4cd2790f17
This CL removes (almost) the last RTP references in VideoReceiveStream. There are still references to RTPFragmentationHeader and SSRCs, which will be dealt with later. There are also new GUARDED_BY and thred checker added to the synchronization class. When there are othre transports than RTP, there will instead be an interface + inheritance for RtpStreamReceiver and RtpStreamSynchronizattion in VideoReceiveStream. This work will be done when we actually know how we want to make thee transport interface. BUG=webrtc:5838 Review-Url: https://codereview.webrtc.org/2216533002 Cr-Commit-Position: refs/heads/master@{#13655}
74 lines
2.4 KiB
C++
74 lines
2.4 KiB
C++
/*
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* Copyright (c) 2012 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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// RtpStreamsSynchronizer is responsible for synchronization audio and video for
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// a given voice engine channel and video receive stream.
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#ifndef WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
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#define WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
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#include <memory>
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/modules/include/module.h"
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#include "webrtc/video/rtp_stream_receiver.h"
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#include "webrtc/video/stream_synchronization.h"
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namespace webrtc {
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class Clock;
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class VideoFrame;
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class VoEVideoSync;
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namespace vcm {
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class VideoReceiver;
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} // namespace vcm
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class RtpStreamsSynchronizer : public Module {
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public:
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RtpStreamsSynchronizer(vcm::VideoReceiver* vcm,
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RtpStreamReceiver* rtp_stream_receiver);
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void ConfigureSync(int voe_channel_id,
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VoEVideoSync* voe_sync_interface);
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// Implements Module.
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int64_t TimeUntilNextProcess() override;
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void Process() override;
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// Gets the sync offset between the current played out audio frame and the
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// video |frame|. Returns true on success, false otherwise.
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bool GetStreamSyncOffsetInMs(const VideoFrame& frame,
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int64_t* stream_offset_ms) const;
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private:
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Clock* const clock_;
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vcm::VideoReceiver* const video_receiver_;
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RtpReceiver* const video_rtp_receiver_;
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RtpRtcp* const video_rtp_rtcp_;
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rtc::CriticalSection crit_;
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int voe_channel_id_ GUARDED_BY(crit_);
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VoEVideoSync* voe_sync_interface_ GUARDED_BY(crit_);
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RtpReceiver* audio_rtp_receiver_ GUARDED_BY(crit_);
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RtpRtcp* audio_rtp_rtcp_ GUARDED_BY(crit_);
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std::unique_ptr<StreamSynchronization> sync_ GUARDED_BY(crit_);
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StreamSynchronization::Measurements audio_measurement_ GUARDED_BY(crit_);
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StreamSynchronization::Measurements video_measurement_ GUARDED_BY(crit_);
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rtc::ThreadChecker process_thread_checker_;
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int64_t last_sync_time_ ACCESS_ON(&process_thread_checker_);
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
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