4ccdf932e1
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}
100 lines
3.7 KiB
C++
100 lines
3.7 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/ref_counted_object.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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// Stops receiving. The track may be reactivated.
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virtual void Stop() = 0;
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// Stops the receiver permanently.
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// Causes the associated track to enter kEnded state. Cannot be reversed.
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virtual void StopAndEndTrack() = 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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// Must call SetMediaChannel(nullptr) before the media channel is destroyed.
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virtual void SetMediaChannel(cricket::MediaChannel* 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 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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