100 lines
3.6 KiB
C++
100 lines
3.6 KiB
C++
/*
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* Copyright (c) 2011 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 sub-API supports the following functionalities:
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//
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// - RTP header modification (time stamp and sequence number fields).
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// - Playout delay tuning to synchronize the voice with video.
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// - Playout delay monitoring.
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//
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// Usage example, omitting error checking:
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//
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// using namespace webrtc;
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// VoiceEngine* voe = VoiceEngine::Create();
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// VoEBase* base = VoEBase::GetInterface(voe);
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// VoEVideoSync* vsync = VoEVideoSync::GetInterface(voe);
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// base->Init();
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// ...
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// int buffer_ms(0);
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// vsync->GetPlayoutBufferSize(buffer_ms);
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// ...
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// base->Terminate();
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// base->Release();
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// vsync->Release();
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// VoiceEngine::Delete(voe);
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//
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#ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
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#define WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
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#include "webrtc/common_types.h"
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namespace webrtc {
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class RtpReceiver;
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class RtpRtcp;
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class VoiceEngine;
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class WEBRTC_DLLEXPORT VoEVideoSync {
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public:
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// Factory for the VoEVideoSync sub-API. Increases an internal
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// reference counter if successful. Returns NULL if the API is not
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// supported or if construction fails.
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static VoEVideoSync* GetInterface(VoiceEngine* voiceEngine);
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// Releases the VoEVideoSync sub-API and decreases an internal
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// reference counter. Returns the new reference count. This value should
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// be zero for all sub-API:s before the VoiceEngine object can be safely
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// deleted.
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virtual int Release() = 0;
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// Gets the current sound card buffer size (playout delay).
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virtual int GetPlayoutBufferSize(int& buffer_ms) = 0;
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// Sets a minimum target delay for the jitter buffer. This delay is
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// maintained by the jitter buffer, unless channel condition (jitter in
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// inter-arrival times) dictates a higher required delay. The overall
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// jitter buffer delay is max of |delay_ms| and the latency that NetEq
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// computes based on inter-arrival times and its playout mode.
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virtual int SetMinimumPlayoutDelay(int channel, int delay_ms) = 0;
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// Gets the |jitter_buffer_delay_ms| (including the algorithmic delay), and
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// the |playout_buffer_delay_ms| for a specified |channel|.
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virtual int GetDelayEstimate(int channel,
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int* jitter_buffer_delay_ms,
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int* playout_buffer_delay_ms) = 0;
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// Returns the least required jitter buffer delay. This is computed by the
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// the jitter buffer based on the inter-arrival time of RTP packets and
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// playout mode. NetEq maintains this latency unless a higher value is
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// requested by calling SetMinimumPlayoutDelay().
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virtual int GetLeastRequiredDelayMs(int channel) const = 0;
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// Manual initialization of the RTP timestamp.
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virtual int SetInitTimestamp(int channel, unsigned int timestamp) = 0;
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// Manual initialization of the RTP sequence number.
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virtual int SetInitSequenceNumber(int channel, short sequenceNumber) = 0;
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// Get the received RTP timestamp
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virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp) = 0;
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virtual int GetRtpRtcp(int channel,
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RtpRtcp** rtpRtcpModule,
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RtpReceiver** rtp_receiver) = 0;
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protected:
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VoEVideoSync() {}
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virtual ~VoEVideoSync() {}
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};
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} // namespace webrtc
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#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
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