Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index e21b0762fe8429f21b87b38916c5c5b772ddb119..6a48d51c8d453df491a60c5805eb6431c9602993 100644 |
--- a/webrtc/call/call.cc |
+++ b/webrtc/call/call.cc |
@@ -1211,22 +1211,15 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, |
if (it != video_receive_ssrcs_.end()) { |
received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
received_video_bytes_per_second_counter_.Add(static_cast<int>(length)); |
- // TODO(brandtr): Notify the BWE of received media packets here. |
- auto status = it->second->DeliverRtp(packet, length, packet_time) |
- ? DELIVERY_OK |
- : DELIVERY_PACKET_ERROR; |
- // Deliver media packets to FlexFEC subsystem. RTP header extensions need |
- // not be parsed, as FlexFEC is oblivious to the semantic meaning of the |
- // packet contents beyond the 12 byte RTP base header. The BWE is fed |
- // information about these media packets from the regular media pipeline. |
+ it->second->OnRtpPacket(*parsed_packet); |
+ // Deliver media packets to FlexFEC subsystem. |
if (parsed_packet) { |
stefan-webrtc
2017/02/10 13:22:34
Isn't this guaranteed to be true given 1190?
nisse-webrtc
2017/02/10 13:27:40
Good catch. Deleted this test and the one a few li
|
auto it_bounds = flexfec_receive_ssrcs_media_.equal_range(ssrc); |
for (auto it = it_bounds.first; it != it_bounds.second; ++it) |
it->second->AddAndProcessReceivedPacket(*parsed_packet); |
} |
- if (status == DELIVERY_OK) |
brandtr
2017/02/10 12:50:55
Maybe check with terelius@ that this change is OK.
nisse-webrtc
2017/02/10 13:20:27
Discussed off-line.
|
- event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length); |
- return status; |
+ event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length); |
+ return DELIVERY_OK; |
} |
} |
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) { |