31 Commits

Author SHA1 Message Date
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
b54bf8a9af Remove pointless Set*Encryptor functions
These functions had dummy implementations, but were not virtual.
The need for those functions seems to be lost in time.

Bug: None
Change-Id: I66dcac4a92f9993d82031f943f2f9ae767156b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330422
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41336}
2023-12-07 13:55:52 +00:00
Harald Alvestrand
d43af9172b Remove internal overrides using old SendRtp and SendRtcp interfaces.
This CL takes away all usages except for Android code.

Low-Coverage-Reason: Refactoring old code
Bug: webrtc:15410
Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40554}
2023-08-15 13:20:21 +00:00
Harald Alvestrand
b38d9d2b6f Add ArrayView versions of SendRtp and SendRtcp
This is part of the long term plan to stop using pointer + length
to pass around buffers.

Bug: webrtc:14870
Change-Id: Ibaf5258fd326b56132b9b5a8a6b1563a763ef2f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314960
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40512}
2023-08-04 11:20:53 +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
84fdf990e8 Convert Media*Channel to contain a webrtc::Transport
Media*Channel objects used to subclass webrtc::Transport.
This was not an optimal design. This CL makes the transport
a member variable of MediaChannelUtil.

Bug: None
Change-Id: I85d33cc1b32b931e563b7bb2d277f1c512600831
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309800
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40328}
2023-06-21 16:13:55 +00:00
Florent Castelli
4e434c313e Remove MediaChannel usage from webrtc_video_engine_unittest
Bug: webrtc:13931
Change-Id: Ie45a25c6b204b38b749381ef5e9403cf036b8126
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40323}
2023-06-21 10:10:56 +00:00
Harald Alvestrand
09e0086d26 Remove ImplForTesting function from MediaChannel
It is not used any more.

Bug: webrtc:13931
Change-Id: I266de41abe239907c6d65f4b182a8dc3aacaba3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308022
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40234}
2023-06-06 16:30:16 +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
Florent Castelli
8c4b9ea535 Remove references to AudioCodec and VideoCodec constructors
The preferred method to create codecs is to use the function
cricket::CreateAudioCodec or cricketCreateVideoCodec.
Empty codec objects are deprecated and should be replaced
with alternatives such as methods returning an
absl::optional object instead.

Bug: webrtc:15214
Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40226}
2023-06-05 23:23:40 +00:00
Harald Alvestrand
9a34d80fc4 Apply the "shim" pattern for WebRtcVoiceEngine
This ensures that the MediaChannel interface is only implemented
through a send/receive shim, splitting channels also when kBoth is
used.

Bug: webrtc:13931
Change-Id: Ie97809597eaae7b4f504939339795432c34e56cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307461
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40210}
2023-06-02 13:56:43 +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
c18f083900 Split MediaChannel concrete functions to MediaChannelUtil
This allows subclasses of MediaSendChannel and MediaReceiveChannel
to derive from MediaChannelUtil without promising to implement
the interfaces.

Bug: webrtc:13931
Change-Id: I998de7566b343032c83cd6e5419f49349f41035f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307140
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40185}
2023-05-31 08:36:25 +00:00
Harald Alvestrand
97c9623839 Make a shim object implementing the VideoMediaChannel interface
The intent is that this object can be used instead of VideoMediaChannel,
clearing the way for decomposing VideoMediaChannel into send and
receive classes.

This CL uses it for the "both" role of WebRtcVideoEngine::CreateMediaChannel; a later CL will use it for all roles on all engines.

Bug: webrtc:13931
Change-Id: Ibd0ca2c3c45b5e3bfcced8f7e30a1edd63cf7654
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306720
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40173}
2023-05-30 08:44:27 +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
cfd4cd0703 Introduce AddDefaultRecvStreamForTesting to VideoReceiveChannel API
This change allows us to remove one static_cast from tests that
was problematic for another refactoring.

Bug: webrtc:13931
Change-Id: I8e1b5cecadd806b266b6c115b56b18b9613cbe82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306500
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40144}
2023-05-25 12:59:53 +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
63551c6f0c Initialize RTP modes from callback
Before the channel split, the RTP modes were set by reading the
configuration of the send codec. After the split, this is done
via the SetReceiverFeedbackParams function.

This CL adds caching those parameters so that they are applied
to receive streams created after the SetReceiverFeedbackParams call.

Bug: webrtc:13931
Change-Id: I92eb651e5dd1ec68aca7f6a162e3521eb835a11d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305021
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40056}
2023-05-12 12:30:15 +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
Harald Alvestrand
16579cc81d Change MediaChannel to have a Role parameter
This allows MediaChannel to know whether it's being used
for sending, receiving, or both. This is a preparatory CL
for landing the split of MediaChannel usage into sending and
receiving objects.

Bug: webrtc:13931
Change-Id: If518c8b53d5256771200a42e1b5f2b3321d26d8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292860
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39283}
2023-02-09 14:29:08 +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
Henrik Boström
175f06f112 Reland "Remove 'trackId' dependency in stats selector algorithm."
This is a reland of commit 81aab488781c1a736c9d85ff1532631be2989523

See diff between Patch Set 1 and latest Patch Set.

The original CL broke this WPT[1] because getStats() with the receiver
as the selector stopped working in the event of unsignalled SSRCs due
to the receiver not knowing what the SSRC was.

This fix is to query media_channel_ for the unsignalled SSRC in the
event that the receiver does not know the SSRC.

[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/simulcast/setParameters-active.https.html

Original change's description:
> Remove 'trackId' dependency in stats selector algorithm.
>
> In preparation for the deletion of deprecated 'track' stats, the
> stats selector algorithm needs to be rewritten not to use 'trackId'.
>
> This is achieved by finding RTP stats by their SSRC, as obtained via
> getParameters(). This unfortunately adds a block-invoke (in the sender
> case the block-invoke happens inside GetParametersInternal and in the
> receiver case the block-invoke is explicit at the calling place), but
> it can't be helped and it's just once per getStats() call and only if
> the selector argument is used.
>
> Bug: webrtc:14175
> Change-Id: If0e14cdbdc76d141e0042e43757970893bf32119
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289101
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38981}

Bug: webrtc:14175, webrtc:14811
Change-Id: I0d16724af4efeb93d50e36dbfcc798564daff5c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290600
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39010}
2023-01-05 09:04:12 +00:00
Harald Alvestrand
1251c6418e Split stats generation for MediaChannel into sender and receiver APIs
This is in preparation for splitting MediaChannel into sender and
receiver channels, with independent objects.

Bug: webrtc:13931
Change-Id: I8e34b0c80b4d76132394efcda658a8face3ab873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288750
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38998}
2023-01-04 14:17:20 +00:00
Harald Alvestrand
d3ac3b668a Remove AsSendChannel/AsReceiveChannel methods
This CL removes a couple more opportunities for client code
to interact directly with the MediaChannel implementation classes.

No-try because of infra failure.

No-Try: true
Bug: webrtc:13931
Change-Id: I658b8b04eff11de7831a1933d16d40fc59c3f0fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288380
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38935}
2022-12-21 07:47:32 +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