blob: d86cc792033f5eb8555a96494fef301bc4b79deb [file] [log] [blame]
/*
* 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(uint32_t video_stream_id,
uint32_t 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) {
NtpTime audio_last_capture_time =
audio_measurement.rtp_to_ntp.Estimate(audio_measurement.latest_timestamp);
if (!audio_last_capture_time.Valid()) {
return false;
}
NtpTime video_last_capture_time =
video_measurement.rtp_to_ntp.Estimate(video_measurement.latest_timestamp);
if (!video_last_capture_time.Valid()) {
return false;
}
int64_t audio_last_capture_time_ms = audio_last_capture_time.ToMs();
int64_t video_last_capture_time_ms = video_last_capture_time.ToMs();
// 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;
}
void StreamSynchronization::ReduceAudioDelay() {
audio_delay_.extra_ms *= 0.9f;
}
void StreamSynchronization::ReduceVideoDelay() {
video_delay_.extra_ms *= 0.9f;
}
} // namespace webrtc