| /* | 
 |  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 
 |  * | 
 |  *  Use of this source code is governed by a BSD-style license | 
 |  *  that can be found in the LICENSE file in the root of the source | 
 |  *  tree. An additional intellectual property rights grant can be found | 
 |  *  in the file PATENTS.  All contributing project authors may | 
 |  *  be found in the AUTHORS file in the root of the source tree. | 
 |  */ | 
 |  | 
 | #include "video/stream_synchronization.h" | 
 |  | 
 | #include <assert.h> | 
 | #include <math.h> | 
 | #include <stdlib.h> | 
 |  | 
 | #include <algorithm> | 
 |  | 
 | #include "rtc_base/logging.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | static const int kMaxChangeMs = 80; | 
 | static const int kMaxDeltaDelayMs = 10000; | 
 | static const int kFilterLength = 4; | 
 | // Minimum difference between audio and video to warrant a change. | 
 | static const int kMinDeltaMs = 30; | 
 |  | 
 | StreamSynchronization::StreamSynchronization(int video_stream_id, | 
 |                                              int audio_stream_id) | 
 |     : video_stream_id_(video_stream_id), | 
 |       audio_stream_id_(audio_stream_id), | 
 |       base_target_delay_ms_(0), | 
 |       avg_diff_ms_(0) { | 
 | } | 
 |  | 
 | bool StreamSynchronization::ComputeRelativeDelay( | 
 |     const Measurements& audio_measurement, | 
 |     const Measurements& video_measurement, | 
 |     int* relative_delay_ms) { | 
 |   assert(relative_delay_ms); | 
 |   int64_t audio_last_capture_time_ms; | 
 |   if (!audio_measurement.rtp_to_ntp.Estimate(audio_measurement.latest_timestamp, | 
 |                                              &audio_last_capture_time_ms)) { | 
 |     return false; | 
 |   } | 
 |   int64_t video_last_capture_time_ms; | 
 |   if (!video_measurement.rtp_to_ntp.Estimate(video_measurement.latest_timestamp, | 
 |                                              &video_last_capture_time_ms)) { | 
 |     return false; | 
 |   } | 
 |   if (video_last_capture_time_ms < 0) { | 
 |     return false; | 
 |   } | 
 |   // Positive diff means that video_measurement is behind audio_measurement. | 
 |   *relative_delay_ms = video_measurement.latest_receive_time_ms - | 
 |       audio_measurement.latest_receive_time_ms - | 
 |       (video_last_capture_time_ms - audio_last_capture_time_ms); | 
 |   if (*relative_delay_ms > kMaxDeltaDelayMs || | 
 |       *relative_delay_ms < -kMaxDeltaDelayMs) { | 
 |     return false; | 
 |   } | 
 |   return true; | 
 | } | 
 |  | 
 | bool StreamSynchronization::ComputeDelays(int relative_delay_ms, | 
 |                                           int current_audio_delay_ms, | 
 |                                           int* total_audio_delay_target_ms, | 
 |                                           int* total_video_delay_target_ms) { | 
 |   assert(total_audio_delay_target_ms && total_video_delay_target_ms); | 
 |  | 
 |   int current_video_delay_ms = *total_video_delay_target_ms; | 
 |   RTC_LOG(LS_VERBOSE) << "Audio delay: " << current_audio_delay_ms | 
 |                       << " current diff: " << relative_delay_ms | 
 |                       << " for stream " << audio_stream_id_; | 
 |   // Calculate the difference between the lowest possible video delay and | 
 |   // the current audio delay. | 
 |   int current_diff_ms = current_video_delay_ms - current_audio_delay_ms + | 
 |       relative_delay_ms; | 
 |  | 
 |   avg_diff_ms_ = ((kFilterLength - 1) * avg_diff_ms_ + | 
 |       current_diff_ms) / kFilterLength; | 
 |   if (abs(avg_diff_ms_) < kMinDeltaMs) { | 
 |     // Don't adjust if the diff is within our margin. | 
 |     return false; | 
 |   } | 
 |  | 
 |   // Make sure we don't move too fast. | 
 |   int diff_ms = avg_diff_ms_ / 2; | 
 |   diff_ms = std::min(diff_ms, kMaxChangeMs); | 
 |   diff_ms = std::max(diff_ms, -kMaxChangeMs); | 
 |  | 
 |   // Reset the average after a move to prevent overshooting reaction. | 
 |   avg_diff_ms_ = 0; | 
 |  | 
 |   if (diff_ms > 0) { | 
 |     // The minimum video delay is longer than the current audio delay. | 
 |     // We need to decrease extra video delay, or add extra audio delay. | 
 |     if (channel_delay_.extra_video_delay_ms > base_target_delay_ms_) { | 
 |       // We have extra delay added to ViE. Reduce this delay before adding | 
 |       // extra delay to VoE. | 
 |       channel_delay_.extra_video_delay_ms -= diff_ms; | 
 |       channel_delay_.extra_audio_delay_ms = base_target_delay_ms_; | 
 |     } else {  // channel_delay_.extra_video_delay_ms > 0 | 
 |       // We have no extra video delay to remove, increase the audio delay. | 
 |       channel_delay_.extra_audio_delay_ms += diff_ms; | 
 |       channel_delay_.extra_video_delay_ms = base_target_delay_ms_; | 
 |     } | 
 |   } else {  // if (diff_ms > 0) | 
 |     // The video delay is lower than the current audio delay. | 
 |     // We need to decrease extra audio delay, or add extra video delay. | 
 |     if (channel_delay_.extra_audio_delay_ms > base_target_delay_ms_) { | 
 |       // We have extra delay in VoiceEngine. | 
 |       // Start with decreasing the voice delay. | 
 |       // Note: diff_ms is negative; add the negative difference. | 
 |       channel_delay_.extra_audio_delay_ms += diff_ms; | 
 |       channel_delay_.extra_video_delay_ms = base_target_delay_ms_; | 
 |     } else {  // channel_delay_.extra_audio_delay_ms > base_target_delay_ms_ | 
 |       // We have no extra delay in VoiceEngine, increase the video delay. | 
 |       // Note: diff_ms is negative; subtract the negative difference. | 
 |       channel_delay_.extra_video_delay_ms -= diff_ms;  // X - (-Y) = X + Y. | 
 |       channel_delay_.extra_audio_delay_ms = base_target_delay_ms_; | 
 |     } | 
 |   } | 
 |  | 
 |   // Make sure that video is never below our target. | 
 |   channel_delay_.extra_video_delay_ms = std::max( | 
 |       channel_delay_.extra_video_delay_ms, base_target_delay_ms_); | 
 |  | 
 |   int new_video_delay_ms; | 
 |   if (channel_delay_.extra_video_delay_ms > base_target_delay_ms_) { | 
 |     new_video_delay_ms = channel_delay_.extra_video_delay_ms; | 
 |   } else { | 
 |     // No change to the extra video delay. We are changing audio and we only | 
 |     // allow to change one at the time. | 
 |     new_video_delay_ms = channel_delay_.last_video_delay_ms; | 
 |   } | 
 |  | 
 |   // Make sure that we don't go below the extra video delay. | 
 |   new_video_delay_ms = std::max( | 
 |       new_video_delay_ms, channel_delay_.extra_video_delay_ms); | 
 |  | 
 |   // Verify we don't go above the maximum allowed video delay. | 
 |   new_video_delay_ms = | 
 |       std::min(new_video_delay_ms, base_target_delay_ms_ + kMaxDeltaDelayMs); | 
 |  | 
 |   int new_audio_delay_ms; | 
 |   if (channel_delay_.extra_audio_delay_ms > base_target_delay_ms_) { | 
 |     new_audio_delay_ms = channel_delay_.extra_audio_delay_ms; | 
 |   } else { | 
 |     // No change to the audio delay. We are changing video and we only | 
 |     // allow to change one at the time. | 
 |     new_audio_delay_ms = channel_delay_.last_audio_delay_ms; | 
 |   } | 
 |  | 
 |   // Make sure that we don't go below the extra audio delay. | 
 |   new_audio_delay_ms = std::max( | 
 |       new_audio_delay_ms, channel_delay_.extra_audio_delay_ms); | 
 |  | 
 |   // Verify we don't go above the maximum allowed audio delay. | 
 |   new_audio_delay_ms = | 
 |       std::min(new_audio_delay_ms, base_target_delay_ms_ + kMaxDeltaDelayMs); | 
 |  | 
 |   // Remember our last audio and video delays. | 
 |   channel_delay_.last_video_delay_ms = new_video_delay_ms; | 
 |   channel_delay_.last_audio_delay_ms = new_audio_delay_ms; | 
 |  | 
 |   RTC_LOG(LS_VERBOSE) << "Sync video delay " << new_video_delay_ms | 
 |                       << " for video stream " << video_stream_id_ | 
 |                       << " and audio delay " | 
 |                       << channel_delay_.extra_audio_delay_ms | 
 |                       << " for audio stream " << audio_stream_id_; | 
 |  | 
 |   // Return values. | 
 |   *total_video_delay_target_ms = new_video_delay_ms; | 
 |   *total_audio_delay_target_ms = new_audio_delay_ms; | 
 |   return true; | 
 | } | 
 |  | 
 | void StreamSynchronization::SetTargetBufferingDelay(int target_delay_ms) { | 
 |   // Initial extra delay for audio (accounting for existing extra delay). | 
 |   channel_delay_.extra_audio_delay_ms += | 
 |       target_delay_ms - base_target_delay_ms_; | 
 |   channel_delay_.last_audio_delay_ms += | 
 |       target_delay_ms - base_target_delay_ms_; | 
 |  | 
 |   // The video delay is compared to the last value (and how much we can update | 
 |   // is limited by that as well). | 
 |   channel_delay_.last_video_delay_ms += | 
 |       target_delay_ms - base_target_delay_ms_; | 
 |  | 
 |   channel_delay_.extra_video_delay_ms += | 
 |       target_delay_ms - base_target_delay_ms_; | 
 |  | 
 |   // Video is already delayed by the desired amount. | 
 |   base_target_delay_ms_ = target_delay_ms; | 
 | } | 
 |  | 
 | }  // namespace webrtc |