blob: 8e51429701ca90d945aa890d893b638b38695dc7 [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 "webrtc/video_engine/stream_synchronization.h"
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include <algorithm>
#include "webrtc/system_wrappers/interface/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;
struct ViESyncDelay {
ViESyncDelay() {
extra_video_delay_ms = 0;
last_video_delay_ms = 0;
extra_audio_delay_ms = 0;
last_audio_delay_ms = 0;
network_delay = 120;
}
int extra_video_delay_ms;
int last_video_delay_ms;
int extra_audio_delay_ms;
int last_audio_delay_ms;
int network_delay;
};
StreamSynchronization::StreamSynchronization(uint32_t video_primary_ssrc,
int audio_channel_id)
: channel_delay_(new ViESyncDelay),
video_primary_ssrc_(video_primary_ssrc),
audio_channel_id_(audio_channel_id),
base_target_delay_ms_(0),
avg_diff_ms_(0) {
}
StreamSynchronization::~StreamSynchronization() {
delete channel_delay_;
}
bool StreamSynchronization::ComputeRelativeDelay(
const Measurements& audio_measurement,
const Measurements& video_measurement,
int* relative_delay_ms) {
assert(relative_delay_ms);
if (audio_measurement.rtcp.size() < 2 || video_measurement.rtcp.size() < 2) {
// We need two RTCP SR reports per stream to do synchronization.
return false;
}
int64_t audio_last_capture_time_ms;
if (!RtpToNtpMs(audio_measurement.latest_timestamp,
audio_measurement.rtcp,
&audio_last_capture_time_ms)) {
return false;
}
int64_t video_last_capture_time_ms;
if (!RtpToNtpMs(video_measurement.latest_timestamp,
video_measurement.rtcp,
&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;
LOG(LS_VERBOSE) << "Audio delay: " << current_audio_delay_ms
<< ", network delay diff: " << channel_delay_->network_delay
<< " current diff: " << relative_delay_ms
<< " for channel " << audio_channel_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;
LOG(LS_VERBOSE) << "Sync video delay " << new_video_delay_ms
<< " for video primary SSRC " << video_primary_ssrc_
<< " and audio delay " << channel_delay_->extra_audio_delay_ms
<< " for audio channel " << audio_channel_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