
Review URL: https://webrtc-codereview.appspot.com/1326008 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3884 4adac7df-926f-26a2-2b94-8c16560cd09d
242 lines
9.6 KiB
C++
242 lines
9.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "video_engine/stream_synchronization.h"
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#include <assert.h>
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#include <algorithm>
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#include <cmath>
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#include "system_wrappers/interface/trace.h"
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namespace webrtc {
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static const int kMaxChangeMs = 80;
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static const int kMaxDeltaDelayMs = 10000;
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static const int kFilterLength = 4;
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// Minimum difference between audio and video to warrant a change.
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static const int kMinDeltaMs = 30;
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struct ViESyncDelay {
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ViESyncDelay() {
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extra_video_delay_ms = 0;
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last_video_delay_ms = 0;
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extra_audio_delay_ms = 0;
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last_sync_delay = 0;
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network_delay = 120;
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}
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int extra_video_delay_ms;
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int last_video_delay_ms;
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int extra_audio_delay_ms;
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int last_sync_delay;
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int network_delay;
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};
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StreamSynchronization::StreamSynchronization(int audio_channel_id,
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int video_channel_id)
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: channel_delay_(new ViESyncDelay),
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audio_channel_id_(audio_channel_id),
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video_channel_id_(video_channel_id),
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base_target_delay_ms_(0),
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avg_diff_ms_(0) {}
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StreamSynchronization::~StreamSynchronization() {
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delete channel_delay_;
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}
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bool StreamSynchronization::ComputeRelativeDelay(
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const Measurements& audio_measurement,
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const Measurements& video_measurement,
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int* relative_delay_ms) {
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assert(relative_delay_ms);
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if (audio_measurement.rtcp.size() < 2 || video_measurement.rtcp.size() < 2) {
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// We need two RTCP SR reports per stream to do synchronization.
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return false;
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}
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int64_t audio_last_capture_time_ms;
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if (!synchronization::RtpToNtpMs(audio_measurement.latest_timestamp,
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audio_measurement.rtcp,
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&audio_last_capture_time_ms)) {
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return false;
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}
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int64_t video_last_capture_time_ms;
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if (!synchronization::RtpToNtpMs(video_measurement.latest_timestamp,
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video_measurement.rtcp,
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&video_last_capture_time_ms)) {
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return false;
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}
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if (video_last_capture_time_ms < 0) {
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return false;
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}
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// Positive diff means that video_measurement is behind audio_measurement.
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*relative_delay_ms = video_measurement.latest_receive_time_ms -
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audio_measurement.latest_receive_time_ms -
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(video_last_capture_time_ms - audio_last_capture_time_ms);
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if (*relative_delay_ms > kMaxDeltaDelayMs ||
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*relative_delay_ms < -kMaxDeltaDelayMs) {
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return false;
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}
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return true;
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}
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bool StreamSynchronization::ComputeDelays(int relative_delay_ms,
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int current_audio_delay_ms,
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int* extra_audio_delay_ms,
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int* total_video_delay_target_ms) {
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assert(extra_audio_delay_ms && total_video_delay_target_ms);
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WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, video_channel_id_,
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"Audio delay is: %d for voice channel: %d",
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current_audio_delay_ms, audio_channel_id_);
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WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, video_channel_id_,
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"Network delay diff is: %d for voice channel: %d",
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channel_delay_->network_delay, audio_channel_id_);
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// Calculate the difference between the lowest possible video delay and
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// the current audio delay.
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WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, video_channel_id_,
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"Current diff is: %d for audio channel: %d",
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relative_delay_ms, audio_channel_id_);
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int current_diff_ms = *total_video_delay_target_ms - current_audio_delay_ms +
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relative_delay_ms;
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avg_diff_ms_ = ((kFilterLength - 1) * avg_diff_ms_ +
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current_diff_ms) / kFilterLength;
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if (abs(avg_diff_ms_) < kMinDeltaMs) {
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// Don't adjust if the diff is within our margin.
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return false;
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}
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// Make sure we don't move too fast.
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int diff_ms = avg_diff_ms_ / 2;
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diff_ms = std::min(diff_ms, kMaxChangeMs);
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diff_ms = std::max(diff_ms, -kMaxChangeMs);
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int video_delay_ms = base_target_delay_ms_;
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if (diff_ms > 0) {
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// The minimum video delay is longer than the current audio delay.
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// We need to decrease extra video delay, if we have added extra delay
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// earlier, or add extra audio delay.
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if (channel_delay_->extra_video_delay_ms > 0) {
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// We have extra delay added to ViE. Reduce this delay before adding
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// extra delay to VoE.
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// This is the desired delay, we can't reduce more than this.
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video_delay_ms = *total_video_delay_target_ms;
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// Check that we don't reduce the delay more than what is allowed.
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if (video_delay_ms < channel_delay_->last_video_delay_ms - diff_ms) {
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video_delay_ms = channel_delay_->last_video_delay_ms - diff_ms;
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channel_delay_->extra_video_delay_ms =
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video_delay_ms - *total_video_delay_target_ms;
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} else {
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channel_delay_->extra_video_delay_ms = 0;
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}
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channel_delay_->last_video_delay_ms = video_delay_ms;
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channel_delay_->last_sync_delay = -1;
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channel_delay_->extra_audio_delay_ms = base_target_delay_ms_;
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} else { // channel_delay_->extra_video_delay_ms > 0
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// We have no extra video delay to remove, increase the audio delay.
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if (channel_delay_->last_sync_delay >= 0) {
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// We have increased the audio delay earlier, increase it even more.
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// Increase the audio delay.
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channel_delay_->extra_audio_delay_ms += diff_ms;
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// Don't set a too high delay.
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channel_delay_->extra_audio_delay_ms = std::min(
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channel_delay_->extra_audio_delay_ms,
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base_target_delay_ms_ + kMaxDeltaDelayMs);
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// Don't add any extra video delay.
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video_delay_ms = *total_video_delay_target_ms;
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channel_delay_->extra_video_delay_ms = 0;
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channel_delay_->last_video_delay_ms = video_delay_ms;
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channel_delay_->last_sync_delay = 1;
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} else { // channel_delay_->last_sync_delay >= 0
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// First time after a delay change, don't add any extra delay.
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// This is to not toggle back and forth too much.
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channel_delay_->extra_audio_delay_ms = base_target_delay_ms_;
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// Set minimum video delay
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video_delay_ms = *total_video_delay_target_ms;
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channel_delay_->extra_video_delay_ms = 0;
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channel_delay_->last_video_delay_ms = video_delay_ms;
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channel_delay_->last_sync_delay = 0;
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}
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}
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} else { // if (diff_ms > 0)
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// The minimum video delay is lower than the current audio delay.
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// We need to decrease possible extra audio delay, or
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// add extra video delay.
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if (channel_delay_->extra_audio_delay_ms > base_target_delay_ms_) {
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// We have extra delay in VoiceEngine.
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// Start with decreasing the voice delay.
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// Note: diff_ms is negative; add the negative difference.
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channel_delay_->extra_audio_delay_ms += diff_ms;
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if (channel_delay_->extra_audio_delay_ms < 0) {
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// Negative values not allowed.
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channel_delay_->extra_audio_delay_ms = base_target_delay_ms_;
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channel_delay_->last_sync_delay = 0;
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} else {
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// There is more audio delay to use for the next round.
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channel_delay_->last_sync_delay = 1;
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}
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// Keep the video delay at the minimum values.
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video_delay_ms = *total_video_delay_target_ms;
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channel_delay_->extra_video_delay_ms = 0;
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channel_delay_->last_video_delay_ms = video_delay_ms;
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} else { // channel_delay_->extra_audio_delay_ms > 0
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// We have no extra delay in VoiceEngine, increase the video delay.
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channel_delay_->extra_audio_delay_ms = base_target_delay_ms_;
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// This is the desired delay.
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// Note: diff_ms is negative.
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video_delay_ms = std::max(*total_video_delay_target_ms,
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channel_delay_->last_video_delay_ms) - diff_ms;
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// Verify we don't go above the maximum allowed delay.
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video_delay_ms = std::min(video_delay_ms,
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base_target_delay_ms_ + kMaxDeltaDelayMs);
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// Store the values.
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channel_delay_->extra_video_delay_ms =
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video_delay_ms - *total_video_delay_target_ms;
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channel_delay_->last_video_delay_ms = video_delay_ms;
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channel_delay_->last_sync_delay = -1;
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}
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}
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avg_diff_ms_ = 0;
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WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, video_channel_id_,
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"Sync video delay %d ms for video channel and audio delay %d for audio "
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"channel %d",
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video_delay_ms, channel_delay_->extra_audio_delay_ms, audio_channel_id_);
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*extra_audio_delay_ms = channel_delay_->extra_audio_delay_ms;
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video_delay_ms = std::max(video_delay_ms, 0);
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*total_video_delay_target_ms = std::max(*total_video_delay_target_ms,
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video_delay_ms);
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return true;
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}
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void StreamSynchronization::SetTargetBufferingDelay(int target_delay_ms) {
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// Initial extra delay for audio (accounting for existing extra delay).
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channel_delay_->extra_audio_delay_ms +=
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target_delay_ms - base_target_delay_ms_;
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// The video delay is compared to the last value (and how much we can update
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// is limited by that as well).
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channel_delay_->last_video_delay_ms +=
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target_delay_ms - base_target_delay_ms_;
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// Video is already delayed by the desired amount.
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base_target_delay_ms_ = target_delay_ms;
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}
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} // namespace webrtc
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