f8cdd184d5
"WebRTC.Video.AVSyncOffsetInMs" The absolute value of the sync offset between a rendered video frame and the latest played audio frame is measured per video frame. The average offset per received video stream is recorded when a stream is removed. Updated sync tests in call_perf_tests.cc to use this implementation. BUG=webrtc:5493 Review URL: https://codereview.webrtc.org/1756193005 Cr-Commit-Position: refs/heads/master@{#11993}
70 lines
2.1 KiB
C++
70 lines
2.1 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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// ViESyncModule is responsible for synchronization audio and video for a given
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// VoE and ViE channel couple.
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#ifndef WEBRTC_VIDEO_VIE_SYNC_MODULE_H_
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#define WEBRTC_VIDEO_VIE_SYNC_MODULE_H_
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#include <memory>
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/modules/include/module.h"
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#include "webrtc/system_wrappers/include/tick_util.h"
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#include "webrtc/video/stream_synchronization.h"
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#include "webrtc/voice_engine/include/voe_video_sync.h"
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namespace webrtc {
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class Clock;
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class RtpRtcp;
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class VideoCodingModule;
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class VideoFrame;
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class ViEChannel;
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class VoEVideoSync;
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class ViESyncModule : public Module {
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public:
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explicit ViESyncModule(VideoCodingModule* vcm);
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~ViESyncModule();
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void ConfigureSync(int voe_channel_id,
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VoEVideoSync* voe_sync_interface,
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RtpRtcp* video_rtcp_module,
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RtpReceiver* video_receiver);
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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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rtc::CriticalSection data_cs_;
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VideoCodingModule* const vcm_;
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Clock* const clock_;
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RtpReceiver* video_receiver_;
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RtpRtcp* video_rtp_rtcp_;
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int voe_channel_id_;
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VoEVideoSync* voe_sync_interface_;
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TickTime last_sync_time_;
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std::unique_ptr<StreamSynchronization> sync_;
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StreamSynchronization::Measurements audio_measurement_;
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StreamSynchronization::Measurements video_measurement_;
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_VIE_SYNC_MODULE_H_
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