VoE changes to allow forwarding of packets from VoE to ViE BWE.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5757 4adac7df-926f-26a2-2b94-8c16560cd09d

1 files changed, 44 insertions, 6 deletions

diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index ea80570a..39effdf5 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -24,6 +24,7 @@
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/video_engine/include/vie_network.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
@@ -911,6 +912,8 @@ Channel::Channel(int32_t channelId,
     _countAliveDetections(0),
     _countDeadDetections(0),
     _outputSpeechType(AudioFrame::kNormalSpeech),
+    vie_network_(NULL),
+    video_channel_(-1),
     _average_jitter_buffer_delay_us(0),
     least_required_delay_ms_(0),
     _previousTimestamp(0),
@@ -1011,6 +1014,10 @@ Channel::~Channel()
     // End of modules shutdown
 
     // Delete other objects
+    if (vie_network_) {
+      vie_network_->Release();
+      vie_network_ = NULL;
+    }
     RtpDump::DestroyRtpDump(&_rtpDumpIn);
     RtpDump::DestroyRtpDump(&_rtpDumpOut);
     delete &_callbackCritSect;
@@ -2025,7 +2032,8 @@ Channel::DeRegisterExternalTransport()
     return 0;
 }
 
-int32_t Channel::ReceivedRTPPacket(const int8_t* data, int32_t length) {
+int32_t Channel::ReceivedRTPPacket(const int8_t* data, int32_t length,
+                                   const PacketTime& packet_time) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::ReceivedRTPPacket()");
 
@@ -2054,6 +2062,23 @@ int32_t Channel::ReceivedRTPPacket(const int8_t* data, int32_t length) {
   rtp_receive_statistics_->IncomingPacket(header, length,
       IsPacketRetransmitted(header, in_order));
   rtp_payload_registry_->SetIncomingPayloadType(header);
+
+  // Forward any packets to ViE bandwidth estimator, if enabled.
+  {
+    CriticalSectionScoped cs(&_callbackCritSect);
+    if (vie_network_) {
+      int64_t arrival_time_ms;
+      if (packet_time.timestamp != -1) {
+        arrival_time_ms = (packet_time.timestamp + 500) / 1000;
+      } else {
+        arrival_time_ms = TickTime::MillisecondTimestamp();
+      }
+      int payload_length = length - header.headerLength;
+      vie_network_->ReceivedBWEPacket(video_channel_, arrival_time_ms,
+                                      payload_length, header);
+    }
+  }
+
   return ReceivePacket(received_packet, length, header, in_order) ? 0 : -1;
 }
 
@@ -3511,11 +3536,9 @@ int Channel::SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id) {
 int Channel::SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id) {
   rtp_header_parser_->DeregisterRtpHeaderExtension(
       kRtpExtensionAbsoluteSendTime);
-  if (enable) {
-    if (!rtp_header_parser_->RegisterRtpHeaderExtension(
-        kRtpExtensionAbsoluteSendTime, id)) {
-      return -1;
-    }
+  if (enable && !rtp_header_parser_->RegisterRtpHeaderExtension(
+      kRtpExtensionAbsoluteSendTime, id)) {
+    return -1;
   }
   return 0;
 }
@@ -4105,6 +4128,21 @@ Channel::RTPDumpIsActive(RTPDirections direction)
     return rtpDumpPtr->IsActive();
 }
 
+void Channel::SetVideoEngineBWETarget(ViENetwork* vie_network,
+                                      int video_channel) {
+  CriticalSectionScoped cs(&_callbackCritSect);
+  if (vie_network_) {
+    vie_network_->Release();
+    vie_network_ = NULL;
+  }
+  video_channel_ = -1;
+
+  if (vie_network != NULL && video_channel != -1) {
+    vie_network_ = vie_network;
+    video_channel_ = video_channel;
+  }
+}
+
 uint32_t
 Channel::Demultiplex(const AudioFrame& audioFrame)
 {