This is a reland of commit 81aab48878
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}
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}
Prior to this CL, calling RtpTransceiver::SetChannel() with null
arguments would cause the receiver's track to end. This is wrong,
because the channel can be nulled for other reasons than the transceiver
being stopped/removed - such as when the transceiver is rolled back but
still in use. Also, stopping a transceiver will end the track, so we
should simply ensure to always stop the transceiver when that is needed.
This CL makes sure that the transceiver is stopped or stopping in all
appropriate places, allowing us to remove the ability to end the source
for any other reason. A side-effect of this is that:
- The track never ends prematurely, fixing https://crbug.com/1315611.
- Removed transceivers are always stopped, fixing
https://crbug.com/webrtc/14005.
This CL fixes the issue of track being ended in the ontrack event when
running https://jsfiddle.net/henbos/nxebusjm/.
- We don't have WPT test coverage for this, so I'll add that separately.
With SetSourceEnded() removed, some stopping/stop in response to
rejecting locally SDP munged content had to be added in order not to
regress the existing test coverage for this:
*PeerConnectionInterfaceTest.RejectMediaContent/1
Bug: chromium:1315611, webrtc:14005.
Change-Id: I21f30a1259e51324066dc84f72a72485b9e0fadc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36669}
This makes SetChannel() consistently make 2 invokes instead of a
multiple of senders+receivers (previous minimum was 4 but could be
larger).
* Stop() doesn't hop to the worker thread.
* SetMediaChannel(), an already-required step on the worker thread for
senders and *sometimes* for receivers[1], is now consistently required
for both. This simplifies transceiver teardown and enables the next
bullet.
* Transceiver stops all senders and receivers in one go rather than
ping ponging between threads.
[1] When not required, it was done implicitly inside of Stop().
See changes in `RtpTransceiver::SetChannel`
Bug: webrtc:13540
Change-Id: Ied61636c8ef09d782bf519524fff2a31e15219a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249797
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36057}
For implementations where the signaling and worker threads are not
the same thread, this significantly cuts down on Thread::Invoke()s that
would block the signaling thread while waiting for the worker thread.
For Audio and Video Rtp receivers, the following methods now do not
block the signaling thread:
* GetParameters
* SetJitterBufferMinimumDelay
* GetSources
* SetFrameDecryptor / GetFrameDecryptor
* SetDepacketizerToDecoderFrameTransformer
Importantly this change also makes the track() accessor accessible
directly from the application thread (bypassing the proxy) since
for receiver objects, the track object is const.
Other changes:
* Remove RefCountedObject inheritance, use make_ref_counted instead.
* Every member variable in the rtp receiver classes is now RTC_GUARDED
* Stop() now fully clears up worker thread state, and Stop() is
consistently called before destruction. This means that there's one
thread hop instead of at least 4 before (sometimes more), per receiver.
* OnChanged triggered volume for audio tracks is done asynchronously.
* Deleted most of the JitterBufferDelay implementation. Turns out that
it was largely unnecessary overhead and complexity.
It seems that these two classes are copy/pasted to a large extent
so further refactoring would be good in the future, as to not have to
fix each issue twice.
Bug: chromium:1184611
Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34022}
Changes Rtp Receivers to use a null value of ssrc to mean a default
receive stream.
Bug: webrtc:8694
Change-Id: I835199345f7add993b9078c8b0e7988d5cdd6646
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152425
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29201}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
This is in preparation for letting Chrome extract DTLSTransport
information after SLD/SRD instead of doing it on-demand.
Bug: chromium:907849
Change-Id: Iac6b174c98d3d14136e1fd25bce4a9292f6c8b41
Reviewed-on: https://webrtc-review.googlesource.com/c/116984
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26289}
