1 /* 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 // This sub-API supports the following functionalities: 12 // 13 // - RTP header modification (time stamp and sequence number fields). 14 // - Playout delay tuning to synchronize the voice with video. 15 // - Playout delay monitoring. 16 // 17 // Usage example, omitting error checking: 18 // 19 // using namespace webrtc; 20 // VoiceEngine* voe = VoiceEngine::Create(); 21 // VoEBase* base = VoEBase::GetInterface(voe); 22 // VoEVideoSync* vsync = VoEVideoSync::GetInterface(voe); 23 // base->Init(); 24 // ... 25 // int buffer_ms(0); 26 // vsync->GetPlayoutBufferSize(buffer_ms); 27 // ... 28 // base->Terminate(); 29 // base->Release(); 30 // vsync->Release(); 31 // VoiceEngine::Delete(voe); 32 // 33 #ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H 34 #define WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H 35 36 #include "webrtc/common_types.h" 37 38 namespace webrtc { 39 40 class RtpReceiver; 41 class RtpRtcp; 42 class VoiceEngine; 43 44 class WEBRTC_DLLEXPORT VoEVideoSync { 45 public: 46 // Factory for the VoEVideoSync sub-API. Increases an internal 47 // reference counter if successful. Returns NULL if the API is not 48 // supported or if construction fails. 49 static VoEVideoSync* GetInterface(VoiceEngine* voiceEngine); 50 51 // Releases the VoEVideoSync sub-API and decreases an internal 52 // reference counter. Returns the new reference count. This value should 53 // be zero for all sub-API:s before the VoiceEngine object can be safely 54 // deleted. 55 virtual int Release() = 0; 56 57 // Gets the current sound card buffer size (playout delay). 58 virtual int GetPlayoutBufferSize(int& buffer_ms) = 0; 59 60 // Sets a minimum target delay for the jitter buffer. This delay is 61 // maintained by the jitter buffer, unless channel condition (jitter in 62 // inter-arrival times) dictates a higher required delay. The overall 63 // jitter buffer delay is max of |delay_ms| and the latency that NetEq 64 // computes based on inter-arrival times and its playout mode. 65 virtual int SetMinimumPlayoutDelay(int channel, int delay_ms) = 0; 66 67 // Gets the |jitter_buffer_delay_ms| (including the algorithmic delay), and 68 // the |playout_buffer_delay_ms| for a specified |channel|. 69 virtual int GetDelayEstimate(int channel, 70 int* jitter_buffer_delay_ms, 71 int* playout_buffer_delay_ms) = 0; 72 73 // Returns the least required jitter buffer delay. This is computed by the 74 // the jitter buffer based on the inter-arrival time of RTP packets and 75 // playout mode. NetEq maintains this latency unless a higher value is 76 // requested by calling SetMinimumPlayoutDelay(). 77 virtual int GetLeastRequiredDelayMs(int channel) const = 0; 78 79 // Manual initialization of the RTP timestamp. 80 virtual int SetInitTimestamp(int channel, unsigned int timestamp) = 0; 81 82 // Manual initialization of the RTP sequence number. 83 virtual int SetInitSequenceNumber(int channel, short sequenceNumber) = 0; 84 85 // Get the received RTP timestamp 86 virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp) = 0; 87 88 virtual int GetRtpRtcp(int channel, 89 RtpRtcp** rtpRtcpModule, 90 RtpReceiver** rtp_receiver) = 0; 91 92 protected: 93 VoEVideoSync() {} 94 virtual ~VoEVideoSync() {} 95 }; 96 97 } // namespace webrtc 98 99 #endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H 100
