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175f06f112
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}
101 lines
3.8 KiB
C++
101 lines
3.8 KiB
C++
/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// This file contains classes that implement RtpReceiverInterface.
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// An RtpReceiver associates a MediaStreamTrackInterface with an underlying
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// transport (provided by cricket::VoiceChannel/cricket::VideoChannel)
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#ifndef PC_RTP_RECEIVER_H_
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#define PC_RTP_RECEIVER_H_
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#include <stdint.h>
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/crypto/frame_decryptor_interface.h"
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#include "api/dtls_transport_interface.h"
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#include "api/media_stream_interface.h"
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#include "api/media_types.h"
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#include "api/rtp_parameters.h"
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#include "api/rtp_receiver_interface.h"
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#include "api/scoped_refptr.h"
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#include "api/video/video_frame.h"
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#include "api/video/video_sink_interface.h"
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#include "api/video/video_source_interface.h"
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#include "media/base/media_channel.h"
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#include "media/base/video_broadcaster.h"
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#include "pc/video_track_source.h"
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#include "rtc_base/thread.h"
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namespace webrtc {
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// Internal class used by PeerConnection.
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class RtpReceiverInternal : public RtpReceiverInterface {
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public:
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// Call on the signaling thread, to let the receiver know that the the
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// embedded source object should enter a stopped/ended state and the track's
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// state set to `kEnded`, a final state that cannot be reversed.
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virtual void Stop() = 0;
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// Sets the underlying MediaEngine channel associated with this RtpSender.
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// A VoiceMediaChannel should be used for audio RtpSenders and
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// a VideoMediaChannel should be used for video RtpSenders.
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// NOTE:
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// * SetMediaChannel(nullptr) must be called before the media channel is
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// destroyed.
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// * This method must be invoked on the worker thread.
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virtual void SetMediaChannel(
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cricket::MediaReceiveChannelInterface* media_channel) = 0;
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// Configures the RtpReceiver with the underlying media channel, with the
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// given SSRC as the stream identifier.
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virtual void SetupMediaChannel(uint32_t ssrc) = 0;
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// Configures the RtpReceiver with the underlying media channel to receive an
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// unsignaled receive stream.
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virtual void SetupUnsignaledMediaChannel() = 0;
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virtual void set_transport(
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rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) = 0;
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// This SSRC is used as an identifier for the receiver between the API layer
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// and the WebRtcVideoEngine, WebRtcVoiceEngine layer.
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virtual absl::optional<uint32_t> ssrc() const = 0;
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// Call this to notify the RtpReceiver when the first packet has been received
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// on the corresponding channel.
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virtual void NotifyFirstPacketReceived() = 0;
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// Set the associated remote media streams for this receiver. The remote track
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// will be removed from any streams that are no longer present and added to
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// any new streams.
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virtual void set_stream_ids(std::vector<std::string> stream_ids) = 0;
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// TODO(https://crbug.com/webrtc/9480): Remove SetStreams() in favor of
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// set_stream_ids() as soon as downstream projects are no longer dependent on
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// stream objects.
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virtual void SetStreams(
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const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) = 0;
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// Returns an ID that changes if the attached track changes, but
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// otherwise remains constant. Used to generate IDs for stats.
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// The special value zero means that no track is attached.
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virtual int AttachmentId() const = 0;
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protected:
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static int GenerateUniqueId();
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static std::vector<rtc::scoped_refptr<MediaStreamInterface>>
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CreateStreamsFromIds(std::vector<std::string> stream_ids);
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};
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} // namespace webrtc
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#endif // PC_RTP_RECEIVER_H_
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