zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Remove UDP transport API from ViE

Browse Source 
Review URL: https://webrtc-codereview.appspot.com/1232004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3754 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org
2013-04-02 20:37:14 +00:00
parent 7b859cc1e9
commit 82dcc9ff11
12 changed files with 67 additions and 1405 deletions
Download Patch File Download Diff File Expand all files Collapse all files

93
webrtc/video_engine/include/vie_network.h
View File

@ -70,34 +70,6 @@ class WEBRTC_DLLEXPORT ViENetwork {
virtual void SetNetworkTransmissionState(const int video_channel,
const bool is_transmitting) = 0;
// Specifies the ports to receive RTP packets on. It is also possible to set
// port for RTCP and local IP address.
virtual int SetLocalReceiver(const int video_channel,
const unsigned short rtp_port,
const unsigned short rtcp_port = 0,
const char* ip_address = NULL) = 0;
// Gets the local receiver ports and address for a specified channel.
virtual int GetLocalReceiver(const int video_channel,
unsigned short& rtp_port,
unsigned short& rtcp_port, char* ip_address) = 0;
// Specifies the destination port and IP address for a specified channel.
virtual int SetSendDestination(const int video_channel,
const char* ip_address,
const unsigned short rtp_port,
const unsigned short rtcp_port = 0,
const unsigned short source_rtp_port = 0,
const unsigned short source_rtcp_port = 0) = 0;
// Get the destination port and address for a specified channel.
virtual int GetSendDestination(const int video_channel,
char* ip_address,
unsigned short& rtp_port,
unsigned short& rtcp_port,
unsigned short& source_rtp_port,
unsigned short& source_rtcp_port) = 0;
// This function registers a user implementation of Transport to use for
// sending RTP and RTCP packets on this channel.
virtual int RegisterSendTransport(const int video_channel,
@ -119,63 +91,6 @@ class WEBRTC_DLLEXPORT ViENetwork {
const void* data,
const int length) = 0;
// Gets the source ports and IP address of the incoming stream for a
// specified channel.
virtual int GetSourceInfo(const int video_channel,
unsigned short& rtp_port,
unsigned short& rtcp_port,
char* ip_address,
unsigned int ip_address_length) = 0;
// Gets the local IP address, in string format.
virtual int GetLocalIP(char ip_address[64], bool ipv6 = false) = 0;
// Enables IPv6, instead of IPv4, for a specified channel.
virtual int EnableIPv6(int video_channel) = 0;
// The function returns true if IPv6 is enabled, false otherwise.
virtual bool IsIPv6Enabled(int video_channel) = 0;
// Enables a port and IP address filtering for incoming packets on a
// specific channel.
virtual int SetSourceFilter(const int video_channel,
const unsigned short rtp_port,
const unsigned short rtcp_port = 0,
const char* ip_address = NULL) = 0;
// Gets current port and IP address filter for a specified channel.
virtual int GetSourceFilter(const int video_channel,
unsigned short& rtp_port,
unsigned short& rtcp_port,
char* ip_address) = 0;
// This function sets the six‐bit Differentiated Services Code Point (DSCP)
// in the IP header of the outgoing stream for a specific channel.
// Windows and Linux only.
virtual int SetSendToS(const int video_channel,
const int DSCP,
const bool use_set_sockOpt = false) = 0;
// Retrieves the six‐bit Differentiated Services Code Point (DSCP) in the IP
// header of the outgoing stream for a specific channel.
virtual int GetSendToS(const int video_channel,
int& DSCP,
bool& use_set_sockOpt) = 0;
// This function sets the Generic Quality of Service (GQoS) service level.
// The Windows operating system then maps to a Differentiated Services Code
// Point (DSCP) and to an 802.1p setting. Windows only.
virtual int SetSendGQoS(const int video_channel, const bool enable,
const int service_type,
const int overrideDSCP = 0) = 0;
// This function retrieves the currently set GQoS service level for a
// specific channel.
virtual int GetSendGQoS(const int video_channel,
bool& enabled,
int& service_type,
int& overrideDSCP) = 0;
// This function sets the Maximum Transition Unit (MTU) for a channel. The
// RTP packet will be packetized based on this MTU to optimize performance
// over the network.
@ -202,14 +117,6 @@ class WEBRTC_DLLEXPORT ViENetwork {
const bool enable,
const unsigned int sample_time_seconds = KDefaultSampleTimeSeconds) = 0;
// This function handles sending a raw UDP data packet over an existing RTP
// or RTCP socket.
virtual int SendUDPPacket(const int video_channel,
const void* data,
const unsigned int length,
int& transmitted_bytes,
bool use_rtcp_socket = false) = 0;
protected:
ViENetwork() {}
virtual ~ViENetwork() {}