1/*
2 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11#include "webrtc/video/vie_sync_module.h"
12
13#include "webrtc/base/logging.h"
14#include "webrtc/base/trace_event.h"
15#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
16#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
17#include "webrtc/modules/video_coding/include/video_coding.h"
18#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
19#include "webrtc/video/stream_synchronization.h"
20#include "webrtc/voice_engine/include/voe_video_sync.h"
21
22namespace webrtc {
23
24int UpdateMeasurements(StreamSynchronization::Measurements* stream,
25                       const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) {
26  if (!receiver.Timestamp(&stream->latest_timestamp))
27    return -1;
28  if (!receiver.LastReceivedTimeMs(&stream->latest_receive_time_ms))
29    return -1;
30
31  uint32_t ntp_secs = 0;
32  uint32_t ntp_frac = 0;
33  uint32_t rtp_timestamp = 0;
34  if (0 != rtp_rtcp.RemoteNTP(&ntp_secs,
35                              &ntp_frac,
36                              NULL,
37                              NULL,
38                              &rtp_timestamp)) {
39    return -1;
40  }
41
42  bool new_rtcp_sr = false;
43  if (!UpdateRtcpList(
44      ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
45    return -1;
46  }
47
48  return 0;
49}
50
51ViESyncModule::ViESyncModule(VideoCodingModule* vcm)
52    : data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
53      vcm_(vcm),
54      video_receiver_(NULL),
55      video_rtp_rtcp_(NULL),
56      voe_channel_id_(-1),
57      voe_sync_interface_(NULL),
58      last_sync_time_(TickTime::Now()),
59      sync_() {
60}
61
62ViESyncModule::~ViESyncModule() {
63}
64
65int ViESyncModule::ConfigureSync(int voe_channel_id,
66                                 VoEVideoSync* voe_sync_interface,
67                                 RtpRtcp* video_rtcp_module,
68                                 RtpReceiver* video_receiver) {
69  CriticalSectionScoped cs(data_cs_.get());
70  // Prevent expensive no-ops.
71  if (voe_channel_id_ == voe_channel_id &&
72      voe_sync_interface_ == voe_sync_interface &&
73      video_receiver_ == video_receiver &&
74      video_rtp_rtcp_ == video_rtcp_module) {
75    return 0;
76  }
77  voe_channel_id_ = voe_channel_id;
78  voe_sync_interface_ = voe_sync_interface;
79  video_receiver_ = video_receiver;
80  video_rtp_rtcp_ = video_rtcp_module;
81  sync_.reset(
82      new StreamSynchronization(video_rtp_rtcp_->SSRC(), voe_channel_id));
83
84  if (!voe_sync_interface) {
85    voe_channel_id_ = -1;
86    if (voe_channel_id >= 0) {
87      // Trying to set a voice channel but no interface exist.
88      return -1;
89    }
90    return 0;
91  }
92  return 0;
93}
94
95int ViESyncModule::VoiceChannel() {
96  return voe_channel_id_;
97}
98
99int64_t ViESyncModule::TimeUntilNextProcess() {
100  const int64_t kSyncIntervalMs = 1000;
101  return kSyncIntervalMs - (TickTime::Now() - last_sync_time_).Milliseconds();
102}
103
104int32_t ViESyncModule::Process() {
105  CriticalSectionScoped cs(data_cs_.get());
106  last_sync_time_ = TickTime::Now();
107
108  const int current_video_delay_ms = vcm_->Delay();
109
110  if (voe_channel_id_ == -1) {
111    return 0;
112  }
113  assert(video_rtp_rtcp_ && voe_sync_interface_);
114  assert(sync_.get());
115
116  int audio_jitter_buffer_delay_ms = 0;
117  int playout_buffer_delay_ms = 0;
118  if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
119                                            &audio_jitter_buffer_delay_ms,
120                                            &playout_buffer_delay_ms) != 0) {
121    return 0;
122  }
123  const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
124      playout_buffer_delay_ms;
125
126  RtpRtcp* voice_rtp_rtcp = NULL;
127  RtpReceiver* voice_receiver = NULL;
128  if (0 != voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &voice_rtp_rtcp,
129                                           &voice_receiver)) {
130    return 0;
131  }
132  assert(voice_rtp_rtcp);
133  assert(voice_receiver);
134
135  if (UpdateMeasurements(&video_measurement_, *video_rtp_rtcp_,
136                         *video_receiver_) != 0) {
137    return 0;
138  }
139
140  if (UpdateMeasurements(&audio_measurement_, *voice_rtp_rtcp,
141                         *voice_receiver) != 0) {
142    return 0;
143  }
144
145  int relative_delay_ms;
146  // Calculate how much later or earlier the audio stream is compared to video.
147  if (!sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_,
148                                   &relative_delay_ms)) {
149    return 0;
150  }
151
152  TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay", current_video_delay_ms);
153  TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay", current_audio_delay_ms);
154  TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
155  int target_audio_delay_ms = 0;
156  int target_video_delay_ms = current_video_delay_ms;
157  // Calculate the necessary extra audio delay and desired total video
158  // delay to get the streams in sync.
159  if (!sync_->ComputeDelays(relative_delay_ms,
160                            current_audio_delay_ms,
161                            &target_audio_delay_ms,
162                            &target_video_delay_ms)) {
163    return 0;
164  }
165
166  if (voe_sync_interface_->SetMinimumPlayoutDelay(
167      voe_channel_id_, target_audio_delay_ms) == -1) {
168    LOG(LS_ERROR) << "Error setting voice delay.";
169  }
170  vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
171  return 0;
172}
173
174}  // namespace webrtc
175