The end goal is to remove AcmReceiver::SetInitialDelay. This change is in preparation for that goal. It turns out that AcmReceiver::SetInitialDelay was only invoked through the following call chain, where each method in the chain is never referenced from anywhere else (except from tests in some cases): ViEChannel::SetReceiverBufferingMode -> ViESyncModule::SetTargetBufferingDelay -> VoEVideoSync::SetInitialPlayoutDelay -> Channel::SetInitialPlayoutDelay -> AudioCodingModule::SetInitialPlayoutDelay -> AcmReceiver::SetInitialDelay The start of the chain, ViEChannel::SetReceiverBufferingMode was never referenced. This change deletes all the methods above except AcmReceiver::SetInitialDelay itself, which will be handled in a follow-up change. BUG=webrtc:3520 Review URL: https://codereview.webrtc.org/1421013006 Cr-Commit-Position: refs/heads/master@{#10471}
63 lines
1.9 KiB
C++
63 lines
1.9 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_ENGINE_VIE_SYNC_MODULE_H_
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#define WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/interface/module.h"
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#include "webrtc/system_wrappers/include/tick_util.h"
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#include "webrtc/video_engine/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 CriticalSectionWrapper;
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class RtpRtcp;
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class VideoCodingModule;
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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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int 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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int VoiceChannel();
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// Implements Module.
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int64_t TimeUntilNextProcess() override;
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int32_t Process() override;
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private:
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rtc::scoped_ptr<CriticalSectionWrapper> data_cs_;
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VideoCodingModule* const vcm_;
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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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rtc::scoped_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_ENGINE_VIE_SYNC_MODULE_H_
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