1 /* 2 * Copyright (c) 2012 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 #ifndef WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_ 12 #define WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_ 13 14 #include <list> 15 16 #include "webrtc/system_wrappers/include/rtp_to_ntp.h" 17 #include "webrtc/typedefs.h" 18 19 namespace webrtc { 20 21 struct ViESyncDelay; 22 23 class StreamSynchronization { 24 public: 25 struct Measurements { 26 Measurements() : rtcp(), latest_receive_time_ms(0), latest_timestamp(0) {} 27 RtcpList rtcp; 28 int64_t latest_receive_time_ms; 29 uint32_t latest_timestamp; 30 }; 31 32 StreamSynchronization(uint32_t video_primary_ssrc, int audio_channel_id); 33 ~StreamSynchronization(); 34 35 bool ComputeDelays(int relative_delay_ms, 36 int current_audio_delay_ms, 37 int* extra_audio_delay_ms, 38 int* total_video_delay_target_ms); 39 40 // On success |relative_delay| contains the number of milliseconds later video 41 // is rendered relative audio. If audio is played back later than video a 42 // |relative_delay| will be negative. 43 static bool ComputeRelativeDelay(const Measurements& audio_measurement, 44 const Measurements& video_measurement, 45 int* relative_delay_ms); 46 // Set target buffering delay - All audio and video will be delayed by at 47 // least target_delay_ms. 48 void SetTargetBufferingDelay(int target_delay_ms); 49 50 private: 51 ViESyncDelay* channel_delay_; 52 const uint32_t video_primary_ssrc_; 53 const int audio_channel_id_; 54 int base_target_delay_ms_; 55 int avg_diff_ms_; 56 }; 57 } // namespace webrtc 58 59 #endif // WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_ 60
