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 { 46 public: 47 // Factory for the VoEVideoSync sub-API. Increases an internal 48 // reference counter if successful. Returns NULL if the API is not 49 // supported or if construction fails. 50 static VoEVideoSync* GetInterface(VoiceEngine* voiceEngine); 51 52 // Releases the VoEVideoSync sub-API and decreases an internal 53 // reference counter. Returns the new reference count. This value should 54 // be zero for all sub-API:s before the VoiceEngine object can be safely 55 // deleted. 56 virtual int Release() = 0; 57 58 // Gets the current sound card buffer size (playout delay). 59 virtual int GetPlayoutBufferSize(int& buffer_ms) = 0; 60 61 // Sets a minimum target delay for the jitter buffer. This delay is 62 // maintained by the jitter buffer, unless channel condition (jitter in 63 // inter-arrival times) dictates a higher required delay. The overall 64 // jitter buffer delay is max of |delay_ms| and the latency that NetEq 65 // computes based on inter-arrival times and its playout mode. 66 virtual int SetMinimumPlayoutDelay(int channel, int delay_ms) = 0; 67 68 // Sets an initial delay for the playout jitter buffer. The playout of the 69 // audio is delayed by |delay_ms| in milliseconds. Thereafter, the delay is 70 // maintained, unless NetEq's internal mechanism requires a higher latency. 71 // Such a latency is computed based on inter-arrival times and NetEq's 72 // playout mode. 73 virtual int SetInitialPlayoutDelay(int channel, int delay_ms) = 0; 74 75 // Gets the |jitter_buffer_delay_ms| (including the algorithmic delay), and 76 // the |playout_buffer_delay_ms| for a specified |channel|. 77 virtual int GetDelayEstimate(int channel, 78 int* jitter_buffer_delay_ms, 79 int* playout_buffer_delay_ms) = 0; 80 81 // Returns the least required jitter buffer delay. This is computed by the 82 // the jitter buffer based on the inter-arrival time of RTP packets and 83 // playout mode. NetEq maintains this latency unless a higher value is 84 // requested by calling SetMinimumPlayoutDelay(). 85 virtual int GetLeastRequiredDelayMs(int channel) const = 0; 86 87 // Manual initialization of the RTP timestamp. 88 virtual int SetInitTimestamp(int channel, unsigned int timestamp) = 0; 89 90 // Manual initialization of the RTP sequence number. 91 virtual int SetInitSequenceNumber(int channel, short sequenceNumber) = 0; 92 93 // Get the received RTP timestamp 94 virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp) = 0; 95 96 virtual int GetRtpRtcp (int channel, RtpRtcp** rtpRtcpModule, 97 RtpReceiver** rtp_receiver) = 0; 98 99 protected: VoEVideoSync()100 VoEVideoSync() { } ~VoEVideoSync()101 virtual ~VoEVideoSync() { } 102 }; 103 104 } // namespace webrtc 105 106 #endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H 107