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}
185 lines
6.8 KiB
C++
185 lines
6.8 KiB
C++
/*
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* Copyright 2019 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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#ifndef PC_VIDEO_RTP_RECEIVER_H_
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#define PC_VIDEO_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/frame_transformer_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/sequence_checker.h"
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#include "api/transport/rtp/rtp_source.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 "pc/jitter_buffer_delay.h"
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#include "pc/media_stream_track_proxy.h"
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#include "pc/rtp_receiver.h"
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#include "pc/video_rtp_track_source.h"
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#include "pc/video_track.h"
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#include "rtc_base/ref_counted_object.h"
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#include "rtc_base/system/no_unique_address.h"
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#include "rtc_base/thread.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class VideoRtpReceiver : public RtpReceiverInternal {
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public:
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// An SSRC of 0 will create a receiver that will match the first SSRC it
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// sees. Must be called on signaling thread.
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VideoRtpReceiver(rtc::Thread* worker_thread,
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std::string receiver_id,
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std::vector<std::string> streams_ids);
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// TODO(hbos): Remove this when streams() is removed.
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// https://crbug.com/webrtc/9480
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VideoRtpReceiver(
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rtc::Thread* worker_thread,
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const std::string& receiver_id,
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const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
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virtual ~VideoRtpReceiver();
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rtc::scoped_refptr<VideoTrackInterface> video_track() const { return track_; }
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// RtpReceiverInterface implementation
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rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
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return track_;
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}
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rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override;
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std::vector<std::string> stream_ids() const override;
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std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
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const override;
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cricket::MediaType media_type() const override {
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return cricket::MEDIA_TYPE_VIDEO;
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}
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std::string id() const override { return id_; }
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RtpParameters GetParameters() const override;
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void SetFrameDecryptor(
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rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
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rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor()
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const override;
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void SetDepacketizerToDecoderFrameTransformer(
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) override;
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// RtpReceiverInternal implementation.
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void Stop() override;
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void SetupMediaChannel(uint32_t ssrc) override;
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void SetupUnsignaledMediaChannel() override;
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uint32_t ssrc() const override;
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void NotifyFirstPacketReceived() override;
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void set_stream_ids(std::vector<std::string> stream_ids) override;
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void set_transport(
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rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override;
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void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
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streams) override;
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void SetObserver(RtpReceiverObserverInterface* observer) override;
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void SetJitterBufferMinimumDelay(
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absl::optional<double> delay_seconds) override;
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void SetMediaChannel(cricket::MediaChannel* media_channel) override;
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int AttachmentId() const override { return attachment_id_; }
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std::vector<RtpSource> GetSources() const override;
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// Combines SetMediaChannel, SetupMediaChannel and
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// SetupUnsignaledMediaChannel.
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void SetupMediaChannel(absl::optional<uint32_t> ssrc,
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cricket::MediaChannel* media_channel);
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private:
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void RestartMediaChannel(absl::optional<uint32_t> ssrc)
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RTC_RUN_ON(&signaling_thread_checker_);
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void RestartMediaChannel_w(absl::optional<uint32_t> ssrc,
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MediaSourceInterface::SourceState state)
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RTC_RUN_ON(worker_thread_);
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void SetSink(rtc::VideoSinkInterface<VideoFrame>* sink)
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RTC_RUN_ON(worker_thread_);
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void SetMediaChannel_w(cricket::MediaChannel* media_channel)
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RTC_RUN_ON(worker_thread_);
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// VideoRtpTrackSource::Callback
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void OnGenerateKeyFrame();
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void OnEncodedSinkEnabled(bool enable);
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void SetEncodedSinkEnabled(bool enable) RTC_RUN_ON(worker_thread_);
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class SourceCallback : public VideoRtpTrackSource::Callback {
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public:
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explicit SourceCallback(VideoRtpReceiver* receiver) : receiver_(receiver) {}
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~SourceCallback() override = default;
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private:
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void OnGenerateKeyFrame() override { receiver_->OnGenerateKeyFrame(); }
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void OnEncodedSinkEnabled(bool enable) override {
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receiver_->OnEncodedSinkEnabled(enable);
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}
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VideoRtpReceiver* const receiver_;
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} source_callback_{this};
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RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_;
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rtc::Thread* const worker_thread_;
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const std::string id_;
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cricket::VideoMediaChannel* media_channel_ RTC_GUARDED_BY(worker_thread_) =
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nullptr;
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absl::optional<uint32_t> ssrc_ RTC_GUARDED_BY(worker_thread_);
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// `source_` is held here to be able to change the state of the source when
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// the VideoRtpReceiver is stopped.
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const rtc::scoped_refptr<VideoRtpTrackSource> source_;
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const rtc::scoped_refptr<VideoTrackProxyWithInternal<VideoTrack>> track_;
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std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_
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RTC_GUARDED_BY(&signaling_thread_checker_);
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RtpReceiverObserverInterface* observer_
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RTC_GUARDED_BY(&signaling_thread_checker_) = nullptr;
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bool received_first_packet_ RTC_GUARDED_BY(&signaling_thread_checker_) =
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false;
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const int attachment_id_;
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rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_
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RTC_GUARDED_BY(worker_thread_);
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rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_
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RTC_GUARDED_BY(&signaling_thread_checker_);
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_
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RTC_GUARDED_BY(worker_thread_);
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// Stores the minimum jitter buffer delay. Handles caching cases
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// if `SetJitterBufferMinimumDelay` is called before start.
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JitterBufferDelay delay_ RTC_GUARDED_BY(worker_thread_);
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// Records if we should generate a keyframe when `media_channel_` gets set up
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// or switched.
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bool saved_generate_keyframe_ RTC_GUARDED_BY(worker_thread_) = false;
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bool saved_encoded_sink_enabled_ RTC_GUARDED_BY(worker_thread_) = false;
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};
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} // namespace webrtc
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#endif // PC_VIDEO_RTP_RECEIVER_H_
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