| Index: webrtc/call/call.cc
|
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
|
| index 6e620b6a6eda0889441f43811f32e6d37db7e12b..9be5d2ee70fc78d5373a6944cbf5b5fb6006964c 100644
|
| --- a/webrtc/call/call.cc
|
| +++ b/webrtc/call/call.cc
|
| @@ -1172,6 +1172,8 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
|
| const PacketTime& packet_time) {
|
| TRACE_EVENT0("webrtc", "Call::DeliverRtp");
|
|
|
| + RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO);
|
| +
|
| ReadLockScoped read_lock(*receive_crit_);
|
| // TODO(nisse): We should parse the RTP header only here, and pass
|
| // on parsed_packet to the receive streams.
|
| @@ -1185,7 +1187,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
|
|
|
| uint32_t ssrc = parsed_packet->Ssrc();
|
|
|
| - if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
|
| + if (media_type == MediaType::AUDIO) {
|
| auto it = audio_receive_ssrcs_.find(ssrc);
|
| if (it != audio_receive_ssrcs_.end()) {
|
| received_bytes_per_second_counter_.Add(static_cast<int>(length));
|
| @@ -1195,7 +1197,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
|
| return DELIVERY_OK;
|
| }
|
| }
|
| - if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
|
| + if (media_type == MediaType::VIDEO) {
|
| auto it = video_receive_ssrcs_.find(ssrc);
|
| if (it != video_receive_ssrcs_.end()) {
|
| received_bytes_per_second_counter_.Add(static_cast<int>(length));
|
| @@ -1211,7 +1213,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
|
| return DELIVERY_OK;
|
| }
|
| }
|
| - if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
|
| + if (media_type == MediaType::VIDEO) {
|
| received_bytes_per_second_counter_.Add(static_cast<int>(length));
|
| // TODO(brandtr): Update here when FlexFEC supports protecting audio.
|
| received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
|
|
|