6b10e2c757f4d406a520a2c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370460
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43528}
2024-12-10 15:55:26 +00:00
Tomas Lundqvist
b40c559858 Set voice RTCP mode based on the RemoteContent and not based on the LocalContent.
The RTCP mode is a send property for both send and receive channels. Send properties should be configured based on what peers support/prefer, which is described by the remote description (content).


Bug: webrtc:340041654
Change-Id: I18cd59e98aecfbbd8f4919b98381836184c10d77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368980
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#43449}
2024-11-25 14:06:39 +00:00
Jakob Ivarsson
68f4e27794 Add RtpSender OnFirstPacketSent callback.
It works in the same way as the first packet received callback and can be used for latency measurements.

One important detail is that RTCP and probe packets are excluded from triggering the callback.

Bug: b/375148360
Change-Id: I5f99b565f96b622e864669cf227be5534aab0fc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366644
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43309}
2024-10-25 16:17:04 +00:00
Harald Alvestrand
dc56a36ff8 Use PayloadTypePicker in WebRtcVoiceEngine
This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code.

Bug: webrtc:360058654
Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-09 18:44:21 +00:00
Harald Alvestrand
927244db7e Set MID in AudioReceiveChannel
This variable was present but unset.

Bug: webrtc:360058654
Change-Id: I492069a1e87208c6fbb5ad5f0a00fcc2ccc0bc25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361824
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42988}
2024-09-09 17:58:33 +00:00
Florent Castelli
8037fc6ffa Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
2024-09-02 12:16:47 +00:00
Harald Alvestrand
d78e30e00b Deprecate cricket::VideoCodec and cricket::AudioCodec
These are aliases for cricket::Codec.
Also remove internal usage

Bug: b/42225532
Change-Id: I220b95260dc942368cb6280432a058159eec8700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349321
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42194}
2024-04-29 16:24:51 +00:00
Tony Herre
a5c8ee1672 Revert "Make Codec::Matches also consider packetization"
This reverts commit 1ae700a9233ed647e1b4080c0fcb48f61a0cca0a.

Reason for revert: Potential root cause of crbug.com/1504351

Original change's description:
> Make Codec::Matches also consider packetization
>
> If it's not considered it can lead to payload IDs erroneously being
> reused if the SDP is munged, see https://crbug.com/webrtc/15473#c10.
>
> Bug: webrtc:15473
> Change-Id: I195a06d556e8a57dbeeb946effc4e0f27cc930b0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326522
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41153}

Bug: webrtc:15473 chromium:1504351
Change-Id: I87fb671d76c3b17beb65124603cc040bb9bf4fa5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329201
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41285}
2023-11-30 14:06:01 +00:00
Philipp Hancke
d0f0f38f72 Remove most usage of MediaContentDescription::as_audio()/as_video()
and unify algorithms a bit more.

BUG=webrtc:15214

Change-Id: Ie9903f3e56d25b1dc026367e8ae6817275faa07b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328442
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41244}
2023-11-27 09:35:39 +00:00
Emil Lundmark
1ae700a923 Make Codec::Matches also consider packetization
If it's not considered it can lead to payload IDs erroneously being
reused if the SDP is munged, see https://crbug.com/webrtc/15473#c10.

Bug: webrtc:15473
Change-Id: I195a06d556e8a57dbeeb946effc4e0f27cc930b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326522
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41153}
2023-11-14 08:14:14 +00:00
Tommi
c3b7a50720 Use webrtc::TaskQueueBase type instead of rtc::Thread
...for signaling and worker thread members in BaseChannel classes.

Bug: webrtc:15099
Change-Id: I83611ed2564e143aca19d0f12ce060b77eb9d2a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325260
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41041}
2023-10-30 14:43:46 +00:00
Harald Alvestrand
ecc38d8d29 Take out callback that modifies voice receive codec based on send codec
No functionality that depends on this information has been identified; no tests break when it is taken out.

Bug: webrtc:15224
Change-Id: I37298479c6b8a4acb82f59d32130c105371936b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324760
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41037}
2023-10-30 12:36:29 +00:00
Philipp Hancke
b527699a53 Reduce usage of audio/video codec specifics
BUG=webrtc:15214

Change-Id: I8e68ac149af53529321ab44776c62afe4cc2f61e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324020
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40960}
2023-10-18 11:34:45 +00:00
Philipp Hancke
19fe2437b7 Remove more codec-related templating
BUG=webrtc:15214

Change-Id: Ia597f674e5650dad31796c9a13769fbe873554fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322122
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40920}
2023-10-12 15:36:42 +00:00
Emil Lundmark
ec8262788b Look through all candidates before falling back to default packetization
It's possible that a peer can signal the same payload with multiple
packetization options. As such, we shouldn't try to fall back to default
packetization until we have considered all the alternatives.

Bug: webrtc:15473
Change-Id: I21772b4d8c53819d1c3105988551ebdbea0df045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320241
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40775}
2023-09-20 12:18:02 +00:00
Emil Lundmark
17304c3bf8 Perform packetization verification until a match is found
If there happens to be an asymmetry between local and remote codecs we
shouldn't validate that there's a 1:1 packetization mapping for every
codec. It's sufficient to check that there's at least one matching
packetization per codec.

Bug: webrtc:15473
Change-Id: Ib4fc8fdd54bb4dccf96f0c802746c848e2deed83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320440
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40760}
2023-09-18 07:43:03 +00:00
Harald Alvestrand
ff281aa328 Convert signals in rtp_transport_internal.h to CallbackList
Bug: webrtc:11943
Change-Id: I8e0839363712d9d8b49c2f6cbdb5f3ac59d79219
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318882
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40700}
2023-09-05 11:37:32 +00:00
Philipp Hancke
5866e1a0ed Rename Set(Send|Recv)Parameters Set(Sender|Receiver)Parameters
following the previous change to rename the classes derived from
  cricket::RtpParameters

Also rename ChangedRecvParameters to ChangedReceiveParameters.

BUG=webrtc:13931

Change-Id: Ia51dd39905a5cbb98162c3948930e43ccaf3786d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314500
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40677}
2023-09-01 08:12:55 +00:00
Philipp Hancke
a9d5141367 Rename cricket::RtpParameters and derived classes
Renames
  cricket::RtpParameters
to
  cricket::MediaChannelParameters
in order to distinguish it better from webrtc::RtpParameters.
This involves renaming
  RtpSendParameters -> SenderParameters
  AudioSendParameters -> AudioSenderParameters
  AudioRecvParameters -> AudioReceiverParameters
  VideoSendParameters -> VideoSenderParameters
  VideoRecvParameters -> VideoReceiverParameters

BUG=webrtc:13931

Change-Id: I664595ee3863418c0c6ca092ca77127be0f9498f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314360
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40497}
2023-08-01 08:55:02 +00:00
Philipp Hancke
4b87d7ac2a Remove Codec template from RtpParameters and helper functions
BUG=webrtc:15214

Change-Id: I3874c4a5089216dab3d072df7854040d5d05bcc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313500
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40492}
2023-07-31 10:49:51 +00:00
Florent Castelli
d797cb6ca7 Remove all split channels related code
Bug: webrtc:13931
Change-Id: I93b8ca0ba1ec15bf260236bbc914b41fbb30aa58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310680
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40376}
2023-06-29 09:32:04 +00:00
Harald Alvestrand
847208e9d6 Remove transitional shim classes
Bug: webrtc:13931
Change-Id: Iaeb0b892aca4b4d64d13a025adc7564e572e0f26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307940
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40232}
2023-06-06 11:58:29 +00:00
Harald Alvestrand
4ad141e69b Add callback for send codec in audio too
It turns out there's a similar linkage as the one for video.
Tests are coming in https://webrtc-review.googlesource.com/c/src/+/307461

Bug: webrtc:13931
Change-Id: I638d1a1907116a71481aa88dce932492323ae5b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307463
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40206}
2023-06-02 11:31:00 +00:00
Harald Alvestrand
5f32fa47a7 Delete MediaBaseChannel class
There are no common functions between MediaSendChannelInterface
and MediaReceiveChannelInterface except media_type().
This allows us to remove the common superclass for the two interfaces,
making for a simpler class structure.

Bug: webrtc:13931
Change-Id: I82a12ca31f0dc62d7bd97bdda34ca37e59a5fd55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306660
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40154}
2023-05-26 10:43:06 +00:00
Harald Alvestrand
13897e67c8 Change SSRC-passing for MediaChannel from external to callback
This makes the handling somewhat more uniform, and is the same
for both video and audio channels.

Bug: webrtc:13931
Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 14:33:59 +00:00
Philipp Hancke
79249155c3 Stop decoding video for m-lines which are sendonly or inactive
by not starting the receive stream whenever it is creating.
Instead, this is controlled by the direction of the media content.

BUG=webrtc:11013

Change-Id: Iaaa0ac0aa9f90a4be776a1348f53a0f9c2b84d99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40064}
2023-05-15 10:54:16 +00:00
Harald Alvestrand
2f55370634 Reland "Use two MediaChannels for 2 directions."
This reverts commit 18c869bc36b342cd4a79947067e52a93a04a7808.

Reason for revert: Added a field trial that allows landing the code without affecting performance in prod.

This CL also incorporates subsequent CLs that also had to be reverted.

Original change's description:
> Revert "Use two MediaChannels for 2 directions."
>
> This reverts commit 8981a6fac3d665beac4a58b9453e6c39988a024f.
>
> Reason for revert: Quality regression detected.
>
> Original change's description:
> > Use two MediaChannels for 2 directions.
> >
> > This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other.
> >
> > The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it.
> >
> > Bug: webrtc:13931
> > Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39340}
>
> No-Try: true
> Bug: webrtc:13931
> Change-Id: I791997ad9eff75c3ac9cd2e4bbacf5bc6c3a3a79
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295663
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39445}

Bug: webrtc:13931
Change-Id: I1318910a685188e2b846c9040e1efc04c2c894ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296080
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39494}
2023-03-07 12:57:35 +00:00
Harald Alvestrand
18c869bc36 Revert "Use two MediaChannels for 2 directions."
This reverts commit 8981a6fac3d665beac4a58b9453e6c39988a024f.

Reason for revert: Quality regression detected.

Original change's description:
> Use two MediaChannels for 2 directions.
>
> This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other.
>
> The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it.
>
> Bug: webrtc:13931
> Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39340}

No-Try: true
Bug: webrtc:13931
Change-Id: I791997ad9eff75c3ac9cd2e4bbacf5bc6c3a3a79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295663
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39445}
2023-03-01 15:57:55 +00:00
Harald Alvestrand
ba088b1dce Revert "Add plumbing for video NACK to be coupled between channels."
This reverts commit a087f6f1c842f1d70ad207b44c48321ab60d2d95.

Reason for revert: Needed to roll back other CL

Original change's description:
> Add plumbing for video NACK to be coupled between channels.
>
> Bug: webrtc:13931, webrtc:14920
> Change-Id: I451869e295e099a1d08c0c80e481decd53149f1b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294382
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39373}

Bug: webrtc:13931, webrtc:14920
Change-Id: I19e176e75630313da470542e7ff1e89b6d717fc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295664
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39432}
2023-03-01 10:49:35 +00:00
Harald Alvestrand
a087f6f1c8 Add plumbing for video NACK to be coupled between channels.
Bug: webrtc:13931, webrtc:14920
Change-Id: I451869e295e099a1d08c0c80e481decd53149f1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294382
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39373}
2023-02-22 14:54:38 +00:00
Harald Alvestrand
8981a6fac3 Use two MediaChannels for 2 directions.
This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other.

The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it.

Bug: webrtc:13931
Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39340}
2023-02-19 10:34:42 +00:00
Per K
89ca299161 Use parsed packet from RtpTransport::DemuxPacket in engine and call
With this cl, a packet is only parsed once in RtpTransport::DemuxPacket and the metadata is reused.
Extensions are still identified twice- one for demuxing based on mid. The second time in Channel::OnReceivedPacket in order to use extensions specific to that mid.

Bug: webrtc:7135, webrtc:14795
Change-Id: I50e3814af92ca4378f148876b20a54bcfac1e146
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290540
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39058}
2023-01-10 15:06:50 +00:00
Harald Alvestrand
50454ef84a Apply PIMPL pattern to MediaSender and Receiver objects
This detaches the implementation (which is still merged)
from the objects used to interface to it.

Bug: webrtc:13931
Change-Id: I872ee10e4ed9fa432bfa231f723af1d3989d79d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288080
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38906}
2022-12-16 09:23:14 +00:00
Harald Alvestrand
c0d44d9d63 Split audio and video channels into Send and Receive APIs.
The implementation here has a number of changes that force the callers
that called the "channel" functions into specific interfaces rather than
just letting C++ take care of it; this should go away once there stops
being a common implementation class for those interfaces.

Bug: webrtc:13931
Change-Id: Ic4e279528a341bc0a0e88d2e1e76c90bc43a1035
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287640
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38888}
2022-12-14 11:00:17 +00:00
Harald Alvestrand
36fafc8827 Split MediaChannel class to sender and receiver
This allows callers to differentiate on whether they need the
channel for sending or receiving purposes.

Note: This CL is incomplete, in that many places cast the pointers
to the concrete subclasses "VideoMediaChannel" and "AudioMediaChannel", which are not split into sending and receiving APIs.

The long term goal is to make two MediaChannel-like class APIs, with distinct implementations, and let the RtpSender and RtpReceiver manage those objects, rather than keeping them in the RtpTransceiver.

Bug: webrtc:13931
Change-Id: I8d56defe2287bd6552b71571cc6a5ec842927fa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287040
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38844}
2022-12-08 10:51:52 +00:00
Danil Chapovalov
9e09a1f327 Replace Thread::Invoke with Thread::BlockingCall
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed

Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
2022-09-09 10:44:17 +00:00
Danil Chapovalov
a30439bbe6 Migrate pc/ to absl::AnyInvocable based TaskQueueBase interface
Bug: webrtc:14245
Change-Id: I9043aa507421a93f0d7ba7406e237f727999b696
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268121
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37478}
2022-07-07 10:33:28 +00:00
Artem Titov
c374d11fac Move to_queued_task.h and pending_task_safety_flag.h into public API
Bug: b/235812579
Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-17 09:20:39 +00:00
Harald Alvestrand
8f42992787 Move channel creation functions into RtpTransceiver
This breaks the link from sdp_offer_answer.cc to channel.h.

Bug: webrtc:13931
Change-Id: I75608f75713bf4e69013ac5f5b17c19e53d07519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261060
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36757}
2022-05-04 11:57:50 +00:00
Harald Alvestrand
9a743179cf Eliminate need for stats to know of channel.h
Also eliminate FillBitrateInfo from the Channel object.

Bug: webrtc:13931
Change-Id: I5265b7629413a1ed04898272adf26708e2ee9b8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260469
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36744}
2022-05-03 12:06:56 +00:00
Harald Alvestrand
c24a2189d7 Update IWYU tool with a mapping file
Also apply IWYU to all .cc files in pc/, and correct BUILD file to match.
Note: Some files came out wrong when iwyu was applied. These are not included.

Bug: none
Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-24 11:05:06 +00:00
Tommi
075db39756 Check for valid transport before setting criteria.
Bug: chromium:1295469
Change-Id: Ib797cf1baf275e3ae1ce167cd4bfbc1161652000
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251620
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35981}
2022-02-11 18:18:27 +00:00
Tommi
2195d54a5c Report an error if applying the demuxer criteria fails.
This is a follow-up to CL [1] to restore the check of whether updating
the demuxer criteria in the channel classes succeeded or failed.

[1] https://webrtc-review.googlesource.com/c/src/+/244700

Bug: webrtc:13536
Change-Id: If0e986bd4e4799e20be2b49f80abcc5a3e6825fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249940
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@google.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35862}
2022-02-01 06:43:58 +00:00
Tomas Gunnarsson
5411b174c8 Add a channel factory interface.
The interface is implemented by the ChannelManager and contains methods
to create and destroy media channel objects as used by a transceiver.

This will subsequently allow us to delete the channel objects from
the transceiver class where ownership really lies rather than from
the outside - which is currently required by some tests that keep
channel objects on the stack. We'll furthermore be able to do the
destruction asynchronously without additional Invoke()s as we do now
which will remove an Invoke when making sdp changes.

With introducing the interface, the following simplifications were made:
* ChannelManager constructed on the signaling thread.
  Before, there was an Invoke in the context class, which existed
  for the purposes of calling MediaEngine::Init() (which in turn is
  only needed for the VoiceEngine). This Invoke has now been moved
  into the CM (more tbd).
* The CM now has a pointer to the signaling thread (since that's the
  construction thread). That allows us to remove the signaling thread
  parameter from the CreateFooChannel methods.
* The ssrc_generator (UniqueRandomIdGenerator) instance for SSRCs moved
  from SdpOfferAnswerHandler to the CM, as it's always used in
  combination with the CM. This simplifies the CreateFooChannel methods
  as well as a couple of other classes that have a CM dependency.
* Removed DestroyFooChannel related code from SdpOfferAnswerHandler since
  the channel type detail can be taken care of by the CM.

Bug: webrtc:11992, webrtc:13540
Change-Id: I04938a803734de8489ba31e6212d9eaecc244126
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247904
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35766}
2022-01-24 08:50:30 +00:00
Tomas Gunnarsson
4f8a58c3d2 Remove 2 Invokes to the network thread when creating a channel.
...and one when destroying a channel object.

This CL removes Init_n() and Deinit_n() from the BaseChannel class.
Channel classes now use SetRtpTransport to do initialization and
uninitialization on the network thread.

Notably if an implementation has called SetRtpTransport() with a valid
transport pointer, it is required that SetRtpTransport be called again
with a nullptr before the channel object can be deleted.

In situations where multiple channels are created, this can mean
a substantial reduction in thread hops. We still hop to the worker
in order to construct the objects - this can probably be avoided
and SetChannel() is still a synchronous operation for the transceivers.
Furthermore, teardown of channel objects also still happens
synchronously and across network/worker/signaling threads.

Bug: webrtc:11992
Change-Id: I68ca7596e181fc82996e3e290733d97381aa5e78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246740
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35738}
2022-01-19 12:17:47 +00:00
Tomas Gunnarsson
1933d3b677 Move network thread invokes for initialization for media channels, out.
Remove Init_w and Deinit(), both of which were wrappers around Invoke()
calls from the worker thread to the network thread.

Instead, replace them with Init_n() and Deinit_n() that are currently*
required to be called by external code in order to associate/disassociate
the channels with the transport.

This CL mostly moves things around in order to prepare for upcoming
changes, but it does change channel destruction in the following way:
- When destroying channels, we don't block the worker thread anymore
  while uninitialization happens on the network thread. Previously
  both signal and worker threads were blocked during the
  uninitialization in the ChannelManager.

* In an upcoming CL, Init_n() and Deinit_n() will be called internally
  from a different method that's always called on the network thread
  when a channel is associated/disassociated with a transceiver. When
  we're there, we will have removed several invokes that currently are
  a part of constructing/destructing channel objects.

Bug: webrtc:11992
Change-Id: Ibc30447a40749ceb36d37834b0cfc5c5ea60e895
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246502
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35707}
2022-01-17 14:06:42 +00:00
Tomas Gunnarsson
cc9b7ec740 Add utility method for consistency checking in BaseChannel classes.
This is a simple check for upcoming changes for media channels to
be able to check if the state on the network thread is consistent.

Bug: webrtc:11992
Change-Id: I8ed2d091ecf3869a66970fc4733aebf209c4ef82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246681
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35706}
2022-01-17 10:19:03 +00:00
Tomas Gunnarsson
a49942af17 Update comments in BaseChannel::SendPacket.
Minor refactoring to reduce logging code.

Bug: none
Change-Id: I0e5883b6cdb356f13939ee64fe002a9ed223ca61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247100
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35705}
2022-01-17 09:45:14 +00:00
Tomas Gunnarsson
e68d874cf1 Update SetPayloadTypeDemuxingEnabled_w to avoid an unnecessary Invoke.
If a call to BaseChannel::SetPayloadTypeDemuxingEnabled does not result
in a demuxer criteria configuration change, we can avoid an Invoke
and speed up negotiations.

Bug: webrtc:11993
Change-Id: I80c894d7cc2a733ab84c1f4978f0c58c09a123a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35649}
2022-01-10 14:16:04 +00:00