Chromium Code Reviews| 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) { |