Index: webrtc/pc/channel.cc |
diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc |
index f5428a43effac7a93fce554e11946c220946337c..01e453397b63321ea38a43d62558f6bbb6c30521 100644 |
--- a/webrtc/pc/channel.cc |
+++ b/webrtc/pc/channel.cc |
@@ -785,7 +785,11 @@ bool BaseChannel::WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet) { |
return true; |
} |
// Check whether we handle this payload. |
- return bundle_filter_.DemuxPacket(packet->data(), packet->size()); |
+ return rtp_transport_.HandlesPacket(packet->data(), packet->size()); |
+} |
+ |
+bool BaseChannel::HandlesPayloadType(int packet_type) { |
+ return rtp_transport_.HandlesPayloadType(packet_type); |
} |
void BaseChannel::HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet, |
@@ -1448,6 +1452,10 @@ void BaseChannel::OnMessage(rtc::Message *pmsg) { |
} |
} |
+void BaseChannel::AddHandledPayloadType(int payload_type) { |
+ rtp_transport_.AddHandledPayloadType(payload_type); |
+} |
+ |
void BaseChannel::FlushRtcpMessages_n() { |
// Flush all remaining RTCP messages. This should only be called in |
// destructor. |
@@ -1766,7 +1774,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content, |
return false; |
} |
for (const AudioCodec& codec : audio->codecs()) { |
- bundle_filter()->AddPayloadType(codec.id); |
+ AddHandledPayloadType(codec.id); |
} |
last_recv_params_ = recv_params; |
@@ -2039,7 +2047,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content, |
return false; |
} |
for (const VideoCodec& codec : video->codecs()) { |
- bundle_filter()->AddPayloadType(codec.id); |
+ AddHandledPayloadType(codec.id); |
} |
last_recv_params_ = recv_params; |
@@ -2234,7 +2242,7 @@ bool RtpDataChannel::SetLocalContent_w(const MediaContentDescription* content, |
return false; |
} |
for (const DataCodec& codec : data->codecs()) { |
- bundle_filter()->AddPayloadType(codec.id); |
+ AddHandledPayloadType(codec.id); |
} |
last_recv_params_ = recv_params; |