Move RTP for synchroninzation and rename classes, files and variables.

webrtc/video/rtp_streams_synchronizer.cc

 /*
  *  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/video/rtp_streams_synchronizer.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.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/video_coding_impl.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video/stream_synchronization.h"
 #include "webrtc/video_frame.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
 namespace {
 int UpdateMeasurements(StreamSynchronization::Measurements* stream,
-                       const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) {
-  if (!receiver.Timestamp(&stream->latest_timestamp))
+                       RtpRtcp* rtp_rtcp, RtpReceiver* receiver) {
+  if (!receiver->Timestamp(&stream->latest_timestamp))
     return -1;
-  if (!receiver.LastReceivedTimeMs(&stream->latest_receive_time_ms))
+  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 (rtp_rtcp.RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
-                         &rtp_timestamp) != 0) {
+  if (rtp_rtcp->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
+                          &rtp_timestamp) != 0) {
     return -1;
   }
 
   bool new_rtcp_sr = false;
   if (!UpdateRtcpList(
       ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
     return -1;
   }
 
   return 0;
 }
 }  // namespace
 
-ViESyncModule::ViESyncModule(vcm::VideoReceiver* video_receiver)
-    : video_receiver_(video_receiver),
-      clock_(Clock::GetRealTimeClock()),
-      rtp_receiver_(nullptr),
-      video_rtp_rtcp_(nullptr),
+RtpStreamsSynchronizer::RtpStreamsSynchronizer(
+    vcm::VideoReceiver* video_receiver,
+    RtpStreamReceiver* rtp_stream_receiver)
+    : clock_(Clock::GetRealTimeClock()),
+      video_receiver_(video_receiver),
+      video_rtp_receiver_(rtp_stream_receiver->GetRtpReceiver()),
+      video_rtp_rtcp_(rtp_stream_receiver->rtp_rtcp()),
       voe_channel_id_(-1),
       voe_sync_interface_(nullptr),
-      last_sync_time_(rtc::TimeNanos()),
-      sync_() {}
-
-ViESyncModule::~ViESyncModule() {
+      audio_rtp_receiver_(nullptr),
+      audio_rtp_rtcp_(nullptr),
+      sync_(),
+      last_sync_time_(rtc::TimeNanos()) {
+  process_thread_checker_.DetachFromThread();
 }
 
-void ViESyncModule::ConfigureSync(int voe_channel_id,
-                                  VoEVideoSync* voe_sync_interface,
-                                  RtpRtcp* video_rtcp_module,
-                                  RtpReceiver* rtp_receiver) {
+void RtpStreamsSynchronizer::ConfigureSync(int voe_channel_id,
+                                           VoEVideoSync* voe_sync_interface) {
   if (voe_channel_id != -1)
     RTC_DCHECK(voe_sync_interface);
-  rtc::CritScope lock(&data_cs_);
-  // Prevent expensive no-ops.
+
+  rtc::CritScope lock(&crit_);
   if (voe_channel_id_ == voe_channel_id &&
-      voe_sync_interface_ == voe_sync_interface &&
-      rtp_receiver_ == rtp_receiver && video_rtp_rtcp_ == video_rtcp_module) {
+      voe_sync_interface_ == voe_sync_interface) {
+    // This prevents expensive no-ops.
     return;
   }
   voe_channel_id_ = voe_channel_id;
   voe_sync_interface_ = voe_sync_interface;
-  rtp_receiver_ = rtp_receiver;
-  video_rtp_rtcp_ = video_rtcp_module;
-  sync_.reset(
-      new StreamSynchronization(video_rtp_rtcp_->SSRC(), voe_channel_id));
+
+  audio_rtp_rtcp_ = nullptr;
+  audio_rtp_receiver_ = nullptr;
+  sync_.reset(nullptr);
+
+  if (voe_channel_id_ != -1) {
+    voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &audio_rtp_rtcp_,
+                                    &audio_rtp_receiver_);
+    RTC_DCHECK(audio_rtp_rtcp_);
+    RTC_DCHECK(audio_rtp_receiver_);
+    sync_.reset(new StreamSynchronization(video_rtp_rtcp_->SSRC(),
+                                          voe_channel_id_));
+  }
 }
 
-int64_t ViESyncModule::TimeUntilNextProcess() {
+int64_t RtpStreamsSynchronizer::TimeUntilNextProcess() {
+  RTC_DCHECK_RUN_ON(&process_thread_checker_);
   const int64_t kSyncIntervalMs = 1000;
   return kSyncIntervalMs -
       (rtc::TimeNanos() - last_sync_time_) / rtc::kNumNanosecsPerMillisec;
 }
 
-void ViESyncModule::Process() {
-  rtc::CritScope lock(&data_cs_);
+void RtpStreamsSynchronizer::Process() {
+  RTC_DCHECK_RUN_ON(&process_thread_checker_);
+
+  const int current_video_delay_ms = video_receiver_->Delay();
   last_sync_time_ = rtc::TimeNanos();
 
-  const int current_video_delay_ms = video_receiver_->Delay();
-
+  rtc::CritScope lock(&crit_);
   if (voe_channel_id_ == -1) {
     return;
   }
-  assert(video_rtp_rtcp_ && voe_sync_interface_);
-  assert(sync_.get());
+  RTC_DCHECK(voe_sync_interface_);
+  RTC_DCHECK(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;
   }
   const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
       playout_buffer_delay_ms;
 
-  RtpRtcp* voice_rtp_rtcp = nullptr;
-  RtpReceiver* voice_receiver = nullptr;
-  if (voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &voice_rtp_rtcp,
-                                      &voice_receiver) != 0) {
-    return;
-  }
-  assert(voice_rtp_rtcp);
-  assert(voice_receiver);
-
-  if (UpdateMeasurements(&video_measurement_, *video_rtp_rtcp_,
-                         *rtp_receiver_) != 0) {
+  if (UpdateMeasurements(&video_measurement_, video_rtp_rtcp_,
+                         video_rtp_receiver_) != 0) {
     return;
   }
 
-  if (UpdateMeasurements(&audio_measurement_, *voice_rtp_rtcp,
-                         *voice_receiver) != 0) {
+  if (UpdateMeasurements(&audio_measurement_, audio_rtp_rtcp_,
+                         audio_rtp_receiver_) != 0) {
     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;
   }
 
@@ skipping 11 matching lines @@
     return;
   }
 
   if (voe_sync_interface_->SetMinimumPlayoutDelay(
       voe_channel_id_, target_audio_delay_ms) == -1) {
     LOG(LS_ERROR) << "Error setting voice delay.";
   }
   video_receiver_->SetMinimumPlayoutDelay(target_video_delay_ms);
 }
 
-bool ViESyncModule::GetStreamSyncOffsetInMs(const VideoFrame& frame,
-                                            int64_t* stream_offset_ms) const {
-  rtc::CritScope lock(&data_cs_);
+bool RtpStreamsSynchronizer::GetStreamSyncOffsetInMs(
+    const VideoFrame& frame, int64_t* stream_offset_ms) const {
+  rtc::CritScope lock(&crit_);
   if (voe_channel_id_ == -1)
     return false;
 
   uint32_t playout_timestamp = 0;
   if (voe_sync_interface_->GetPlayoutTimestamp(voe_channel_id_,
                                                playout_timestamp) != 0) {
     return false;
   }
 
   int64_t latest_audio_ntp;
@@ skipping 11 matching lines @@
   int64_t time_to_render_ms =
       frame.render_time_ms() - clock_->TimeInMilliseconds();
   if (time_to_render_ms > 0)
     latest_video_ntp += time_to_render_ms;
 
   *stream_offset_ms = latest_audio_ntp - latest_video_ntp;
   return true;
 }
 
 }  // namespace webrtc