Stop using VoEVideoSync in Call/VideoReceiveStream.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
@@ skipping 37 matching lines @@
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/send_delay_stats.h"
 #include "webrtc/video/stats_counter.h"
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/video/video_send_stream.h"
 #include "webrtc/video/vie_remb.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
 
 namespace webrtc {
 
 const int Call::Config::kDefaultStartBitrateBps = 300000;
 
 namespace internal {
 
 class Call : public webrtc::Call,
              public PacketReceiver,
              public RecoveredPacketReceiver,
@@ skipping 70 matching lines @@
  private:
   DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
                              size_t length);
   DeliveryStatus DeliverRtp(MediaType media_type,
                             const uint8_t* packet,
                             size_t length,
                             const PacketTime& packet_time);
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
 
-  VoiceEngine* voice_engine() {
-    internal::AudioState* audio_state =
-        static_cast<internal::AudioState*>(config_.audio_state.get());
-    if (audio_state)
-      return audio_state->voice_engine();
-    else
-      return nullptr;
-  }
-
   rtc::Optional<RtpPacketReceived> ParseRtpPacket(const uint8_t* packet,
                                                   size_t length,
                                                   const PacketTime& packet_time)
       SHARED_LOCKS_REQUIRED(receive_crit_);
 
   void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
   void UpdateReceiveHistograms();
   void UpdateHistograms();
   void UpdateAggregateNetworkState();
 
@@ skipping 473 matching lines @@
 
   bool protected_by_flexfec = false;
   {
     ReadLockScoped read_lock(*receive_crit_);
     protected_by_flexfec =
         flexfec_receive_ssrcs_media_.find(configuration.rtp.remote_ssrc) !=
         flexfec_receive_ssrcs_media_.end();
   }
   VideoReceiveStream* receive_stream = new VideoReceiveStream(
       num_cpu_cores_, protected_by_flexfec, congestion_controller_.get(),
-      &packet_router_, std::move(configuration), voice_engine(),
-      module_process_thread_.get(), call_stats_.get(), &remb_);
+      &packet_router_, std::move(configuration), module_process_thread_.get(),
+      call_stats_.get(), &remb_);
 
   const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
                video_receive_ssrcs_.end());
     video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
     if (config.rtp.rtx_ssrc)
       video_receive_ssrcs_[config.rtp.rtx_ssrc] = receive_stream;
     video_receive_streams_.insert(receive_stream);
@@ skipping 349 matching lines @@
                                      uint32_t max_padding_bitrate_bps) {
   congestion_controller_->SetAllocatedSendBitrateLimits(
       min_send_bitrate_bps, max_padding_bitrate_bps);
   rtc::CritScope lock(&bitrate_crit_);
   min_allocated_send_bitrate_bps_ = min_send_bitrate_bps;
   configured_max_padding_bitrate_bps_ = max_padding_bitrate_bps;
 }
 
 void Call::ConfigureSync(const std::string& sync_group) {
   // Set sync only if there was no previous one.
-  if (voice_engine() == nullptr || sync_group.empty())
+  if (sync_group.empty())
     return;
 
   AudioReceiveStream* sync_audio_stream = nullptr;
   // Find existing audio stream.
   const auto it = sync_stream_mapping_.find(sync_group);
   if (it != sync_stream_mapping_.end()) {
     sync_audio_stream = it->second;
   } else {
     // No configured audio stream, see if we can find one.
     for (const auto& kv : audio_receive_ssrcs_) {
@@ skipping 16 matching lines @@
       continue;
     ++num_synced_streams;
     if (num_synced_streams > 1) {
       // TODO(pbos): Support synchronizing more than one A/V pair.
       // https://code.google.com/p/webrtc/issues/detail?id=4762
       LOG(LS_WARNING) << "Attempting to sync more than one audio/video pair "
                          "within the same sync group. This is not supported in "
                          "the current implementation.";
     }
     // Only sync the first A/V pair within this sync group.
-    if (sync_audio_stream != nullptr && num_synced_streams == 1) {
-      video_stream->SetSyncChannel(voice_engine(),
-                                   sync_audio_stream->config().voe_channel_id);
+    if (num_synced_streams == 1) {
+      // sync_audio_stream may be null and that's ok.
+      video_stream->SetSync(sync_audio_stream);
     } else {
-      video_stream->SetSyncChannel(voice_engine(), -1);
+      video_stream->SetSync(nullptr);
     }
   }
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
                                                  const uint8_t* packet,
                                                  size_t length) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtcp");
   // TODO(pbos): Make sure it's a valid packet.
   //             Return DELIVERY_UNKNOWN_SSRC if it can be determined that
@@ skipping 135 matching lines @@
 
 void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet) {
   RTPHeader header;
   packet.GetHeader(&header);
   congestion_controller_->OnReceivedPacket(packet.arrival_time_ms(),
                                            packet.payload_size(), header);
 }
 
 }  // namespace internal
 }  // namespace webrtc