Adding upstream version 115.8.0esr.upstream/115.8.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 195 insertions, 0 deletions

diff --git a/third_party/libwebrtc/video/stream_synchronization.cc b/third_party/libwebrtc/video/stream_synchronization.cc
new file mode 100644
index 0000000000..d86cc79203
--- /dev/null
+++ b/third_party/libwebrtc/video/stream_synchronization.cc
@@ -0,0 +1,195 @@
+/*
+ *  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