Revert of Merge webrtc/video_engine/ into webrtc/video/

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 9ca9a9480e81a8b59b8a61edacb0d2e35879346b..0000000000000000000000000000000000000000
--- a/webrtc/video/vie_sync_module.cc
+++ /dev/null
@@ -1,174 +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/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