1 files changed, 174 insertions, 0 deletions
diff --git a/webrtc/video/vie_sync_module.cc b/webrtc/video/vie_sync_module.cc
new file mode 100644
index 0000000000..9ca9a9480e
--- /dev/null
+++ b/webrtc/video/vie_sync_module.cc
@@ -0,0 +1,174 @@
+/*
+ *  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/vie_sync_module.h"
+
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/video/stream_synchronization.h"
+#include "webrtc/voice_engine/include/voe_video_sync.h"
+
+namespace webrtc {
+
+int UpdateMeasurements(StreamSynchronization::Measurements* stream,
+                       const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) {
+  if (!receiver.Timestamp(&stream->latest_timestamp))
+    return -1;
+  if (!receiver.LastReceivedTimeMs(&stream->latest_receive_time_ms))
+    return -1;
+
+  uint32_t ntp_secs = 0;
+  uint32_t ntp_frac = 0;
+  uint32_t rtp_timestamp = 0;
+  if (0 != rtp_rtcp.RemoteNTP(&ntp_secs,
+                              &ntp_frac,
+                              NULL,
+                              NULL,
+                              &rtp_timestamp)) {
+    return -1;
+  }
+
+  bool new_rtcp_sr = false;
+  if (!UpdateRtcpList(
+      ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
+    return -1;
+  }
+
+  return 0;
+}
+
+ViESyncModule::ViESyncModule(VideoCodingModule* vcm)
+    : data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+      vcm_(vcm),
+      video_receiver_(NULL),
+      video_rtp_rtcp_(NULL),
+      voe_channel_id_(-1),
+      voe_sync_interface_(NULL),
+      last_sync_time_(TickTime::Now()),
+      sync_() {
+}
+
+ViESyncModule::~ViESyncModule() {
+}
+
+int ViESyncModule::ConfigureSync(int voe_channel_id,
+                                 VoEVideoSync* voe_sync_interface,
+                                 RtpRtcp* video_rtcp_module,
+                                 RtpReceiver* video_receiver) {
+  CriticalSectionScoped cs(data_cs_.get());
+  // Prevent expensive no-ops.
+  if (voe_channel_id_ == voe_channel_id &&
+      voe_sync_interface_ == voe_sync_interface &&
+      video_receiver_ == video_receiver &&
+      video_rtp_rtcp_ == video_rtcp_module) {
+    return 0;
+  }
+  voe_channel_id_ = voe_channel_id;
+  voe_sync_interface_ = voe_sync_interface;
+  video_receiver_ = video_receiver;
+  video_rtp_rtcp_ = video_rtcp_module;
+  sync_.reset(
+      new StreamSynchronization(video_rtp_rtcp_->SSRC(), voe_channel_id));
+
+  if (!voe_sync_interface) {
+    voe_channel_id_ = -1;
+    if (voe_channel_id >= 0) {
+      // Trying to set a voice channel but no interface exist.
+      return -1;
+    }
+    return 0;
+  }
+  return 0;
+}
+
+int ViESyncModule::VoiceChannel() {
+  return voe_channel_id_;
+}
+
+int64_t ViESyncModule::TimeUntilNextProcess() {
+  const int64_t kSyncIntervalMs = 1000;
+  return kSyncIntervalMs - (TickTime::Now() - last_sync_time_).Milliseconds();
+}
+
+int32_t ViESyncModule::Process() {
+  CriticalSectionScoped cs(data_cs_.get());
+  last_sync_time_ = TickTime::Now();
+
+  const int current_video_delay_ms = vcm_->Delay();
+
+  if (voe_channel_id_ == -1) {
+    return 0;
+  }
+  assert(video_rtp_rtcp_ && voe_sync_interface_);
+  assert(sync_.get());
+
+  int audio_jitter_buffer_delay_ms = 0;
+  int playout_buffer_delay_ms = 0;
+  if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
+                                            &audio_jitter_buffer_delay_ms,
+                                            &playout_buffer_delay_ms) != 0) {
+    return 0;
+  }
+  const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
+      playout_buffer_delay_ms;
+
+  RtpRtcp* voice_rtp_rtcp = NULL;
+  RtpReceiver* voice_receiver = NULL;
+  if (0 != voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &voice_rtp_rtcp,
+                                           &voice_receiver)) {
+    return 0;
+  }
+  assert(voice_rtp_rtcp);
+  assert(voice_receiver);
+
+  if (UpdateMeasurements(&video_measurement_, *video_rtp_rtcp_,
+                         *video_receiver_) != 0) {
+    return 0;
+  }
+
+  if (UpdateMeasurements(&audio_measurement_, *voice_rtp_rtcp,
+                         *voice_receiver) != 0) {
+    return 0;
+  }
+
+  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 0;
+  }
+
+  TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay", current_video_delay_ms);
+  TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay", current_audio_delay_ms);
+  TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
+  int target_audio_delay_ms = 0;
+  int target_video_delay_ms = current_video_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,
+                            current_audio_delay_ms,
+                            &target_audio_delay_ms,
+                            &target_video_delay_ms)) {
+    return 0;
+  }
+
+  if (voe_sync_interface_->SetMinimumPlayoutDelay(
+      voe_channel_id_, target_audio_delay_ms) == -1) {
+    LOG(LS_ERROR) << "Error setting voice delay.";
+  }
+  vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
+  return 0;
+}
+
+}  // namespace webrtc