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

webrtc/video/vie_sync_module.cc

Index: webrtc/video/vie_sync_module.cc
diff --git a/webrtc/video/vie_sync_module.cc b/webrtc/video/vie_sync_module.cc
deleted file mode 100644
index 2e62ff8143938d3591925056657b8a84bfbb082e..0000000000000000000000000000000000000000
--- a/webrtc/video/vie_sync_module.cc
+++ /dev/null
@@ -1,194 +0,0 @@
-/*
- *  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/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))
-    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 (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),
-      voe_channel_id_(-1),
-      voe_sync_interface_(nullptr),
-      last_sync_time_(rtc::TimeNanos()),
-      sync_() {}
-
-ViESyncModule::~ViESyncModule() {
-}
-
-void ViESyncModule::ConfigureSync(int voe_channel_id,
-                                  VoEVideoSync* voe_sync_interface,
-                                  RtpRtcp* video_rtcp_module,
-                                  RtpReceiver* rtp_receiver) {
-  if (voe_channel_id != -1)
-    RTC_DCHECK(voe_sync_interface);
-  rtc::CritScope lock(&data_cs_);
-  // Prevent expensive no-ops.
-  if (voe_channel_id_ == voe_channel_id &&
-      voe_sync_interface_ == voe_sync_interface &&
-      rtp_receiver_ == rtp_receiver && video_rtp_rtcp_ == video_rtcp_module) {
-    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));
-}
-
-int64_t ViESyncModule::TimeUntilNextProcess() {
-  const int64_t kSyncIntervalMs = 1000;
-  return kSyncIntervalMs -
-      (rtc::TimeNanos() - last_sync_time_) / rtc::kNumNanosecsPerMillisec;
-}
-
-void ViESyncModule::Process() {
-  rtc::CritScope lock(&data_cs_);
-  last_sync_time_ = rtc::TimeNanos();
-
-  const int current_video_delay_ms = video_receiver_->Delay();
-
-  if (voe_channel_id_ == -1) {
-    return;
-  }
-  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;
-  }
-  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) {
-    return;
-  }
-
-  if (UpdateMeasurements(&audio_measurement_, *voice_rtp_rtcp,
-                         *voice_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;
-  }
-
-  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;
-  }
-
-  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_);
-  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;
-  if (!RtpToNtpMs(playout_timestamp, audio_measurement_.rtcp,
-                  &latest_audio_ntp)) {
-    return false;
-  }
-
-  int64_t latest_video_ntp;
-  if (!RtpToNtpMs(frame.timestamp(), video_measurement_.rtcp,
-                  &latest_video_ntp)) {
-    return false;
-  }
-
-  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