webrtc_m130/webrtc/video/stream_synchronization.cc
mflodman 4cd2790f17 Move RTP for synchroninzation and rename classes, files and variables.
This CL removes (almost) the last RTP references in VideoReceiveStream.
There are still references to RTPFragmentationHeader and SSRCs, which
will be dealt with later.

There are also new GUARDED_BY and thred checker added to the
synchronization class.

When there are othre transports than RTP, there will instead be an
interface + inheritance for RtpStreamReceiver and
RtpStreamSynchronizattion in VideoReceiveStream. This work will be done
when we actually know how we want to make thee transport interface.

BUG=webrtc:5838

Review-Url: https://codereview.webrtc.org/2216533002
Cr-Commit-Position: refs/heads/master@{#13655}
2016-08-05 13:28:50 +00:00

201 lines
7.7 KiB
C++

/*
* 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/stream_synchronization.h"
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include <algorithm>
#include "webrtc/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_primary_ssrc,
int audio_channel_id)
: video_primary_ssrc_(video_primary_ssrc),
audio_channel_id_(audio_channel_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 (!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
<< " 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