Adding upstream version 115.8.0esr.upstream/115.8.0esr

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

1 files changed, 219 insertions, 0 deletions

diff --git a/third_party/libwebrtc/video/rtp_streams_synchronizer2.cc b/third_party/libwebrtc/video/rtp_streams_synchronizer2.cc
new file mode 100644
index 0000000000..0fbb3916cb
--- /dev/null
+++ b/third_party/libwebrtc/video/rtp_streams_synchronizer2.cc
@@ -0,0 +1,219 @@
+/*
+ *  Copyright (c) 2020 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/rtp_streams_synchronizer2.h"
+
+#include "absl/types/optional.h"
+#include "call/syncable.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/rtp_to_ntp_estimator.h"
+
+namespace webrtc {
+namespace internal {
+namespace {
+// Time interval for logging stats.
+constexpr int64_t kStatsLogIntervalMs = 10000;
+constexpr TimeDelta kSyncInterval = TimeDelta::Millis(1000);
+
+bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
+                        const Syncable::Info& info) {
+  stream->latest_timestamp = info.latest_received_capture_timestamp;
+  stream->latest_receive_time_ms = info.latest_receive_time_ms;
+  return stream->rtp_to_ntp.UpdateMeasurements(
+             NtpTime(info.capture_time_ntp_secs, info.capture_time_ntp_frac),
+             info.capture_time_source_clock) !=
+         RtpToNtpEstimator::kInvalidMeasurement;
+}
+
+}  // namespace
+
+RtpStreamsSynchronizer::RtpStreamsSynchronizer(TaskQueueBase* main_queue,
+                                               Syncable* syncable_video)
+    : task_queue_(main_queue),
+      syncable_video_(syncable_video),
+      last_stats_log_ms_(rtc::TimeMillis()) {
+  RTC_DCHECK(syncable_video);
+}
+
+RtpStreamsSynchronizer::~RtpStreamsSynchronizer() {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  repeating_task_.Stop();
+}
+
+void RtpStreamsSynchronizer::ConfigureSync(Syncable* syncable_audio) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+
+  // Prevent expensive no-ops.
+  if (syncable_audio == syncable_audio_)
+    return;
+
+  syncable_audio_ = syncable_audio;
+  sync_.reset(nullptr);
+  if (!syncable_audio_) {
+    repeating_task_.Stop();
+    return;
+  }
+
+  sync_.reset(
+      new StreamSynchronization(syncable_video_->id(), syncable_audio_->id()));
+
+  if (repeating_task_.Running())
+    return;
+
+  repeating_task_ =
+      RepeatingTaskHandle::DelayedStart(task_queue_, kSyncInterval, [this]() {
+        UpdateDelay();
+        return kSyncInterval;
+      });
+}
+
+void RtpStreamsSynchronizer::UpdateDelay() {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+
+  if (!syncable_audio_)
+    return;
+
+  RTC_DCHECK(sync_.get());
+
+  bool log_stats = false;
+  const int64_t now_ms = rtc::TimeMillis();
+  if (now_ms - last_stats_log_ms_ > kStatsLogIntervalMs) {
+    last_stats_log_ms_ = now_ms;
+    log_stats = true;
+  }
+
+  int64_t last_audio_receive_time_ms =
+      audio_measurement_.latest_receive_time_ms;
+  absl::optional<Syncable::Info> audio_info = syncable_audio_->GetInfo();
+  if (!audio_info || !UpdateMeasurements(&audio_measurement_, *audio_info)) {
+    return;
+  }
+
+  if (last_audio_receive_time_ms == audio_measurement_.latest_receive_time_ms) {
+    // No new audio packet has been received since last update.
+    return;
+  }
+
+  int64_t last_video_receive_ms = video_measurement_.latest_receive_time_ms;
+  absl::optional<Syncable::Info> video_info = syncable_video_->GetInfo();
+  if (!video_info || !UpdateMeasurements(&video_measurement_, *video_info)) {
+    return;
+  }
+
+  if (last_video_receive_ms == video_measurement_.latest_receive_time_ms) {
+    // No new video packet has been received since last update.
+    return;
+  }
+
+  int relative_delay_ms;
+  // Calculate how much later or earlier the audio stream is compared to video.
+  if (!sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_,
+                                   &relative_delay_ms)) {
+    return;
+  }
+
+  if (log_stats) {
+    RTC_LOG(LS_INFO) << "Sync info stats: " << now_ms
+                     << ", {ssrc: " << sync_->audio_stream_id() << ", "
+                     << "cur_delay_ms: " << audio_info->current_delay_ms
+                     << "} {ssrc: " << sync_->video_stream_id() << ", "
+                     << "cur_delay_ms: " << video_info->current_delay_ms
+                     << "} {relative_delay_ms: " << relative_delay_ms << "} ";
+  }
+
+  TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay",
+                 video_info->current_delay_ms);
+  TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay",
+                 audio_info->current_delay_ms);
+  TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
+
+  int target_audio_delay_ms = 0;
+  int target_video_delay_ms = video_info->current_delay_ms;
+  // Calculate the necessary extra audio delay and desired total video
+  // delay to get the streams in sync.
+  if (!sync_->ComputeDelays(relative_delay_ms, audio_info->current_delay_ms,
+                            &target_audio_delay_ms, &target_video_delay_ms)) {
+    return;
+  }
+
+  if (log_stats) {
+    RTC_LOG(LS_INFO) << "Sync delay stats: " << now_ms
+                     << ", {ssrc: " << sync_->audio_stream_id() << ", "
+                     << "target_delay_ms: " << target_audio_delay_ms
+                     << "} {ssrc: " << sync_->video_stream_id() << ", "
+                     << "target_delay_ms: " << target_video_delay_ms << "} ";
+  }
+
+  if (!syncable_audio_->SetMinimumPlayoutDelay(target_audio_delay_ms)) {
+    sync_->ReduceAudioDelay();
+  }
+  if (!syncable_video_->SetMinimumPlayoutDelay(target_video_delay_ms)) {
+    sync_->ReduceVideoDelay();
+  }
+}
+
+// TODO(https://bugs.webrtc.org/7065): Move RtpToNtpEstimator out of
+// RtpStreamsSynchronizer and into respective receive stream to always populate
+// the estimated playout timestamp.
+bool RtpStreamsSynchronizer::GetStreamSyncOffsetInMs(
+    uint32_t rtp_timestamp,
+    int64_t render_time_ms,
+    int64_t* video_playout_ntp_ms,
+    int64_t* stream_offset_ms,
+    double* estimated_freq_khz) const {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+
+  if (!syncable_audio_)
+    return false;
+
+  uint32_t audio_rtp_timestamp;
+  int64_t time_ms;
+  if (!syncable_audio_->GetPlayoutRtpTimestamp(&audio_rtp_timestamp,
+                                               &time_ms)) {
+    return false;
+  }
+
+  NtpTime latest_audio_ntp =
+      audio_measurement_.rtp_to_ntp.Estimate(audio_rtp_timestamp);
+  if (!latest_audio_ntp.Valid()) {
+    return false;
+  }
+  int64_t latest_audio_ntp_ms = latest_audio_ntp.ToMs();
+
+  syncable_audio_->SetEstimatedPlayoutNtpTimestampMs(latest_audio_ntp_ms,
+                                                     time_ms);
+
+  NtpTime latest_video_ntp =
+      video_measurement_.rtp_to_ntp.Estimate(rtp_timestamp);
+  if (!latest_video_ntp.Valid()) {
+    return false;
+  }
+  int64_t latest_video_ntp_ms = latest_video_ntp.ToMs();
+
+  // Current audio ntp.
+  int64_t now_ms = rtc::TimeMillis();
+  latest_audio_ntp_ms += (now_ms - time_ms);
+
+  // Remove video playout delay.
+  int64_t time_to_render_ms = render_time_ms - now_ms;
+  if (time_to_render_ms > 0)
+    latest_video_ntp_ms -= time_to_render_ms;
+
+  *video_playout_ntp_ms = latest_video_ntp_ms;
+  *stream_offset_ms = latest_audio_ntp_ms - latest_video_ntp_ms;
+  *estimated_freq_khz = video_measurement_.rtp_to_ntp.EstimatedFrequencyKhz();
+  return true;
+}
+
+}  // namespace internal
+}  // namespace webrtc