media/webrtc/trunk/src/video_engine/vie_sync_module.cc

   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 "video_engine/vie_sync_module.h"
  12
  13 #include "modules/rtp_rtcp/interface/rtp_rtcp.h"
  14 #include "modules/video_coding/main/interface/video_coding.h"
  15 #include "system_wrappers/interface/critical_section_wrapper.h"
  16 #include "system_wrappers/interface/trace.h"
  17 #include "video_engine/stream_synchronization.h"
  18 #include "voice_engine/include/voe_video_sync.h"
  19
  20 namespace webrtc {
  21
  22 enum { kSyncInterval = 1000};
  23
  24 ViESyncModule::ViESyncModule(const int32_t channel_id, VideoCodingModule* vcm)
  25     : data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
  26       channel_id_(channel_id),
  27       vcm_(vcm),
  28       video_rtcp_module_(NULL),
  29       voe_channel_id_(-1),
  30       voe_sync_interface_(NULL),
  31       last_sync_time_(TickTime::Now()),
  32       sync_() {
  33 }
  34
  35 ViESyncModule::~ViESyncModule() {
  36 }
  37
  38 int ViESyncModule::ConfigureSync(int voe_channel_id,
  39                                  VoEVideoSync* voe_sync_interface,
  40                                  RtpRtcp* video_rtcp_module) {
  41   CriticalSectionScoped cs(data_cs_.get());
  42   voe_channel_id_ = voe_channel_id;
  43   voe_sync_interface_ = voe_sync_interface;
  44   video_rtcp_module_ = video_rtcp_module;
  45   sync_.reset(new StreamSynchronization(voe_channel_id, channel_id_));
  46
  47   if (!voe_sync_interface) {
  48     voe_channel_id_ = -1;
  49     if (voe_channel_id >= 0) {
  50       // Trying to set a voice channel but no interface exist.
  51       return -1;
  52     }
  53     return 0;
  54   }
  55   return 0;
  56 }
  57
  58 int ViESyncModule::VoiceChannel() {
  59   return voe_channel_id_;
  60 }
  61
  62 WebRtc_Word32 ViESyncModule::TimeUntilNextProcess() {
  63   return static_cast<WebRtc_Word32>(kSyncInterval -
  64                          (TickTime::Now() - last_sync_time_).Milliseconds());
  65 }
  66
  67 WebRtc_Word32 ViESyncModule::Process() {
  68   CriticalSectionScoped cs(data_cs_.get());
  69   last_sync_time_ = TickTime::Now();
  70
  71   int total_video_delay_target_ms = vcm_->Delay();
  72   WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, channel_id_,
  73                "Video delay (JB + decoder) is %d ms",
  74                total_video_delay_target_ms);
  75
  76   if (voe_channel_id_ == -1) {
  77     return 0;
  78   }
  79   assert(video_rtcp_module_ && voe_sync_interface_);
  80   assert(sync_.get());
  81
  82   int current_audio_delay_ms = 0;
  83   if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
  84                                             current_audio_delay_ms) != 0) {
  85     // Could not get VoE delay value, probably not a valid channel Id.
  86     WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo, channel_id_,
  87                  "%s: VE_GetDelayEstimate error for voice_channel %d",
  88                  __FUNCTION__, total_video_delay_target_ms, voe_channel_id_);
  89     return 0;
  90   }
  91
  92   // VoiceEngine report delay estimates even when not started, ignore if the
  93   // reported value is lower than 40 ms.
  94   if (current_audio_delay_ms < 40) {
  95     WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, channel_id_,
  96                  "A/V Sync: Audio delay < 40, skipping.");
  97     return 0;
  98   }
  99
 100   RtpRtcp* voice_rtcp_module = NULL;
 101   if (0 != voe_sync_interface_->GetRtpRtcp(voe_channel_id_,
 102                                            voice_rtcp_module)) {
 103     return 0;
 104   }
 105   assert(voice_rtcp_module);
 106
 107   StreamSynchronization::Measurements video;
 108   if (0 != video_rtcp_module_->RemoteNTP(&video.received_ntp_secs,
 109                                          &video.received_ntp_frac,
 110                                          &video.rtcp_arrivaltime_secs,
 111                                          &video.rtcp_arrivaltime_frac)) {
 112     // Failed to get video NTP.
 113     return 0;
 114   }
 115
 116   StreamSynchronization::Measurements audio;
 117   if (0 != voice_rtcp_module->RemoteNTP(&audio.received_ntp_secs,
 118                                         &audio.received_ntp_frac,
 119                                         &audio.rtcp_arrivaltime_secs,
 120                                         &audio.rtcp_arrivaltime_frac)) {
 121     // Failed to get audio NTP.
 122     return 0;
 123   }
 124   int extra_audio_delay_ms = 0;
 125   if (sync_->ComputeDelays(audio, current_audio_delay_ms, &extra_audio_delay_ms,
 126                           video, &total_video_delay_target_ms) != 0) {
 127     return 0;
 128   }
 129   // Set the extra audio delay.synchronization
 130   if (voe_sync_interface_->SetMinimumPlayoutDelay(
 131       voe_channel_id_, extra_audio_delay_ms) == -1) {
 132     WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, channel_id_,
 133                  "Error setting voice delay");
 134   }
 135   vcm_->SetMinimumPlayoutDelay(total_video_delay_target_ms);
 136   WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, channel_id_,
 137                "New Video delay target is: %d", total_video_delay_target_ms);
 138   return 0;
 139 }
 140
 141 }  // namespace webrtc