Index: webrtc/pc/channel.cc |
diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc |
index 80fd75b5e8441d2a93aa895481195e4c5d1c8bc7..f6568c3c18f0335dc1080f762ac22fdefa3a97f1 100644 |
--- a/webrtc/pc/channel.cc |
+++ b/webrtc/pc/channel.cc |
@@ -748,7 +748,7 @@ bool BaseChannel::HandlesPayloadType(int packet_type) const { |
} |
void BaseChannel::OnPacketReceived(bool rtcp, |
- rtc::CopyOnWriteBuffer& packet, |
+ rtc::CopyOnWriteBuffer* packet, |
const rtc::PacketTime& packet_time) { |
if (!has_received_packet_ && !rtcp) { |
has_received_packet_ = true; |
@@ -758,8 +758,8 @@ void BaseChannel::OnPacketReceived(bool rtcp, |
// Unprotect the packet, if needed. |
if (srtp_filter_.IsActive()) { |
TRACE_EVENT0("webrtc", "SRTP Decode"); |
- char* data = packet.data<char>(); |
- int len = static_cast<int>(packet.size()); |
+ char* data = packet->data<char>(); |
+ int len = static_cast<int>(packet->size()); |
bool res; |
if (!rtcp) { |
res = srtp_filter_.UnprotectRtp(data, len, &len); |
@@ -784,7 +784,7 @@ void BaseChannel::OnPacketReceived(bool rtcp, |
} |
} |
- packet.SetSize(len); |
+ packet->SetSize(len); |
} else if (srtp_required_) { |
// Our session description indicates that SRTP is required, but we got a |
// packet before our SRTP filter is active. This means either that |
@@ -804,7 +804,7 @@ void BaseChannel::OnPacketReceived(bool rtcp, |
invoker_.AsyncInvoke<void>( |
RTC_FROM_HERE, worker_thread_, |
- Bind(&BaseChannel::ProcessPacket, this, rtcp, packet, packet_time)); |
+ Bind(&BaseChannel::ProcessPacket, this, rtcp, *packet, packet_time)); |
} |
void BaseChannel::ProcessPacket(bool rtcp, |
@@ -1678,7 +1678,7 @@ void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) { |
} |
void VoiceChannel::OnPacketReceived(bool rtcp, |
- rtc::CopyOnWriteBuffer& packet, |
+ rtc::CopyOnWriteBuffer* packet, |
const rtc::PacketTime& packet_time) { |
BaseChannel::OnPacketReceived(rtcp, packet, packet_time); |
// Set a flag when we've received an RTP packet. If we're waiting for early |