| Index: webrtc/media/base/mediachannel.h
|
| diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
|
| index d7aefe54ec4f238b0bb006a83968c69cdc0ab10b..6a8c0b296917cfee531fca4b3c9ffd0bb93b0636 100644
|
| --- a/webrtc/media/base/mediachannel.h
|
| +++ b/webrtc/media/base/mediachannel.h
|
| @@ -16,7 +16,7 @@
|
| #include <vector>
|
|
|
| #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"
|
| @@ -344,9 +344,9 @@ 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;
|
| @@ -368,10 +368,10 @@ 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;
|
| @@ -396,11 +396,13 @@ 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);
|
| }
|
|
|
| @@ -429,7 +431,7 @@ 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_);
|
| @@ -1096,7 +1098,7 @@ 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&,
|
|
|