| Index: webrtc/media/base/mediachannel.h
|
| diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
|
| index ed565ee5b9df876080beb2e4afd5ff2f17246683..95355c9576a1afd117a4a5e4af43065b1a2cd654 100644
|
| --- a/webrtc/media/base/mediachannel.h
|
| +++ b/webrtc/media/base/mediachannel.h
|
| @@ -17,7 +17,7 @@
|
|
|
| #include "webrtc/api/rtpparameters.h"
|
| #include "webrtc/base/basictypes.h"
|
| -#include "webrtc/base/buffer.h"
|
| +#include "webrtc/base/copyonwritebuffer.h"
|
| #include "webrtc/base/dscp.h"
|
| #include "webrtc/base/logging.h"
|
| #include "webrtc/base/optional.h"
|
| @@ -356,9 +356,9 @@ class MediaChannel : public sigslot::has_slots<> {
|
| class NetworkInterface {
|
| public:
|
| enum SocketType { ST_RTP, ST_RTCP };
|
| - virtual bool SendPacket(rtc::Buffer* packet,
|
| + virtual bool SendPacket(rtc::CopyOnWriteBuffer* packet,
|
| const rtc::PacketOptions& options) = 0;
|
| - virtual bool SendRtcp(rtc::Buffer* packet,
|
| + virtual bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
|
| const rtc::PacketOptions& options) = 0;
|
| virtual int SetOption(SocketType type, rtc::Socket::Option opt,
|
| int option) = 0;
|
| @@ -380,10 +380,10 @@ class MediaChannel : public sigslot::has_slots<> {
|
| return rtc::DSCP_DEFAULT;
|
| }
|
| // Called when a RTP packet is received.
|
| - virtual void OnPacketReceived(rtc::Buffer* packet,
|
| + virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
|
| const rtc::PacketTime& packet_time) = 0;
|
| // Called when a RTCP packet is received.
|
| - virtual void OnRtcpReceived(rtc::Buffer* packet,
|
| + virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
|
| const rtc::PacketTime& packet_time) = 0;
|
| // Called when the socket's ability to send has changed.
|
| virtual void OnReadyToSend(bool ready) = 0;
|
| @@ -408,11 +408,13 @@ class MediaChannel : public sigslot::has_slots<> {
|
| }
|
|
|
| // Base method to send packet using NetworkInterface.
|
| - bool SendPacket(rtc::Buffer* packet, const rtc::PacketOptions& options) {
|
| + bool SendPacket(rtc::CopyOnWriteBuffer* packet,
|
| + const rtc::PacketOptions& options) {
|
| return DoSendPacket(packet, false, options);
|
| }
|
|
|
| - bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options) {
|
| + bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
|
| + const rtc::PacketOptions& options) {
|
| return DoSendPacket(packet, true, options);
|
| }
|
|
|
| @@ -441,7 +443,7 @@ class MediaChannel : public sigslot::has_slots<> {
|
| return ret;
|
| }
|
|
|
| - bool DoSendPacket(rtc::Buffer* packet,
|
| + bool DoSendPacket(rtc::CopyOnWriteBuffer* packet,
|
| bool rtcp,
|
| const rtc::PacketOptions& options) {
|
| rtc::CritScope cs(&network_interface_crit_);
|
| @@ -1104,7 +1106,7 @@ class DataMediaChannel : public MediaChannel {
|
|
|
| virtual bool SendData(
|
| const SendDataParams& params,
|
| - const rtc::Buffer& payload,
|
| + const rtc::CopyOnWriteBuffer& payload,
|
| SendDataResult* result = NULL) = 0;
|
| // Signals when data is received (params, data, len)
|
| sigslot::signal3<const ReceiveDataParams&,
|
|
|