| Index: webrtc/pc/channel.h
|
| diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h
|
| index ae3a29a2ffb1de42f48340f57a264cb3292e0906..11d21212f8ee4ecb2d847431cea3433d5e32c638 100644
|
| --- a/webrtc/pc/channel.h
|
| +++ b/webrtc/pc/channel.h
|
| @@ -271,7 +271,7 @@ class BaseChannel
|
| const rtc::PacketTime& packet_time);
|
| // TODO(zstein): packet can be const once the RtpTransport handles protection.
|
| virtual void OnPacketReceived(bool rtcp,
|
| - rtc::CopyOnWriteBuffer& packet,
|
| + rtc::CopyOnWriteBuffer* packet,
|
| const rtc::PacketTime& packet_time);
|
| void ProcessPacket(bool rtcp,
|
| const rtc::CopyOnWriteBuffer& packet,
|
| @@ -505,7 +505,7 @@ class VoiceChannel : public BaseChannel {
|
| private:
|
| // overrides from BaseChannel
|
| void OnPacketReceived(bool rtcp,
|
| - rtc::CopyOnWriteBuffer& packet,
|
| + rtc::CopyOnWriteBuffer* packet,
|
| const rtc::PacketTime& packet_time) override;
|
| void UpdateMediaSendRecvState_w() override;
|
| const ContentInfo* GetFirstContent(const SessionDescription* sdesc) override;
|
|
|