Index: webrtc/pc/channel.cc |
diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc |
index f22bdc4ca41029155f2fe4fe511bb9c2765ee679..2f9447d3ce15a5ce907befd154fabd1e328a1534 100644 |
--- a/webrtc/pc/channel.cc |
+++ b/webrtc/pc/channel.cc |
@@ -24,6 +24,7 @@ |
#include "webrtc/base/trace_event.h" |
#include "webrtc/media/base/mediaconstants.h" |
#include "webrtc/media/base/rtputils.h" |
+#include "webrtc/p2p/base/packettransport.h" |
#include "webrtc/p2p/base/transportchannel.h" |
#include "webrtc/pc/channelmanager.h" |
@@ -527,17 +528,20 @@ bool BaseChannel::SetCryptoOptions(const rtc::CryptoOptions& crypto_options) { |
return true; |
} |
-void BaseChannel::OnWritableState(TransportChannel* channel) { |
+void BaseChannel::OnWritableState(rtc::PacketTransport* pt) { |
+ TransportChannel* channel = static_cast<TransportChannel*>(pt); |
RTC_DCHECK(channel == transport_channel_ || |
channel == rtcp_transport_channel_); |
RTC_DCHECK(network_thread_->IsCurrent()); |
UpdateWritableState_n(); |
} |
-void BaseChannel::OnChannelRead(TransportChannel* channel, |
- const char* data, size_t len, |
+void BaseChannel::OnChannelRead(rtc::PacketTransport* pt, |
+ const char* data, |
+ size_t len, |
const rtc::PacketTime& packet_time, |
int flags) { |
+ TransportChannel* channel = static_cast<TransportChannel*>(pt); |
TRACE_EVENT0("webrtc", "BaseChannel::OnChannelRead"); |
// OnChannelRead gets called from P2PSocket; now pass data to MediaEngine |
RTC_DCHECK(network_thread_->IsCurrent()); |
@@ -549,7 +553,8 @@ void BaseChannel::OnChannelRead(TransportChannel* channel, |
HandlePacket(rtcp, &packet, packet_time); |
} |
-void BaseChannel::OnReadyToSend(TransportChannel* channel) { |
+void BaseChannel::OnReadyToSend(rtc::PacketTransport* pt) { |
+ TransportChannel* channel = static_cast<TransportChannel*>(pt); |
RTC_DCHECK(channel == transport_channel_ || |
channel == rtcp_transport_channel_); |
SetTransportChannelReadyToSend(channel == rtcp_transport_channel_, true); |
@@ -1445,7 +1450,7 @@ void BaseChannel::FlushRtcpMessages_n() { |
} |
} |
-void BaseChannel::SignalSentPacket_n(TransportChannel* /* channel */, |
+void BaseChannel::SignalSentPacket_n(rtc::PacketTransport* /* pt */, |
const rtc::SentPacket& sent_packet) { |
RTC_DCHECK(network_thread_->IsCurrent()); |
invoker_.AsyncInvoke<void>( |
@@ -1641,12 +1646,13 @@ void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) { |
media_channel()->GetActiveStreams(actives); |
} |
-void VoiceChannel::OnChannelRead(TransportChannel* channel, |
- const char* data, size_t len, |
+void VoiceChannel::OnChannelRead(rtc::PacketTransport* pt, |
+ const char* data, |
+ size_t len, |
const rtc::PacketTime& packet_time, |
- int flags) { |
- BaseChannel::OnChannelRead(channel, data, len, packet_time, flags); |
- |
+ int flags) { |
+ BaseChannel::OnChannelRead(pt, data, len, packet_time, flags); |
+ TransportChannel* channel = static_cast<TransportChannel*>(pt); |
// Set a flag when we've received an RTP packet. If we're waiting for early |
// media, this will disable the timeout. |
if (!received_media_ && !PacketIsRtcp(channel, data, len)) { |