| /* |
| * 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 <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) { |
| 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) { |
| 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 (video_delay_.extra_ms > base_target_delay_ms_) { |
| // We have extra delay added to ViE. Reduce this delay before adding |
| // extra delay to VoE. |
| video_delay_.extra_ms -= diff_ms; |
| audio_delay_.extra_ms = base_target_delay_ms_; |
| } else { // video_delay_.extra_ms > 0 |
| // We have no extra video delay to remove, increase the audio delay. |
| audio_delay_.extra_ms += diff_ms; |
| video_delay_.extra_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 (audio_delay_.extra_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. |
| audio_delay_.extra_ms += diff_ms; |
| video_delay_.extra_ms = base_target_delay_ms_; |
| } else { // audio_delay_.extra_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. |
| video_delay_.extra_ms -= diff_ms; // X - (-Y) = X + Y. |
| audio_delay_.extra_ms = base_target_delay_ms_; |
| } |
| } |
| |
| // Make sure that video is never below our target. |
| video_delay_.extra_ms = |
| std::max(video_delay_.extra_ms, base_target_delay_ms_); |
| |
| int new_video_delay_ms; |
| if (video_delay_.extra_ms > base_target_delay_ms_) { |
| new_video_delay_ms = video_delay_.extra_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 = video_delay_.last_ms; |
| } |
| |
| // Make sure that we don't go below the extra video delay. |
| new_video_delay_ms = std::max(new_video_delay_ms, video_delay_.extra_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 (audio_delay_.extra_ms > base_target_delay_ms_) { |
| new_audio_delay_ms = audio_delay_.extra_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 = audio_delay_.last_ms; |
| } |
| |
| // Make sure that we don't go below the extra audio delay. |
| new_audio_delay_ms = std::max(new_audio_delay_ms, audio_delay_.extra_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); |
| |
| video_delay_.last_ms = new_video_delay_ms; |
| audio_delay_.last_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 " << audio_delay_.extra_ms |
| << " for audio stream " << audio_stream_id_; |
| |
| *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). |
| audio_delay_.extra_ms += target_delay_ms - base_target_delay_ms_; |
| audio_delay_.last_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). |
| video_delay_.last_ms += target_delay_ms - base_target_delay_ms_; |
| |
| video_delay_.extra_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 |